diff options
Diffstat (limited to 'sound')
238 files changed, 13448 insertions, 5425 deletions
diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile index 31cbe68fd42f..c3ee77fc4b2d 100644 --- a/sound/aoa/codecs/Makefile +++ b/sound/aoa/codecs/Makefile @@ -1,3 +1,7 @@ +snd-aoa-codec-onyx-objs := onyx.o +snd-aoa-codec-tas-objs := tas.o +snd-aoa-codec-toonie-objs := toonie.o + obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.c b/sound/aoa/codecs/onyx.c index 6a3837d480e5..15500b9d2da0 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -37,7 +37,7 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); -#include "snd-aoa-codec-onyx.h" +#include "onyx.h" #include "../aoa.h" #include "../soundbus/soundbus.h" @@ -292,7 +292,7 @@ static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new capture_source_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* If we name this 'Input Source', it properly shows up in - * alsamixer as a selection, * but it's shown under the + * alsamixer as a selection, * but it's shown under the * 'Playback' category. * If I name it 'Capture Source', it shows up in strange * ways (two bools of which one can be selected at a @@ -477,7 +477,7 @@ static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol, ucontrol->value.iec958.status[3] = 0x3f; ucontrol->value.iec958.status[4] = 0x0f; - + return 0; } @@ -682,7 +682,7 @@ static int onyx_usable(struct codec_info_item *cii, onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); spdif_enabled = !!(v & ONYX_SPDIF_ENABLE); onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); - analog_enabled = + analog_enabled = (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT)) != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT); mutex_unlock(&onyx->mutex); @@ -882,7 +882,7 @@ static int onyx_init_codec(struct aoa_codec *codec) msleep(1); onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); msleep(1); - + if (onyx_register_init(onyx)) { printk(KERN_ERR PFX "failed to initialise onyx registers\n"); return -ENODEV; @@ -1069,7 +1069,7 @@ static int onyx_i2c_attach(struct i2c_adapter *adapter) /* if that didn't work, try desperate mode for older * machines that have stuff missing from the device tree */ - + if (!of_device_is_compatible(busnode, "k2-i2c")) return -ENODEV; diff --git a/sound/aoa/codecs/snd-aoa-codec-onyx.h b/sound/aoa/codecs/onyx.h index ffd20254ff76..ffd20254ff76 100644 --- a/sound/aoa/codecs/snd-aoa-codec-onyx.h +++ b/sound/aoa/codecs/onyx.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h b/sound/aoa/codecs/tas-basstreble.h index 69b61136fd54..69b61136fd54 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas-basstreble.h +++ b/sound/aoa/codecs/tas-basstreble.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h b/sound/aoa/codecs/tas-gain-table.h index 4cfa6757715e..4cfa6757715e 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h +++ b/sound/aoa/codecs/tas-gain-table.h diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/tas.c index 6c515b2b8bbd..008e0f85097d 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/tas.c @@ -71,9 +71,9 @@ MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("tas codec driver for snd-aoa"); -#include "snd-aoa-codec-tas.h" -#include "snd-aoa-codec-tas-gain-table.h" -#include "snd-aoa-codec-tas-basstreble.h" +#include "tas.h" +#include "tas-gain-table.h" +#include "tas-basstreble.h" #include "../aoa.h" #include "../soundbus/soundbus.h" @@ -880,7 +880,7 @@ static void tas_exit_codec(struct aoa_codec *codec) return; tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); } - + static struct i2c_driver tas_driver; diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.h b/sound/aoa/codecs/tas.h index ae177e3466e6..ae177e3466e6 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.h +++ b/sound/aoa/codecs/tas.h diff --git a/sound/aoa/codecs/snd-aoa-codec-toonie.c b/sound/aoa/codecs/toonie.c index 3c7d1d8a9a6f..f13827e17562 100644 --- a/sound/aoa/codecs/snd-aoa-codec-toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -131,7 +131,7 @@ static int __init toonie_init(void) toonie->codec.owner = THIS_MODULE; toonie->codec.init = toonie_init_codec; toonie->codec.exit = toonie_exit_codec; - + if (aoa_codec_register(&toonie->codec)) { kfree(toonie); return -EINVAL; diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile index 62dc7287f663..a1596e88c718 100644 --- a/sound/aoa/core/Makefile +++ b/sound/aoa/core/Makefile @@ -1,5 +1,5 @@ obj-$(CONFIG_SND_AOA) += snd-aoa.o -snd-aoa-objs := snd-aoa-core.o \ - snd-aoa-alsa.o \ - snd-aoa-gpio-pmf.o \ - snd-aoa-gpio-feature.o +snd-aoa-objs := core.o \ + alsa.o \ + gpio-pmf.o \ + gpio-feature.o diff --git a/sound/aoa/core/snd-aoa-alsa.c b/sound/aoa/core/alsa.c index 17fe689ed287..617850463582 100644 --- a/sound/aoa/core/snd-aoa-alsa.c +++ b/sound/aoa/core/alsa.c @@ -6,7 +6,7 @@ * GPL v2, can be found in COPYING. */ #include <linux/module.h> -#include "snd-aoa-alsa.h" +#include "alsa.h" static int index = -1; module_param(index, int, 0444); @@ -64,7 +64,7 @@ int aoa_snd_device_new(snd_device_type_t type, { struct snd_card *card = aoa_get_card(); int err; - + if (!card) return -ENOMEM; err = snd_device_new(card, type, device_data, ops); diff --git a/sound/aoa/core/snd-aoa-alsa.h b/sound/aoa/core/alsa.h index 9669e4489cab..9669e4489cab 100644 --- a/sound/aoa/core/snd-aoa-alsa.h +++ b/sound/aoa/core/alsa.h diff --git a/sound/aoa/core/snd-aoa-core.c b/sound/aoa/core/core.c index 19fdae400687..10bec6c61382 100644 --- a/sound/aoa/core/snd-aoa-core.c +++ b/sound/aoa/core/core.c @@ -10,7 +10,7 @@ #include <linux/module.h> #include <linux/list.h> #include "../aoa.h" -#include "snd-aoa-alsa.h" +#include "alsa.h" MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/gpio-feature.c index 805dcbff2257..c93ad5dec66b 100644 --- a/sound/aoa/core/snd-aoa-gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -5,7 +5,7 @@ * * GPL v2, can be found in COPYING. * - * This file contains the GPIO control routines for + * This file contains the GPIO control routines for * direct (through feature calls) access to the GPIO * registers. */ diff --git a/sound/aoa/core/snd-aoa-gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 5ca2220eac7d..5ca2220eac7d 100644 --- a/sound/aoa/core/snd-aoa-gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile index 55fc5e7e52cf..da37c10eca51 100644 --- a/sound/aoa/fabrics/Makefile +++ b/sound/aoa/fabrics/Makefile @@ -1 +1,3 @@ +snd-aoa-fabric-layout-objs += layout.o + obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/layout.c index dea7abb082cd..ad60f5d10e82 100644 --- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c +++ b/sound/aoa/fabrics/layout.c @@ -66,7 +66,7 @@ struct layout { unsigned int layout_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; - + /* if busname is not assigned, we use 'Master' below, * so that our layout table doesn't need to be filled * too much. diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index f84f3e505788..fa8ab2815a98 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -176,7 +176,7 @@ int soundbus_add_one(struct soundbus_dev *dev) return -EINVAL; } - snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount); + dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount); dev->ofdev.dev.bus = &soundbus_bus_type; return of_device_register(&dev->ofdev); } diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile index e57a5cf65655..1b949b2a4028 100644 --- a/sound/aoa/soundbus/i2sbus/Makefile +++ b/sound/aoa/soundbus/i2sbus/Makefile @@ -1,2 +1,2 @@ obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o -snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o +snd-aoa-i2sbus-objs := core.o pcm.o control.o diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-control.c b/sound/aoa/soundbus/i2sbus/control.c index 87beb4ad4d63..87beb4ad4d63 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-control.c +++ b/sound/aoa/soundbus/i2sbus/control.c diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/core.c index e6beb92c6933..be468edf3ecb 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -64,7 +64,7 @@ static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r) { if (!r->space) return; - + dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, r->size, r->space, r->bus_addr); } @@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_dev *dev; struct device_node *child = NULL, *sound = NULL; struct resource *r; - int i, layout = 0, rlen; + int i, layout = 0, rlen, ok = force; static const char *rnames[] = { "i2sbus: %s (control)", "i2sbus: %s (tx)", "i2sbus: %s (rx)" }; @@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, layout = *layout_id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); - force = 1; + ok = 1; } } /* for the time being, until we can handle non-layout-id @@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * When there are two i2s busses and only one has a layout-id, * then this depends on the order, but that isn't important * either as the second one in that case is just a modem. */ - if (!force) { + if (!ok) { kfree(dev); return -ENODEV; } @@ -247,7 +247,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * but request_resource doesn't know about parents and * contained resources... */ - dev->allocated_resource[i] = + dev->allocated_resource[i] = request_mem_region(dev->resources[i].start, dev->resources[i].end - dev->resources[i].start + 1, diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h index ff29654782c9..befefd99e271 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus.h +++ b/sound/aoa/soundbus/i2sbus/i2sbus.h @@ -18,7 +18,7 @@ #include <asm/pmac_feature.h> #include <asm/dbdma.h> -#include "i2sbus-interface.h" +#include "interface.h" #include "../soundbus.h" struct i2sbus_control { diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h b/sound/aoa/soundbus/i2sbus/interface.h index c6b5f5452d20..c6b5f5452d20 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-interface.h +++ b/sound/aoa/soundbus/i2sbus/interface.h diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 59bacd365733..59bacd365733 100644 --- a/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 622cd37a0118..a0f223c13f66 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -8,7 +8,7 @@ #ifndef __SOUNDBUS_H #define __SOUNDBUS_H -#include <asm/of_device.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <linux/list.h> diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 99026dfb81ea..34c1d94f921e 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -50,7 +50,7 @@ unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -90,7 +90,7 @@ void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec space */ - if ((cpu_is_pxa21x() || cpu_is_pxa25x()) && reg == AC97_GPIO_STATUS) + if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; else reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -200,7 +200,7 @@ static inline void pxa_ac97_cold_pxa3xx(void) bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_warm_pxa25x(); else #endif @@ -230,7 +230,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { #ifdef CONFIG_PXA25x - if (cpu_is_pxa21x() || cpu_is_pxa25x()) + if (cpu_is_pxa25x()) pxa_ac97_cold_pxa25x(); else #endif @@ -301,7 +301,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_suspend); int pxa2xx_ac97_hw_resume(void) { - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); @@ -325,7 +325,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret < 0) goto err; - if (cpu_is_pxa21x() || cpu_is_pxa25x() || cpu_is_pxa27x()) { + if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index cba71d867542..c2635beb4c88 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -44,7 +44,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { .name = "AC97 PCM out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -52,7 +52,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_in = { .name = "AC97 PCM in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb77cb99..75a0d746fb60 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 66348c92f88d..7bbdda041a99 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -95,6 +95,26 @@ config SND_SEQUENCER_OSS this will be compiled as a module. The module will be called snd-seq-oss. +config SND_HRTIMER + tristate "HR-timer backend support" + depends on HIGH_RES_TIMERS + select SND_TIMER + help + Say Y here to enable HR-timer backend for ALSA timer. ALSA uses + the hrtimer as a precise timing source. The ALSA sequencer code + also can use this timing source. + + To compile this driver as a module, choose M here: the module + will be called snd-hrtimer. + +config SND_SEQ_HRTIMER_DEFAULT + bool "Use HR-timer as default sequencer timer" + depends on SND_HRTIMER && SND_SEQUENCER + default y + help + Say Y here to use the HR-timer backend as the default sequencer + timer. + config SND_RTCTIMER tristate "RTC Timer support" depends on RTC @@ -114,6 +134,7 @@ config SND_RTCTIMER config SND_SEQ_RTCTIMER_DEFAULT bool "Use RTC as default sequencer timer" depends on SND_RTCTIMER && SND_SEQUENCER + depends on !SND_SEQ_HRTIMER_DEFAULT default y help Say Y here to use the RTC timer as the default sequencer diff --git a/sound/core/Makefile b/sound/core/Makefile index d57125a5687d..4229052e7b91 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -17,12 +17,14 @@ snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o +snd-hrtimer-objs := hrtimer.o snd-rtctimer-objs := rtctimer.o snd-hwdep-objs := hwdep.o obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o +obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/control.c b/sound/core/control.c index 6d71f9a7ccbb..636b3b52ef8b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -113,7 +113,6 @@ static int snd_ctl_release(struct inode *inode, struct file *file) unsigned int idx; ctl = file->private_data; - fasync_helper(-1, file, 0, &ctl->fasync); file->private_data = NULL; card = ctl->card; write_lock_irqsave(&card->ctl_files_rwlock, flags); @@ -225,8 +224,13 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, kctl.id.iface = ncontrol->iface; kctl.id.device = ncontrol->device; kctl.id.subdevice = ncontrol->subdevice; - if (ncontrol->name) + if (ncontrol->name) { strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); + if (strcmp(ncontrol->name, kctl.id.name) != 0) + snd_printk(KERN_WARNING + "Control name '%s' truncated to '%s'\n", + ncontrol->name, kctl.id.name); + } kctl.id.index = ncontrol->index; kctl.count = ncontrol->count ? ncontrol->count : 1; access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : diff --git a/sound/core/device.c b/sound/core/device.c index c58d8227254c..a67dfac08c03 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -98,7 +98,7 @@ int snd_device_free(struct snd_card *card, void *device_data) kfree(dev); return 0; } - snd_printd("device free %p (from %p), not found\n", device_data, + snd_printd("device free %p (from %pF), not found\n", device_data, __builtin_return_address(0)); return -ENXIO; } @@ -135,7 +135,7 @@ int snd_device_disconnect(struct snd_card *card, void *device_data) } return 0; } - snd_printd("device disconnect %p (from %p), not found\n", device_data, + snd_printd("device disconnect %p (from %pF), not found\n", device_data, __builtin_return_address(0)); return -ENXIO; } diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c new file mode 100644 index 000000000000..b712d7f211ff --- /dev/null +++ b/sound/core/hrtimer.c @@ -0,0 +1,140 @@ +/* + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/hrtimer.h> +#include <sound/core.h> +#include <sound/timer.h> + +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); +MODULE_DESCRIPTION("ALSA hrtimer backend"); +MODULE_LICENSE("GPL"); + +MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_HRTIMER)); + +#define NANO_SEC 1000000000UL /* 10^9 in sec */ +static unsigned int resolution; + +struct snd_hrtimer { + struct snd_timer *timer; + struct hrtimer hrt; +}; + +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) +{ + struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); + struct snd_timer *t = stime->timer; + hrtimer_forward_now(hrt, ktime_set(0, t->sticks * resolution)); + snd_timer_interrupt(stime->timer, t->sticks); + return HRTIMER_RESTART; +} + +static int snd_hrtimer_open(struct snd_timer *t) +{ + struct snd_hrtimer *stime; + + stime = kmalloc(sizeof(*stime), GFP_KERNEL); + if (!stime) + return -ENOMEM; + hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + stime->timer = t; + stime->hrt.cb_mode = HRTIMER_CB_SOFTIRQ; + stime->hrt.function = snd_hrtimer_callback; + t->private_data = stime; + return 0; +} + +static int snd_hrtimer_close(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + if (stime) { + hrtimer_cancel(&stime->hrt); + kfree(stime); + t->private_data = NULL; + } + return 0; +} + +static int snd_hrtimer_start(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + hrtimer_start(&stime->hrt, ktime_set(0, t->sticks * resolution), + HRTIMER_MODE_REL); + return 0; +} + +static int snd_hrtimer_stop(struct snd_timer *t) +{ + struct snd_hrtimer *stime = t->private_data; + + hrtimer_cancel(&stime->hrt); + return 0; +} + +static struct snd_timer_hardware hrtimer_hw = { + .flags = (SNDRV_TIMER_HW_AUTO | + /*SNDRV_TIMER_HW_FIRST |*/ + SNDRV_TIMER_HW_TASKLET), + .open = snd_hrtimer_open, + .close = snd_hrtimer_close, + .start = snd_hrtimer_start, + .stop = snd_hrtimer_stop, +}; + +/* + * entry functions + */ + +static struct snd_timer *mytimer; + +static int __init snd_hrtimer_init(void) +{ + struct snd_timer *timer; + struct timespec tp; + int err; + + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec > 0 || !tp.tv_nsec) { + snd_printk(KERN_ERR + "snd-hrtimer: Invalid resolution %u.%09u", + (unsigned)tp.tv_sec, (unsigned)tp.tv_nsec); + return -EINVAL; + } + resolution = tp.tv_nsec; + + /* Create a new timer and set up the fields */ + err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER, + &timer); + if (err < 0) + return err; + + timer->module = THIS_MODULE; + strcpy(timer->name, "HR timer"); + timer->hw = hrtimer_hw; + timer->hw.resolution = resolution; + timer->hw.ticks = NANO_SEC / resolution; + + err = snd_timer_global_register(timer); + if (err < 0) { + snd_timer_global_free(timer); + return err; + } + mytimer = timer; /* remember this */ + + return 0; +} + +static void __exit snd_hrtimer_exit(void) +{ + if (mytimer) { + snd_timer_global_free(mytimer); + mytimer = NULL; + } +} + +module_init(snd_hrtimer_init); +module_exit(snd_hrtimer_exit); diff --git a/sound/core/info.c b/sound/core/info.c index 527b207462b0..70fa87189f36 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -653,6 +653,23 @@ int snd_info_card_register(struct snd_card *card) } /* + * called on card->id change + */ +void snd_info_card_id_change(struct snd_card *card) +{ + mutex_lock(&info_mutex); + if (card->proc_root_link) { + snd_remove_proc_entry(snd_proc_root, card->proc_root_link); + card->proc_root_link = NULL; + } + if (strcmp(card->id, card->proc_root->name)) + card->proc_root_link = proc_symlink(card->id, + snd_proc_root, + card->proc_root->name); + mutex_unlock(&info_mutex); +} + +/* * de-register the card proc file * called from init.c */ diff --git a/sound/core/init.c b/sound/core/init.c index 8af467df9245..0d5520c415d3 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -264,8 +264,11 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) } spin_unlock(&shutdown_lock); - if (likely(df)) + if (likely(df)) { + if ((file->f_flags & FASYNC) && df->disconnected_f_op->fasync) + df->disconnected_f_op->fasync(-1, file, 0); return df->disconnected_f_op->release(inode, file); + } panic("%s(%p, %p) failed!", __func__, inode, file); } @@ -530,6 +533,65 @@ static void choose_default_id(struct snd_card *card) } } +#ifndef CONFIG_SYSFS_DEPRECATED +static ssize_t +card_id_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%s\n", card ? card->id : "(null)"); +} + +static ssize_t +card_id_store_attr(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_card *card = dev_get_drvdata(dev); + char buf1[sizeof(card->id)]; + size_t copy = count > sizeof(card->id) - 1 ? + sizeof(card->id) - 1 : count; + size_t idx; + int c; + + for (idx = 0; idx < copy; idx++) { + c = buf[idx]; + if (!isalnum(c) && c != '_' && c != '-') + return -EINVAL; + } + memcpy(buf1, buf, copy); + buf1[copy] = '\0'; + mutex_lock(&snd_card_mutex); + if (!snd_info_check_reserved_words(buf1)) { + __exist: + mutex_unlock(&snd_card_mutex); + return -EEXIST; + } + for (idx = 0; idx < snd_ecards_limit; idx++) { + if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) + goto __exist; + } + strcpy(card->id, buf1); + snd_info_card_id_change(card); + mutex_unlock(&snd_card_mutex); + + return count; +} + +static struct device_attribute card_id_attrs = + __ATTR(id, S_IRUGO | S_IWUSR, card_id_show_attr, card_id_store_attr); + +static ssize_t +card_number_show_attr(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_card *card = dev_get_drvdata(dev); + return snprintf(buf, PAGE_SIZE, "%i\n", card ? card->number : -1); +} + +static struct device_attribute card_number_attrs = + __ATTR(number, S_IRUGO, card_number_show_attr, NULL); +#endif /* CONFIG_SYSFS_DEPRECATED */ + /** * snd_card_register - register the soundcard * @card: soundcard structure @@ -549,9 +611,9 @@ int snd_card_register(struct snd_card *card) return -EINVAL; #ifndef CONFIG_SYSFS_DEPRECATED if (!card->card_dev) { - card->card_dev = device_create_drvdata(sound_class, card->dev, - MKDEV(0, 0), NULL, - "card%i", card->number); + card->card_dev = device_create(sound_class, card->dev, + MKDEV(0, 0), card, + "card%i", card->number); if (IS_ERR(card->card_dev)) card->card_dev = NULL; } @@ -573,6 +635,16 @@ int snd_card_register(struct snd_card *card) if (snd_mixer_oss_notify_callback) snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER); #endif +#ifndef CONFIG_SYSFS_DEPRECATED + if (card->card_dev) { + err = device_create_file(card->card_dev, &card_id_attrs); + if (err < 0) + return err; + err = device_create_file(card->card_dev, &card_number_attrs); + if (err < 0) + return err; + } +#endif return 0; } diff --git a/sound/core/jack.c b/sound/core/jack.c index 438445f77d6d..284432f427f4 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -151,6 +151,9 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + if (!jack) + return; + if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index a7b46ec72f32..1b3534d67686 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -33,9 +33,6 @@ #include <linux/moduleparam.h> #include <linux/mutex.h> #include <sound/memalloc.h> -#ifdef CONFIG_SBUS -#include <asm/sbus.h> -#endif MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>, Jaroslav Kysela <perex@perex.cz>"); @@ -162,39 +159,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, } #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SBUS - -static void *snd_malloc_sbus_pages(struct device *dev, size_t size, - dma_addr_t *dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - void *res; - - if (WARN_ON(!dma_addr)) - return NULL; - pg = get_order(size); - res = sbus_alloc_consistent(sdev, PAGE_SIZE * (1 << pg), dma_addr); - if (res != NULL) - inc_snd_pages(pg); - return res; -} - -static void snd_free_sbus_pages(struct device *dev, size_t size, - void *ptr, dma_addr_t dma_addr) -{ - struct sbus_dev *sdev = (struct sbus_dev *)dev; - int pg; - - if (ptr == NULL) - return; - pg = get_order(size); - dec_snd_pages(pg); - sbus_free_consistent(sdev, PAGE_SIZE * (1 << pg), ptr, dma_addr); -} - -#endif /* CONFIG_SBUS */ - /* * * ALSA generic memory management @@ -231,11 +195,6 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, dmab->area = snd_malloc_pages(size, (unsigned long)device); dmab->addr = 0; break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - dmab->area = snd_malloc_sbus_pages(device, size, &dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); @@ -306,11 +265,6 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_CONTINUOUS: snd_free_pages(dmab->area, dmab->bytes); break; -#ifdef CONFIG_SBUS - case SNDRV_DMA_TYPE_SBUS: - snd_free_sbus_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); - break; -#endif #ifdef CONFIG_HAS_DMA case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); @@ -419,7 +373,7 @@ static int snd_mem_proc_read(struct seq_file *seq, void *offset) long pages = snd_allocated_pages >> (PAGE_SHIFT-12); struct snd_mem_list *mem; int devno; - static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "SBUS" }; + static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG" }; mutex_lock(&list_mutex); seq_printf(seq, "pages : %li bytes (%li pages per %likB)\n", diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 1af62b8b86c6..e17836680f49 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2283,7 +2283,7 @@ static int snd_pcm_oss_open_file(struct file *file, int idx, err; struct snd_pcm_oss_file *pcm_oss_file; struct snd_pcm_substream *substream; - unsigned int f_mode = file->f_mode; + fmode_t f_mode = file->f_mode; if (rpcm_oss_file) *rpcm_oss_file = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6ea5cfb83998..921691080f35 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -908,12 +908,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, EXPORT_SYMBOL(snd_pcm_hw_rule_add); /** - * snd_pcm_hw_constraint_mask + * snd_pcm_hw_constraint_mask - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 32-bit mask parameter. */ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int32_t mask) @@ -928,12 +928,12 @@ int snd_pcm_hw_constraint_mask(struct snd_pcm_runtime *runtime, snd_pcm_hw_param } /** - * snd_pcm_hw_constraint_mask64 + * snd_pcm_hw_constraint_mask64 - apply the given bitmap mask constraint * @runtime: PCM runtime instance * @var: hw_params variable to apply the mask * @mask: the 64bit bitmap mask * - * Apply the constraint of the given bitmap mask to a mask parameter. + * Apply the constraint of the given bitmap mask to a 64-bit mask parameter. */ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, u_int64_t mask) @@ -949,7 +949,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par } /** - * snd_pcm_hw_constraint_integer + * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the integer constraint * @@ -964,7 +964,7 @@ int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_pa EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); /** - * snd_pcm_hw_constraint_minmax + * snd_pcm_hw_constraint_minmax - apply a min/max range constraint to an interval * @runtime: PCM runtime instance * @var: hw_params variable to apply the range * @min: the minimal value @@ -995,7 +995,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_constraint_list + * snd_pcm_hw_constraint_list - apply a list of constraints to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the list constraint @@ -1031,7 +1031,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratnums + * snd_pcm_hw_constraint_ratnums - apply ratnums constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratnums constraint @@ -1064,7 +1064,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_ratdens + * snd_pcm_hw_constraint_ratdens - apply ratdens constraint to a parameter * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the ratdens constraint @@ -1095,7 +1095,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_msbits + * snd_pcm_hw_constraint_msbits - add a hw constraint msbits rule * @runtime: PCM runtime instance * @cond: condition bits * @width: sample bits width @@ -1123,7 +1123,7 @@ static int snd_pcm_hw_rule_step(struct snd_pcm_hw_params *params, } /** - * snd_pcm_hw_constraint_step + * snd_pcm_hw_constraint_step - add a hw constraint step rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the step constraint @@ -1154,7 +1154,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm } /** - * snd_pcm_hw_constraint_pow2 + * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule * @runtime: PCM runtime instance * @cond: condition bits * @var: hw_params variable to apply the power-of-2 constraint @@ -1202,13 +1202,13 @@ void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params) EXPORT_SYMBOL(_snd_pcm_hw_params_any); /** - * snd_pcm_hw_param_value + * snd_pcm_hw_param_value - return @params field @var value * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Return the value for field PAR if it's fixed in configuration space - * defined by PARAMS. Return -EINVAL otherwise + * Return the value for field @var if it's fixed in configuration space + * defined by @params. Return -%EINVAL otherwise. */ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) @@ -1271,13 +1271,13 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_first + * snd_pcm_hw_param_first - refine config space and return minimum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values > minimum. Reduce configuration space accordingly. * Return the minimum. */ @@ -1317,13 +1317,13 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, /** - * snd_pcm_hw_param_last + * snd_pcm_hw_param_last - refine config space and return maximum value * @pcm: PCM instance * @params: the hw_params instance * @var: parameter to retrieve - * @dir: pointer to the direction (-1,0,1) or NULL + * @dir: pointer to the direction (-1,0,1) or %NULL * - * Inside configuration space defined by PARAMS remove from PAR all + * Inside configuration space defined by @params remove from @var all * values < maximum. Reduce configuration space accordingly. * Return the maximum. */ @@ -1345,11 +1345,11 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, EXPORT_SYMBOL(snd_pcm_hw_param_last); /** - * snd_pcm_hw_param_choose + * snd_pcm_hw_param_choose - choose a configuration defined by @params * @pcm: PCM instance * @params: the hw_params instance * - * Choose one configuration from configuration space defined by PARAMS + * Choose one configuration from configuration space defined by @params. * The configuration chosen is that obtained fixing in this order: * first access, first format, first subformat, min channels, * min rate, min period time, max buffer size, min tick time diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 89b7f549bebd..ea2bf82c9373 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -319,6 +319,7 @@ EXPORT_SYMBOL(snd_pcm_format_physical_width); /** * snd_pcm_format_size - return the byte size of samples on the given format * @format: the format to check + * @samples: sampling rate * * Returns the byte size of the given samples for the format, or a * negative error code if unknown format. diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index e61e12506ded..a789efc9df39 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -875,10 +875,8 @@ static struct action_ops snd_pcm_action_start = { }; /** - * snd_pcm_start + * snd_pcm_start - start all linked streams * @substream: the PCM substream instance - * - * Start all linked streams. */ int snd_pcm_start(struct snd_pcm_substream *substream) { @@ -926,12 +924,11 @@ static struct action_ops snd_pcm_action_stop = { }; /** - * snd_pcm_stop + * snd_pcm_stop - try to stop all running streams in the substream group * @substream: the PCM substream instance * @state: PCM state after stopping the stream * - * Try to stop all running streams in the substream group. - * The state of each stream is changed to the given value after that unconditionally. + * The state of each stream is then changed to the given state unconditionally. */ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) { @@ -941,11 +938,10 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, int state) EXPORT_SYMBOL(snd_pcm_stop); /** - * snd_pcm_drain_done + * snd_pcm_drain_done - stop the DMA only when the given stream is playback * @substream: the PCM substream * - * Stop the DMA only when the given stream is playback. - * The state is changed to SETUP. + * After stopping, the state is changed to SETUP. * Unlike snd_pcm_stop(), this affects only the given stream. */ int snd_pcm_drain_done(struct snd_pcm_substream *substream) @@ -1065,10 +1061,9 @@ static struct action_ops snd_pcm_action_suspend = { }; /** - * snd_pcm_suspend + * snd_pcm_suspend - trigger SUSPEND to all linked streams * @substream: the PCM substream * - * Trigger SUSPEND to all linked streams. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend(struct snd_pcm_substream *substream) @@ -1088,10 +1083,9 @@ int snd_pcm_suspend(struct snd_pcm_substream *substream) EXPORT_SYMBOL(snd_pcm_suspend); /** - * snd_pcm_suspend_all + * snd_pcm_suspend_all - trigger SUSPEND to all substreams in the given pcm * @pcm: the PCM instance * - * Trigger SUSPEND to all substreams in the given pcm. * After this call, all streams are changed to SUSPENDED state. */ int snd_pcm_suspend_all(struct snd_pcm *pcm) @@ -1313,11 +1307,9 @@ static struct action_ops snd_pcm_action_prepare = { }; /** - * snd_pcm_prepare + * snd_pcm_prepare - prepare the PCM substream to be triggerable * @substream: the PCM substream instance * @file: file to refer f_flags - * - * Prepare the PCM substream to be triggerable. */ static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) @@ -2177,7 +2169,6 @@ static int snd_pcm_release(struct inode *inode, struct file *file) if (snd_BUG_ON(!substream)) return -ENXIO; pcm = substream->pcm; - fasync_helper(-1, file, 0, &substream->runtime->fasync); mutex_lock(&pcm->open_mutex); snd_pcm_release_substream(substream); kfree(pcm_file); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index c4995c9f5730..39672f68ce5d 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -148,6 +148,8 @@ static int snd_rawmidi_runtime_free(struct snd_rawmidi_substream *substream) static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *substream,int up) { + if (!substream->opened) + return; if (up) { tasklet_hi_schedule(&substream->runtime->tasklet); } else { @@ -158,6 +160,8 @@ static inline void snd_rawmidi_output_trigger(struct snd_rawmidi_substream *subs static void snd_rawmidi_input_trigger(struct snd_rawmidi_substream *substream, int up) { + if (!substream->opened) + return; substream->ops->trigger(substream, up); if (!up && substream->runtime->event) tasklet_kill(&substream->runtime->tasklet); @@ -857,6 +861,8 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; + if (!substream->opened) + return -EBADFD; if (runtime->buffer == NULL) { snd_printd("snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; @@ -1126,6 +1132,8 @@ int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count) int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count) { + if (!substream->opened) + return -EBADFD; count = snd_rawmidi_transmit_peek(substream, buffer, count); if (count < 0) return count; diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index ee0f8405ab35..bf09a5ad1865 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -43,7 +43,9 @@ int seq_default_timer_class = SNDRV_TIMER_CLASS_GLOBAL; int seq_default_timer_sclass = SNDRV_TIMER_SCLASS_NONE; int seq_default_timer_card = -1; int seq_default_timer_device = -#ifdef CONFIG_SND_SEQ_RTCTIMER_DEFAULT +#ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT + SNDRV_TIMER_GLOBAL_HRTIMER +#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM diff --git a/sound/core/sound.c b/sound/core/sound.c index c0685e2f0afa..44a69bb8d4f0 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -274,9 +274,8 @@ int snd_register_device_for_dev(int type, struct snd_card *card, int dev, return minor; } snd_minors[minor] = preg; - preg->dev = device_create_drvdata(sound_class, device, - MKDEV(major, minor), - private_data, "%s", name); + preg->dev = device_create(sound_class, device, MKDEV(major, minor), + private_data, "%s", name); if (IS_ERR(preg->dev)) { snd_minors[minor] = NULL; mutex_unlock(&sound_mutex); diff --git a/sound/core/timer.c b/sound/core/timer.c index e582face89d2..c584408c9f17 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1263,7 +1263,6 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (file->private_data) { tu = file->private_data; file->private_data = NULL; - fasync_helper(-1, file, 0, &tu->fasync); if (tu->timeri) snd_timer_close(tu->timeri); kfree(tu->queue); diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index e5e749f3e0ef..73be7e14a603 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -51,7 +51,7 @@ static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) if (err < 0) return err; err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err) < 0) + if (err < 0) return err; return 0; } diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index ecdbeb6d3603..7783843ca9ae 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, /* get irq */ irq = platform_get_irq(pfdev, 0); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, - pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", irq); diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 1899cf0685bc..2a02f704f366 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -96,7 +96,7 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) return -EINVAL; hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); - pcsp_chip.timer.cb_mode = HRTIMER_CB_SOFTIRQ; + pcsp_chip.timer.cb_mode = HRTIMER_CB_IRQSAFE_UNLOCKED; pcsp_chip.timer.function = pcsp_do_timer; card = snd_card_new(index, id, THIS_MODULE, 0); @@ -188,10 +188,8 @@ static int __devexit pcsp_remove(struct platform_device *dev) static void pcsp_stop_beep(struct snd_pcsp *chip) { - spin_lock_irq(&chip->substream_lock); - if (!chip->playback_substream) - pcspkr_stop_sound(); - spin_unlock_irq(&chip->substream_lock); + pcsp_sync_stop(chip); + pcspkr_stop_sound(); } #ifdef CONFIG_PM diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index 1d661f795e8c..cdef2664218f 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -62,6 +62,8 @@ struct snd_pcsp { unsigned short port, irq, dma; spinlock_t substream_lock; struct snd_pcm_substream *playback_substream; + unsigned int fmt_size; + unsigned int is_signed; size_t playback_ptr; size_t period_ptr; atomic_t timer_active; @@ -77,6 +79,7 @@ struct snd_pcsp { extern struct snd_pcsp pcsp_chip; extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); +extern void pcsp_sync_stop(struct snd_pcsp *chip); extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c index cd9b83e7f7d1..0444cdeb4bec 100644 --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -24,13 +24,13 @@ static void pcspkr_do_sound(unsigned int count) spin_lock_irqsave(&i8253_lock, flags); if (count) { - /* enable counter 2 */ - outb_p(inb_p(0x61) | 3, 0x61); /* set command for counter 2, 2 byte write */ outb_p(0xB6, 0x43); /* select desired HZ */ outb_p(count & 0xff, 0x42); outb((count >> 8) & 0xff, 0x42); + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); } else { /* disable counter 2 */ outb(inb_p(0x61) & 0xFC, 0x61); diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e341f3f83b6a..84cc2658c05b 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/interrupt.h> #include <sound/pcm.h> #include <asm/io.h> #include "pcsp.h" @@ -19,61 +20,57 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround " #define DMIX_WANTS_S16 1 -enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +/* + * Call snd_pcm_period_elapsed in a tasklet + * This avoids spinlock messes and long-running irq contexts + */ +static void pcsp_call_pcm_elapsed(unsigned long priv) +{ + if (atomic_read(&pcsp_chip.timer_active)) { + struct snd_pcm_substream *substream; + substream = pcsp_chip.playback_substream; + if (substream) + snd_pcm_period_elapsed(substream); + } +} + +static DECLARE_TASKLET(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed, 0); + +/* write the port and returns the next expire time in ns; + * called at the trigger-start and in hrtimer callback + */ +static unsigned long pcsp_timer_update(struct hrtimer *handle) { unsigned char timer_cnt, val; - int fmt_size, periods_elapsed; u64 ns; - size_t period_bytes, buffer_bytes; struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + unsigned long flags; if (chip->thalf) { outb(chip->val61, 0x61); chip->thalf = 0; if (!atomic_read(&chip->timer_active)) - return HRTIMER_NORESTART; - hrtimer_forward(&chip->timer, chip->timer.expires, - ktime_set(0, chip->ns_rem)); - return HRTIMER_RESTART; + return 0; + return chip->ns_rem; } - spin_lock_irq(&chip->substream_lock); - /* Takashi Iwai says regarding this extra lock: - - If the irq handler handles some data on the DMA buffer, it should - do snd_pcm_stream_lock(). - That protects basically against all races among PCM callbacks, yes. - However, there are two remaining issues: - 1. The substream pointer you try to lock isn't protected _before_ - this lock yet. - 2. snd_pcm_period_elapsed() itself acquires the lock. - The requirement of another lock is because of 1. When you get - chip->playback_substream, it's not protected. - Keeping this lock while snd_pcm_period_elapsed() assures the substream - is still protected (at least, not released). And the other status is - handled properly inside snd_pcm_stream_lock() in - snd_pcm_period_elapsed(). - - */ - if (!chip->playback_substream) - goto exit_nr_unlock1; - substream = chip->playback_substream; - snd_pcm_stream_lock(substream); if (!atomic_read(&chip->timer_active)) - goto exit_nr_unlock2; + return 0; + substream = chip->playback_substream; + if (!substream) + return 0; runtime = substream->runtime; - fmt_size = snd_pcm_format_physical_width(runtime->format) >> 3; /* assume it is mono! */ - val = runtime->dma_area[chip->playback_ptr + fmt_size - 1]; - if (snd_pcm_format_signed(runtime->format)) + val = runtime->dma_area[chip->playback_ptr + chip->fmt_size - 1]; + if (chip->is_signed) val ^= 0x80; timer_cnt = val * CUR_DIV() / 256; if (timer_cnt && chip->enable) { - spin_lock(&i8253_lock); + spin_lock_irqsave(&i8253_lock, flags); if (!nforce_wa) { outb_p(chip->val61, 0x61); outb_p(timer_cnt, 0x42); @@ -82,12 +79,39 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) outb(chip->val61 ^ 2, 0x61); chip->thalf = 1; } - spin_unlock(&i8253_lock); + spin_unlock_irqrestore(&i8253_lock, flags); } + chip->ns_rem = PCSP_PERIOD_NS(); + ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); + chip->ns_rem -= ns; + return ns; +} + +enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) +{ + struct snd_pcsp *chip = container_of(handle, struct snd_pcsp, timer); + struct snd_pcm_substream *substream; + int periods_elapsed, pointer_update; + size_t period_bytes, buffer_bytes; + unsigned long ns; + unsigned long flags; + + pointer_update = !chip->thalf; + ns = pcsp_timer_update(handle); + if (!ns) + return HRTIMER_NORESTART; + + /* update the playback position */ + substream = chip->playback_substream; + if (!substream) + return HRTIMER_NORESTART; + period_bytes = snd_pcm_lib_period_bytes(substream); buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - chip->playback_ptr += PCSP_INDEX_INC() * fmt_size; + + spin_lock_irqsave(&chip->substream_lock, flags); + chip->playback_ptr += PCSP_INDEX_INC() * chip->fmt_size; periods_elapsed = chip->playback_ptr - chip->period_ptr; if (periods_elapsed < 0) { #if PCSP_DEBUG @@ -102,40 +126,30 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) * or ALSA will BUG on us. */ chip->playback_ptr %= buffer_bytes; - snd_pcm_stream_unlock(substream); - if (periods_elapsed) { - snd_pcm_period_elapsed(substream); chip->period_ptr += periods_elapsed * period_bytes; chip->period_ptr %= buffer_bytes; } + spin_unlock_irqrestore(&chip->substream_lock, flags); - spin_unlock_irq(&chip->substream_lock); + if (periods_elapsed) + tasklet_schedule(&pcsp_pcm_tasklet); - if (!atomic_read(&chip->timer_active)) - return HRTIMER_NORESTART; + hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); - chip->ns_rem = PCSP_PERIOD_NS(); - ns = (chip->thalf ? PCSP_CALC_NS(timer_cnt) : chip->ns_rem); - chip->ns_rem -= ns; - hrtimer_forward(&chip->timer, chip->timer.expires, ktime_set(0, ns)); return HRTIMER_RESTART; - -exit_nr_unlock2: - snd_pcm_stream_unlock(substream); -exit_nr_unlock1: - spin_unlock_irq(&chip->substream_lock); - return HRTIMER_NORESTART; } -static void pcsp_start_playing(struct snd_pcsp *chip) +static int pcsp_start_playing(struct snd_pcsp *chip) { + unsigned long ns; + #if PCSP_DEBUG printk(KERN_INFO "PCSP: start_playing called\n"); #endif if (atomic_read(&chip->timer_active)) { printk(KERN_ERR "PCSP: Timer already active\n"); - return; + return -EIO; } spin_lock(&i8253_lock); @@ -145,7 +159,12 @@ static void pcsp_start_playing(struct snd_pcsp *chip) atomic_set(&chip->timer_active, 1); chip->thalf = 0; - hrtimer_start(&pcsp_chip.timer, ktime_set(0, 0), HRTIMER_MODE_REL); + ns = pcsp_timer_update(&pcsp_chip.timer); + if (!ns) + return -EIO; + + hrtimer_start(&pcsp_chip.timer, ktime_set(0, ns), HRTIMER_MODE_REL); + return 0; } static void pcsp_stop_playing(struct snd_pcsp *chip) @@ -164,26 +183,35 @@ static void pcsp_stop_playing(struct snd_pcsp *chip) spin_unlock(&i8253_lock); } +/* + * Force to stop and sync the stream + */ +void pcsp_sync_stop(struct snd_pcsp *chip) +{ + local_irq_disable(); + pcsp_stop_playing(chip); + local_irq_enable(); + hrtimer_cancel(&chip->timer); + tasklet_kill(&pcsp_pcm_tasklet); +} + static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG printk(KERN_INFO "PCSP: close called\n"); #endif - if (atomic_read(&chip->timer_active)) { - printk(KERN_ERR "PCSP: timer still active\n"); - pcsp_stop_playing(chip); - } - spin_lock_irq(&chip->substream_lock); + pcsp_sync_stop(chip); chip->playback_substream = NULL; - spin_unlock_irq(&chip->substream_lock); return 0; } static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); int err; + pcsp_sync_stop(chip); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) @@ -193,9 +221,11 @@ static int snd_pcsp_playback_hw_params(struct snd_pcm_substream *substream, static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) { + struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG printk(KERN_INFO "PCSP: hw_free called\n"); #endif + pcsp_sync_stop(chip); return snd_pcm_lib_free_pages(substream); } @@ -211,8 +241,12 @@ static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) snd_pcm_lib_period_bytes(substream), substream->runtime->periods); #endif + pcsp_sync_stop(chip); chip->playback_ptr = 0; chip->period_ptr = 0; + chip->fmt_size = + snd_pcm_format_physical_width(substream->runtime->format) >> 3; + chip->is_signed = snd_pcm_format_signed(substream->runtime->format); return 0; } @@ -225,8 +259,7 @@ static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: - pcsp_start_playing(chip); - break; + return pcsp_start_playing(chip); case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: pcsp_stop_playing(chip); @@ -241,7 +274,11 @@ static snd_pcm_uframes_t snd_pcsp_playback_pointer(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); - return bytes_to_frames(substream->runtime, chip->playback_ptr); + unsigned int pos; + spin_lock(&chip->substream_lock); + pos = chip->playback_ptr; + spin_unlock(&chip->substream_lock); + return bytes_to_frames(substream->runtime, pos); } static struct snd_pcm_hardware snd_pcsp_playback = { @@ -278,9 +315,7 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) return -EBUSY; } runtime->hw = snd_pcsp_playback; - spin_lock_irq(&chip->substream_lock); chip->playback_substream = substream; - spin_unlock_irq(&chip->substream_lock); return 0; } diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 83e90057270e..c13a178383ba 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -87,8 +87,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) static int snd_tea575x_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long data) { - struct video_device *dev = video_devdata(file); - struct snd_tea575x *tea = video_get_drvdata(dev); + struct snd_tea575x *tea = video_drvdata(file); void __user *arg = (void __user *)data; switch(cmd) { @@ -175,6 +174,21 @@ static void snd_tea575x_release(struct video_device *vfd) { } +static int snd_tea575x_exclusive_open(struct inode *inode, struct file *file) +{ + struct snd_tea575x *tea = video_drvdata(file); + + return test_and_set_bit(0, &tea->in_use) ? -EBUSY : 0; +} + +static int snd_tea575x_exclusive_release(struct inode *inode, struct file *file) +{ + struct snd_tea575x *tea = video_drvdata(file); + + clear_bit(0, &tea->in_use); + return 0; +} + /* * initialize all the tea575x chips */ @@ -193,9 +207,10 @@ void snd_tea575x_init(struct snd_tea575x *tea) tea->vd.release = snd_tea575x_release; video_set_drvdata(&tea->vd, tea); tea->vd.fops = &tea->fops; + tea->in_use = 0; tea->fops.owner = tea->card->module; - tea->fops.open = video_exclusive_open; - tea->fops.release = video_exclusive_release; + tea->fops.open = snd_tea575x_exclusive_open; + tea->fops.release = snd_tea575x_exclusive_release; tea->fops.ioctl = snd_tea575x_ioctl; if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) { snd_printk(KERN_ERR "unable to register tea575x tuner\n"); diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 660beb41f767..ce0aa044e274 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -211,7 +211,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" - select SND_HWDEP + select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM help diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index b68d20edc20f..223a6c038819 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c index efa8c80d05b6..374b7177e111 100644 --- a/sound/isa/adlib.c +++ b/sound/isa/adlib.c @@ -36,7 +36,7 @@ static int __devinit snd_adlib_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } return 1; @@ -55,13 +55,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) card = snd_card_new(index[n], id[n], THIS_MODULE, 0); if (!card) { - snd_printk(KERN_ERR "%s: could not create card\n", dev->bus_id); + dev_err(dev, "could not create card\n"); return -EINVAL; } card->private_data = request_region(port[n], 4, CRD_NAME); if (!card->private_data) { - snd_printk(KERN_ERR "%s: could not grab ports\n", dev->bus_id); + dev_err(dev, "could not grab ports\n"); error = -EBUSY; goto out; } @@ -73,13 +73,13 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_opl3_create(card, port[n], port[n] + 2, OPL3_HW_AUTO, 1, &opl3); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create OPL\n", dev->bus_id); + dev_err(dev, "could not create OPL\n"); goto out; } error = snd_opl3_hwdep_new(opl3, 0, 0, NULL); if (error < 0) { - snd_printk(KERN_ERR "%s: could not create FM\n", dev->bus_id); + dev_err(dev, "could not create FM\n"); goto out; } @@ -87,7 +87,7 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n) error = snd_card_register(card); if (error < 0) { - snd_printk(KERN_ERR "%s: could not register card\n", dev->bus_id); + dev_err(dev, "could not register card\n"); goto out; } diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ddd289120aa8..f019d449e2d6 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -74,15 +74,15 @@ static int __devinit snd_cs4231_match(struct device *dev, unsigned int n) return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + dev_err(dev, "please specify port\n"); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + dev_err(dev, "please specify irq\n"); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + dev_err(dev, "please specify dma1\n"); return 0; } return 1; @@ -133,7 +133,7 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) mpu_port[n], 0, mpu_irq[n], mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) - printk(KERN_WARNING "%s: MPU401 not detected\n", dev->bus_id); + dev_warn(dev, "MPU401 not detected\n"); } snd_card_set_dev(card, dev); diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 91f9c15d3e30..019c9401663e 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -488,19 +488,19 @@ static int __devinit snd_cs423x_isa_match(struct device *pdev, return 0; if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", pdev->bus_id); + dev_err(pdev, "please specify port\n"); return 0; } if (cport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify cport\n", pdev->bus_id); + dev_err(pdev, "please specify cport\n"); return 0; } if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma1[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", pdev->bus_id); + dev_err(pdev, "please specify dma1\n"); return 0; } return 1; diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index f88639ea64b2..b46377139cf8 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -88,16 +88,14 @@ static int __devinit snd_es1688_legacy_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA\n"); return -EBUSY; } } @@ -147,8 +145,7 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_WARNING "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, chip->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 8f914b37bf89..426532a4d730 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -90,24 +90,21 @@ static int __devinit snd_gusclassic_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ\n", - dev->bus_id); + dev_err(dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA1\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[n] == SNDRV_AUTO_DMA) { dma2[n] = snd_legacy_find_free_dma(possible_dmas); if (dma2[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA2\n", - dev->bus_id); + dev_err(dev, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -174,8 +171,8 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n) error = -ENODEV; if (gus->max_flag || gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Classic or ACE soundcard was " - "not detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Classic or ACE soundcard was " + "not detected at 0x%lx\n", gus->gf1.port); goto out; } diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index da13185eb0a0..7ad4c3b41a84 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -106,16 +106,14 @@ static int __devinit snd_gusextreme_es1688_create(struct snd_card *card, if (irq[n] == SNDRV_AUTO_IRQ) { irq[n] = snd_legacy_find_free_irq(possible_irqs); if (irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for ES1688\n"); return -EBUSY; } } if (dma8[n] == SNDRV_AUTO_DMA) { dma8[n] = snd_legacy_find_free_dma(possible_dmas); if (dma8[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for ES1688\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for ES1688\n"); return -EBUSY; } } @@ -143,16 +141,14 @@ static int __devinit snd_gusextreme_gus_card_create(struct snd_card *card, if (gf1_irq[n] == SNDRV_AUTO_IRQ) { gf1_irq[n] = snd_legacy_find_free_irq(possible_irqs); if (gf1_irq[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free IRQ " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free IRQ for GF1\n"); return -EBUSY; } } if (dma1[n] == SNDRV_AUTO_DMA) { dma1[n] = snd_legacy_find_free_dma(possible_dmas); if (dma1[n] < 0) { - snd_printk(KERN_ERR "%s: unable to find a free DMA " - "for GF1\n", dev->bus_id); + dev_err(dev, "unable to find a free DMA for GF1\n"); return -EBUSY; } } @@ -278,8 +274,8 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) error = -ENODEV; if (!gus->ess_flag) { - snd_printk(KERN_ERR "%s: GUS Extreme soundcard was not " - "detected at 0x%lx\n", dev->bus_id, gus->gf1.port); + dev_err(dev, "GUS Extreme soundcard was not " + "detected at 0x%lx\n", gus->gf1.port); goto out; } gus->codec_flag = 1; @@ -310,8 +306,7 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) if (snd_opl3_create(card, es1688->port, es1688->port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) - printk(KERN_ERR "%s: opl3 not detected at 0x%lx\n", - dev->bus_id, es1688->port); + dev_warn(dev, "opl3 not detected at 0x%lx\n", es1688->port); else { error = snd_opl3_hwdep_new(opl3, 0, 2, NULL); if (error < 0) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 336a34277907..667eccc676a4 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -85,11 +85,11 @@ static int __devinit snd_sb8_match(struct device *pdev, unsigned int dev) if (!enable[dev]) return 0; if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", pdev->bus_id); + dev_err(pdev, "please specify irq\n"); return 0; } if (dma8[dev] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma8\n", pdev->bus_id); + dev_err(pdev, "please specify dma8\n"); return 0; } return 1; diff --git a/sound/oss/ac97_codec.c b/sound/oss/ac97_codec.c index b63839e8f9bd..456a1b4d7832 100644 --- a/sound/oss/ac97_codec.c +++ b/sound/oss/ac97_codec.c @@ -30,7 +30,7 @@ ************************************************************************** * * History - * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * Removed non existant WM9700 * Added support for WM9705, WM9708, WM9709, WM9710, WM9711 * WM9712 and WM9717 diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 23018a7c063a..81e1f443d094 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -93,7 +93,7 @@ static struct au1550_state { spinlock_t lock; struct mutex open_mutex; struct mutex sem; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; struct dmabuf { diff --git a/sound/oss/dmasound/dmasound.h b/sound/oss/dmasound/dmasound.h index d978b0096564..1308d8d34186 100644 --- a/sound/oss/dmasound/dmasound.h +++ b/sound/oss/dmasound/dmasound.h @@ -129,7 +129,7 @@ typedef struct { int (*mixer_ioctl)(u_int, u_long); /* optional */ int (*write_sq_setup)(void); /* optional */ int (*read_sq_setup)(void); /* optional */ - int (*sq_open)(mode_t); /* optional */ + int (*sq_open)(fmode_t); /* optional */ int (*state_info)(char *, size_t); /* optional */ void (*abort_read)(void); /* optional */ int min_dsp_speed; @@ -235,7 +235,7 @@ struct sound_queue { */ int active; wait_queue_head_t action_queue, open_queue, sync_queue; - int open_mode; + int non_blocking; int busy, syncing, xruns, died; }; diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 285239d64b82..4d45bd63718b 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -143,7 +143,7 @@ static int AtaMixerIoctl(u_int cmd, u_long arg); static int TTMixerIoctl(u_int cmd, u_long arg); static int FalconMixerIoctl(u_int cmd, u_long arg); static int AtaWriteSqSetup(void); -static int AtaSqOpen(mode_t mode); +static int AtaSqOpen(fmode_t mode); static int TTStateInfo(char *buffer, size_t space); static int FalconStateInfo(char *buffer, size_t space); @@ -1461,7 +1461,7 @@ static int AtaWriteSqSetup(void) return 0 ; } -static int AtaSqOpen(mode_t mode) +static int AtaSqOpen(fmode_t mode) { write_sq_ignore_int = 1; return 0 ; diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 95fc5c681755..793b7f478433 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -212,7 +212,7 @@ static int irq_installed; #endif /* MODULE */ /* control over who can modify resources shared between play/record */ -static mode_t shared_resource_owner; +static fmode_t shared_resource_owner; static int shared_resources_initialised; /* @@ -603,7 +603,7 @@ static ssize_t sq_write(struct file *file, const char __user *src, size_t uLeft, while (uLeft) { while (write_sq.count >= write_sq.max_active) { sq_play(); - if (write_sq.open_mode & O_NONBLOCK) + if (write_sq.non_blocking) return uWritten > 0 ? uWritten : -EAGAIN; SLEEP(write_sq.action_queue); if (signal_pending(current)) @@ -668,7 +668,7 @@ static inline void sq_init_waitqueue(struct sound_queue *sq) #if 0 /* blocking open() */ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, - mode_t mode) + fmode_t mode) { if (file->f_mode & mode) { sq->busy = 0; /* CHECK: IS THIS OK??? */ @@ -677,7 +677,7 @@ static inline void sq_wake_up(struct sound_queue *sq, struct file *file, } #endif -static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, +static int sq_open2(struct sound_queue *sq, struct file *file, fmode_t mode, int numbufs, int bufsize) { int rc = 0; @@ -718,7 +718,7 @@ static int sq_open2(struct sound_queue *sq, struct file *file, mode_t mode, return rc; } - sq->open_mode = file->f_mode; + sq->non_blocking = file->f_flags & O_NONBLOCK; } return rc; } @@ -891,10 +891,10 @@ static int sq_release(struct inode *inode, struct file *file) is the owner - if we have problems. */ -static int shared_resources_are_mine(mode_t md) +static int shared_resources_are_mine(fmode_t md) { if (shared_resource_owner) - return (shared_resource_owner & md ) ; + return (shared_resource_owner & md) != 0; else { shared_resource_owner = md ; return 1 ; diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index eb9bc365530d..c180598f1710 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -1,7 +1,7 @@ /* * Initialisation code for Cyrix/NatSemi VSA1 softaudio * - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. * The older version (VSA1) provides fairly good soundblaster emulation diff --git a/sound/oss/msnd.h b/sound/oss/msnd.h index 61b3955481c5..c8be47ec2b7e 100644 --- a/sound/oss/msnd.h +++ b/sound/oss/msnd.h @@ -211,7 +211,7 @@ typedef struct multisound_dev { /* State variables */ enum { msndClassic, msndPinnacle } type; - mode_t mode; + fmode_t mode; unsigned long flags; #define F_RESETTING 0 #define F_HAVEDIGITAL 1 diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b493660deb36..e5d423994918 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -26,7 +26,7 @@ #include <asm/cpu/dac.h> #include <asm/cpu/timer.h> #include <asm/machvec.h> -#include <asm/hp6xx.h> +#include <mach/hp6xx.h> #include <asm/hd64461.h> #define MODNAME "sh_dac_audio" diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 1a00a3210616..55271fbe7f49 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -110,24 +110,16 @@ struct channel_info { #define OPEN_WRITE PCM_ENABLE_OUTPUT #define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) -#if OPEN_READ == FMODE_READ && OPEN_WRITE == FMODE_WRITE - -static inline int translate_mode(struct file *file) -{ - return file->f_mode; -} - -#else - static inline int translate_mode(struct file *file) { - return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | - ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); + if (OPEN_READ == (__force int)FMODE_READ && + OPEN_WRITE == (__force int)FMODE_WRITE) + return (__force int)(file->f_mode & (FMODE_READ | FMODE_WRITE)); + else + return ((file->f_mode & FMODE_READ) ? OPEN_READ : 0) | + ((file->f_mode & FMODE_WRITE) ? OPEN_WRITE : 0); } -#endif - - #include "sound_calls.h" #include "dev_table.h" diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 7d89c081a086..61aaedae6b7e 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -560,19 +560,18 @@ static int __init oss_init(void) sound_dmap_flag = (dmabuf > 0 ? 1 : 0); for (i = 0; i < ARRAY_SIZE(dev_list); i++) { - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, dev_list[i].minor), - NULL, "%s", dev_list[i].name); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, dev_list[i].minor), NULL, + "%s", dev_list[i].name); if (!dev_list[i].num) continue; for (j = 1; j < *dev_list[i].num; j++) - device_create_drvdata(sound_class, NULL, - MKDEV(SOUND_MAJOR, - dev_list[i].minor + (j*0x10)), - NULL, - "%s%d", dev_list[i].name, j); + device_create(sound_class, NULL, + MKDEV(SOUND_MAJOR, + dev_list[i].minor + (j*0x10)), + NULL, "%s%d", dev_list[i].name, j); } if (sound_nblocks >= 1024) diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 044453a4ee5b..41562ecde5bb 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -295,7 +295,7 @@ struct cs4297a_state { struct mutex open_mutex; struct mutex open_sem_adc; struct mutex open_sem_dac; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; wait_queue_head_t open_wait_adc; wait_queue_head_t open_wait_dac; diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index dcbb3f739e61..78b8acc7c3b9 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -1509,7 +1509,7 @@ typedef struct vwsnd_dev { struct mutex open_mutex; struct mutex io_mutex; struct mutex mix_mutex; - mode_t open_mode; + fmode_t open_mode; wait_queue_head_t open_wait; lithium_t lith; diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index caebf296b62b..6e3a1848447c 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -208,7 +208,8 @@ config SND_OXYGEN * AuzenTech X-Meridian * Bgears b-Enspirer * Club3D Theatron DTS - * HT-Omega Claro + * HT-Omega Claro (plus) + * HT-Omega Claro halo (XT) * Razer Barracuda AC-1 * Sondigo Inferno diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acbca8c0..bd510eceff1f 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1927,9 +1927,9 @@ static int snd_ac97_dev_register(struct snd_device *device) ac97->dev.bus = &ac97_bus_type; ac97->dev.parent = ac97->bus->card->dev; ac97->dev.release = ac97_device_release; - snprintf(ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - ac97->bus->card->number, ac97->num, - snd_ac97_get_short_name(ac97)); + dev_set_name(&ac97->dev, "%d-%d:%s", + ac97->bus->card->number, ac97->num, + snd_ac97_get_short_name(ac97)); if ((err = device_register(&ac97->dev)) < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); ac97->dev.bus = NULL; diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 6ce3cbe98a6a..2c7cd97d2234 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97) } /* - * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk> * removed broken wolfson00 patch. * added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717. */ @@ -2832,6 +2832,8 @@ static int patch_alc655(struct snd_ac97 * ac97) val &= ~(1 << 1); /* Pin 47 is EAPD (for internal speaker) */ else val |= (1 << 1); /* Pin 47 is spdif input pin */ + /* this seems missing on some hardwares */ + ac97->ext_id |= AC97_EI_SPDIF; } val &= ~(1 << 12); /* vref enable */ snd_ac97_write_cache(ac97, 0x7a, val); diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a976ef2e..a7f38e63303f 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b8c807..226fe8237d31 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f106306c7fe..0e6e5cc1c501 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e490074..a36d4d1fd419 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d973540a..1aa1c0402540 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index a7d89662acf6..88fbf285d2b7 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -759,7 +759,6 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | SPCS_GENERATIONSTATUS | 0x00001200 | 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); - } #endif return 0; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f7c45b..192e7842e181 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68e997e..6dea5b5cc774 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -2,7 +2,7 @@ * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio * * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net> - * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk> * * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did * mess with it a bit. The chip seems to have to have trouble with full duplex @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 2f283ea6ad9a..7958006a1d66 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -69,7 +69,7 @@ MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); * EMU10K1 init / done *************************************************************************/ -void snd_emu10k1_voice_init(struct snd_emu10k1 * emu, int ch) +void snd_emu10k1_voice_init(struct snd_emu10k1 *emu, int ch) { snd_emu10k1_ptr_write(emu, DCYSUSV, ch, 0); snd_emu10k1_ptr_write(emu, IP, ch, 0); @@ -151,9 +151,9 @@ static unsigned int i2c_adc_init[][2] = { { 0x12, 0x32 }, /* ALC Control 3 */ { 0x13, 0x00 }, /* Noise gate control */ { 0x14, 0xa6 }, /* Limiter control */ - { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for Audigy 2 ZS Notebook */ + { 0x15, ADC_MUX_2 }, /* ADC Mixer control. Mic for A2ZS Notebook */ }; - + static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) { unsigned int silent_page; @@ -161,8 +161,8 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) u32 tmp; /* disable audio and lock cache */ - outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | HCFG_MUTEBUTTONENABLE, - emu->port + HCFG); + outl(HCFG_LOCKSOUNDCACHE | HCFG_LOCKTANKCACHE_MASK | + HCFG_MUTEBUTTONENABLE, emu->port + HCFG); /* reset recording buffers */ snd_emu10k1_ptr_write(emu, MICBS, 0, ADCBS_BUFSIZE_NONE); @@ -179,7 +179,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) snd_emu10k1_ptr_write(emu, SOLEL, 0, 0); snd_emu10k1_ptr_write(emu, SOLEH, 0, 0); - if (emu->audigy){ + if (emu->audigy) { /* set SPDIF bypass mode */ snd_emu10k1_ptr_write(emu, SPBYPASS, 0, SPBYPASS_FORMAT); /* enable rear left + rear right AC97 slots */ @@ -197,12 +197,12 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Hacks for Alice3 to work independent of haP16V driver */ - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); snd_emu10k1_ptr_write(emu, A_SPDIF_SAMPLERATE, 0, tmp); - + /* Setup SRCSel (Enable Spdif,I2S SRCMulti) */ snd_emu10k1_ptr20_write(emu, SRCSel, 0, 0x14); /* Setup SRCMulti Input Audio Enable */ @@ -217,7 +217,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) if (emu->card_capabilities->ca0108_chip) { /* audigy2 Value */ /* Hacks for Alice3 to work independent of haP16V driver */ snd_printk(KERN_INFO "Audigy2 value: Special config.\n"); - //Setup SRCMulti_I2S SamplingRate + /* Setup SRCMulti_I2S SamplingRate */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, 0); tmp &= 0xfffff1ff; tmp |= (0x2<<9); @@ -270,13 +270,13 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) size = ARRAY_SIZE(i2c_adc_init); for (n = 0; n < size; n++) snd_emu10k1_i2c_write(emu, i2c_adc_init[n][0], i2c_adc_init[n][1]); - for (n=0; n < 4; n++) { - emu->i2c_capture_volume[n][0]= 0xcf; - emu->i2c_capture_volume[n][1]= 0xcf; + for (n = 0; n < 4; n++) { + emu->i2c_capture_volume[n][0] = 0xcf; + emu->i2c_capture_volume[n][1] = 0xcf; } } - + snd_emu10k1_ptr_write(emu, PTB, 0, emu->ptb_pages.addr); snd_emu10k1_ptr_write(emu, TCB, 0, 0); /* taken from original driver */ snd_emu10k1_ptr_write(emu, TCBS, 0, 4); /* taken from original driver */ @@ -313,7 +313,7 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) (emu->model == 0x21 && emu->revision < 6)) outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE, emu->port + HCFG); else - // With on-chip joystick + /* With on-chip joystick */ outl(HCFG_LOCKTANKCACHE_MASK | HCFG_AUTOMUTE | HCFG_JOYENABLE, emu->port + HCFG); if (enable_ir) { /* enable IR for SB Live */ @@ -335,9 +335,9 @@ static int snd_emu10k1_init(struct snd_emu10k1 *emu, int enable_ir, int resume) outl(reg | HCFG_GPOUT1 | HCFG_GPOUT2, emu->port + HCFG); udelay(100); outl(reg, emu->port + HCFG); - } + } } - + if (emu->card_capabilities->emu_model) { ; /* Disable all access to A_IOCFG for the emu1010 */ } else if (emu->card_capabilities->i2c_adc) { @@ -364,7 +364,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) ; /* Disable A_IOCFG for Audigy 2 ZS Notebook */ } else if (emu->audigy) { outl(inl(emu->port + A_IOCFG) & ~0x44, emu->port + A_IOCFG); - + if (emu->card_capabilities->ca0151_chip) { /* audigy2 */ /* Unmute Analog now. Set GPO6 to 1 for Apollo. * This has to be done after init ALice3 I2SOut beyond 48KHz. @@ -378,12 +378,12 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) outl(inl(emu->port + A_IOCFG) | 0x0080, emu->port + A_IOCFG); } } - + #if 0 { unsigned int tmp; /* FIXME: the following routine disables LiveDrive-II !! */ - // TOSLink detection + /* TOSLink detection */ emu->tos_link = 0; tmp = inl(emu->port + HCFG); if (tmp & (HCFG_GPINPUT0 | HCFG_GPINPUT1)) { @@ -400,7 +400,7 @@ static void snd_emu10k1_audio_enable(struct snd_emu10k1 *emu) snd_emu10k1_intr_enable(emu, INTE_PCIERRORENABLE); } -int snd_emu10k1_done(struct snd_emu10k1 * emu) +int snd_emu10k1_done(struct snd_emu10k1 *emu) { int ch; @@ -495,7 +495,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_LAST_PROMFILE_ADDR 0x2f -#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The +#define EC_SERIALNUM_ADDR 0x30 /* First word of serial number. The * can be up to 30 characters in length * and is stored as a NULL-terminated * ASCII string. Any unused bytes must be @@ -503,8 +503,8 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) #define EC_CHECKSUM_ADDR 0x3f /* Location at which checksum is stored */ -/* Most of this stuff is pretty self-evident. According to the hardware - * dudes, we need to leave the ADCCAL bit low in order to avoid a DC +/* Most of this stuff is pretty self-evident. According to the hardware + * dudes, we need to leave the ADCCAL bit low in order to avoid a DC * offset problem. Weird. */ #define EC_RAW_RUN_MODE (EC_DACMUTEN | EC_ADCRSTN | EC_TRIM_MUTEN | \ @@ -523,7 +523,7 @@ int snd_emu10k1_done(struct snd_emu10k1 * emu) * register. */ -static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value) +static void snd_emu10k1_ecard_write(struct snd_emu10k1 *emu, unsigned int value) { unsigned short count; unsigned int data; @@ -561,7 +561,7 @@ static void snd_emu10k1_ecard_write(struct snd_emu10k1 * emu, unsigned int value * channel. */ -static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, +static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 *emu, unsigned short gain) { unsigned int bit; @@ -574,7 +574,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, for (bit = (1 << 15); bit; bit >>= 1) { unsigned int value; - + value = emu->ecard_ctrl & ~(EC_TRIM_CSN | EC_TRIM_SDATA); if (gain & bit) @@ -589,7 +589,7 @@ static void snd_emu10k1_ecard_setadcgain(struct snd_emu10k1 * emu, snd_emu10k1_ecard_write(emu, emu->ecard_ctrl); } -static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_ecard_init(struct snd_emu10k1 *emu) { unsigned int hc_value; @@ -598,7 +598,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) EC_SPDIF0_SELECT(EC_DEFAULT_SPDIF0_SEL) | EC_SPDIF1_SELECT(EC_DEFAULT_SPDIF1_SEL); - /* Step 0: Set the codec type in the hardware control register + /* Step 0: Set the codec type in the hardware control register * and enable audio output */ hc_value = inl(emu->port + HCFG); outl(hc_value | HCFG_AUDIOENABLE | HCFG_CODECFORMAT_I2S, emu->port + HCFG); @@ -629,7 +629,7 @@ static int snd_emu10k1_ecard_init(struct snd_emu10k1 * emu) return 0; } -static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_cardbus_init(struct snd_emu10k1 *emu) { unsigned long special_port; unsigned int value; @@ -656,7 +656,7 @@ static int snd_emu10k1_cardbus_init(struct snd_emu10k1 * emu) return 0; } -static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * filename) +static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filename) { int err; int n, i; @@ -666,11 +666,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file unsigned long flags; const struct firmware *fw_entry; - if ((err = request_firmware(&fw_entry, filename, &emu->pci->dev)) != 0) { - snd_printk(KERN_ERR "firmware: %s not found. Err=%d\n",filename, err); + err = request_firmware(&fw_entry, filename, &emu->pci->dev); + if (err != 0) { + snd_printk(KERN_ERR "firmware: %s not found. Err = %d\n", filename, err); return err; } - snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); + snd_printk(KERN_INFO "firmware size = 0x%zx\n", fw_entry->size); /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -685,13 +686,13 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file outl(0x80, emu->port + A_IOCFG); /* Leave bit 7 set during netlist setup. */ write_post = inl(emu->port + A_IOCFG); udelay(100); /* Allow FPGA memory to clean */ - for(n = 0; n < fw_entry->size; n++) { - value=fw_entry->data[n]; - for(i = 0; i < 8; i++) { + for (n = 0; n < fw_entry->size; n++) { + value = fw_entry->data[n]; + for (i = 0; i < 8; i++) { reg = 0x80; if (value & 0x1) reg = reg | 0x20; - value = value >> 1; + value = value >> 1; outl(reg, emu->port + A_IOCFG); write_post = inl(emu->port + A_IOCFG); outl(reg | 0x40, emu->port + A_IOCFG); @@ -703,14 +704,14 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file write_post = inl(emu->port + A_IOCFG); spin_unlock_irqrestore(&emu->emu_lock, flags); - release_firmware(fw_entry); + release_firmware(fw_entry); return 0; } static int emu1010_firmware_thread(void *data) { - struct snd_emu10k1 * emu = data; - int tmp,tmp2; + struct snd_emu10k1 *emu = data; + int tmp, tmp2; int reg; int err; @@ -719,50 +720,50 @@ static int emu1010_firmware_thread(void *data) msleep_interruptible(1000); if (kthread_should_stop()) break; - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp ); /* IRQ Status */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); /* OPTIONS: Which cards are attached to the EMU */ + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, &tmp); /* IRQ Status */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); /* OPTIONS: Which cards are attached to the EMU */ if (reg & EMU_HANA_OPTION_DOCK_OFFLINE) { /* Audio Dock attached */ /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK); if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, DOCK_FILENAME); + if (err != 0) continue; - } } else if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010B) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME); + if (err != 0) continue; - } } else if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1616) { - if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME); + if (err != 0) continue; - } } - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); - snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0); + snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ®); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS = 0x%x\n", reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID = 0x%x\n", reg); if ((reg & 0x1f) != 0x15) { /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); + snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg = 0x%x\n", reg); continue; } snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2 ); - snd_printk("Audio Dock ver:%d.%d\n",tmp ,tmp2); + snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); + snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); /* Unmute all. Default is muted after a firmware load */ - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE ); + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } } snd_printk(KERN_INFO "emu1010: firmware thread stopping\n"); @@ -800,10 +801,10 @@ static int emu1010_firmware_thread(void *data) * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops */ -static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) +static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) { unsigned int i; - int tmp,tmp2; + int tmp, tmp2; int reg; int err; const char *filename = NULL; @@ -818,7 +819,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0005a004, emu->port + HCFG); + outl(0x0005a004, emu->port + HCFG); /* AC97 2.1, Any 16Meg of 4Gig address, Auto-Mute, EMU32 Slave, * Mute all codecs. */ @@ -829,25 +830,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) outl(0x0005a000, emu->port + HCFG); /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printdd("reg1=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printdd("reg1 = 0x%x\n", reg); if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ - snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02 ); + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0x02); } - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - snd_printdd("reg2=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); + snd_printdd("reg2 = 0x%x\n", reg); if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } - snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); + snd_printk(KERN_INFO "emu1010: EMU_HANA_ID = 0x%x\n", reg); switch (emu->card_capabilities->emu_model) { case EMU_MODEL_EMU1010: filename = HANA_FILENAME; @@ -876,25 +877,25 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ - snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); + snd_emu1010_fpga_read(emu, EMU_HANA_ID, ®); if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg = 0x%x\n", reg); return -ENODEV; } snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); - snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2 ); - snd_printk("Hana ver:%d.%d\n",tmp ,tmp2); + snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); + snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options=0x%x\n",reg); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options = 0x%x\n", reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTICAL_TYPE, &tmp); /* Optical -> ADAT I/O */ /* 0 : SPDIF * 1 : ADAT @@ -904,41 +905,42 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) tmp = 0; tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); - snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp ); - snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + snd_emu1010_fpga_read(emu, EMU_HANA_ADC_PADS, &tmp); /* Set no attenuation on Audio Dock pads. */ - snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_ADC_PADS, 0x00); emu->emu1010.adc_pads = 0x00; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); /* Unmute Audio dock DACs, Headphone source DAC-4. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); - snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); + snd_emu1010_fpga_read(emu, EMU_HANA_DAC_PADS, &tmp); /* DAC PADs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f ); + snd_emu1010_fpga_write(emu, EMU_HANA_DAC_PADS, 0x0f); emu->emu1010.dac_pads = 0x0f; - snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30 ); - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); + snd_emu1010_fpga_read(emu, EMU_HANA_DOCK_MISC, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_MISC, 0x30); + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); /* SPDIF Format. Set Consumer mode, 24bit, copy enable */ - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* MIDI routing */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* Unknown. */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); - /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); // IRQ Enable: All on */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); + /* IRQ Enable: Alll on */ + /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); */ /* IRQ Enable: All off */ - snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ® ); - snd_printk(KERN_INFO "emu1010: Card options3=0x%x\n",reg); + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, ®); + snd_printk(KERN_INFO "emu1010: Card options3 = 0x%x\n", reg); /* Default WCLK set to 48kHz. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x00); /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ /* Audio Dock LEDs. */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); #if 0 /* For 96kHz */ @@ -992,7 +994,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) * Defaults only, users will set their own values anyways, let's * just copy/paste. */ - + snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); snd_emu1010_fpga_link_dst_src_write(emu, @@ -1037,19 +1039,19 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_F, EMU_SRC_HAMOA_ADC_LEFT2); #endif - for (i = 0;i < 0x20; i++ ) { - /* AudioDock Elink <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0100+i, EMU_SRC_SILENCE); + for (i = 0; i < 0x20; i++) { + /* AudioDock Elink <- Silence */ + snd_emu1010_fpga_link_dst_src_write(emu, 0x0100 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 4; i++) { + for (i = 0; i < 4; i++) { /* Hana SPDIF Out <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0200+i, EMU_SRC_SILENCE); + snd_emu1010_fpga_link_dst_src_write(emu, 0x0200 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hamoa DAC <- Silence */ - snd_emu1010_fpga_link_dst_src_write(emu, 0x0300+i, EMU_SRC_SILENCE); + snd_emu1010_fpga_link_dst_src_write(emu, 0x0300 + i, EMU_SRC_SILENCE); } - for (i = 0;i < 7; i++) { + for (i = 0; i < 7; i++) { /* Hana ADAT Out <- Silence */ snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_HANA_ADAT + i, EMU_SRC_SILENCE); } @@ -1065,30 +1067,30 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE_I2S2_LEFT, EMU_SRC_DOCK_ADC3_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE_I2S2_RIGHT, EMU_SRC_DOCK_ADC3_RIGHT1); - snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01 ); // Unmute all + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x01); /* Unmute all */ + + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Mute all codecs. */ - outl(0x0000a000, emu->port + HCFG); + outl(0x0000a000, emu->port + HCFG); /* AC97 1.03, Any 32Meg of 2Gig address, Auto-Mute, EMU32 Slave, * Lock Sound Memory Cache, Lock Tank Memory Cache, * Un-Mute all codecs. */ outl(0x0000a001, emu->port + HCFG); - + /* Initial boot complete. Now patches */ - snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19 ); /* MIDI Route */ - snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c ); /* Unknown */ - snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp ); - snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10 ); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ + snd_emu1010_fpga_read(emu, EMU_HANA_OPTION_CARDS, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* MIDI Route */ + snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); /* Unknown */ + snd_emu1010_fpga_read(emu, EMU_HANA_SPDIF_MODE, &tmp); + snd_emu1010_fpga_write(emu, EMU_HANA_SPDIF_MODE, 0x10); /* SPDIF Format spdif (or 0x11 for aes/ebu) */ /* Start Micro/Audio Dock firmware loader thread */ if (!emu->emu1010.firmware_thread) { @@ -1218,20 +1220,20 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) emu->emu1010.output_source[23] = 28; } /* TEMP: Select SPDIF in/out */ - //snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); /* Output spdif */ + /* snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, 0x0); */ /* Output spdif */ /* TEMP: Select 48kHz SPDIF out */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x0); /* Mute all */ snd_emu1010_fpga_write(emu, EMU_HANA_DEFCLOCK, 0x0); /* Default fallback clock 48kHz */ /* Word Clock source, Internal 48kHz x1 */ - snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K ); - //snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X ); + snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K); + /* snd_emu1010_fpga_write(emu, EMU_HANA_WCLOCK, EMU_HANA_WCLOCK_INT_48K | EMU_HANA_WCLOCK_4X); */ emu->emu1010.internal_clock = 1; /* 48000 */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12);/* Set LEDs on Audio Dock */ + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_LEDS_2, 0x12); /* Set LEDs on Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x0); /* Mute all */ - //snd_emu1010_fpga_write(emu, 0x7, 0x1); /* Unmute all */ - //snd_emu1010_fpga_write(emu, 0xe, 0x12); /* Set LEDs on Audio Dock */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x0); */ /* Mute all */ + /* snd_emu1010_fpga_write(emu, 0x7, 0x1); */ /* Unmute all */ + /* snd_emu1010_fpga_write(emu, 0xe, 0x12); */ /* Set LEDs on Audio Dock */ return 0; } @@ -1247,13 +1249,13 @@ static void free_pm_buffer(struct snd_emu10k1 *emu); static int snd_emu10k1_free(struct snd_emu10k1 *emu) { if (emu->port) { /* avoid access to already used hardware */ - snd_emu10k1_fx8010_tram_setup(emu, 0); + snd_emu10k1_fx8010_tram_setup(emu, 0); snd_emu10k1_done(emu); snd_emu10k1_free_efx(emu); - } + } if (emu->card_capabilities->emu_model == EMU_MODEL_EMU1010) { /* Disable 48Volt power to Audio Dock */ - snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0 ); + snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, 0); } if (emu->emu1010.firmware_thread) kthread_stop(emu->emu1010.firmware_thread); @@ -1278,7 +1280,7 @@ static int snd_emu10k1_free(struct snd_emu10k1 *emu) #endif if (emu->port) pci_release_regions(emu->pci); - if (emu->card_capabilities->ca0151_chip) /* P16V */ + if (emu->card_capabilities->ca0151_chip) /* P16V */ snd_p16v_free(emu); pci_disable_device(emu->pci); kfree(emu); @@ -1292,21 +1294,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device) } static struct snd_emu_chip_details emu_chip_details[] = { - /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ - /* Tested by James@superbug.co.uk 3rd July 2005 */ - /* DSP: CA0108-IAT - * DAC: CS4382-KQ - * ADC: Philips 1361T - * AC97: STAC9750 - * CA0151: None - */ - {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, - .driver = "Audigy2", .name = "Audigy 2 Value [SB0400]", - .id = "Audigy2", - .emu10k2_chip = 1, - .ca0108_chip = 1, - .spk71 = 1, - .ac97_chip = 1} , /* Audigy4 (Not PRO) SB0610 */ /* Tested by James@superbug.co.uk 4th April 2006 */ /* A_IOCFG bits @@ -1346,20 +1333,37 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: None */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, - .driver = "Audigy2", .name = "Audigy 4 [SB0610]", + .driver = "Audigy2", .name = "SB Audigy 4 [SB0610]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , + /* Audigy 2 Value AC3 out does not work yet. + * Need to find out how to turn off interpolators. + */ + /* Tested by James@superbug.co.uk 3rd July 2005 */ + /* DSP: CA0108-IAT + * DAC: CS4382-KQ + * ADC: Philips 1361T + * AC97: STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10011102, + .driver = "Audigy2", .name = "SB Audigy 2 Value [SB0400]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .ac97_chip = 1} , /* Audigy 2 ZS Notebook Cardbus card.*/ /* Tested by James@superbug.co.uk 6th November 2006 */ /* Audio output 7.1/Headphones working. * Digital output working. (AC3 not checked, only PCM) * Audio Mic/Line inputs working. * Digital input not tested. - */ + */ /* DSP: Tina2 * DAC: Wolfson WM8768/WM8568 * ADC: Wolfson WM8775 @@ -1386,7 +1390,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS Notebook [SB0530]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS Notebook [SB0530]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1396,7 +1400,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1} , /* Tested by James@superbug.co.uk 4th Nov 2007. */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x42011102, - .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", + .driver = "Audigy2", .name = "E-mu 1010 Notebook [MAEM8950]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, @@ -1404,47 +1408,49 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spk71 = 1 , .emu_model = EMU_MODEL_EMU1616}, /* Tested by James@superbug.co.uk 4th Nov 2007. */ + /* This is MAEM8960, 0202 is MAEM 8980 */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, - .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM8960]", .id = "EMU1010", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU1010B}, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ /* Tested by James@superbug.co.uk 8th July 2005. */ + /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102, - .driver = "Audigy2", .name = "E-mu 1010 [4001]", + .driver = "Audigy2", .name = "E-mu 1010 [MAEM8810]", .id = "EMU1010", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + .emu_model = EMU_MODEL_EMU1010}, /* EMU 1010 old revision */ /* EMU0404b */ {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .driver = "Audigy2", .name = "E-mu 0404b PCI [MAEM8852]", .id = "EMU0404", .emu10k2_chip = 1, .ca0108_chip = 1, .spk71 = 1, - .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 new revision */ /* Tested by James@superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, - .driver = "Audigy2", .name = "E-mu 0404 [4002]", + .driver = "Audigy2", .name = "E-mu 0404 [MAEM8850]", .id = "EMU0404", .emu10k2_chip = 1, .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ - /* Audigy4 (Not PRO) SB0610 */ - {.vendor = 0x1102, .device = 0x0008, - .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", + /* Note that all E-mu cards require kernel 2.6 or newer. */ + {.vendor = 0x1102, .device = 0x0008, + .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0108_chip = 1, .ac97_chip = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20071102, - .driver = "Audigy2", .name = "Audigy 4 PRO [SB0380]", + .driver = "Audigy2", .name = "SB Audigy 4 PRO [SB0380]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1457,31 +1463,34 @@ static struct snd_emu_chip_details emu_chip_details[] = { * Just like 0x20021102 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20061102, - .driver = "Audigy2", .name = "Audigy 2 [SB0350b]", + .driver = "Audigy2", .name = "SB Audigy 2 [SB0350b]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20021102, - .driver = "Audigy2", .name = "Audigy 2 ZS [SB0350]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0350]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x20011102, - .driver = "Audigy2", .name = "Audigy 2 ZS [2001]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0360]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , /* Audigy 2 */ /* Tested by James@superbug.co.uk 3rd July 2005 */ @@ -1492,7 +1501,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { * CA0151: Yes */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10071102, - .driver = "Audigy2", .name = "Audigy 2 [SB0240]", + .driver = "Audigy2", .name = "SB Audigy 2 [SB0240]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1502,7 +1511,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10051102, - .driver = "Audigy2", .name = "Audigy 2 EX [1005]", + .driver = "Audigy2", .name = "SB Audigy 2 Platinum EX [SB0280]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1512,7 +1521,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { /* Dell OEM/Creative Labs Audigy 2 ZS */ /* See ALSA bug#1365 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10031102, - .driver = "Audigy2", .name = "Audigy 2 ZS [SB0353]", + .driver = "Audigy2", .name = "SB Audigy 2 ZS [SB0353]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1521,7 +1530,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, - .driver = "Audigy2", .name = "Audigy 2 Platinum [SB0240P]", + .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1532,7 +1541,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .adc_1361t = 1, /* 24 bit capture instead of 16bit. Fixes ALSA bug#324 */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .revision = 0x04, - .driver = "Audigy2", .name = "Audigy 2 [Unknown]", + .driver = "Audigy2", .name = "SB Audigy 2 [Unknown]", .id = "Audigy2", .emu10k2_chip = 1, .ca0102_chip = 1, @@ -1540,78 +1549,79 @@ static struct snd_emu_chip_details emu_chip_details[] = { .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00531102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "SB Audigy 1 [SB0092]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00521102, - .driver = "Audigy", .name = "Audigy 1 ES [SB0160]", + .driver = "Audigy", .name = "SB Audigy 1 ES [SB0160]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .spdif_bug = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x00511102, - .driver = "Audigy", .name = "Audigy 1 [SB0090]", + .driver = "Audigy", .name = "SB Audigy 1 [SB0090]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, - .driver = "Audigy", .name = "Audigy 1 [Unknown]", + .driver = "Audigy", .name = "Audigy 1 [Unknown]", .id = "Audigy", .emu10k2_chip = 1, .ca0102_chip = 1, .ac97_chip = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806B1102, - .driver = "EMU10K1", .name = "SBLive! [SB0105]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806A1102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0103]", + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806b1102, + .driver = "EMU10K1", .name = "SB Live! [SB0105]", + .id = "Live", + .emu10k1_chip = 1, + .ac97_chip = 1, + .sblive51 = 1} , + {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x806a1102, + .driver = "EMU10K1", .name = "SB Live! Value [SB0103]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80691102, - .driver = "EMU10K1", .name = "SBLive! Value [SB0101]", + .driver = "EMU10K1", .name = "SB Live! Value [SB0101]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by ALSA bug#1680 26th December 2005 */ - /* note: It really has SB0220 written on the card. */ + /* note: It really has SB0220 written on the card, */ + /* but it's SB0228 according to kx.inf */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80661102, - .driver = "EMU10K1", .name = "SB Live 5.1 Dell OEM [SB0220]", + .driver = "EMU10K1", .name = "SB Live! 5.1 Dell OEM [SB0228]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by Thomas Zehetbauer 27th Aug 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80651102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", - .id = "Live", - .emu10k1_chip = 1, - .ac97_chip = 1, - .sblive51 = 1} , - {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x100a1102, - .driver = "EMU10K1", .name = "SB Live 5.1 [SB0220]", + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0220]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80641102, - .driver = "EMU10K1", .name = "SB Live 5.1", + .driver = "EMU10K1", .name = "SB Live! 5.1", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by alsa bugtrack user "hus" bug #1297 12th Aug 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80611102, - .driver = "EMU10K1", .name = "SBLive 5.1 [SB0060]", + .driver = "EMU10K1", .name = "SB Live! 5.1 [SB0060]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 2, /* ac97 is optional; both SBLive 5.1 and platinum @@ -1619,78 +1629,78 @@ static struct snd_emu_chip_details emu_chip_details[] = { */ .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80511102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4850]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4850]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80401102, - .driver = "EMU10K1", .name = "SBLive! Platinum [CT4760P]", + .driver = "EMU10K1", .name = "SB Live! Platinum [CT4760P]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80321102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4871]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4871]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80311102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4831]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4831]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80281102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4870]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4870]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80271102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4832]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4832]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80261102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4830]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4830]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80231102, - .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", + .driver = "EMU10K1", .name = "SB PCI512 [CT4790]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x80221102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4780]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4780]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x40011102, - .driver = "EMU10K1", .name = "E-mu APS [4001]", + .driver = "EMU10K1", .name = "E-mu APS [PC545]", .id = "APS", .emu10k1_chip = 1, .ecard = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00211102, - .driver = "EMU10K1", .name = "SBLive! [CT4620]", + .driver = "EMU10K1", .name = "SB Live! [CT4620]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, .subsystem = 0x00201102, - .driver = "EMU10K1", .name = "SBLive! Value [CT4670]", + .driver = "EMU10K1", .name = "SB Live! Value [CT4670]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, .sblive51 = 1} , {.vendor = 0x1102, .device = 0x0002, - .driver = "EMU10K1", .name = "SB Live [Unknown]", + .driver = "EMU10K1", .name = "SB Live! [Unknown]", .id = "Live", .emu10k1_chip = 1, .ac97_chip = 1, @@ -1699,13 +1709,13 @@ static struct snd_emu_chip_details emu_chip_details[] = { }; int __devinit snd_emu10k1_create(struct snd_card *card, - struct pci_dev * pci, + struct pci_dev *pci, unsigned short extin_mask, unsigned short extout_mask, long max_cache_bytes, int enable_ir, uint subsystem, - struct snd_emu10k1 ** remu) + struct snd_emu10k1 **remu) { struct snd_emu10k1 *emu; int idx, err; @@ -1715,11 +1725,12 @@ int __devinit snd_emu10k1_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_emu10k1_dev_free, }; - + *remu = NULL; /* enable PCI device */ - if ((err = pci_enable_device(pci)) < 0) + err = pci_enable_device(pci); + if (err < 0) return err; emu = kzalloc(sizeof(*emu), GFP_KERNEL); @@ -1746,16 +1757,17 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model); - snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model); + snd_printdd("vendor = 0x%x, device = 0x%x, subsystem_vendor_id = 0x%x, subsystem_id = 0x%x\n", pci->vendor, pci->device, emu->serial, emu->model); for (c = emu_chip_details; c->vendor; c++) { if (c->vendor == pci->vendor && c->device == pci->device) { if (subsystem) { - if (c->subsystem && (c->subsystem == subsystem) ) { + if (c->subsystem && (c->subsystem == subsystem)) break; - } else continue; + else + continue; } else { - if (c->subsystem && (c->subsystem != emu->serial) ) + if (c->subsystem && (c->subsystem != emu->serial)) continue; if (c->revision && c->revision != emu->revision) continue; @@ -1771,14 +1783,18 @@ int __devinit snd_emu10k1_create(struct snd_card *card, } emu->card_capabilities = c; if (c->subsystem && !subsystem) - snd_printdd("Sound card name=%s\n", c->name); - else if (subsystem) - snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x. Forced to subsytem=0x%x\n", - c->name, pci->vendor, pci->device, emu->serial, c->subsystem); - else - snd_printdd("Sound card name=%s, vendor=0x%x, device=0x%x, subsystem=0x%x.\n", - c->name, pci->vendor, pci->device, emu->serial); - + snd_printdd("Sound card name = %s\n", c->name); + else if (subsystem) + snd_printdd("Sound card name = %s, " + "vendor = 0x%x, device = 0x%x, subsystem = 0x%x. " + "Forced to subsytem = 0x%x\n", c->name, + pci->vendor, pci->device, emu->serial, c->subsystem); + else + snd_printdd("Sound card name = %s, " + "vendor = 0x%x, device = 0x%x, subsystem = 0x%x.\n", + c->name, pci->vendor, pci->device, + emu->serial); + if (!*card->id && c->id) { int i, n = 0; strlcpy(card->id, c->id, sizeof(card->id)); @@ -1812,7 +1828,8 @@ int __devinit snd_emu10k1_create(struct snd_card *card, else emu->gpr_base = FXGPREGBASE; - if ((err = pci_request_regions(pci, "EMU10K1")) < 0) { + err = pci_request_regions(pci, "EMU10K1"); + if (err < 0) { kfree(emu); pci_disable_device(pci); return err; @@ -1859,21 +1876,25 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->enable_ir = enable_ir; if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + err = snd_emu10k1_cardbus_init(emu); + if (err < 0) goto error; } if (emu->card_capabilities->ecard) { - if ((err = snd_emu10k1_ecard_init(emu)) < 0) + err = snd_emu10k1_ecard_init(emu); + if (err < 0) goto error; } else if (emu->card_capabilities->emu_model) { - if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { - snd_emu10k1_free(emu); - return err; - } + err = snd_emu10k1_emu1010_init(emu); + if (err < 0) { + snd_emu10k1_free(emu); + return err; + } } else { /* 5.1: Enable the additional AC97 Slots. If the emu10k1 version does not support this, it shouldn't do any harm */ - snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); + snd_emu10k1_ptr_write(emu, AC97SLOT, 0, + AC97SLOT_CNTR|AC97SLOT_LFE); } /* initialize TRAM setup */ @@ -1913,7 +1934,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, snd_emu10k1_synth_alloc(emu, 4096); if (emu->reserved_page) emu->reserved_page->map_locked = 1; - + /* Clear silent pages and set up pointers */ memset(emu->silent_page.area, 0, PAGE_SIZE); silent_page = emu->silent_page.addr << 1; @@ -1926,19 +1947,23 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->voices[idx].number = idx; } - if ((err = snd_emu10k1_init(emu, enable_ir, 0)) < 0) + err = snd_emu10k1_init(emu, enable_ir, 0); + if (err < 0) goto error; #ifdef CONFIG_PM - if ((err = alloc_pm_buffer(emu)) < 0) + err = alloc_pm_buffer(emu); + if (err < 0) goto error; #endif /* Initialize the effect engine */ - if ((err = snd_emu10k1_init_efx(emu)) < 0) + err = snd_emu10k1_init_efx(emu); + if (err < 0) goto error; snd_emu10k1_audio_enable(emu); - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops)) < 0) + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, emu, &ops); + if (err < 0) goto error; #ifdef CONFIG_PROC_FS @@ -1978,7 +2003,7 @@ static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); emu->saved_ptr = vmalloc(4 * NUM_G * size); - if (! emu->saved_ptr) + if (!emu->saved_ptr) return -ENOMEM; if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) return -ENOMEM; @@ -2023,7 +2048,7 @@ void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); else if (emu->card_capabilities->emu_model) - snd_emu10k1_emu1010_init(emu); + snd_emu10k1_emu1010_init(emu); else snd_emu10k1_ptr_write(emu, AC97SLOT, 0, AC97SLOT_CNTR|AC97SLOT_LFE); snd_emu10k1_init(emu, emu->enable_ir, 1); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8aee322313ed..9350089e1083 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2183,7 +2183,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 5b442383fcda..58d7cda03de5 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2688,12 +2688,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } err = snd_ice1712_ac97_mixer(ice); if (err < 0) { @@ -2715,12 +2716,13 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } - if (ice_has_con_ac97(ice)) + if (ice_has_con_ac97(ice)) { err = snd_ice1712_pcm_ds(ice, pcm_dev++, NULL); if (err < 0) { snd_card_free(card); return err; } + } if (!c->no_mpu401) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 1b3f11702713..0dfa0540ce2c 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -382,23 +382,25 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) unsigned char status_mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX | VT1724_IRQ_MTPCM; int handled = 0; -#ifdef CONFIG_SND_DEBUG int timeout = 0; -#endif while (1) { status = inb(ICEREG1724(ice, IRQSTAT)); status &= status_mask; if (status == 0) break; -#ifdef CONFIG_SND_DEBUG if (++timeout > 10) { - printk(KERN_ERR - "ice1724: Too long irq loop, status = 0x%x\n", - status); + status = inb(ICEREG1724(ice, IRQSTAT)); + printk(KERN_ERR "ice1724: Too long irq loop, " + "status = 0x%x\n", status); + if (status & VT1724_IRQ_MPU_TX) { + printk(KERN_ERR "ice1724: Disabling MPU_TX\n"); + outb(inb(ICEREG1724(ice, IRQMASK)) | + VT1724_IRQ_MPU_TX, + ICEREG1724(ice, IRQMASK)); + } break; } -#endif handled = 1; if (status & VT1724_IRQ_MPU_TX) { spin_lock(&ice->reg_lock); @@ -2351,7 +2353,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card, { struct snd_ice1712 *ice; int err; - unsigned char mask; static struct snd_device_ops ops = { .dev_free = snd_vt1724_dev_free, }; @@ -2412,9 +2413,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, return -EIO; } - /* unmask used interrupts */ - mask = VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX; - outb(mask, ICEREG1724(ice, IRQMASK)); + /* MPU_RX and TX irq masks are cleared later dynamically */ + outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); + /* don't handle FIFO overrun/underruns (just yet), * since they cause machine lockups */ diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index c88d1eace1c4..19d3391e229f 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2702,6 +2702,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] __devinitdata = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), + SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000), SND_PCI_QUIRK(0x1043, 0x80f3, "AD1985", 48000), { } /* terminator */ }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce649a64..f23a73577c22 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1010,7 +1010,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * .dev_free = snd_mixart_chip_dev_free, }; - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (! chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; @@ -1025,6 +1025,7 @@ static int __devinit snd_mixart_create(struct mixart_mgr *mgr, struct snd_card * return err; } + mgr->chip[idx] = chip; snd_card_set_dev(card, &mgr->pci->dev); return 0; @@ -1314,8 +1315,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); @@ -1378,6 +1378,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); if ((err = snd_mixart_create(mgr, card, i)) < 0) { + snd_card_free(card); snd_mixart_free(mgr); return err; } diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index b60f6212745a..de999c6d6dd3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -61,6 +61,7 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_CMEDIA_REF, /* C-Media's reference design */ MODEL_MERIDIAN, /* AuzenTech X-Meridian */ + MODEL_HALO, /* HT-Omega Claro halo */ }; static struct pci_device_id oxygen_ids[] __devinitdata = { @@ -74,6 +75,7 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF }, + { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO }, { } }; MODULE_DEVICE_TABLE(pci, oxygen_ids); @@ -301,6 +303,8 @@ static int generic_probe(struct oxygen *chip, unsigned long driver_data) PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2 | CAPTURE_1_FROM_SPDIF; + } + if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) { chip->model.misc_flags = OXYGEN_MISC_MIDI; chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; } diff --git a/sound/pci/pcxhr/Makefile b/sound/pci/pcxhr/Makefile index 10473c05918d..b06128e918ca 100644 --- a/sound/pci/pcxhr/Makefile +++ b/sound/pci/pcxhr/Makefile @@ -1,2 +1,2 @@ -snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o +snd-pcxhr-objs := pcxhr.o pcxhr_hwdep.o pcxhr_mixer.o pcxhr_core.o pcxhr_mix22.o obj-$(CONFIG_SND_PCXHR) += snd-pcxhr.o diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 0e06c6c9fcc0..05a1b6cbd72b 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -40,18 +40,20 @@ #include "pcxhr_mixer.h" #include "pcxhr_hwdep.h" #include "pcxhr_core.h" +#include "pcxhr_mix22.h" #define DRIVER_NAME "pcxhr" -MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>"); +MODULE_AUTHOR("Markus Bollinger <bollinger@digigram.com>, " + "Marc Titinger <titinger@digigram.com>"); MODULE_DESCRIPTION("Digigram " DRIVER_NAME " " PCXHR_DRIVER_VERSION_STRING); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Digigram," DRIVER_NAME "}}"); -static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ -static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ -static int mono[SNDRV_CARDS]; /* capture in mono only */ +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ +static int mono[SNDRV_CARDS]; /* capture mono only */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Digigram " DRIVER_NAME " soundcard"); @@ -67,18 +69,58 @@ enum { PCI_ID_PCX882HR, PCI_ID_VX881HR, PCI_ID_PCX881HR, + PCI_ID_VX882E, + PCI_ID_PCX882E, + PCI_ID_VX881E, + PCI_ID_PCX881E, + PCI_ID_VX1222HR, PCI_ID_PCX1222HR, + PCI_ID_VX1221HR, PCI_ID_PCX1221HR, + PCI_ID_VX1222E, + PCI_ID_PCX1222E, + PCI_ID_VX1221E, + PCI_ID_PCX1221E, + PCI_ID_VX222HR, + PCI_ID_VX222E, + PCI_ID_PCX22HR, + PCI_ID_PCX22E, + PCI_ID_VX222HRMIC, + PCI_ID_VX222E_MIC, + PCI_ID_PCX924HR, + PCI_ID_PCX924E, + PCI_ID_PCX924HRMIC, + PCI_ID_PCX924E_MIC, PCI_ID_LAST }; static struct pci_device_id pcxhr_ids[] = { - { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, /* VX882HR */ - { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, /* PCX882HR */ - { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, /* VX881HR */ - { 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, }, /* PCX881HR */ - { 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, /* PCX1222HR */ - { 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, /* PCX1221HR */ + { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb301, 0, 0, PCI_ID_PCX881HR, }, + { 0x10b5, 0x9056, 0x1369, 0xb021, 0, 0, PCI_ID_VX882E, }, + { 0x10b5, 0x9056, 0x1369, 0xb121, 0, 0, PCI_ID_PCX882E, }, + { 0x10b5, 0x9056, 0x1369, 0xb221, 0, 0, PCI_ID_VX881E, }, + { 0x10b5, 0x9056, 0x1369, 0xb321, 0, 0, PCI_ID_PCX881E, }, + { 0x10b5, 0x9656, 0x1369, 0xb401, 0, 0, PCI_ID_VX1222HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb501, 0, 0, PCI_ID_PCX1222HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb601, 0, 0, PCI_ID_VX1221HR, }, + { 0x10b5, 0x9656, 0x1369, 0xb701, 0, 0, PCI_ID_PCX1221HR, }, + { 0x10b5, 0x9056, 0x1369, 0xb421, 0, 0, PCI_ID_VX1222E, }, + { 0x10b5, 0x9056, 0x1369, 0xb521, 0, 0, PCI_ID_PCX1222E, }, + { 0x10b5, 0x9056, 0x1369, 0xb621, 0, 0, PCI_ID_VX1221E, }, + { 0x10b5, 0x9056, 0x1369, 0xb721, 0, 0, PCI_ID_PCX1221E, }, + { 0x10b5, 0x9056, 0x1369, 0xba01, 0, 0, PCI_ID_VX222HR, }, + { 0x10b5, 0x9056, 0x1369, 0xba21, 0, 0, PCI_ID_VX222E, }, + { 0x10b5, 0x9056, 0x1369, 0xbd01, 0, 0, PCI_ID_PCX22HR, }, + { 0x10b5, 0x9056, 0x1369, 0xbd21, 0, 0, PCI_ID_PCX22E, }, + { 0x10b5, 0x9056, 0x1369, 0xbc01, 0, 0, PCI_ID_VX222HRMIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbc21, 0, 0, PCI_ID_VX222E_MIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbb01, 0, 0, PCI_ID_PCX924HR, }, + { 0x10b5, 0x9056, 0x1369, 0xbb21, 0, 0, PCI_ID_PCX924E, }, + { 0x10b5, 0x9056, 0x1369, 0xbf01, 0, 0, PCI_ID_PCX924HRMIC, }, + { 0x10b5, 0x9056, 0x1369, 0xbf21, 0, 0, PCI_ID_PCX924E_MIC, }, { 0, } }; @@ -88,27 +130,55 @@ struct board_parameters { char* board_name; short playback_chips; short capture_chips; + short fw_file_set; short firmware_num; }; static struct board_parameters pcxhr_board_params[] = { -[PCI_ID_VX882HR] = { "VX882HR", 4, 4, 41, }, -[PCI_ID_PCX882HR] = { "PCX882HR", 4, 4, 41, }, -[PCI_ID_VX881HR] = { "VX881HR", 4, 4, 41, }, -[PCI_ID_PCX881HR] = { "PCX881HR", 4, 4, 41, }, -[PCI_ID_PCX1222HR] = { "PCX1222HR", 6, 1, 42, }, -[PCI_ID_PCX1221HR] = { "PCX1221HR", 6, 1, 42, }, +[PCI_ID_VX882HR] = { "VX882HR", 4, 4, 0, 41 }, +[PCI_ID_PCX882HR] = { "PCX882HR", 4, 4, 0, 41 }, +[PCI_ID_VX881HR] = { "VX881HR", 4, 4, 0, 41 }, +[PCI_ID_PCX881HR] = { "PCX881HR", 4, 4, 0, 41 }, +[PCI_ID_VX882E] = { "VX882e", 4, 4, 1, 41 }, +[PCI_ID_PCX882E] = { "PCX882e", 4, 4, 1, 41 }, +[PCI_ID_VX881E] = { "VX881e", 4, 4, 1, 41 }, +[PCI_ID_PCX881E] = { "PCX881e", 4, 4, 1, 41 }, +[PCI_ID_VX1222HR] = { "VX1222HR", 6, 1, 2, 42 }, +[PCI_ID_PCX1222HR] = { "PCX1222HR", 6, 1, 2, 42 }, +[PCI_ID_VX1221HR] = { "VX1221HR", 6, 1, 2, 42 }, +[PCI_ID_PCX1221HR] = { "PCX1221HR", 6, 1, 2, 42 }, +[PCI_ID_VX1222E] = { "VX1222e", 6, 1, 3, 42 }, +[PCI_ID_PCX1222E] = { "PCX1222e", 6, 1, 3, 42 }, +[PCI_ID_VX1221E] = { "VX1221e", 6, 1, 3, 42 }, +[PCI_ID_PCX1221E] = { "PCX1221e", 6, 1, 3, 42 }, +[PCI_ID_VX222HR] = { "VX222HR", 1, 1, 4, 44 }, +[PCI_ID_VX222E] = { "VX222e", 1, 1, 4, 44 }, +[PCI_ID_PCX22HR] = { "PCX22HR", 1, 0, 4, 44 }, +[PCI_ID_PCX22E] = { "PCX22e", 1, 0, 4, 44 }, +[PCI_ID_VX222HRMIC] = { "VX222HR-Mic", 1, 1, 5, 44 }, +[PCI_ID_VX222E_MIC] = { "VX222e-Mic", 1, 1, 5, 44 }, +[PCI_ID_PCX924HR] = { "PCX924HR", 1, 1, 5, 44 }, +[PCI_ID_PCX924E] = { "PCX924e", 1, 1, 5, 44 }, +[PCI_ID_PCX924HRMIC] = { "PCX924HR-Mic", 1, 1, 5, 44 }, +[PCI_ID_PCX924E_MIC] = { "PCX924e-Mic", 1, 1, 5, 44 }, }; +/* boards without hw AES1 and SRC onboard are all using fw_file_set==4 */ +/* VX222HR, VX222e, PCX22HR and PCX22e */ +#define PCXHR_BOARD_HAS_AES1(x) (x->fw_file_set != 4) +/* some boards do not support 192kHz on digital AES input plugs */ +#define PCXHR_BOARD_AESIN_NO_192K(x) ((x->capture_chips == 0) || \ + (x->fw_file_set == 0) || \ + (x->fw_file_set == 2)) static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, unsigned int* realfreq) { unsigned int reg; - if (freq < 6900 || freq > 110250) + if (freq < 6900 || freq > 110000) return -EINVAL; - reg = (28224000 * 10) / freq; - reg = (reg + 5) / 10; + reg = (28224000 * 2) / freq; + reg = (reg - 1) / 2; if (reg < 0x200) *pllreg = reg + 0x800; else if (reg < 0x400) @@ -121,7 +191,7 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, reg &= ~3; } if (realfreq) - *realfreq = ((28224000 * 10) / reg + 5) / 10; + *realfreq = (28224000 / (reg + 1)); return 0; } @@ -151,11 +221,6 @@ static int pcxhr_pll_freq_register(unsigned int freq, unsigned int* pllreg, #define PCXHR_FREQ_AES_3 0x03 #define PCXHR_FREQ_AES_4 0x0d -#define PCXHR_MODIFY_CLOCK_S_BIT 0x04 - -#define PCXHR_IRQ_TIMER_FREQ 92000 -#define PCXHR_IRQ_TIMER_PERIOD 48 - static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, unsigned int *reg, unsigned int *freq) { @@ -196,19 +261,32 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, err = pcxhr_send_msg(mgr, &rmh); if (err < 0) { snd_printk(KERN_ERR - "error CMD_ACCESS_IO_WRITE for PLL register : %x!\n", - err ); + "error CMD_ACCESS_IO_WRITE " + "for PLL register : %x!\n", err); return err; } } break; - case PCXHR_CLOCK_TYPE_WORD_CLOCK : val = PCXHR_FREQ_WORD_CLOCK; break; - case PCXHR_CLOCK_TYPE_AES_SYNC : val = PCXHR_FREQ_SYNC_AES; break; - case PCXHR_CLOCK_TYPE_AES_1 : val = PCXHR_FREQ_AES_1; break; - case PCXHR_CLOCK_TYPE_AES_2 : val = PCXHR_FREQ_AES_2; break; - case PCXHR_CLOCK_TYPE_AES_3 : val = PCXHR_FREQ_AES_3; break; - case PCXHR_CLOCK_TYPE_AES_4 : val = PCXHR_FREQ_AES_4; break; - default : return -EINVAL; + case PCXHR_CLOCK_TYPE_WORD_CLOCK: + val = PCXHR_FREQ_WORD_CLOCK; + break; + case PCXHR_CLOCK_TYPE_AES_SYNC: + val = PCXHR_FREQ_SYNC_AES; + break; + case PCXHR_CLOCK_TYPE_AES_1: + val = PCXHR_FREQ_AES_1; + break; + case PCXHR_CLOCK_TYPE_AES_2: + val = PCXHR_FREQ_AES_2; + break; + case PCXHR_CLOCK_TYPE_AES_3: + val = PCXHR_FREQ_AES_3; + break; + case PCXHR_CLOCK_TYPE_AES_4: + val = PCXHR_FREQ_AES_4; + break; + default: + return -EINVAL; } *reg = val; *freq = realfreq; @@ -216,14 +294,13 @@ static int pcxhr_get_clock_reg(struct pcxhr_mgr *mgr, unsigned int rate, } -int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) +static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr, + unsigned int rate, + int *changed) { unsigned int val, realfreq, speed; struct pcxhr_rmh rmh; - int err, changed; - - if (rate == 0) - return 0; /* nothing to do */ + int err; err = pcxhr_get_clock_reg(mgr, rate, &val, &realfreq); if (err) @@ -237,13 +314,17 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) else speed = 2; /* quad speed */ if (mgr->codec_speed != speed) { - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* mute outputs */ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set speed ratio */ rmh.cmd[0] |= IO_NUM_SPEED_RATIO; rmh.cmd[1] = speed; rmh.cmd_len = 2; @@ -253,25 +334,57 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) } /* set the new frequency */ snd_printdd("clock register : set %x\n", val); - err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, val, &changed); + err = pcxhr_write_io_num_reg_cont(mgr, PCXHR_FREQ_REG_MASK, + val, changed); if (err) return err; + mgr->sample_rate_real = realfreq; mgr->cur_clock_type = mgr->use_clock_type; /* unmute after codec speed modes */ if (mgr->codec_speed != speed) { - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* unmute outputs */ rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - mgr->codec_speed = speed; /* save new codec speed */ + mgr->codec_speed = speed; /* save new codec speed */ } + snd_printdd("pcxhr_sub_set_clock to %dHz (realfreq=%d)\n", + rate, realfreq); + return 0; +} + +#define PCXHR_MODIFY_CLOCK_S_BIT 0x04 + +#define PCXHR_IRQ_TIMER_FREQ 92000 +#define PCXHR_IRQ_TIMER_PERIOD 48 + +int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) +{ + struct pcxhr_rmh rmh; + int err, changed; + + if (rate == 0) + return 0; /* nothing to do */ + + if (mgr->is_hr_stereo) + err = hr222_sub_set_clock(mgr, rate, &changed); + else + err = pcxhr_sub_set_clock(mgr, rate, &changed); + + if (err) + return err; + if (changed) { pcxhr_init_rmh(&rmh, CMD_MODIFY_CLOCK); - rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos */ + rmh.cmd[0] |= PCXHR_MODIFY_CLOCK_S_BIT; /* resync fifos */ if (rate < PCXHR_IRQ_TIMER_FREQ) rmh.cmd[1] = PCXHR_IRQ_TIMER_PERIOD; else @@ -282,26 +395,39 @@ int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate) if (err) return err; } - snd_printdd("pcxhr_set_clock to %dHz (realfreq=%d)\n", rate, realfreq); return 0; } -int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, - int *sample_rate) +static int pcxhr_sub_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) { struct pcxhr_rmh rmh; unsigned char reg; int err, rate; switch (clock_type) { - case PCXHR_CLOCK_TYPE_WORD_CLOCK : reg = REG_STATUS_WORD_CLOCK; break; - case PCXHR_CLOCK_TYPE_AES_SYNC : reg = REG_STATUS_AES_SYNC; break; - case PCXHR_CLOCK_TYPE_AES_1 : reg = REG_STATUS_AES_1; break; - case PCXHR_CLOCK_TYPE_AES_2 : reg = REG_STATUS_AES_2; break; - case PCXHR_CLOCK_TYPE_AES_3 : reg = REG_STATUS_AES_3; break; - case PCXHR_CLOCK_TYPE_AES_4 : reg = REG_STATUS_AES_4; break; - default : return -EINVAL; + case PCXHR_CLOCK_TYPE_WORD_CLOCK: + reg = REG_STATUS_WORD_CLOCK; + break; + case PCXHR_CLOCK_TYPE_AES_SYNC: + reg = REG_STATUS_AES_SYNC; + break; + case PCXHR_CLOCK_TYPE_AES_1: + reg = REG_STATUS_AES_1; + break; + case PCXHR_CLOCK_TYPE_AES_2: + reg = REG_STATUS_AES_2; + break; + case PCXHR_CLOCK_TYPE_AES_3: + reg = REG_STATUS_AES_3; + break; + case PCXHR_CLOCK_TYPE_AES_4: + reg = REG_STATUS_AES_4; + break; + default: + return -EINVAL; } pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd_len = 2; @@ -311,7 +437,7 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_ err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - udelay(100); /* wait minimum 2 sample_frames at 32kHz ! */ + udelay(100); /* wait minimum 2 sample_frames at 32kHz ! */ mgr->last_reg_stat = reg; } rmh.cmd[1] = REG_STATUS_CURRENT; @@ -336,6 +462,18 @@ int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_ } +int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) +{ + if (mgr->is_hr_stereo) + return hr222_get_external_clock(mgr, clock_type, + sample_rate); + else + return pcxhr_sub_get_external_clock(mgr, clock_type, + sample_rate); +} + /* * start or stop playback/capture substream */ @@ -350,7 +488,8 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) start = 1; else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { - snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state CANNOT be stopped\n"); + snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state " + "CANNOT be stopped\n"); return -EINVAL; } start = 0; @@ -359,11 +498,12 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) return -EINVAL; stream->timer_abs_periods = 0; - stream->timer_period_frag = 0; /* reset theoretical stream pos */ + stream->timer_period_frag = 0; /* reset theoretical stream pos */ stream->timer_buf_periods = 0; stream->timer_is_synced = 0; - stream_mask = stream->pipe->is_capture ? 1 : 1<<stream->substream->number; + stream_mask = + stream->pipe->is_capture ? 1 : 1<<stream->substream->number; pcxhr_init_rmh(&rmh, start ? CMD_START_STREAM : CMD_STOP_STREAM); pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, @@ -373,8 +513,10 @@ static int pcxhr_set_stream_state(struct pcxhr_stream *stream) err = pcxhr_send_msg(chip->mgr, &rmh); if (err) - snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", err); - stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; + snd_printk(KERN_ERR "ERROR pcxhr_set_stream_state err=%x;\n", + err); + stream->status = + start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; return err; } @@ -399,13 +541,15 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) header = HEADER_FMT_BASE_LIN; break; case SNDRV_PCM_FORMAT_S16_LE: - header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS | HEADER_FMT_INTEL; + header = HEADER_FMT_BASE_LIN | + HEADER_FMT_16BITS | HEADER_FMT_INTEL; break; case SNDRV_PCM_FORMAT_S16_BE: header = HEADER_FMT_BASE_LIN | HEADER_FMT_16BITS; break; case SNDRV_PCM_FORMAT_S24_3LE: - header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS | HEADER_FMT_INTEL; + header = HEADER_FMT_BASE_LIN | + HEADER_FMT_24BITS | HEADER_FMT_INTEL; break; case SNDRV_PCM_FORMAT_S24_3BE: header = HEADER_FMT_BASE_LIN | HEADER_FMT_24BITS; @@ -414,7 +558,8 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) header = HEADER_FMT_BASE_FLOAT | HEADER_FMT_INTEL; break; default: - snd_printk(KERN_ERR "error pcxhr_set_format() : unknown format\n"); + snd_printk(KERN_ERR + "error pcxhr_set_format() : unknown format\n"); return -EINVAL; } chip = snd_pcm_substream_chip(stream->substream); @@ -432,14 +577,31 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) is_capture = stream->pipe->is_capture; stream_num = is_capture ? 0 : stream->substream->number; - pcxhr_init_rmh(&rmh, is_capture ? CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0); - if (is_capture) - rmh.cmd[0] |= 1<<12; + pcxhr_init_rmh(&rmh, is_capture ? + CMD_FORMAT_STREAM_IN : CMD_FORMAT_STREAM_OUT); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, + stream_num, 0); + if (is_capture) { + /* bug with old dsp versions: */ + /* bit 12 also sets the format of the playback stream */ + if (DSP_EXT_CMD_SET(chip->mgr)) + rmh.cmd[0] |= 1<<10; + else + rmh.cmd[0] |= 1<<12; + } rmh.cmd[1] = 0; - rmh.cmd[2] = header >> 8; - rmh.cmd[3] = (header & 0xff) << 16; - rmh.cmd_len = 4; + rmh.cmd_len = 2; + if (DSP_EXT_CMD_SET(chip->mgr)) { + /* add channels and set bit 19 if channels>2 */ + rmh.cmd[1] = stream->channels; + if (!is_capture) { + /* playback : add channel mask to command */ + rmh.cmd[2] = (stream->channels == 1) ? 0x01 : 0x03; + rmh.cmd_len = 3; + } + } + rmh.cmd[rmh.cmd_len++] = header >> 8; + rmh.cmd[rmh.cmd_len++] = (header & 0xff) << 16; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) snd_printk(KERN_ERR "ERROR pcxhr_set_format err=%x;\n", err); @@ -456,30 +618,38 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) is_capture = (subs->stream == SNDRV_PCM_STREAM_CAPTURE); stream_num = is_capture ? 0 : subs->number; - snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + snd_printdd("pcxhr_update_r_buffer(pcm%c%d) : " + "addr(%p) bytes(%zx) subs(%d)\n", is_capture ? 'c' : 'p', chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); pcxhr_init_rmh(&rmh, CMD_UPDATE_R_BUFFERS); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, stream_num, 0); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, stream->pipe->first_audio, + stream_num, 0); /* max buffer size is 2 MByte */ snd_BUG_ON(subs->runtime->dma_bytes >= 0x200000); - rmh.cmd[1] = subs->runtime->dma_bytes * 8; /* size in bits */ - rmh.cmd[2] = subs->runtime->dma_addr >> 24; /* most significant byte */ - rmh.cmd[2] |= 1<<19; /* this is a circular buffer */ - rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD; /* least 3 significant bytes */ + /* size in bits */ + rmh.cmd[1] = subs->runtime->dma_bytes * 8; + /* most significant byte */ + rmh.cmd[2] = subs->runtime->dma_addr >> 24; + /* this is a circular buffer */ + rmh.cmd[2] |= 1<<19; + /* least 3 significant bytes */ + rmh.cmd[3] = subs->runtime->dma_addr & MASK_DSP_WORD; rmh.cmd_len = 4; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) - snd_printk(KERN_ERR "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err); + snd_printk(KERN_ERR + "ERROR CMD_UPDATE_R_BUFFERS err=%x;\n", err); return err; } #if 0 -static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, snd_pcm_uframes_t *sample_count) +static int pcxhr_pipe_sample_count(struct pcxhr_stream *stream, + snd_pcm_uframes_t *sample_count) { struct pcxhr_rmh rmh; int err; @@ -533,8 +703,8 @@ static void pcxhr_trigger_tasklet(unsigned long arg) for (j = 0; j < chip->nb_streams_play; j++) { if (pcxhr_stream_scheduled_get_pipe(&chip->playback_stream[j], &pipe)) { playback_mask |= (1 << pipe->first_audio); - break; /* add only once, as all playback streams of - * one chip use the same pipe + break; /* add only once, as all playback + * streams of one chip use the same pipe */ } } @@ -545,19 +715,21 @@ static void pcxhr_trigger_tasklet(unsigned long arg) return; } - snd_printdd("pcxhr_trigger_tasklet : playback_mask=%x capture_mask=%x\n", + snd_printdd("pcxhr_trigger_tasklet : " + "playback_mask=%x capture_mask=%x\n", playback_mask, capture_mask); /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { mutex_unlock(&mgr->setup_mutex); - snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error stop pipes (P%x C%x)\n", + snd_printk(KERN_ERR "pcxhr_trigger_tasklet : " + "error stop pipes (P%x C%x)\n", playback_mask, capture_mask); return; } - /* unfortunately the dsp lost format and buffer info with the stop pipe */ + /* the dsp lost format and buffer info with the stop pipe */ for (i = 0; i < mgr->num_cards; i++) { struct pcxhr_stream *stream; chip = mgr->chip[i]; @@ -596,12 +768,15 @@ static void pcxhr_trigger_tasklet(unsigned long arg) err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { mutex_unlock(&mgr->setup_mutex); - snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error start pipes (P%x C%x)\n", + snd_printk(KERN_ERR "pcxhr_trigger_tasklet : " + "error start pipes (P%x C%x)\n", playback_mask, capture_mask); return; } - /* put the streams into the running state now (increment pointer by interrupt) */ + /* put the streams into the running state now + * (increment pointer by interrupt) + */ spin_lock_irqsave(&mgr->lock, flags); for ( i =0; i < mgr->num_cards; i++) { struct pcxhr_stream *stream; @@ -615,7 +790,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) stream = &chip->playback_stream[j]; if (stream->status == PCXHR_STREAM_STATUS_STARTED) { /* playback will already have advanced ! */ - stream->timer_period_frag += PCXHR_GRANULARITY; + stream->timer_period_frag += mgr->granularity; stream->status = PCXHR_STREAM_STATUS_RUNNING; } } @@ -697,12 +872,14 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start) pcxhr_init_rmh(&rmh, CMD_SET_TIMER_INTERRUPT); if (start) { - mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; /* last dsp time invalid */ - rmh.cmd[0] |= PCXHR_GRANULARITY; + /* last dsp time invalid */ + mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; + rmh.cmd[0] |= mgr->granularity; } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", err); + snd_printk(KERN_ERR "error pcxhr_hardware_timer err(%x)\n", + err); return err; } @@ -713,38 +890,16 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) { struct snd_pcxhr *chip = snd_pcm_substream_chip(subs); struct pcxhr_mgr *mgr = chip->mgr; - /* - struct pcxhr_stream *stream = (pcxhr_stream_t*)subs->runtime->private_data; - */ int err = 0; snd_printdd("pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", subs->runtime->period_size, subs->runtime->periods, subs->runtime->buffer_size); - /* - if(subs->runtime->period_size <= PCXHR_GRANULARITY) { - snd_printk(KERN_ERR "pcxhr_prepare : error period_size too small (%x)\n", - (unsigned int)subs->runtime->period_size); - return -EINVAL; - } - */ - mutex_lock(&mgr->setup_mutex); do { - /* if the stream was stopped before, format and buffer were reset */ - /* - if(stream->status == PCXHR_STREAM_STATUS_STOPPED) { - err = pcxhr_set_format(stream); - if(err) break; - err = pcxhr_update_r_buffer(stream); - if(err) break; - } - */ - /* only the first stream can choose the sample rate */ - /* the further opened streams will be limited to its frequency (see open) */ /* set the clock only once (first stream) */ if (mgr->sample_rate != subs->runtime->rate) { err = pcxhr_set_clock(mgr, subs->runtime->rate); @@ -787,22 +942,9 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, stream->channels = channels; stream->format = format; - /* set the format to the board */ - /* - err = pcxhr_set_format(stream); - if(err) { - mutex_unlock(&mgr->setup_mutex); - return err; - } - */ /* allocate buffer */ err = snd_pcm_lib_malloc_pages(subs, params_buffer_bytes(hw)); - /* - if (err > 0) { - err = pcxhr_update_r_buffer(stream); - } - */ mutex_unlock(&mgr->setup_mutex); return err; @@ -820,14 +962,18 @@ static int pcxhr_hw_free(struct snd_pcm_substream *subs) */ static struct snd_pcm_hardware pcxhr_caps = { - .info = ( SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | - 0 /*SNDRV_PCM_INFO_PAUSE*/), - .formats = ( SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | - SNDRV_PCM_FMTBIT_FLOAT_LE ), - .rates = SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000, + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_3BE | + SNDRV_PCM_FMTBIT_FLOAT_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_192000), .rate_min = 8000, .rate_max = 192000, .channels_min = 1, @@ -847,6 +993,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) struct pcxhr_mgr *mgr = chip->mgr; struct snd_pcm_runtime *runtime = subs->runtime; struct pcxhr_stream *stream; + int err; mutex_lock(&mgr->setup_mutex); @@ -874,6 +1021,18 @@ static int pcxhr_open(struct snd_pcm_substream *subs) return -EBUSY; } + /* float format support is in some cases buggy on stereo cards */ + if (mgr->is_hr_stereo) + runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_FLOAT_LE; + + /* buffer-size should better be multiple of period-size */ + err = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + mutex_unlock(&mgr->setup_mutex); + return err; + } + /* if a sample rate is already used or fixed by external clock, * the stream cannot change */ @@ -889,7 +1048,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs) mutex_unlock(&mgr->setup_mutex); return -EBUSY; } - runtime->hw.rate_min = runtime->hw.rate_max = external_rate; + runtime->hw.rate_min = external_rate; + runtime->hw.rate_max = external_rate; } } @@ -899,9 +1059,11 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->private_data = stream; - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4); - snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 4); - + /* better get a divisor of granularity values (96 or 192) */ + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); snd_pcm_set_sync(subs); mgr->ref_count_rate++; @@ -919,11 +1081,12 @@ static int pcxhr_close(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - snd_printdd("pcxhr_close chip%d subs%d\n", chip->chip_idx, subs->number); + snd_printdd("pcxhr_close chip%d subs%d\n", + chip->chip_idx, subs->number); /* sample rate released */ if (--mgr->ref_count_rate == 0) { - mgr->sample_rate = 0; /* the sample rate is no more locked */ + mgr->sample_rate = 0; /* the sample rate is no more locked */ pcxhr_hardware_timer(mgr, 0); /* stop the DSP-timer */ } @@ -1016,7 +1179,8 @@ static int pcxhr_chip_dev_free(struct snd_device *device) /* */ -static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, int idx) +static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, + struct snd_card *card, int idx) { int err; struct snd_pcxhr *chip; @@ -1024,7 +1188,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, .dev_free = pcxhr_chip_dev_free, }; - mgr->chip[idx] = chip = kzalloc(sizeof(*chip), GFP_KERNEL); + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (! chip) { snd_printk(KERN_ERR "cannot allocate chip\n"); return -ENOMEM; @@ -1040,7 +1204,7 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, if (idx < mgr->capture_chips) { if (mgr->mono_capture) - chip->nb_streams_capt = 2; /* 2 mono streams (left+right) */ + chip->nb_streams_capt = 2; /* 2 mono streams */ else chip->nb_streams_capt = 1; /* or 1 stereo stream */ } @@ -1050,13 +1214,15 @@ static int __devinit pcxhr_create(struct pcxhr_mgr *mgr, struct snd_card *card, return err; } + mgr->chip[idx] = chip; snd_card_set_dev(card, &mgr->pci->dev); return 0; } /* proc interface */ -static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void pcxhr_proc_info(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct snd_pcxhr *chip = entry->private_data; struct pcxhr_mgr *mgr = chip->mgr; @@ -1069,8 +1235,10 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer short ver_maj = (mgr->dsp_version >> 16) & 0xff; short ver_min = (mgr->dsp_version >> 8) & 0xff; short ver_build = mgr->dsp_version & 0xff; - snd_iprintf(buffer, "module version %s\n", PCXHR_DRIVER_VERSION_STRING); - snd_iprintf(buffer, "dsp version %d.%d.%d\n", ver_maj, ver_min, ver_build); + snd_iprintf(buffer, "module version %s\n", + PCXHR_DRIVER_VERSION_STRING); + snd_iprintf(buffer, "dsp version %d.%d.%d\n", + ver_maj, ver_min, ver_build); if (mgr->board_has_analog) snd_iprintf(buffer, "analog io available\n"); else @@ -1084,18 +1252,22 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer if (ref > 0) { if (mgr->sample_rate_real != 0 && mgr->sample_rate_real != 48000) { - ref = (ref * 48000) / mgr->sample_rate_real; - if (mgr->sample_rate_real >= PCXHR_IRQ_TIMER_FREQ) + ref = (ref * 48000) / + mgr->sample_rate_real; + if (mgr->sample_rate_real >= + PCXHR_IRQ_TIMER_FREQ) ref *= 2; } cur = 100 - (100 * cur) / ref; snd_iprintf(buffer, "cpu load %d%%\n", cur); - snd_iprintf(buffer, "buffer pool %d/%d kWords\n", + snd_iprintf(buffer, "buffer pool %d/%d\n", rmh.stat[2], rmh.stat[3]); } } - snd_iprintf(buffer, "dma granularity : %d\n", PCXHR_GRANULARITY); - snd_iprintf(buffer, "dsp time errors : %d\n", mgr->dsp_time_err); + snd_iprintf(buffer, "dma granularity : %d\n", + mgr->granularity); + snd_iprintf(buffer, "dsp time errors : %d\n", + mgr->dsp_time_err); snd_iprintf(buffer, "dsp async pipe xrun errors : %d\n", mgr->async_err_pipe_xrun); snd_iprintf(buffer, "dsp async stream xrun errors : %d\n", @@ -1110,33 +1282,52 @@ static void pcxhr_proc_info(struct snd_info_entry *entry, struct snd_info_buffer rmh.cmd_idx = CMD_LAST_INDEX; if( ! pcxhr_send_msg(mgr, &rmh) ) { int i; + if (rmh.stat_len > 8) + rmh.stat_len = 8; for (i = 0; i < rmh.stat_len; i++) - snd_iprintf(buffer, "debug[%02d] = %06x\n", i, rmh.stat[i]); + snd_iprintf(buffer, "debug[%02d] = %06x\n", + i, rmh.stat[i]); } } else snd_iprintf(buffer, "no firmware loaded\n"); snd_iprintf(buffer, "\n"); } -static void pcxhr_proc_sync(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +static void pcxhr_proc_sync(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct snd_pcxhr *chip = entry->private_data; struct pcxhr_mgr *mgr = chip->mgr; - static char *texts[7] = { - "Internal", "Word", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4" + static const char *textsHR22[3] = { + "Internal", "AES Sync", "AES 1" + }; + static const char *textsPCXHR[7] = { + "Internal", "Word", "AES Sync", + "AES 1", "AES 2", "AES 3", "AES 4" }; + const char **texts; + int max_clock; + if (mgr->is_hr_stereo) { + texts = textsHR22; + max_clock = HR22_CLOCK_TYPE_MAX; + } else { + texts = textsPCXHR; + max_clock = PCXHR_CLOCK_TYPE_MAX; + } snd_iprintf(buffer, "\n%s\n", mgr->longname); - snd_iprintf(buffer, "Current Sample Clock\t: %s\n", texts[mgr->cur_clock_type]); - snd_iprintf(buffer, "Current Sample Rate\t= %d\n", mgr->sample_rate_real); - + snd_iprintf(buffer, "Current Sample Clock\t: %s\n", + texts[mgr->cur_clock_type]); + snd_iprintf(buffer, "Current Sample Rate\t= %d\n", + mgr->sample_rate_real); /* commands available when embedded DSP is running */ if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { int i, err, sample_rate; - for (i = PCXHR_CLOCK_TYPE_WORD_CLOCK; i< (3 + mgr->capture_chips); i++) { + for (i = 1; i <= max_clock; i++) { err = pcxhr_get_external_clock(mgr, i, &sample_rate); if (err) break; - snd_iprintf(buffer, "%s Clock\t\t= %d\n", texts[i], sample_rate); + snd_iprintf(buffer, "%s Clock\t\t= %d\n", + texts[i], sample_rate); } } else snd_iprintf(buffer, "no firmware loaded\n"); @@ -1194,7 +1385,8 @@ static int pcxhr_free(struct pcxhr_mgr *mgr) /* * probe function - creates the card manager */ -static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +static int __devinit pcxhr_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) { static int dev; struct pcxhr_mgr *mgr; @@ -1217,7 +1409,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id /* check if we can restrict PCI DMA transfers to 32 bits */ if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { - snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); + snd_printk(KERN_ERR "architecture does not support " + "32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; } @@ -1229,13 +1422,30 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id return -ENOMEM; } - if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) + if (snd_BUG_ON(pci_id->driver_data >= PCI_ID_LAST)) { + kfree(mgr); + pci_disable_device(pci); return -ENODEV; - card_name = pcxhr_board_params[pci_id->driver_data].board_name; - mgr->playback_chips = pcxhr_board_params[pci_id->driver_data].playback_chips; - mgr->capture_chips = pcxhr_board_params[pci_id->driver_data].capture_chips; - mgr->firmware_num = pcxhr_board_params[pci_id->driver_data].firmware_num; + } + card_name = + pcxhr_board_params[pci_id->driver_data].board_name; + mgr->playback_chips = + pcxhr_board_params[pci_id->driver_data].playback_chips; + mgr->capture_chips = + pcxhr_board_params[pci_id->driver_data].capture_chips; + mgr->fw_file_set = + pcxhr_board_params[pci_id->driver_data].fw_file_set; + mgr->firmware_num = + pcxhr_board_params[pci_id->driver_data].firmware_num; mgr->mono_capture = mono[dev]; + mgr->is_hr_stereo = (mgr->playback_chips == 1); + mgr->board_has_aes1 = PCXHR_BOARD_HAS_AES1(mgr); + mgr->board_aes_in_192k = !PCXHR_BOARD_AESIN_NO_192K(mgr); + + if (mgr->is_hr_stereo) + mgr->granularity = PCXHR_GRANULARITY_HR22; + else + mgr->granularity = PCXHR_GRANULARITY; /* resource assignment */ if ((err = pci_request_regions(pci, card_name)) < 0) { @@ -1258,7 +1468,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id mgr->irq = pci->irq; sprintf(mgr->shortname, "Digigram %s", card_name); - sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", mgr->shortname, + sprintf(mgr->longname, "%s at 0x%lx & 0x%lx, 0x%lx irq %i", + mgr->shortname, mgr->port[0], mgr->port[1], mgr->port[2], mgr->irq); /* ISR spinlock */ @@ -1269,10 +1480,14 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id mutex_init(&mgr->setup_mutex); /* init taslket */ - tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr); - tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, (unsigned long) mgr); + tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, + (unsigned long) mgr); + tasklet_init(&mgr->trigger_taskq, pcxhr_trigger_tasklet, + (unsigned long) mgr); + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + - sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - + PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); if (! mgr->prmh) { pcxhr_free(mgr); @@ -1293,7 +1508,8 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id else idx = index[dev] + i; - snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : card_name, i); + snprintf(tmpid, sizeof(tmpid), "%s-%d", + id[dev] ? id[dev] : card_name, i); card = snd_card_new(idx, tmpid, THIS_MODULE, 0); if (! card) { @@ -1307,6 +1523,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id sprintf(card->longname, "%s [PCM #%d]", mgr->longname, i); if ((err = pcxhr_create(mgr, card, i)) < 0) { + snd_card_free(card); pcxhr_free(mgr); return err; } diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 652064787a55..84131a916c92 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -27,15 +27,18 @@ #include <linux/mutex.h> #include <sound/pcm.h> -#define PCXHR_DRIVER_VERSION 0x000804 /* 0.8.4 */ -#define PCXHR_DRIVER_VERSION_STRING "0.8.4" /* 0.8.4 */ +#define PCXHR_DRIVER_VERSION 0x000905 /* 0.9.5 */ +#define PCXHR_DRIVER_VERSION_STRING "0.9.5" /* 0.9.5 */ -#define PCXHR_MAX_CARDS 6 -#define PCXHR_PLAYBACK_STREAMS 4 +#define PCXHR_MAX_CARDS 6 +#define PCXHR_PLAYBACK_STREAMS 4 -#define PCXHR_GRANULARITY 96 /* transfer granularity (should be min 96 and multiple of 48) */ -#define PCXHR_GRANULARITY_MIN 96 /* transfer granularity of pipes and the dsp time (MBOX4) */ +#define PCXHR_GRANULARITY 96 /* min 96 and multiple of 48 */ +/* transfer granularity of pipes and the dsp time (MBOX4) */ +#define PCXHR_GRANULARITY_MIN 96 +/* TODO : granularity could be 64 or 128 */ +#define PCXHR_GRANULARITY_HR22 192 /* granularity for stereo cards */ struct snd_pcxhr; struct pcxhr_mgr; @@ -51,6 +54,11 @@ enum pcxhr_clock_type { PCXHR_CLOCK_TYPE_AES_2, PCXHR_CLOCK_TYPE_AES_3, PCXHR_CLOCK_TYPE_AES_4, + PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4, + HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL, + HR22_CLOCK_TYPE_AES_SYNC, + HR22_CLOCK_TYPE_AES_1, + HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1, }; struct pcxhr_mgr { @@ -61,6 +69,8 @@ struct pcxhr_mgr { int irq; + int granularity; + /* card access with 1 mem bar and 2 io bar's */ unsigned long port[3]; @@ -83,11 +93,16 @@ struct pcxhr_mgr { /* hardware interface */ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */ unsigned int dsp_version; /* read from embedded once firmware is loaded */ - int board_has_analog; /* if 0 the board is digital only */ - int mono_capture; /* if 1 the board does mono capture */ - int playback_chips; /* 4 or 6 */ - int capture_chips; /* 4 or 1 */ - int firmware_num; /* 41 or 42 */ + int playback_chips; + int capture_chips; + int fw_file_set; + int firmware_num; + int is_hr_stereo:1; + int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + int board_has_analog:1; /* if 0 the board is digital only */ + int board_has_mic:1; /* if 1 the board has microphone input */ + int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; @@ -106,6 +121,9 @@ struct pcxhr_mgr { int async_err_stream_xrun; int async_err_pipe_xrun; int async_err_other_last; + + unsigned char xlx_cfg; /* copy of PCXHR_XLX_CFG register */ + unsigned char xlx_selmic; /* copy of PCXHR_XLX_SELMIC register */ }; @@ -155,24 +173,30 @@ struct snd_pcxhr { struct snd_pcm *pcm; /* PCM */ - struct pcxhr_pipe playback_pipe; /* 1 stereo pipe only */ - struct pcxhr_pipe capture_pipe[2]; /* 1 stereo pipe or 2 mono pipes */ + struct pcxhr_pipe playback_pipe; /* 1 stereo pipe only */ + struct pcxhr_pipe capture_pipe[2]; /* 1 stereo or 2 mono pipes */ struct pcxhr_stream playback_stream[PCXHR_PLAYBACK_STREAMS]; - struct pcxhr_stream capture_stream[2]; /* 1 stereo stream or 2 mono streams */ + struct pcxhr_stream capture_stream[2]; /* 1 stereo or 2 mono streams */ int nb_streams_play; int nb_streams_capt; - int analog_playback_active[2]; /* Mixer : Master Playback active (!mute) */ - int analog_playback_volume[2]; /* Mixer : Master Playback Volume */ - int analog_capture_volume[2]; /* Mixer : Master Capture Volume */ - int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Active [streams][stereo]*/ - int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2]; /* Mixer : Digital Playback Volume [streams][stereo]*/ - int digital_capture_volume[2]; /* Mixer : Digital Capture Volume [stereo] */ - int monitoring_active[2]; /* Mixer : Monitoring Active */ - int monitoring_volume[2]; /* Mixer : Monitoring Volume */ - int audio_capture_source; /* Mixer : Audio Capture Source */ - unsigned char aes_bits[5]; /* Mixer : IEC958_AES bits */ + int analog_playback_active[2]; /* Mixer : Master Playback !mute */ + int analog_playback_volume[2]; /* Mixer : Master Playback Volume */ + int analog_capture_volume[2]; /* Mixer : Master Capture Volume */ + int digital_playback_active[PCXHR_PLAYBACK_STREAMS][2]; + int digital_playback_volume[PCXHR_PLAYBACK_STREAMS][2]; + int digital_capture_volume[2]; /* Mixer : Digital Capture Volume */ + int monitoring_active[2]; /* Mixer : Monitoring Active */ + int monitoring_volume[2]; /* Mixer : Monitoring Volume */ + int audio_capture_source; /* Mixer : Audio Capture Source */ + int mic_volume; /* used by cards with MIC only */ + int mic_boost; /* used by cards with MIC only */ + int mic_active; /* used by cards with MIC only */ + int analog_capture_active; /* used by cards with MIC only */ + int phantom_power; /* used by cards with MIC only */ + + unsigned char aes_bits[5]; /* Mixer : IEC958_AES bits */ }; struct pcxhr_hostport @@ -184,6 +208,8 @@ struct pcxhr_hostport /* exported */ int pcxhr_create_pcm(struct snd_pcxhr *chip); int pcxhr_set_clock(struct pcxhr_mgr *mgr, unsigned int rate); -int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate); +int pcxhr_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate); #endif /* __SOUND_PCXHR_H */ diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 7143259cfe34..d5f18226261d 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -132,13 +132,15 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg, *read = PCXHR_INPB(mgr, reg); if ((*read & mask) == bit) { if (i > 100) - snd_printdd("ATTENTION! check_reg(%x) loopcount=%d\n", + snd_printdd("ATTENTION! check_reg(%x) " + "loopcount=%d\n", reg, i); return 0; } i++; } while (time_after_eq(end_time, jiffies)); - snd_printk(KERN_ERR "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=0x%x\n", + snd_printk(KERN_ERR + "pcxhr_check_reg_bit: timeout, reg=%x, mask=0x%x, val=%x\n", reg, mask, *read); return -EIO; } @@ -159,18 +161,22 @@ static int pcxhr_check_reg_bit(struct pcxhr_mgr *mgr, unsigned int reg, #define PCXHR_IT_TEST_XILINX (0x0000003C | PCXHR_MASK_IT_HF1 | \ PCXHR_MASK_IT_MANAGE_HF5) #define PCXHR_IT_DOWNLOAD_BOOT (0x0000000C | PCXHR_MASK_IT_HF1 | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT) #define PCXHR_IT_RESET_BOARD_FUNC (0x0000000C | PCXHR_MASK_IT_HF0 | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT_EXTRA) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT_EXTRA) #define PCXHR_IT_DOWNLOAD_DSP (0x0000000C | \ - PCXHR_MASK_IT_MANAGE_HF5 | PCXHR_MASK_IT_WAIT) + PCXHR_MASK_IT_MANAGE_HF5 | \ + PCXHR_MASK_IT_WAIT) #define PCXHR_IT_DEBUG (0x0000005A | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_RESET_SEMAPHORE (0x0000005C | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_MESSAGE (0x00000074 | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_RESET_CHK (0x00000076 | PCXHR_MASK_IT_NO_HF0_HF1) #define PCXHR_IT_UPDATE_RBUFFER (0x00000078 | PCXHR_MASK_IT_NO_HF0_HF1) -static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atomic) +static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, + unsigned int itdsp, int atomic) { int err; unsigned char reg; @@ -178,17 +184,21 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) { /* clear hf5 bit */ PCXHR_OUTPL(mgr, PCXHR_PLX_MBOX0, - PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & ~PCXHR_MBOX0_HF5); + PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & + ~PCXHR_MBOX0_HF5); } if ((itdsp & PCXHR_MASK_IT_NO_HF0_HF1) == 0) { - reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ; + reg = (PCXHR_ICR_HI08_RREQ | + PCXHR_ICR_HI08_TREQ | + PCXHR_ICR_HI08_HDRQ); if (itdsp & PCXHR_MASK_IT_HF0) reg |= PCXHR_ICR_HI08_HF0; if (itdsp & PCXHR_MASK_IT_HF1) reg |= PCXHR_ICR_HI08_HF1; PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg); } - reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | PCXHR_CVR_HI08_HC); + reg = (unsigned char)(((itdsp & PCXHR_MASK_EXTRA_INFO) >> 1) | + PCXHR_CVR_HI08_HC); PCXHR_OUTPB(mgr, PCXHR_DSP_CVR, reg); if (itdsp & PCXHR_MASK_IT_WAIT) { if (atomic) @@ -211,10 +221,14 @@ static int pcxhr_send_it_dsp(struct pcxhr_mgr *mgr, unsigned int itdsp, int atom } if (itdsp & PCXHR_MASK_IT_MANAGE_HF5) { /* wait for hf5 bit */ - err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, PCXHR_MBOX0_HF5, - PCXHR_MBOX0_HF5, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_PLX_MBOX0, + PCXHR_MBOX0_HF5, + PCXHR_MBOX0_HF5, + PCXHR_TIMEOUT_DSP, + ®); if (err) { - snd_printk(KERN_ERR "pcxhr_send_it_dsp : TIMEOUT HF5\n"); + snd_printk(KERN_ERR + "pcxhr_send_it_dsp : TIMEOUT HF5\n"); return err; } } @@ -263,7 +277,8 @@ void pcxhr_enable_dsp(struct pcxhr_mgr *mgr) /* * load the xilinx image */ -int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilinx, int second) +int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, + const struct firmware *xilinx, int second) { unsigned int i; unsigned int chipsc; @@ -274,7 +289,9 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin /* test first xilinx */ chipsc = PCXHR_INPL(mgr, PCXHR_PLX_CHIPSC); /* REV01 cards do not support the PCXHR_CHIPSC_GPI_USERI bit anymore */ - /* this bit will always be 1; no possibility to test presence of first xilinx */ + /* this bit will always be 1; + * no possibility to test presence of first xilinx + */ if(second) { if ((chipsc & PCXHR_CHIPSC_GPI_USERI) == 0) { snd_printk(KERN_ERR "error loading first xilinx\n"); @@ -290,7 +307,8 @@ int pcxhr_load_xilinx_binary(struct pcxhr_mgr *mgr, const struct firmware *xilin data = *image; mask = 0x80; while (mask) { - chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | PCXHR_CHIPSC_DATA_IN); + chipsc &= ~(PCXHR_CHIPSC_DATA_CLK | + PCXHR_CHIPSC_DATA_IN); if (data & mask) chipsc |= PCXHR_CHIPSC_DATA_IN; PCXHR_OUTPL(mgr, PCXHR_PLX_CHIPSC, chipsc); @@ -330,15 +348,20 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp) data = dsp->data + i; if (i == 0) { /* test data header consistency */ - len = (unsigned int)((data[0]<<16) + (data[1]<<8) + data[2]); - if (len && dsp->size != (len + 2) * 3) + len = (unsigned int)((data[0]<<16) + + (data[1]<<8) + + data[2]); + if (len && (dsp->size != (len + 2) * 3)) return -EINVAL; } /* wait DSP ready for new transfer */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, - PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, &dummy); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_TRDY, + PCXHR_ISR_HI08_TRDY, + PCXHR_TIMEOUT_DSP, &dummy); if (err) { - snd_printk(KERN_ERR "dsp loading error at position %d\n", i); + snd_printk(KERN_ERR + "dsp loading error at position %d\n", i); return err; } /* send host data */ @@ -357,7 +380,8 @@ static int pcxhr_download_dsp(struct pcxhr_mgr *mgr, const struct firmware *dsp) /* * load the eeprom image */ -int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eeprom) +int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, + const struct firmware *eeprom) { int err; unsigned char reg; @@ -365,7 +389,9 @@ int pcxhr_load_eeprom_binary(struct pcxhr_mgr *mgr, const struct firmware *eepro /* init value of the ICR register */ reg = PCXHR_ICR_HI08_RREQ | PCXHR_ICR_HI08_TREQ | PCXHR_ICR_HI08_HDRQ; if (PCXHR_INPL(mgr, PCXHR_PLX_MBOX0) & PCXHR_MBOX0_BOOT_HERE) { - /* no need to load the eeprom binary, but init the HI08 interface */ + /* no need to load the eeprom binary, + * but init the HI08 interface + */ PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg | PCXHR_ICR_HI08_INIT); msleep(PCXHR_WAIT_DEFAULT); PCXHR_OUTPB(mgr, PCXHR_DSP_ICR, reg); @@ -429,8 +455,10 @@ int pcxhr_load_dsp_binary(struct pcxhr_mgr *mgr, const struct firmware *dsp) if (err) return err; /* wait for chk bit */ - return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_CHK, - PCXHR_ISR_HI08_CHK, PCXHR_TIMEOUT_DSP, &dummy); + return pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_CHK, + PCXHR_ISR_HI08_CHK, + PCXHR_TIMEOUT_DSP, &dummy); } @@ -443,8 +471,8 @@ struct pcxhr_cmd_info { /* RMH status type */ enum { RMH_SSIZE_FIXED = 0, /* status size fix (st_length = 0..x) */ - RMH_SSIZE_ARG = 1, /* status size given in the LSB byte (used with st_length = 1) */ - RMH_SSIZE_MASK = 2, /* status size given in bitmask (used with st_length = 1) */ + RMH_SSIZE_ARG = 1, /* status size given in the LSB byte */ + RMH_SSIZE_MASK = 2, /* status size given in bitmask */ }; /* @@ -474,7 +502,7 @@ static struct pcxhr_cmd_info pcxhr_dsp_cmds[] = { [CMD_UPDATE_R_BUFFERS] = { 0x840000, 0, RMH_SSIZE_FIXED }, [CMD_FORMAT_STREAM_OUT] = { 0x860000, 0, RMH_SSIZE_FIXED }, [CMD_FORMAT_STREAM_IN] = { 0x870000, 0, RMH_SSIZE_FIXED }, -[CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, /* stat_len = nb_streams * 2 */ +[CMD_STREAM_SAMPLE_COUNT] = { 0x902000, 2, RMH_SSIZE_FIXED }, [CMD_AUDIO_LEVEL_ADJUST] = { 0xc22000, 0, RMH_SSIZE_FIXED }, }; @@ -524,10 +552,13 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) for (i = 0; i < rmh->stat_len; i++) { /* wait for receiver full */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF, - PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_RXDF, + PCXHR_ISR_HI08_RXDF, + PCXHR_TIMEOUT_DSP, ®); if (err) { - snd_printk(KERN_ERR "ERROR RMH stat: ISR:RXDF=1 (ISR = %x; i=%d )\n", + snd_printk(KERN_ERR "ERROR RMH stat: " + "ISR:RXDF=1 (ISR = %x; i=%d )\n", reg, i); return err; } @@ -537,10 +568,10 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL); /* need to update rmh->stat_len on the fly ?? */ - if (i==0) { + if (!i) { if (rmh->dsp_stat != RMH_SSIZE_FIXED) { if (rmh->dsp_stat == RMH_SSIZE_ARG) { - rmh->stat_len = (u16)(data & 0x0000ff) + 1; + rmh->stat_len = (data & 0x0000ff) + 1; data &= 0xffff00; } else { /* rmh->dsp_stat == RMH_SSIZE_MASK */ @@ -562,7 +593,8 @@ static int pcxhr_read_rmh_status(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) rmh->stat[i] = data; } if (rmh->stat_len > max_stat_len) { - snd_printdd("PCXHR : rmh->stat_len=%x too big\n", rmh->stat_len); + snd_printdd("PCXHR : rmh->stat_len=%x too big\n", + rmh->stat_len); rmh->stat_len = max_stat_len; } return 0; @@ -605,7 +637,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data &= 0xff7fff; /* MASK_1_WORD_COMMAND */ #ifdef CONFIG_SND_DEBUG_VERBOSE if (rmh->cmd_idx < CMD_LAST_INDEX) - snd_printdd("MSG cmd[0]=%x (%s)\n", data, cmd_names[rmh->cmd_idx]); + snd_printdd("MSG cmd[0]=%x (%s)\n", + data, cmd_names[rmh->cmd_idx]); #endif err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, @@ -619,8 +652,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) if (rmh->cmd_len > 1) { /* send length */ data = rmh->cmd_len - 1; - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_TRDY, - PCXHR_ISR_HI08_TRDY, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_TRDY, + PCXHR_ISR_HI08_TRDY, + PCXHR_TIMEOUT_DSP, ®); if (err) return err; PCXHR_OUTPB(mgr, PCXHR_DSP_TXH, (data>>16)&0xFF); @@ -653,8 +688,10 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) /* test status ISR */ if (reg & PCXHR_ISR_HI08_ERR) { /* ERROR, wait for receiver full */ - err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, PCXHR_ISR_HI08_RXDF, - PCXHR_ISR_HI08_RXDF, PCXHR_TIMEOUT_DSP, ®); + err = pcxhr_check_reg_bit(mgr, PCXHR_DSP_ISR, + PCXHR_ISR_HI08_RXDF, + PCXHR_ISR_HI08_RXDF, + PCXHR_TIMEOUT_DSP, ®); if (err) { snd_printk(KERN_ERR "ERROR RMH: ISR:RXDF=1 (ISR = %x)\n", reg); return err; @@ -663,7 +700,8 @@ static int pcxhr_send_msg_nolock(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) data = PCXHR_INPB(mgr, PCXHR_DSP_TXH) << 16; data |= PCXHR_INPB(mgr, PCXHR_DSP_TXM) << 8; data |= PCXHR_INPB(mgr, PCXHR_DSP_TXL); - snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", rmh->cmd_idx, data); + snd_printk(KERN_ERR "ERROR RMH(%d): 0x%x\n", + rmh->cmd_idx, data); err = -EINVAL; } else { /* read the response data */ @@ -732,8 +770,9 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh) static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr) { int start_mask = PCXHR_INPL(mgr, PCXHR_PLX_MBOX2); - /* least segnificant 12 bits are the pipe states for the playback audios */ - /* next 12 bits are the pipe states for the capture audios + /* least segnificant 12 bits are the pipe states + * for the playback audios + * next 12 bits are the pipe states for the capture audios * (PCXHR_PIPE_STATE_CAPTURE_OFFSET) */ start_mask &= 0xffffff; @@ -744,7 +783,8 @@ static inline int pcxhr_pipes_running(struct pcxhr_mgr *mgr) #define PCXHR_PIPE_STATE_CAPTURE_OFFSET 12 #define MAX_WAIT_FOR_DSP 20 -static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int *retry) +static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, + int audio_mask, int *retry) { struct pcxhr_rmh rmh; int err; @@ -760,17 +800,20 @@ static int pcxhr_prepair_pipe_start(struct pcxhr_mgr *mgr, int audio_mask, int * } else { /* can start capture pipe */ pcxhr_set_pipe_cmd_params(&rmh, 1, audio - - PCXHR_PIPE_STATE_CAPTURE_OFFSET, - 0, 0); + PCXHR_PIPE_STATE_CAPTURE_OFFSET, + 0, 0); } err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe start (CMD_CAN_START_PIPE) err=%x!\n", + "error pipe start " + "(CMD_CAN_START_PIPE) err=%x!\n", err); return err; } - /* if the pipe couldn't be prepaired for start, retry it later */ + /* if the pipe couldn't be prepaired for start, + * retry it later + */ if (rmh.stat[0] == 0) *retry |= (1<<audio); } @@ -795,14 +838,14 @@ static int pcxhr_stop_pipes(struct pcxhr_mgr *mgr, int audio_mask) } else { /* stop capture pipe */ pcxhr_set_pipe_cmd_params(&rmh, 1, audio - - PCXHR_PIPE_STATE_CAPTURE_OFFSET, - 0, 0); + PCXHR_PIPE_STATE_CAPTURE_OFFSET, + 0, 0); } err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe stop (CMD_STOP_PIPE) err=%x!\n", - err); + "error pipe stop " + "(CMD_STOP_PIPE) err=%x!\n", err); return err; } } @@ -822,15 +865,16 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) if (audio_mask & 1) { pcxhr_init_rmh(&rmh, CMD_CONF_PIPE); if (audio < PCXHR_PIPE_STATE_CAPTURE_OFFSET) - pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, 1 << audio); + pcxhr_set_pipe_cmd_params(&rmh, 0, 0, 0, + 1 << audio); else pcxhr_set_pipe_cmd_params(&rmh, 1, 0, 0, 1 << (audio - PCXHR_PIPE_STATE_CAPTURE_OFFSET)); err = pcxhr_send_msg(mgr, &rmh); if (err) { snd_printk(KERN_ERR - "error pipe start (CMD_CONF_PIPE) err=%x!\n", - err); + "error pipe start " + "(CMD_CONF_PIPE) err=%x!\n", err); return err; } } @@ -841,7 +885,9 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) pcxhr_init_rmh(&rmh, CMD_SEND_IRQA); err = pcxhr_send_msg(mgr, &rmh); if (err) { - snd_printk(KERN_ERR "error pipe start (CMD_SEND_IRQA) err=%x!\n", err ); + snd_printk(KERN_ERR + "error pipe start (CMD_SEND_IRQA) err=%x!\n", + err); return err; } return 0; @@ -849,7 +895,8 @@ static int pcxhr_toggle_pipes(struct pcxhr_mgr *mgr, int audio_mask) -int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_mask, int start) +int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, + int capture_mask, int start) { int state, i, err; int audio_mask; @@ -858,21 +905,23 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif - audio_mask = (playback_mask | (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); + audio_mask = (playback_mask | + (capture_mask << PCXHR_PIPE_STATE_CAPTURE_OFFSET)); /* current pipe state (playback + record) */ state = pcxhr_pipes_running(mgr); snd_printdd("pcxhr_set_pipe_state %s (mask %x current %x)\n", start ? "START" : "STOP", audio_mask, state); if (start) { - audio_mask &= ~state; /* start only pipes that are not yet started */ + /* start only pipes that are not yet started */ + audio_mask &= ~state; state = audio_mask; for (i = 0; i < MAX_WAIT_FOR_DSP; i++) { err = pcxhr_prepair_pipe_start(mgr, state, &state); if (err) return err; if (state == 0) - break; /* success, all pipes prepaired for start */ - mdelay(1); /* otherwise wait 1 millisecond and retry */ + break; /* success, all pipes prepaired */ + mdelay(1); /* wait 1 millisecond and retry */ } } else { audio_mask &= state; /* stop only pipes that are started */ @@ -891,7 +940,7 @@ int pcxhr_set_pipe_state(struct pcxhr_mgr *mgr, int playback_mask, int capture_m if ((state & audio_mask) == (start ? audio_mask : 0)) break; if (++i >= MAX_WAIT_FOR_DSP * 100) { - snd_printk(KERN_ERR "error pipe start/stop (ED_NO_RESPONSE_AT_IRQA)\n"); + snd_printk(KERN_ERR "error pipe start/stop\n"); return -EBUSY; } udelay(10); /* wait 10 microseconds */ @@ -918,7 +967,8 @@ int pcxhr_write_io_num_reg_cont(struct pcxhr_mgr *mgr, unsigned int mask, spin_lock_irqsave(&mgr->msg_lock, flags); if ((mgr->io_num_reg_cont & mask) == value) { - snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", mask, value); + snd_printdd("IO_NUM_REG_CONT mask %x already is set to %x\n", + mask, value); if (changed) *changed = 0; spin_unlock_irqrestore(&mgr->msg_lock, flags); @@ -971,7 +1021,8 @@ static int pcxhr_handle_async_err(struct pcxhr_mgr *mgr, u32 err, err = ((err >> 12) & 0xfff); if (!err) return 0; - snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", err_src_name[err_src], + snd_printdd("CMD_ASYNC : Error %s %s Pipe %d err=%x\n", + err_src_name[err_src], is_capture ? "Record" : "Play", pipe, err); if (err == 0xe01) mgr->async_err_stream_xrun++; @@ -996,6 +1047,13 @@ void pcxhr_msg_tasklet(unsigned long arg) snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n"); if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY) snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n"); + if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) { + /* clear events FREQ_CHANGE and TIME_CODE */ + pcxhr_init_rmh(prmh, CMD_TEST_IT); + err = pcxhr_send_msg(mgr, prmh); + snd_printdd("CMD_TEST_IT : err=%x, stat=%x\n", + err, prmh->stat[0]); + } if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) { snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n"); @@ -1005,18 +1063,22 @@ void pcxhr_msg_tasklet(unsigned long arg) prmh->stat_len = PCXHR_SIZE_MAX_LONG_STATUS; err = pcxhr_send_msg(mgr, prmh); if (err) - snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", err); + snd_printk(KERN_ERR "ERROR pcxhr_msg_tasklet=%x;\n", + err); i = 1; while (i < prmh->stat_len) { - int nb_audio = (prmh->stat[i] >> FIELD_SIZE) & MASK_FIRST_FIELD; - int nb_stream = (prmh->stat[i] >> (2*FIELD_SIZE)) & MASK_FIRST_FIELD; + int nb_audio = ((prmh->stat[i] >> FIELD_SIZE) & + MASK_FIRST_FIELD); + int nb_stream = ((prmh->stat[i] >> (2*FIELD_SIZE)) & + MASK_FIRST_FIELD); int pipe = prmh->stat[i] & MASK_FIRST_FIELD; int is_capture = prmh->stat[i] & 0x400000; u32 err2; if (prmh->stat[i] & 0x800000) { /* if BIT_END */ snd_printdd("TASKLET : End%sPipe %d\n", - is_capture ? "Record" : "Play", pipe); + is_capture ? "Record" : "Play", + pipe); } i++; err2 = prmh->stat[i] ? prmh->stat[i] : prmh->stat[i+1]; @@ -1062,7 +1124,7 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, pcxhr_init_rmh(&rmh, CMD_STREAM_SAMPLE_COUNT); pcxhr_set_pipe_cmd_params(&rmh, stream->pipe->is_capture, stream->pipe->first_audio, 0, stream_mask); - /* rmh.stat_len = 2; */ /* 2 resp data for each stream of the pipe */ + /* rmh.stat_len = 2; */ /* 2 resp data for each stream of the pipe */ err = pcxhr_send_msg(mgr, &rmh); if (err) @@ -1072,18 +1134,21 @@ static u_int64_t pcxhr_stream_read_position(struct pcxhr_mgr *mgr, hw_sample_count += (u_int64_t)rmh.stat[1]; snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n", - stream->pipe->is_capture ? 'C':'P', stream->substream->number, + stream->pipe->is_capture ? 'C' : 'P', + stream->substream->number, (long unsigned int)hw_sample_count, (long unsigned int)(stream->timer_abs_periods + - stream->timer_period_frag + PCXHR_GRANULARITY)); - + stream->timer_period_frag + + mgr->granularity)); return hw_sample_count; } static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, - struct pcxhr_stream *stream, int samples_to_add) + struct pcxhr_stream *stream, + int samples_to_add) { - if (stream->substream && (stream->status == PCXHR_STREAM_STATUS_RUNNING)) { + if (stream->substream && + (stream->status == PCXHR_STREAM_STATUS_RUNNING)) { u_int64_t new_sample_count; int elapsed = 0; int hardware_read = 0; @@ -1092,20 +1157,22 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, if (samples_to_add < 0) { stream->timer_is_synced = 0; /* add default if no hardware_read possible */ - samples_to_add = PCXHR_GRANULARITY; + samples_to_add = mgr->granularity; } if (!stream->timer_is_synced) { - if (stream->timer_abs_periods != 0 || - stream->timer_period_frag + PCXHR_GRANULARITY >= - runtime->period_size) { - new_sample_count = pcxhr_stream_read_position(mgr, stream); + if ((stream->timer_abs_periods != 0) || + ((stream->timer_period_frag + samples_to_add) >= + runtime->period_size)) { + new_sample_count = + pcxhr_stream_read_position(mgr, stream); hardware_read = 1; - if (new_sample_count >= PCXHR_GRANULARITY_MIN) { - /* sub security offset because of jitter and - * finer granularity of dsp time (MBOX4) + if (new_sample_count >= mgr->granularity) { + /* sub security offset because of + * jitter and finer granularity of + * dsp time (MBOX4) */ - new_sample_count -= PCXHR_GRANULARITY_MIN; + new_sample_count -= mgr->granularity; stream->timer_is_synced = 1; } } @@ -1128,12 +1195,15 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, stream->timer_buf_periods = 0; stream->timer_abs_periods = new_elapse_pos; } - if (new_sample_count >= stream->timer_abs_periods) - stream->timer_period_frag = (u_int32_t)(new_sample_count - - stream->timer_abs_periods); - else - snd_printk(KERN_ERR "ERROR new_sample_count too small ??? %lx\n", + if (new_sample_count >= stream->timer_abs_periods) { + stream->timer_period_frag = + (u_int32_t)(new_sample_count - + stream->timer_abs_periods); + } else { + snd_printk(KERN_ERR + "ERROR new_sample_count too small ??? %ld\n", (long unsigned int)new_sample_count); + } if (elapsed) { spin_unlock(&mgr->lock); @@ -1143,7 +1213,6 @@ static void pcxhr_update_timer_pos(struct pcxhr_mgr *mgr, } } - irqreturn_t pcxhr_interrupt(int irq, void *dev_id) { struct pcxhr_mgr *mgr = dev_id; @@ -1156,7 +1225,8 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) reg = PCXHR_INPL(mgr, PCXHR_PLX_IRQCS); if (! (reg & PCXHR_IRQCS_ACTIVE_PCIDB)) { spin_unlock(&mgr->lock); - return IRQ_NONE; /* this device did not cause the interrupt */ + /* this device did not cause the interrupt */ + return IRQ_NONE; } /* clear interrupt */ @@ -1167,10 +1237,12 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if (reg & PCXHR_IRQ_TIMER) { int timer_toggle = reg & PCXHR_IRQ_TIMER; /* is a 24 bit counter */ - int dsp_time_new = PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK; + int dsp_time_new = + PCXHR_INPL(mgr, PCXHR_PLX_MBOX4) & PCXHR_DSP_TIME_MASK; int dsp_time_diff = dsp_time_new - mgr->dsp_time_last; - if (dsp_time_diff < 0 && mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID) { + if ((dsp_time_diff < 0) && + (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) { snd_printdd("ERROR DSP TIME old(%d) new(%d) -> " "resynchronize all streams\n", mgr->dsp_time_last, dsp_time_new); @@ -1178,40 +1250,49 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) } #ifdef CONFIG_SND_DEBUG_VERBOSE if (dsp_time_diff == 0) - snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", dsp_time_new); - else if (dsp_time_diff >= (2*PCXHR_GRANULARITY)) + snd_printdd("ERROR DSP TIME NO DIFF time(%d)\n", + dsp_time_new); + else if (dsp_time_diff >= (2*mgr->granularity)) snd_printdd("ERROR DSP TIME TOO BIG old(%d) add(%d)\n", - mgr->dsp_time_last, dsp_time_new - mgr->dsp_time_last); + mgr->dsp_time_last, + dsp_time_new - mgr->dsp_time_last); + else if (dsp_time_diff % mgr->granularity) + snd_printdd("ERROR DSP TIME increased by %d\n", + dsp_time_diff); #endif mgr->dsp_time_last = dsp_time_new; - if (timer_toggle == mgr->timer_toggle) + if (timer_toggle == mgr->timer_toggle) { snd_printdd("ERROR TIMER TOGGLE\n"); + mgr->dsp_time_err++; + } mgr->timer_toggle = timer_toggle; reg &= ~PCXHR_IRQ_TIMER; for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; for (j = 0; j < chip->nb_streams_capt; j++) - pcxhr_update_timer_pos(mgr, &chip->capture_stream[j], - dsp_time_diff); + pcxhr_update_timer_pos(mgr, + &chip->capture_stream[j], + dsp_time_diff); } for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; for (j = 0; j < chip->nb_streams_play; j++) - pcxhr_update_timer_pos(mgr, &chip->playback_stream[j], - dsp_time_diff); + pcxhr_update_timer_pos(mgr, + &chip->playback_stream[j], + dsp_time_diff); } } /* other irq's handled in the tasklet */ if (reg & PCXHR_IRQ_MASK) { - - /* as we didn't request any notifications, some kind of xrun error - * will probably occured - */ - /* better resynchronize all streams next interrupt : */ - mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; - + if (reg & PCXHR_IRQ_ASYNC) { + /* as we didn't request any async notifications, + * some kind of xrun error will probably occured + */ + /* better resynchronize all streams next interrupt : */ + mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; + } mgr->src_it_dsp = reg; tasklet_hi_schedule(&mgr->msg_taskq); } diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index d9a4ab609875..bbbd66d13a64 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -65,7 +65,7 @@ enum { CMD_RESYNC_AUDIO_INPUTS, /* cmd_len = 1 stat_len = 0 */ CMD_GET_DSP_RESOURCES, /* cmd_len = 1 stat_len = 4 */ CMD_SET_TIMER_INTERRUPT, /* cmd_len = 1 stat_len = 0 */ - CMD_RES_PIPE, /* cmd_len = 2 stat_len = 0 */ + CMD_RES_PIPE, /* cmd_len >=2 stat_len = 0 */ CMD_FREE_PIPE, /* cmd_len = 1 stat_len = 0 */ CMD_CONF_PIPE, /* cmd_len = 2 stat_len = 0 */ CMD_STOP_PIPE, /* cmd_len = 1 stat_len = 0 */ @@ -96,6 +96,8 @@ void pcxhr_init_rmh(struct pcxhr_rmh *rmh, int cmd); void pcxhr_set_pipe_cmd_params(struct pcxhr_rmh* rmh, int capture, unsigned int param1, unsigned int param2, unsigned int param3); +#define DSP_EXT_CMD_SET(x) (x->dsp_version > 0x012800) + /* send the rmh */ @@ -110,6 +112,7 @@ int pcxhr_send_msg(struct pcxhr_mgr *mgr, struct pcxhr_rmh *rmh); #define IO_NUM_REG_STATUS 5 #define IO_NUM_REG_CUER 10 #define IO_NUM_UER_CHIP_REG 11 +#define IO_NUM_REG_CONFIG_SRC 12 #define IO_NUM_REG_OUT_ANA_LEVEL 20 #define IO_NUM_REG_IN_ANA_LEVEL 21 diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 96640d9c227d..ea50018d785b 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -31,6 +31,7 @@ #include "pcxhr_mixer.h" #include "pcxhr_hwdep.h" #include "pcxhr_core.h" +#include "pcxhr_mix22.h" #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE) @@ -40,10 +41,10 @@ #endif +static int pcxhr_sub_init(struct pcxhr_mgr *mgr); /* * get basic information and init pcxhr card */ - static int pcxhr_init_board(struct pcxhr_mgr *mgr) { int err; @@ -68,7 +69,7 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) if ((rmh.stat[0] & MASK_FIRST_FIELD) != mgr->playback_chips * 2) return -EINVAL; /* test 8 or 2 phys in */ - if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) != + if (((rmh.stat[0] >> (2 * FIELD_SIZE)) & MASK_FIRST_FIELD) < mgr->capture_chips * 2) return -EINVAL; /* test max nb substream per board */ @@ -77,20 +78,34 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) /* test max nb substream per pipe */ if (((rmh.stat[1] >> 7) & 0x5F) < PCXHR_PLAYBACK_STREAMS) return -EINVAL; + snd_printdd("supported formats : playback=%x capture=%x\n", + rmh.stat[2], rmh.stat[3]); pcxhr_init_rmh(&rmh, CMD_VERSION); /* firmware num for DSP */ rmh.cmd[0] |= mgr->firmware_num; /* transfer granularity in samples (should be multiple of 48) */ - rmh.cmd[1] = (1<<23) + PCXHR_GRANULARITY; + rmh.cmd[1] = (1<<23) + mgr->granularity; rmh.cmd_len = 2; err = pcxhr_send_msg(mgr, &rmh); if (err) return err; - snd_printdd("PCXHR DSP version is %d.%d.%d\n", - (rmh.stat[0]>>16)&0xff, (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff); + snd_printdd("PCXHR DSP version is %d.%d.%d\n", (rmh.stat[0]>>16)&0xff, + (rmh.stat[0]>>8)&0xff, rmh.stat[0]&0xff); mgr->dsp_version = rmh.stat[0]; + if (mgr->is_hr_stereo) + err = hr222_sub_init(mgr); + else + err = pcxhr_sub_init(mgr); + return err; +} + +static int pcxhr_sub_init(struct pcxhr_mgr *mgr) +{ + int err; + struct pcxhr_rmh rmh; + /* get options */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_REG_STATUS; @@ -100,20 +115,22 @@ static int pcxhr_init_board(struct pcxhr_mgr *mgr) if (err) return err; - if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == REG_STATUS_OPT_ANALOG_BOARD) - mgr->board_has_analog = 1; /* analog addon board available */ - else - /* analog addon board not available -> no support for instance */ - return -EINVAL; + if ((rmh.stat[1] & REG_STATUS_OPT_DAUGHTER_MASK) == + REG_STATUS_OPT_ANALOG_BOARD) + mgr->board_has_analog = 1; /* analog addon board found */ /* unmute inputs */ err = pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, REG_CONT_UNMUTE_INPUTS, NULL); if (err) return err; - /* unmute outputs */ - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); /* a write to IO_NUM_REG_MUTE_OUT mutes! */ + /* unmute outputs (a write to IO_NUM_REG_MUTE_OUT mutes!) */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; + if (DSP_EXT_CMD_SET(mgr)) { + rmh.cmd[1] = 1; /* unmute digital plugs */ + rmh.cmd_len = 2; + } err = pcxhr_send_msg(mgr, &rmh); return err; } @@ -124,19 +141,25 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr) if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { /* mute outputs */ + if (!mgr->is_hr_stereo) { /* a read to IO_NUM_REG_MUTE_OUT register unmutes! */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); rmh.cmd[0] |= IO_NUM_REG_MUTE_OUT; pcxhr_send_msg(mgr, &rmh); /* mute inputs */ - pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, 0, NULL); + pcxhr_write_io_num_reg_cont(mgr, REG_CONT_UNMUTE_INPUTS, + 0, NULL); + } + /* stereo cards mute with reset of dsp */ } /* reset pcxhr dsp */ - if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX)) + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_EPRM_INDEX)) pcxhr_reset_dsp(mgr); /* reset second xilinx */ - if (mgr->dsp_loaded & ( 1 << PCXHR_FIRMWARE_XLX_COM_INDEX)) + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_XLX_COM_INDEX)) { pcxhr_reset_xilinx_com(mgr); + mgr->dsp_loaded = 1; + } return; } @@ -144,8 +167,9 @@ void pcxhr_reset_board(struct pcxhr_mgr *mgr) /* * allocate a playback/capture pipe (pcmp0/pcmc0) */ -static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe, - int is_capture, int pin) +static int pcxhr_dsp_allocate_pipe(struct pcxhr_mgr *mgr, + struct pcxhr_pipe *pipe, + int is_capture, int pin) { int stream_count, audio_count; int err; @@ -161,15 +185,23 @@ static int pcxhr_dsp_allocate_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pi stream_count = PCXHR_PLAYBACK_STREAMS; audio_count = 2; /* always stereo */ } - snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", pin, is_capture ? 'c' : 'p'); + snd_printdd("snd_add_ref_pipe pin(%d) pcm%c0\n", + pin, is_capture ? 'c' : 'p'); pipe->is_capture = is_capture; pipe->first_audio = pin; /* define pipe (P_PCM_ONLY_MASK (0x020000) is not necessary) */ pcxhr_init_rmh(&rmh, CMD_RES_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, audio_count, stream_count); + pcxhr_set_pipe_cmd_params(&rmh, is_capture, pin, + audio_count, stream_count); + rmh.cmd[1] |= 0x020000; /* add P_PCM_ONLY_MASK */ + if (DSP_EXT_CMD_SET(mgr)) { + /* add channel mask to command */ + rmh.cmd[rmh.cmd_len++] = (audio_count == 1) ? 0x01 : 0x03; + } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) { - snd_printk(KERN_ERR "error pipe allocation (CMD_RES_PIPE) err=%x!\n", err ); + snd_printk(KERN_ERR "error pipe allocation " + "(CMD_RES_PIPE) err=%x!\n", err); return err; } pipe->status = PCXHR_PIPE_DEFINED; @@ -199,10 +231,12 @@ static int pcxhr_dsp_free_pipe( struct pcxhr_mgr *mgr, struct pcxhr_pipe *pipe) snd_printk(KERN_ERR "error stopping pipe!\n"); /* release the pipe */ pcxhr_init_rmh(&rmh, CMD_FREE_PIPE); - pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, 0, 0); + pcxhr_set_pipe_cmd_params(&rmh, pipe->is_capture, pipe->first_audio, + 0, 0); err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - snd_printk(KERN_ERR "error pipe release (CMD_FREE_PIPE) err(%x)\n", err); + snd_printk(KERN_ERR "error pipe release " + "(CMD_FREE_PIPE) err(%x)\n", err); pipe->status = PCXHR_PIPE_UNDEFINED; return err; } @@ -248,15 +282,16 @@ static int pcxhr_start_pipes(struct pcxhr_mgr *mgr) for (i = 0; i < mgr->num_cards; i++) { chip = mgr->chip[i]; if (chip->nb_streams_play) - playback_mask |= (1 << chip->playback_pipe.first_audio); + playback_mask |= 1 << chip->playback_pipe.first_audio; for (j = 0; j < chip->nb_streams_capt; j++) - capture_mask |= (1 << chip->capture_pipe[j].first_audio); + capture_mask |= 1 << chip->capture_pipe[j].first_audio; } return pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); } -static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmware *dsp) +static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, + const struct firmware *dsp) { int err, card_index; @@ -330,22 +365,33 @@ static int pcxhr_dsp_load(struct pcxhr_mgr *mgr, int index, const struct firmwar int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) { - static char *fw_files[5] = { - "xi_1_882.dat", - "xc_1_882.dat", - "e321_512.e56", - "b321_512.b56", - "d321_512.d56" + static char *fw_files[][5] = { + [0] = { "xi_1_882.dat", "xc_1_882.dat", + "e321_512.e56", "b321_512.b56", "d321_512.d56" }, + [1] = { "xi_1_882.dat", "xc_882e.dat", + "e321_512.e56", "b882e.b56", "d321_512.d56" }, + [2] = { "xi_1_882.dat", "xc_1222.dat", + "e321_512.e56", "b1222.b56", "d1222.d56" }, + [3] = { "xi_1_882.dat", "xc_1222e.dat", + "e321_512.e56", "b1222e.b56", "d1222.d56" }, + [4] = { NULL, "x1_222hr.dat", + "e924.e56", "b924.b56", "l_1_222.d56" }, + [5] = { NULL, "x1_924hr.dat", + "e924.e56", "b924.b56", "l_1_222.d56" }, }; char path[32]; const struct firmware *fw_entry; int i, err; + int fw_set = mgr->fw_file_set; - for (i = 0; i < ARRAY_SIZE(fw_files); i++) { - sprintf(path, "pcxhr/%s", fw_files[i]); + for (i = 0; i < 5; i++) { + if (!fw_files[fw_set][i]) + continue; + sprintf(path, "pcxhr/%s", fw_files[fw_set][i]); if (request_firmware(&fw_entry, path, &mgr->pci->dev)) { - snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", path); + snd_printk(KERN_ERR "pcxhr: can't load firmware %s\n", + path); return -ENOENT; } /* fake hwdep dsp record */ @@ -360,10 +406,25 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) MODULE_FIRMWARE("pcxhr/xi_1_882.dat"); MODULE_FIRMWARE("pcxhr/xc_1_882.dat"); +MODULE_FIRMWARE("pcxhr/xc_882e.dat"); MODULE_FIRMWARE("pcxhr/e321_512.e56"); MODULE_FIRMWARE("pcxhr/b321_512.b56"); +MODULE_FIRMWARE("pcxhr/b882e.b56"); MODULE_FIRMWARE("pcxhr/d321_512.d56"); +MODULE_FIRMWARE("pcxhr/xc_1222.dat"); +MODULE_FIRMWARE("pcxhr/xc_1222e.dat"); +MODULE_FIRMWARE("pcxhr/b1222.b56"); +MODULE_FIRMWARE("pcxhr/b1222e.b56"); +MODULE_FIRMWARE("pcxhr/d1222.d56"); + +MODULE_FIRMWARE("pcxhr/x1_222hr.dat"); +MODULE_FIRMWARE("pcxhr/x1_924hr.dat"); +MODULE_FIRMWARE("pcxhr/e924.e56"); +MODULE_FIRMWARE("pcxhr/b924.b56"); +MODULE_FIRMWARE("pcxhr/l_1_222.d56"); + + #else /* old style firmware loading */ /* pcxhr hwdep interface id string */ @@ -373,7 +434,8 @@ MODULE_FIRMWARE("pcxhr/d321_512.d56"); static int pcxhr_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { - strcpy(info->id, "pcxhr"); + struct pcxhr_mgr *mgr = hw->private_data; + sprintf(info->id, "pcxhr%d", mgr->fw_file_set); info->num_dsps = PCXHR_FIRMWARE_FILES_MAX_INDEX; if (hw->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) @@ -393,8 +455,8 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, fw.size = dsp->length; fw.data = vmalloc(fw.size); if (! fw.data) { - snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image (%lu bytes)\n", - (unsigned long)fw.size); + snd_printk(KERN_ERR "pcxhr: cannot allocate dsp image " + "(%lu bytes)\n", (unsigned long)fw.size); return -ENOMEM; } if (copy_from_user((void *)fw.data, dsp->image, dsp->length)) { @@ -424,8 +486,11 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) int err; struct snd_hwdep *hw; - /* only create hwdep interface for first cardX (see "index" module parameter)*/ - if ((err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw)) < 0) + /* only create hwdep interface for first cardX + * (see "index" module parameter) + */ + err = snd_hwdep_new(mgr->chip[0]->card, PCXHR_HWDEP_ID, 0, &hw); + if (err < 0) return err; hw->iface = SNDRV_HWDEP_IFACE_PCXHR; @@ -435,10 +500,13 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) hw->ops.dsp_status = pcxhr_hwdep_dsp_status; hw->ops.dsp_load = pcxhr_hwdep_dsp_load; hw->exclusive = 1; + /* stereo cards don't need fw_file_0 -> dsp_loaded = 1 */ + hw->dsp_loaded = mgr->is_hr_stereo ? 1 : 0; mgr->dsp_loaded = 0; sprintf(hw->name, PCXHR_HWDEP_ID); - if ((err = snd_card_register(mgr->chip[0]->card)) < 0) + err = snd_card_register(mgr->chip[0]->card); + if (err < 0) return err; return 0; } diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c new file mode 100644 index 000000000000..ff019126b672 --- /dev/null +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -0,0 +1,820 @@ +/* + * Driver for Digigram pcxhr compatible soundcards + * + * mixer interface for stereo cards + * + * Copyright (c) 2004 by Digigram <alsa@digigram.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/delay.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/asoundef.h> +#include "pcxhr.h" +#include "pcxhr_core.h" +#include "pcxhr_mix22.h" + + +/* registers used on the DSP and Xilinx (port 2) : HR stereo cards only */ +#define PCXHR_DSP_RESET 0x20 +#define PCXHR_XLX_CFG 0x24 +#define PCXHR_XLX_RUER 0x28 +#define PCXHR_XLX_DATA 0x2C +#define PCXHR_XLX_STATUS 0x30 +#define PCXHR_XLX_LOFREQ 0x34 +#define PCXHR_XLX_HIFREQ 0x38 +#define PCXHR_XLX_CSUER 0x3C +#define PCXHR_XLX_SELMIC 0x40 + +#define PCXHR_DSP 2 + +/* byte access only ! */ +#define PCXHR_INPB(mgr, x) inb((mgr)->port[PCXHR_DSP] + (x)) +#define PCXHR_OUTPB(mgr, x, data) outb((data), (mgr)->port[PCXHR_DSP] + (x)) + + +/* values for PCHR_DSP_RESET register */ +#define PCXHR_DSP_RESET_DSP 0x01 +#define PCXHR_DSP_RESET_MUTE 0x02 +#define PCXHR_DSP_RESET_CODEC 0x08 + +/* values for PCHR_XLX_CFG register */ +#define PCXHR_CFG_SYNCDSP_MASK 0x80 +#define PCXHR_CFG_DEPENDENCY_MASK 0x60 +#define PCXHR_CFG_INDEPENDANT_SEL 0x00 +#define PCXHR_CFG_MASTER_SEL 0x40 +#define PCXHR_CFG_SLAVE_SEL 0x20 +#define PCXHR_CFG_DATA_UER1_SEL_MASK 0x10 /* 0 (UER0), 1(UER1) */ +#define PCXHR_CFG_DATAIN_SEL_MASK 0x08 /* 0 (ana), 1 (UER) */ +#define PCXHR_CFG_SRC_MASK 0x04 /* 0 (Bypass), 1 (SRC Actif) */ +#define PCXHR_CFG_CLOCK_UER1_SEL_MASK 0x02 /* 0 (UER0), 1(UER1) */ +#define PCXHR_CFG_CLOCKIN_SEL_MASK 0x01 /* 0 (internal), 1 (AES/EBU) */ + +/* values for PCHR_XLX_DATA register */ +#define PCXHR_DATA_CODEC 0x80 +#define AKM_POWER_CONTROL_CMD 0xA007 +#define AKM_RESET_ON_CMD 0xA100 +#define AKM_RESET_OFF_CMD 0xA103 +#define AKM_CLOCK_INF_55K_CMD 0xA240 +#define AKM_CLOCK_SUP_55K_CMD 0xA24D +#define AKM_MUTE_CMD 0xA38D +#define AKM_UNMUTE_CMD 0xA30D +#define AKM_LEFT_LEVEL_CMD 0xA600 +#define AKM_RIGHT_LEVEL_CMD 0xA700 + +/* values for PCHR_XLX_STATUS register - READ */ +#define PCXHR_STAT_SRC_LOCK 0x01 +#define PCXHR_STAT_LEVEL_IN 0x02 +#define PCXHR_STAT_MIC_CAPS 0x10 +/* values for PCHR_XLX_STATUS register - WRITE */ +#define PCXHR_STAT_FREQ_SYNC_MASK 0x01 +#define PCXHR_STAT_FREQ_UER1_MASK 0x02 +#define PCXHR_STAT_FREQ_SAVE_MASK 0x80 + +/* values for PCHR_XLX_CSUER register */ +#define PCXHR_SUER1_BIT_U_READ_MASK 0x80 +#define PCXHR_SUER1_BIT_C_READ_MASK 0x40 +#define PCXHR_SUER1_DATA_PRESENT_MASK 0x20 +#define PCXHR_SUER1_CLOCK_PRESENT_MASK 0x10 +#define PCXHR_SUER_BIT_U_READ_MASK 0x08 +#define PCXHR_SUER_BIT_C_READ_MASK 0x04 +#define PCXHR_SUER_DATA_PRESENT_MASK 0x02 +#define PCXHR_SUER_CLOCK_PRESENT_MASK 0x01 + +#define PCXHR_SUER_BIT_U_WRITE_MASK 0x02 +#define PCXHR_SUER_BIT_C_WRITE_MASK 0x01 + +/* values for PCXHR_XLX_SELMIC register - WRITE */ +#define PCXHR_SELMIC_PREAMPLI_OFFSET 2 +#define PCXHR_SELMIC_PREAMPLI_MASK 0x0C +#define PCXHR_SELMIC_PHANTOM_ALIM 0x80 + + +static const unsigned char g_hr222_p_level[] = { + 0x00, /* [000] -49.5 dB: AKM[000] = -1.#INF dB (mute) */ + 0x01, /* [001] -49.0 dB: AKM[001] = -48.131 dB (diff=0.86920 dB) */ + 0x01, /* [002] -48.5 dB: AKM[001] = -48.131 dB (diff=0.36920 dB) */ + 0x01, /* [003] -48.0 dB: AKM[001] = -48.131 dB (diff=0.13080 dB) */ + 0x01, /* [004] -47.5 dB: AKM[001] = -48.131 dB (diff=0.63080 dB) */ + 0x01, /* [005] -46.5 dB: AKM[001] = -48.131 dB (diff=1.63080 dB) */ + 0x01, /* [006] -47.0 dB: AKM[001] = -48.131 dB (diff=1.13080 dB) */ + 0x01, /* [007] -46.0 dB: AKM[001] = -48.131 dB (diff=2.13080 dB) */ + 0x01, /* [008] -45.5 dB: AKM[001] = -48.131 dB (diff=2.63080 dB) */ + 0x02, /* [009] -45.0 dB: AKM[002] = -42.110 dB (diff=2.88980 dB) */ + 0x02, /* [010] -44.5 dB: AKM[002] = -42.110 dB (diff=2.38980 dB) */ + 0x02, /* [011] -44.0 dB: AKM[002] = -42.110 dB (diff=1.88980 dB) */ + 0x02, /* [012] -43.5 dB: AKM[002] = -42.110 dB (diff=1.38980 dB) */ + 0x02, /* [013] -43.0 dB: AKM[002] = -42.110 dB (diff=0.88980 dB) */ + 0x02, /* [014] -42.5 dB: AKM[002] = -42.110 dB (diff=0.38980 dB) */ + 0x02, /* [015] -42.0 dB: AKM[002] = -42.110 dB (diff=0.11020 dB) */ + 0x02, /* [016] -41.5 dB: AKM[002] = -42.110 dB (diff=0.61020 dB) */ + 0x02, /* [017] -41.0 dB: AKM[002] = -42.110 dB (diff=1.11020 dB) */ + 0x02, /* [018] -40.5 dB: AKM[002] = -42.110 dB (diff=1.61020 dB) */ + 0x03, /* [019] -40.0 dB: AKM[003] = -38.588 dB (diff=1.41162 dB) */ + 0x03, /* [020] -39.5 dB: AKM[003] = -38.588 dB (diff=0.91162 dB) */ + 0x03, /* [021] -39.0 dB: AKM[003] = -38.588 dB (diff=0.41162 dB) */ + 0x03, /* [022] -38.5 dB: AKM[003] = -38.588 dB (diff=0.08838 dB) */ + 0x03, /* [023] -38.0 dB: AKM[003] = -38.588 dB (diff=0.58838 dB) */ + 0x03, /* [024] -37.5 dB: AKM[003] = -38.588 dB (diff=1.08838 dB) */ + 0x04, /* [025] -37.0 dB: AKM[004] = -36.090 dB (diff=0.91040 dB) */ + 0x04, /* [026] -36.5 dB: AKM[004] = -36.090 dB (diff=0.41040 dB) */ + 0x04, /* [027] -36.0 dB: AKM[004] = -36.090 dB (diff=0.08960 dB) */ + 0x04, /* [028] -35.5 dB: AKM[004] = -36.090 dB (diff=0.58960 dB) */ + 0x05, /* [029] -35.0 dB: AKM[005] = -34.151 dB (diff=0.84860 dB) */ + 0x05, /* [030] -34.5 dB: AKM[005] = -34.151 dB (diff=0.34860 dB) */ + 0x05, /* [031] -34.0 dB: AKM[005] = -34.151 dB (diff=0.15140 dB) */ + 0x05, /* [032] -33.5 dB: AKM[005] = -34.151 dB (diff=0.65140 dB) */ + 0x06, /* [033] -33.0 dB: AKM[006] = -32.568 dB (diff=0.43222 dB) */ + 0x06, /* [034] -32.5 dB: AKM[006] = -32.568 dB (diff=0.06778 dB) */ + 0x06, /* [035] -32.0 dB: AKM[006] = -32.568 dB (diff=0.56778 dB) */ + 0x07, /* [036] -31.5 dB: AKM[007] = -31.229 dB (diff=0.27116 dB) */ + 0x07, /* [037] -31.0 dB: AKM[007] = -31.229 dB (diff=0.22884 dB) */ + 0x08, /* [038] -30.5 dB: AKM[008] = -30.069 dB (diff=0.43100 dB) */ + 0x08, /* [039] -30.0 dB: AKM[008] = -30.069 dB (diff=0.06900 dB) */ + 0x09, /* [040] -29.5 dB: AKM[009] = -29.046 dB (diff=0.45405 dB) */ + 0x09, /* [041] -29.0 dB: AKM[009] = -29.046 dB (diff=0.04595 dB) */ + 0x0a, /* [042] -28.5 dB: AKM[010] = -28.131 dB (diff=0.36920 dB) */ + 0x0a, /* [043] -28.0 dB: AKM[010] = -28.131 dB (diff=0.13080 dB) */ + 0x0b, /* [044] -27.5 dB: AKM[011] = -27.303 dB (diff=0.19705 dB) */ + 0x0b, /* [045] -27.0 dB: AKM[011] = -27.303 dB (diff=0.30295 dB) */ + 0x0c, /* [046] -26.5 dB: AKM[012] = -26.547 dB (diff=0.04718 dB) */ + 0x0d, /* [047] -26.0 dB: AKM[013] = -25.852 dB (diff=0.14806 dB) */ + 0x0e, /* [048] -25.5 dB: AKM[014] = -25.208 dB (diff=0.29176 dB) */ + 0x0e, /* [049] -25.0 dB: AKM[014] = -25.208 dB (diff=0.20824 dB) */ + 0x0f, /* [050] -24.5 dB: AKM[015] = -24.609 dB (diff=0.10898 dB) */ + 0x10, /* [051] -24.0 dB: AKM[016] = -24.048 dB (diff=0.04840 dB) */ + 0x11, /* [052] -23.5 dB: AKM[017] = -23.522 dB (diff=0.02183 dB) */ + 0x12, /* [053] -23.0 dB: AKM[018] = -23.025 dB (diff=0.02535 dB) */ + 0x13, /* [054] -22.5 dB: AKM[019] = -22.556 dB (diff=0.05573 dB) */ + 0x14, /* [055] -22.0 dB: AKM[020] = -22.110 dB (diff=0.11020 dB) */ + 0x15, /* [056] -21.5 dB: AKM[021] = -21.686 dB (diff=0.18642 dB) */ + 0x17, /* [057] -21.0 dB: AKM[023] = -20.896 dB (diff=0.10375 dB) */ + 0x18, /* [058] -20.5 dB: AKM[024] = -20.527 dB (diff=0.02658 dB) */ + 0x1a, /* [059] -20.0 dB: AKM[026] = -19.831 dB (diff=0.16866 dB) */ + 0x1b, /* [060] -19.5 dB: AKM[027] = -19.504 dB (diff=0.00353 dB) */ + 0x1d, /* [061] -19.0 dB: AKM[029] = -18.883 dB (diff=0.11716 dB) */ + 0x1e, /* [062] -18.5 dB: AKM[030] = -18.588 dB (diff=0.08838 dB) */ + 0x20, /* [063] -18.0 dB: AKM[032] = -18.028 dB (diff=0.02780 dB) */ + 0x22, /* [064] -17.5 dB: AKM[034] = -17.501 dB (diff=0.00123 dB) */ + 0x24, /* [065] -17.0 dB: AKM[036] = -17.005 dB (diff=0.00475 dB) */ + 0x26, /* [066] -16.5 dB: AKM[038] = -16.535 dB (diff=0.03513 dB) */ + 0x28, /* [067] -16.0 dB: AKM[040] = -16.090 dB (diff=0.08960 dB) */ + 0x2b, /* [068] -15.5 dB: AKM[043] = -15.461 dB (diff=0.03857 dB) */ + 0x2d, /* [069] -15.0 dB: AKM[045] = -15.067 dB (diff=0.06655 dB) */ + 0x30, /* [070] -14.5 dB: AKM[048] = -14.506 dB (diff=0.00598 dB) */ + 0x33, /* [071] -14.0 dB: AKM[051] = -13.979 dB (diff=0.02060 dB) */ + 0x36, /* [072] -13.5 dB: AKM[054] = -13.483 dB (diff=0.01707 dB) */ + 0x39, /* [073] -13.0 dB: AKM[057] = -13.013 dB (diff=0.01331 dB) */ + 0x3c, /* [074] -12.5 dB: AKM[060] = -12.568 dB (diff=0.06778 dB) */ + 0x40, /* [075] -12.0 dB: AKM[064] = -12.007 dB (diff=0.00720 dB) */ + 0x44, /* [076] -11.5 dB: AKM[068] = -11.481 dB (diff=0.01937 dB) */ + 0x48, /* [077] -11.0 dB: AKM[072] = -10.984 dB (diff=0.01585 dB) */ + 0x4c, /* [078] -10.5 dB: AKM[076] = -10.515 dB (diff=0.01453 dB) */ + 0x51, /* [079] -10.0 dB: AKM[081] = -9.961 dB (diff=0.03890 dB) */ + 0x55, /* [080] -9.5 dB: AKM[085] = -9.542 dB (diff=0.04243 dB) */ + 0x5a, /* [081] -9.0 dB: AKM[090] = -9.046 dB (diff=0.04595 dB) */ + 0x60, /* [082] -8.5 dB: AKM[096] = -8.485 dB (diff=0.01462 dB) */ + 0x66, /* [083] -8.0 dB: AKM[102] = -7.959 dB (diff=0.04120 dB) */ + 0x6c, /* [084] -7.5 dB: AKM[108] = -7.462 dB (diff=0.03767 dB) */ + 0x72, /* [085] -7.0 dB: AKM[114] = -6.993 dB (diff=0.00729 dB) */ + 0x79, /* [086] -6.5 dB: AKM[121] = -6.475 dB (diff=0.02490 dB) */ + 0x80, /* [087] -6.0 dB: AKM[128] = -5.987 dB (diff=0.01340 dB) */ + 0x87, /* [088] -5.5 dB: AKM[135] = -5.524 dB (diff=0.02413 dB) */ + 0x8f, /* [089] -5.0 dB: AKM[143] = -5.024 dB (diff=0.02408 dB) */ + 0x98, /* [090] -4.5 dB: AKM[152] = -4.494 dB (diff=0.00607 dB) */ + 0xa1, /* [091] -4.0 dB: AKM[161] = -3.994 dB (diff=0.00571 dB) */ + 0xaa, /* [092] -3.5 dB: AKM[170] = -3.522 dB (diff=0.02183 dB) */ + 0xb5, /* [093] -3.0 dB: AKM[181] = -2.977 dB (diff=0.02277 dB) */ + 0xbf, /* [094] -2.5 dB: AKM[191] = -2.510 dB (diff=0.01014 dB) */ + 0xcb, /* [095] -2.0 dB: AKM[203] = -1.981 dB (diff=0.01912 dB) */ + 0xd7, /* [096] -1.5 dB: AKM[215] = -1.482 dB (diff=0.01797 dB) */ + 0xe3, /* [097] -1.0 dB: AKM[227] = -1.010 dB (diff=0.01029 dB) */ + 0xf1, /* [098] -0.5 dB: AKM[241] = -0.490 dB (diff=0.00954 dB) */ + 0xff, /* [099] +0.0 dB: AKM[255] = +0.000 dB (diff=0.00000 dB) */ +}; + + +static void hr222_config_akm(struct pcxhr_mgr *mgr, unsigned short data) +{ + unsigned short mask = 0x8000; + /* activate access to codec registers */ + PCXHR_INPB(mgr, PCXHR_XLX_HIFREQ); + + while (mask) { + PCXHR_OUTPB(mgr, PCXHR_XLX_DATA, + data & mask ? PCXHR_DATA_CODEC : 0); + mask >>= 1; + } + /* termiate access to codec registers */ + PCXHR_INPB(mgr, PCXHR_XLX_RUER); +} + + +static int hr222_set_hw_playback_level(struct pcxhr_mgr *mgr, + int idx, int level) +{ + unsigned short cmd; + if (idx > 1 || + level < 0 || + level >= ARRAY_SIZE(g_hr222_p_level)) + return -EINVAL; + + if (idx == 0) + cmd = AKM_LEFT_LEVEL_CMD; + else + cmd = AKM_RIGHT_LEVEL_CMD; + + /* conversion from PmBoardCodedLevel to AKM nonlinear programming */ + cmd += g_hr222_p_level[level]; + + hr222_config_akm(mgr, cmd); + return 0; +} + + +static int hr222_set_hw_capture_level(struct pcxhr_mgr *mgr, + int level_l, int level_r, int level_mic) +{ + /* program all input levels at the same time */ + unsigned int data; + int i; + + if (!mgr->capture_chips) + return -EINVAL; /* no PCX22 */ + + data = ((level_mic & 0xff) << 24); /* micro is mono, but apply */ + data |= ((level_mic & 0xff) << 16); /* level on both channels */ + data |= ((level_r & 0xff) << 8); /* line input right channel */ + data |= (level_l & 0xff); /* line input left channel */ + + PCXHR_INPB(mgr, PCXHR_XLX_DATA); /* activate input codec */ + /* send 32 bits (4 x 8 bits) */ + for (i = 0; i < 32; i++, data <<= 1) { + PCXHR_OUTPB(mgr, PCXHR_XLX_DATA, + (data & 0x80000000) ? PCXHR_DATA_CODEC : 0); + } + PCXHR_INPB(mgr, PCXHR_XLX_RUER); /* close input level codec */ + return 0; +} + +static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level); + +int hr222_sub_init(struct pcxhr_mgr *mgr) +{ + unsigned char reg; + + mgr->board_has_analog = 1; /* analog always available */ + mgr->xlx_cfg = PCXHR_CFG_SYNCDSP_MASK; + + reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS); + if (reg & PCXHR_STAT_MIC_CAPS) + mgr->board_has_mic = 1; /* microphone available */ + snd_printdd("MIC input available = %d\n", mgr->board_has_mic); + + /* reset codec */ + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, + PCXHR_DSP_RESET_DSP); + msleep(5); + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, + PCXHR_DSP_RESET_DSP | + PCXHR_DSP_RESET_MUTE | + PCXHR_DSP_RESET_CODEC); + msleep(5); + + /* config AKM */ + hr222_config_akm(mgr, AKM_POWER_CONTROL_CMD); + hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD); + hr222_config_akm(mgr, AKM_UNMUTE_CMD); + hr222_config_akm(mgr, AKM_RESET_OFF_CMD); + + /* init micro boost */ + hr222_micro_boost(mgr, 0); + + return 0; +} + + +/* calc PLL register */ +/* TODO : there is a very similar fct in pcxhr.c */ +static int hr222_pll_freq_register(unsigned int freq, + unsigned int *pllreg, + unsigned int *realfreq) +{ + unsigned int reg; + + if (freq < 6900 || freq > 219000) + return -EINVAL; + reg = (28224000 * 2) / freq; + reg = (reg - 1) / 2; + if (reg < 0x100) + *pllreg = reg + 0xC00; + else if (reg < 0x200) + *pllreg = reg + 0x800; + else if (reg < 0x400) + *pllreg = reg & 0x1ff; + else if (reg < 0x800) { + *pllreg = ((reg >> 1) & 0x1ff) + 0x200; + reg &= ~1; + } else { + *pllreg = ((reg >> 2) & 0x1ff) + 0x400; + reg &= ~3; + } + if (realfreq) + *realfreq = (28224000 / (reg + 1)); + return 0; +} + +int hr222_sub_set_clock(struct pcxhr_mgr *mgr, + unsigned int rate, + int *changed) +{ + unsigned int speed, pllreg = 0; + int err; + unsigned realfreq = rate; + + switch (mgr->use_clock_type) { + case HR22_CLOCK_TYPE_INTERNAL: + err = hr222_pll_freq_register(rate, &pllreg, &realfreq); + if (err) + return err; + + mgr->xlx_cfg &= ~(PCXHR_CFG_CLOCKIN_SEL_MASK | + PCXHR_CFG_CLOCK_UER1_SEL_MASK); + break; + case HR22_CLOCK_TYPE_AES_SYNC: + mgr->xlx_cfg |= PCXHR_CFG_CLOCKIN_SEL_MASK; + mgr->xlx_cfg &= ~PCXHR_CFG_CLOCK_UER1_SEL_MASK; + break; + case HR22_CLOCK_TYPE_AES_1: + if (!mgr->board_has_aes1) + return -EINVAL; + + mgr->xlx_cfg |= (PCXHR_CFG_CLOCKIN_SEL_MASK | + PCXHR_CFG_CLOCK_UER1_SEL_MASK); + break; + default: + return -EINVAL; + } + hr222_config_akm(mgr, AKM_MUTE_CMD); + + if (mgr->use_clock_type == HR22_CLOCK_TYPE_INTERNAL) { + PCXHR_OUTPB(mgr, PCXHR_XLX_HIFREQ, pllreg >> 8); + PCXHR_OUTPB(mgr, PCXHR_XLX_LOFREQ, pllreg & 0xff); + } + + /* set clock source */ + PCXHR_OUTPB(mgr, PCXHR_XLX_CFG, mgr->xlx_cfg); + + /* codec speed modes */ + speed = rate < 55000 ? 0 : 1; + if (mgr->codec_speed != speed) { + mgr->codec_speed = speed; + if (speed == 0) + hr222_config_akm(mgr, AKM_CLOCK_INF_55K_CMD); + else + hr222_config_akm(mgr, AKM_CLOCK_SUP_55K_CMD); + } + + mgr->sample_rate_real = realfreq; + mgr->cur_clock_type = mgr->use_clock_type; + + if (changed) + *changed = 1; + + hr222_config_akm(mgr, AKM_UNMUTE_CMD); + + snd_printdd("set_clock to %dHz (realfreq=%d pllreg=%x)\n", + rate, realfreq, pllreg); + return 0; +} + +int hr222_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate) +{ + int rate, calc_rate = 0; + unsigned int ticks; + unsigned char mask, reg; + + if (clock_type == HR22_CLOCK_TYPE_AES_SYNC) { + + mask = (PCXHR_SUER_CLOCK_PRESENT_MASK | + PCXHR_SUER_DATA_PRESENT_MASK); + reg = PCXHR_STAT_FREQ_SYNC_MASK; + + } else if (clock_type == HR22_CLOCK_TYPE_AES_1 && mgr->board_has_aes1) { + + mask = (PCXHR_SUER1_CLOCK_PRESENT_MASK | + PCXHR_SUER1_DATA_PRESENT_MASK); + reg = PCXHR_STAT_FREQ_UER1_MASK; + + } else { + snd_printdd("get_external_clock : type %d not supported\n", + clock_type); + return -EINVAL; /* other clocks not supported */ + } + + if ((PCXHR_INPB(mgr, PCXHR_XLX_CSUER) & mask) != mask) { + snd_printdd("get_external_clock(%d) = 0 Hz\n", clock_type); + *sample_rate = 0; + return 0; /* no external clock locked */ + } + + PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* calculate freq */ + + /* save the measured clock frequency */ + reg |= PCXHR_STAT_FREQ_SAVE_MASK; + + if (mgr->last_reg_stat != reg) { + udelay(500); /* wait min 2 cycles of lowest freq (8000) */ + mgr->last_reg_stat = reg; + } + + PCXHR_OUTPB(mgr, PCXHR_XLX_STATUS, reg); /* save */ + + /* get the frequency */ + ticks = (unsigned int)PCXHR_INPB(mgr, PCXHR_XLX_CFG); + ticks = (ticks & 0x03) << 8; + ticks |= (unsigned int)PCXHR_INPB(mgr, PCXHR_DSP_RESET); + + if (ticks != 0) + calc_rate = 28224000 / ticks; + /* rounding */ + if (calc_rate > 184200) + rate = 192000; + else if (calc_rate > 152200) + rate = 176400; + else if (calc_rate > 112000) + rate = 128000; + else if (calc_rate > 92100) + rate = 96000; + else if (calc_rate > 76100) + rate = 88200; + else if (calc_rate > 56000) + rate = 64000; + else if (calc_rate > 46050) + rate = 48000; + else if (calc_rate > 38050) + rate = 44100; + else if (calc_rate > 28000) + rate = 32000; + else if (calc_rate > 23025) + rate = 24000; + else if (calc_rate > 19025) + rate = 22050; + else if (calc_rate > 14000) + rate = 16000; + else if (calc_rate > 11512) + rate = 12000; + else if (calc_rate > 9512) + rate = 11025; + else if (calc_rate > 7000) + rate = 8000; + else + rate = 0; + + snd_printdd("External clock is at %d Hz (measured %d Hz)\n", + rate, calc_rate); + *sample_rate = rate; + return 0; +} + + +int hr222_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, int channel) +{ + snd_printdd("hr222_update_analog_audio_level(%s chan=%d)\n", + is_capture ? "capture" : "playback", channel); + if (is_capture) { + int level_l, level_r, level_mic; + /* we have to update all levels */ + if (chip->analog_capture_active) { + level_l = chip->analog_capture_volume[0]; + level_r = chip->analog_capture_volume[1]; + } else { + level_l = HR222_LINE_CAPTURE_LEVEL_MIN; + level_r = HR222_LINE_CAPTURE_LEVEL_MIN; + } + if (chip->mic_active) + level_mic = chip->mic_volume; + else + level_mic = HR222_MICRO_CAPTURE_LEVEL_MIN; + return hr222_set_hw_capture_level(chip->mgr, + level_l, level_r, level_mic); + } else { + int vol; + if (chip->analog_playback_active[channel]) + vol = chip->analog_playback_volume[channel]; + else + vol = HR222_LINE_PLAYBACK_LEVEL_MIN; + return hr222_set_hw_playback_level(chip->mgr, channel, vol); + } +} + + +/*texts[5] = {"Line", "Digital", "Digi+SRC", "Mic", "Line+Mic"}*/ +#define SOURCE_LINE 0 +#define SOURCE_DIGITAL 1 +#define SOURCE_DIGISRC 2 +#define SOURCE_MIC 3 +#define SOURCE_LINEMIC 4 + +int hr222_set_audio_source(struct snd_pcxhr *chip) +{ + int digital = 0; + /* default analog source */ + chip->mgr->xlx_cfg &= ~(PCXHR_CFG_SRC_MASK | + PCXHR_CFG_DATAIN_SEL_MASK | + PCXHR_CFG_DATA_UER1_SEL_MASK); + + if (chip->audio_capture_source == SOURCE_DIGISRC) { + chip->mgr->xlx_cfg |= PCXHR_CFG_SRC_MASK; + digital = 1; + } else { + if (chip->audio_capture_source == SOURCE_DIGITAL) + digital = 1; + } + if (digital) { + chip->mgr->xlx_cfg |= PCXHR_CFG_DATAIN_SEL_MASK; + if (chip->mgr->board_has_aes1) { + /* get data from the AES1 plug */ + chip->mgr->xlx_cfg |= PCXHR_CFG_DATA_UER1_SEL_MASK; + } + /* chip->mic_active = 0; */ + /* chip->analog_capture_active = 0; */ + } else { + int update_lvl = 0; + chip->analog_capture_active = 0; + chip->mic_active = 0; + if (chip->audio_capture_source == SOURCE_LINE || + chip->audio_capture_source == SOURCE_LINEMIC) { + if (chip->analog_capture_active == 0) + update_lvl = 1; + chip->analog_capture_active = 1; + } + if (chip->audio_capture_source == SOURCE_MIC || + chip->audio_capture_source == SOURCE_LINEMIC) { + if (chip->mic_active == 0) + update_lvl = 1; + chip->mic_active = 1; + } + if (update_lvl) { + /* capture: update all 3 mutes/unmutes with one call */ + hr222_update_analog_audio_level(chip, 1, 0); + } + } + /* set the source infos (max 3 bits modified) */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CFG, chip->mgr->xlx_cfg); + return 0; +} + + +int hr222_iec958_capture_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char *aes_bits) +{ + unsigned char idx = (unsigned char)(aes_idx * 8); + unsigned char temp = 0; + unsigned char mask = chip->mgr->board_has_aes1 ? + PCXHR_SUER1_BIT_C_READ_MASK : PCXHR_SUER_BIT_C_READ_MASK; + int i; + for (i = 0; i < 8; i++) { + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx++); /* idx < 192 */ + temp <<= 1; + if (PCXHR_INPB(chip->mgr, PCXHR_XLX_CSUER) & mask) + temp |= 1; + } + snd_printdd("read iec958 AES %d byte %d = 0x%x\n", + chip->chip_idx, aes_idx, temp); + *aes_bits = temp; + return 0; +} + + +int hr222_iec958_update_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char aes_bits) +{ + int i; + unsigned char new_bits = aes_bits; + unsigned char old_bits = chip->aes_bits[aes_idx]; + unsigned char idx = (unsigned char)(aes_idx * 8); + for (i = 0; i < 8; i++) { + if ((old_bits & 0x01) != (new_bits & 0x01)) { + /* idx < 192 */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_RUER, idx); + /* write C and U bit */ + PCXHR_OUTPB(chip->mgr, PCXHR_XLX_CSUER, new_bits&0x01 ? + PCXHR_SUER_BIT_C_WRITE_MASK : 0); + } + idx++; + old_bits >>= 1; + new_bits >>= 1; + } + chip->aes_bits[aes_idx] = aes_bits; + return 0; +} + +static void hr222_micro_boost(struct pcxhr_mgr *mgr, int level) +{ + unsigned char boost_mask; + boost_mask = (unsigned char) (level << PCXHR_SELMIC_PREAMPLI_OFFSET); + if (boost_mask & (~PCXHR_SELMIC_PREAMPLI_MASK)) + return; /* only values form 0 to 3 accepted */ + + mgr->xlx_selmic &= ~PCXHR_SELMIC_PREAMPLI_MASK; + mgr->xlx_selmic |= boost_mask; + + PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic); + + snd_printdd("hr222_micro_boost : set %x\n", boost_mask); +} + +static void hr222_phantom_power(struct pcxhr_mgr *mgr, int power) +{ + if (power) + mgr->xlx_selmic |= PCXHR_SELMIC_PHANTOM_ALIM; + else + mgr->xlx_selmic &= ~PCXHR_SELMIC_PHANTOM_ALIM; + + PCXHR_OUTPB(mgr, PCXHR_XLX_SELMIC, mgr->xlx_selmic); + + snd_printdd("hr222_phantom_power : set %d\n", power); +} + + +/* mic level */ +static const DECLARE_TLV_DB_SCALE(db_scale_mic_hr222, -9850, 50, 650); + +static int hr222_mic_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = HR222_MICRO_CAPTURE_LEVEL_MIN; /* -98 dB */ + /* gains from 9 dB to 31.5 dB not recommended; use micboost instead */ + uinfo->value.integer.max = HR222_MICRO_CAPTURE_LEVEL_MAX; /* +7 dB */ + return 0; +} + +static int hr222_mic_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->mic_volume; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_mic_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + mutex_lock(&chip->mgr->mixer_mutex); + if (chip->mic_volume != ucontrol->value.integer.value[0]) { + changed = 1; + chip->mic_volume = ucontrol->value.integer.value[0]; + hr222_update_analog_audio_level(chip, 1, 0); + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_control_mic_level = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Mic Capture Volume", + .info = hr222_mic_vol_info, + .get = hr222_mic_vol_get, + .put = hr222_mic_vol_put, + .tlv = { .p = db_scale_mic_hr222 }, +}; + + +/* mic boost level */ +static const DECLARE_TLV_DB_SCALE(db_scale_micboost_hr222, 0, 1800, 5400); + +static int hr222_mic_boost_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; /* 0 dB */ + uinfo->value.integer.max = 3; /* 54 dB */ + return 0; +} + +static int hr222_mic_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->mic_boost; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_mic_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int changed = 0; + mutex_lock(&chip->mgr->mixer_mutex); + if (chip->mic_boost != ucontrol->value.integer.value[0]) { + changed = 1; + chip->mic_boost = ucontrol->value.integer.value[0]; + hr222_micro_boost(chip->mgr, chip->mic_boost); + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_control_mic_boost = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "MicBoost Capture Volume", + .info = hr222_mic_boost_info, + .get = hr222_mic_boost_get, + .put = hr222_mic_boost_put, + .tlv = { .p = db_scale_micboost_hr222 }, +}; + + +/******************* Phantom power switch *******************/ +#define hr222_phantom_power_info snd_ctl_boolean_mono_info + +static int hr222_phantom_power_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + mutex_lock(&chip->mgr->mixer_mutex); + ucontrol->value.integer.value[0] = chip->phantom_power; + mutex_unlock(&chip->mgr->mixer_mutex); + return 0; +} + +static int hr222_phantom_power_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + int power, changed = 0; + + mutex_lock(&chip->mgr->mixer_mutex); + power = !!ucontrol->value.integer.value[0]; + if (chip->phantom_power != power) { + hr222_phantom_power(chip->mgr, power); + chip->phantom_power = power; + changed = 1; + } + mutex_unlock(&chip->mgr->mixer_mutex); + return changed; +} + +static struct snd_kcontrol_new hr222_phantom_power_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phantom Power Switch", + .info = hr222_phantom_power_info, + .get = hr222_phantom_power_get, + .put = hr222_phantom_power_put, +}; + + +int hr222_add_mic_controls(struct snd_pcxhr *chip) +{ + int err; + if (!chip->mgr->board_has_mic) + return 0; + + /* controls */ + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_level, + chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_control_mic_boost, + chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&hr222_phantom_power_switch, + chip)); + return err; +} diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h new file mode 100644 index 000000000000..6b318b2f0100 --- /dev/null +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -0,0 +1,56 @@ +/* + * Driver for Digigram pcxhr compatible soundcards + * + * low level interface with interrupt ans message handling + * + * Copyright (c) 2004 by Digigram <alsa@digigram.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_PCXHR_MIX22_H +#define __SOUND_PCXHR_MIX22_H + +struct pcxhr_mgr; + +int hr222_sub_init(struct pcxhr_mgr *mgr); +int hr222_sub_set_clock(struct pcxhr_mgr *mgr, unsigned int rate, + int *changed); +int hr222_get_external_clock(struct pcxhr_mgr *mgr, + enum pcxhr_clock_type clock_type, + int *sample_rate); + +#define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ +#define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ +#define HR222_LINE_PLAYBACK_LEVEL_MAX 99 /* +24.0 dB */ + +#define HR222_LINE_CAPTURE_LEVEL_MIN 0 /* -111.5 dB */ +#define HR222_LINE_CAPTURE_ZERO_LEVEL 223 /* 0.0 dB */ +#define HR222_LINE_CAPTURE_LEVEL_MAX 255 /* +16 dB */ +#define HR222_MICRO_CAPTURE_LEVEL_MIN 0 /* -98.5 dB */ +#define HR222_MICRO_CAPTURE_LEVEL_MAX 210 /* +6.5 dB */ + +int hr222_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, + int channel); +int hr222_set_audio_source(struct snd_pcxhr *chip); +int hr222_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, + unsigned char *aes_bits); +int hr222_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, + unsigned char aes_bits); + +int hr222_add_mic_controls(struct snd_pcxhr *chip); + +#endif /* __SOUND_PCXHR_MIX22_H */ diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index aabc7bc5321e..2436e374586f 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -33,20 +33,24 @@ #include <sound/tlv.h> #include <sound/asoundef.h> #include "pcxhr_mixer.h" +#include "pcxhr_mix22.h" +#define PCXHR_LINE_CAPTURE_LEVEL_MIN 0 /* -112.0 dB */ +#define PCXHR_LINE_CAPTURE_LEVEL_MAX 255 /* +15.5 dB */ +#define PCXHR_LINE_CAPTURE_ZERO_LEVEL 224 /* 0.0 dB ( 0 dBu -> 0 dBFS ) */ -#define PCXHR_ANALOG_CAPTURE_LEVEL_MIN 0 /* -96.0 dB */ -#define PCXHR_ANALOG_CAPTURE_LEVEL_MAX 255 /* +31.5 dB */ -#define PCXHR_ANALOG_CAPTURE_ZERO_LEVEL 224 /* +16.0 dB ( +31.5 dB - fix level +15.5 dB ) */ +#define PCXHR_LINE_PLAYBACK_LEVEL_MIN 0 /* -104.0 dB */ +#define PCXHR_LINE_PLAYBACK_LEVEL_MAX 128 /* +24.0 dB */ +#define PCXHR_LINE_PLAYBACK_ZERO_LEVEL 104 /* 0.0 dB ( 0 dBFS -> 0 dBu ) */ -#define PCXHR_ANALOG_PLAYBACK_LEVEL_MIN 0 /* -128.0 dB */ -#define PCXHR_ANALOG_PLAYBACK_LEVEL_MAX 128 /* 0.0 dB */ -#define PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL 104 /* -24.0 dB ( 0.0 dB - fix level +24.0 dB ) */ - -static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -9600, 50, 3150); +static const DECLARE_TLV_DB_SCALE(db_scale_analog_capture, -11200, 50, 1550); static const DECLARE_TLV_DB_SCALE(db_scale_analog_playback, -10400, 100, 2400); -static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) +static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_capture, -11150, 50, 1600); +static const DECLARE_TLV_DB_SCALE(db_scale_a_hr222_playback, -2550, 50, 2400); + +static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, + int is_capture, int channel) { int err, vol; struct pcxhr_rmh rmh; @@ -60,15 +64,17 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur if (chip->analog_playback_active[channel]) vol = chip->analog_playback_volume[channel]; else - vol = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN; - rmh.cmd[2] = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX - vol; /* playback analog levels are inversed */ + vol = PCXHR_LINE_PLAYBACK_LEVEL_MIN; + /* playback analog levels are inversed */ + rmh.cmd[2] = PCXHR_LINE_PLAYBACK_LEVEL_MAX - vol; } rmh.cmd[1] = 1 << ((2 * chip->chip_idx) + channel); /* audio mask */ rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); if (err < 0) { - snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d) " - "is_capture(%d) err(%x)\n", chip->chip_idx, is_capture, err); + snd_printk(KERN_DEBUG "error update_analog_audio_level card(%d)" + " is_capture(%d) err(%x)\n", + chip->chip_idx, is_capture, err); return -EINVAL; } return 0; @@ -80,14 +86,34 @@ static int pcxhr_update_analog_audio_level(struct snd_pcxhr *chip, int is_captur static int pcxhr_analog_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; if (kcontrol->private_value == 0) { /* playback */ - uinfo->value.integer.min = PCXHR_ANALOG_PLAYBACK_LEVEL_MIN; /* -128 dB */ - uinfo->value.integer.max = PCXHR_ANALOG_PLAYBACK_LEVEL_MAX; /* 0 dB */ + if (chip->mgr->is_hr_stereo) { + uinfo->value.integer.min = + HR222_LINE_PLAYBACK_LEVEL_MIN; /* -25 dB */ + uinfo->value.integer.max = + HR222_LINE_PLAYBACK_LEVEL_MAX; /* +24 dB */ + } else { + uinfo->value.integer.min = + PCXHR_LINE_PLAYBACK_LEVEL_MIN; /*-104 dB */ + uinfo->value.integer.max = + PCXHR_LINE_PLAYBACK_LEVEL_MAX; /* +24 dB */ + } } else { /* capture */ - uinfo->value.integer.min = PCXHR_ANALOG_CAPTURE_LEVEL_MIN; /* -96 dB */ - uinfo->value.integer.max = PCXHR_ANALOG_CAPTURE_LEVEL_MAX; /* 31.5 dB */ + if (chip->mgr->is_hr_stereo) { + uinfo->value.integer.min = + HR222_LINE_CAPTURE_LEVEL_MIN; /*-112 dB */ + uinfo->value.integer.max = + HR222_LINE_CAPTURE_LEVEL_MAX; /* +15.5 dB */ + } else { + uinfo->value.integer.min = + PCXHR_LINE_CAPTURE_LEVEL_MIN; /*-112 dB */ + uinfo->value.integer.max = + PCXHR_LINE_CAPTURE_LEVEL_MAX; /* +15.5 dB */ + } } return 0; } @@ -98,11 +124,11 @@ static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); mutex_lock(&chip->mgr->mixer_mutex); if (kcontrol->private_value == 0) { /* playback */ - ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; - ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; + ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; + ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; } else { /* capture */ - ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; - ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; + ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; + ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; } mutex_unlock(&chip->mgr->mixer_mutex); return 0; @@ -123,18 +149,35 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, &chip->analog_capture_volume[i] : &chip->analog_playback_volume[i]; if (is_capture) { - if (new_volume < PCXHR_ANALOG_CAPTURE_LEVEL_MIN || - new_volume > PCXHR_ANALOG_CAPTURE_LEVEL_MAX) - continue; + if (chip->mgr->is_hr_stereo) { + if (new_volume < HR222_LINE_CAPTURE_LEVEL_MIN || + new_volume > HR222_LINE_CAPTURE_LEVEL_MAX) + continue; + } else { + if (new_volume < PCXHR_LINE_CAPTURE_LEVEL_MIN || + new_volume > PCXHR_LINE_CAPTURE_LEVEL_MAX) + continue; + } } else { - if (new_volume < PCXHR_ANALOG_PLAYBACK_LEVEL_MIN || - new_volume > PCXHR_ANALOG_PLAYBACK_LEVEL_MAX) - continue; + if (chip->mgr->is_hr_stereo) { + if (new_volume < HR222_LINE_PLAYBACK_LEVEL_MIN || + new_volume > HR222_LINE_PLAYBACK_LEVEL_MAX) + continue; + } else { + if (new_volume < PCXHR_LINE_PLAYBACK_LEVEL_MIN || + new_volume > PCXHR_LINE_PLAYBACK_LEVEL_MAX) + continue; + } } if (*stored_volume != new_volume) { *stored_volume = new_volume; changed = 1; - pcxhr_update_analog_audio_level(chip, is_capture, i); + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, + is_capture, i); + else + pcxhr_update_analog_audio_level(chip, + is_capture, i); } } mutex_unlock(&chip->mgr->mixer_mutex); @@ -153,6 +196,7 @@ static struct snd_kcontrol_new pcxhr_control_analog_level = { }; /* shared */ + #define pcxhr_sw_info snd_ctl_boolean_stereo_info static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol, @@ -180,7 +224,10 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, !!ucontrol->value.integer.value[i]; changed = 1; /* update playback levels */ - pcxhr_update_analog_audio_level(chip, 0, i); + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 0, i); + else + pcxhr_update_analog_audio_level(chip, 0, i); } } mutex_unlock(&chip->mgr->mixer_mutex); @@ -251,7 +298,8 @@ static int pcxhr_update_playback_stream_level(struct snd_pcxhr* chip, int idx) #define VALID_AUDIO_IO_MUTE_LEVEL 0x000004 #define VALID_AUDIO_IO_MUTE_MONITOR_1 0x000008 -static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, int channel) +static int pcxhr_update_audio_pipe_level(struct snd_pcxhr *chip, + int capture, int channel) { int err; struct pcxhr_rmh rmh; @@ -264,18 +312,20 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in pcxhr_init_rmh(&rmh, CMD_AUDIO_LEVEL_ADJUST); /* add channel mask */ - pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, 1 << (channel + pipe->first_audio)); - /* TODO : if mask (3 << pipe->first_audio) is used, left and right channel - * will be programmed to the same params - */ + pcxhr_set_pipe_cmd_params(&rmh, capture, 0, 0, + 1 << (channel + pipe->first_audio)); + /* TODO : if mask (3 << pipe->first_audio) is used, left and right + * channel will be programmed to the same params */ if (capture) { rmh.cmd[0] |= VALID_AUDIO_IO_DIGITAL_LEVEL; - /* VALID_AUDIO_IO_MUTE_LEVEL not yet handled (capture pipe level) */ + /* VALID_AUDIO_IO_MUTE_LEVEL not yet handled + * (capture pipe level) */ rmh.cmd[2] = chip->digital_capture_volume[channel]; } else { - rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | VALID_AUDIO_IO_MUTE_MONITOR_1; - /* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL not yet - * handled (playback pipe level) + rmh.cmd[0] |= VALID_AUDIO_IO_MONITOR_LEVEL | + VALID_AUDIO_IO_MUTE_MONITOR_1; + /* VALID_AUDIO_IO_DIGITAL_LEVEL and VALID_AUDIO_IO_MUTE_LEVEL + * not yet handled (playback pipe level) */ rmh.cmd[2] = chip->monitoring_volume[channel] << 10; if (chip->monitoring_active[channel] == 0) @@ -284,8 +334,8 @@ static int pcxhr_update_audio_pipe_level(struct snd_pcxhr* chip, int capture, in rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); - if(err<0) { - snd_printk(KERN_DEBUG "error update_audio_level card(%d) err(%x)\n", + if (err < 0) { + snd_printk(KERN_DEBUG "error update_audio_level(%d) err=%x\n", chip->chip_idx, err); return -EINVAL; } @@ -309,15 +359,15 @@ static int pcxhr_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int *stored_volume; int is_capture = kcontrol->private_value; mutex_lock(&chip->mgr->mixer_mutex); - if (is_capture) - stored_volume = chip->digital_capture_volume; /* digital capture */ - else - stored_volume = chip->digital_playback_volume[idx]; /* digital playback */ + if (is_capture) /* digital capture */ + stored_volume = chip->digital_capture_volume; + else /* digital playback */ + stored_volume = chip->digital_playback_volume[idx]; ucontrol->value.integer.value[0] = stored_volume[0]; ucontrol->value.integer.value[1] = stored_volume[1]; mutex_unlock(&chip->mgr->mixer_mutex); @@ -328,7 +378,7 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ + int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int changed = 0; int is_capture = kcontrol->private_value; int *stored_volume; @@ -384,7 +434,8 @@ static int pcxhr_pcm_sw_get(struct snd_kcontrol *kcontrol, return 0; } -static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int changed = 0; @@ -444,8 +495,8 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, if (chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { chip->monitoring_volume[i] = - !!ucontrol->value.integer.value[i]; - if(chip->monitoring_active[i]) + ucontrol->value.integer.value[i]; + if (chip->monitoring_active[i]) /* update monitoring volume and mute */ /* do only when monitoring is unmuted */ pcxhr_update_audio_pipe_level(chip, 0, i); @@ -460,7 +511,7 @@ static struct snd_kcontrol_new pcxhr_control_monitor_vol = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .name = "Monitoring Volume", + .name = "Monitoring Playback Volume", .info = pcxhr_digital_vol_info, /* shared */ .get = pcxhr_monitor_vol_get, .put = pcxhr_monitor_vol_put, @@ -511,7 +562,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new pcxhr_control_monitor_sw = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Monitoring Switch", + .name = "Monitoring Playback Switch", .info = pcxhr_sw_info, /* shared */ .get = pcxhr_monitor_sw_get, .put = pcxhr_monitor_sw_put @@ -533,7 +584,7 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) struct pcxhr_rmh rmh; unsigned int mask, reg; unsigned int codec; - int err, use_src, changed; + int err, changed; switch (chip->chip_idx) { case 0 : mask = PCXHR_SOURCE_AUDIO01_UER; codec = CS8420_01_CS; break; @@ -542,13 +593,10 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) case 3 : mask = PCXHR_SOURCE_AUDIO67_UER; codec = CS8420_67_CS; break; default: return -EINVAL; } - reg = 0; /* audio source from analog plug */ - use_src = 0; /* do not activate codec SRC */ - if (chip->audio_capture_source != 0) { reg = mask; /* audio source from digital plug */ - if (chip->audio_capture_source == 2) - use_src = 1; + } else { + reg = 0; /* audio source from analog plug */ } /* set the input source */ pcxhr_write_io_num_reg_cont(chip->mgr, mask, reg, &changed); @@ -560,29 +608,61 @@ static int pcxhr_set_audio_source(struct snd_pcxhr* chip) if (err) return err; } - pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); /* set codec SRC on off */ - rmh.cmd_len = 3; - rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; - rmh.cmd[1] = codec; - rmh.cmd[2] = (CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x54); - err = pcxhr_send_msg(chip->mgr, &rmh); - if(err) - return err; - rmh.cmd[2] = (CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | (use_src ? 0x41 : 0x49); - err = pcxhr_send_msg(chip->mgr, &rmh); + if (chip->mgr->board_aes_in_192k) { + int i; + unsigned int src_config = 0xC0; + /* update all src configs with one call */ + for (i = 0; (i < 4) && (i < chip->mgr->capture_chips); i++) { + if (chip->mgr->chip[i]->audio_capture_source == 2) + src_config |= (1 << (3 - i)); + } + /* set codec SRC on off */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); + rmh.cmd_len = 2; + rmh.cmd[0] |= IO_NUM_REG_CONFIG_SRC; + rmh.cmd[1] = src_config; + err = pcxhr_send_msg(chip->mgr, &rmh); + } else { + int use_src = 0; + if (chip->audio_capture_source == 2) + use_src = 1; + /* set codec SRC on off */ + pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); + rmh.cmd_len = 3; + rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; + rmh.cmd[1] = codec; + rmh.cmd[2] = ((CS8420_DATA_FLOW_CTL & CHIP_SIG_AND_MAP_SPI) | + (use_src ? 0x41 : 0x54)); + err = pcxhr_send_msg(chip->mgr, &rmh); + if (err) + return err; + rmh.cmd[2] = ((CS8420_CLOCK_SRC_CTL & CHIP_SIG_AND_MAP_SPI) | + (use_src ? 0x41 : 0x49)); + err = pcxhr_send_msg(chip->mgr, &rmh); + } return err; } static int pcxhr_audio_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = {"Analog", "Digital", "Digi+SRC"}; + static const char *texts[5] = { + "Line", "Digital", "Digi+SRC", "Mic", "Line+Mic" + }; + int i; + struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); + i = 2; /* no SRC, no Mic available */ + if (chip->mgr->board_has_aes1) { + i = 3; /* SRC available */ + if (chip->mgr->board_has_mic) + i = 5; /* Mic and MicroMix available */ + } uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 3; - if (uinfo->value.enumerated.item > 2) - uinfo->value.enumerated.item = 2; + uinfo->value.enumerated.items = i; + if (uinfo->value.enumerated.item > (i-1)) + uinfo->value.enumerated.item = i-1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); return 0; @@ -601,13 +681,21 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol, { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int ret = 0; - - if (ucontrol->value.enumerated.item[0] >= 3) + int i = 2; /* no SRC, no Mic available */ + if (chip->mgr->board_has_aes1) { + i = 3; /* SRC available */ + if (chip->mgr->board_has_mic) + i = 5; /* Mic and MicroMix available */ + } + if (ucontrol->value.enumerated.item[0] >= i) return -EINVAL; mutex_lock(&chip->mgr->mixer_mutex); if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) { chip->audio_capture_source = ucontrol->value.enumerated.item[0]; - pcxhr_set_audio_source(chip); + if (chip->mgr->is_hr_stereo) + hr222_set_audio_source(chip); + else + pcxhr_set_audio_source(chip); ret = 1; } mutex_unlock(&chip->mgr->mixer_mutex); @@ -626,25 +714,46 @@ static struct snd_kcontrol_new pcxhr_control_audio_src = { /* * clock type selection * enum pcxhr_clock_type { - * PCXHR_CLOCK_TYPE_INTERNAL = 0, - * PCXHR_CLOCK_TYPE_WORD_CLOCK, - * PCXHR_CLOCK_TYPE_AES_SYNC, - * PCXHR_CLOCK_TYPE_AES_1, - * PCXHR_CLOCK_TYPE_AES_2, - * PCXHR_CLOCK_TYPE_AES_3, - * PCXHR_CLOCK_TYPE_AES_4, - * }; + * PCXHR_CLOCK_TYPE_INTERNAL = 0, + * PCXHR_CLOCK_TYPE_WORD_CLOCK, + * PCXHR_CLOCK_TYPE_AES_SYNC, + * PCXHR_CLOCK_TYPE_AES_1, + * PCXHR_CLOCK_TYPE_AES_2, + * PCXHR_CLOCK_TYPE_AES_3, + * PCXHR_CLOCK_TYPE_AES_4, + * PCXHR_CLOCK_TYPE_MAX = PCXHR_CLOCK_TYPE_AES_4, + * HR22_CLOCK_TYPE_INTERNAL = PCXHR_CLOCK_TYPE_INTERNAL, + * HR22_CLOCK_TYPE_AES_SYNC, + * HR22_CLOCK_TYPE_AES_1, + * HR22_CLOCK_TYPE_MAX = HR22_CLOCK_TYPE_AES_1, + * }; */ static int pcxhr_clock_type_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[7] = { - "Internal", "WordClock", "AES Sync", "AES 1", "AES 2", "AES 3", "AES 4" + static const char *textsPCXHR[7] = { + "Internal", "WordClock", "AES Sync", + "AES 1", "AES 2", "AES 3", "AES 4" + }; + static const char *textsHR22[3] = { + "Internal", "AES Sync", "AES 1" }; + const char **texts; struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); - int clock_items = 3 + mgr->capture_chips; - + int clock_items = 2; /* at least Internal and AES Sync clock */ + if (mgr->board_has_aes1) { + clock_items += mgr->capture_chips; /* add AES x */ + if (!mgr->is_hr_stereo) + clock_items += 1; /* add word clock */ + } + if (mgr->is_hr_stereo) { + texts = textsHR22; + snd_BUG_ON(clock_items > (HR22_CLOCK_TYPE_MAX+1)); + } else { + texts = textsPCXHR; + snd_BUG_ON(clock_items > (PCXHR_CLOCK_TYPE_MAX+1)); + } uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = clock_items; @@ -667,9 +776,13 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); - unsigned int clock_items = 3 + mgr->capture_chips; int rate, ret = 0; - + unsigned int clock_items = 2; /* at least Internal and AES Sync clock */ + if (mgr->board_has_aes1) { + clock_items += mgr->capture_chips; /* add AES x */ + if (!mgr->is_hr_stereo) + clock_items += 1; /* add word clock */ + } if (ucontrol->value.enumerated.item[0] >= clock_items) return -EINVAL; mutex_lock(&mgr->mixer_mutex); @@ -677,7 +790,8 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, mutex_lock(&mgr->setup_mutex); mgr->use_clock_type = ucontrol->value.enumerated.item[0]; if (mgr->use_clock_type) - pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate); + pcxhr_get_external_clock(mgr, mgr->use_clock_type, + &rate); else rate = mgr->sample_rate; if (rate) { @@ -686,7 +800,7 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, mgr->sample_rate = rate; } mutex_unlock(&mgr->setup_mutex); - ret = 1; /* return 1 even if the set was not done. ok ? */ + ret = 1; /* return 1 even if the set was not done. ok ? */ } mutex_unlock(&mgr->mixer_mutex); return ret; @@ -747,14 +861,16 @@ static struct snd_kcontrol_new pcxhr_control_clock_rate = { /* * IEC958 status bits */ -static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int pcxhr_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; uinfo->count = 1; return 0; } -static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char* aes_bits) +static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char *aes_bits) { int i, err; unsigned char temp; @@ -763,39 +879,61 @@ static int pcxhr_iec958_capture_byte(struct snd_pcxhr *chip, int aes_idx, unsign pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_READ); rmh.cmd[0] |= IO_NUM_UER_CHIP_REG; switch (chip->chip_idx) { - case 0: rmh.cmd[1] = CS8420_01_CS; break; /* use CS8416_01_CS for AES SYNC plug */ + /* instead of CS8420_01_CS use CS8416_01_CS for AES SYNC plug */ + case 0: rmh.cmd[1] = CS8420_01_CS; break; case 1: rmh.cmd[1] = CS8420_23_CS; break; case 2: rmh.cmd[1] = CS8420_45_CS; break; case 3: rmh.cmd[1] = CS8420_67_CS; break; default: return -EINVAL; } - switch (aes_idx) { - case 0: rmh.cmd[2] = CS8420_CSB0; break; /* use CS8416_CSBx for AES SYNC plug */ - case 1: rmh.cmd[2] = CS8420_CSB1; break; - case 2: rmh.cmd[2] = CS8420_CSB2; break; - case 3: rmh.cmd[2] = CS8420_CSB3; break; - case 4: rmh.cmd[2] = CS8420_CSB4; break; - default: return -EINVAL; + if (chip->mgr->board_aes_in_192k) { + switch (aes_idx) { + case 0: rmh.cmd[2] = CS8416_CSB0; break; + case 1: rmh.cmd[2] = CS8416_CSB1; break; + case 2: rmh.cmd[2] = CS8416_CSB2; break; + case 3: rmh.cmd[2] = CS8416_CSB3; break; + case 4: rmh.cmd[2] = CS8416_CSB4; break; + default: return -EINVAL; + } + } else { + switch (aes_idx) { + /* instead of CS8420_CSB0 use CS8416_CSBx for AES SYNC plug */ + case 0: rmh.cmd[2] = CS8420_CSB0; break; + case 1: rmh.cmd[2] = CS8420_CSB1; break; + case 2: rmh.cmd[2] = CS8420_CSB2; break; + case 3: rmh.cmd[2] = CS8420_CSB3; break; + case 4: rmh.cmd[2] = CS8420_CSB4; break; + default: return -EINVAL; + } } - rmh.cmd[1] &= 0x0fffff; /* size and code the chip id for the fpga */ - rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI; /* chip signature + map for spi read */ + /* size and code the chip id for the fpga */ + rmh.cmd[1] &= 0x0fffff; + /* chip signature + map for spi read */ + rmh.cmd[2] &= CHIP_SIG_AND_MAP_SPI; rmh.cmd_len = 3; err = pcxhr_send_msg(chip->mgr, &rmh); if (err) return err; - temp = 0; - for (i = 0; i < 8; i++) { - /* attention : reversed bit order (not with CS8416_01_CS) */ - temp <<= 1; - if (rmh.stat[1] & (1 << i)) - temp |= 1; + + if (chip->mgr->board_aes_in_192k) { + temp = (unsigned char)rmh.stat[1]; + } else { + temp = 0; + /* reversed bit order (not with CS8416_01_CS) */ + for (i = 0; i < 8; i++) { + temp <<= 1; + if (rmh.stat[1] & (1 << i)) + temp |= 1; + } } - snd_printdd("read iec958 AES %d byte %d = 0x%x\n", chip->chip_idx, aes_idx, temp); + snd_printdd("read iec958 AES %d byte %d = 0x%x\n", + chip->chip_idx, aes_idx, temp); *aes_bits = temp; return 0; } -static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); unsigned char aes_bits; @@ -806,7 +944,12 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v if (kcontrol->private_value == 0) /* playback */ aes_bits = chip->aes_bits[i]; else { /* capture */ - err = pcxhr_iec958_capture_byte(chip, i, &aes_bits); + if (chip->mgr->is_hr_stereo) + err = hr222_iec958_capture_byte(chip, i, + &aes_bits); + else + err = pcxhr_iec958_capture_byte(chip, i, + &aes_bits); if (err) break; } @@ -825,7 +968,8 @@ static int pcxhr_iec958_mask_get(struct snd_kcontrol *kcontrol, return 0; } -static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigned char aes_bits) +static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, + int aes_idx, unsigned char aes_bits) { int i, err, cmd; unsigned char new_bits = aes_bits; @@ -834,12 +978,12 @@ static int pcxhr_iec958_update_byte(struct snd_pcxhr *chip, int aes_idx, unsigne for (i = 0; i < 8; i++) { if ((old_bits & 0x01) != (new_bits & 0x01)) { - cmd = chip->chip_idx & 0x03; /* chip index 0..3 */ - if(chip->chip_idx > 3) + cmd = chip->chip_idx & 0x03; /* chip index 0..3 */ + if (chip->chip_idx > 3) /* new bit used if chip_idx>3 (PCX1222HR) */ cmd |= 1 << 22; - cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */ - cmd |= (new_bits & 0x01) << 23; /* add bit value */ + cmd |= ((aes_idx << 3) + i) << 2; /* add bit offset */ + cmd |= (new_bits & 0x01) << 23; /* add bit value */ pcxhr_init_rmh(&rmh, CMD_ACCESS_IO_WRITE); rmh.cmd[0] |= IO_NUM_REG_CUER; rmh.cmd[1] = cmd; @@ -867,7 +1011,12 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol, mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 5; i++) { if (ucontrol->value.iec958.status[i] != chip->aes_bits[i]) { - pcxhr_iec958_update_byte(chip, i, ucontrol->value.iec958.status[i]); + if (chip->mgr->is_hr_stereo) + hr222_iec958_update_byte(chip, i, + ucontrol->value.iec958.status[i]); + else + pcxhr_iec958_update_byte(chip, i, + ucontrol->value.iec958.status[i]); changed = 1; } } @@ -917,29 +1066,53 @@ static void pcxhr_init_audio_levels(struct snd_pcxhr *chip) /* at boot time the digital volumes are unmuted 0dB */ for (j = 0; j < PCXHR_PLAYBACK_STREAMS; j++) { chip->digital_playback_active[j][i] = 1; - chip->digital_playback_volume[j][i] = PCXHR_DIGITAL_ZERO_LEVEL; + chip->digital_playback_volume[j][i] = + PCXHR_DIGITAL_ZERO_LEVEL; } - /* after boot, only two bits are set on the uer interface */ - chip->aes_bits[0] = IEC958_AES0_PROFESSIONAL | IEC958_AES0_PRO_FS_48000; -/* only for test purpose, remove later */ + /* after boot, only two bits are set on the uer + * interface + */ + chip->aes_bits[0] = (IEC958_AES0_PROFESSIONAL | + IEC958_AES0_PRO_FS_48000); #ifdef CONFIG_SND_DEBUG - /* analog volumes for playback (is LEVEL_MIN after boot) */ + /* analog volumes for playback + * (is LEVEL_MIN after boot) + */ chip->analog_playback_active[i] = 1; - chip->analog_playback_volume[i] = PCXHR_ANALOG_PLAYBACK_ZERO_LEVEL; - pcxhr_update_analog_audio_level(chip, 0, i); + if (chip->mgr->is_hr_stereo) + chip->analog_playback_volume[i] = + HR222_LINE_PLAYBACK_ZERO_LEVEL; + else { + chip->analog_playback_volume[i] = + PCXHR_LINE_PLAYBACK_ZERO_LEVEL; + pcxhr_update_analog_audio_level(chip, 0, i); + } #endif -/* test end */ + /* stereo cards need to be initialised after boot */ + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 0, i); } if (chip->nb_streams_capt) { /* at boot time the digital volumes are unmuted 0dB */ - chip->digital_capture_volume[i] = PCXHR_DIGITAL_ZERO_LEVEL; -/* only for test purpose, remove later */ + chip->digital_capture_volume[i] = + PCXHR_DIGITAL_ZERO_LEVEL; + chip->analog_capture_active = 1; #ifdef CONFIG_SND_DEBUG - /* analog volumes for playback (is LEVEL_MIN after boot) */ - chip->analog_capture_volume[i] = PCXHR_ANALOG_CAPTURE_ZERO_LEVEL; - pcxhr_update_analog_audio_level(chip, 1, i); + /* analog volumes for playback + * (is LEVEL_MIN after boot) + */ + if (chip->mgr->is_hr_stereo) + chip->analog_capture_volume[i] = + HR222_LINE_CAPTURE_ZERO_LEVEL; + else { + chip->analog_capture_volume[i] = + PCXHR_LINE_CAPTURE_ZERO_LEVEL; + pcxhr_update_analog_audio_level(chip, 1, i); + } #endif -/* test end */ + /* stereo cards need to be initialised after boot */ + if (chip->mgr->is_hr_stereo) + hr222_update_analog_audio_level(chip, 1, i); } } @@ -963,90 +1136,125 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) temp = pcxhr_control_analog_level; temp.name = "Master Playback Volume"; temp.private_value = 0; /* playback */ - temp.tlv.p = db_scale_analog_playback; - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + if (mgr->is_hr_stereo) + temp.tlv.p = db_scale_a_hr222_playback; + else + temp.tlv.p = db_scale_analog_playback; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; + /* output mute controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_output_switch, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_output_switch, + chip)); + if (err < 0) return err; - + temp = snd_pcxhr_pcm_vol; temp.name = "PCM Playback Volume"; temp.count = PCXHR_PLAYBACK_STREAMS; temp.private_value = 0; /* playback */ - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_pcm_switch, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_pcm_switch, chip)); + if (err < 0) return err; /* IEC958 controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_playback_iec958_mask, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_playback_iec958_mask, + chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_playback_iec958, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_playback_iec958, + chip)); + if (err < 0) return err; } if (chip->nb_streams_capt) { - /* analog input level control only on first two chips !*/ + /* analog input level control */ temp = pcxhr_control_analog_level; - temp.name = "Master Capture Volume"; + temp.name = "Line Capture Volume"; temp.private_value = 1; /* capture */ - temp.tlv.p = db_scale_analog_capture; - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + if (mgr->is_hr_stereo) + temp.tlv.p = db_scale_a_hr222_capture; + else + temp.tlv.p = db_scale_analog_capture; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; temp = snd_pcxhr_pcm_vol; temp.name = "PCM Capture Volume"; temp.count = 1; temp.private_value = 1; /* capture */ - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&temp, chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&temp, chip)); + if (err < 0) return err; + /* Audio source */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_audio_src, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_audio_src, chip)); + if (err < 0) return err; + /* IEC958 controls */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_capture_iec958_mask, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_capture_iec958_mask, + chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_capture_iec958, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_capture_iec958, + chip)); + if (err < 0) return err; + + if (mgr->is_hr_stereo) { + err = hr222_add_mic_controls(chip); + if (err < 0) + return err; + } } /* monitoring only if playback and capture device available */ if (chip->nb_streams_capt > 0 && chip->nb_streams_play > 0) { /* monitoring */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_monitor_vol, - chip))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_monitor_vol, chip)); + if (err < 0) return err; - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_monitor_sw, - chip))) < 0) + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_monitor_sw, chip)); + if (err < 0) return err; } if (i == 0) { /* clock mode only one control per pcxhr */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_clock_type, - mgr))) < 0) + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_clock_type, mgr)); + if (err < 0) return err; - /* non standard control used to scan the external clock presence/frequencies */ - if ((err = snd_ctl_add(chip->card, - snd_ctl_new1(&pcxhr_control_clock_rate, - mgr))) < 0) + /* non standard control used to scan + * the external clock presence/frequencies + */ + err = snd_ctl_add(chip->card, + snd_ctl_new1(&pcxhr_control_clock_rate, mgr)); + if (err < 0) return err; } diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index d723543beadd..fdd3be5b439d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1452,7 +1452,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1; - sprintf (hdsp->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1); + sprintf(hdsp->midi[id].rmidi->name, "HDSP MIDI %d", id+1); hdsp->midi[id].rmidi->private_data = &hdsp->midi[id]; snd_rawmidi_set_ops (hdsp->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_hdsp_midi_output); @@ -4548,11 +4548,20 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne { struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; + int err; switch (cmd) { case SNDRV_HDSP_IOCTL_GET_PEAK_RMS: { struct hdsp_peak_rms __user *peak_rms = (struct hdsp_peak_rms __user *)arg; + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + if (!(hdsp->state & HDSP_FirmwareLoaded)) { snd_printk(KERN_ERR "Hammerfall-DSP: firmware needs to be uploaded to the card.\n"); return -EINVAL; @@ -4572,10 +4581,14 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne unsigned long flags; int i; - if (!(hdsp->state & HDSP_FirmwareLoaded)) { - snd_printk(KERN_ERR "Hammerfall-DSP: Firmware needs to be uploaded to the card.\n"); - return -EINVAL; - } + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + + err = hdsp_check_for_firmware(hdsp, 1); + if (err < 0) + return err; + spin_lock_irqsave(&hdsp->lock, flags); info.pref_sync_ref = (unsigned char)hdsp_pref_sync_ref(hdsp); info.wordclock_sync_check = (unsigned char)hdsp_wc_sync_check(hdsp); @@ -5045,6 +5058,10 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ ssleep(2); + err = hdsp_check_for_iobox(hdsp); + if (err < 0) + return err; + if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { #ifdef HDSP_FW_LOADER if ((err = hdsp_request_fw_loader(hdsp)) < 0) @@ -5057,7 +5074,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, /* init is complete, we return */ return 0; #endif - /* no iobox connected, we defer initialization */ + /* we defer initialization */ snd_printk(KERN_INFO "Hammerfall-DSP: card initialization pending : waiting for firmware\n"); if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 98762f909d64..dc5c4baa1e64 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1293,7 +1293,7 @@ static int __devinit snd_hdspm_create_midi (struct snd_card *card, if (err < 0) return err; - sprintf (hdspm->midi[id].rmidi->name, "%s MIDI %d", card->id, id+1); + sprintf(hdspm->midi[id].rmidi->name, "HDSPM MIDI %d", id+1); hdspm->midi[id].rmidi->private_data = &hdspm->midi[id]; snd_rawmidi_set_ops(hdspm->midi[id].rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index a38c0c790d2b..af76ee862d27 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1033,7 +1033,7 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) } if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = 0; /* no capture */ + chip->can_capture = machine_is_compatible("PowerMac4,2"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 20d0e328288a..8f9e3859c37c 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -666,6 +666,7 @@ static int snd_ps3_init_avsetting(struct snd_ps3_card_info *card) card->avs.avs_audio_width = PS3AV_CMD_AUDIO_WORD_BITS_16; card->avs.avs_audio_format = PS3AV_CMD_AUDIO_FORMAT_PCM; card->avs.avs_audio_source = PS3AV_CMD_AUDIO_SOURCE_SERIAL; + memcpy(card->avs.avs_cs_info, ps3av_mode_cs_info, 8); ret = snd_ps3_change_avsetting(card); @@ -685,6 +686,7 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) { struct snd_ps3_card_info *card = snd_pcm_substream_chip(substream); struct snd_ps3_avsetting_info avs; + int ret; avs = card->avs; @@ -729,19 +731,92 @@ static int snd_ps3_set_avsetting(struct snd_pcm_substream *substream) return 1; } - if ((card->avs.avs_audio_width != avs.avs_audio_width) || - (card->avs.avs_audio_rate != avs.avs_audio_rate)) { - card->avs = avs; - snd_ps3_change_avsetting(card); + memcpy(avs.avs_cs_info, ps3av_mode_cs_info, 8); + if (memcmp(&card->avs, &avs, sizeof(avs))) { pr_debug("%s: after freq=%d width=%d\n", __func__, card->avs.avs_audio_rate, card->avs.avs_audio_width); - return 0; + card->avs = avs; + snd_ps3_change_avsetting(card); + ret = 0; } else + ret = 1; + + /* check CS non-audio bit and mute accordingly */ + if (avs.avs_cs_info[0] & 0x02) + ps3av_audio_mute_analog(1); /* mute if non-audio */ + else + ps3av_audio_mute_analog(0); + + return ret; +} + +/* + * SPDIF status bits controls + */ +static int snd_ps3_spdif_mask_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +/* FIXME: ps3av_set_audio_mode() assumes only consumer mode */ +static int snd_ps3_spdif_cmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memset(ucontrol->value.iec958.status, 0xff, 8); + return 0; +} + +static int snd_ps3_spdif_pmask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int snd_ps3_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memcpy(ucontrol->value.iec958.status, ps3av_mode_cs_info, 8); + return 0; +} + +static int snd_ps3_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (memcmp(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8)) { + memcpy(ps3av_mode_cs_info, ucontrol->value.iec958.status, 8); return 1; + } + return 0; } +static struct snd_kcontrol_new spdif_ctls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_cmask_get, + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_pmask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = snd_ps3_spdif_mask_info, + .get = snd_ps3_spdif_default_get, + .put = snd_ps3_spdif_default_put, + }, +}; static int snd_ps3_map_mmio(void) @@ -842,7 +917,7 @@ static void snd_ps3_audio_set_base_addr(uint64_t ioaddr_start) static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) { - int ret; + int i, ret; u64 lpar_addr, lpar_size; BUG_ON(!firmware_has_feature(FW_FEATURE_PS3_LV1)); @@ -903,6 +978,15 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) strcpy(the_card.card->driver, "PS3"); strcpy(the_card.card->shortname, "PS3"); strcpy(the_card.card->longname, "PS3 sound"); + + /* create control elements */ + for (i = 0; i < ARRAY_SIZE(spdif_ctls); i++) { + ret = snd_ctl_add(the_card.card, + snd_ctl_new1(&spdif_ctls[i], &the_card)); + if (ret < 0) + goto clean_card; + } + /* create PCM devices instance */ /* NOTE:this driver works assuming pcm:substream = 1:1 */ ret = snd_pcm_new(the_card.card, diff --git a/sound/ppc/snd_ps3.h b/sound/ppc/snd_ps3.h index 4b7e6fbbe500..326fb29e82d8 100644 --- a/sound/ppc/snd_ps3.h +++ b/sound/ppc/snd_ps3.h @@ -51,6 +51,7 @@ struct snd_ps3_avsetting_info { uint32_t avs_audio_width; uint32_t avs_audio_format; /* fixed */ uint32_t avs_audio_source; /* fixed */ + unsigned char avs_cs_info[8]; }; /* * PS3 audio 'card' instance diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index f746e15b8481..3eb223385416 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -875,7 +875,8 @@ static struct snd_kcontrol_new snapper_mixers[] __initdata = { .put = tumbler_put_master_switch }, DEFINE_SNAPPER_MIX("PCM Playback Volume", 0, VOL_IDX_PCM), - DEFINE_SNAPPER_MIX("PCM Playback Volume", 1, VOL_IDX_PCM2), + /* Alternative PCM is assigned to Mic analog loopback on iBook G4 */ + DEFINE_SNAPPER_MIX("Mic Playback Volume", 0, VOL_IDX_PCM2), DEFINE_SNAPPER_MIX("Monitor Mix Volume", 0, VOL_IDX_ADC), DEFINE_SNAPPER_MONO("Tone Control - Bass", bass), DEFINE_SNAPPER_MONO("Tone Control - Treble", treble), diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 4dfda6674bec..615ebf0b76e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,8 +23,7 @@ config SND_SOC_AC97_BUS bool # All the supported Soc's -source "sound/soc/at32/Kconfig" -source "sound/soc/at91/Kconfig" +source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d849349f2c66..4d475c3ceb91 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ +obj-$(CONFIG_SND_SOC) += codecs/ atmel/ pxa/ s3c24xx/ sh/ fsl/ davinci/ obj-$(CONFIG_SND_SOC) += omap/ au1x/ blackfin/ diff --git a/sound/soc/at32/Kconfig b/sound/soc/at32/Kconfig deleted file mode 100644 index b0765e86c085..000000000000 --- a/sound/soc/at32/Kconfig +++ /dev/null @@ -1,34 +0,0 @@ -config SND_AT32_SOC - tristate "SoC Audio for the Atmel AT32 System-on-a-Chip" - depends on AVR32 && SND_SOC - help - Say Y or M if you want to add support for codecs attached to - the AT32 SSC interface. You will also need to - to select the audio interfaces to support below. - - -config SND_AT32_SOC_SSC - tristate - - - -config SND_AT32_SOC_PLAYPAQ - tristate "SoC Audio support for PlayPaq with WM8510" - depends on SND_AT32_SOC && BOARD_PLAYPAQ - select SND_AT32_SOC_SSC - select SND_SOC_WM8510 - help - Say Y or M here if you want to add support for SoC audio - on the LRS PlayPaq. - - - -config SND_AT32_SOC_PLAYPAQ_SLAVE - bool "Run CODEC on PlayPaq in slave mode" - depends on SND_AT32_SOC_PLAYPAQ - default n - help - Say Y if you want to run with the AT32 SSC generating the BCLK - and FRAME signals on the PlayPaq. Unless you want to play - with the AT32 as the SSC master, you probably want to say N here, - as this will give you better sound quality. diff --git a/sound/soc/at32/Makefile b/sound/soc/at32/Makefile deleted file mode 100644 index c03e55ececeb..000000000000 --- a/sound/soc/at32/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT32 Platform Support -snd-soc-at32-objs := at32-pcm.o -snd-soc-at32-ssc-objs := at32-ssc.o - -obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o -obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o - -# AT32 Machine Support -snd-soc-playpaq-objs := playpaq_wm8510.o - -obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.h b/sound/soc/at32/at32-pcm.h deleted file mode 100644 index 2a52430417da..000000000000 --- a/sound/soc/at32/at32-pcm.h +++ /dev/null @@ -1,79 +0,0 @@ -/* sound/soc/at32/at32-pcm.h - * ASoC PCM interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_PCM_H -#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__ - -#include <linux/atmel-ssc.h> - - -/* - * Registers and status bits that are required by the PCM driver - * TODO: Is ptcr really used? - */ -struct at32_pdc_regs { - u32 xpr; /* PDC RX/TX pointer */ - u32 xcr; /* PDC RX/TX counter */ - u32 xnpr; /* PDC next RX/TX pointer */ - u32 xncr; /* PDC next RX/TX counter */ - u32 ptcr; /* PDC transfer control */ -}; - - - -/* - * SSC mask info - */ -struct at32_ssc_mask { - u32 ssc_enable; /* SSC RX/TX enable */ - u32 ssc_disable; /* SSC RX/TX disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFF or RXBUFF */ - u32 pdc_enable; /* PDC RX/TX enable */ - u32 pdc_disable; /* PDC RX/TX disable */ -}; - - - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at32_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - struct ssc_device *ssc; /* SSC device for stream */ - struct at32_pdc_regs *pdc; /* PDC register info */ - struct at32_ssc_mask *mask; /* SSC mask info */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler) (u32, struct snd_pcm_substream *); -}; - - - -/* - * The AT32 ASoC platform driver - */ -extern struct snd_soc_platform at32_soc_platform; - - - -/* - * SSC register access (since ssc_writel() / ssc_readl() require literal name) - */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) - -#endif /* __SOUND_SOC_AT32_AT32_PCM_H */ diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c deleted file mode 100644 index 4ef6492c902e..000000000000 --- a/sound/soc/at32/at32-ssc.c +++ /dev/null @@ -1,849 +0,0 @@ -/* sound/soc/at32/at32-ssc.c - * ASoC platform driver for AT32 using SSC as DAI - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * Note that this is basically a port of the sound/soc/at91-ssc.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. - */ - -/* #define DEBUG */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> -#include <linux/atmel_pdc.h> -#include <linux/atmel-ssc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include "at32-pcm.h" -#include "at32-ssc.h" - - - -/*-------------------------------------------------------------------------*\ - * Constants -\*-------------------------------------------------------------------------*/ -#define NUM_SSC_DEVICES 3 - -/* - * SSC direction masks - */ -#define SSC_DIR_MASK_UNUSED 0 -#define SSC_DIR_MASK_PLAYBACK 1 -#define SSC_DIR_MASK_CAPTURE 2 - -/* - * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These - * are expected to be used with SSC_BF - */ -/* START bit field values */ -#define SSC_START_CONTINUOUS 0 -#define SSC_START_TX_RX 1 -#define SSC_START_LOW_RF 2 -#define SSC_START_HIGH_RF 3 -#define SSC_START_FALLING_RF 4 -#define SSC_START_RISING_RF 5 -#define SSC_START_LEVEL_RF 6 -#define SSC_START_EDGE_RF 7 -#define SSS_START_COMPARE_0 8 - -/* CKI bit field values */ -#define SSC_CKI_FALLING 0 -#define SSC_CKI_RISING 1 - -/* CKO bit field values */ -#define SSC_CKO_NONE 0 -#define SSC_CKO_CONTINUOUS 1 -#define SSC_CKO_TRANSFER 2 - -/* CKS bit field values */ -#define SSC_CKS_DIV 0 -#define SSC_CKS_CLOCK 1 -#define SSC_CKS_PIN 2 - -/* FSEDGE bit field values */ -#define SSC_FSEDGE_POSITIVE 0 -#define SSC_FSEDGE_NEGATIVE 1 - -/* FSOS bit field values */ -#define SSC_FSOS_NONE 0 -#define SSC_FSOS_NEGATIVE 1 -#define SSC_FSOS_POSITIVE 2 -#define SSC_FSOS_LOW 3 -#define SSC_FSOS_HIGH 4 -#define SSC_FSOS_TOGGLE 5 - -#define START_DELAY 1 - - - -/*-------------------------------------------------------------------------*\ - * Module data -\*-------------------------------------------------------------------------*/ -/* - * SSC PDC registered required by the PCM DMA engine - */ -static struct at32_pdc_regs pdc_tx_reg = { - .xpr = SSC_PDC_TPR, - .xcr = SSC_PDC_TCR, - .xnpr = SSC_PDC_TNPR, - .xncr = SSC_PDC_TNCR, -}; - - - -static struct at32_pdc_regs pdc_rx_reg = { - .xpr = SSC_PDC_RPR, - .xcr = SSC_PDC_RCR, - .xnpr = SSC_PDC_RNPR, - .xncr = SSC_PDC_RNCR, -}; - - - -/* - * SSC and PDC status bits for transmit and receive - */ -static struct at32_ssc_mask ssc_tx_mask = { - .ssc_enable = SSC_BIT(CR_TXEN), - .ssc_disable = SSC_BIT(CR_TXDIS), - .ssc_endx = SSC_BIT(SR_ENDTX), - .ssc_endbuf = SSC_BIT(SR_TXBUFE), - .pdc_enable = SSC_BIT(PDC_PTCR_TXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_TXTDIS), -}; - - - -static struct at32_ssc_mask ssc_rx_mask = { - .ssc_enable = SSC_BIT(CR_RXEN), - .ssc_disable = SSC_BIT(CR_RXDIS), - .ssc_endx = SSC_BIT(SR_ENDRX), - .ssc_endbuf = SSC_BIT(SR_RXBUFF), - .pdc_enable = SSC_BIT(PDC_PTCR_RXTEN), - .pdc_disable = SSC_BIT(PDC_PTCR_RXTDIS), -}; - - - -/* - * DMA parameters for each SSC - */ -static struct at32_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - { - { - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, - { - { - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }, - }, -}; - - - -static struct at32_ssc_info ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = SSC_DIR_MASK_UNUSED, - .initialized = 0, - }, -}; - - - - -/*-------------------------------------------------------------------------*\ - * ISR -\*-------------------------------------------------------------------------*/ -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA interrupt - * handler in the PCM driver. - */ -static irqreturn_t at32_ssc_interrupt(int irq, void *dev_id) -{ - struct at32_ssc_info *ssc_p = dev_id; - struct at32_pcm_dma_params *dma_params; - u32 ssc_sr; - u32 ssc_substream_mask; - int i; - - ssc_sr = (ssc_readl(ssc_p->ssc->regs, SR) & - ssc_readl(ssc_p->ssc->regs, IMR)); - - /* - * Loop through substreams attached to this SSC. If a DMA-related - * interrupt occured on that substream, call the DMA interrupt - * handler function, if one has been registered in the dma_param - * structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if ((dma_params != NULL) && - (dma_params->dma_intr_handler != NULL)) { - ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); - if (ssc_sr & ssc_substream_mask) { - dma_params->dma_intr_handler(ssc_sr, - dma_params-> - substream); - } - } - } - - - return IRQ_HANDLED; -} - -/*-------------------------------------------------------------------------*\ - * DAI functions -\*-------------------------------------------------------------------------*/ -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at32_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - dir_mask = ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - SSC_DIR_MASK_PLAYBACK : SSC_DIR_MASK_CAPTURE); - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - - - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at32_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - int dir_mask; - - dma_params = ssc_p->dma_params[substream->stream]; - - if (dma_params != NULL) { - ssc_writel(dma_params->ssc->regs, CR, - dma_params->mask->ssc_disable); - pr_debug("%s disabled SSC_SR=0x%08x\n", - (substream->stream ? "receiver" : "transmit"), - ssc_readl(ssc_p->ssc->regs, SR)); - - dma_params->ssc = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[substream->stream] = NULL; - } - - - dir_mask = 1 << substream->stream; - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock */ - pr_debug("at32-ssc: Stopping user %d clock\n", - ssc_p->ssc->user); - clk_disable(ssc_p->ssc->clk); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc->irq, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - /* clear the SSC dividers */ - ssc_p->cmr_div = 0; - ssc_p->tcmr_period = 0; - ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - - - -/* - * Set the SSC system clock rate - */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* TODO: What the heck do I do here? */ - return 0; -} - - - -/* - * Record DAI format for use by hw_params() - */ -static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - - - -/* - * Record SSC clock dividers for use in hw_params() - */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT32_SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT32_SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT32_SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - - - -/* - * Configure the SSC - */ -static int at32_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at32_ssc_info *ssc_p = &ssc_info[id]; - struct at32_pcm_dma_params *dma_params; - int channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - - /* - * Currently, there is only one set of dma_params for each direction. - * If more are added, this code will have to be changed to select - * the proper set - */ - dma_params = &ssc_dma_params[id][substream->stream]; - dma_params->ssc = ssc_p->ssc; - dma_params->substream = substream; - - ssc_p->dma_params[substream->stream] = dma_params; - - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the PCM driver's hw_params() - * function. It should not be used for other purposes as it - * is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - - /* - * Determine sample size in bits and the PDC increment - */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - - case SNDRV_PCM_FORMAT_S16: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - - case SNDRV_PCM_FORMAT_S24: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - - case SNDRV_PCM_FORMAT_S32: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - - default: - pr_warning("at32-ssc: Unsupported PCM format %d", - params_format(params)); - return -EINVAL; - } - pr_debug("at32-ssc: bits = %d, pdc_xfer_size = %d, channels = %d\n", - bits, dma_params->pdc_xfer_size, channels); - - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) - if (bits > 16) { - pr_warning("at32-ssc: " - "sample size %d is too large for I2S\n", - bits); - return -EINVAL; - } - - - /* - * Compute the SSC register settings - */ - switch (ssc_p->daifmt & (SND_SOC_DAIFMT_FORMAT_MASK | - SND_SOC_DAIFMT_MASTER_MASK)) { - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRS clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, SSC_START_FALLING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(RFMR_FSLEN, bits - 1) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, SSC_START_FALLING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) | - SSC_BF(TFMR_FSLEN, bits - 1) | - SSC_BF(TFMR_DATNB, channels - 1) | SSC_BIT(TFMR_MSBF) | - SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clock. - * - * The SSC transmit clock is obtained from the BCLK signal - * on the TK line, and the SSC receive clock is generated from - * the transmit clock. - * - * For single channel data, one sample is transferred on the - * falling edge of the LRC clock. For two channel data, one - * sample is transferred on both edges of the LRC clock. - */ - pr_debug("at32-ssc: SSC mode is I2S BCLK / FRAME slave\n"); - start_event = ((channels == 1) ? - SSC_START_FALLING_RF : SSC_START_EDGE_RF); - - rcmr = (SSC_BF(RCMR_STTDLY, START_DELAY) | - SSC_BF(RCMR_START, start_event) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_CLOCK)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_STTDLY, START_DELAY) | - SSC_BF(TCMR_START, start_event) | - SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | - SSC_BF(TCMR_CKO, SSC_CKO_NONE) | - SSC_BF(TCMR_CKS, SSC_CKS_PIN)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line - */ - pr_debug("at32-ssc: SSC mode is DSP A BCLK / FRAME master\n"); - rcmr = (SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | - SSC_BF(RCMR_STTDLY, 1) | - SSC_BF(RCMR_START, SSC_START_RISING_RF) | - SSC_BF(RCMR_CKI, SSC_CKI_RISING) | - SSC_BF(RCMR_CKO, SSC_CKO_NONE) | - SSC_BF(RCMR_CKS, SSC_CKS_DIV)); - - rfmr = (SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(RFMR_DATNB, channels - 1) | - SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_DATLEN, bits - 1)); - - tcmr = (SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) | - SSC_BF(TCMR_STTDLY, 1) | - SSC_BF(TCMR_START, SSC_START_RISING_RF) | - SSC_BF(TCMR_CKI, SSC_CKI_RISING) | - SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) | - SSC_BF(TCMR_CKS, SSC_CKS_DIV)); - - tfmr = (SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | - SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) | - SSC_BF(TFMR_DATNB, channels - 1) | - SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATLEN, bits - 1)); - break; - - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - pr_warning("at32-ssc: unsupported DAI format 0x%x\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - pr_debug("at32-ssc: RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", - rcmr, rfmr, tcmr, tfmr); - - - if (!ssc_p->initialized) { - /* enable peripheral clock */ - pr_debug("at32-ssc: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); - - /* Reset the SSC and its PDC registers */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); - - ret = request_irq(ssc_p->ssc->irq, at32_ssc_interrupt, 0, - ssc_p->name, ssc_p); - if (ret < 0) { - pr_warning("at32-ssc: request irq failed (%d)\n", ret); - pr_debug("at32-ssc: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - return ret; - } - - ssc_p->initialized = 1; - } - - /* Set SSC clock mode register */ - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); - - /* set transmit clock mode and format */ - ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); - ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); - - pr_debug("at32-ssc: SSC initialized\n"); - return 0; -} - - - -static int at32_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at32_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at32_pcm_dma_params *dma_params; - - dma_params = ssc_p->dma_params[substream->stream]; - - ssc_writel(dma_params->ssc->regs, CR, dma_params->mask->ssc_enable); - - return 0; -} - - - -#ifdef CONFIG_PM -static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive */ - ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); - - /* Save the current interrupt mask, then disable unmasked interrupts */ - ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); - ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); - ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); - ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); - ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); - ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); - - return 0; -} - - - -static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at32_ssc_info *ssc_p; - u32 cr; - - if (!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* restore SSC register settings */ - ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); - ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); - ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); - ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); - ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); - - /* re-enable interrupts */ - ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); - - /* Re-enable recieve and transmit as appropriate */ - cr = 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; - cr |= - (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; - ssc_writel(ssc_p->ssc->regs, CR, cr); - - return 0; -} -#else /* CONFIG_PM */ -# define at32_ssc_suspend NULL -# define at32_ssc_resume NULL -#endif /* CONFIG_PM */ - - -#define AT32_SSC_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - - -#define AT32_SSC_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16 | \ - SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) - - -struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { - { - .name = "at32-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[0], - }, - { - .name = "at32-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[1], - }, - { - .name = "at32-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at32_ssc_suspend, - .resume = at32_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT32_SSC_RATES, - .formats = AT32_SSC_FORMATS, - }, - .ops = { - .startup = at32_ssc_startup, - .shutdown = at32_ssc_shutdown, - .prepare = at32_ssc_prepare, - .hw_params = at32_ssc_hw_params, - }, - .dai_ops = { - .set_sysclk = at32_ssc_set_dai_sysclk, - .set_fmt = at32_ssc_set_dai_fmt, - .set_clkdiv = at32_ssc_set_dai_clkdiv, - }, - .private_data = &ssc_info[2], - }, -}; -EXPORT_SYMBOL_GPL(at32_ssc_dai); - - -MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); -MODULE_DESCRIPTION("AT32 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h deleted file mode 100644 index 3c052dbbe460..000000000000 --- a/sound/soc/at32/at32-ssc.h +++ /dev/null @@ -1,59 +0,0 @@ -/* sound/soc/at32/at32-ssc.h - * ASoC SSC interface for Atmel AT32 SoC - * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SOC_AT32_AT32_SSC_H -#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__ - -#include <linux/types.h> -#include <linux/atmel-ssc.h> - -#include "at32-pcm.h" - - - -struct at32_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - - - -struct at32_ssc_info { - char *name; - struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* true if SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at32_pcm_dma_params *dma_params[2]; - struct at32_ssc_state ssc_state; -}; - - -/* SSC divider ids */ -#define AT32_SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT32_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - - -extern struct snd_soc_dai at32_ssc_dai[]; - - - -#endif /* __SOUND_SOC_AT32_AT32_SSC_H */ diff --git a/sound/soc/at91/Kconfig b/sound/soc/at91/Kconfig deleted file mode 100644 index 905186502e00..000000000000 --- a/sound/soc/at91/Kconfig +++ /dev/null @@ -1,27 +0,0 @@ -config SND_AT91_SOC - tristate "SoC Audio for the Atmel AT91 System-on-Chip" - depends on ARCH_AT91 - help - Say Y or M if you want to add support for codecs attached to - the AT91 SSC interface. You will also need - to select the audio interfaces to support below. - -config SND_AT91_SOC_SSC - tristate - -config SND_AT91_SOC_ETI_B1_WM8731 - tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards" - depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1) - select SND_AT91_SOC_SSC - select SND_SOC_WM8731 - help - Say Y if you want to add support for SoC audio on WM8731-based - Endrelia Technologies Inc ETI-B1 or ETI-C1 boards. - -config SND_AT91_SOC_ETI_SLAVE - bool "Run codec in slave Mode on Endrelia boards" - depends on SND_AT91_SOC_ETI_B1_WM8731 - default n - help - Say Y if you want to run with the AT91 SSC generating the BCLK - and LRC signals on Endrelia boards. diff --git a/sound/soc/at91/Makefile b/sound/soc/at91/Makefile deleted file mode 100644 index f23da17cc328..000000000000 --- a/sound/soc/at91/Makefile +++ /dev/null @@ -1,11 +0,0 @@ -# AT91 Platform Support -snd-soc-at91-objs := at91-pcm.o -snd-soc-at91-ssc-objs := at91-ssc.o - -obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o -obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o - -# AT91 Machine Support -snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o - -obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c deleted file mode 100644 index 7ab48bd25e4c..000000000000 --- a/sound/soc/at91/at91-pcm.c +++ /dev/null @@ -1,434 +0,0 @@ -/* - * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "at91-pcm: " x) -#else -#define DBG(x...) -#endif - -static const struct snd_pcm_hardware at91_pcm_hardware = { - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .period_bytes_min = 32, - .period_bytes_max = 8192, - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, -}; - -struct at91_runtime_data { - struct at91_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of dma buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ - size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - u32 pdc_xpr_save; /* PDC register save */ - u32 pdc_xcr_save; - u32 pdc_xnpr_save; - u32 pdc_xncr_save; -}; - -static void at91_pcm_dma_irq(u32 ssc_sr, - struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - static int count = 0; - - count++; - - if (ssc_sr & params->mask->ssc_endbuf) { - - printk(KERN_WARNING - "at91-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK - ? "underrun" : "overrun", - params->name, ssc_sr, count); - - /* re-start the PDC */ - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - } - - if (ssc_sr & params->mask->ssc_endx) { - - /* Load the PDC next pointer and counter registers */ - prtd->period_ptr += prtd->period_size; - if (prtd->period_ptr >= prtd->dma_buffer_end) { - prtd->period_ptr = prtd->dma_buffer; - } - at91_ssc_write(params->ssc_base + params->pdc->xnpr, - prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - } - - snd_pcm_period_elapsed(substream); -} - -static int at91_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - /* this may get called several times by oss emulation - * with different params */ - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at91_pcm_dma_irq; - - prtd->dma_buffer = runtime->dma_addr; - prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; - prtd->period_size = params_period_bytes(params); - - DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - return 0; -} - -static int at91_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - if (params != NULL) { - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - prtd->params->dma_intr_handler = NULL; - } - - return 0; -} - -static int at91_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - - at91_ssc_write(params->ssc_base + AT91_SSC_IDR, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; -} - -static int at91_pcm_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - prtd->period_ptr = prtd->dma_buffer; - - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); - - prtd->period_ptr += prtd->period_size; - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->period_ptr); - at91_ssc_write(params->ssc_base + params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); - - DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", - (unsigned long) prtd->period_ptr, - at91_ssc_read(params->ssc_base + params->pdc->xpr), - at91_ssc_read(params->ssc_base + params->pdc->xcr), - at91_ssc_read(params->ssc_base + params->pdc->xnpr), - at91_ssc_read(params->ssc_base + params->pdc->xncr)); - - at91_ssc_write(params->ssc_base + AT91_SSC_IER, - params->mask->ssc_endx | params->mask->ssc_endbuf); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, - params->mask->pdc_enable); - - DBG("sr=%lx imr=%lx\n", - at91_ssc_read(params->ssc_base + AT91_SSC_SR), - at91_ssc_read(params->ssc_base + AT91_SSC_IMR)); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - break; - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - break; - - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t at91_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd = runtime->private_data; - struct at91_pcm_dma_params *params = prtd->params; - dma_addr_t ptr; - snd_pcm_uframes_t x; - - ptr = (dma_addr_t) at91_ssc_read(params->ssc_base + params->pdc->xpr); - x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); - - if (x == runtime->buffer_size) - x = 0; - return x; -} - -static int at91_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct at91_runtime_data *prtd; - int ret = 0; - - snd_soc_set_runtime_hwparams(substream, &at91_pcm_hardware); - - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - prtd = kzalloc(sizeof(struct at91_runtime_data), GFP_KERNEL); - if (prtd == NULL) { - ret = -ENOMEM; - goto out; - } - runtime->private_data = prtd; - - out: - return ret; -} - -static int at91_pcm_close(struct snd_pcm_substream *substream) -{ - struct at91_runtime_data *prtd = substream->runtime->private_data; - - kfree(prtd); - return 0; -} - -static int at91_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -struct snd_pcm_ops at91_pcm_ops = { - .open = at91_pcm_open, - .close = at91_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at91_pcm_hw_params, - .hw_free = at91_pcm_hw_free, - .prepare = at91_pcm_prepare, - .trigger = at91_pcm_trigger, - .pointer = at91_pcm_pointer, - .mmap = at91_pcm_mmap, -}; - -static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = at91_pcm_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", - (void *) buf->area, - (void *) buf->addr, - size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static u64 at91_pcm_dmamask = 0xffffffff; - -static int at91_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &at91_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = at91_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -#ifdef CONFIG_PM -static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* disable the PDC and save the PDC registers */ - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_disable); - - prtd->pdc_xpr_save = at91_ssc_read(params->ssc_base + params->pdc->xpr); - prtd->pdc_xcr_save = at91_ssc_read(params->ssc_base + params->pdc->xcr); - prtd->pdc_xnpr_save = at91_ssc_read(params->ssc_base + params->pdc->xnpr); - prtd->pdc_xncr_save = at91_ssc_read(params->ssc_base + params->pdc->xncr); - - return 0; -} - -static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - struct snd_pcm_runtime *runtime = dai->runtime; - struct at91_runtime_data *prtd; - struct at91_pcm_dma_params *params; - - if (!runtime) - return 0; - - prtd = runtime->private_data; - params = prtd->params; - - /* restore the PDC registers and enable the PDC */ - at91_ssc_write(params->ssc_base + params->pdc->xpr, prtd->pdc_xpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xcr, prtd->pdc_xcr_save); - at91_ssc_write(params->ssc_base + params->pdc->xnpr, prtd->pdc_xnpr_save); - at91_ssc_write(params->ssc_base + params->pdc->xncr, prtd->pdc_xncr_save); - - at91_ssc_write(params->ssc_base + ATMEL_PDC_PTCR, params->mask->pdc_enable); - return 0; -} -#else -#define at91_pcm_suspend NULL -#define at91_pcm_resume NULL -#endif - -struct snd_soc_platform at91_soc_platform = { - .name = "at91-audio", - .pcm_ops = &at91_pcm_ops, - .pcm_new = at91_pcm_new, - .pcm_free = at91_pcm_free_dma_buffers, - .suspend = at91_pcm_suspend, - .resume = at91_pcm_resume, -}; - -EXPORT_SYMBOL_GPL(at91_soc_platform); - -MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -MODULE_DESCRIPTION("Atmel AT91 PCM module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h deleted file mode 100644 index e5aada2cb102..000000000000 --- a/sound/soc/at91/at91-pcm.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 3, 2006 - * - * Based on pxa2xx-pcm.h by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_PCM_H -#define _AT91_PCM_H - -#include <mach/hardware.h> - -struct at91_ssc_periph { - void __iomem *base; - u32 pid; -}; - -/* - * Registers and status bits that are required by the PCM driver. - */ -struct at91_pdc_regs { - unsigned int xpr; /* PDC recv/trans pointer */ - unsigned int xcr; /* PDC recv/trans counter */ - unsigned int xnpr; /* PDC next recv/trans pointer */ - unsigned int xncr; /* PDC next recv/trans counter */ - unsigned int ptcr; /* PDC transfer control */ -}; - -struct at91_ssc_mask { - u32 ssc_enable; /* SSC recv/trans enable */ - u32 ssc_disable; /* SSC recv/trans disable */ - u32 ssc_endx; /* SSC ENDTX or ENDRX */ - u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ - u32 pdc_enable; /* PDC recv/trans enable */ - u32 pdc_disable; /* PDC recv/trans disable */ -}; - -/* - * This structure, shared between the PCM driver and the interface, - * contains all information required by the PCM driver to perform the - * PDC DMA operation. All fields except dma_intr_handler() are initialized - * by the interface. The dms_intr_handler() pointer is set by the PCM - * driver and called by the interface SSC interrupt handler if it is - * non-NULL. - */ -struct at91_pcm_dma_params { - char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ - void __iomem *ssc_base; /* SSC base address */ - struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */ - struct at91_ssc_mask *mask;/* SSC & PDC status bits */ - struct snd_pcm_substream *substream; - void (*dma_intr_handler)(u32, struct snd_pcm_substream *); -}; - -extern struct snd_soc_platform at91_soc_platform; - -#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) -#define at91_ssc_write(a,v) __raw_writel((v),(a)) - -#endif /* _AT91_PCM_H */ diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c deleted file mode 100644 index a5b1a79ebffb..000000000000 --- a/sound/soc/at91/at91-ssc.c +++ /dev/null @@ -1,791 +0,0 @@ -/* - * at91-ssc.c -- ALSA SoC AT91 SSC Audio Layer Platform driver - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * - * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/interrupt.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/atmel_pdc.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/at91_pmc.h> -#include <mach/at91_ssc.h> - -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_DEBUG "at91-ssc:" x) -#else -#define DBG(x...) -#endif - -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) -#define NUM_SSC_DEVICES 1 -#else -#define NUM_SSC_DEVICES 3 -#endif - - -/* - * SSC PDC registers required by the PCM DMA engine. - */ -static struct at91_pdc_regs pdc_tx_reg = { - .xpr = ATMEL_PDC_TPR, - .xcr = ATMEL_PDC_TCR, - .xnpr = ATMEL_PDC_TNPR, - .xncr = ATMEL_PDC_TNCR, -}; - -static struct at91_pdc_regs pdc_rx_reg = { - .xpr = ATMEL_PDC_RPR, - .xcr = ATMEL_PDC_RCR, - .xnpr = ATMEL_PDC_RNPR, - .xncr = ATMEL_PDC_RNCR, -}; - -/* - * SSC & PDC status bits for transmit and receive. - */ -static struct at91_ssc_mask ssc_tx_mask = { - .ssc_enable = AT91_SSC_TXEN, - .ssc_disable = AT91_SSC_TXDIS, - .ssc_endx = AT91_SSC_ENDTX, - .ssc_endbuf = AT91_SSC_TXBUFE, - .pdc_enable = ATMEL_PDC_TXTEN, - .pdc_disable = ATMEL_PDC_TXTDIS, -}; - -static struct at91_ssc_mask ssc_rx_mask = { - .ssc_enable = AT91_SSC_RXEN, - .ssc_disable = AT91_SSC_RXDIS, - .ssc_endx = AT91_SSC_ENDRX, - .ssc_endbuf = AT91_SSC_RXBUFF, - .pdc_enable = ATMEL_PDC_RXTEN, - .pdc_disable = ATMEL_PDC_RXTDIS, -}; - - -/* - * DMA parameters. - */ -static struct at91_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { - {{ - .name = "SSC0 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC0 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#if NUM_SSC_DEVICES == 3 - {{ - .name = "SSC1 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC1 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, - {{ - .name = "SSC2 PCM out", - .pdc = &pdc_tx_reg, - .mask = &ssc_tx_mask, - }, - { - .name = "SSC2 PCM in", - .pdc = &pdc_rx_reg, - .mask = &ssc_rx_mask, - }}, -#endif -}; - -struct at91_ssc_state { - u32 ssc_cmr; - u32 ssc_rcmr; - u32 ssc_rfmr; - u32 ssc_tcmr; - u32 ssc_tfmr; - u32 ssc_sr; - u32 ssc_imr; -}; - -static struct at91_ssc_info { - char *name; - struct at91_ssc_periph ssc; - spinlock_t lock; /* lock for dir_mask */ - unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ - unsigned short initialized; /* 1=SSC has been initialized */ - unsigned short daifmt; - unsigned short cmr_div; - unsigned short tcmr_period; - unsigned short rcmr_period; - struct at91_pcm_dma_params *dma_params[2]; - struct at91_ssc_state ssc_state; - -} ssc_info[NUM_SSC_DEVICES] = { - { - .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), - .dir_mask = 0, - .initialized = 0, - }, -#if NUM_SSC_DEVICES == 3 - { - .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), - .dir_mask = 0, - .initialized = 0, - }, - { - .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), - .dir_mask = 0, - .initialized = 0, - }, -#endif -}; - -static unsigned int at91_ssc_sysclk; - -/* - * SSC interrupt handler. Passes PDC interrupts to the DMA - * interrupt handler in the PCM driver. - */ -static irqreturn_t at91_ssc_interrupt(int irq, void *dev_id) -{ - struct at91_ssc_info *ssc_p = dev_id; - struct at91_pcm_dma_params *dma_params; - u32 ssc_sr; - int i; - - ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR) - & at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - - /* - * Loop through the substreams attached to this SSC. If - * a DMA-related interrupt occurred on that substream, call - * the DMA interrupt handler function, if one has been - * registered in the dma_params structure by the PCM driver. - */ - for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { - dma_params = ssc_p->dma_params[i]; - - if (dma_params != NULL && dma_params->dma_intr_handler != NULL && - (ssc_sr & - (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) - - dma_params->dma_intr_handler(ssc_sr, dma_params->substream); - } - - return IRQ_HANDLED; -} - -/* - * Startup. Only that one substream allowed in each direction. - */ -static int at91_ssc_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - int dir_mask; - - DBG("ssc_startup: SSC_SR=0x%08lx\n", - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; - - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); - return -EBUSY; - } - ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); - - return 0; -} - -/* - * Shutdown. Clear DMA parameters and shutdown the SSC if there - * are no other substreams open. - */ -static void at91_ssc_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir, dir_mask; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - if (dma_params != NULL) { - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_disable); - DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), - at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR)); - - dma_params->ssc_base = NULL; - dma_params->substream = NULL; - ssc_p->dma_params[dir] = NULL; - } - - dir_mask = 1 << dir; - - spin_lock_irq(&ssc_p->lock); - ssc_p->dir_mask &= ~dir_mask; - if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - - if (ssc_p->initialized) { - free_irq(ssc_p->ssc.pid, ssc_p); - ssc_p->initialized = 0; - } - - /* Reset the SSC */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - /* Clear the SSC dividers */ - ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; - } - spin_unlock_irq(&ssc_p->lock); -} - -/* - * Record the SSC system clock rate. - */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - /* - * The only clock supplied to the SSC is the AT91 master clock, - * which is only used if the SSC is generating BCLK and/or - * LRC clocks. - */ - switch (clk_id) { - case AT91_SYSCLK_MCK: - at91_ssc_sysclk = freq; - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * Record the DAI format for use in hw_params(). - */ -static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - ssc_p->daifmt = fmt; - return 0; -} - -/* - * Record SSC clock dividers for use in hw_params(). - */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; - - switch (div_id) { - case AT91SSC_CMR_DIV: - /* - * The same master clock divider is used for both - * transmit and receive, so if a value has already - * been set, it must match this value. - */ - if (ssc_p->cmr_div == 0) - ssc_p->cmr_div = div; - else - if (div != ssc_p->cmr_div) - return -EBUSY; - break; - - case AT91SSC_TCMR_PERIOD: - ssc_p->tcmr_period = div; - break; - - case AT91SSC_RCMR_PERIOD: - ssc_p->rcmr_period = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -/* - * Configure the SSC. - */ -static int at91_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - int id = rtd->dai->cpu_dai->id; - struct at91_ssc_info *ssc_p = &ssc_info[id]; - struct at91_pcm_dma_params *dma_params; - int dir, channels, bits; - u32 tfmr, rfmr, tcmr, rcmr; - int start_event; - int ret; - - /* - * Currently, there is only one set of dma params for - * each direction. If more are added, this code will - * have to be changed to select the proper set. - */ - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - - dma_params = &ssc_dma_params[id][dir]; - dma_params->ssc_base = ssc_p->ssc.base; - dma_params->substream = substream; - - ssc_p->dma_params[dir] = dma_params; - - /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() - * function. It should not be used for other purposes - * as it is common to all substreams. - */ - rtd->dai->cpu_dai->dma_data = dma_params; - - channels = params_channels(params); - - /* - * Determine sample size in bits and the PDC increment. - */ - switch(params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bits = 8; - dma_params->pdc_xfer_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bits = 16; - dma_params->pdc_xfer_size = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bits = 24; - dma_params->pdc_xfer_size = 4; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bits = 32; - dma_params->pdc_xfer_size = 4; - break; - default: - printk(KERN_WARNING "at91-ssc: unsupported PCM format\n"); - return -EINVAL; - } - - /* - * The SSC only supports up to 16-bit samples in I2S format, due - * to the size of the Frame Mode Register FSLEN field. - */ - if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S - && bits > 16) { - printk(KERN_WARNING - "at91-ssc: sample size %d is too large for I2S\n", bits); - return -EINVAL; - } - - /* - * Compute SSC register settings. - */ - switch (ssc_p->daifmt - & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: - /* - * I2S format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) - | (((bits - 1) << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* - * I2S format, CODEC supplies BCLK and LRC clocks. - * - * The SSC transmit clock is obtained from the BCLK signal on - * on the TK line, and the SSC receive clock is generated from the - * transmit clock. - * - * For single channel data, one sample is transferred on the falling - * edge of the LRC clock. For two channel data, one sample is - * transferred on both edges of the LRC clock. - */ - start_event = channels == 1 - ? AT91_SSC_START_FALLING_RF - : AT91_SSC_START_EDGE_RF; - - rcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( 0 << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( start_event ) & AT91_SSC_START) - | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_PIN ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (( 0 << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: - /* - * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. - * - * The SSC transmit and receive clocks are generated from the - * MCK divider, and the BCLK signal is output on the SSC TK line. - */ - rcmr = (( ssc_p->rcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - rfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_LOOP) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - tcmr = (( ssc_p->tcmr_period << 24) & AT91_SSC_PERIOD) - | (( 1 << 16) & AT91_SSC_STTDLY) - | (( AT91_SSC_START_RISING_RF ) & AT91_SSC_START) - | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) - | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) - | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); - - tfmr = (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) - | (( 0 << 23) & AT91_SSC_FSDEN) - | (( AT91_SSC_FSOS_POSITIVE ) & AT91_SSC_FSOS) - | (( 0 << 16) & AT91_SSC_FSLEN) - | (((channels - 1) << 8) & AT91_SSC_DATNB) - | (( 1 << 7) & AT91_SSC_MSBF) - | (( 0 << 5) & AT91_SSC_DATDEF) - | (((bits - 1) << 0) & AT91_SSC_DATALEN); - - - - break; - - case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: - default: - printk(KERN_WARNING "at91-ssc: unsupported DAI format 0x%x.\n", - ssc_p->daifmt); - return -EINVAL; - break; - } - DBG("RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); - - if (!ssc_p->initialized) { - - /* Enable PMC peripheral clock for this SSC */ - DBG("Starting pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->ssc.pid); - - /* Reset the SSC and its PDC registers */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, AT91_SSC_SWRST); - - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_RNCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TCR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNPR, 0); - at91_ssc_write(ssc_p->ssc.base + ATMEL_PDC_TNCR, 0); - - if ((ret = request_irq(ssc_p->ssc.pid, at91_ssc_interrupt, - 0, ssc_p->name, ssc_p)) < 0) { - printk(KERN_WARNING "at91-ssc: request_irq failure\n"); - - DBG("Stopping pid %d clock\n", ssc_p->ssc.pid); - at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->ssc.pid); - return ret; - } - - ssc_p->initialized = 1; - } - - /* set SSC clock mode register */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->cmr_div); - - /* set receive clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, rfmr); - - /* set transmit clock mode and format */ - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, tfmr); - - DBG("hw_params: SSC initialized\n"); - return 0; -} - - -static int at91_ssc_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct at91_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; - struct at91_pcm_dma_params *dma_params; - int dir; - - dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; - dma_params = ssc_p->dma_params[dir]; - - at91_ssc_write(dma_params->ssc_base + AT91_SSC_CR, - dma_params->mask->ssc_enable); - - DBG("%s enabled SSC_SR=0x%08lx\n", dir ? "receive" : "transmit", - at91_ssc_read(dma_params->ssc_base + AT91_SSC_SR)); - return 0; -} - - -#ifdef CONFIG_PM -static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - /* Save the status register before disabling transmit and receive. */ - ssc_p->ssc_state.ssc_sr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_SR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - AT91_SSC_TXDIS | AT91_SSC_RXDIS); - - /* Save the current interrupt mask, then disable unmasked interrupts. */ - ssc_p->ssc_state.ssc_imr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_IMR); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IDR, ssc_p->ssc_state.ssc_imr); - - ssc_p->ssc_state.ssc_cmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_CMR); - ssc_p->ssc_state.ssc_rcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RCMR); - ssc_p->ssc_state.ssc_rfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_RFMR); - ssc_p->ssc_state.ssc_tcmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TCMR); - ssc_p->ssc_state.ssc_tfmr = at91_ssc_read(ssc_p->ssc.base + AT91_SSC_TFMR); - - return 0; -} - -static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_dai *cpu_dai) -{ - struct at91_ssc_info *ssc_p; - - if(!cpu_dai->active) - return 0; - - ssc_p = &ssc_info[cpu_dai->id]; - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TFMR, ssc_p->ssc_state.ssc_tfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_TCMR, ssc_p->ssc_state.ssc_tcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RFMR, ssc_p->ssc_state.ssc_rfmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_RCMR, ssc_p->ssc_state.ssc_rcmr); - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CMR, ssc_p->ssc_state.ssc_cmr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_IER, ssc_p->ssc_state.ssc_imr); - - at91_ssc_write(ssc_p->ssc.base + AT91_SSC_CR, - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | - ((ssc_p->ssc_state.ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); - - return 0; -} - -#else -#define at91_ssc_suspend NULL -#define at91_ssc_resume NULL -#endif - -#define AT91_SSC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - -#define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { - { .name = "at91-ssc0", - .id = 0, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[0].ssc, - }, -#if NUM_SSC_DEVICES == 3 - { .name = "at91-ssc1", - .id = 1, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[1].ssc, - }, - { .name = "at91-ssc2", - .id = 2, - .type = SND_SOC_DAI_PCM, - .suspend = at91_ssc_suspend, - .resume = at91_ssc_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = AT91_SSC_RATES, - .formats = AT91_SSC_FORMATS,}, - .ops = { - .startup = at91_ssc_startup, - .shutdown = at91_ssc_shutdown, - .prepare = at91_ssc_prepare, - .hw_params = at91_ssc_hw_params,}, - .dai_ops = { - .set_sysclk = at91_ssc_set_dai_sysclk, - .set_fmt = at91_ssc_set_dai_fmt, - .set_clkdiv = at91_ssc_set_dai_clkdiv,}, - .private_data = &ssc_info[2].ssc, - }, -#endif -}; - -EXPORT_SYMBOL_GPL(at91_ssc_dai); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com"); -MODULE_DESCRIPTION("AT91 SSC ASoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h deleted file mode 100644 index 6b7bf382d06f..000000000000 --- a/sound/soc/at91/at91-ssc.h +++ /dev/null @@ -1,27 +0,0 @@ -/* - * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Jan 9, 2007 - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _AT91_SSC_H -#define _AT91_SSC_H - -/* SSC system clock ids */ -#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ - -/* SSC divider ids */ -#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */ -#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ -#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ - -extern struct snd_soc_dai at91_ssc_dai[]; - -#endif /* _AT91_SSC_H */ - diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c deleted file mode 100644 index 684781e4088b..000000000000 --- a/sound/soc/at91/eti_b1_wm8731.c +++ /dev/null @@ -1,349 +0,0 @@ -/* - * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board. - * - * Author: Frank Mandarino <fmandarino@endrelia.com> - * Endrelia Technologies Inc. - * Created: Mar 29, 2006 - * - * Based on corgi.c by: - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> - * Richard Purdie <richard@openedhand.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <mach/hardware.h> -#include <mach/gpio.h> - -#include "../codecs/wm8731.h" -#include "at91-pcm.h" -#include "at91-ssc.h" - -#if 0 -#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x) -#else -#define DBG(x...) -#endif - -static struct clk *pck1_clk; -static struct clk *pllb_clk; - - -static int eti_b1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* cpu clock is the AT91 master clock sent to the SSC */ - ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, - 60000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Start PCK1 clock. */ - clk_enable(pck1_clk); - DBG("pck1 started\n"); - - return 0; -} - -static void eti_b1_shutdown(struct snd_pcm_substream *substream) -{ - /* Stop PCK1 clock. */ - clk_disable(pck1_clk); - DBG("pck1 stopped\n"); -} - -static int eti_b1_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - unsigned int rate; - int cmr_div, period; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */ - break; - case 32000: - cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */ - period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */ - break; - case 48000: - cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */ - period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - AT91SSC_RCMR_PERIOD, period); - } - if (ret < 0) - return ret; - -#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - /* - * Codec in Master Mode. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - -#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */ - - return 0; -} - -static struct snd_soc_ops eti_b1_ops = { - .startup = eti_b1_startup, - .hw_params = eti_b1_hw_params, - .shutdown = eti_b1_shutdown, -}; - - -static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Int Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), -}; - -static const struct snd_soc_dapm_route intercon[] = { - - /* speaker connected to LHPOUT */ - {"Ext Spk", NULL, "LHPOUT"}, - - /* mic is connected to Mic Jack, with WM8731 Mic Bias */ - {"MICIN", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Int Mic"}, -}; - -/* - * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. - */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) -{ - DBG("eti_b1_wm8731_init() called\n"); - - /* Add specific widgets */ - snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets, - ARRAY_SIZE(eti_b1_dapm_widgets)); - - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); - - /* not connected */ - snd_soc_dapm_disable_pin(codec, "RLINEIN"); - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - - /* always connected */ - snd_soc_dapm_enable_pin(codec, "Int Mic"); - snd_soc_dapm_enable_pin(codec, "Ext Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link eti_b1_dai = { - .name = "WM8731", - .stream_name = "WM8731 PCM", - .cpu_dai = &at91_ssc_dai[1], - .codec_dai = &wm8731_dai, - .init = eti_b1_wm8731_init, - .ops = &eti_b1_ops, -}; - -static struct snd_soc_machine snd_soc_machine_eti_b1 = { - .name = "ETI_B1_WM8731", - .dai_link = &eti_b1_dai, - .num_links = 1, -}; - -static struct wm8731_setup_data eti_b1_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - -static struct snd_soc_device eti_b1_snd_devdata = { - .machine = &snd_soc_machine_eti_b1, - .platform = &at91_soc_platform, - .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &eti_b1_wm8731_setup, -}; - -static struct platform_device *eti_b1_snd_device; - -static int __init eti_b1_init(void) -{ - int ret; - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) { - DBG("SSC1 memory region is busy\n"); - return -EBUSY; - } - - ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K); - if (!ssc->base) { - DBG("SSC1 memory ioremap failed\n"); - ret = -ENOMEM; - goto fail_release_mem; - } - - ssc->pid = AT91RM9200_ID_SSC1; - - eti_b1_snd_device = platform_device_alloc("soc-audio", -1); - if (!eti_b1_snd_device) { - DBG("platform device allocation failed\n"); - ret = -ENOMEM; - goto fail_io_unmap; - } - - platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); - eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; - - ret = platform_device_add(eti_b1_snd_device); - if (ret) { - DBG("platform device add failed\n"); - platform_device_put(eti_b1_snd_device); - goto fail_io_unmap; - } - - at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */ - at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */ - at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */ - at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */ -/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */ - at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */ - - /* - * Set PCK1 parent to PLLB and its rate to 12 Mhz. - */ - pllb_clk = clk_get(NULL, "pllb"); - pck1_clk = clk_get(NULL, "pck1"); - - clk_set_parent(pck1_clk, pllb_clk); - clk_set_rate(pck1_clk, 12000000); - - DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); - - /* assign the GPIO pin to PCK1 */ - at91_set_B_periph(AT91_PIN_PA24, 0); - -#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE - printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n"); -#else - printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n"); -#endif - return ret; - -fail_io_unmap: - iounmap(ssc->base); -fail_release_mem: - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); - return ret; -} - -static void __exit eti_b1_exit(void) -{ - struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data; - - clk_put(pck1_clk); - clk_put(pllb_clk); - - platform_device_unregister(eti_b1_snd_device); - - iounmap(ssc->base); - release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K); -} - -module_init(eti_b1_init); -module_exit(eti_b1_exit); - -/* Module information */ -MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); -MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig new file mode 100644 index 000000000000..a608d7009dbd --- /dev/null +++ b/sound/soc/atmel/Kconfig @@ -0,0 +1,43 @@ +config SND_ATMEL_SOC + tristate "SoC Audio for the Atmel System-on-Chip" + depends on ARCH_AT91 || AVR32 + help + Say Y or M if you want to add support for codecs attached to + the ATMEL SSC interface. You will also need + to select the audio interfaces to support below. + +config SND_ATMEL_SOC_SSC + tristate + depends on SND_ATMEL_SOC + help + Say Y or M if you want to add support for codecs the + ATMEL SSC interface. You will also needs to select the individual + machine drivers to support below. + +config SND_AT91_SOC_SAM9G20_WM8731 + tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" + depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8731 + help + Say Y if you want to add support for SoC audio on WM8731-based + AT91sam9g20 evaluation board. + +config SND_AT32_SOC_PLAYPAQ + tristate "SoC Audio support for PlayPaq with WM8510" + depends on SND_ATMEL_SOC && BOARD_PLAYPAQ + select SND_ATMEL_SOC_SSC + select SND_SOC_WM8510 + help + Say Y or M here if you want to add support for SoC audio + on the LRS PlayPaq. + +config SND_AT32_SOC_PLAYPAQ_SLAVE + bool "Run CODEC on PlayPaq in slave mode" + depends on SND_AT32_SOC_PLAYPAQ + default n + help + Say Y if you want to run with the AT32 SSC generating the BCLK + and FRAME signals on the PlayPaq. Unless you want to play + with the AT32 as the SSC master, you probably want to say N here, + as this will give you better sound quality. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile new file mode 100644 index 000000000000..f54a7cc68e66 --- /dev/null +++ b/sound/soc/atmel/Makefile @@ -0,0 +1,15 @@ +# AT91 Platform Support +snd-soc-atmel-pcm-objs := atmel-pcm.o +snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o + +obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o + +# AT91 Machine Support +snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o + +# AT32 Machine Support +snd-soc-playpaq-objs := playpaq_wm8510.o + +obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o +obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/atmel/atmel-pcm.c index c83584f989a9..394412fb396f 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -1,15 +1,34 @@ -/* sound/soc/at32/at32-pcm.c - * ASoC PCM interface for Atmel AT32 SoC +/* + * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. * - * Copyright (C) 2008 Long Range Systems - * Geoffrey Wossum <gwossum@acm.org> + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. * * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. * - * Note that this is basically a port of the sound/soc/at91-pcm.c to - * the AVR32 kernel. Thanks to Frank Mandarino for that code. + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> @@ -18,14 +37,16 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/atmel_pdc.h> +#include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include "at32-pcm.h" +#include <mach/hardware.h> +#include "atmel-pcm.h" /*--------------------------------------------------------------------------*\ @@ -34,36 +55,33 @@ /* TODO: These values were taken from the AT91 platform driver, check * them against real values for AT32 */ -static const struct snd_pcm_hardware at32_pcm_hardware = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - - .formats = SNDRV_PCM_FMTBIT_S16, - .period_bytes_min = 32, - .period_bytes_max = 8192, /* 512 frames * 16 bytes / frame */ - .periods_min = 2, - .periods_max = 1024, - .buffer_bytes_max = 32 * 1024, +static const struct snd_pcm_hardware atmel_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 32 * 1024, }; - /*--------------------------------------------------------------------------*\ * Data types \*--------------------------------------------------------------------------*/ -struct at32_runtime_data { - struct at32_pcm_dma_params *params; - dma_addr_t dma_buffer; /* physical address of DMA buffer */ - dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ +struct atmel_runtime_data { + struct atmel_pcm_dma_params *params; + dma_addr_t dma_buffer; /* physical address of dma buffer */ + dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ size_t period_size; - dma_addr_t period_ptr; /* physical address of next period */ - int periods; /* period index of period_ptr */ + dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ - /* Save PDC registers (for power management) */ + /* PDC register save */ u32 pdc_xpr_save; u32 pdc_xcr_save; u32 pdc_xnpr_save; @@ -71,49 +89,51 @@ struct at32_runtime_data { }; - /*--------------------------------------------------------------------------*\ * Helper functions \*--------------------------------------------------------------------------*/ -static int at32_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *dmabuf = &substream->dma_buffer; - size_t size = at32_pcm_hardware.buffer_bytes_max; - - dmabuf->dev.type = SNDRV_DMA_TYPE_DEV; - dmabuf->dev.dev = pcm->card->dev; - dmabuf->private_data = NULL; - dmabuf->area = dma_alloc_coherent(pcm->card->dev, size, - &dmabuf->addr, GFP_KERNEL); - pr_debug("at32_pcm: preallocate_dma_buffer: " - "area=%p, addr=%p, size=%ld\n", - (void *)dmabuf->area, (void *)dmabuf->addr, size); - - if (!dmabuf->area) + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = atmel_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm:" + "preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, + (void *) buf->addr, + size); + + if (!buf->area) return -ENOMEM; - dmabuf->bytes = size; + buf->bytes = size; return 0; } - - - /*--------------------------------------------------------------------------*\ * ISR \*--------------------------------------------------------------------------*/ -static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; static int count; count++; + if (ssc_sr & params->mask->ssc_endbuf) { - pr_warning("at32-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "underrun" : "overrun", params->name, ssc_sr, count); + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x, count=%d)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr, count); /* re-start the PDC */ ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, @@ -122,7 +142,6 @@ static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) if (prtd->period_ptr >= prtd->dma_buffer_end) prtd->period_ptr = prtd->dma_buffer; - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xcr, @@ -131,60 +150,58 @@ static void at32_pcm_dma_irq(u32 ssc_sr, struct snd_pcm_substream *substream) params->mask->pdc_enable); } - if (ssc_sr & params->mask->ssc_endx) { /* Load the PDC next pointer and counter registers */ prtd->period_ptr += prtd->period_size; if (prtd->period_ptr >= prtd->dma_buffer_end) prtd->period_ptr = prtd->dma_buffer; + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->period_size / params->pdc_xfer_size); } - snd_pcm_period_elapsed(substream); } - /*--------------------------------------------------------------------------*\ * PCM operations \*--------------------------------------------------------------------------*/ -static int at32_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; + struct atmel_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; /* this may get called several times by oss emulation - * with different params - */ + * with different params */ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); prtd->params = rtd->dai->cpu_dai->dma_data; - prtd->params->dma_intr_handler = at32_pcm_dma_irq; + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; prtd->period_size = params_period_bytes(params); - pr_debug("hw_params: DMA for %s initialized " - "(dma_bytes=%ld, period_size=%ld)\n", - prtd->params->name, runtime->dma_bytes, prtd->period_size); - + pr_debug("atmel-pcm: " + "hw_params: DMA for %s initialized " + "(dma_bytes=%u, period_size=%u)\n", + prtd->params->name, + runtime->dma_bytes, + prtd->period_size); return 0; } - - -static int at32_pcm_hw_free(struct snd_pcm_substream *substream) +static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) { - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; if (params != NULL) { ssc_writex(params->ssc->regs, SSC_PDC_PTCR, @@ -195,32 +212,29 @@ static int at32_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } - - -static int at32_pcm_prepare(struct snd_pcm_substream *substream) +static int atmel_pcm_prepare(struct snd_pcm_substream *substream) { - struct at32_runtime_data *prtd = substream->runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; ssc_writex(params->ssc->regs, SSC_IDR, params->mask->ssc_endx | params->mask->ssc_endbuf); ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_disable); - return 0; } - -static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int atmel_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) { struct snd_pcm_runtime *rtd = substream->runtime; - struct at32_runtime_data *prtd = rtd->private_data; - struct at32_pcm_dma_params *params = prtd->params; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; int ret = 0; - pr_debug("at32_pcm_trigger: buffer_size = %ld, " - "dma_area = %p, dma_bytes = %ld\n", - rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + pr_debug("atmel-pcm:buffer_size = %ld," + "dma_area = %p, dma_bytes = %u\n", + rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -237,26 +251,25 @@ static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->period_size / params->pdc_xfer_size); - pr_debug("trigger: period_ptr=%lx, xpr=%x, " - "xcr=%d, xnpr=%x, xncr=%d\n", - (unsigned long)prtd->period_ptr, - ssc_readx(params->ssc->regs, params->pdc->xpr), - ssc_readx(params->ssc->regs, params->pdc->xcr), - ssc_readx(params->ssc->regs, params->pdc->xnpr), - ssc_readx(params->ssc->regs, params->pdc->xncr)); + pr_debug("atmel-pcm: trigger: " + "period_ptr=%lx, xpr=%u, " + "xcr=%u, xnpr=%u, xncr=%u\n", + (unsigned long)prtd->period_ptr, + ssc_readx(params->ssc->regs, params->pdc->xpr), + ssc_readx(params->ssc->regs, params->pdc->xcr), + ssc_readx(params->ssc->regs, params->pdc->xnpr), + ssc_readx(params->ssc->regs, params->pdc->xncr)); ssc_writex(params->ssc->regs, SSC_IER, params->mask->ssc_endx | params->mask->ssc_endbuf); ssc_writex(params->ssc->regs, SSC_PDC_PTCR, params->mask->pdc_enable); - pr_debug("sr=%x, imr=%x\n", - ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); + pr_debug("sr=%u imr=%u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IER)); break; /* SNDRV_PCM_TRIGGER_START */ - - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -264,7 +277,6 @@ static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) params->mask->pdc_disable); break; - case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, @@ -278,13 +290,12 @@ static int at32_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - - -static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t atmel_pcm_pointer( + struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd = runtime->private_data; - struct at32_pcm_dma_params *params = prtd->params; + struct atmel_runtime_data *prtd = runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; dma_addr_t ptr; snd_pcm_uframes_t x; @@ -297,108 +308,95 @@ static snd_pcm_uframes_t at32_pcm_pointer(struct snd_pcm_substream *substream) return x; } - - -static int at32_pcm_open(struct snd_pcm_substream *substream) +static int atmel_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct at32_runtime_data *prtd; + struct atmel_runtime_data *prtd; int ret = 0; - snd_soc_set_runtime_hwparams(substream, &at32_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); + SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) goto out; - prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + prtd = kzalloc(sizeof(struct atmel_runtime_data), GFP_KERNEL); if (prtd == NULL) { ret = -ENOMEM; goto out; } runtime->private_data = prtd; - -out: + out: return ret; } - - -static int at32_pcm_close(struct snd_pcm_substream *substream) +static int atmel_pcm_close(struct snd_pcm_substream *substream) { - struct at32_runtime_data *prtd = substream->runtime->private_data; + struct atmel_runtime_data *prtd = substream->runtime->private_data; kfree(prtd); return 0; } - -static int at32_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) { return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); } - - -static struct snd_pcm_ops at32_pcm_ops = { - .open = at32_pcm_open, - .close = at32_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = at32_pcm_hw_params, - .hw_free = at32_pcm_hw_free, - .prepare = at32_pcm_prepare, - .trigger = at32_pcm_trigger, - .pointer = at32_pcm_pointer, - .mmap = at32_pcm_mmap, +struct snd_pcm_ops atmel_pcm_ops = { + .open = atmel_pcm_open, + .close = atmel_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_pcm_hw_params, + .hw_free = atmel_pcm_hw_free, + .prepare = atmel_pcm_prepare, + .trigger = atmel_pcm_trigger, + .pointer = atmel_pcm_pointer, + .mmap = atmel_pcm_mmap, }; - /*--------------------------------------------------------------------------*\ * ASoC platform driver \*--------------------------------------------------------------------------*/ -static u64 at32_pcm_dmamask = 0xffffffff; +static u64 atmel_pcm_dmamask = 0xffffffff; -static int at32_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int atmel_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; if (!card->dev->dma_mask) - card->dev->dma_mask = &at32_pcm_dmamask; + card->dev->dma_mask = &atmel_pcm_dmamask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_PLAYBACK); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - pr_debug("at32-pcm: Allocating PCM capture DMA buffer\n"); - ret = at32_pcm_preallocate_dma_buffer( - pcm, SNDRV_PCM_STREAM_CAPTURE); + pr_debug("at32-pcm:" + "Allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; } - - -out: + out: return ret; } - - -static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -406,7 +404,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; - if (substream == NULL) + if (!substream) continue; buf = &substream->dma_buffer; @@ -418,24 +416,23 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) } } - - #ifdef CONFIG_PM -static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int atmel_pcm_suspend(struct platform_device *pdev, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; - if (runtime == NULL) + if (!runtime) return 0; + prtd = runtime->private_data; params = prtd->params; - /* Disable the PDC and save the PDC registers */ - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, - params->mask->pdc_disable); + /* disable the PDC and save the PDC registers */ + + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); @@ -445,48 +442,43 @@ static int at32_pcm_suspend(struct platform_device *pdev, return 0; } - - -static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int atmel_pcm_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; - struct at32_runtime_data *prtd; - struct at32_pcm_dma_params *params; + struct atmel_runtime_data *prtd; + struct atmel_pcm_dma_params *params; - if (runtime == NULL) + if (!runtime) return 0; + prtd = runtime->private_data; params = prtd->params; - /* Restore the PDC registers and enable the PDC */ + /* restore the PDC registers and enable the PDC */ ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); return 0; } -#else /* CONFIG_PM */ -# define at32_pcm_suspend NULL -# define at32_pcm_resume NULL -#endif /* CONFIG_PM */ - - - -struct snd_soc_platform at32_soc_platform = { - .name = "at32-audio", - .pcm_ops = &at32_pcm_ops, - .pcm_new = at32_pcm_new, - .pcm_free = at32_pcm_free_dma_buffers, - .suspend = at32_pcm_suspend, - .resume = at32_pcm_resume, +#else +#define atmel_pcm_suspend NULL +#define atmel_pcm_resume NULL +#endif + +struct snd_soc_platform atmel_soc_platform = { + .name = "atmel-audio", + .pcm_ops = &atmel_pcm_ops, + .pcm_new = atmel_pcm_new, + .pcm_free = atmel_pcm_free_dma_buffers, + .suspend = atmel_pcm_suspend, + .resume = atmel_pcm_resume, }; -EXPORT_SYMBOL_GPL(at32_soc_platform); - - +EXPORT_SYMBOL_GPL(atmel_soc_platform); -MODULE_AUTHOR("Geoffrey Wossum <gwossum@acm.org>"); -MODULE_DESCRIPTION("Atmel AT32 PCM module"); +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>"); +MODULE_DESCRIPTION("Atmel PCM module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h new file mode 100644 index 000000000000..ec9b2824b663 --- /dev/null +++ b/sound/soc/atmel/atmel-pcm.h @@ -0,0 +1,86 @@ +/* + * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on at91-pcm. by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * + * Based on pxa2xx-pcm.c by: + * + * Author: Nicolas Pitre + * Created: Nov 30, 2004 + * Copyright: (C) 2004 MontaVista Software, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_PCM_H +#define _ATMEL_PCM_H + +#include <linux/atmel-ssc.h> + +/* + * Registers and status bits that are required by the PCM driver. + */ +struct atmel_pdc_regs { + unsigned int xpr; /* PDC recv/trans pointer */ + unsigned int xcr; /* PDC recv/trans counter */ + unsigned int xnpr; /* PDC next recv/trans pointer */ + unsigned int xncr; /* PDC next recv/trans counter */ + unsigned int ptcr; /* PDC transfer control */ +}; + +struct atmel_ssc_mask { + u32 ssc_enable; /* SSC recv/trans enable */ + u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_endx; /* SSC ENDTX or ENDRX */ + u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ + u32 pdc_enable; /* PDC recv/trans enable */ + u32 pdc_disable; /* PDC recv/trans disable */ +}; + +/* + * This structure, shared between the PCM driver and the interface, + * contains all information required by the PCM driver to perform the + * PDC DMA operation. All fields except dma_intr_handler() are initialized + * by the interface. The dms_intr_handler() pointer is set by the PCM + * driver and called by the interface SSC interrupt handler if it is + * non-NULL. + */ +struct atmel_pcm_dma_params { + char *name; /* stream identifier */ + int pdc_xfer_size; /* PDC counter increment in bytes */ + struct ssc_device *ssc; /* SSC device for stream */ + struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ + struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ + struct snd_pcm_substream *substream; + void (*dma_intr_handler)(u32, struct snd_pcm_substream *); +}; + +extern struct snd_soc_platform atmel_soc_platform; + + +/* + * SSC register access (since ssc_writel() / ssc_readl() require literal name) + */ +#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) +#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) + +#endif /* _ATMEL_PCM_H */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c new file mode 100644 index 000000000000..0bb18dfa9495 --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -0,0 +1,780 @@ +/* + * atmel_ssc_dai.c -- ALSA SoC ATMEL SSC Audio Layer Platform driver + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino@endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/atmel_pdc.h> + +#include <linux/atmel-ssc.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> + +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#define NUM_SSC_DEVICES 1 +#else +#define NUM_SSC_DEVICES 3 +#endif + +/* + * SSC PDC registers required by the PCM DMA engine. + */ +static struct atmel_pdc_regs pdc_tx_reg = { + .xpr = ATMEL_PDC_TPR, + .xcr = ATMEL_PDC_TCR, + .xnpr = ATMEL_PDC_TNPR, + .xncr = ATMEL_PDC_TNCR, +}; + +static struct atmel_pdc_regs pdc_rx_reg = { + .xpr = ATMEL_PDC_RPR, + .xcr = ATMEL_PDC_RCR, + .xnpr = ATMEL_PDC_RNPR, + .xncr = ATMEL_PDC_RNCR, +}; + +/* + * SSC & PDC status bits for transmit and receive. + */ +static struct atmel_ssc_mask ssc_tx_mask = { + .ssc_enable = SSC_BIT(CR_TXEN), + .ssc_disable = SSC_BIT(CR_TXDIS), + .ssc_endx = SSC_BIT(SR_ENDTX), + .ssc_endbuf = SSC_BIT(SR_TXBUFE), + .pdc_enable = ATMEL_PDC_TXTEN, + .pdc_disable = ATMEL_PDC_TXTDIS, +}; + +static struct atmel_ssc_mask ssc_rx_mask = { + .ssc_enable = SSC_BIT(CR_RXEN), + .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_endx = SSC_BIT(SR_ENDRX), + .ssc_endbuf = SSC_BIT(SR_RXBUFF), + .pdc_enable = ATMEL_PDC_RXTEN, + .pdc_disable = ATMEL_PDC_RXTDIS, +}; + + +/* + * DMA parameters. + */ +static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { + {{ + .name = "SSC0 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC0 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#if NUM_SSC_DEVICES == 3 + {{ + .name = "SSC1 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC1 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, + {{ + .name = "SSC2 PCM out", + .pdc = &pdc_tx_reg, + .mask = &ssc_tx_mask, + }, + { + .name = "SSC2 PCM in", + .pdc = &pdc_rx_reg, + .mask = &ssc_rx_mask, + } }, +#endif +}; + + +static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { + { + .name = "ssc0", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#if NUM_SSC_DEVICES == 3 + { + .name = "ssc1", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, + { + .name = "ssc2", + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), + .dir_mask = SSC_DIR_MASK_UNUSED, + .initialized = 0, + }, +#endif +}; + + +/* + * SSC interrupt handler. Passes PDC interrupts to the DMA + * interrupt handler in the PCM driver. + */ +static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) +{ + struct atmel_ssc_info *ssc_p = dev_id; + struct atmel_pcm_dma_params *dma_params; + u32 ssc_sr; + u32 ssc_substream_mask; + int i; + + ssc_sr = (unsigned long)ssc_readl(ssc_p->ssc->regs, SR) + & (unsigned long)ssc_readl(ssc_p->ssc->regs, IMR); + + /* + * Loop through the substreams attached to this SSC. If + * a DMA-related interrupt occurred on that substream, call + * the DMA interrupt handler function, if one has been + * registered in the dma_params structure by the PCM driver. + */ + for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { + dma_params = ssc_p->dma_params[i]; + + if ((dma_params != NULL) && + (dma_params->dma_intr_handler != NULL)) { + ssc_substream_mask = (dma_params->mask->ssc_endx | + dma_params->mask->ssc_endbuf); + if (ssc_sr & ssc_substream_mask) { + dma_params->dma_intr_handler(ssc_sr, + dma_params-> + substream); + } + } + } + + return IRQ_HANDLED; +} + + +/*-------------------------------------------------------------------------*\ + * DAI functions +\*-------------------------------------------------------------------------*/ +/* + * Startup. Only that one substream allowed in each direction. + */ +static int atmel_ssc_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + int dir_mask; + + pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + ssc_readl(ssc_p->ssc->regs, SR)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir_mask = SSC_DIR_MASK_PLAYBACK; + else + dir_mask = SSC_DIR_MASK_CAPTURE; + + spin_lock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) { + spin_unlock_irq(&ssc_p->lock); + return -EBUSY; + } + ssc_p->dir_mask |= dir_mask; + spin_unlock_irq(&ssc_p->lock); + + return 0; +} + +/* + * Shutdown. Clear DMA parameters and shutdown the SSC if there + * are no other substreams open. + */ +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir, dir_mask; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + if (dma_params != NULL) { + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", + (dir ? "receive" : "transmit"), + ssc_readl(ssc_p->ssc->regs, SR)); + + dma_params->ssc = NULL; + dma_params->substream = NULL; + ssc_p->dma_params[dir] = NULL; + } + + dir_mask = 1 << dir; + + spin_lock_irq(&ssc_p->lock); + ssc_p->dir_mask &= ~dir_mask; + if (!ssc_p->dir_mask) { + if (ssc_p->initialized) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dau: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + + free_irq(ssc_p->ssc->irq, ssc_p); + ssc_p->initialized = 0; + } + + /* Reset the SSC */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Clear the SSC dividers */ + ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; + } + spin_unlock_irq(&ssc_p->lock); +} + + +/* + * Record the DAI format for use in hw_params(). + */ +static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + ssc_p->daifmt = fmt; + return 0; +} + +/* + * Record SSC clock dividers for use in hw_params(). + */ +static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + + switch (div_id) { + case ATMEL_SSC_CMR_DIV: + /* + * The same master clock divider is used for both + * transmit and receive, so if a value has already + * been set, it must match this value. + */ + if (ssc_p->cmr_div == 0) + ssc_p->cmr_div = div; + else + if (div != ssc_p->cmr_div) + return -EBUSY; + break; + + case ATMEL_SSC_TCMR_PERIOD: + ssc_p->tcmr_period = div; + break; + + case ATMEL_SSC_RCMR_PERIOD: + ssc_p->rcmr_period = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* + * Configure the SSC. + */ +static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + int id = rtd->dai->cpu_dai->id; + struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct atmel_pcm_dma_params *dma_params; + int dir, channels, bits; + u32 tfmr, rfmr, tcmr, rcmr; + int start_event; + int ret; + + /* + * Currently, there is only one set of dma params for + * each direction. If more are added, this code will + * have to be changed to select the proper set. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = &ssc_dma_params[id][dir]; + dma_params->ssc = ssc_p->ssc; + dma_params->substream = substream; + + ssc_p->dma_params[dir] = dma_params; + + /* + * The cpu_dai->dma_data field is only used to communicate the + * appropriate DMA parameters to the pcm driver hw_params() + * function. It should not be used for other purposes + * as it is common to all substreams. + */ + rtd->dai->cpu_dai->dma_data = dma_params; + + channels = params_channels(params); + + /* + * Determine sample size in bits and the PDC increment. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + bits = 8; + dma_params->pdc_xfer_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + bits = 16; + dma_params->pdc_xfer_size = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = 24; + dma_params->pdc_xfer_size = 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = 32; + dma_params->pdc_xfer_size = 4; + break; + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); + return -EINVAL; + } + + /* + * The SSC only supports up to 16-bit samples in I2S format, due + * to the size of the Frame Mode Register FSLEN field. + */ + if ((ssc_p->daifmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S + && bits > 16) { + printk(KERN_WARNING + "atmel_ssc_dai: sample size %d" + "is too large for I2S\n", bits); + return -EINVAL; + } + + /* + * Compute SSC register settings. + */ + switch (ssc_p->daifmt + & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_MASTER_MASK)) { + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS: + /* + * I2S format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated + * from the MCK divider, and the BCLK signal + * is output on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(RFMR_FSLEN, (bits - 1)) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NEGATIVE) + | SSC_BF(TFMR_FSLEN, (bits - 1)) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * + * The SSC transmit clock is obtained from the BCLK signal on + * on the TK line, and the SSC receive clock is + * generated from the transmit clock. + * + * For single channel data, one sample is transferred + * on the falling edge of the LRC clock. + * For two channel data, one sample is + * transferred on both edges of the LRC clock. + */ + start_event = ((channels == 1) + ? SSC_START_FALLING_RF + : SSC_START_EDGE_RF); + + rcmr = SSC_BF(RCMR_PERIOD, 0) + | SSC_BF(RCMR_STTDLY, START_DELAY) + | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, 0) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, 0) + | SSC_BF(TCMR_STTDLY, START_DELAY) + | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(TCMR_CKO, SSC_CKO_NONE) + | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, 0) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS: + /* + * DSP/PCM Mode A format, SSC provides BCLK and LRC clocks. + * + * The SSC transmit and receive clocks are generated from the + * MCK divider, and the BCLK signal is output + * on the SSC TK line. + */ + rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) + | SSC_BF(RCMR_STTDLY, 1) + | SSC_BF(RCMR_START, SSC_START_RISING_RF) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) + | SSC_BF(RCMR_CKO, SSC_CKO_NONE) + | SSC_BF(RCMR_CKS, SSC_CKS_DIV); + + rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(RFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(RFMR_FSLEN, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) + | SSC_BIT(RFMR_MSBF) + | SSC_BF(RFMR_LOOP, 0) + | SSC_BF(RFMR_DATLEN, (bits - 1)); + + tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_START, SSC_START_RISING_RF) + | SSC_BF(TCMR_CKI, SSC_CKI_RISING) + | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) + | SSC_BF(TCMR_CKS, SSC_CKS_DIV); + + tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) + | SSC_BF(TFMR_FSDEN, 0) + | SSC_BF(TFMR_FSOS, SSC_FSOS_POSITIVE) + | SSC_BF(TFMR_FSLEN, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATDEF, 0) + | SSC_BF(TFMR_DATLEN, (bits - 1)); + break; + + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM: + default: + printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", + ssc_p->daifmt); + return -EINVAL; + break; + } + pr_debug("atmel_ssc_hw_params: " + "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", + rcmr, rfmr, tcmr, tfmr); + + if (!ssc_p->initialized) { + + /* Enable PMC peripheral clock for this SSC */ + pr_debug("atmel_ssc_dai: Starting clock\n"); + clk_enable(ssc_p->ssc->clk); + + /* Reset the SSC and its PDC registers */ + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + + ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, + ssc_p->name, ssc_p); + if (ret < 0) { + printk(KERN_WARNING + "atmel_ssc_dai: request_irq failure\n"); + pr_debug("Atmel_ssc_dai: Stoping clock\n"); + clk_disable(ssc_p->ssc->clk); + return ret; + } + + ssc_p->initialized = 1; + } + + /* set SSC clock mode register */ + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->cmr_div); + + /* set receive clock mode and format */ + ssc_writel(ssc_p->ssc->regs, RCMR, rcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, rfmr); + + /* set transmit clock mode and format */ + ssc_writel(ssc_p->ssc->regs, TCMR, tcmr); + ssc_writel(ssc_p->ssc->regs, TFMR, tfmr); + + pr_debug("atmel_ssc_dai,hw_params: SSC initialized\n"); + return 0; +} + + +static int atmel_ssc_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + + pr_debug("%s enabled SSC_SR=0x%08x\n", + dir ? "receive" : "transmit", + ssc_readl(ssc_p->ssc->regs, SR)); + return 0; +} + + +#ifdef CONFIG_PM +static int atmel_ssc_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* Save the status register before disabling transmit and receive */ + ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_TXDIS) | SSC_BIT(CR_RXDIS)); + + /* Save the current interrupt mask, then disable unmasked interrupts */ + ssc_p->ssc_state.ssc_imr = ssc_readl(ssc_p->ssc->regs, IMR); + ssc_writel(ssc_p->ssc->regs, IDR, ssc_p->ssc_state.ssc_imr); + + ssc_p->ssc_state.ssc_cmr = ssc_readl(ssc_p->ssc->regs, CMR); + ssc_p->ssc_state.ssc_rcmr = ssc_readl(ssc_p->ssc->regs, RCMR); + ssc_p->ssc_state.ssc_rfmr = ssc_readl(ssc_p->ssc->regs, RFMR); + ssc_p->ssc_state.ssc_tcmr = ssc_readl(ssc_p->ssc->regs, TCMR); + ssc_p->ssc_state.ssc_tfmr = ssc_readl(ssc_p->ssc->regs, TFMR); + + return 0; +} + + + +static int atmel_ssc_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct atmel_ssc_info *ssc_p; + u32 cr; + + if (!cpu_dai->active) + return 0; + + ssc_p = &ssc_info[cpu_dai->id]; + + /* restore SSC register settings */ + ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); + ssc_writel(ssc_p->ssc->regs, TCMR, ssc_p->ssc_state.ssc_tcmr); + ssc_writel(ssc_p->ssc->regs, RFMR, ssc_p->ssc_state.ssc_rfmr); + ssc_writel(ssc_p->ssc->regs, RCMR, ssc_p->ssc_state.ssc_rcmr); + ssc_writel(ssc_p->ssc->regs, CMR, ssc_p->ssc_state.ssc_cmr); + + /* re-enable interrupts */ + ssc_writel(ssc_p->ssc->regs, IER, ssc_p->ssc_state.ssc_imr); + + /* Re-enable recieve and transmit as appropriate */ + cr = 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_RXEN)) ? SSC_BIT(CR_RXEN) : 0; + cr |= + (ssc_p->ssc_state.ssc_sr & SSC_BIT(SR_TXEN)) ? SSC_BIT(CR_TXEN) : 0; + ssc_writel(ssc_p->ssc->regs, CR, cr); + + return 0; +} +#else /* CONFIG_PM */ +# define atmel_ssc_suspend NULL +# define atmel_ssc_resume NULL +#endif /* CONFIG_PM */ + + +#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) + +#define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { + { .name = "atmel-ssc0", + .id = 0, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[0], + }, +#if NUM_SSC_DEVICES == 3 + { .name = "atmel-ssc1", + .id = 1, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[1], + }, + { .name = "atmel-ssc2", + .id = 2, + .suspend = atmel_ssc_suspend, + .resume = atmel_ssc_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = ATMEL_SSC_RATES, + .formats = ATMEL_SSC_FORMATS,}, + .ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .private_data = &ssc_info[2], + }, +#endif +}; +EXPORT_SYMBOL_GPL(atmel_ssc_dai); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou, sedji.gaouaou@atmel.com, www.atmel.com"); +MODULE_DESCRIPTION("ATMEL SSC ASoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h new file mode 100644 index 000000000000..a828746e8a2f --- /dev/null +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -0,0 +1,121 @@ +/* + * atmel_ssc_dai.h - ALSA SSC interface for the Atmel SoC + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * ATMEL CORP. + * + * Based on at91-ssc.c by + * Frank Mandarino <fmandarino@endrelia.com> + * Based on pxa2xx Platform drivers by + * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef _ATMEL_SSC_DAI_H +#define _ATMEL_SSC_DAI_H + +#include <linux/types.h> +#include <linux/atmel-ssc.h> + +#include "atmel-pcm.h" + +/* SSC system clock ids */ +#define ATMEL_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */ + +/* SSC divider ids */ +#define ATMEL_SSC_CMR_DIV 0 /* MCK divider for BCLK */ +#define ATMEL_SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ +#define ATMEL_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ +/* + * SSC direction masks + */ +#define SSC_DIR_MASK_UNUSED 0 +#define SSC_DIR_MASK_PLAYBACK 1 +#define SSC_DIR_MASK_CAPTURE 2 + +/* + * SSC register values that Atmel left out of <linux/atmel-ssc.h>. These + * are expected to be used with SSC_BF + */ +/* START bit field values */ +#define SSC_START_CONTINUOUS 0 +#define SSC_START_TX_RX 1 +#define SSC_START_LOW_RF 2 +#define SSC_START_HIGH_RF 3 +#define SSC_START_FALLING_RF 4 +#define SSC_START_RISING_RF 5 +#define SSC_START_LEVEL_RF 6 +#define SSC_START_EDGE_RF 7 +#define SSS_START_COMPARE_0 8 + +/* CKI bit field values */ +#define SSC_CKI_FALLING 0 +#define SSC_CKI_RISING 1 + +/* CKO bit field values */ +#define SSC_CKO_NONE 0 +#define SSC_CKO_CONTINUOUS 1 +#define SSC_CKO_TRANSFER 2 + +/* CKS bit field values */ +#define SSC_CKS_DIV 0 +#define SSC_CKS_CLOCK 1 +#define SSC_CKS_PIN 2 + +/* FSEDGE bit field values */ +#define SSC_FSEDGE_POSITIVE 0 +#define SSC_FSEDGE_NEGATIVE 1 + +/* FSOS bit field values */ +#define SSC_FSOS_NONE 0 +#define SSC_FSOS_NEGATIVE 1 +#define SSC_FSOS_POSITIVE 2 +#define SSC_FSOS_LOW 3 +#define SSC_FSOS_HIGH 4 +#define SSC_FSOS_TOGGLE 5 + +#define START_DELAY 1 + +struct atmel_ssc_state { + u32 ssc_cmr; + u32 ssc_rcmr; + u32 ssc_rfmr; + u32 ssc_tcmr; + u32 ssc_tfmr; + u32 ssc_sr; + u32 ssc_imr; +}; + + +struct atmel_ssc_info { + char *name; + struct ssc_device *ssc; + spinlock_t lock; /* lock for dir_mask */ + unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ + unsigned short initialized; /* true if SSC has been initialized */ + unsigned short daifmt; + unsigned short cmr_div; + unsigned short tcmr_period; + unsigned short rcmr_period; + struct atmel_pcm_dma_params *dma_params[2]; + struct atmel_ssc_state ssc_state; +}; +extern struct snd_soc_dai atmel_ssc_dai[]; + +#endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 98a2d5826a85..d40b5a52a8d2 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -22,7 +22,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/errno.h> #include <linux/clk.h> @@ -40,8 +39,8 @@ #include <mach/portmux.h> #include "../codecs/wm8510.h" -#include "at32-pcm.h" -#include "at32-ssc.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" /*-------------------------------------------------------------------------*\ @@ -304,7 +303,7 @@ static const struct snd_soc_dapm_widget playpaq_dapm_widgets[] = { -static const char *intercon[][3] = { +static const struct snd_soc_dapm_route intercon[] = { /* speaker connected to SPKOUT */ {"Ext Spk", NULL, "SPKOUTP"}, {"Ext Spk", NULL, "SPKOUTN"}, @@ -312,9 +311,6 @@ static const char *intercon[][3] = { {"Mic Bias", NULL, "Int Mic"}, {"MICN", NULL, "Mic Bias"}, {"MICP", NULL, "Mic Bias"}, - - /* Terminator */ - {NULL, NULL, NULL}, }; @@ -334,11 +330,8 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) /* * Setup audio path interconnects */ - for (i = 0; intercon[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, - intercon[i][0], - intercon[i][1], intercon[i][2]); - } + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + /* always connected pins */ @@ -368,7 +361,7 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = { -static struct snd_soc_machine snd_soc_machine_playpaq = { +static struct snd_soc_card snd_soc_playpaq = { .name = "LRS_PlayPaq_WM8510", .dai_link = &playpaq_wm8510_dai, .num_links = 1, @@ -384,7 +377,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = { static struct snd_soc_device playpaq_wm8510_snd_devdata = { - .machine = &snd_soc_machine_playpaq, + .card = &snd_soc_playpaq, .platform = &at32_soc_platform, .codec_dev = &soc_codec_dev_wm8510, .codec_data = &playpaq_wm8510_setup, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c new file mode 100644 index 000000000000..fdc1d0206e0b --- /dev/null +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -0,0 +1,328 @@ +/* + * sam9g20_wm8731 -- SoC audio for AT91SAM9G20-based + * ATMEL AT91SAM9G20ek board. + * + * Copyright (C) 2005 SAN People + * Copyright (C) 2008 Atmel + * + * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com> + * + * Based on ati_b1_wm8731.c by: + * Frank Mandarino <fmandarino@endrelia.com> + * Copyright 2006 Endrelia Technologies Inc. + * Based on corgi.c by: + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/clk.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> + +#include <linux/atmel-ssc.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/hardware.h> +#include <mach/gpio.h> + +#include "../codecs/wm8731.h" +#include "atmel-pcm.h" +#include "atmel_ssc_dai.h" + + +static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret; + + /* codec system clock is supplied by PCK0, set to 12MHz */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); + + dev_dbg(rtd->socdev->dev, "shutdown"); +} + +static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct atmel_ssc_info *ssc_p = cpu_dai->private_data; + struct ssc_device *ssc = ssc_p->ssc; + int ret; + + unsigned int rate; + int cmr_div, period; + + if (ssc == NULL) { + printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); + return -EINVAL; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* + * The SSC clock dividers depend on the sample rate. The CMR.DIV + * field divides the system master clock MCK to drive the SSC TK + * signal which provides the codec BCLK. The TCMR.PERIOD and + * RCMR.PERIOD fields further divide the BCLK signal to drive + * the SSC TF and RF signals which provide the codec DACLRC and + * ADCLRC clocks. + * + * The dividers were determined through trial and error, where a + * CMR.DIV value is chosen such that the resulting BCLK value is + * divisible, or almost divisible, by (2 * sample rate), and then + * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. + */ + rate = params_rate(params); + + switch (rate) { + case 8000: + cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ + period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ + break; + case 11025: + cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ + period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ + break; + case 16000: + cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ + period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ + break; + case 22050: + cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ + period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ + break; + case 32000: + cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ + period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ + break; + case 44100: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ + break; + case 48000: + cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ + period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ + break; + case 88200: + cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ + period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ + break; + case 96000: + cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ + period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ + break; + default: + printk(KERN_WARNING "unsupported rate %d" + " on at91sam9g20ek board\n", rate); + return -EINVAL; + } + + /* set the MCK divider for BCLK */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set the BCLK divider for DACLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_TCMR_PERIOD, period); + } else { + /* set the BCLK divider for ADCLRC */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, + ATMEL_SSC_RCMR_PERIOD, period); + } + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops at91sam9g20ek_ops = { + .startup = at91sam9g20ek_startup, + .hw_params = at91sam9g20ek_hw_params, + .shutdown = at91sam9g20ek_shutdown, +}; + + +static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + + /* speaker connected to LHPOUT */ + {"Ext Spk", NULL, "LHPOUT"}, + + /* mic is connected to Mic Jack, with WM8731 Mic Bias */ + {"MICIN", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Int Mic"}, +}; + +/* + * Logic for a wm8731 as connected on a at91sam9g20ek board. + */ +static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) +{ + printk(KERN_DEBUG + "at91sam9g20ek_wm8731 " + ": at91sam9g20ek_wm8731_init() called\n"); + + /* Add specific widgets */ + snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, + ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); + /* Set up specific audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + /* not connected */ + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link at91sam9g20ek_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &atmel_ssc_dai[0], + .codec_dai = &wm8731_dai, + .init = at91sam9g20ek_wm8731_init, + .ops = &at91sam9g20ek_ops, +}; + +static struct snd_soc_card snd_soc_at91sam9g20ek = { + .name = "WM8731", + .dai_link = &at91sam9g20ek_dai, + .num_links = 1, +}; + +static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { + .i2c_bus = 0, + .i2c_address = 0x1b, +}; + +static struct snd_soc_device at91sam9g20ek_snd_devdata = { + .card = &snd_soc_at91sam9g20ek, + .platform = &atmel_soc_platform, + .codec_dev = &soc_codec_dev_wm8731, + .codec_data = &at91sam9g20ek_wm8731_setup, +}; + +static struct platform_device *at91sam9g20ek_snd_device; + +static int __init at91sam9g20ek_init(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc = NULL; + int ret; + + /* + * Request SSC device + */ + ssc = ssc_request(0); + if (IS_ERR(ssc)) { + ret = PTR_ERR(ssc); + ssc = NULL; + goto err_ssc; + } + ssc_p->ssc = ssc; + + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); + if (!at91sam9g20ek_snd_device) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + ret = -ENOMEM; + } + + platform_set_drvdata(at91sam9g20ek_snd_device, + &at91sam9g20ek_snd_devdata); + at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev; + + ret = platform_device_add(at91sam9g20ek_snd_device); + if (ret) { + printk(KERN_DEBUG + "platform device allocation failed\n"); + platform_device_put(at91sam9g20ek_snd_device); + } + + return ret; + +err_ssc: + return ret; +} + +static void __exit at91sam9g20ek_exit(void) +{ + struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; + struct ssc_device *ssc; + + if (ssc_p != NULL) { + ssc = ssc_p->ssc; + if (ssc != NULL) + ssc_free(ssc); + ssc_p->ssc = NULL; + } + + platform_device_unregister(at91sam9g20ek_snd_device); + at91sam9g20ek_snd_device = NULL; +} + +module_init(at91sam9g20ek_init); +module_exit(at91sam9g20ek_exit); + +/* Module information */ +MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>"); +MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 57facbad6825..a0bcfeaf5f86 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; @@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, } static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; @@ -345,7 +346,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = au1xpsc_ac97_probe, .remove = au1xpsc_ac97_remove, .suspend = au1xpsc_ac97_suspend, diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 9384702c7ebd..f4217e70a787 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -116,7 +116,8 @@ out: } static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; @@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) return 0; } -static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; int ret, stype = SUBSTREAM_TYPE(substream); @@ -369,7 +371,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", - .type = SND_SOC_DAI_I2S, .probe = au1xpsc_i2s_probe, .remove = au1xpsc_i2s_remove, .suspend = au1xpsc_i2s_suspend, @@ -389,8 +390,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, - }, - .dai_ops = { .set_fmt = au1xpsc_i2s_set_fmt, }, }; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c index f75ae7f62c3d..27683eb7905e 100644 --- a/sound/soc/au1x/sample-ac97.c +++ b/sound/soc/au1x/sample-ac97.c @@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { .ops = NULL, }; -static struct snd_soc_machine au1xpsc_sample_ac97_machine = { +static struct snd_soc_card au1xpsc_sample_ac97_machine = { .name = "Au1xxx PSC AC97 Audio", .dai_link = &au1xpsc_sample_ac97_dai, .num_links = 1, }; static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .machine = &au1xpsc_sample_ac97_machine, + .card = &au1xpsc_sample_ac97_machine, .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index f98331d099e7..e162cbbd3f3b 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -13,10 +13,25 @@ config SND_BF5XX_SOC_SSM2602 select SND_BF5XX_SOC_I2S select SND_SOC_SSM2602 select I2C - select I2C_BLACKFIN_TWI help Say Y if you want to add support for SoC audio on BF527-EZKIT. +config SND_BF5XX_SOC_AD73311 + tristate "SoC AD73311 Audio support for Blackfin" + depends on SND_BF5XX_I2S + select SND_BF5XX_SOC_I2S + select SND_SOC_AD73311 + help + Say Y if you want to add support for AD73311 codec on Blackfin. + +config SND_BFIN_AD73311_SE + int "PF pin for AD73311L Chip Select" + depends on SND_BF5XX_SOC_AD73311 + default 4 + help + Enter the GPIO used to control AD73311's SE pin. Acceptable + values are 0 to 7 + config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" depends on BLACKFIN && SND_SOC @@ -31,7 +46,7 @@ config SND_BF5XX_AC97 properly with this driver. This driver is known to work with the Analog Devices line of AC97 codecs. -config SND_MMAP_SUPPORT +config SND_BF5XX_MMAP_SUPPORT bool "Enable MMAP Support" depends on SND_BF5XX_AC97 default y @@ -39,9 +54,17 @@ config SND_MMAP_SUPPORT Say y if you want AC97 driver to support mmap mode. We introduce an intermediate buffer to simulate mmap. +config SND_BF5XX_MULTICHAN_SUPPORT + bool "Enable Multichannel Support" + depends on SND_BF5XX_AC97 + default n + help + Say y if you want AC97 driver to support up to 5.1 channel audio. + this mode will consume much more memory for DMA. + config SND_BF5XX_SOC_SPORT tristate - + config SND_BF5XX_SOC_I2S tristate select SND_BF5XX_SOC_SPORT @@ -64,7 +87,7 @@ config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" depends on (SND_BF5XX_I2S || SND_BF5XX_AC97) range 0 3 if BF54x - range 0 1 if (BF53x || BF561) + range 0 1 if !BF54x default 0 help Set the correct SPORT for sound chip. @@ -74,12 +97,13 @@ config SND_BF5XX_HAVE_COLD_RESET depends on SND_BF5XX_AC97 default y if BFIN548_EZKIT default n if !BFIN548_EZKIT - + config SND_BF5XX_RESET_GPIO_NUM int "Set a GPIO for cold reset" depends on SND_BF5XX_HAVE_COLD_RESET range 0 159 default 19 if BFIN548_EZKIT default 5 if BFIN537_STAMP + default 0 help Set the correct GPIO for RESET the sound chip. diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 9ea8bd9e0ba3..97bb37a6359c 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o # Blackfin Machine Support snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o - +snd-ad73311-objs := bf5xx-ad73311.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o +obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 51f4907c4831..d3d51bcb4569 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -43,23 +43,34 @@ #include "bf5xx-ac97.h" #include "bf5xx-sport.h" -#if defined(CONFIG_SND_MMAP_SUPPORT) +static unsigned int ac97_chan_mask[] = { + SP_FL, /* Mono */ + SP_STEREO, /* Stereo */ + SP_2DOT1, /* 2.1*/ + SP_QUAD,/*Quadraquic*/ + SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */ + SP_5DOT1, /* 5.1 */ +}; + +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, snd_pcm_uframes_t count) { struct snd_pcm_runtime *runtime = substream->runtime; struct sport_device *sport = runtime->private_data; + unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - bf5xx_pcm_to_ac97( - (struct ac97_frame *)sport->tx_dma_buf + sport->tx_pos, - (__u32 *)runtime->dma_area + sport->tx_pos, count); + bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf + + sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos * + runtime->channels, count, chan_mask); sport->tx_pos += runtime->period_size; if (sport->tx_pos >= runtime->buffer_size) sport->tx_pos %= runtime->buffer_size; + sport->tx_delay_pos = sport->tx_pos; } else { - bf5xx_ac97_to_pcm( - (struct ac97_frame *)sport->rx_dma_buf + sport->rx_pos, - (__u32 *)runtime->dma_area + sport->rx_pos, count); + bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf + + sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos * + runtime->channels, count); sport->rx_pos += runtime->period_size; if (sport->rx_pos >= runtime->buffer_size) sport->rx_pos %= runtime->buffer_size; @@ -70,9 +81,17 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, static void bf5xx_dma_irq(void *data) { struct snd_pcm_substream *pcm = data; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) struct snd_pcm_runtime *runtime = pcm->runtime; + struct sport_device *sport = runtime->private_data; bf5xx_mmap_copy(pcm, runtime->period_size); + if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sport->once == 0) { + snd_pcm_period_elapsed(pcm); + bf5xx_mmap_copy(pcm, runtime->period_size); + sport->once = 1; + } + } #endif snd_pcm_period_elapsed(pcm); } @@ -81,17 +100,14 @@ static void bf5xx_dma_irq(void *data) * The total rx/tx buffer is for ac97 frame to hold all pcm data * is 0x20000 * sizeof(struct ac97_frame) / 4. */ -#ifdef CONFIG_SND_MMAP_SUPPORT static const struct snd_pcm_hardware bf5xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER, -#else -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER, #endif + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 0x10000, @@ -114,6 +130,20 @@ static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) { +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport->once = 0; + if (runtime->dma_area) + memset(runtime->dma_area, 0, runtime->buffer_size); + memset(sport->tx_dma_buf, 0, runtime->buffer_size * + sizeof(struct ac97_frame)); + } else + memset(sport->rx_dma_buf, 0, runtime->buffer_size * + sizeof(struct ac97_frame)); +#endif snd_pcm_lib_free_pages(substream); return 0; } @@ -126,17 +156,12 @@ static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) /* An intermediate buffer is introduced for implementing mmap for * SPORT working in TMD mode(include AC97). */ -#if defined(CONFIG_SND_MMAP_SUPPORT) - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - /*clean up intermediate buffer*/ +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - memset(sport->tx_dma_buf, 0, size); sport_set_tx_callback(sport, bf5xx_dma_irq, substream); sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); } else { - memset(sport->rx_dma_buf, 0, size); sport_set_rx_callback(sport, bf5xx_dma_irq, substream); sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, runtime->period_size * sizeof(struct ac97_frame)); @@ -164,21 +189,25 @@ static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) pr_debug("%s enter\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) + bf5xx_mmap_copy(substream, runtime->period_size); + sport->tx_delay_pos = 0; +#endif sport_tx_start(sport); - else + } else sport_rx_start(sport); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) sport->tx_pos = 0; #endif sport_tx_stop(sport); } else { -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) sport->rx_pos = 0; #endif sport_rx_stop(sport); @@ -196,9 +225,9 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) struct sport_device *sport = runtime->private_data; unsigned int curr; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_pos; + curr = sport->tx_delay_pos; else curr = sport->rx_pos; #else @@ -237,7 +266,7 @@ static int bf5xx_pcm_open(struct snd_pcm_substream *substream) return ret; } -#ifdef CONFIG_SND_MMAP_SUPPORT +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -254,18 +283,16 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, void __user *buf, snd_pcm_uframes_t count) { struct snd_pcm_runtime *runtime = substream->runtime; - + unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; pr_debug("%s copy pos:0x%lx count:0x%lx\n", substream->stream ? "Capture" : "Playback", pos, count); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bf5xx_pcm_to_ac97( - (struct ac97_frame *)runtime->dma_area + pos, - buf, count); + bf5xx_pcm_to_ac97((struct ac97_frame *)runtime->dma_area + pos, + (__u16 *)buf, count, chan_mask); else - bf5xx_ac97_to_pcm( - (struct ac97_frame *)runtime->dma_area + pos, - buf, count); + bf5xx_ac97_to_pcm((struct ac97_frame *)runtime->dma_area + pos, + (__u16 *)buf, count); return 0; } #endif @@ -278,7 +305,7 @@ struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .prepare = bf5xx_pcm_prepare, .trigger = bf5xx_pcm_trigger, .pointer = bf5xx_pcm_pointer, -#ifdef CONFIG_SND_MMAP_SUPPORT +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) .mmap = bf5xx_pcm_mmap, #else .copy = bf5xx_pcm_copy, @@ -316,7 +343,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) * Need to allocate local buffer when enable * MMAP for SPORT working in TMD mode (include AC97). */ -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!sport_handle->tx_dma_buf) { sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ @@ -353,7 +380,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; int stream; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) size_t size = bf5xx_pcm_hardware.buffer_bytes_max * sizeof(struct ac97_frame) / 4; #endif @@ -367,7 +394,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) continue; dma_free_coherent(NULL, buf->bytes, buf->area, 0); buf->area = NULL; -#if defined(CONFIG_SND_MMAP_SUPPORT) +#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (sport_handle->tx_dma_buf) dma_free_coherent(NULL, size, \ diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index c782e311fd56..709bdf08e398 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -54,71 +54,103 @@ static int *cmd_count; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; -#if defined(CONFIG_BF54x) +static u16 sport_req[][7] = { + PIN_REQ_SPORT_0, +#ifdef PIN_REQ_SPORT_1 + PIN_REQ_SPORT_1, +#endif +#ifdef PIN_REQ_SPORT_2 + PIN_REQ_SPORT_2, +#endif +#ifdef PIN_REQ_SPORT_3 + PIN_REQ_SPORT_3, +#endif + }; + static struct sport_param sport_params[4] = { { .dma_rx_chan = CH_SPORT0_RX, .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERR, + .err_irq = IRQ_SPORT0_ERROR, .regs = (struct sport_register *)SPORT0_TCR1, }, +#ifdef PIN_REQ_SPORT_1 { .dma_rx_chan = CH_SPORT1_RX, .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERR, + .err_irq = IRQ_SPORT1_ERROR, .regs = (struct sport_register *)SPORT1_TCR1, }, +#endif +#ifdef PIN_REQ_SPORT_2 { .dma_rx_chan = CH_SPORT2_RX, .dma_tx_chan = CH_SPORT2_TX, - .err_irq = IRQ_SPORT2_ERR, + .err_irq = IRQ_SPORT2_ERROR, .regs = (struct sport_register *)SPORT2_TCR1, }, +#endif +#ifdef PIN_REQ_SPORT_3 { .dma_rx_chan = CH_SPORT3_RX, .dma_tx_chan = CH_SPORT3_TX, - .err_irq = IRQ_SPORT3_ERR, + .err_irq = IRQ_SPORT3_ERROR, .regs = (struct sport_register *)SPORT3_TCR1, } -}; -#else -static struct sport_param sport_params[2] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - } -}; #endif +}; -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \ - size_t count) +void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, + size_t count, unsigned int chan_mask) { while (count--) { - dst->ac97_tag = TAG_VALID | TAG_PCM; - (dst++)->ac97_pcm = *src++; + dst->ac97_tag = TAG_VALID; + if (chan_mask & SP_FL) { + dst->ac97_pcm_r = *src++; + dst->ac97_tag |= TAG_PCM_RIGHT; + } + if (chan_mask & SP_FR) { + dst->ac97_pcm_l = *src++; + dst->ac97_tag |= TAG_PCM_LEFT; + + } +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + if (chan_mask & SP_SR) { + dst->ac97_sl = *src++; + dst->ac97_tag |= TAG_PCM_SL; + } + if (chan_mask & SP_SL) { + dst->ac97_sr = *src++; + dst->ac97_tag |= TAG_PCM_SR; + } + if (chan_mask & SP_LFE) { + dst->ac97_lfe = *src++; + dst->ac97_tag |= TAG_PCM_LFE; + } + if (chan_mask & SP_FC) { + dst->ac97_center = *src++; + dst->ac97_tag |= TAG_PCM_CENTER; + } +#endif + dst++; } } EXPORT_SYMBOL(bf5xx_pcm_to_ac97); -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \ +void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, size_t count) { - while (count--) - *(dst++) = (src++)->ac97_pcm; + while (count--) { + *(dst++) = src->ac97_pcm_l; + *(dst++) = src->ac97_pcm_r; + src++; + } } EXPORT_SYMBOL(bf5xx_ac97_to_pcm); static unsigned int sport_tx_curr_frag(struct sport_device *sport) { - return sport->tx_curr_frag = sport_curr_offset_tx(sport) / \ + return sport->tx_curr_frag = sport_curr_offset_tx(sport) / sport->tx_fragsize; } @@ -129,9 +161,8 @@ static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) struct ac97_frame *nextwrite; sport_incfrag(sport, &nextfrag, 1); - sport_incfrag(sport, &nextfrag, 1); - nextwrite = (struct ac97_frame *)(sport->tx_buf + \ + nextwrite = (struct ac97_frame *)(sport->tx_buf + nextfrag * sport->tx_fragsize); pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n", sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]); @@ -298,20 +329,15 @@ static int bf5xx_ac97_resume(struct platform_device *pdev, static int bf5xx_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - int ret; -#if defined(CONFIG_BF54x) - u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1, - PIN_REQ_SPORT_2, PIN_REQ_SPORT_3}; -#else - u16 sport_req[][7] = {PIN_REQ_SPORT_0, PIN_REQ_SPORT_1}; -#endif + int ret = 0; cmd_count = (int *)get_zeroed_page(GFP_KERNEL); if (cmd_count == NULL) return -ENOMEM; if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); - return -EFAULT; + ret = -EFAULT; + goto peripheral_err; } #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET @@ -319,54 +345,54 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, if (gpio_request(CONFIG_SND_BF5XX_RESET_GPIO_NUM, "SND_AD198x RESET")) { pr_err("Failed to request GPIO_%d for reset\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM); - peripheral_free_list(&sport_req[sport_num][0]); - return -1; + ret = -1; + goto gpio_err; } gpio_direction_output(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); #endif sport_handle = sport_init(&sport_params[sport_num], 2, \ sizeof(struct ac97_frame), NULL); if (!sport_handle) { - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -ENODEV; + ret = -ENODEV; + goto sport_err; } /*SPORT works in TDM mode to simulate AC97 transfers*/ ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); - kfree(sport_handle); - peripheral_free_list(&sport_req[sport_num][0]); -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif - return -EBUSY; + ret = -EBUSY; + goto sport_config_err; } + return 0; + +sport_config_err: + kfree(sport_handle); +sport_err: +#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET + gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); +#endif +gpio_err: + peripheral_free_list(&sport_req[sport_num][0]); +peripheral_err: + free_page((unsigned long)cmd_count); + cmd_count = NULL; + + return ret; } static void bf5xx_ac97_remove(struct platform_device *pdev, @@ -374,6 +400,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, { free_page((unsigned long)cmd_count); cmd_count = NULL; + peripheral_free_list(&sport_req[sport_num][0]); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif @@ -382,7 +409,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai bfin_ac97_dai = { .name = "bf5xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = bf5xx_ac97_probe, .remove = bf5xx_ac97_remove, .suspend = bf5xx_ac97_suspend, @@ -390,7 +417,11 @@ struct snd_soc_dai bfin_ac97_dai = { .playback = { .stream_name = "AC97 Playback", .channels_min = 2, +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + .channels_max = 6, +#else .channels_max = 2, +#endif .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 3f77cc558dc0..3f2a911fe0cb 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -16,21 +16,46 @@ struct ac97_frame { u16 ac97_tag; /* slot 0 */ u16 ac97_addr; /* slot 1 */ u16 ac97_data; /* slot 2 */ - u32 ac97_pcm; /* slot 3 and 4: left and right pcm data */ + u16 ac97_pcm_l; /*slot 3:front left*/ + u16 ac97_pcm_r; /*slot 4:front left*/ +#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) + u16 ac97_mdm_l1; + u16 ac97_center; /*slot 6:center*/ + u16 ac97_sl; /*slot 7:surround left*/ + u16 ac97_sr; /*slot 8:surround right*/ + u16 ac97_lfe; /*slot 9:lfe*/ +#endif } __attribute__ ((packed)); +/* Speaker location */ +#define SP_FL 0x0001 +#define SP_FR 0x0010 +#define SP_FC 0x0002 +#define SP_LFE 0x0020 +#define SP_SL 0x0004 +#define SP_SR 0x0040 + +#define SP_STEREO (SP_FL | SP_FR) +#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE) +#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR) +#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR) + #define TAG_VALID 0x8000 #define TAG_CMD 0x6000 #define TAG_PCM_LEFT 0x1000 #define TAG_PCM_RIGHT 0x0800 -#define TAG_PCM (TAG_PCM_LEFT | TAG_PCM_RIGHT) +#define TAG_PCM_MDM_L1 0x0400 +#define TAG_PCM_CENTER 0x0200 +#define TAG_PCM_SL 0x0100 +#define TAG_PCM_SR 0x0080 +#define TAG_PCM_LFE 0x0040 extern struct snd_soc_dai bfin_ac97_dai; -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \ - size_t count); +void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \ + size_t count, unsigned int chan_mask); -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \ +void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \ size_t count); #endif diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index 124425d22320..36c569a43ce1 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -43,7 +43,7 @@ #include "bf5xx-ac97-pcm.h" #include "bf5xx-ac97.h" -static struct snd_soc_machine bf5xx_board; +static struct snd_soc_card bf5xx_board; static int bf5xx_board_startup(struct snd_pcm_substream *substream) { @@ -67,14 +67,14 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .ops = &bf5xx_board_ops, }; -static struct snd_soc_machine bf5xx_board = { +static struct snd_soc_card bf5xx_board = { .name = "bf5xx-board", .dai_link = &bf5xx_board_dai, .num_links = 1, }; static struct snd_soc_device bf5xx_board_snd_devdata = { - .machine = &bf5xx_board, + .card = &bf5xx_board, .platform = &bf5xx_ac97_soc_platform, .codec_dev = &soc_codec_dev_ad1980, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c new file mode 100644 index 000000000000..57da14799375 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -0,0 +1,240 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad73311.c + * Author: Cliff Cai <Cliff.Cai@analog.com> + * + * Created: Thur Sep 25 2008 + * Description: Board driver for ad73311 sound chip + * + * Modified: + * Copyright 2008 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad73311.h" +#include "bf5xx-sport.h" +#include "bf5xx-i2s-pcm.h" +#include "bf5xx-i2s.h" + +#if CONFIG_SND_BF5XX_SPORT_NUM == 0 +#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT +#else +#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 +#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 +#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 +#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 +#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT +#endif + +#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE + +static struct snd_soc_card bf5xx_ad73311; + +static int snd_ad73311_startup(void) +{ + pr_debug("%s enter\n", __func__); + + /* Pull up SE pin on AD73311L */ + gpio_set_value(GPIO_SE, 1); + return 0; +} + +static int snd_ad73311_configure(void) +{ + unsigned short ctrl_regs[6]; + unsigned short status = 0; + int count = 0; + + /* DMCLK = MCLK = 16.384 MHz + * SCLK = DMCLK/8 = 2.048 MHz + * Sample Rate = DMCLK/2048 = 8 KHz + */ + ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ + REGB_SCDIV(0) | REGB_DIRATE(0); + ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ + REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; + ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ + REGD_IGS(2); + ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); + ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; + ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; + + local_irq_disable(); + snd_ad73311_startup(); + udelay(1); + + bfin_write_SPORT_TCR1(TFSR); + bfin_write_SPORT_TCR2(0xF); + SSYNC(); + + /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to + * FIFO before enable SPORT to transfer the data + */ + for (count = 0; count < 6; count++) + bfin_write_SPORT_TX16(ctrl_regs[count]); + SSYNC(); + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); + SSYNC(); + + /* When TUVF is set, the data is already send out */ + while (!(status & TUVF) && count++ < 10000) { + udelay(1); + status = bfin_read_SPORT_STAT(); + SSYNC(); + } + bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); + SSYNC(); + local_irq_enable(); + + if (count == 10000) { + printk(KERN_ERR "ad73311: failed to configure codec\n"); + return -1; + } + return 0; +} + +static int bf5xx_probe(struct platform_device *pdev) +{ + int err; + if (gpio_request(GPIO_SE, "AD73311_SE")) { + printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); + return -EBUSY; + } + + gpio_direction_output(GPIO_SE, 0); + + err = snd_ad73311_configure(); + if (err < 0) + return -EFAULT; + + return 0; +} + +static int bf5xx_ad73311_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + pr_debug("%s enter\n", __func__); + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + pr_debug("%s rate %d format %x\n", __func__, params_rate(params), + params_format(params)); + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + + +static struct snd_soc_ops bf5xx_ad73311_ops = { + .startup = bf5xx_ad73311_startup, + .hw_params = bf5xx_ad73311_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad73311_dai = { + .name = "ad73311", + .stream_name = "AD73311", + .cpu_dai = &bf5xx_i2s_dai, + .codec_dai = &ad73311_dai, + .ops = &bf5xx_ad73311_ops, +}; + +static struct snd_soc_card bf5xx_ad73311 = { + .name = "bf5xx_ad73311", + .probe = bf5xx_probe, + .dai_link = &bf5xx_ad73311_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad73311_snd_devdata = { + .card = &bf5xx_ad73311, + .platform = &bf5xx_i2s_soc_platform, + .codec_dev = &soc_codec_dev_ad73311, +}; + +static struct platform_device *bf52x_ad73311_snd_device; + +static int __init bf5xx_ad73311_init(void) +{ + int ret; + + pr_debug("%s enter\n", __func__); + bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf52x_ad73311_snd_device) + return -ENOMEM; + + platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; + ret = platform_device_add(bf52x_ad73311_snd_device); + + if (ret) + platform_device_put(bf52x_ad73311_snd_device); + + return ret; +} + +static void __exit bf5xx_ad73311_exit(void) +{ + pr_debug("%s enter\n", __func__); + platform_device_unregister(bf52x_ad73311_snd_device); +} + +module_init(bf5xx_ad73311_init); +module_exit(bf5xx_ad73311_exit); + +/* Module information */ +MODULE_AUTHOR("Cliff Cai"); +MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 43a4092eeb89..6e5036bf9245 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,6 +70,25 @@ static struct sport_param sport_params[2] = { } }; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; + static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -78,28 +97,34 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* interface format:support I2S,slave mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + bf5xx_i2s.tcr1 |= TFSR | TCKFE; + bf5xx_i2s.rcr1 |= RFSR | RCKFE; + bf5xx_i2s.tcr2 |= TSFSE; + bf5xx_i2s.rcr2 |= RSFSE; + break; + case SND_SOC_DAIFMT_DSP_A: + bf5xx_i2s.tcr1 |= TFSR; + bf5xx_i2s.rcr1 |= RFSR; break; case SND_SOC_DAIFMT_LEFT_J: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } @@ -107,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) +static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); @@ -117,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) } static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { int ret = 0; @@ -127,14 +154,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: bf5xx_i2s.tcr2 |= 15; bf5xx_i2s.rcr2 |= 15; + sport_handle->wdsize = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bf5xx_i2s.tcr2 |= 23; bf5xx_i2s.rcr2 |= 23; + sport_handle->wdsize = 3; break; case SNDRV_PCM_FORMAT_S32_LE: bf5xx_i2s.tcr2 |= 31; bf5xx_i2s.rcr2 |= 31; + sport_handle->wdsize = 4; break; } @@ -145,17 +175,17 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, * need to configure both of them at the time when the first * stream is opened. * - * CPU DAI format:I2S, slave mode. + * CPU DAI:slave mode. */ - ret = sport_config_rx(sport_handle, RFSR | RCKFE, - RSFSE|bf5xx_i2s.rcr2, 0, 0); + ret = sport_config_rx(sport_handle, bf5xx_i2s.rcr1, + bf5xx_i2s.rcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, - TSFSE|bf5xx_i2s.tcr2, 0, 0); + ret = sport_config_tx(sport_handle, bf5xx_i2s.tcr1, + bf5xx_i2s.tcr2, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; @@ -165,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); bf5xx_i2s.counter--; @@ -174,13 +205,6 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) static int bf5xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, - }; - pr_debug("%s enter\n", __func__); if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); @@ -198,6 +222,13 @@ static int bf5xx_i2s_probe(struct platform_device *pdev, return 0; } +static void bf5xx_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("%s enter\n", __func__); + peripheral_free_list(&sport_req[sport_num][0]); +} + #ifdef CONFIG_PM static int bf5xx_i2s_suspend(struct platform_device *dev, struct snd_soc_dai *dai) @@ -261,25 +292,24 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, + .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, .resume = bf5xx_i2s_resume, .playback = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .capture = { - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, .ops = { .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, }, }; diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 4c163454bbf8..2e63dea73e9c 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -116,13 +116,15 @@ struct sport_device { void *err_data; unsigned char *tx_dma_buf; unsigned char *rx_dma_buf; -#ifdef CONFIG_SND_MMAP_SUPPORT +#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT dma_addr_t tx_dma_phy; dma_addr_t rx_dma_phy; int tx_pos;/*pcm sample count*/ int rx_pos; unsigned int tx_buffer_size; unsigned int rx_buffer_size; + int tx_delay_pos; + int once; #endif void *private_data; }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index e15f67fd7769..0078dfcd95b9 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -44,7 +44,7 @@ #include "bf5xx-i2s-pcm.h" #include "bf5xx-i2s.h" -static struct snd_soc_machine bf5xx_ssm2602; +static struct snd_soc_card bf5xx_ssm2602; static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream) { @@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; - ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -135,14 +135,14 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = { .i2c_address = 0x1b, }; -static struct snd_soc_machine bf5xx_ssm2602 = { +static struct snd_soc_card bf5xx_ssm2602 = { .name = "bf5xx_ssm2602", .dai_link = &bf5xx_ssm2602_dai, .num_links = 1, }; static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { - .machine = &bf5xx_ssm2602, + .card = &bf5xx_ssm2602, .platform = &bf5xx_i2s_soc_platform, .codec_dev = &soc_codec_dev_ssm2602, .codec_data = &bf5xx_ssm2602_setup, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e0b9869df0f1..bf68052d6924 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,29 +1,39 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" - depends on I2C - select SPI - select SPI_MASTER - select SND_SOC_AK4535 - select SND_SOC_CS4270 - select SND_SOC_SSM2602 - select SND_SOC_TLV320AIC26 - select SND_SOC_TLV320AIC3X - select SND_SOC_UDA1380 - select SND_SOC_WM8510 - select SND_SOC_WM8580 - select SND_SOC_WM8731 - select SND_SOC_WM8750 - select SND_SOC_WM8753 - select SND_SOC_WM8900 - select SND_SOC_WM8903 - select SND_SOC_WM8971 - select SND_SOC_WM8990 + select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_AD73311 if I2C + select SND_SOC_AK4535 if I2C + select SND_SOC_CS4270 if I2C + select SND_SOC_PCM3008 + select SND_SOC_SSM2602 if I2C + select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_UDA134X + select SND_SOC_UDA1380 if I2C + select SND_SOC_WM8510 if (I2C || SPI_MASTER) + select SND_SOC_WM8580 if I2C + select SND_SOC_WM8728 if (I2C || SPI_MASTER) + select SND_SOC_WM8731 if (I2C || SPI_MASTER) + select SND_SOC_WM8750 if (I2C || SPI_MASTER) + select SND_SOC_WM8753 if (I2C || SPI_MASTER) + select SND_SOC_WM8900 if I2C + select SND_SOC_WM8903 if I2C + select SND_SOC_WM8971 if I2C + select SND_SOC_WM8990 if I2C + select SND_SOC_WM9712 if SND_SOC_AC97_BUS + select SND_SOC_WM9713 if SND_SOC_AC97_BUS help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine driver. Selecting this option will allow these drivers to be built without an explicit machine driver for test and development purposes. + Support for the bus types used to access the codecs to be built must + be selected separately. + If unsure select "N". @@ -34,6 +44,9 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1980 tristate +config SND_SOC_AD73311 + tristate + config SND_SOC_AK4535 tristate @@ -55,17 +68,35 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_L3 + tristate + +config SND_SOC_PCM3008 + tristate + config SND_SOC_SSM2602 tristate +config SND_SOC_TLV320AIC23 + tristate + depends on I2C + config SND_SOC_TLV320AIC26 - tristate "TI TLV320AIC26 Codec support" - depends on SND_SOC && SPI + tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE + depends on SPI config SND_SOC_TLV320AIC3X tristate depends on I2C +config SND_SOC_TWL4030 + tristate + depends on TWL4030_CORE + +config SND_SOC_UDA134X + tristate + select SND_SOC_L3 + config SND_SOC_UDA1380 tristate @@ -75,6 +106,9 @@ config SND_SOC_WM8510 config SND_SOC_WM8580 tristate +config SND_SOC_WM8728 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f977978a3409..9a20fddd09c7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,13 +1,20 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1980-objs := ad1980.o +snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-l3-objs := l3.o +snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-twl4030-objs := twl4030.o +snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -20,14 +27,21 @@ snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o +obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 61fd96ca7bc7..c4208c8210c8 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -2,8 +2,7 @@ * ac97.c -- ALSA Soc AC97 codec support * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -25,7 +24,8 @@ #define AC97_VERSION "0.6" -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -43,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Playback", .channels_min = 1, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 4e09c1f2c063..a9a268112d3f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -13,7 +13,6 @@ #include <linux/init.h> #include <linux/module.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/device.h> #include <sound/core.h> @@ -86,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), + SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), @@ -143,10 +145,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, struct snd_soc_dai ad1980_dai = { .name = "AC97", + .ac97_control = 1, .playback = { .stream_name = "Playback", .channels_min = 2, - .channels_max = 2, + .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { @@ -193,6 +196,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; u16 vendor_id2; + u16 ext_status; printk(KERN_INFO "AD1980 SoC Audio Codec\n"); @@ -235,7 +239,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) ret = ad1980_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n"); goto reset_err; } @@ -254,9 +258,16 @@ static int ad1980_soc_probe(struct platform_device *pdev) "supported\n"); } - ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */ - ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */ - ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */ + /* unmute captures and playbacks volume */ + ac97_write(codec, AC97_MASTER, 0x0000); + ac97_write(codec, AC97_PCM, 0x0000); + ac97_write(codec, AC97_REC_GAIN, 0x0000); + ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); + ac97_write(codec, AC97_SURROUND_MASTER, 0x0000); + + /*power on LFE/CENTER/Surround DACs*/ + ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); ad1980_add_controls(codec); ret = snd_soc_register_card(socdev); diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c new file mode 100644 index 000000000000..59c4c8f18cbb --- /dev/null +++ b/sound/soc/codecs/ad73311.c @@ -0,0 +1,106 @@ +/* + * ad73311.c -- ALSA Soc AD73311 codec support + * + * Copyright: Analog Device Inc. + * Author: Cliff Cai <cliff.cai@analog.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 25th Sep 2008 Initial version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ad73311.h" + +struct snd_soc_dai ad73311_dai = { + .name = "AD73311", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, +}; +EXPORT_SYMBOL_GPL(ad73311_dai); + +static int ad73311_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "AD73311"; + codec->owner = THIS_MODULE; + codec->dai = &ad73311_dai; + codec->num_dai = 1; + socdev->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ad73311: failed to register card\n"); + goto register_err; + } + + return ret; + +register_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int ad73311_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad73311 = { + .probe = ad73311_soc_probe, + .remove = ad73311_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); + +MODULE_DESCRIPTION("ASoC ad73311 driver"); +MODULE_AUTHOR("Cliff Cai "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h new file mode 100644 index 000000000000..507ce0c30edf --- /dev/null +++ b/sound/soc/codecs/ad73311.h @@ -0,0 +1,90 @@ +/* + * File: sound/soc/codec/ad73311.h + * Based on: + * Author: Cliff Cai <cliff.cai@analog.com> + * + * Created: Thur Sep 25, 2008 + * Description: definitions for AD73311 registers + * + * + * Modified: + * Copyright 2006 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD73311_H__ +#define __AD73311_H__ + +#define AD_CONTROL 0x8000 +#define AD_DATA 0x0000 +#define AD_READ 0x4000 +#define AD_WRITE 0x0000 + +/* Control register A */ +#define CTRL_REG_A (0 << 8) + +#define REGA_MODE_PRO 0x00 +#define REGA_MODE_DATA 0x01 +#define REGA_MODE_MIXED 0x03 +#define REGA_DLB 0x04 +#define REGA_SLB 0x08 +#define REGA_DEVC(x) ((x & 0x7) << 4) +#define REGA_RESET 0x80 + +/* Control register B */ +#define CTRL_REG_B (1 << 8) + +#define REGB_DIRATE(x) (x & 0x3) +#define REGB_SCDIV(x) ((x & 0x3) << 2) +#define REGB_MCDIV(x) ((x & 0x7) << 4) +#define REGB_CEE (1 << 7) + +/* Control register C */ +#define CTRL_REG_C (2 << 8) + +#define REGC_PUDEV (1 << 0) +#define REGC_PUADC (1 << 3) +#define REGC_PUDAC (1 << 4) +#define REGC_PUREF (1 << 5) +#define REGC_REFUSE (1 << 6) + +/* Control register D */ +#define CTRL_REG_D (3 << 8) + +#define REGD_IGS(x) (x & 0x7) +#define REGD_RMOD (1 << 3) +#define REGD_OGS(x) ((x & 0x7) << 4) +#define REGD_MUTE (x << 7) + +/* Control register E */ +#define CTRL_REG_E (4 << 8) + +#define REGE_DA(x) (x & 0x1f) +#define REGE_IBYP (1 << 5) + +/* Control register F */ +#define CTRL_REG_F (5 << 8) + +#define REGF_SEEN (1 << 5) +#define REGF_INV (1 << 6) +#define REGF_ALB (1 << 7) + +extern struct snd_soc_dai ad73311_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad73311; +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 088cf9927720..c742290e5533 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -28,7 +28,6 @@ #include "ak4535.h" -#define AUDIO_NAME "ak4535" #define AK4535_VERSION "0.3" struct snd_soc_codec_device soc_codec_dev_ak4535; @@ -340,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int ak4535_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -452,8 +452,6 @@ struct snd_soc_dai ak4535_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .hw_params = ak4535_hw_params, - }, - .dai_ops = { .set_fmt = ak4535_set_dai_fmt, .digital_mute = ak4535_mute, .set_sysclk = ak4535_set_dai_sysclk, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0bbd94501d7e..7507d468b200 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, /* * Program the CS4270 with the given hardware parameters. * - * The .dai_ops functions are used to provide board-specific data, like + * The .ops functions are used to provide board-specific data, like * input frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ static int cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -450,6 +451,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Disable automatic volume control. It's enabled by default, and + * it causes volume change commands to be delayed, sometimes until + * after playback has started. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + printk(KERN_ERR "I2C write failed\n"); + return ret; + } + /* Thaw and power-up the codec */ ret = snd_soc_write(codec, CS4270_PWRCTL, 0); @@ -697,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.dai_ops.digital_mute = cs4270_mute; + cs4270_dai.ops.digital_mute = cs4270_mute; #endif } else printk(KERN_INFO "cs4270: no I2C device found, " diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c new file mode 100644 index 000000000000..5353af58862c --- /dev/null +++ b/sound/soc/codecs/l3.c @@ -0,0 +1,91 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin <chripell@evolware.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/delay.h> + +#include <sound/l3.h> + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct l3_pins *adap, unsigned int byte) +{ + int i; + + for (i = 0; i < 8; i++) { + adap->setclk(0); + udelay(adap->data_hold); + adap->setdat(byte & 1); + udelay(adap->data_setup); + adap->setclk(1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct l3_pins *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + adap->setmode(0); + udelay(adap->mode); + } + adap->setmode(1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) +{ + adap->setclk(1); + adap->setdat(1); + adap->setmode(1); + udelay(adap->mode); + + adap->setmode(0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + adap->setclk(1); + adap->setdat(1); + adap->setmode(0); + + return len; +} +EXPORT_SYMBOL_GPL(l3_write); + +MODULE_DESCRIPTION("L3 bit-banging driver"); +MODULE_AUTHOR("Christian Pellegrin <chripell@evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c new file mode 100644 index 000000000000..651a15eb8c2c --- /dev/null +++ b/sound/soc/codecs/pcm3008.c @@ -0,0 +1,200 @@ +/* + * ALSA Soc PCM3008 codec support + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on AC97 Soc codec, original copyright follow: + * Copyright 2005 Wolfson Microelectronics PLC. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Generic PCM3008 support. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "pcm3008.h" + +#define PCM3008_VERSION "0.2" + +#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +struct snd_soc_dai pcm3008_dai = { + .name = "PCM3008 HiFi", + .playback = { + .stream_name = "PCM3008 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM3008 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(pcm3008_dai); + +static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) +{ + gpio_free(setup->dem0_pin); + gpio_free(setup->dem1_pin); + gpio_free(setup->pdad_pin); + gpio_free(setup->pdda_pin); +} + +static int pcm3008_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + int ret = 0; + + printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->codec) + return -ENOMEM; + + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->name = "PCM3008"; + codec->owner = THIS_MODULE; + codec->dai = &pcm3008_dai; + codec->num_dai = 1; + codec->write = NULL; + codec->read = NULL; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* Register PCMs. */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to create pcms\n"); + goto pcm_err; + } + + /* Register Card. */ + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to register card\n"); + goto card_err; + } + + /* DEM1 DEM0 DE-EMPHASIS_MODE + * Low Low De-emphasis 44.1 kHz ON + * Low High De-emphasis OFF + * High Low De-emphasis 48 kHz ON + * High High De-emphasis 32 kHz ON + */ + + /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem0_pin, "codec_dem0"); + if (ret == 0) + ret = gpio_direction_output(setup->dem0_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ + ret = gpio_request(setup->dem1_pin, "codec_dem1"); + if (ret == 0) + ret = gpio_direction_output(setup->dem1_pin, 0); + if (ret != 0) + goto gpio_err; + + /* Configure PDAD GPIO. */ + ret = gpio_request(setup->pdad_pin, "codec_pdad"); + if (ret == 0) + ret = gpio_direction_output(setup->pdad_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure PDDA GPIO. */ + ret = gpio_request(setup->pdda_pin, "codec_pdda"); + if (ret == 0) + ret = gpio_direction_output(setup->pdda_pin, 1); + if (ret != 0) + goto gpio_err; + + return ret; + +gpio_err: + pcm3008_gpio_free(setup); +card_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + + return ret; +} + +static int pcm3008_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + + if (!codec) + return 0; + + pcm3008_gpio_free(setup); + snd_soc_free_pcms(socdev); + kfree(socdev->codec); + + return 0; +} + +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_pcm3008 = { + .probe = pcm3008_soc_probe, + .remove = pcm3008_soc_remove, + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); + +MODULE_DESCRIPTION("Soc PCM3008 driver"); +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h new file mode 100644 index 000000000000..d04e87d3c060 --- /dev/null +++ b/sound/soc/codecs/pcm3008.h @@ -0,0 +1,25 @@ +/* + * PCM3008 ALSA SoC Layer + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_SOC_PCM3008_H +#define __LINUX_SND_SOC_PCM3008_H + +struct pcm3008_setup_data { + unsigned dem0_pin; + unsigned dem1_pin; + unsigned pdad_pin; + unsigned pdda_pin; +}; + +extern struct snd_soc_codec_device soc_codec_dev_pcm3008; +extern struct snd_soc_dai pcm3008_dai; + +#endif diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 940ce1c3522e..0c5884ea1b00 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -42,7 +42,6 @@ #include "ssm2602.h" -#define AUDIO_NAME "ssm2602" #define SSM2602_VERSION "0.1" struct snd_soc_codec_device soc_codec_dev_ssm2602; @@ -286,16 +285,23 @@ static inline int get_coeff(int mclk, int rate) } static int ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -325,19 +331,26 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream) +static int ssm2602_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -355,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) return 0; } -static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -366,14 +380,21 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream) +static void ssm2602_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) @@ -497,6 +518,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -504,20 +528,18 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, - }, - .dai_ops = { .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c new file mode 100644 index 000000000000..a4e13d0688c9 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.c @@ -0,0 +1,852 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Notes: + * The AIC23 is a driver for a low power stereo audio + * codec tlv320aic23 + * + * The machine layer should disable unsupported inputs/outputs by + * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320aic23.h" + +#define AIC23_VERSION "0.1" + +/* + * AIC23 register cache + */ +static const u16 tlv320aic23_reg[] = { + 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ + 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ + 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +}; + +/* + * read tlv320aic23 register cache + */ +static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec + *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return -1; + return cache[reg]; +} + +/* + * write tlv320aic23 register cache + */ +static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u16 value) +{ + u16 *cache = codec->reg_cache; + if (reg >= ARRAY_SIZE(tlv320aic23_reg)) + return; + cache[reg] = value; +} + +/* + * write to the tlv320aic23 register space + */ +static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + + u8 data[2]; + + /* TLV320AIC23 has 7 bit address and 9 bits of data + * so we need to switch one data bit into reg and rest + * of data into val + */ + + if ((reg < 0 || reg > 9) && (reg != 15)) { + printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg); + return -1; + } + + data[0] = (reg << 1) | (value >> 8 & 0x01); + data[1] = value & 0xff; + + tlv320aic23_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + + printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__, + value, reg); + + return -EIO; +} + +static const char *rec_src_text[] = { "Line", "Mic" }; +static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; + +static const struct soc_enum rec_src_enum = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); + +static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = +SOC_DAPM_ENUM("Input Select", rec_src_enum); + +static const struct soc_enum tlv320aic23_rec_src = + SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static const struct soc_enum tlv320aic23_deemph = + SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); +static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0); + +static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val, reg; + + val = (ucontrol->value.integer.value[0] & 0x07); + + /* linear conversion to userspace + * 000 = -6db + * 001 = -9db + * 010 = -12db + * 011 = -18db (Min) + * 100 = 0db (Max) + */ + val = (val >= 4) ? 4 : (3 - val); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0); + tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6)); + + return 0; +} + +static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 val; + + val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0); + val = val >> 6; + val = (val >= 4) ? 4 : (3 - val); + ucontrol->value.integer.value[0] = val; + return 0; + +} + +#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\ + .put = snd_soc_tlv320aic23_put_volsw, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + +static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL, + TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv), + SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1), + SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 7, 1, 0), + SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL, + TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv), + SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1), + SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0), + SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG, + 6, 4, 0, sidetone_vol_tlv), + SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), +}; + +/* add non dapm controls */ +static int tlv320aic23_add_controls(struct snd_soc_codec *codec) +{ + + int err, i; + + for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&tlv320aic23_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; + +} + +/* PGA Mixer controls for Line and Mic switch */ +static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { + SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), + SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1), + SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0, + &tlv320aic23_rec_src_mux_controls), + SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1, + &tlv320aic23_output_mixer_controls[0], + ARRAY_SIZE(tlv320aic23_output_mixer_controls)), + SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LHPOUT"), + SND_SOC_DAPM_OUTPUT("RHPOUT"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("LLINEIN"), + SND_SOC_DAPM_INPUT("RLINEIN"), + + SND_SOC_DAPM_INPUT("MICIN"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Output Mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "Playback Switch", "DAC"}, + {"Output Mixer", "Mic Sidetone Switch", "Mic Input"}, + + /* Outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + + /* Inputs */ + {"Line Input", "NULL", "LLINEIN"}, + {"Line Input", "NULL", "RLINEIN"}, + + {"Mic Input", "NULL", "MICIN"}, + + /* input mux */ + {"Capture Source", "Line", "Line Input"}, + {"Capture Source", "Mic", "Mic Input"}, + {"ADC", NULL, "Capture Source"}, + +}; + +/* AIC23 driver data */ +struct aic23 { + struct snd_soc_codec codec; + int mclk; + int requested_adc; + int requested_dac; +}; + +/* + * Common Crystals used + * 11.2896 Mhz /128 = *88.2k /192 = 58.8k + * 12.0000 Mhz /125 = *96k /136 = 88.235K + * 12.2880 Mhz /128 = *96k /192 = 64k + * 16.9344 Mhz /128 = 132.3k /192 = *88.2k + * 18.4320 Mhz /128 = 144k /192 = *96k + */ + +/* + * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 + * USB BOSR 0-250/2 = 125, 1-272/2 = 136 + */ +static const int bosr_usb_divisor_table[] = { + 128, 125, 192, 136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ + LOWER_GROUP, /* Usb, bosr - 0*/ + UPPER_GROUP, /* Usb, bosr - 1*/ +}; +/* + * Every divisor is a factor of 11*12 + */ +#define SR_MULT (11*12) +#define A(x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { + A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), + A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) +}; +static const unsigned char sr_dac_mult_table[] = { + A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), + A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) +}; + +static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, + int dac, int dac_l, int dac_h, int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { + int diff_adc = need_adc - adc; + int diff_dac = need_dac - dac; + return abs(diff_adc) + abs(diff_dac); + } + return UINT_MAX; +} + +static int find_rate(int mclk, u32 need_adc, u32 need_dac) +{ + int i, j; + int best_i = -1; + int best_j = -1; + int best_div = 0; + unsigned best_score = UINT_MAX; + int adc_l, adc_h, dac_l, dac_h; + + need_adc *= SR_MULT; + need_dac *= SR_MULT; + /* + * rates given are +/- 1/32 + */ + adc_l = need_adc - (need_adc >> 5); + adc_h = need_adc + (need_adc >> 5); + dac_l = need_dac - (need_dac >> 5); + dac_h = need_dac + (need_dac >> 5); + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { + int base = mclk / bosr_usb_divisor_table[i]; + int mask = sr_valid_mask[i]; + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { + int adc; + int dac; + int score; + if ((mask & 1) == 0) + continue; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score = get_score(adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); + if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div = 0; + } + score = get_score((adc >> 1), adc_l, adc_h, need_adc, + (dac >> 1), dac_l, dac_h, need_dac); + /* prefer to have a /2 */ + if ((score != UINT_MAX) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div = 1; + } + } + } + return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); +} + +#ifdef DEBUG +static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, + u32 *sample_rate_adc, u32 *sample_rate_dac) +{ + int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int sr = (src >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[src & 3]); + int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; + int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; + if (src & TLV320AIC23_CLKIN_HALF) { + adc >>= 1; + dac >>= 1; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} +#endif + +static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, + u32 sample_rate_adc, u32 sample_rate_dac) +{ + /* Search for the right sample rate */ + int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); + if (data < 0) { + printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", + __func__, sample_rate_adc, sample_rate_dac); + return -EINVAL; + } + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); +#ifdef DEBUG + { + u32 adc, dac; + get_current_sample_rates(codec, mclk, &adc, &dac); + printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", + adc, dac, data); + } +#endif + return 0; +} + +static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 iface_reg; + int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + u32 sample_rate_adc = aic23->requested_adc; + u32 sample_rate_dac = aic23->requested_dac; + u32 sample_rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aic23->requested_dac = sample_rate_dac = sample_rate; + if (!sample_rate_adc) + sample_rate_adc = sample_rate; + } else { + aic23->requested_adc = sample_rate_adc = sample_rate; + if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); + if (ret < 0) + return ret; + + iface_reg = + tlv320aic23_read_reg_cache(codec, + TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_reg |= (0x01 << 2); + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_reg |= (0x02 << 2); + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_reg |= (0x03 << 2); + break; + } + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + + /* set active */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); + + return 0; +} + +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + + /* deactivate */ + if (!codec->active) { + udelay(50); + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aic23->requested_dac = 0; + else + aic23->requested_adc = 0; +} + +static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT); + if (mute) + reg |= TLV320AIC23_DACM_MUTE; + + else + reg &= ~TLV320AIC23_DACM_MUTE; + + tlv320aic23_write(codec, TLV320AIC23_DIGT, reg); + + return 0; +} + +static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface_reg; + + iface_reg = + tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg |= TLV320AIC23_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface_reg |= TLV320AIC23_FOR_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_FOR_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg |= TLV320AIC23_FOR_LJUST; + break; + default: + return -EINVAL; + + } + + tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg); + + return 0; +} + +static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; + return 0; +} + +static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + /* vref/mid, osc on, dac unmute */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC23_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai tlv320aic23_dai = { + .name = "tlv320aic23", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AIC23_RATES, + .formats = AIC23_FORMATS,}, + .ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } +}; +EXPORT_SYMBOL_GPL(tlv320aic23_dai); + +static int tlv320aic23_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int tlv320aic23_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + int i; + u16 reg; + + /* Sync reg_cache with the hardware */ + for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) { + u16 val = tlv320aic23_read_reg_cache(codec, reg); + tlv320aic23_write(codec, reg, val); + } + + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + tlv320aic23_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the AIC23 driver + * register the mixer and dsp interfaces with the kernel + */ +static int tlv320aic23_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + u16 reg; + + codec->name = "tlv320aic23"; + codec->owner = THIS_MODULE; + codec->read = tlv320aic23_read_reg_cache; + codec->write = tlv320aic23_write; + codec->set_bias_level = tlv320aic23_set_bias_level; + codec->dai = &tlv320aic23_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg); + codec->reg_cache = + kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset codec */ + tlv320aic23_write(codec, TLV320AIC23_RESET, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K); + + /* Unmute input */ + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL); + tlv320aic23_write(codec, TLV320AIC23_LINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + (TLV320AIC23_LRS_ENABLED)); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL); + tlv320aic23_write(codec, TLV320AIC23_RINVOL, + (reg & (~TLV320AIC23_LIM_MUTED)) | + TLV320AIC23_LRS_ENABLED); + + reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG); + tlv320aic23_write(codec, TLV320AIC23_ANLG, + (reg) & (~TLV320AIC23_BYPASS_ON) & + (~TLV320AIC23_MICM_MUTED)); + + /* Default output volume */ + tlv320aic23_write(codec, TLV320AIC23_LCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + tlv320aic23_write(codec, TLV320AIC23_RCHNVOL, + TLV320AIC23_DEFAULT_OUT_VOL & + TLV320AIC23_OUT_VOL_MASK); + + tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); + + tlv320aic23_add_controls(codec); + tlv320aic23_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} +static struct snd_soc_device *tlv320aic23_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +/* + * If the i2c layer weren't so broken, we could pass this kind of data + * around + */ +static int tlv320aic23_codec_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct snd_soc_device *socdev = tlv320aic23_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = tlv320aic23_init(socdev); + if (ret < 0) { + printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + put_device(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23", + }, + .probe = tlv320aic23_codec_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +#endif + +static int tlv320aic23_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct aic23 *aic23; + int ret = 0; + + printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); + + aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + if (aic23 == NULL) + return -ENOMEM; + codec = &aic23->codec; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + tlv320aic23_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = NULL; + ret = i2c_add_driver(&tlv320aic23_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); +#endif + return ret; +} + +static int tlv320aic23_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + + if (codec->control_data) + tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&tlv320aic23_i2c_driver); +#endif + kfree(codec->reg_cache); + kfree(aic23); + + return 0; +} +struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { + .probe = tlv320aic23_probe, + .remove = tlv320aic23_remove, + .suspend = tlv320aic23_suspend, + .resume = tlv320aic23_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h new file mode 100644 index 000000000000..79d1faf8e570 --- /dev/null +++ b/sound/soc/codecs/tlv320aic23.h @@ -0,0 +1,122 @@ +/* + * ALSA SoC TLV320AIC23 codec driver + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _TLV320AIC23_H +#define _TLV320AIC23_H + +/* Codec TLV320AIC23 */ +#define TLV320AIC23_LINVOL 0x00 +#define TLV320AIC23_RINVOL 0x01 +#define TLV320AIC23_LCHNVOL 0x02 +#define TLV320AIC23_RCHNVOL 0x03 +#define TLV320AIC23_ANLG 0x04 +#define TLV320AIC23_DIGT 0x05 +#define TLV320AIC23_PWR 0x06 +#define TLV320AIC23_DIGT_FMT 0x07 +#define TLV320AIC23_SRATE 0x08 +#define TLV320AIC23_ACTIVE 0x09 +#define TLV320AIC23_RESET 0x0F + +/* Left (right) line input volume control register */ +#define TLV320AIC23_LRS_ENABLED 0x0100 +#define TLV320AIC23_LIM_MUTED 0x0080 +#define TLV320AIC23_LIV_DEFAULT 0x0017 +#define TLV320AIC23_LIV_MAX 0x001f +#define TLV320AIC23_LIV_MIN 0x0000 + +/* Left (right) channel headphone volume control register */ +#define TLV320AIC23_LZC_ON 0x0080 +#define TLV320AIC23_LHV_DEFAULT 0x0079 +#define TLV320AIC23_LHV_MAX 0x007f +#define TLV320AIC23_LHV_MIN 0x0000 + +/* Analog audio path control register */ +#define TLV320AIC23_STA_REG(x) ((x)<<6) +#define TLV320AIC23_STE_ENABLED 0x0020 +#define TLV320AIC23_DAC_SELECTED 0x0010 +#define TLV320AIC23_BYPASS_ON 0x0008 +#define TLV320AIC23_INSEL_MIC 0x0004 +#define TLV320AIC23_MICM_MUTED 0x0002 +#define TLV320AIC23_MICB_20DB 0x0001 + +/* Digital audio path control register */ +#define TLV320AIC23_DACM_MUTE 0x0008 +#define TLV320AIC23_DEEMP_32K 0x0002 +#define TLV320AIC23_DEEMP_44K 0x0004 +#define TLV320AIC23_DEEMP_48K 0x0006 +#define TLV320AIC23_ADCHP_ON 0x0001 + +/* Power control down register */ +#define TLV320AIC23_DEVICE_PWR_OFF 0x0080 +#define TLV320AIC23_CLK_OFF 0x0040 +#define TLV320AIC23_OSC_OFF 0x0020 +#define TLV320AIC23_OUT_OFF 0x0010 +#define TLV320AIC23_DAC_OFF 0x0008 +#define TLV320AIC23_ADC_OFF 0x0004 +#define TLV320AIC23_MIC_OFF 0x0002 +#define TLV320AIC23_LINE_OFF 0x0001 + +/* Digital audio interface register */ +#define TLV320AIC23_MS_MASTER 0x0040 +#define TLV320AIC23_LRSWAP_ON 0x0020 +#define TLV320AIC23_LRP_ON 0x0010 +#define TLV320AIC23_IWL_16 0x0000 +#define TLV320AIC23_IWL_20 0x0004 +#define TLV320AIC23_IWL_24 0x0008 +#define TLV320AIC23_IWL_32 0x000C +#define TLV320AIC23_FOR_I2S 0x0002 +#define TLV320AIC23_FOR_DSP 0x0003 +#define TLV320AIC23_FOR_LJUST 0x0001 + +/* Sample rate control register */ +#define TLV320AIC23_CLKOUT_HALF 0x0080 +#define TLV320AIC23_CLKIN_HALF 0x0040 +#define TLV320AIC23_BOSR_384fs 0x0002 /* BOSR_272fs in USB mode */ +#define TLV320AIC23_USB_CLK_ON 0x0001 +#define TLV320AIC23_SR_MASK 0xf +#define TLV320AIC23_CLKOUT_SHIFT 7 +#define TLV320AIC23_CLKIN_SHIFT 6 +#define TLV320AIC23_SR_SHIFT 2 +#define TLV320AIC23_BOSR_SHIFT 1 + +/* Digital interface register */ +#define TLV320AIC23_ACT_ON 0x0001 + +/* + * AUDIO related MACROS + */ + +#define TLV320AIC23_DEFAULT_OUT_VOL 0x70 +#define TLV320AIC23_DEFAULT_IN_VOLUME 0x10 + +#define TLV320AIC23_OUT_VOL_MIN TLV320AIC23_LHV_MIN +#define TLV320AIC23_OUT_VOL_MAX TLV320AIC23_LHV_MAX +#define TLV320AIC23_OUT_VO_RANGE (TLV320AIC23_OUT_VOL_MAX - \ + TLV320AIC23_OUT_VOL_MIN) +#define TLV320AIC23_OUT_VOL_MASK TLV320AIC23_OUT_VOL_MAX + +#define TLV320AIC23_IN_VOL_MIN TLV320AIC23_LIV_MIN +#define TLV320AIC23_IN_VOL_MAX TLV320AIC23_LIV_MAX +#define TLV320AIC23_IN_VOL_RANGE (TLV320AIC23_IN_VOL_MAX - \ + TLV320AIC23_IN_VOL_MIN) +#define TLV320AIC23_IN_VOL_MASK TLV320AIC23_IN_VOL_MAX + +#define TLV320AIC23_SIDETONE_MASK 0x1c0 +#define TLV320AIC23_SIDETONE_0 0x100 +#define TLV320AIC23_SIDETONE_6 0x000 +#define TLV320AIC23_SIDETONE_9 0x040 +#define TLV320AIC23_SIDETONE_12 0x080 +#define TLV320AIC23_SIDETONE_18 0x0c0 + +extern struct snd_soc_dai tlv320aic23_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23; + +#endif /* _TLV320AIC23_H */ diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index bed8a9e63ddc..6b7ddfc92573 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, * Digital Audio Interface Operations */ static int aic26_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = { }, .ops = { .hw_params = aic26_hw_params, - }, - .dai_ops = { .digital_mute = aic26_mute, .set_sysclk = aic26_set_sysclk, .set_fmt = aic26_set_fmt, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 566a427c928f..255e784c805b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -48,7 +48,6 @@ #include "tlv320aic3x.h" -#define AUDIO_NAME "aic3x" #define AIC3X_VERSION "0.2" /* codec private data */ @@ -273,8 +272,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), @@ -282,8 +283,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - PGAR_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), @@ -334,7 +337,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); /* Left DAC_L1 Mixer */ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), @@ -342,7 +346,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { /* Right DAC_R1 Mixer */ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), @@ -351,14 +356,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { /* Left PGA Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), }; /* Right PGA Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; @@ -380,34 +389,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); /* Left PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), }; /* Right PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), }; /* Left Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), }; /* Right Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { @@ -440,22 +457,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Mono Output */ SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Left Inputs to Left ADC */ + /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_pga_mixer_controls[0], ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1_mux_controls), + SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1_mux_controls), SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line2_mux_controls), - /* Right Inputs to Right ADC */ + /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_pga_mixer_controls[0], ARRAY_SIZE(aic3x_right_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, @@ -532,7 +553,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left DAC Mux", "DAC_L2", "Left DAC"}, {"Left DAC Mux", "DAC_L3", "Left DAC"}, - {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"}, @@ -558,7 +580,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R2", "Right DAC"}, {"Right DAC Mux", "DAC_R3", "Right DAC"}, - {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"}, @@ -593,8 +616,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line2L Mux", "differential", "LINE2L"}, {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, {"Left ADC", NULL, "GPIO1 dmic modclk"}, @@ -606,18 +631,23 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, + {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ - {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"}, {"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"}, @@ -628,10 +658,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left PGA Bypass Mixer"}, /* Right PGA Bypass */ - {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"}, {"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"}, {"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"}, @@ -644,10 +677,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right HP Out", NULL, "Right PGA Bypass Mixer"}, /* Left Line2 Bypass */ - {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"}, - {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"}, @@ -658,10 +692,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left Line2 Bypass Mixer"}, /* Right Line2 Bypass */ - {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"}, - {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"}, {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"}, @@ -695,7 +730,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) } static int aic3x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -864,17 +900,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* interface format */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; - case SND_SOC_DAIFMT_DSP_A: + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; - case SND_SOC_DAIFMT_RIGHT_J: + case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x02 << 6); break; - case SND_SOC_DAIFMT_LEFT_J: + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): iface_breg |= (0x03 << 6); break; default: @@ -991,7 +1031,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) struct snd_soc_dai aic3x_dai = { - .name = "aic3x", + .name = "tlv320aic3x", .playback = { .stream_name = "Playback", .channels_min = 1, @@ -1006,8 +1046,6 @@ struct snd_soc_dai aic3x_dai = { .formats = AIC3X_FORMATS,}, .ops = { .hw_params = aic3x_hw_params, - }, - .dai_ops = { .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, @@ -1055,7 +1093,7 @@ static int aic3x_init(struct snd_soc_device *socdev) struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; - codec->name = "aic3x"; + codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; codec->read = aic3x_read_reg_cache; codec->write = aic3x_write; diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195aa02e4..7e982acf3996 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -48,7 +48,9 @@ #define MIC3LR_2_RADC_CTRL 18 /* Line1 Input control registers */ #define LINE1L_2_LADC_CTRL 19 +#define LINE1R_2_LADC_CTRL 21 #define LINE1R_2_RADC_CTRL 22 +#define LINE1L_2_RADC_CTRL 24 /* Line2 Input control registers */ #define LINE2L_2_LADC_CTRL 20 #define LINE2R_2_RADC_CTRL 23 @@ -79,6 +81,8 @@ #define LINE2L_2_HPLOUT_VOL 45 #define LINE2R_2_HPROUT_VOL 62 #define PGAL_2_HPLOUT_VOL 46 +#define PGAL_2_HPROUT_VOL 60 +#define PGAR_2_HPLOUT_VOL 49 #define PGAR_2_HPROUT_VOL 63 #define DACL1_2_HPLOUT_VOL 47 #define DACR1_2_HPROUT_VOL 64 @@ -88,6 +92,8 @@ #define LINE2L_2_HPLCOM_VOL 52 #define LINE2R_2_HPRCOM_VOL 69 #define PGAL_2_HPLCOM_VOL 53 +#define PGAR_2_HPLCOM_VOL 56 +#define PGAL_2_HPRCOM_VOL 67 #define PGAR_2_HPRCOM_VOL 70 #define DACL1_2_HPLCOM_VOL 54 #define DACR1_2_HPRCOM_VOL 71 @@ -103,11 +109,17 @@ #define MONOLOPM_CTRL 79 /* Line Output Plus/Minus control registers */ #define LINE2L_2_LLOPM_VOL 80 +#define LINE2L_2_RLOPM_VOL 87 +#define LINE2R_2_LLOPM_VOL 83 #define LINE2R_2_RLOPM_VOL 90 #define PGAL_2_LLOPM_VOL 81 +#define PGAL_2_RLOPM_VOL 88 +#define PGAR_2_LLOPM_VOL 84 #define PGAR_2_RLOPM_VOL 91 #define DACL1_2_LLOPM_VOL 82 +#define DACL1_2_RLOPM_VOL 89 #define DACR1_2_RLOPM_VOL 92 +#define DACR1_2_LLOPM_VOL 85 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 /* GPIO/IRQ registers */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c new file mode 100644 index 000000000000..413623147891 --- /dev/null +++ b/sound/soc/codecs/twl4030.c @@ -0,0 +1,826 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman, <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl4030.h" + +/* + * twl4030 register cache & default register settings + */ +static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { + 0x00, /* this register not used */ + 0x93, /* REG_CODEC_MODE (0x1) */ + 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_UNKNOWN (0x3) */ + 0x00, /* REG_MICBIAS_CTL (0x4) */ + 0x24, /* REG_ANAMICL (0x5) */ + 0x04, /* REG_ANAMICR (0x6) */ + 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x00, /* REG_ADCMICSEL (0x8) */ + 0x00, /* REG_DIGMIXING (0x9) */ + 0x0c, /* REG_ATXL1PGA (0xA) */ + 0x0c, /* REG_ATXR1PGA (0xB) */ + 0x00, /* REG_AVTXL2PGA (0xC) */ + 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_VOICE_IF (0xF) */ + 0x00, /* REG_ARXR1PGA (0x10) */ + 0x00, /* REG_ARXL1PGA (0x11) */ + 0x6c, /* REG_ARXR2PGA (0x12) */ + 0x6c, /* REG_ARXL2PGA (0x13) */ + 0x00, /* REG_VRXPGA (0x14) */ + 0x00, /* REG_VSTPGA (0x15) */ + 0x00, /* REG_VRX2ARXPGA (0x16) */ + 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_ARX2VTXPGA (0x18) */ + 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x00, /* REG_ATX2ARXPGA (0x1D) */ + 0x00, /* REG_BT_IF (0x1E) */ + 0x00, /* REG_BTPGA (0x1F) */ + 0x00, /* REG_BTSTPGA (0x20) */ + 0x00, /* REG_EAR_CTL (0x21) */ + 0x24, /* REG_HS_SEL (0x22) */ + 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_POPN_SET (0x24) */ + 0x00, /* REG_PREDL_CTL (0x25) */ + 0x00, /* REG_PREDR_CTL (0x26) */ + 0x00, /* REG_PRECKL_CTL (0x27) */ + 0x00, /* REG_PRECKR_CTL (0x28) */ + 0x00, /* REG_HFL_CTL (0x29) */ + 0x00, /* REG_HFR_CTL (0x2A) */ + 0x00, /* REG_ALC_CTL (0x2B) */ + 0x00, /* REG_ALC_SET1 (0x2C) */ + 0x00, /* REG_ALC_SET2 (0x2D) */ + 0x00, /* REG_BOOST_CTL (0x2E) */ + 0x00, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x00, /* REG_DTMF_TONEXT1H (0x31) */ + 0x00, /* REG_DTMF_TONEXT1L (0x32) */ + 0x00, /* REG_DTMF_TONEXT2H (0x33) */ + 0x00, /* REG_DTMF_TONEXT2L (0x34) */ + 0x00, /* REG_DTMF_TONOFF (0x35) */ + 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ + 0x16, /* REG_APLL_CTL (0x3A) */ + 0x00, /* REG_DTMF_CTL (0x3B) */ + 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x00, /* REG_MISC_SET_1 (0x3E) */ + 0x00, /* REG_PCMBTMUX (0x3F) */ + 0x00, /* not used (0x40) */ + 0x00, /* not used (0x41) */ + 0x00, /* not used (0x42) */ + 0x00, /* REG_RX_PATH_SEL (0x43) */ + 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x00, /* REG_VIBRA_CTL (0x45) */ + 0x00, /* REG_VIBRA_SET (0x46) */ + 0x00, /* REG_VIBRA_PWM_SET (0x47) */ + 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x00, /* REG_MISC_SET_2 (0x49) */ +}; + +/* + * read twl4030 register cache + */ +static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + return cache[reg]; +} + +/* + * write twl4030 register cache + */ +static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL4030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl4030 register space + */ +static int twl4030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + twl4030_write_reg_cache(codec, reg, value); + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode & ~TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_set_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode | TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_init_chip(struct snd_soc_codec *codec) +{ + int i; + + /* clear CODECPDZ prior to setting register defaults */ + twl4030_clear_codecpdz(codec); + + /* set all audio section registers to reasonable defaults */ + for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) + twl4030_write(codec, i, twl4030_reg[i]); + +} + +/* + * Some of the gain controls in TWL (mostly those which are associated with + * the outputs) are implemented in an interesting way: + * 0x0 : Power down (mute) + * 0x1 : 6dB + * 0x2 : 0 dB + * 0x3 : -6 dB + * Inverting not going to help with these. + * Custom volsw and volsw_2r get/put functions to handle these gain bits. + */ +#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_twl4030, \ + .put = snd_soc_put_volsw_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = snd_soc_get_volsw_r2_twl4030,\ + .put = snd_soc_put_volsw_r2_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ + SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ + xinvert, tlv_array) + +static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + + if (shift != rshift) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg) >> rshift) & mask; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + } + + return 0; +} + +static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + unsigned short val, val2, val_mask; + + val = (ucontrol->value.integer.value[0] & mask); + + val_mask = mask << shift; + if (val) + val = max + 1 - val; + val = val << shift; + if (shift != rshift) { + val2 = (ucontrol->value.integer.value[1] & mask); + val_mask |= mask << rshift; + if (val2) + val2 = max + 1 - val2; + val |= val2 << rshift; + } + return snd_soc_update_bits(codec, reg, val_mask, val); +} + +static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1<<fls(max))-1; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg2) >> shift) & mask; + + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + + return 0; +} + +static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + int err; + unsigned short val, val2, val_mask; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] & mask); + val2 = (ucontrol->value.integer.value[1] & mask); + + if (val) + val = max + 1 - val; + if (val2) + val2 = max + 1 - val2; + + val = val << shift; + val2 = val2 << shift; + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + +/* + * FGAIN volume control: + * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) + */ +static DECLARE_TLV_DB_SCALE(master_tlv, -6300, 100, 1); + +/* + * CGAIN volume control: + * 0 dB to 12 dB in 6 dB steps + * value 2 and 3 means 12 dB + */ +static DECLARE_TLV_DB_SCALE(master_coarse_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new twl4030_snd_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 0x3f, 0, master_tlv), + SOC_DOUBLE_R_TLV("Master PCM Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 0x2, 0, master_coarse_tlv), + SOC_DOUBLE_R("Capture Volume", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 0x1f, 0), +}; + +/* add non dapm controls */ +static int twl4030_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&twl4030_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* outputs */ + {"OUTL", NULL, "DACL"}, + {"OUTR", NULL, "DACR"}, + + /* inputs */ + {"ADCL", NULL, "INL"}, + {"ADCR", NULL, "INR"}, +}; + +static int twl4030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets, + ARRAY_SIZE(twl4030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + u8 anamicl, regmisc1, byte, popn, hsgain; + int i = 0; + + /* set CODECPDZ to turn on codec */ + twl4030_set_codecpdz(codec); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_clear_codecpdz(codec); + twl4030_set_codecpdz(codec); + + /* program anti-pop with bias ramp delay */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= TWL4030_RAMP_DELAY; + popn |= TWL4030_RAMP_DELAY_645MS; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + popn |= TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* enable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_0DB | TWL4030_HSL_GAIN_0DB; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* enable anti-pop ramp */ + popn |= TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); +} + +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + u8 popn, hsgain; + + /* disable anti-pop ramp */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= ~TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* disable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_PWR_DOWN | TWL4030_HSL_GAIN_PWR_DOWN; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* disable bias out */ + popn &= ~TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* power down */ + twl4030_clear_codecpdz(codec); +} + +static int twl4030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl4030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + /* TODO: develop a twl4030_prepare function */ + break; + case SND_SOC_BIAS_STANDBY: + /* TODO: develop a twl4030_standby function */ + twl4030_power_down(codec); + break; + case SND_SOC_BIAS_OFF: + twl4030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u8 mode, old_mode, format, old_format; + + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; + + switch (params_rate(params)) { + case 8000: + mode |= TWL4030_APLL_RATE_8000; + break; + case 11025: + mode |= TWL4030_APLL_RATE_11025; + break; + case 12000: + mode |= TWL4030_APLL_RATE_12000; + break; + case 16000: + mode |= TWL4030_APLL_RATE_16000; + break; + case 22050: + mode |= TWL4030_APLL_RATE_22050; + break; + case 24000: + mode |= TWL4030_APLL_RATE_24000; + break; + case 32000: + mode |= TWL4030_APLL_RATE_32000; + break; + case 44100: + mode |= TWL4030_APLL_RATE_44100; + break; + case 48000: + mode |= TWL4030_APLL_RATE_48000; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_set_codecpdz(codec); + } + + /* sample size */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + format &= ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format |= TWL4030_DATA_WIDTH_16S_16W; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format |= TWL4030_DATA_WIDTH_32S_24W; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + return 0; +} + +static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 19200000: + infreq = TWL4030_APLL_INFREQ_19200KHZ; + break; + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + case 38400000: + infreq = TWL4030_APLL_INFREQ_38400KHZ; + break; + default: + printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + format &= ~(TWL4030_AIF_SLAVE_EN); + format &= ~(TWL4030_CLK256FS_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_AIF_SLAVE_EN; + format |= TWL4030_CLK256FS_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + format &= ~TWL4030_AIF_FORMAT; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= TWL4030_AIF_FORMAT_CODEC; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + + return 0; +} + +#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) +#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_dai twl4030_dai = { + .name = "twl4030", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(twl4030_dai); + +static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl4030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl4030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + * register the mixer and dsp interfaces with the kernel + */ + +static int twl4030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + printk(KERN_INFO "TWL4030 Audio Codec init \n"); + + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = &twl4030_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to create pcms\n"); + goto pcm_err; + } + + twl4030_init_chip(codec); + + /* power on device */ + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl4030_add_controls(codec); + twl4030_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl4030_socdev; + +static int twl4030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl4030_socdev = socdev; + twl4030_init(socdev); + + return 0; +} + +static int twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_probe, + .remove = twl4030_remove, + .suspend = twl4030_suspend, + .resume = twl4030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + +MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); +MODULE_AUTHOR("Steve Sakoman"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h new file mode 100644 index 000000000000..09865d9f5203 --- /dev/null +++ b/sound/soc/codecs/twl4030.h @@ -0,0 +1,197 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL4030_AUDIO_H__ +#define __TWL4030_AUDIO_H__ + +#define TWL4030_REG_CODEC_MODE 0x1 +#define TWL4030_REG_OPTION 0x2 +#define TWL4030_REG_UNKNOWN 0x3 +#define TWL4030_REG_MICBIAS_CTL 0x4 +#define TWL4030_REG_ANAMICL 0x5 +#define TWL4030_REG_ANAMICR 0x6 +#define TWL4030_REG_AVADC_CTL 0x7 +#define TWL4030_REG_ADCMICSEL 0x8 +#define TWL4030_REG_DIGMIXING 0x9 +#define TWL4030_REG_ATXL1PGA 0xA +#define TWL4030_REG_ATXR1PGA 0xB +#define TWL4030_REG_AVTXL2PGA 0xC +#define TWL4030_REG_AVTXR2PGA 0xD +#define TWL4030_REG_AUDIO_IF 0xE +#define TWL4030_REG_VOICE_IF 0xF +#define TWL4030_REG_ARXR1PGA 0x10 +#define TWL4030_REG_ARXL1PGA 0x11 +#define TWL4030_REG_ARXR2PGA 0x12 +#define TWL4030_REG_ARXL2PGA 0x13 +#define TWL4030_REG_VRXPGA 0x14 +#define TWL4030_REG_VSTPGA 0x15 +#define TWL4030_REG_VRX2ARXPGA 0x16 +#define TWL4030_REG_AVDAC_CTL 0x17 +#define TWL4030_REG_ARX2VTXPGA 0x18 +#define TWL4030_REG_ARXL1_APGA_CTL 0x19 +#define TWL4030_REG_ARXR1_APGA_CTL 0x1A +#define TWL4030_REG_ARXL2_APGA_CTL 0x1B +#define TWL4030_REG_ARXR2_APGA_CTL 0x1C +#define TWL4030_REG_ATX2ARXPGA 0x1D +#define TWL4030_REG_BT_IF 0x1E +#define TWL4030_REG_BTPGA 0x1F +#define TWL4030_REG_BTSTPGA 0x20 +#define TWL4030_REG_EAR_CTL 0x21 +#define TWL4030_REG_HS_SEL 0x22 +#define TWL4030_REG_HS_GAIN_SET 0x23 +#define TWL4030_REG_HS_POPN_SET 0x24 +#define TWL4030_REG_PREDL_CTL 0x25 +#define TWL4030_REG_PREDR_CTL 0x26 +#define TWL4030_REG_PRECKL_CTL 0x27 +#define TWL4030_REG_PRECKR_CTL 0x28 +#define TWL4030_REG_HFL_CTL 0x29 +#define TWL4030_REG_HFR_CTL 0x2A +#define TWL4030_REG_ALC_CTL 0x2B +#define TWL4030_REG_ALC_SET1 0x2C +#define TWL4030_REG_ALC_SET2 0x2D +#define TWL4030_REG_BOOST_CTL 0x2E +#define TWL4030_REG_SOFTVOL_CTL 0x2F +#define TWL4030_REG_DTMF_FREQSEL 0x30 +#define TWL4030_REG_DTMF_TONEXT1H 0x31 +#define TWL4030_REG_DTMF_TONEXT1L 0x32 +#define TWL4030_REG_DTMF_TONEXT2H 0x33 +#define TWL4030_REG_DTMF_TONEXT2L 0x34 +#define TWL4030_REG_DTMF_TONOFF 0x35 +#define TWL4030_REG_DTMF_WANONOFF 0x36 +#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 +#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 +#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 +#define TWL4030_REG_APLL_CTL 0x3A +#define TWL4030_REG_DTMF_CTL 0x3B +#define TWL4030_REG_DTMF_PGA_CTL2 0x3C +#define TWL4030_REG_DTMF_PGA_CTL1 0x3D +#define TWL4030_REG_MISC_SET_1 0x3E +#define TWL4030_REG_PCMBTMUX 0x3F +#define TWL4030_REG_RX_PATH_SEL 0x43 +#define TWL4030_REG_VDL_APGA_CTL 0x44 +#define TWL4030_REG_VIBRA_CTL 0x45 +#define TWL4030_REG_VIBRA_SET 0x46 +#define TWL4030_REG_VIBRA_PWM_SET 0x47 +#define TWL4030_REG_ANAMIC_GAIN 0x48 +#define TWL4030_REG_MISC_SET_2 0x49 + +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) + +/* Bitfield Definitions */ + +/* TWL4030_CODEC_MODE (0x01) Fields */ + +#define TWL4030_APLL_RATE 0xF0 +#define TWL4030_APLL_RATE_8000 0x00 +#define TWL4030_APLL_RATE_11025 0x10 +#define TWL4030_APLL_RATE_12000 0x20 +#define TWL4030_APLL_RATE_16000 0x40 +#define TWL4030_APLL_RATE_22050 0x50 +#define TWL4030_APLL_RATE_24000 0x60 +#define TWL4030_APLL_RATE_32000 0x80 +#define TWL4030_APLL_RATE_44100 0x90 +#define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_SEL_16K 0x04 +#define TWL4030_CODECPDZ 0x02 +#define TWL4030_OPT_MODE 0x01 + +/* ANAMICL (0x05) Fields */ +#define TWL4030_CNCL_OFFSET_START 0x80 +#define TWL4030_OFFSET_CNCL_SEL 0x60 +#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 +#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 +#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 +#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 +#define TWL4030_MICAMPL_EN 0x10 +#define TWL4030_CKMIC_EN 0x08 +#define TWL4030_AUXL_EN 0x04 +#define TWL4030_HSMIC_EN 0x02 +#define TWL4030_MAINMIC_EN 0x01 + +/* ANAMICR (0x06) Fields */ +#define TWL4030_MICAMPR_EN 0x10 +#define TWL4030_AUXR_EN 0x04 +#define TWL4030_SUBMIC_EN 0x01 + +/* AUDIO_IF (0x0E) Fields */ + +#define TWL4030_AIF_SLAVE_EN 0x80 +#define TWL4030_DATA_WIDTH 0x60 +#define TWL4030_DATA_WIDTH_16S_16W 0x00 +#define TWL4030_DATA_WIDTH_32S_16W 0x40 +#define TWL4030_DATA_WIDTH_32S_24W 0x60 +#define TWL4030_AIF_FORMAT 0x18 +#define TWL4030_AIF_FORMAT_CODEC 0x00 +#define TWL4030_AIF_FORMAT_LEFT 0x08 +#define TWL4030_AIF_FORMAT_RIGHT 0x10 +#define TWL4030_AIF_FORMAT_TDM 0x18 +#define TWL4030_AIF_TRI_EN 0x04 +#define TWL4030_CLK256FS_EN 0x02 +#define TWL4030_AIF_EN 0x01 + +/* HS_GAIN_SET (0x23) Fields */ + +#define TWL4030_HSR_GAIN 0x0C +#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 +#define TWL4030_HSR_GAIN_0DB 0x08 +#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C +#define TWL4030_HSL_GAIN 0x03 +#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 +#define TWL4030_HSL_GAIN_0DB 0x02 +#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 + +/* HS_POPN_SET (0x24) Fields */ + +#define TWL4030_VMID_EN 0x40 +#define TWL4030_EXTMUTE 0x20 +#define TWL4030_RAMP_DELAY 0x1C +#define TWL4030_RAMP_DELAY_20MS 0x00 +#define TWL4030_RAMP_DELAY_40MS 0x04 +#define TWL4030_RAMP_DELAY_81MS 0x08 +#define TWL4030_RAMP_DELAY_161MS 0x0C +#define TWL4030_RAMP_DELAY_323MS 0x10 +#define TWL4030_RAMP_DELAY_645MS 0x14 +#define TWL4030_RAMP_DELAY_1291MS 0x18 +#define TWL4030_RAMP_DELAY_2581MS 0x1C +#define TWL4030_RAMP_EN 0x02 + +/* APLL_CTL (0x3A) Fields */ + +#define TWL4030_APLL_EN 0x10 +#define TWL4030_APLL_INFREQ 0x0F +#define TWL4030_APLL_INFREQ_19200KHZ 0x05 +#define TWL4030_APLL_INFREQ_26000KHZ 0x06 +#define TWL4030_APLL_INFREQ_38400KHZ 0x0F + +/* REG_MISC_SET_1 (0x3E) Fields */ + +#define TWL4030_CLK64_EN 0x80 +#define TWL4030_SCRAMBLE_EN 0x40 +#define TWL4030_FMLOOP_EN 0x20 +#define TWL4030_SMOOTH_ANAVOL_EN 0x02 +#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 + +extern struct snd_soc_dai twl4030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl4030; + +#endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c new file mode 100644 index 000000000000..91f333cdc7cf --- /dev/null +++ b/sound/soc/codecs/uda134x.c @@ -0,0 +1,656 @@ +/* + * uda134x.c -- UDA134X ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin <chripell@evolware.org> + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <sound/uda134x.h> +#include <sound/l3.h> + +#include "uda134x.h" + + +#define POWER_OFF_ON_STANDBY 1 +/* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you define POWER_OFF_ON_STANDBY the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + just comment this line, you will have slightly higher power + consumption . Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + +#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda134x_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda134x_reg[UDA134X_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda134x registers + * + */ +static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + struct uda134x_platform_data *pd = codec->control_data; + + pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); + + if (reg >= UDA134X_REGS_NUM) { + printk(KERN_ERR "%s unkown register: reg: %d", + __func__, reg); + return -EINVAL; + } + + uda134x_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA134X_STATUS0: + case UDA134X_STATUS1: + addr = UDA134X_STATUS_ADDR; + break; + case UDA134X_DATA000: + case UDA134X_DATA001: + case UDA134X_DATA010: + addr = UDA134X_DATA0_ADDR; + break; + case UDA134X_DATA1: + addr = UDA134X_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA134X_EXTADDR_PREFIX); + + ret = l3_write(&pd->l3, + UDA134X_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA134X_DATA0_ADDR; + data = (value | UDA134X_EXTDATA_PREFIX); + break; + } + + ret = l3_write(&pd->l3, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda134x_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6)); + msleep(1); + uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda134x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010); + + pr_debug("%s mute: %d\n", __func__, mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda134x_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + if (uda134x->master_substream) { + master_runtime = uda134x->master_substream->runtime; + + pr_debug("%s constraining to %d bits at %d\n", __func__, + master_runtime->sample_bits, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + uda134x->slave_substream = substream; + } else + uda134x->master_substream = substream; + + return 0; +} + +static void uda134x_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + + if (uda134x->master_substream == substream) + uda134x->master_substream = uda134x->slave_substream; + + uda134x->slave_substream = NULL; +} + +static int uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + u8 hw_params; + + if (substream == uda134x->slave_substream) { + pr_debug("%s ignoring hw_params for slave substream\n", + __func__); + return 0; + } + + hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + pr_debug("%s sysclk: %d, rate:%d\n", __func__, + uda134x->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda134x->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + printk(KERN_ERR "%s unsupported fs\n", __func__); + return -EINVAL; + } + + pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, + uda134x->dai_fmt, params_format(params)); + + /* set DAI format and word length */ + switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + printk(KERN_ERR "%s unsupported format (right)\n", + __func__); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + printk(KERN_ERR "%s unsupported format\n", __func__); + return -EINVAL; + } + + uda134x_write(codec, UDA134X_STATUS0, hw_params); + + return 0; +} + +static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + because the codec is slave. Of course limitations of the clock + master (the IIS controller) apply. + We'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda134x->sysclk = freq; + return 0; + } + + printk(KERN_ERR "%s unsupported sysclk\n", __func__); + return -EINVAL; +} + +static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s fmt: %08X\n", __func__, fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + printk(KERN_ERR "%s unsupported slave mode\n", __func__); + return -EINVAL; + } + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + printk(KERN_ERR "%s unsupported clock inversion\n", __func__); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda134x->dai_fmt = fmt; + + return 0; +} + +static int uda134x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + struct uda134x_platform_data *pd = codec->control_data; + int i; + u8 *cache = codec->reg_cache; + + pr_debug("%s bias level %d\n", __func__, level); + + switch (level) { + case SND_SOC_BIAS_ON: + /* ADC, DAC on */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + case SND_SOC_BIAS_PREPARE: + /* power on */ + if (pd->power) { + pd->power(1); + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++) + codec->write(codec, i, *cache++); + } + break; + case SND_SOC_BIAS_STANDBY: + /* ADC, DAC power off */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (pd->power) + pd->power(0); + break; + } + codec->bias_level = level; + return 0; +} + +static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda134x_mixmode[] = {"Differential", "Analog1", + "Analog2", "Both"}; + +static const struct soc_enum uda134x_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), +SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), +SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), +SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), +SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), + +SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), +SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), +SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), + +SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), +SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), +SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static const struct snd_kcontrol_new uda1340_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static int uda134x_add_controls(struct snd_soc_codec *codec) +{ + int err, i, n; + const struct snd_kcontrol_new *ctrls; + struct uda134x_platform_data *pd = codec->control_data; + + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + n = ARRAY_SIZE(uda1340_snd_controls); + ctrls = uda1340_snd_controls; + break; + case UDA134X_UDA1341: + n = ARRAY_SIZE(uda1341_snd_controls); + ctrls = uda1341_snd_controls; + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + + for (i = 0; i < n; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ctrls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda134x_dai = { + .name = "UDA134X", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda134x_dai); + + +static int uda134x_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda134x_priv *uda134x; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda134x_platform_data *pd; + + printk(KERN_INFO "UDA134X SoC Audio Codec\n"); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA134X SoC codec: " + "missing L3 bitbang function\n"); + return -ENODEV; + } + + pd = codec_setup_data; + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1341: + case UDA134X_UDA1344: + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", + pd->model); + return -EINVAL; + } + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); + if (uda134x == NULL) + goto priv_err; + codec->private_data = uda134x; + + codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda134x_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA134X"; + codec->owner = THIS_MODULE; + codec->dai = &uda134x_dai; + codec->num_dai = 1; + codec->read = uda134x_read_reg_cache; + codec->write = uda134x_write; +#ifdef POWER_OFF_ON_STANDBY + codec->set_bias_level = uda134x_set_bias_level; +#endif + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->control_data = codec_setup_data; + + if (pd->power) + pd->power(1); + + uda134x_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda134x_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda134x_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda134x_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda134x_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + return 0; +} +#else +#define uda134x_soc_suspend NULL +#define uda134x_soc_resume NULL +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda134x = { + .probe = uda134x_soc_probe, + .remove = uda134x_soc_remove, + .suspend = uda134x_soc_suspend, + .resume = uda134x_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); + +MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h new file mode 100644 index 000000000000..94f440490b31 --- /dev/null +++ b/sound/soc/codecs/uda134x.h @@ -0,0 +1,36 @@ +#ifndef _UDA134X_CODEC_H +#define _UDA134X_CODEC_H + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +#endif diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index d206d7f892b6..330877c70699 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -36,7 +36,6 @@ #include "uda1380.h" #define UDA1380_VERSION "0.6" -#define AUDIO_NAME "uda1380" /* * uda1380 register cache @@ -408,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, * when the DAI is being clocked by the CPU DAI. It's up to the * machine and cpu DAI driver to do this before we are called. */ -static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -440,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) } static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -478,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -561,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -580,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -599,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .set_fmt = uda1380_set_dai_fmt, }, }, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 9a37c8d95ed2..173b66c0c766 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -3,7 +3,7 @@ * * Copyright 2006 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -18,6 +18,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,7 +28,6 @@ #include "wm8510.h" -#define AUDIO_NAME "wm8510" #define WM8510_VERSION "0.6" struct snd_soc_codec_device soc_codec_dev_wm8510; @@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { 0x0001, }; +#define WM8510_POWER1_BIASEN 0x08 +#define WM8510_POWER1_BUFIOEN 0x10 + /* * read wm8510 register cache */ @@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), -SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, - &wm8510_micpga_controls[0], - ARRAY_SIZE(wm8510_micpga_controls)), +SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0, + &wm8510_micpga_controls[0], + ARRAY_SIZE(wm8510_micpga_controls)), SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, &wm8510_boost_controls[0], ARRAY_SIZE(wm8510_boost_controls)), @@ -460,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -526,23 +530,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: - wm8510_write(codec, WM8510_POWER1, 0x1ff); - wm8510_write(codec, WM8510_POWER2, 0x1ff); - wm8510_write(codec, WM8510_POWER3, 0x1ff); - break; case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + wm8510_write(codec, WM8510_POWER1, power1); + break; + case SND_SOC_BIAS_STANDBY: + power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + wm8510_write(codec, WM8510_POWER1, power1); break; + case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ - wm8510_write(codec, WM8510_POWER1, 0x0); - wm8510_write(codec, WM8510_POWER2, 0x0); - wm8510_write(codec, WM8510_POWER3, 0x0); + wm8510_write(codec, WM8510_POWER1, 0); + wm8510_write(codec, WM8510_POWER2, 0); + wm8510_write(codec, WM8510_POWER3, 0); break; } + codec->bias_level = level; return 0; } @@ -570,8 +586,6 @@ struct snd_soc_dai wm8510_dai = { .formats = WM8510_FORMATS,}, .ops = { .hw_params = wm8510_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8510_mute, .set_fmt = wm8510_set_dai_fmt, .set_clkdiv = wm8510_set_dai_clkdiv, @@ -640,6 +654,7 @@ static int wm8510_init(struct snd_soc_device *socdev) } /* power on device */ + codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); @@ -747,6 +762,62 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8510_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8510_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8510_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8510\n"); + + return ret; +} + +static int __devexit wm8510_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8510_spi_driver = { + .driver = { + .name = "wm8510", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8510_spi_probe, + .remove = __devexit_p(wm8510_spi_remove), +}; + +static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + static int wm8510_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -772,8 +843,14 @@ static int wm8510_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8510_spi_write; + ret = spi_register_driver(&wm8510_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) @@ -796,6 +873,9 @@ static int wm8510_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8510_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8510_spi_driver); +#endif kfree(codec); return 0; diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c53683960456..bdefcf5c69ff 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -94,6 +94,7 @@ #define WM8510_MCLKDIV_12 (7 << 5) struct wm8510_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index df1ffbe305bf..220d4b68904a 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -36,7 +35,6 @@ #include "wm8580.h" -#define AUDIO_NAME "wm8580" #define WM8580_VERSION "0.1" struct pll_state { @@ -550,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id); + u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ @@ -576,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb); + wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb); return 0; } @@ -800,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, @@ -820,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, @@ -902,85 +896,85 @@ static struct snd_soc_device *wm8580_socdev; * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; -static struct i2c_driver wm8580_i2c_driver; -static struct i2c_client client_template; - -static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8580_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8580_socdev; - struct wm8580_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8580_init(socdev); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "failed to initialise WM8580\n"); - goto err; - } - - return ret; - -err: - kfree(codec); - kfree(i2c); return ret; } -static int wm8580_i2c_detach(struct i2c_client *client) +static int wm8580_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8580_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8580_codec_probe); -} +static const struct i2c_device_id wm8580_i2c_id[] = { + { "wm8580", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "WM8580 I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8580_i2c_attach, - .detach_client = wm8580_i2c_detach, - .command = NULL, + .probe = wm8580_i2c_probe, + .remove = wm8580_i2c_remove, + .id_table = wm8580_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8580", - .driver = &wm8580_i2c_driver, -}; +static int wm8580_add_i2c_device(struct platform_device *pdev, + const struct wm8580_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8580_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8580", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8580_i2c_driver); + return -ENODEV; +} #endif static int wm8580_probe(struct platform_device *pdev) @@ -1013,11 +1007,8 @@ static int wm8580_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8580_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8580_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1036,6 +1027,7 @@ static int wm8580_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8580_i2c_driver); #endif kfree(codec->private_data); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 589ddaba21d7..09e4422f6f2f 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -29,6 +29,7 @@ #define WM8580_CLKSRC_NONE 5 struct wm8580_setup_data { + int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c new file mode 100644 index 000000000000..71949bd320d3 --- /dev/null +++ b/sound/soc/codecs/wm8728.c @@ -0,0 +1,573 @@ +/* + * wm8728.c -- WM8728 ALSA SoC Audio driver + * + * Copyright 2008 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8728.h" + +struct snd_soc_codec_device soc_codec_dev_wm8728; + +/* + * We can't read the WM8728 register space so we cache them instead. + * Note that the defaults here aren't the physical defaults, we latch + * the volume update bits, mute the output and enable infinite zero + * detect. + */ +static const u16 wm8728_reg_defaults[] = { + 0x1ff, + 0x1ff, + 0x001, + 0x100, +}; + +static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + return cache[reg]; +} + +static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + cache[reg] = value; +} + +/* + * write to the WM8728 register space + */ +static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8728 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8728_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new wm8728_snd_controls[] = { + +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, + 0, 255, 0, wm8728_tlv), + +SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), +}; + +static int wm8728_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8728_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* + * DAPM controls. + */ +static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"VOUTL", NULL, "DAC"}, + {"VOUTR", NULL, "DAC"}, +}; + +static int wm8728_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets, + ARRAY_SIZE(wm8728_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int wm8728_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + if (mute) + wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + else + wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + + return 0; +} + +static int wm8728_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + dac &= ~0x18; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dac |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dac |= 0x08; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_DACCTL, dac); + + return 0; +} + +static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + + /* Currently only I2S is supported by the driver, though the + * hardware is more flexible. + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 1; + break; + default: + return -EINVAL; + } + + /* The hardware only support full slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface &= ~0x22; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x20; + iface &= ~0x02; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x02; + iface &= ~0x20; + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x22; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_IFCTL, iface); + return 0; +} + +static int wm8728_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + int i; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Power everything up... */ + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + + /* ..then sync in the register cache. */ + for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) + wm8728_write(codec, i, + wm8728_read_reg_cache(codec, i)); + } + break; + + case SND_SOC_BIAS_OFF: + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000) + +#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai wm8728_dai = { + .name = "WM8728", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8728_RATES, + .formats = WM8728_FORMATS, + }, + .ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(wm8728_dai); + +static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8728_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the WM8728 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8728_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8728"; + codec->owner = THIS_MODULE; + codec->read = wm8728_read_reg_cache; + codec->write = wm8728_write; + codec->set_bias_level = wm8728_set_bias_level; + codec->dai = &wm8728_dai; + codec->num_dai = 1; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults); + codec->reg_cache = kmemdup(wm8728_reg_defaults, + sizeof(wm8728_reg_defaults), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8728_add_controls(codec); + wm8728_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8728_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8728 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ + +static int wm8728_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = wm8728_init(socdev); + if (ret < 0) + pr_err("failed to initialise WM8728\n"); + + return ret; +} + +static int wm8728_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id wm8728_i2c_id[] = { + { "wm8728", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); + +static struct i2c_driver wm8728_i2c_driver = { + .driver = { + .name = "WM8728 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8728_i2c_probe, + .remove = wm8728_i2c_remove, + .id_table = wm8728_i2c_id, +}; + +static int wm8728_add_i2c_device(struct platform_device *pdev, + const struct wm8728_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8728_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8728", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8728_i2c_driver); + return -ENODEV; +} +#endif + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8728_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8728_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8728\n"); + + return ret; +} + +static int __devexit wm8728_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8728_spi_driver = { + .driver = { + .name = "wm8728", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8728_spi_probe, + .remove = __devexit_p(wm8728_spi_remove), +}; + +static int wm8728_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + +static int wm8728_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8728_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8728_socdev = socdev; + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = wm8728_add_i2c_device(pdev, setup); + } +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8728_spi_write; + ret = spi_register_driver(&wm8728_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } +#endif + + if (ret != 0) + kfree(codec); + + return ret; +} + +/* power down chip */ +static int wm8728_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&wm8728_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8728_spi_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8728 = { + .probe = wm8728_probe, + .remove = wm8728_remove, + .suspend = wm8728_suspend, + .resume = wm8728_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728); + +MODULE_DESCRIPTION("ASoC WM8728 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h new file mode 100644 index 000000000000..d269c132474b --- /dev/null +++ b/sound/soc/codecs/wm8728.h @@ -0,0 +1,30 @@ +/* + * wm8728.h -- WM8728 ASoC codec driver + * + * Copyright 2008 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8728_H +#define _WM8728_H + +#define WM8728_DACLVOL 0x00 +#define WM8728_DACRVOL 0x01 +#define WM8728_DACCTL 0x02 +#define WM8728_IFCTL 0x03 + +struct wm8728_setup_data { + int spi; + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8728_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8728; + +#endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b64d9a7ff76..c0f277053bb2 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,7 +29,6 @@ #include "wm8731.h" -#define AUDIO_NAME "wm8731" #define WM8731_VERSION "0.13" struct snd_soc_codec_device soc_codec_dev_wm8731; @@ -265,7 +264,8 @@ static inline int get_coeff(int mclk, int rate) } static int wm8731_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -294,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) +static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -306,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void wm8731_shutdown(struct snd_pcm_substream *substream) +static void wm8731_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -462,8 +464,6 @@ struct snd_soc_dai wm8731_dai = { .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, - }, - .dai_ops = { .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4892e398a598..860a1d56830a 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,7 +29,6 @@ #include "wm8750.h" -#define AUDIO_NAME "WM8750" #define WM8750_VERSION "0.12" /* codec private data */ @@ -615,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -710,8 +710,6 @@ struct snd_soc_dai wm8750_dai = { .formats = WM8750_FORMATS,}, .ops = { .hw_params = wm8750_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8750_mute, .set_fmt = wm8750_set_dai_fmt, .set_sysclk = wm8750_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8c4df44f3345..5e4cd3bb824a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -2,8 +2,7 @@ * wm8753.c -- WM8753 ALSA Soc Audio driver * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -40,6 +39,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -51,7 +51,6 @@ #include "wm8753.h" -#define AUDIO_NAME "wm8753" #define WM8753_VERSION "0.16" static int caps_charge = 2000; @@ -923,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1156,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1324,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1h_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1357,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1v_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1386,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode2_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1411,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1440,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1719,6 +1715,63 @@ err_driver: } #endif +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8753_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8753_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8753_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8753\n"); + + return ret; +} + +static int __devexit wm8753_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; + +static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif + + static int wm8753_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -1753,8 +1806,14 @@ static int wm8753_probe(struct platform_device *pdev) codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8753_add_i2c_device(pdev, setup); } -#else - /* Add other interfaces here */ +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8753_spi_write; + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } #endif if (ret != 0) { @@ -1798,6 +1857,9 @@ static int wm8753_remove(struct platform_device *pdev) i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8753_i2c_driver); #endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif kfree(codec->private_data); kfree(codec); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 7defde069f1d..f55704ce931b 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -2,8 +2,7 @@ * wm8753.h -- audio driver for WM8753 * * Copyright 2003 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -79,6 +78,7 @@ #define WM8753_ADCTL2 0x3f struct wm8753_setup_data { + int spi; int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 0b8c6d38b48f..d1326be91c8b 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -18,7 +18,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> @@ -728,7 +727,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) } static int wm8900_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1118,8 +1118,6 @@ struct snd_soc_dai wm8900_dai = { }, .ops = { .hw_params = wm8900_hw_params, - }, - .dai_ops = { .set_clkdiv = wm8900_set_dai_clkdiv, .set_pll = wm8900_set_dai_pll, .set_fmt = wm8900_set_dai_fmt, @@ -1388,89 +1386,86 @@ static struct snd_soc_device *wm8900_socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8900_i2c_driver; -static struct i2c_client client_template; - /* If the i2c layer weren't so broken, we could pass this kind of data around */ -static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8900_socdev; - struct wm8900_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - dev_err(&adap->dev, "Probe on %x\n", addr); - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&adap->dev, - "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8900_init(socdev); - if (ret < 0) { - dev_err(&adap->dev, "failed to initialise WM8900\n"); - goto err; - } - return ret; - -err: - kfree(codec); - kfree(i2c); + if (ret < 0) + dev_err(&i2c->dev, "failed to initialise WM8900\n"); return ret; } -static int wm8900_i2c_detach(struct i2c_client *client) +static int wm8900_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8900_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8900_codec_probe); -} +static const struct i2c_device_id wm8900_i2c_id[] = { + { "wm8900", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8900_i2c_driver = { .driver = { .name = "WM8900 I2C codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8900_i2c_attach, - .detach_client = wm8900_i2c_detach, - .command = NULL, + .probe = wm8900_i2c_probe, + .remove = wm8900_i2c_remove, + .id_table = wm8900_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8900", - .driver = &wm8900_i2c_driver, -}; +static int wm8900_add_i2c_device(struct platform_device *pdev, + const struct wm8900_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8900_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8900", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8900_i2c_driver); + return -ENODEV; +} #endif static int wm8900_probe(struct platform_device *pdev) @@ -1498,11 +1493,8 @@ static int wm8900_probe(struct platform_device *pdev) wm8900_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8900_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8900_add_i2c_device(pdev, setup); } #else #error Non-I2C interfaces not yet supported @@ -1522,6 +1514,7 @@ static int wm8900_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8900_i2c_driver); #endif kfree(codec); diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h index ba450d99e902..2249a446ad37 100644 --- a/sound/soc/codecs/wm8900.h +++ b/sound/soc/codecs/wm8900.h @@ -55,6 +55,7 @@ #define WM8900_ struct wm8900_setup_data { + int i2c_bus; unsigned short i2c_address; }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a3f54ec4226e..efbe8927b7d2 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = { /* Input PGAs - No TLV since the scale depends on PGA mode */ SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0, 0, 31, 0), SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1, 6, 1, 0), SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0, - 7, 1, 0), + 7, 1, 1), SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0), SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, @@ -1257,7 +1257,8 @@ static struct { { 0, 0 }, }; -static int wm8903_startup(struct snd_pcm_substream *substream) +static int wm8903_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1298,7 +1299,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream) return 0; } -static void wm8903_shutdown(struct snd_pcm_substream *substream) +static void wm8903_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1317,7 +1319,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream) } static int wm8903_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1515,8 +1518,6 @@ struct snd_soc_dai wm8903_dai = { .startup = wm8903_startup, .shutdown = wm8903_shutdown, .hw_params = wm8903_hw_params, - }, - .dai_ops = { .digital_mute = wm8903_digital_mute, .set_fmt = wm8903_set_dai_fmt, .set_sysclk = wm8903_set_dai_sysclk diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 974a4cd0f3fd..26edcc9d6e87 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,7 +29,6 @@ #include "wm8971.h" -#define AUDIO_NAME "wm8971" #define WM8971_VERSION "0.9" #define WM8971_REG_COUNT 43 @@ -542,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -635,8 +635,6 @@ struct snd_soc_dai wm8971_dai = { .formats = WM8971_FORMATS,}, .ops = { .hw_params = wm8971_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8971_mute, .set_fmt = wm8971_set_dai_fmt, .set_sysclk = wm8971_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 63410d7b5efb..13926516d16e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -30,7 +30,6 @@ #include "wm8990.h" -#define AUDIO_NAME "wm8990" #define WM8990_VERSION "0.2" /* codec private data */ @@ -107,6 +106,7 @@ static const u16 wm8990_reg[] = { 0x0008, /* R60 - PLL1 */ 0x0031, /* R61 - PLL2 */ 0x0026, /* R62 - PLL3 */ + 0x0000, /* R63 - Driver internal */ }; /* @@ -127,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); - /* Reset register is uncached */ - if (reg == 0) + /* Reset register and reserved registers are uncached */ + if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) return; cache[reg] = value; @@ -1173,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8990_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1223,8 +1223,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: break; + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); break; + case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable all output discharge bits */ @@ -1273,10 +1279,17 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); - } else { - /* ON -> standby */ + /* Enable workaround for ADC clocking issue. */ + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2); + wm8990_write(codec, WM8990_EXT_CTL1, 0xa003); + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0); } + + /* VMID=2*250k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: @@ -1350,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = { .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, .ops = { - .hw_params = wm8990_hw_params,}, - .dai_ops = { + .hw_params = wm8990_hw_params, .digital_mute = wm8990_mute, .set_fmt = wm8990_set_dai_fmt, .set_clkdiv = wm8990_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 0e192f3b0788..7114ddc88b4b 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -80,8 +80,8 @@ #define WM8990_PLL3 0x3E #define WM8990_INTDRIVBITS 0x3F -#define WM8990_REGISTER_COUNT 60 -#define WM8990_MAX_REGISTER 0x3F +#define WM8990_EXT_ACCESS_ENA 0x75 +#define WM8990_EXT_CTL1 0x7a /* * Field Definitions. diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 2f1c91b1d556..40f14061fb72 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -2,8 +2,7 @@ * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -488,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -508,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -534,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, @@ -689,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) ret = wm9712_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n"); goto reset_err; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 441d0580db1f..9dad0bffcb05 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -2,8 +2,7 @@ * wm9713.c -- ALSA Soc WM9713 codec support * * Copyright 2006 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -141,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), -SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), SOC_ENUM("ALC Function", wm9713_enum[6]), SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), @@ -929,11 +928,10 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; switch (params_format(params)) { @@ -955,11 +953,10 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 status; /* Gracefully shut down the voice interface. */ @@ -970,12 +967,11 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) ac97_write(codec, AC97_EXTENDED_MID, status); } -static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +static int ac97_hifi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; int reg; u16 vra; @@ -990,12 +986,11 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); @@ -1029,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, @@ -1043,8 +1038,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_hifi_prepare,}, - .dai_ops = { + .prepare = ac97_hifi_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1057,8 +1051,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_aux_prepare,}, - .dai_ops = { + .prepare = ac97_aux_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1078,8 +1071,7 @@ struct snd_soc_dai wm9713_dai[] = { .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown,}, - .dai_ops = { + .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, @@ -1098,6 +1090,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; @@ -1241,7 +1235,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n"); goto reset_err; } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8f7e33834902..b502741692d6 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -17,3 +17,13 @@ config SND_DAVINCI_SOC_EVM help Say Y if you want to add support for SoC audio on TI DaVinci EVM platform. + +config SND_DAVINCI_SOC_SFFSDR + tristate "SoC Audio support for SFFSDR" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + select SND_DAVINCI_SOC_I2S + select SND_SOC_PCM3008 + select SFFSDR_FPGA + help + Say Y if you want to add support for SoC audio on + Lyrtech SFFSDR board. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca772e5b4637..ca8bae1fc3f6 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -7,5 +7,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o +snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 9e6062cd6b59..2ce34d44b15c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -128,7 +128,7 @@ static struct snd_soc_dai_link evm_dai = { }; /* davinci-evm audio machine driver */ -static struct snd_soc_machine snd_soc_machine_evm = { +static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .dai_link = &evm_dai, .num_links = 1, @@ -142,7 +142,7 @@ static struct aic3x_setup_data evm_aic3x_setup = { /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { - .machine = &snd_soc_machine_evm, + .card = &snd_soc_card_evm, .platform = &davinci_soc_platform, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &evm_aic3x_setup, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index abb5fedb0b1e..cf31b3bb96cf 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -59,6 +59,7 @@ #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) #define DAVINCI_MCBSP_PCR_FSRP (1 << 2) #define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7) #define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) #define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) #define DAVINCI_MCBSP_PCR_FSRM (1 << 10) @@ -110,16 +111,59 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_platform *platform = socdev->platform; u32 w; + int ret; /* Start the sample generator and enable transmitter/receiver */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Stop the DMA to avoid data loss */ + /* while the transmitter is out of reset to handle XSYNCERR */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA stop failed\n"); + } + + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Restart the DMA */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA start failed\n"); + } + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + } else { + + /* Enable the reciever */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } + /* Start frame sync */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -144,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream) +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -171,6 +216,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, DAVINCI_MCBSP_SRGR_FSGM); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP CLKR pin is the input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate Generator. */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_SCLKME | + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; case SND_SOC_DAIFMT_CBM_CFM: davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); break; @@ -205,11 +260,34 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(0)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(0) | + DAVINCI_MCBSP_XCR_XFIG); + break; + case SND_SOC_DAIFMT_I2S: + default: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + break; + } + return 0; } static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; @@ -219,17 +297,14 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, u32 w; /* general line settings */ - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, - DAVINCI_MCBSP_SPCR_RINTM(3) | - DAVINCI_MCBSP_SPCR_XINTM(3) | - DAVINCI_MCBSP_SPCR_FREE); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, - DAVINCI_MCBSP_RCR_RFRLEN1(1) | - DAVINCI_MCBSP_RCR_RDATDLY(1)); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, - DAVINCI_MCBSP_XCR_XFRLEN1(1) | - DAVINCI_MCBSP_XCR_XDATDLY(1) | - DAVINCI_MCBSP_XCR_XFIG); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } else { + w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); @@ -260,20 +335,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } else { + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + } return 0; } -static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -299,8 +378,8 @@ static int davinci_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -361,8 +440,8 @@ static void davinci_i2s_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai *cpu_dai = card->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -381,7 +460,6 @@ static void davinci_i2s_remove(struct platform_device *pdev, struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .playback = { @@ -397,8 +475,7 @@ struct snd_soc_dai davinci_i2s_dai = { .ops = { .startup = davinci_i2s_startup, .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params,}, - .dai_ops = { + .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, }, }; diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c new file mode 100644 index 000000000000..e95fde1766b5 --- /dev/null +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -0,0 +1,157 @@ +/* + * ASoC driver for Lyrtech SFFSDR board. + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow: + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/dma.h> +#include <asm/plat-sffsdr/sffsdr-fpga.h> + +#include <mach/mcbsp.h> +#include <mach/edma.h> + +#include "../codecs/pcm3008.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +static int sffsdr_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fs; + int ret = 0; + + /* Set cpu DAI configuration: + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_RIGHT_J | + SND_SOC_DAIFMT_CBM_CFS | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + + pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); + + return sffsdr_fpga_set_codec_fs(fs); +} + +static struct snd_soc_ops sffsdr_ops = { + .hw_params = sffsdr_hw_params, +}; + +/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sffsdr_dai = { + .name = "PCM3008", /* Codec name */ + .stream_name = "PCM3008 HiFi", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &pcm3008_dai, + .ops = &sffsdr_ops, +}; + +/* davinci-sffsdr audio machine driver */ +static struct snd_soc_card snd_soc_sffsdr = { + .name = "DaVinci SFFSDR", + .dai_link = &sffsdr_dai, + .num_links = 1, +}; + +/* sffsdr audio private data */ +static struct pcm3008_setup_data sffsdr_pcm3008_setup = { + .dem0_pin = GPIO(45), + .dem1_pin = GPIO(46), + .pdad_pin = GPIO(47), + .pdda_pin = GPIO(38), +}; + +/* sffsdr audio subsystem */ +static struct snd_soc_device sffsdr_snd_devdata = { + .card = &snd_soc_sffsdr, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_pcm3008, + .codec_data = &sffsdr_pcm3008_setup, +}; + +static struct resource sffsdr_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data sffsdr_snd_data = { + .tx_dma_ch = DAVINCI_DMA_MCBSP_TX, + .rx_dma_ch = DAVINCI_DMA_MCBSP_RX, +}; + +static struct platform_device *sffsdr_snd_device; + +static int __init sffsdr_init(void) +{ + int ret; + + sffsdr_snd_device = platform_device_alloc("soc-audio", 0); + if (!sffsdr_snd_device) { + printk(KERN_ERR "platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); + sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; + sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + + ret = platform_device_add_resources(sffsdr_snd_device, + sffsdr_snd_resources, + ARRAY_SIZE(sffsdr_snd_resources)); + if (ret) { + printk(KERN_ERR "platform device add ressources failed\n"); + goto error; + } + + ret = platform_device_add(sffsdr_snd_device); + if (ret) + goto error; + + return ret; + +error: + platform_device_put(sffsdr_snd_device); + return ret; +} + +static void __exit sffsdr_exit(void) +{ + platform_device_unregister(sffsdr_snd_device); +} + +module_init(sffsdr_init); +module_exit(sffsdr_exit); + +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index d2d3da9729f2..bf92331b4768 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -284,7 +284,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * fsl_dma_new: initialize this PCM driver. * * This function is called when the codec driver calls snd_soc_new_pcms(), - * once for each .dai_link in the machine driver's snd_soc_machine + * once for each .dai_link in the machine driver's snd_soc_card * structure. */ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 157a7895ffa1..52c290bb47bf 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) * If this is the first stream open, then grab the IRQ and program most of * the SSI registers. */ -static int fsl_ssi_startup(struct snd_pcm_substream *substream) +static int fsl_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +static int fsl_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) * The DMA channel is in external master start and pause mode, which * means the SSI completely controls the flow of data. */ -static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the SSI if there are no other substreams open. */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .prepare = fsl_ssi_prepare, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, - }, - .dai_ops = { .set_sysclk = fsl_ssi_set_sysclk, .set_fmt = fsl_ssi_set_fmt, }, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 86923299bc10..9ad8f9a2d8e9 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ -static int psc_i2s_startup(struct snd_pcm_substream *substream) +static int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream) } static int psc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +static int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream) * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -277,7 +281,7 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) struct mpc52xx_psc __iomem *regs = psc_i2s->psc_regs; u16 imr; u8 psc_cmd; - long flags; + unsigned long flags; if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) s = &psc_i2s->capture; @@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the PSC if there are no other substreams open. */ -static void psc_i2s_shutdown(struct snd_pcm_substream *substream) +static void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -464,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai psc_i2s_dai_template = { - .type = SND_SOC_DAI_I2S, .playback = { .channels_min = 2, .channels_max = 2, @@ -483,8 +487,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { .hw_free = psc_i2s_hw_free, .shutdown = psc_i2s_shutdown, .trigger = psc_i2s_trigger, - }, - .dai_ops = { .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }, @@ -699,9 +701,11 @@ static ssize_t psc_i2s_stat_store(struct device *dev, return count; } -DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); -DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); -DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, psc_i2s_stat_store); +static DEVICE_ATTR(status, 0644, psc_i2s_status_show, NULL); +static DEVICE_ATTR(playback_underrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); +static DEVICE_ATTR(capture_overrun, 0644, psc_i2s_stat_show, + psc_i2s_stat_store); /* --------------------------------------------------------------------- * OF platform bus binding code: @@ -819,8 +823,8 @@ static int __devinit psc_i2s_of_probe(struct of_device *op, /* Register the SYSFS files */ rc = device_create_file(psc_i2s->dev, &dev_attr_status); - rc = device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); - rc = device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_capture_overrun); + rc |= device_create_file(psc_i2s->dev, &dev_attr_playback_underrun); if (rc) dev_info(psc_i2s->dev, "error creating sysfs files\n"); diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 94f89debde1f..1cf4d6eeb538 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -29,7 +29,7 @@ struct mpc8610_hpcd_data { struct snd_soc_device sound_devdata; struct snd_soc_dai_link dai; - struct snd_soc_machine machine; + struct snd_soc_card machine; unsigned int dai_format; unsigned int codec_clk_direction; unsigned int cpu_clk_direction; @@ -185,7 +185,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { /** * mpc8610_hpcd_machine: ASoC machine data */ -static struct snd_soc_machine mpc8610_hpcd_machine = { +static struct snd_soc_card mpc8610_hpcd_machine = { .probe = mpc8610_hpcd_machine_probe, .remove = mpc8610_hpcd_machine_remove, .name = "MPC8610 HPCD", @@ -465,7 +465,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - machine_data->sound_devdata.machine = &mpc8610_hpcd_machine; + machine_data->sound_devdata.card = &mpc8610_hpcd_machine; machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; machine_data->sound_devdata.platform = &fsl_soc_platform; diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c index 0382fdac51cd..53be6491320a 100644 --- a/sound/soc/fsl/soc-of-simple.c +++ b/sound/soc/fsl/soc-of-simple.c @@ -31,7 +31,7 @@ struct of_snd_soc_device { int id; struct list_head list; struct snd_soc_device device; - struct snd_soc_machine machine; + struct snd_soc_card card; struct snd_soc_dai_link dai_link; struct platform_device *pdev; struct device_node *platform_node; @@ -58,9 +58,9 @@ of_snd_soc_get_device(struct device_node *codec_node) /* Initialize the structure and add it to the global list */ of_soc->codec_node = codec_node; of_soc->id = of_snd_soc_next_index++; - of_soc->machine.dai_link = &of_soc->dai_link; - of_soc->machine.num_links = 1; - of_soc->device.machine = &of_soc->machine; + of_soc->card.dai_link = &of_soc->dai_link; + of_soc->card.num_links = 1; + of_soc->device.card = &of_soc->card; of_soc->dai_link.ops = &of_snd_soc_ops; list_add(&of_soc->list, &of_snd_soc_device_list); @@ -159,7 +159,7 @@ int of_snd_soc_register_platform(struct snd_soc_platform *platform, of_soc->platform_node = node; of_soc->dai_link.cpu_dai = cpu_dai; of_soc->device.platform = platform; - of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + of_soc->card.name = of_soc->dai_link.cpu_dai->name; /* Now try to register the SoC device */ of_snd_soc_register_device(of_soc); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index aea27e70043c..9a869390abb9 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -13,3 +13,36 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. + +config SND_OMAP_SOC_OSK5912 + tristate "SoC Audio support for omap osk5912" + depends on SND_OMAP_SOC && MACH_OMAP_OSK + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on osk5912. + +config SND_OMAP_SOC_OVERO + tristate "SoC Audio support for Gumstix Overo" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Gumstix Overo. + +config SND_OMAP_SOC_OMAP2EVM + tristate "SoC Audio support for OMAP2EVM board" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the omap2evm board. + +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments + SDP3430. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d8d8d58075e3..29cf3a856c89 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,5 +7,13 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-osk5912-objs := osk5912.o +snd-soc-overo-objs := overo.o +snd-soc-omap2evm-objs := omap2evm.o +snd-soc-sdp3430-objs := sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o +obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d166b6b2a60d..18e2062e3a11 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -70,9 +70,13 @@ static void n810_ext_control(struct snd_soc_codec *codec) static int n810_startup(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->socdev->codec; + snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); + n810_ext_control(codec); return clk_enable(sys_clkout2); } @@ -247,9 +251,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); - snd_soc_dapm_disable_pin(codec, "HPLCOM"); - snd_soc_dapm_disable_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "HPLCOM"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -282,7 +286,7 @@ static struct snd_soc_dai_link n810_dai = { }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_n810 = { +static struct snd_soc_card snd_soc_n810 = { .name = "N810", .dai_link = &n810_dai, .num_links = 1, @@ -298,7 +302,7 @@ static struct aic3x_setup_data n810_aic33_setup = { /* Audio subsystem */ static struct snd_soc_device n810_snd_devdata = { - .machine = &snd_soc_machine_n810, + .card = &snd_soc_n810, .platform = &omap_soc_platform, .codec_dev = &soc_codec_dev_aic3x, .codec_data = &n810_aic33_setup, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 35310e16d7f3..e8f1314762d7 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -36,13 +36,12 @@ #include "omap-mcbsp.h" #include "omap-pcm.h" -#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_KNOT) +#define OMAP_MCBSP_RATES (SNDRV_PCM_RATE_8000_96000) struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -59,12 +58,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = { -{ - { .name = "I2S PCM Stereo out", }, - { .name = "I2S PCM Stereo in", }, -}, -}; +static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) static const int omap1_dma_reqs[][2] = { @@ -84,11 +78,22 @@ static const unsigned long omap1_mcbsp_port[][2] = { static const int omap1_dma_reqs[][2] = {}; static const unsigned long omap1_mcbsp_port[][2] = {}; #endif -#if defined(CONFIG_ARCH_OMAP2420) -static const int omap2420_dma_reqs[][2] = { + +#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX) +static const int omap24xx_dma_reqs[][2] = { { OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX }, { OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX }, +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) + { OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX }, + { OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX }, + { OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX }, +#endif }; +#else +static const int omap24xx_dma_reqs[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 }, @@ -96,11 +101,45 @@ static const unsigned long omap2420_mcbsp_port[][2] = { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 }, }; #else -static const int omap2420_dma_reqs[][2] = {}; static const unsigned long omap2420_mcbsp_port[][2] = {}; #endif -static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) +#if defined(CONFIG_ARCH_OMAP2430) +static const unsigned long omap2430_mcbsp_port[][2] = { + { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap2430_mcbsp_port[][2] = {}; +#endif + +#if defined(CONFIG_ARCH_OMAP34XX) +static const unsigned long omap34xx_mcbsp_port[][2] = { + { OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR, + OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap34xx_mcbsp_port[][2] = {}; +#endif + +static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -113,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) return err; } -static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) +static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -125,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) } } -static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -154,27 +195,34 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) } static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen, channels; unsigned long port; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; port = omap1_mcbsp_port[bus_id][substream->stream]; } else if (cpu_is_omap2420()) { - dma = omap2420_dma_reqs[bus_id][substream->stream]; + dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap2420_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap2430()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap2430_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap343x()) { + dma = omap24xx_dma_reqs[bus_id][substream->stream]; + port = omap34xx_mcbsp_port[bus_id][substream->stream]; } else { - /* - * TODO: Add support for 2430 and 3430 - */ return -ENODEV; } + omap_mcbsp_dai_dma_params[id][substream->stream].name = + substream->stream ? "Audio Capture" : "Audio Playback"; omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; @@ -184,12 +232,17 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } - switch (params_channels(params)) { + channels = params_channels(params); + switch (channels) { case 2: - /* Set 1 word per (McBPSP) frame and use dual-phase frames */ - regs->rcr2 |= RFRLEN2(1 - 1) | RPHASE; + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + case 1: + /* Set 1 word per (McBSP) frame */ + regs->rcr2 |= RFRLEN2(1 - 1); regs->rcr1 |= RFRLEN1(1 - 1); - regs->xcr2 |= XFRLEN2(1 - 1) | XPHASE; + regs->xcr2 |= XFRLEN2(1 - 1); regs->xcr1 |= XFRLEN1(1 - 1); break; default: @@ -200,19 +253,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr1 |= FWID(wlen * channels - 2); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -232,6 +295,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -245,6 +309,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + break; default: /* Unsupported data format */ return -EINVAL; @@ -310,7 +379,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, int clk_id) { int sel_bit; - u16 reg; + u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1; if (cpu_class_is_omap1()) { /* OMAP1's can use only external source clock */ @@ -320,6 +389,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } + if (cpu_is_omap2420() && mcbsp_data->bus_id > 1) + return -EINVAL; + + if (cpu_is_omap343x()) + reg_devconf1 = OMAP343X_CONTROL_DEVCONF1; + switch (mcbsp_data->bus_id) { case 0: reg = OMAP2_CONTROL_DEVCONF0; @@ -329,20 +404,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, reg = OMAP2_CONTROL_DEVCONF0; sel_bit = 6; break; - /* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */ + case 2: + reg = reg_devconf1; + sel_bit = 0; + break; + case 3: + reg = reg_devconf1; + sel_bit = 2; + break; + case 4: + reg = reg_devconf1; + sel_bit = 4; + break; default: return -EINVAL; } - if (cpu_class_is_omap2()) { - if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) { - omap_ctrl_writel(omap_ctrl_readl(reg) & - ~(1 << sel_bit), reg); - } else { - omap_ctrl_writel(omap_ctrl_readl(reg) | - (1 << sel_bit), reg); - } - } + if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) + omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg); + else + omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg); return 0; } @@ -376,37 +457,46 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } -struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { -{ - .name = "omap-mcbsp-dai", - .id = 0, - .type = SND_SOC_DAI_I2S, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = { - .startup = omap_mcbsp_dai_startup, - .shutdown = omap_mcbsp_dai_shutdown, - .trigger = omap_mcbsp_dai_trigger, - .hw_params = omap_mcbsp_dai_hw_params, - }, - .dai_ops = { - .set_fmt = omap_mcbsp_dai_set_dai_fmt, - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, - }, - .private_data = &mcbsp_data[0].bus_id, -}, +#define OMAP_MCBSP_DAI_BUILDER(link_id) \ +{ \ + .name = "omap-mcbsp-dai-"#link_id, \ + .id = (link_id), \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = OMAP_MCBSP_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .ops = { \ + .startup = omap_mcbsp_dai_startup, \ + .shutdown = omap_mcbsp_dai_shutdown, \ + .trigger = omap_mcbsp_dai_trigger, \ + .hw_params = omap_mcbsp_dai_hw_params, \ + .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ + }, \ + .private_data = &mcbsp_data[(link_id)].bus_id, \ +} + +struct snd_soc_dai omap_mcbsp_dai[] = { + OMAP_MCBSP_DAI_BUILDER(0), + OMAP_MCBSP_DAI_BUILDER(1), +#if NUM_LINKS >= 3 + OMAP_MCBSP_DAI_BUILDER(2), +#endif +#if NUM_LINKS == 5 + OMAP_MCBSP_DAI_BUILDER(3), + OMAP_MCBSP_DAI_BUILDER(4), +#endif }; + EXPORT_SYMBOL_GPL(omap_mcbsp_dai); MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>"); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ed8afb550671..df7ad13ba73d 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -38,11 +38,17 @@ enum omap_mcbsp_div { OMAP_MCBSP_CLKGDV, /* Sample rate generator divider */ }; -/* - * REVISIT: Preparation for the ASoC v2. Let the number of available links to - * be same than number of McBSP ports found in OMAP(s) we are compiling for. - */ -#define NUM_LINKS 1 +#if defined(CONFIG_ARCH_OMAP2420) +#define NUM_LINKS 2 +#endif +#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX) +#undef NUM_LINKS +#define NUM_LINKS 3 +#endif +#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX) +#undef NUM_LINKS +#define NUM_LINKS 5 +#endif extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 690bfeaec4a0..e9084fdd2082 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!cpu_is_omap1510()) { + if (!err & !cpu_is_omap1510()) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_or_dst_synch = OMAP_DMA_DST_SYNC; dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; + dma_params.dst_port = OMAP_DMA_PORT_MPUI; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC; dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; + dma_params.src_port = OMAP_DMA_PORT_MPUI; } /* * Set DMA transfer frame size equal to ALSA period size and frame diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c new file mode 100644 index 000000000000..7b160f9d83f9 --- /dev/null +++ b/sound/soc/omap/omap2evm.c @@ -0,0 +1,151 @@ +/* + * omap2evm.c -- SoC audio machine driver for omap2evm board + * + * Author: Arun KS <arunks@mistralsolutions.com> + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap2evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap2evm_ops = { + .hw_params = omap2evm_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap2evm_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap2evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap2evm = { + .name = "omap2evm", + .dai_link = &omap2evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap2evm_snd_devdata = { + .card = &snd_soc_omap2evm, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap2evm_snd_device; + +static int __init omap2evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap2evm()) { + pr_debug("Not omap2evm!\n"); + return -ENODEV; + } + printk(KERN_INFO "omap2evm SoC init\n"); + + omap2evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap2evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap2evm_snd_device, &omap2evm_snd_devdata); + omap2evm_snd_devdata.dev = &omap2evm_snd_device->dev; + *(unsigned int *)omap2evm_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap2evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap2evm_snd_device); + + return ret; +} +module_init(omap2evm_soc_init); + +static void __exit omap2evm_soc_exit(void) +{ + platform_device_unregister(omap2evm_snd_device); +} +module_exit(omap2evm_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC omap2evm"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c new file mode 100644 index 000000000000..3ed25464627f --- /dev/null +++ b/sound/soc/omap/omap3beagle.c @@ -0,0 +1,149 @@ +/* + * omap3beagle.c -- SoC audio for OMAP3 Beagle + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int omap3beagle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops omap3beagle_ops = { + .hw_params = omap3beagle_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3beagle_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3beagle_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_omap3beagle = { + .name = "omap3beagle", + .dai_link = &omap3beagle_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3beagle_snd_devdata = { + .card = &snd_soc_omap3beagle, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3beagle_snd_device; + +static int __init omap3beagle_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_beagle()) { + pr_debug("Not OMAP3 Beagle!\n"); + return -ENODEV; + } + pr_info("OMAP3 Beagle SoC init\n"); + + omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap3beagle_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(omap3beagle_snd_device, &omap3beagle_snd_devdata); + omap3beagle_snd_devdata.dev = &omap3beagle_snd_device->dev; + *(unsigned int *)omap3beagle_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(omap3beagle_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(omap3beagle_snd_device); + + return ret; +} + +static void __exit omap3beagle_soc_exit(void) +{ + platform_device_unregister(omap3beagle_snd_device); +} + +module_init(omap3beagle_soc_init); +module_exit(omap3beagle_soc_exit); + +MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c new file mode 100644 index 000000000000..7a8f14d0c772 --- /dev/null +++ b/sound/soc/omap/osk5912.c @@ -0,0 +1,232 @@ +/* + * osk5912.c -- SoC audio for OSK 5912 + * + * Copyright (C) 2008 Mistral Solutions + * + * Contact: Arun KS <arunks@mistralsolutions.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <linux/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static struct clk *tlv320aic23_mclk; + +static int osk_startup(struct snd_pcm_substream *substream) +{ + return clk_enable(tlv320aic23_mclk); +} + +static void osk_shutdown(struct snd_pcm_substream *substream) +{ + clk_disable(tlv320aic23_mclk); +} + +static int osk_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err; + + /* Set codec DAI configuration */ + err = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return err; + } + + /* Set cpu DAI configuration */ + err = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM); + if (err < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return err; + } + + /* Set the codec system clock for DAC and ADC */ + err = + snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); + + if (err < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return err; + } + + return err; +} + +static struct snd_soc_ops osk_ops = { + .startup = osk_startup, + .hw_params = osk_hw_params, + .shutdown = osk_shutdown, +}; + +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "LHPOUT"}, + {"Headphone Jack", NULL, "RHPOUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic Jack"}, +}; + +static int osk_tlv320aic23_init(struct snd_soc_codec *codec) +{ + + /* Add osk5912 specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up osk5912 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link osk_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = osk_tlv320aic23_init, + .ops = &osk_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_osk = { + .name = "OSK5912", + .dai_link = &osk_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device osk_snd_devdata = { + .card = &snd_soc_card_osk, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *osk_snd_device; + +static int __init osk_soc_init(void) +{ + int err; + u32 curRate; + struct device *dev; + + if (!(machine_is_omap_osk())) + return -ENODEV; + + osk_snd_device = platform_device_alloc("soc-audio", -1); + if (!osk_snd_device) + return -ENOMEM; + + platform_set_drvdata(osk_snd_device, &osk_snd_devdata); + osk_snd_devdata.dev = &osk_snd_device->dev; + *(unsigned int *)osk_dai.cpu_dai->private_data = 0; /* McBSP1 */ + err = platform_device_add(osk_snd_device); + if (err) + goto err1; + + dev = &osk_snd_device->dev; + + tlv320aic23_mclk = clk_get(dev, "mclk"); + if (IS_ERR(tlv320aic23_mclk)) { + printk(KERN_ERR "Could not get mclk clock\n"); + return -ENODEV; + } + + if (clk_get_usecount(tlv320aic23_mclk) > 0) { + /* MCLK is already in use */ + printk(KERN_WARNING + "MCLK in use at %d Hz. We change it to %d Hz\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); + } + + /* + * Configure 12 MHz output on MCLK. + */ + curRate = (uint) clk_get_rate(tlv320aic23_mclk); + if (curRate != CODEC_CLOCK) { + if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) { + printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n"); + err = -ECANCELED; + goto err1; + } + } + + printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, + clk_get_usecount(tlv320aic23_mclk)); + + return 0; +err1: + clk_put(tlv320aic23_mclk); + platform_device_del(osk_snd_device); + platform_device_put(osk_snd_device); + + return err; + +} + +static void __exit osk_soc_exit(void) +{ + platform_device_unregister(osk_snd_device); +} + +module_init(osk_soc_init); +module_exit(osk_soc_exit); + +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_DESCRIPTION("ALSA SoC OSK 5912"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c new file mode 100644 index 000000000000..eea0c372bb3f --- /dev/null +++ b/sound/soc/omap/overo.c @@ -0,0 +1,148 @@ +/* + * overo.c -- SoC audio for Gumstix Overo + * + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int overo_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops overo_ops = { + .hw_params = overo_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link overo_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &overo_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_overo = { + .name = "overo", + .dai_link = &overo_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device overo_snd_devdata = { + .card = &snd_soc_card_overo, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *overo_snd_device; + +static int __init overo_soc_init(void) +{ + int ret; + + if (!machine_is_overo()) { + pr_debug("Not Overo!\n"); + return -ENODEV; + } + printk(KERN_INFO "overo SoC init\n"); + + overo_snd_device = platform_device_alloc("soc-audio", -1); + if (!overo_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(overo_snd_device, &overo_snd_devdata); + overo_snd_devdata.dev = &overo_snd_device->dev; + *(unsigned int *)overo_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(overo_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(overo_snd_device); + + return ret; +} +module_init(overo_soc_init); + +static void __exit overo_soc_exit(void) +{ + platform_device_unregister(overo_snd_device); +} +module_exit(overo_soc_exit); + +MODULE_AUTHOR("Steve Sakoman <steve@sakoman.com>"); +MODULE_DESCRIPTION("ALSA SoC overo"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 000000000000..85fd160bca17 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729@ti.com> + * + * Based on: + * Author: Steve Sakoman <steve@sakoman.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f8c1cdd940ac..f82e10699471 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,6 +21,9 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate +config SND_PXA_SOC_SSP + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx @@ -75,3 +78,22 @@ config SND_PXA2XX_SOC_EM_X270 help Say Y if you want to add support for SoC audio on CompuLab EM-x270. + +config SND_PXA2XX_SOC_PALM27X + bool "SoC Audio support for Palm T|X, T5 and LifeDrive" + depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5) + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + Palm T|X, T5 or LifeDrive handheld computer. + +config SND_SOC_ZYLONITE + tristate "SoC Audio support for Marvell Zylonite" + depends on SND_PXA2XX_SOC && MACH_ZYLONITE + select SND_PXA2XX_SOC_AC97 + select SND_PXA_SOC_SSP + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + Marvell Zylonite reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 5bc8edf9dca9..08a9f2797729 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,10 +2,12 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o +snd-soc-pxa-ssp-objs := pxa-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o +obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o @@ -14,6 +16,8 @@ snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o +snd-soc-palm27x-objs := palm27x.o +snd-soc-zylonite-objs := zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -21,3 +25,5 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o +obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 72b7a5140bf8..647f056a3cb3 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -18,13 +18,13 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> #include <mach/corgi.h> @@ -54,8 +54,8 @@ static void corgi_ext_control(struct snd_soc_codec *codec) switch (corgi_jack_func) { case CORGI_HP: /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); @@ -63,24 +63,24 @@ static void corgi_ext_control(struct snd_soc_codec *codec) break; case CORGI_MIC: /* reset = mute headphone */ - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 0); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 0); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_disable_pin(codec, "Headphone Jack"); @@ -114,8 +114,8 @@ static int corgi_shutdown(struct snd_pcm_substream *substream) struct snd_soc_codec *codec = rtd->socdev->codec; /* set = unmute headphone */ - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + gpio_set_value(CORGI_GPIO_MUTE_L, 1); + gpio_set_value(CORGI_GPIO_MUTE_R, 1); return 0; } @@ -218,22 +218,14 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol, static int corgi_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); - + gpio_set_value(CORGI_GPIO_APM_ON, SND_SOC_DAPM_EVENT_ON(event)); return 0; } static int corgi_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - else - reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); - + gpio_set_value(CORGI_GPIO_MIC_BIAS, SND_SOC_DAPM_EVENT_ON(event)); return 0; } @@ -289,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -322,7 +314,7 @@ static struct snd_soc_dai_link corgi_dai = { }; /* corgi audio machine driver */ -static struct snd_soc_machine snd_soc_machine_corgi = { +static struct snd_soc_card snd_soc_corgi = { .name = "Corgi", .dai_link = &corgi_dai, .num_links = 1, @@ -336,7 +328,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = { /* corgi audio subsystem */ static struct snd_soc_device corgi_snd_devdata = { - .machine = &snd_soc_machine_corgi, + .card = &snd_soc_corgi, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm8731, .codec_data = &corgi_wm8731_setup, diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 6781c5be242f..60c64861512a 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -29,7 +29,7 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_machine e800; +static struct snd_soc_card e800; static struct snd_soc_dai_link e800_dai[] = { { @@ -40,14 +40,14 @@ static struct snd_soc_dai_link e800_dai[] = { }, }; -static struct snd_soc_machine e800 = { +static struct snd_soc_card e800 = { .name = "Toshiba e800", .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), }; static struct snd_soc_device e800_snd_devdata = { - .machine = &e800, + .card = &e800, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm9712, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index d9c3f7b28be2..4a61925c3104 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -9,7 +9,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -53,14 +53,14 @@ static struct snd_soc_dai_link em_x270_dai[] = { }, }; -static struct snd_soc_machine em_x270 = { +static struct snd_soc_card em_x270 = { .name = "EM-X270", .dai_link = em_x270_dai, .num_links = ARRAY_SIZE(em_x270_dai), }; static struct snd_soc_device em_x270_snd_devdata = { - .machine = &em_x270, + .card = &em_x270, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm9712, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c new file mode 100644 index 000000000000..3bb8879ac8a2 --- /dev/null +++ b/sound/soc/pxa/palm27x.c @@ -0,0 +1,269 @@ +/* + * linux/sound/soc/pxa/palm27x.c + * + * SoC Audio driver for Palm T|X, T5 and LifeDrive + * + * based on tosa.c + * + * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/irq.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> +#include <mach/palmasoc.h> + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int palm27x_jack_func = 1; +static int palm27x_spk_func = 1; +static int palm27x_ep_gpio = -1; + +static void palm27x_ext_control(struct snd_soc_codec *codec) +{ + if (!palm27x_spk_func) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + + if (!palm27x_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + + snd_soc_dapm_sync(codec); +} + +static int palm27x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->codec; + + /* check the jack status at stream startup */ + palm27x_ext_control(codec); + return 0; +} + +static struct snd_soc_ops palm27x_ops = { + .startup = palm27x_startup, +}; + +static irqreturn_t palm27x_interrupt(int irq, void *v) +{ + palm27x_spk_func = gpio_get_value(palm27x_ep_gpio); + palm27x_jack_func = !palm27x_spk_func; + return IRQ_HANDLED; +} + +static int palm27x_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = palm27x_jack_func; + return 0; +} + +static int palm27x_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (palm27x_jack_func == ucontrol->value.integer.value[0]) + return 0; + + palm27x_jack_func = ucontrol->value.integer.value[0]; + palm27x_ext_control(codec); + return 1; +} + +static int palm27x_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = palm27x_spk_func; + return 0; +} + +static int palm27x_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (palm27x_spk_func == ucontrol->value.integer.value[0]) + return 0; + + palm27x_spk_func = ucontrol->value.integer.value[0]; + palm27x_ext_control(codec); + return 1; +} + +/* PalmTX machine dapm widgets */ +static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* PalmTX audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* headphone connected to HPOUTL, HPOUTR */ + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + + /* ext speaker connected to ROUT2, LOUT2 */ + {"Speaker", NULL, "LOUT2"}, + {"Speaker", NULL, "ROUT2"}, +}; + +static const char *jack_function[] = {"Headphone", "Off"}; +static const char *spk_function[] = {"On", "Off"}; +static const struct soc_enum palm27x_enum[] = { + SOC_ENUM_SINGLE_EXT(2, jack_function), + SOC_ENUM_SINGLE_EXT(2, spk_function), +}; + +static const struct snd_kcontrol_new palm27x_controls[] = { + SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack, + palm27x_set_jack), + SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk, + palm27x_set_spk), +}; + +static int palm27x_ac97_init(struct snd_soc_codec *codec) +{ + int i, err; + + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); + + /* add palm27x specific controls */ + for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&palm27x_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + /* add palm27x specific widgets */ + snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, + ARRAY_SIZE(palm27x_dapm_widgets)); + + /* set up palm27x specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link palm27x_dai[] = { +{ + .name = "AC97 HiFi", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = palm27x_ac97_init, + .ops = &palm27x_ops, +}, +{ + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + .ops = &palm27x_ops, +}, +}; + +static struct snd_soc_card palm27x_asoc = { + .name = "Palm/PXA27x", + .dai_link = palm27x_dai, + .num_links = ARRAY_SIZE(palm27x_dai), +}; + +static struct snd_soc_device palm27x_snd_devdata = { + .card = &palm27x_asoc, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *palm27x_snd_device; + +static int __init palm27x_asoc_init(void) +{ + int ret; + + if (!(machine_is_palmtx() || machine_is_palmt5() || + machine_is_palmld())) + return -ENODEV; + + ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_input(palm27x_ep_gpio); + if (ret) + goto err_alloc; + + if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + "Headphone jack", NULL)) + goto err_alloc; + + palm27x_snd_device = platform_device_alloc("soc-audio", -1); + if (!palm27x_snd_device) { + ret = -ENOMEM; + goto err_dev; + } + + platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata); + palm27x_snd_devdata.dev = &palm27x_snd_device->dev; + ret = platform_device_add(palm27x_snd_device); + + if (ret != 0) + goto put_device; + + return 0; + +put_device: + platform_device_put(palm27x_snd_device); +err_dev: + free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); +err_alloc: + gpio_free(palm27x_ep_gpio); + + return ret; +} + +static void __exit palm27x_asoc_exit(void) +{ + free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); + gpio_free(palm27x_ep_gpio); + platform_device_unregister(palm27x_snd_device); +} + +void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) +{ + palm27x_ep_gpio = data->jack_gpio; +} + +module_init(palm27x_asoc_init); +module_exit(palm27x_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index f84f7d8db09a..03b510ab2824 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "LLINEIN"); - snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_nc_pin(codec, "LLINEIN"); + snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ @@ -276,7 +276,7 @@ static struct snd_soc_dai_link poodle_dai = { }; /* poodle audio machine driver */ -static struct snd_soc_machine snd_soc_machine_poodle = { +static struct snd_soc_card snd_soc_poodle = { .name = "Poodle", .dai_link = &poodle_dai, .num_links = 1, @@ -290,7 +290,7 @@ static struct wm8731_setup_data poodle_wm8731_setup = { /* poodle audio subsystem */ static struct snd_soc_device poodle_snd_devdata = { - .machine = &snd_soc_machine_poodle, + .card = &snd_soc_poodle, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm8731, .codec_data = &poodle_wm8731_setup, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c new file mode 100644 index 000000000000..402fc5ba65e7 --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.c @@ -0,0 +1,921 @@ +#define DEBUG +/* + * pxa-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/pxa-regs.h> +#include <mach/regs-ssp.h> +#include <mach/audio.h> +#include <mach/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +/* + * SSP audio private data + */ +struct ssp_priv { + struct ssp_dev dev; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + struct ssp_state state; +#endif +}; + +#define PXA2xx_SSP1_BASE 0x41000000 +#define PXA27x_SSP2_BASE 0x41700000 +#define PXA27x_SSP3_BASE 0x41900000 +#define PXA3xx_SSP4_BASE 0x41a00000 + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_out = { + .name = "SSP1 PCM Mono out", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(14), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_mono_in = { + .name = "SSP1 PCM Mono in", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(13), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_out = { + .name = "SSP1 PCM Stereo out", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(14), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp1_pcm_stereo_in = { + .name = "SSP1 PCM Stereo in", + .dev_addr = PXA2xx_SSP1_BASE + SSDR, + .drcmr = &DRCMR(13), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_out = { + .name = "SSP2 PCM Mono out", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(16), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_mono_in = { + .name = "SSP2 PCM Mono in", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(15), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_out = { + .name = "SSP2 PCM Stereo out", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(16), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp2_pcm_stereo_in = { + .name = "SSP2 PCM Stereo in", + .dev_addr = PXA27x_SSP2_BASE + SSDR, + .drcmr = &DRCMR(15), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_out = { + .name = "SSP3 PCM Mono out", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_mono_in = { + .name = "SSP3 PCM Mono in", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_out = { + .name = "SSP3 PCM Stereo out", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp3_pcm_stereo_in = { + .name = "SSP3 PCM Stereo in", + .dev_addr = PXA27x_SSP3_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_out = { + .name = "SSP4 PCM Mono out", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_mono_in = { + .name = "SSP4 PCM Mono in", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH2, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_out = { + .name = "SSP4 PCM Stereo out", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(67), + .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static struct pxa2xx_pcm_dma_params pxa_ssp4_pcm_stereo_in = { + .name = "SSP4 PCM Stereo in", + .dev_addr = PXA3xx_SSP4_BASE + SSDR, + .drcmr = &DRCMR(66), + .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | + DCMD_BURST16 | DCMD_WIDTH4, +}; + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), + ssp_read_reg(ssp, SSTO)); + + dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", + ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), + ssp_read_reg(ssp, SSACD)); +} + +static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { + { + &pxa_ssp1_pcm_mono_out, &pxa_ssp1_pcm_mono_in, + &pxa_ssp1_pcm_stereo_out, &pxa_ssp1_pcm_stereo_in, + }, + { + &pxa_ssp2_pcm_mono_out, &pxa_ssp2_pcm_mono_in, + &pxa_ssp2_pcm_stereo_out, &pxa_ssp2_pcm_stereo_in, + }, + { + &pxa_ssp3_pcm_mono_out, &pxa_ssp3_pcm_mono_in, + &pxa_ssp3_pcm_stereo_out, &pxa_ssp3_pcm_stereo_in, + }, + { + &pxa_ssp4_pcm_mono_out, &pxa_ssp4_pcm_mono_in, + &pxa_ssp4_pcm_stereo_out, &pxa_ssp4_pcm_stereo_in, + }, +}; + +static int pxa_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + int ret = 0; + + if (!cpu_dai->active) { + ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); + if (ret < 0) + return ret; + ssp_disable(&priv->dev); + } + return ret; +} + +static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) { + ssp_disable(&priv->dev); + ssp_exit(&priv->dev); + } +} + +#ifdef CONFIG_PM + +static int pxa_ssp_suspend(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) + return 0; + + ssp_save_state(&priv->dev, &priv->state); + clk_disable(priv->dev.ssp->clk); + return 0; +} + +static int pxa_ssp_resume(struct platform_device *pdev, + struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) + return 0; + + clk_enable(priv->dev.ssp->clk); + ssp_restore_state(&priv->dev, &priv->state); + ssp_enable(&priv->dev); + + return 0; +} + +#else +#define pxa_ssp_suspend NULL +#define pxa_ssp_resume NULL +#endif + +/** + * ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void ssp_set_scr(struct ssp_dev *dev, u32 div) +{ + struct ssp_device *ssp = dev->ssp; + u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + + ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA_SSP_CLK_AUDIO: + priv->sysclk = 0; + ssp_set_scr(&priv->dev, 1); + sscr0 |= SSCR0_ADC; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(priv->dev.ssp->clk); + val = ssp_read_reg(ssp, SSCR0) | sscr0; + ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(priv->dev.ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + switch (div_id) { + case PXA_SSP_AUDIO_DIV_ACDS: + val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + ssp_write_reg(ssp, SSACD, val); + break; + case PXA_SSP_AUDIO_DIV_SCDB: + val = ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + ssp_write_reg(ssp, SSACD, val); + break; + case PXA_SSP_DIV_SCR: + ssp_set_scr(&priv->dev, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64;; + ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %dHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +/* + * Set the active slots in TDM/Network mode + */ +static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int mask, int slots) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr0; + + sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, mask); + ssp_write_reg(ssp, SSRSA, mask); + return 0; +} + +/* + * Tristate the SSP DAI lines + */ +static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, + int tristate) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr1; + + sscr1 = ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* reset port settings */ + sscr0 = ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_FSRT; + break; + case SND_SOC_DAIFMT_NB_IF: + sspsp |= SSPSP_SFRMP | SSPSP_FSRT; + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + + dump_registers(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->dai_fmt = fmt; + + return 0; +} + +/* + * Set the SSP audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int dma = 0, chn = params_channels(params); + u32 sscr0; + u32 sspsp; + int width = snd_pcm_format_physical_width(params_format(params)); + + /* select correct DMA params */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + dma = 1; /* capture DMA offset is 1,3 */ + if (chn == 2) + dma += 2; /* stereo DMA offset is 2, mono is 0 */ + cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma]; + + dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma); + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + return 0; + + /* clear selected SSP bits */ + sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + ssp_write_reg(ssp, SSCR0, sscr0); + + /* bit size */ + sscr0 = ssp_read_reg(ssp, SSCR0); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: +#ifdef CONFIG_PXA3xx + if (cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; +#endif + sscr0 |= SSCR0_DataSize(16); + if (params_channels(params) > 1) + sscr0 |= SSCR0_EDSS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); + /* we must be in network mode (2 slots) for 24 bit stereo */ + break; + case SNDRV_PCM_FORMAT_S32_LE: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); + /* we must be in network mode (2 slots) for 32 bit stereo */ + break; + } + ssp_write_reg(ssp, SSCR0, sscr0); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Cleared when the DAI format is set */ + sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + ssp_write_reg(ssp, SSPSP, sspsp); + break; + default: + break; + } + + /* We always use a network mode so we always require TDM slots + * - complain loudly and fail if they've not been set up yet. + */ + if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); + return -EINVAL; + } + + dump_registers(ssp); + + return 0; +} + +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->dev.ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ssp_enable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + val = ssp_read_reg(ssp, SSSR); + ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + ssp_enable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_STOP: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ssp_disable(&priv->dev); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + + default: + ret = -EINVAL; + } + + dump_registers(ssp); + + return ret; +} + +static int pxa_ssp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv; + int ret; + + priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + if (priv->dev.ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + + dai->private_data = priv; + + return 0; + +err_priv: + kfree(priv); + return ret; +} + +static void pxa_ssp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv = dai->private_data; + ssp_free(priv->dev.ssp); +} + +#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai pxa_ssp_dai[] = { + { + .name = "pxa2xx-ssp1", + .id = 0, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { .name = "pxa2xx-ssp2", + .id = 1, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { + .name = "pxa2xx-ssp3", + .id = 2, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, + { + .name = "pxa2xx-ssp4", + .id = 3, + .probe = pxa_ssp_probe, + .remove = pxa_ssp_remove, + .suspend = pxa_ssp_suspend, + .resume = pxa_ssp_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = PXA_SSP_RATES, + .formats = PXA_SSP_FORMATS, + }, + .ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(pxa_ssp_dai); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h new file mode 100644 index 000000000000..91deadd55675 --- /dev/null +++ b/sound/soc/pxa/pxa-ssp.h @@ -0,0 +1,47 @@ +/* + * ASoC PXA SSP port support + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _PXA_SSP_H +#define _PXA_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA_DAI_SSP1 0 +#define PXA_DAI_SSP2 1 +#define PXA_DAI_SSP3 2 +#define PXA_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA_SSP_CLK_PLL 0 +#define PXA_SSP_CLK_EXT 1 +#define PXA_SSP_CLK_NET 2 +#define PXA_SSP_CLK_AUDIO 3 +#define PXA_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA_SSP_AUDIO_DIV_ACDS 0 +#define PXA_SSP_AUDIO_DIV_SCDB 1 +#define PXA_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA_SSP_CLK_AUDIO_DIV_1 0 +#define PXA_SSP_CLK_AUDIO_DIV_2 1 +#define PXA_SSP_CLK_AUDIO_DIV_4 2 +#define PXA_SSP_CLK_AUDIO_DIV_8 3 +#define PXA_SSP_CLK_AUDIO_DIV_16 4 +#define PXA_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA_SSP_CLK_SCDB_4 0 +#define PXA_SSP_CLK_SCDB_1 1 +#define PXA_SSP_CLK_SCDB_8 2 + +#define PXA_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa_ssp_dai[4]; + +#endif diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a80ae074b090..bffbe288634c 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -49,7 +49,7 @@ struct snd_ac97_bus_ops soc_ac97_ops = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { .name = "AC97 PCM Stereo out", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRTXPCDR, + .drcmr = &DRCMR(12), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -57,7 +57,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { .name = "AC97 PCM Stereo in", .dev_addr = __PREG(PCDR), - .drcmr = &DRCMRRXPCDR, + .drcmr = &DRCMR(11), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -65,7 +65,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { .name = "AC97 Aux PCM (Slot 5) Mono out", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRTXMODR, + .drcmr = &DRCMR(10), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -73,7 +73,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { .name = "AC97 Aux PCM (Slot 5) Mono in", .dev_addr = __PREG(MODR), - .drcmr = &DRCMRRXMODR, + .drcmr = &DRCMR(9), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -81,7 +81,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { .name = "AC97 Mic PCM (Slot 6) Mono in", .dev_addr = __PREG(MCDR), - .drcmr = &DRCMRRXMCDR, + .drcmr = &DRCMR(8), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST16 | DCMD_WIDTH2, }; @@ -117,7 +117,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev, } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -131,7 +132,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -145,7 +147,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -170,7 +173,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = pxa2xx_ac97_probe, .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, @@ -193,7 +196,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-aux", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -212,7 +215,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 2, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 39d19212f6d3..f9a9e2ebafa1 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -3,7 +3,7 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * lrg@slimlogic.co.uk * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -39,6 +39,45 @@ struct pxa2xx_gpio { u32 frm; }; +/* + * I2S Controller Register and Bit Definitions + */ +#define SACR0 __REG(0x40400000) /* Global Control Register */ +#define SACR1 __REG(0x40400004) /* Serial Audio I 2 S/MSB-Justified Control Register */ +#define SASR0 __REG(0x4040000C) /* Serial Audio I 2 S/MSB-Justified Interface and FIFO Status Register */ +#define SAIMR __REG(0x40400014) /* Serial Audio Interrupt Mask Register */ +#define SAICR __REG(0x40400018) /* Serial Audio Interrupt Clear Register */ +#define SADIV __REG(0x40400060) /* Audio Clock Divider Register. */ +#define SADR __REG(0x40400080) /* Serial Audio Data Register (TX and RX FIFO access Register). */ + +#define SACR0_RFTH(x) ((x) << 12) /* Rx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_TFTH(x) ((x) << 8) /* Tx FIFO Interrupt or DMA Trigger Threshold */ +#define SACR0_STRF (1 << 5) /* FIFO Select for EFWR Special Function */ +#define SACR0_EFWR (1 << 4) /* Enable EFWR Function */ +#define SACR0_RST (1 << 3) /* FIFO, i2s Register Reset */ +#define SACR0_BCKD (1 << 2) /* Bit Clock Direction */ +#define SACR0_ENB (1 << 0) /* Enable I2S Link */ +#define SACR1_ENLBF (1 << 5) /* Enable Loopback */ +#define SACR1_DRPL (1 << 4) /* Disable Replaying Function */ +#define SACR1_DREC (1 << 3) /* Disable Recording Function */ +#define SACR1_AMSL (1 << 0) /* Specify Alternate Mode */ + +#define SASR0_I2SOFF (1 << 7) /* Controller Status */ +#define SASR0_ROR (1 << 6) /* Rx FIFO Overrun */ +#define SASR0_TUR (1 << 5) /* Tx FIFO Underrun */ +#define SASR0_RFS (1 << 4) /* Rx FIFO Service Request */ +#define SASR0_TFS (1 << 3) /* Tx FIFO Service Request */ +#define SASR0_BSY (1 << 2) /* I2S Busy */ +#define SASR0_RNE (1 << 1) /* Rx FIFO Not Empty */ +#define SASR0_TNF (1 << 0) /* Tx FIFO Not Empty */ + +#define SAICR_ROR (1 << 6) /* Clear Rx FIFO Overrun Interrupt */ +#define SAICR_TUR (1 << 5) /* Clear Tx FIFO Underrun Interrupt */ + +#define SAIMR_ROR (1 << 6) /* Enable Rx FIFO Overrun Condition Interrupt */ +#define SAIMR_TUR (1 << 5) /* Enable Tx FIFO Underrun Condition Interrupt */ +#define SAIMR_RFS (1 << 4) /* Enable Rx FIFO Service Interrupt */ +#define SAIMR_TFS (1 << 3) /* Enable Tx FIFO Service Interrupt */ struct pxa_i2s_port { u32 sadiv; @@ -54,7 +93,7 @@ static struct clk *clk_i2s; static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { .name = "I2S PCM Stereo out", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRTXSADR, + .drcmr = &DRCMR(3), .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -62,7 +101,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { .name = "I2S PCM Stereo in", .dev_addr = __PREG(SADR), - .drcmr = &DRCMRRXSADR, + .drcmr = &DRCMR(2), .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | DCMD_BURST32 | DCMD_WIDTH4, }; @@ -82,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = { }, }; -static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) +static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -148,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -209,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -230,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { SACR1 |= SACR1_DRPL; @@ -297,7 +340,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .suspend = pxa2xx_i2s_suspend, .resume = pxa2xx_i2s_resume, .playback = { @@ -314,8 +356,7 @@ struct snd_soc_dai pxa_i2s_dai = { .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = pxa2xx_i2s_hw_params, .set_fmt = pxa2xx_i2s_set_dai_fmt, .set_sysclk = pxa2xx_i2s_set_dai_sysclk, }, @@ -366,6 +407,6 @@ module_init(pxa2xx_i2s_init); module_exit(pxa2xx_i2s_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index afcd892cd2fa..0f6b7bb2d44b 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -69,7 +69,7 @@ static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -struct snd_pcm_ops pxa2xx_pcm_ops = { +static struct snd_pcm_ops pxa2xx_pcm_ops = { .open = __pxa2xx_pcm_open, .close = __pxa2xx_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 3d4738c06e7e..579d93368f14 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -19,16 +19,15 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <asm/hardware/scoop.h> #include <mach/pxa-regs.h> #include <mach/hardware.h> -#include <mach/akita.h> #include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" @@ -63,8 +62,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 1); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ @@ -72,8 +71,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Mic Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ @@ -81,8 +80,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_enable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ @@ -90,8 +89,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); snd_soc_dapm_enable_pin(codec, "Headset Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_HP_OFF: @@ -100,8 +99,8 @@ static void spitz_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "Headset Jack"); snd_soc_dapm_disable_pin(codec, "Mic Jack"); snd_soc_dapm_disable_pin(codec, "Line Jack"); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); - reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); + gpio_set_value(SPITZ_GPIO_MUTE_L, 0); + gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; } snd_soc_dapm_sync(codec); @@ -215,23 +214,14 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol, static int spitz_mic_bias(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - if (machine_is_borzoi() || machine_is_spitz()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - set_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - else - reset_scoop_gpio(&spitzscoop2_device.dev, - SPITZ_SCP2_MIC_BIAS); - } + if (machine_is_borzoi() || machine_is_spitz()) + gpio_set_value(SPITZ_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); + + if (machine_is_akita()) + gpio_set_value(AKITA_GPIO_MIC_BIAS, + SND_SOC_DAPM_EVENT_ON(event)); - if (machine_is_akita()) { - if (SND_SOC_DAPM_EVENT_ON(event)) - akita_set_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - else - akita_reset_ioexp(&akitaioexp_device.dev, - AKITA_IOEXP_MIC_BIAS); - } return 0; } @@ -291,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_disable_pin(codec, "RINPUT1"); - snd_soc_dapm_disable_pin(codec, "LINPUT2"); - snd_soc_dapm_disable_pin(codec, "RINPUT2"); - snd_soc_dapm_disable_pin(codec, "LINPUT3"); - snd_soc_dapm_disable_pin(codec, "RINPUT3"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "RINPUT1"); + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -329,7 +319,7 @@ static struct snd_soc_dai_link spitz_dai = { }; /* spitz audio machine driver */ -static struct snd_soc_machine snd_soc_machine_spitz = { +static struct snd_soc_card snd_soc_spitz = { .name = "Spitz", .dai_link = &spitz_dai, .num_links = 1, @@ -343,7 +333,7 @@ static struct wm8750_setup_data spitz_wm8750_setup = { /* spitz audio subsystem */ static struct snd_soc_device spitz_snd_devdata = { - .machine = &snd_soc_machine_spitz, + .card = &snd_soc_spitz, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm8750, .codec_data = &spitz_wm8750_setup, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 2baaa750f123..48242b32a28b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -4,7 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Authors: Liam Girdwood <lrg@slimlogic.co.uk> * Richard Purdie <richard@openedhand.com> * * This program is free software; you can redistribute it and/or modify it @@ -38,7 +38,7 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_machine tosa; +static struct snd_soc_card tosa; #define TOSA_HP 0 #define TOSA_MIC_INT 1 @@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -230,14 +230,36 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; -static struct snd_soc_machine tosa = { +static int tosa_probe(struct platform_device *dev) +{ + int ret; + + ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); + if (ret) + gpio_free(TOSA_GPIO_L_MUTE); + + return ret; +} + +static int tosa_remove(struct platform_device *dev) +{ + gpio_free(TOSA_GPIO_L_MUTE); + return 0; +} + +static struct snd_soc_card tosa = { .name = "Tosa", .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + .probe = tosa_probe, + .remove = tosa_remove, }; static struct snd_soc_device tosa_snd_devdata = { - .machine = &tosa, + .card = &tosa, .platform = &pxa2xx_soc_platform, .codec_dev = &soc_codec_dev_wm9712, }; @@ -251,11 +273,6 @@ static int __init tosa_init(void) if (!machine_is_tosa()) return -ENODEV; - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - tosa_snd_device = platform_device_alloc("soc-audio", -1); if (!tosa_snd_device) { ret = -ENOMEM; @@ -272,15 +289,12 @@ static int __init tosa_init(void) platform_device_put(tosa_snd_device); err_alloc: - gpio_free(TOSA_GPIO_L_MUTE); - return ret; } static void __exit tosa_exit(void) { platform_device_unregister(tosa_snd_device); - gpio_free(TOSA_GPIO_L_MUTE); } module_init(tosa_init); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c new file mode 100644 index 000000000000..842d6500d61f --- /dev/null +++ b/sound/soc/pxa/zylonite.c @@ -0,0 +1,219 @@ +/* + * zylonite.c -- SoC audio for Zylonite + * + * Copyright 2008 Wolfson Microelectronics PLC. + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm9713.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "pxa-ssp.h" + +static struct snd_soc_card zylonite; + +static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Microphone", NULL), + SND_SOC_DAPM_MIC("Handset Microphone", NULL), + SND_SOC_DAPM_SPK("Multiactor", NULL), + SND_SOC_DAPM_SPK("Headset Earpiece", NULL), +}; + +/* Currently supported audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Headphone output connected to HPL/HPR */ + { "Headphone", NULL, "HPL" }, + { "Headphone", NULL, "HPR" }, + + /* On-board earpiece */ + { "Headset Earpiece", NULL, "OUT3" }, + + /* Headphone mic */ + { "MIC2A", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Headset Microphone" }, + + /* On-board mic */ + { "MIC1", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "Handset Microphone" }, + + /* Multiactor differentially connected over SPKL/SPKR */ + { "Multiactor", NULL, "SPKL" }, + { "Multiactor", NULL, "SPKR" }, +}; + +static int zylonite_wm9713_init(struct snd_soc_codec *codec) +{ + /* Currently we only support use of the AC97 clock here. If + * CLK_POUT is selected by SW15 then the clock API will need + * to be used to request and enable it here. + */ + + snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, + ARRAY_SIZE(zylonite_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Static setup for now */ + snd_soc_dapm_enable_pin(codec, "Headphone"); + snd_soc_dapm_enable_pin(codec, "Headset Earpiece"); + + snd_soc_dapm_sync(codec); + return 0; +} + +static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0; + unsigned int acds = 0; + unsigned int wm9713_div = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + wm9713_div = 12; + pll_out = 2048000; + break; + case 16000: + wm9713_div = 6; + pll_out = 4096000; + break; + case 48000: + default: + wm9713_div = 2; + pll_out = 12288000; + acds = 1; + break; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, + params_channels(params), + params_channels(params)); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + if (ret < 0) + return ret; + + /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs + * to be set instead. + */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, + WM9713_PCMDIV(wm9713_div)); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops zylonite_voice_ops = { + .hw_params = zylonite_voice_hw_params, +}; + +static struct snd_soc_dai_link zylonite_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = zylonite_wm9713_init, +}, +{ + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], +}, +{ + .name = "WM9713 Voice", + .stream_name = "WM9713 Voice", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], + .ops = &zylonite_voice_ops, +}, +}; + +static struct snd_soc_card zylonite = { + .name = "Zylonite", + .dai_link = zylonite_dai, + .num_links = ARRAY_SIZE(zylonite_dai), +}; + +static struct snd_soc_device zylonite_snd_ac97_devdata = { + .card = &zylonite, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *zylonite_snd_ac97_device; + +static int __init zylonite_init(void) +{ + int ret; + + zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!zylonite_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(zylonite_snd_ac97_device, + &zylonite_snd_ac97_devdata); + zylonite_snd_ac97_devdata.dev = &zylonite_snd_ac97_device->dev; + + ret = platform_device_add(zylonite_snd_ac97_device); + if (ret != 0) + platform_device_put(zylonite_snd_ac97_device); + + return ret; +} + +static void __exit zylonite_exit(void) +{ + platform_device_unregister(zylonite_snd_ac97_device); +} + +module_init(zylonite_init); +module_exit(zylonite_exit); + +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index b9f2353effeb..fcd03acf10f6 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -44,3 +44,8 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 Say Y if you want to add support for SoC audio on ln2440sbc with the ALC650. +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_UDA134X diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 0aa5fb0b9700..96b3f3f617d4 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -13,7 +13,9 @@ obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 4eab2c19c454..a70cbc0fa070 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -27,7 +27,7 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" -static struct snd_soc_machine ln2440sbc; +static struct snd_soc_card ln2440sbc; static struct snd_soc_dai_link ln2440sbc_dai[] = { { @@ -38,14 +38,14 @@ static struct snd_soc_dai_link ln2440sbc_dai[] = { }, }; -static struct snd_soc_machine ln2440sbc = { +static struct snd_soc_card ln2440sbc = { .name = "LN2440SBC", .dai_link = ln2440sbc_dai, .num_links = ARRAY_SIZE(ln2440sbc_dai), }; static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { - .machine = &ln2440sbc, + .card = &ln2440sbc, .platform = &s3c24xx_soc_platform, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 73a50e93a9a2..3df2224a6723 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -59,7 +59,7 @@ #define NEO_CAPTURE_HEADSET 7 #define NEO_CAPTURE_BLUETOOTH 8 -static struct snd_soc_machine neo1973; +static struct snd_soc_card neo1973; static struct i2c_client *i2c; static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, @@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_disable_pin(codec, "LOUT2"); - snd_soc_dapm_disable_pin(codec, "ROUT2"); - snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "OUT4"); - snd_soc_dapm_disable_pin(codec, "LINE1"); - snd_soc_dapm_disable_pin(codec, "LINE2"); - - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); /* Add neo1973 specific widgets */ snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, ARRAY_SIZE(wm8753_dapm_widgets)); + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { err = snd_ctl_add(codec->card, @@ -549,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, - .type = SND_SOC_DAI_PCM, .playback = { .channels_min = 1, .channels_max = 1, @@ -580,7 +578,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { }, }; -static struct snd_soc_machine neo1973 = { +static struct snd_soc_card neo1973 = { .name = "neo1973", .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), @@ -592,7 +590,7 @@ static struct wm8753_setup_data neo1973_wm8753_setup = { }; static struct snd_soc_device neo1973_snd_devdata = { - .machine = &neo1973, + .card = &neo1973, .platform = &s3c24xx_soc_platform, .codec_dev = &soc_codec_dev_wm8753, .codec_data = &neo1973_wm8753_setup, @@ -603,6 +601,8 @@ static int lm4857_i2c_probe(struct i2c_client *client, { DBG("Entered %s\n", __func__); + i2c = client; + lm4857_write_regs(); return 0; } @@ -611,6 +611,8 @@ static int lm4857_i2c_remove(struct i2c_client *client) { DBG("Entered %s\n", __func__); + i2c = NULL; + return 0; } @@ -650,7 +652,7 @@ static void lm4857_shutdown(struct i2c_client *dev) } static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 } + { "neo1973_lm4857", 0 }, { } }; @@ -668,48 +670,6 @@ static struct i2c_driver lm4857_i2c_driver = { }; static struct platform_device *neo1973_snd_device; -static struct i2c_client *lm4857_client; - -static int __init neo1973_add_lm4857_device(struct platform_device *pdev, - int i2c_bus, - unsigned short i2c_address) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&lm4857_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add lm4857 driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = i2c_address; - strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - lm4857_client = client; - return 0; - -err_driver: - i2c_del_driver(&lm4857_i2c_driver); - return -ENODEV; -} static int __init neo1973_init(void) { @@ -736,8 +696,8 @@ static int __init neo1973_init(void) return ret; } - ret = neo1973_add_lm4857_device(neo1973_snd_device, - neo1973_wm8753_setup, 0x7C); + ret = i2c_add_driver(&lm4857_i2c_driver); + if (ret != 0) platform_device_unregister(neo1973_snd_device); @@ -748,7 +708,6 @@ static void __exit neo1973_exit(void) { DBG("Entered %s\n", __func__); - i2c_unregister_device(lm4857_client); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ded7d995a922..1c741047ae35 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; @@ -711,7 +713,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c2412_i2s_probe, .suspend = s3c2412_i2s_suspend, .resume = s3c2412_i2s_resume, @@ -730,8 +731,6 @@ struct snd_soc_dai s3c2412_i2s_dai = { .ops = { .trigger = s3c2412_i2s_trigger, .hw_params = s3c2412_i2s_hw_params, - }, - .dai_ops = { .set_fmt = s3c2412_i2s_set_fmt, .set_clkdiv = s3c2412_i2s_set_clkdiv, .set_sysclk = s3c2412_i2s_set_sysclk, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 19c5c3cf5d8c..41bde6a3883b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev, } static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -313,7 +314,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) } static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -327,7 +329,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, } static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; @@ -356,7 +358,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = s3c2443_ac97_probe, .remove = s3c2443_ac97_remove, .playback = { @@ -378,7 +380,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b55fbc..8d9135f41bc9 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -261,10 +262,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; + default: + return -EINVAL; } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -272,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -452,7 +461,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c24xx_i2s_probe, .suspend = s3c24xx_i2s_suspend, .resume = s3c24xx_i2s_resume, @@ -468,8 +476,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = s3c24xx_i2s_hw_params, .set_fmt = s3c24xx_i2s_set_fmt, .set_clkdiv = s3c24xx_i2s_set_clkdiv, .set_sysclk = s3c24xx_i2s_set_sysclk, diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c new file mode 100644 index 000000000000..23325fca1f64 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -0,0 +1,373 @@ +/* + * Modifications by Christian Pellegrin <chripell@evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime;; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_card snd_soc_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .card = &snd_soc_s3c24xx_uda134x, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell@evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index 8515d6ff03f2..3d2e6a0417ec 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -23,7 +23,7 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" -static struct snd_soc_machine smdk2443; +static struct snd_soc_card smdk2443; static struct snd_soc_dai_link smdk2443_dai[] = { { @@ -34,14 +34,14 @@ static struct snd_soc_dai_link smdk2443_dai[] = { }, }; -static struct snd_soc_machine smdk2443 = { +static struct snd_soc_card smdk2443 = { .name = "SMDK2443", .dai_link = smdk2443_dai, .num_links = ARRAY_SIZE(smdk2443_dai), }; static struct snd_soc_device smdk2443_snd_ac97_devdata = { - .machine = &smdk2443, + .card = &smdk2443, .platform = &s3c24xx_soc_platform, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index df7bc345c320..c435913c60eb 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; @@ -270,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, @@ -290,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = { #ifdef CONFIG_CPU_SUBTYPE_SH7760 { .name = "HAC1", + .ac97_control = 1, .id = 1, - .type = SND_SOC_DAI_AC97, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 92bfaf4774a7..8b44f9c8a9ff 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -38,14 +38,14 @@ static struct snd_soc_dai_link sh7760_ac97_dai = { .ops = NULL, }; -static struct snd_soc_machine sh7760_ac97_soc_machine = { +static struct snd_soc_card sh7760_ac97_soc_machine = { .name = "SH7760 AC97", .dai_link = &sh7760_ac97_dai, .num_links = 1, }; static struct snd_soc_device sh7760_ac97_snd_devdata = { - .machine = &sh7760_ac97_soc_machine, + .card = &sh7760_ac97_soc_machine, .platform = &sh7760_soc_platform, .codec_dev = &soc_codec_dev_ac97, }; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 55c3464163ab..fed544a3deff 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -89,7 +89,8 @@ struct ssi_priv { * track usage of the SSI; it is simplex-only so prevent attempts of * concurrent playback + capture. FIXME: any locking required? */ -static int ssi_startup(struct snd_pcm_substream *substream) +static int ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream) return 0; } -static void ssi_shutdown(struct snd_pcm_substream *substream) +static void ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream) ssi->inuse = 0; } -static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) } static int ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -336,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, @@ -354,8 +357,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, @@ -365,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI1", .id = 1, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, @@ -383,8 +383,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad381138fc2e..13b4aaff0e9c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4,8 +4,7 @@ * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * with code, comments and ideas from :- * Richard Purdie <richard@openedhand.com> * @@ -27,6 +26,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/bitops.h> +#include <linux/debugfs.h> #include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -35,14 +35,6 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> -/* debug */ -#define SOC_DEBUG 0 -#if SOC_DEBUG -#define dbg(format, arg...) printk(format, ## arg) -#else -#define dbg(format, arg...) -#endif - static DEFINE_MUTEX(pcm_mutex); static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); @@ -96,8 +88,8 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.parent = NULL; codec->ac97->dev.release = soc_ac97_device_release; - snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", - codec->card->number, 0, codec->name); + dev_set_name(&codec->ac97->dev, "%d-%d:%s", + codec->card->number, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); @@ -108,20 +100,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char *get_dai_name(int type) -{ - switch (type) { - case SND_SOC_DAI_AC97_BUS: - case SND_SOC_DAI_AC97: - return "AC97"; - case SND_SOC_DAI_I2S: - return "I2S"; - case SND_SOC_DAI_PCM: - return "PCM"; - } - return NULL; -} - /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -142,7 +120,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* startup the audio subsystem */ if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream); + ret = cpu_dai->ops.startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -159,7 +137,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream); + ret = codec_dai->ops.startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -229,12 +207,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } - dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); - dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); - dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, - runtime->hw.channels_max); - dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, - runtime->hw.rate_max); + pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); + pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); + pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, + runtime->hw.channels_max); + pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, + runtime->hw.rate_max); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->playback.active = codec_dai->playback.active = 1; @@ -256,7 +234,7 @@ codec_dai_err: platform_err: if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -279,18 +257,18 @@ static void close_delayed_work(struct work_struct *work) for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - dbg("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->playback.stream_name, - codec_dai->playback.active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->playback.stream_name, + codec_dai->playback.active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { /* Reduce power if no longer active */ if (codec->active == 0) { - dbg("pop wq D1 %s %s\n", codec->name, - codec_dai->playback.stream_name); + pr_debug("pop wq D1 %s %s\n", codec->name, + codec_dai->playback.stream_name); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); } @@ -302,8 +280,8 @@ static void close_delayed_work(struct work_struct *work) /* Fall into standby if no longer active */ if (codec->active == 0) { - dbg("pop wq D3 %s %s\n", codec->name, - codec_dai->playback.stream_name); + pr_debug("pop wq D3 %s %s\n", codec->name, + codec_dai->playback.stream_name); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); } @@ -347,10 +325,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 1); if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream); + codec_dai->ops.shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -414,7 +392,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream); + ret = codec_dai->ops.prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; @@ -422,58 +400,49 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream); + ret = cpu_dai->ops.prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; } } - /* we only want to start a DAPM playback stream if we are not waiting - * on an existing one stopping */ - if (codec_dai->pop_wait) { - /* we are waiting for the delayed work to start */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - snd_soc_dapm_stream_event(socdev->codec, - codec_dai->capture.stream_name, - SND_SOC_DAPM_STREAM_START); - else { - codec_dai->pop_wait = 0; - cancel_delayed_work(&socdev->delayed_work); - snd_soc_dai_digital_mute(codec_dai, 0); - } - } else { - /* no delayed work - do we need to power up codec */ - if (codec->bias_level != SND_SOC_BIAS_ON) { + /* cancel any delayed stream shutdown that is pending */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + codec_dai->pop_wait) { + codec_dai->pop_wait = 0; + cancel_delayed_work(&socdev->delayed_work); + } - snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); + /* do we need to power up codec */ + if (codec->bias_level != SND_SOC_BIAS_ON) { + snd_soc_dapm_set_bias_level(socdev, + SND_SOC_BIAS_PREPARE); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, + else + snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - snd_soc_dai_digital_mute(codec_dai, 0); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); + snd_soc_dai_digital_mute(codec_dai, 0); - } else { - /* codec already powered - power on widgets */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, + } else { + /* codec already powered - power on widgets */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(codec, + else + snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - snd_soc_dai_digital_mute(codec_dai, 0); - } + snd_soc_dai_digital_mute(codec_dai, 0); } out: @@ -508,7 +477,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params); + ret = codec_dai->ops.hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -517,7 +486,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params); + ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -540,11 +509,11 @@ out: platform_err: if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); interface_err: if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -583,10 +552,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAI's */ if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -603,7 +572,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) int ret; if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd); + ret = codec_dai->ops.trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -615,7 +584,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd); + ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -637,7 +606,7 @@ static struct snd_pcm_ops soc_pcm_ops = { static int soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; @@ -654,22 +623,22 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); /* mute any active DAC's */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 1); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 1); } /* suspend all pcms */ - for (i = 0; i < machine->num_links; i++) - snd_pcm_suspend_all(machine->dai_link[i].pcm); + for (i = 0; i < card->num_links; i++) + snd_pcm_suspend_all(card->dai_link[i].pcm); - if (machine->suspend_pre) - machine->suspend_pre(pdev, state); + if (card->suspend_pre) + card->suspend_pre(pdev, state); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->suspend && !cpu_dai->ac97_control) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) platform->suspend(pdev, cpu_dai); @@ -693,14 +662,14 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) if (codec_dev->suspend) codec_dev->suspend(pdev, state); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->suspend && cpu_dai->ac97_control) cpu_dai->suspend(pdev, cpu_dai); } - if (machine->suspend_post) - machine->suspend_post(pdev, state); + if (card->suspend_post) + card->suspend_post(pdev, state); return 0; } @@ -713,7 +682,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, deferred_resume_work); - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; struct snd_soc_codec *codec = socdev->codec; @@ -724,14 +693,14 @@ static void soc_resume_deferred(struct work_struct *work) * so userspace apps are blocked from touching us */ - dev_info(socdev->dev, "starting resume work\n"); + dev_dbg(socdev->dev, "starting resume work\n"); - if (machine->resume_pre) - machine->resume_pre(pdev); + if (card->resume_pre) + card->resume_pre(pdev); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->resume && cpu_dai->ac97_control) cpu_dai->resume(pdev, cpu_dai); } @@ -750,24 +719,24 @@ static void soc_resume_deferred(struct work_struct *work) } /* unmute any active DACs */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 0); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 0); } - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->resume && !cpu_dai->ac97_control) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) platform->resume(pdev, cpu_dai); } - if (machine->resume_post) - machine->resume_post(pdev); + if (card->resume_post) + card->resume_post(pdev); - dev_info(socdev->dev, "resume work completed\n"); + dev_dbg(socdev->dev, "resume work completed\n"); /* userspace can access us now we are back as we were before */ snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); @@ -778,10 +747,10 @@ static int soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - dev_info(socdev->dev, "scheduling resume work\n"); + dev_dbg(socdev->dev, "scheduling resume work\n"); if (!schedule_work(&socdev->deferred_resume_work)) - dev_err(socdev->dev, "work item may be lost\n"); + dev_err(socdev->dev, "resume work item may be lost\n"); return 0; } @@ -796,18 +765,18 @@ static int soc_probe(struct platform_device *pdev) { int ret = 0, i; struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (machine->probe) { - ret = machine->probe(pdev); + if (card->probe) { + ret = card->probe(pdev); if (ret < 0) return ret; } - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -842,13 +811,13 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } - if (machine->remove) - machine->remove(pdev); + if (card->remove) + card->remove(pdev); return ret; } @@ -858,7 +827,7 @@ static int soc_remove(struct platform_device *pdev) { int i; struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = socdev->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; @@ -870,14 +839,14 @@ static int soc_remove(struct platform_device *pdev) if (codec_dev->remove) codec_dev->remove(pdev); - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } - if (machine->remove) - machine->remove(pdev); + if (card->remove) + card->remove(pdev); return 0; } @@ -915,8 +884,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, - get_dai_name(cpu_dai->type), num); + sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, + num); if (codec_dai->playback.channels_min) playback = 1; @@ -962,10 +931,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; @@ -1002,8 +969,117 @@ static ssize_t codec_reg_show(struct device *dev, return count; } +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata, buf); +} + static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); +#ifdef CONFIG_DEBUG_FS +static int codec_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_device *devdata = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(devdata, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t codec_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_device *devdata = file->private_data; + struct snd_soc_codec *codec = devdata->codec; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = codec_reg_open_file, + .read = codec_reg_read_file, + .write = codec_reg_write_file, +}; + +static void soc_init_debugfs(struct snd_soc_device *socdev) +{ + struct dentry *root, *file; + struct snd_soc_codec *codec = socdev->codec; + root = debugfs_create_dir(dev_name(socdev->dev), NULL); + if (IS_ERR(root) || !root) + goto exit1; + + file = debugfs_create_file("codec_reg", 0644, + root, socdev, &codec_reg_fops); + if (!file) + goto exit2; + + file = debugfs_create_u32("dapm_pop_time", 0744, + root, &codec->pop_time); + if (!file) + goto exit2; + socdev->debugfs_root = root; + return; +exit2: + debugfs_remove_recursive(root); +exit1: + dev_err(socdev->dev, "debugfs is not available\n"); +} + +static void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ + debugfs_remove_recursive(socdev->debugfs_root); + socdev->debugfs_root = NULL; +} + +#else + +static inline void soc_init_debugfs(struct snd_soc_device *socdev) +{ +} + +static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev) +{ +} +#endif + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1122,7 +1198,7 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0, i; mutex_lock(&codec->mutex); @@ -1141,11 +1217,11 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); /* create the pcms */ - for (i = 0; i < machine->num_links; i++) { - ret = soc_new_pcm(socdev, &machine->dai_link[i], i); + for (i = 0; i < card->num_links; i++) { + ret = soc_new_pcm(socdev, &card->dai_link[i], i); if (ret < 0) { printk(KERN_ERR "asoc: can't create pcm %s\n", - machine->dai_link[i].stream_name); + card->dai_link[i].stream_name); mutex_unlock(&codec->mutex); return ret; } @@ -1168,26 +1244,25 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms); int snd_soc_register_card(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0, i, ac97 = 0, err = 0; - for (i = 0; i < machine->num_links; i++) { - if (socdev->machine->dai_link[i].init) { - err = socdev->machine->dai_link[i].init(codec); + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].init) { + err = card->dai_link[i].init(codec); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", - socdev->machine->dai_link[i].stream_name); + card->dai_link[i].stream_name); continue; } } - if (socdev->machine->dai_link[i].codec_dai->type == - SND_SOC_DAI_AC97_BUS) + if (card->dai_link[i].codec_dai->ac97_control) ac97 = 1; } snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", machine->name); + "%s", card->name); snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", machine->name, codec->name); + "%s (%s)", card->name, codec->name); ret = snd_card_register(codec->card); if (ret < 0) { @@ -1217,6 +1292,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + soc_init_debugfs(socdev); mutex_unlock(&codec->mutex); out: @@ -1240,10 +1316,11 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #endif mutex_lock(&codec->mutex); + soc_cleanup_debugfs(socdev); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { + if (codec_dai->ac97_control && codec->ac97) { soc_ac97_dev_unregister(codec); goto free_card; } @@ -1463,7 +1540,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - unsigned int shift = mc->min; + unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; if (max == 1) @@ -1757,8 +1834,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops.set_sysclk) + return dai->ops.set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -1777,8 +1854,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->dai_ops.set_clkdiv) - return dai->dai_ops.set_clkdiv(dai, div_id, div); + if (dai->ops.set_clkdiv) + return dai->ops.set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -1796,8 +1873,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->dai_ops.set_pll) - return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops.set_pll) + return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -1806,15 +1883,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @clk_id: DAI specific clock ID * @fmt: SND_SOC_DAIFMT_ format value. * * Configures the DAI hardware format and clocking. */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->dai_ops.set_fmt) - return dai->dai_ops.set_fmt(dai, fmt); + if (dai->ops.set_fmt) + return dai->ops.set_fmt(dai, fmt); else return -EINVAL; } @@ -1832,8 +1908,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tdm_slot(dai, mask, slots); + if (dai->ops.set_sysclk) + return dai->ops.set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -1848,8 +1924,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tristate(dai, tristate); + if (dai->ops.set_sysclk) + return dai->ops.set_tristate(dai, tristate); else return -EINVAL; } @@ -1864,8 +1940,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->dai_ops.digital_mute) - return dai->dai_ops.digital_mute(dai, mute); + if (dai->ops.digital_mute) + return dai->ops.digital_mute(dai, mute); else return -EINVAL; } @@ -1873,7 +1949,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); static int __devinit snd_soc_init(void) { - printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); return platform_driver_register(&soc_driver); } @@ -1886,7 +1961,7 @@ module_init(snd_soc_init); module_exit(snd_soc_exit); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9ca9c08610fa..61d7d85aa578 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2,8 +2,7 @@ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management * * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -38,7 +37,6 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> -#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -68,17 +66,13 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); -static struct dentry *asoc_debugfs; - -static u32 pop_time; - -static void pop_wait(void) +static void pop_wait(u32 pop_time) { if (pop_time) schedule_timeout_uninterruptible(msecs_to_jiffies(pop_time)); } -static void pop_dbg(const char *fmt, ...) +static void pop_dbg(u32 pop_time, const char *fmt, ...) { va_list args; @@ -86,7 +80,7 @@ static void pop_dbg(const char *fmt, ...) if (pop_time) { vprintk(fmt, args); - pop_wait(); + pop_wait(pop_time); } va_end(args); @@ -231,10 +225,11 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) change = old != new; if (change) { - pop_dbg("pop test %s : %s in %d ms\n", widget->name, - widget->power ? "on" : "off", pop_time); + pop_dbg(codec->pop_time, "pop test %s : %s in %d ms\n", + widget->name, widget->power ? "on" : "off", + codec->pop_time); snd_soc_write(codec, widget->reg, new); - pop_wait(); + pop_wait(codec->pop_time); } pr_debug("reg %x old %x new %x change %d\n", widget->reg, old, new, change); @@ -294,7 +289,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *w) { int i, ret = 0; - char name[32]; + size_t name_len; struct snd_soc_dapm_path *path; /* add kcontrol */ @@ -308,11 +303,16 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, continue; /* add dapm control with long name */ - snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); - path->long_name = kstrdup (name, GFP_KERNEL); + name_len = 2 + strlen(w->name) + + strlen(w->kcontrols[i].name); + path->long_name = kmalloc(name_len, GFP_KERNEL); if (path->long_name == NULL) return -ENOMEM; + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); + path->long_name[name_len - 1] = '\0'; + path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, path->long_name); ret = snd_ctl_add(codec->card, path->kcontrol); @@ -822,23 +822,9 @@ static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); int snd_soc_dapm_sys_add(struct device *dev) { - int ret = 0; - if (!dapm_status) return 0; - - ret = device_create_file(dev, &dev_attr_dapm_widget); - if (ret != 0) - return ret; - - asoc_debugfs = debugfs_create_dir("asoc", NULL); - if (!IS_ERR(asoc_debugfs)) - debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, - &pop_time); - else - asoc_debugfs = NULL; - - return 0; + return device_create_file(dev, &dev_attr_dapm_widget); } static void snd_soc_dapm_sys_remove(struct device *dev) @@ -846,9 +832,6 @@ static void snd_soc_dapm_sys_remove(struct device *dev) if (dapm_status) { device_remove_file(dev, &dev_attr_dapm_widget); } - - if (asoc_debugfs) - debugfs_remove_recursive(asoc_debugfs); } /* free all dapm widgets and resources */ @@ -1008,28 +991,6 @@ err: } /** - * snd_soc_dapm_connect_input - connect dapm widgets - * @codec: audio codec - * @sink: name of target widget - * @control: mixer control name - * @source: name of source name - * - * Connects 2 dapm widgets together via a named audio path. The sink is - * the widget receiving the audio signal, whilst the source is the sender - * of the audio signal. - * - * This function has been deprecated in favour of snd_soc_dapm_add_routes(). - * - * Returns 0 for success else error. - */ -int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, - const char *control, const char *source) -{ - return snd_soc_dapm_add_route(codec, sink, control, source); -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); - -/** * snd_soc_dapm_add_routes - Add routes between DAPM widgets * @codec: codec * @route: audio routes @@ -1441,11 +1402,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, enum snd_soc_bias_level level) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_card *card = socdev->card; int ret = 0; - if (machine->set_bias_level) - ret = machine->set_bias_level(machine, level); + if (card->set_bias_level) + ret = card->set_bias_level(card, level); if (ret == 0 && codec->set_bias_level) ret = codec->set_bias_level(codec, level); @@ -1484,6 +1445,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** + * snd_soc_dapm_nc_pin - permanently disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); + +/** * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec * @pin: audio signal pin endpoint (or start point) @@ -1521,6 +1502,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) EXPORT_SYMBOL_GPL(snd_soc_dapm_free); /* Module information */ -MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); MODULE_LICENSE("GPL"); diff --git a/sound/sound_core.c b/sound/sound_core.c index 4ae07e236b36..10ba4218161b 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -57,7 +57,7 @@ module_exit(cleanup_soundcore); /* * OSS sound core handling. Breaks out sound functions to submodules * - * Author: Alan Cox <alan.cox@linux.org> + * Author: Alan Cox <alan@lxorguk.ukuu.org.uk> * * Fixes: * @@ -220,9 +220,8 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati else sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP); - device_create_drvdata(sound_class, dev, - MKDEV(SOUND_MAJOR, s->unit_minor), - NULL, s->name+6); + device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), + NULL, s->name+6); return r; fail: @@ -458,7 +457,7 @@ EXPORT_SYMBOL(unregister_sound_mixer); void unregister_sound_midi(int unit) { - return sound_remove_unit(&chains[2], unit); + sound_remove_unit(&chains[2], unit); } EXPORT_SYMBOL(unregister_sound_midi); @@ -475,7 +474,7 @@ EXPORT_SYMBOL(unregister_sound_midi); void unregister_sound_dsp(int unit) { - return sound_remove_unit(&chains[3], unit); + sound_remove_unit(&chains[3], unit); } @@ -508,7 +507,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit) return NULL; } -int soundcore_open(struct inode *inode, struct file *file) +static int soundcore_open(struct inode *inode, struct file *file) { int chain; int unit = iminor(inode); diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 49acee0c4840..f87933e48812 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1,6 +1,6 @@ /* * Driver for AMD7930 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller <davem@redhat.com> + * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net> * * Based entirely upon drivers/sbus/audio/amd7930.c which is: * Copyright (C) 1996,1997 Thomas K. Dyas (tdyas@eden.rutgers.edu) @@ -35,6 +35,8 @@ #include <linux/init.h> #include <linux/interrupt.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -44,7 +46,6 @@ #include <asm/io.h> #include <asm/irq.h> -#include <asm/sbus.h> #include <asm/prom.h> static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ @@ -335,8 +336,8 @@ struct snd_amd7930 { int pgain; int mgain; + struct of_device *op; unsigned int irq; - unsigned int regs_size; struct snd_amd7930 *next; }; @@ -905,13 +906,16 @@ static int __devinit snd_amd7930_mixer(struct snd_amd7930 *amd) static int snd_amd7930_free(struct snd_amd7930 *amd) { + struct of_device *op = amd->op; + amd7930_idle(amd); if (amd->irq) free_irq(amd->irq, amd); if (amd->regs) - sbus_iounmap(amd->regs, amd->regs_size); + of_iounmap(&op->resource[0], amd->regs, + resource_size(&op->resource[0])); kfree(amd); @@ -930,13 +934,12 @@ static struct snd_device_ops snd_amd7930_dev_ops = { }; static int __devinit snd_amd7930_create(struct snd_card *card, - struct resource *rp, - unsigned int reg_size, + struct of_device *op, int irq, int dev, struct snd_amd7930 **ramd) { - unsigned long flags; struct snd_amd7930 *amd; + unsigned long flags; int err; *ramd = NULL; @@ -946,9 +949,10 @@ static int __devinit snd_amd7930_create(struct snd_card *card, spin_lock_init(&amd->lock); amd->card = card; - amd->regs_size = reg_size; + amd->op = op; - amd->regs = sbus_ioremap(rp, 0, amd->regs_size, "amd7930"); + amd->regs = of_ioremap(&op->resource[0], 0, + resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -997,12 +1001,15 @@ static int __devinit snd_amd7930_create(struct snd_card *card, return 0; } -static int __devinit amd7930_attach_common(struct resource *rp, int irq) +static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_device_id *match) { + struct resource *rp = &op->resource[0]; static int dev_num; struct snd_card *card; struct snd_amd7930 *amd; - int err; + int err, irq; + + irq = op->irqs[0]; if (dev_num >= SNDRV_CARDS) return -ENODEV; @@ -1023,8 +1030,7 @@ static int __devinit amd7930_attach_common(struct resource *rp, int irq) (unsigned long long)rp->start, irq); - if ((err = snd_amd7930_create(card, rp, - (rp->end - rp->start) + 1, + if ((err = snd_amd7930_create(card, op, irq, dev_num, &amd)) < 0) goto out_err; @@ -1049,43 +1055,7 @@ out_err: return err; } -static int __devinit amd7930_obio_attach(struct device_node *dp) -{ - const struct linux_prom_registers *regs; - const struct linux_prom_irqs *irqp; - struct resource res, *rp; - int len; - - irqp = of_get_property(dp, "intr", &len); - if (!irqp) { - snd_printk("%s: Firmware node lacks IRQ property.\n", - dp->full_name); - return -ENODEV; - } - - regs = of_get_property(dp, "reg", &len); - if (!regs) { - snd_printk("%s: Firmware node lacks register property.\n", - dp->full_name); - return -ENODEV; - } - - rp = &res; - rp->start = regs->phys_addr; - rp->end = rp->start + regs->reg_size - 1; - rp->flags = IORESOURCE_IO | (regs->which_io & 0xff); - - return amd7930_attach_common(rp, irqp->pri); -} - -static int __devinit amd7930_sbus_probe(struct of_device *dev, const struct of_device_id *match) -{ - struct sbus_dev *sdev = to_sbus_device(&dev->dev); - - return amd7930_attach_common(&sdev->resource[0], sdev->irqs[0]); -} - -static struct of_device_id amd7930_match[] = { +static const struct of_device_id amd7930_match[] = { { .name = "audio", }, @@ -1100,20 +1070,7 @@ static struct of_platform_driver amd7930_sbus_driver = { static int __init amd7930_init(void) { - struct device_node *dp; - - /* Try to find the sun4c "audio" node first. */ - dp = of_find_node_by_path("/"); - dp = dp->child; - while (dp) { - if (!strcmp(dp->name, "audio")) - amd7930_obio_attach(dp); - - dp = dp->sibling; - } - - /* Probe each SBUS for amd7930 chips. */ - return of_register_driver(&amd7930_sbus_driver, &sbus_bus_type); + return of_register_driver(&amd7930_sbus_driver, &of_bus_type); } static void __exit amd7930_exit(void) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 791d2fb821d1..d44bf98e965e 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -1,6 +1,6 @@ /* * Driver for CS4231 sound chips found on Sparcs. - * Copyright (C) 2002 David S. Miller <davem@redhat.com> + * Copyright (C) 2002, 2008 David S. Miller <davem@davemloft.net> * * Based entirely upon drivers/sbus/audio/cs4231.c which is: * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow@andrew.cmu.edu) @@ -17,7 +17,8 @@ #include <linux/moduleparam.h> #include <linux/irq.h> #include <linux/io.h> - +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,13 +30,12 @@ #ifdef CONFIG_SBUS #define SBUS_SUPPORT -#include <asm/sbus.h> #endif #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) #define EBUS_SUPPORT #include <linux/pci.h> -#include <asm/ebus.h> +#include <asm/ebus_dma.h> #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ @@ -70,8 +70,6 @@ struct cs4231_dma_control { int (*request)(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len); unsigned int (*address)(struct cs4231_dma_control *dma_cont); - void (*preallocate)(struct snd_cs4231 *chip, - struct snd_pcm *pcm); #ifdef EBUS_SUPPORT struct ebus_dma_info ebus_info; #endif @@ -114,21 +112,12 @@ struct snd_cs4231 { struct mutex mce_mutex; /* mutex for mce register */ struct mutex open_mutex; /* mutex for ALSA open/close */ - union { -#ifdef SBUS_SUPPORT - struct sbus_dev *sdev; -#endif -#ifdef EBUS_SUPPORT - struct pci_dev *pdev; -#endif - } dev_u; + struct of_device *op; unsigned int irq[2]; unsigned int regs_size; struct snd_cs4231 *next; }; -static struct snd_cs4231 *cs4231_list; - /* Eventually we can use sound/isa/cs423x/cs4231_lib.c directly, but for * now.... -DaveM */ @@ -267,27 +256,19 @@ static unsigned char snd_cs4231_original_image[32] = static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return readb(reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_readb(reg_addr); -#endif } static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) { -#ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) return writeb(val, reg_addr); else -#endif -#ifdef SBUS_SUPPORT return sbus_writeb(val, reg_addr); -#endif } /* @@ -1258,7 +1239,9 @@ static int __init snd_cs4231_pcm(struct snd_card *card) pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; strcpy(pcm->name, "CS4231"); - chip->p_dma.preallocate(chip, pcm); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->op->dev, + 64 * 1024, 128 * 1024); chip->pcm = pcm; @@ -1627,8 +1610,7 @@ static int __init cs4231_attach_finish(struct snd_card *card) if (err < 0) goto out_err; - chip->next = cs4231_list; - cs4231_list = chip; + dev_set_drvdata(&chip->op->dev, chip); dev++; return 0; @@ -1783,24 +1765,19 @@ static unsigned int sbus_dma_addr(struct cs4231_dma_control *dma_cont) return sbus_readl(base->regs + base->dir + APCVA); } -static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_SBUS, - snd_dma_sbus_data(chip->dev_u.sdev), - 64 * 1024, 128 * 1024); -} - /* * Init and exit routines */ static int snd_cs4231_sbus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->op; + if (chip->irq[0]) free_irq(chip->irq[0], chip); if (chip->port) - sbus_iounmap(chip->port, chip->regs_size); + of_iounmap(&op->resource[0], chip->port, chip->regs_size); return 0; } @@ -1817,7 +1794,7 @@ static struct snd_device_ops snd_cs4231_sbus_dev_ops = { }; static int __init snd_cs4231_sbus_create(struct snd_card *card, - struct sbus_dev *sdev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -1828,13 +1805,13 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->dev_u.sdev = sdev; - chip->regs_size = sdev->reg_addrs[0].reg_size; + chip->op = op; + chip->regs_size = resource_size(&op->resource[0]); memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - chip->port = sbus_ioremap(&sdev->resource[0], 0, - chip->regs_size, "cs4231"); + chip->port = of_ioremap(&op->resource[0], 0, + chip->regs_size, "cs4231"); if (!chip->port) { snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -1849,22 +1826,20 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, chip->p_dma.enable = sbus_dma_enable; chip->p_dma.request = sbus_dma_request; chip->p_dma.address = sbus_dma_addr; - chip->p_dma.preallocate = sbus_dma_preallocate; chip->c_dma.prepare = sbus_dma_prepare; chip->c_dma.enable = sbus_dma_enable; chip->c_dma.request = sbus_dma_request; chip->c_dma.address = sbus_dma_addr; - chip->c_dma.preallocate = sbus_dma_preallocate; - if (request_irq(sdev->irqs[0], snd_cs4231_sbus_interrupt, + if (request_irq(op->irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n", - dev, sdev->irqs[0]); + dev, op->irqs[0]); snd_cs4231_sbus_free(chip); return -EBUSY; } - chip->irq[0] = sdev->irqs[0]; + chip->irq[0] = op->irqs[0]; if (snd_cs4231_probe(chip) < 0) { snd_cs4231_sbus_free(chip); @@ -1881,9 +1856,9 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, return 0; } -static int __init cs4231_sbus_attach(struct sbus_dev *sdev) +static int __devinit cs4231_sbus_probe(struct of_device *op, const struct of_device_id *match) { - struct resource *rp = &sdev->resource[0]; + struct resource *rp = &op->resource[0]; struct snd_card *card; int err; @@ -1895,9 +1870,9 @@ static int __init cs4231_sbus_attach(struct sbus_dev *sdev) card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, - sdev->irqs[0]); + op->irqs[0]); - err = snd_cs4231_sbus_create(card, sdev, dev); + err = snd_cs4231_sbus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -1950,30 +1925,25 @@ static unsigned int _ebus_dma_addr(struct cs4231_dma_control *dma_cont) return ebus_dma_addr(&dma_cont->ebus_info); } -static void _ebus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) -{ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->dev_u.pdev), - 64*1024, 128*1024); -} - /* * Init and exit routines */ static int snd_cs4231_ebus_free(struct snd_cs4231 *chip) { + struct of_device *op = chip->op; + if (chip->c_dma.ebus_info.regs) { ebus_dma_unregister(&chip->c_dma.ebus_info); - iounmap(chip->c_dma.ebus_info.regs); + of_iounmap(&op->resource[2], chip->c_dma.ebus_info.regs, 0x10); } if (chip->p_dma.ebus_info.regs) { ebus_dma_unregister(&chip->p_dma.ebus_info); - iounmap(chip->p_dma.ebus_info.regs); + of_iounmap(&op->resource[1], chip->p_dma.ebus_info.regs, 0x10); } if (chip->port) - iounmap(chip->port); + of_iounmap(&op->resource[0], chip->port, 0x10); return 0; } @@ -1990,7 +1960,7 @@ static struct snd_device_ops snd_cs4231_ebus_dev_ops = { }; static int __init snd_cs4231_ebus_create(struct snd_card *card, - struct linux_ebus_device *edev, + struct of_device *op, int dev) { struct snd_cs4231 *chip = card->private_data; @@ -2002,35 +1972,35 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; - chip->dev_u.pdev = edev->bus->self; + chip->op = op; memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); strcpy(chip->c_dma.ebus_info.name, "cs4231(capture)"); chip->c_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->c_dma.ebus_info.callback = snd_cs4231_ebus_capture_callback; chip->c_dma.ebus_info.client_cookie = chip; - chip->c_dma.ebus_info.irq = edev->irqs[0]; + chip->c_dma.ebus_info.irq = op->irqs[0]; strcpy(chip->p_dma.ebus_info.name, "cs4231(play)"); chip->p_dma.ebus_info.flags = EBUS_DMA_FLAG_USE_EBDMA_HANDLER; chip->p_dma.ebus_info.callback = snd_cs4231_ebus_play_callback; chip->p_dma.ebus_info.client_cookie = chip; - chip->p_dma.ebus_info.irq = edev->irqs[1]; + chip->p_dma.ebus_info.irq = op->irqs[1]; chip->p_dma.prepare = _ebus_dma_prepare; chip->p_dma.enable = _ebus_dma_enable; chip->p_dma.request = _ebus_dma_request; chip->p_dma.address = _ebus_dma_addr; - chip->p_dma.preallocate = _ebus_dma_preallocate; chip->c_dma.prepare = _ebus_dma_prepare; chip->c_dma.enable = _ebus_dma_enable; chip->c_dma.request = _ebus_dma_request; chip->c_dma.address = _ebus_dma_addr; - chip->c_dma.preallocate = _ebus_dma_preallocate; - chip->port = ioremap(edev->resource[0].start, 0x10); - chip->p_dma.ebus_info.regs = ioremap(edev->resource[1].start, 0x10); - chip->c_dma.ebus_info.regs = ioremap(edev->resource[2].start, 0x10); + chip->port = of_ioremap(&op->resource[0], 0, 0x10, "cs4231"); + chip->p_dma.ebus_info.regs = + of_ioremap(&op->resource[1], 0, 0x10, "cs4231_pdma"); + chip->c_dma.ebus_info.regs = + of_ioremap(&op->resource[2], 0, 0x10, "cs4231_cdma"); if (!chip->port || !chip->p_dma.ebus_info.regs || !chip->c_dma.ebus_info.regs) { snd_cs4231_ebus_free(chip); @@ -2078,7 +2048,7 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, return 0; } -static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) +static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_device_id *match) { struct snd_card *card; int err; @@ -2089,10 +2059,10 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, - edev->resource[0].start, - edev->irqs[0]); + op->resource[0].start, + op->irqs[0]); - err = snd_cs4231_ebus_create(card, edev, dev); + err = snd_cs4231_ebus_create(card, op, dev); if (err < 0) { snd_card_free(card); return err; @@ -2102,68 +2072,57 @@ static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) } #endif -static int __init cs4231_init(void) +static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match) { -#ifdef SBUS_SUPPORT - struct sbus_bus *sbus; - struct sbus_dev *sdev; -#endif #ifdef EBUS_SUPPORT - struct linux_ebus *ebus; - struct linux_ebus_device *edev; + if (!strcmp(op->node->parent->name, "ebus")) + return cs4231_ebus_probe(op, match); #endif - int found; - - found = 0; - #ifdef SBUS_SUPPORT - for_all_sbusdev(sdev, sbus) { - if (!strcmp(sdev->prom_name, "SUNW,CS4231")) { - if (cs4231_sbus_attach(sdev) == 0) - found++; - } - } + if (!strcmp(op->node->parent->name, "sbus") || + !strcmp(op->node->parent->name, "sbi")) + return cs4231_sbus_probe(op, match); #endif -#ifdef EBUS_SUPPORT - for_each_ebus(ebus) { - for_each_ebusdev(edev, ebus) { - int match = 0; - - if (!strcmp(edev->prom_node->name, "SUNW,CS4231")) { - match = 1; - } else if (!strcmp(edev->prom_node->name, "audio")) { - const char *compat; - - compat = of_get_property(edev->prom_node, - "compatible", NULL); - if (compat && !strcmp(compat, "SUNW,CS4231")) - match = 1; - } + return -ENODEV; +} - if (match && - cs4231_ebus_attach(edev) == 0) - found++; - } - } -#endif +static int __devexit cs4231_remove(struct of_device *op) +{ + struct snd_cs4231 *chip = dev_get_drvdata(&op->dev); + snd_card_free(chip->card); - return (found > 0) ? 0 : -EIO; + return 0; } -static void __exit cs4231_exit(void) -{ - struct snd_cs4231 *p = cs4231_list; +static const struct of_device_id cs4231_match[] = { + { + .name = "SUNW,CS4231", + }, + { + .name = "audio", + .compatible = "SUNW,CS4231", + }, + {}, +}; - while (p != NULL) { - struct snd_cs4231 *next = p->next; +MODULE_DEVICE_TABLE(of, cs4231_match); - snd_card_free(p->card); +static struct of_platform_driver cs4231_driver = { + .name = "audio", + .match_table = cs4231_match, + .probe = cs4231_probe, + .remove = __devexit_p(cs4231_remove), +}; - p = next; - } +static int __init cs4231_init(void) +{ + return of_register_driver(&cs4231_driver, &of_bus_type); +} - cs4231_list = NULL; +static void __exit cs4231_exit(void) +{ + of_unregister_driver(&cs4231_driver); } module_init(cs4231_init); diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index c534a2a849fa..23ed6f04a718 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -57,6 +57,7 @@ #include <linux/delay.h> #include <linux/irq.h> #include <linux/io.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -66,7 +67,7 @@ #include <sound/initval.h> #include <linux/of.h> -#include <asm/sbus.h> +#include <linux/of_device.h> #include <asm/atomic.h> MODULE_AUTHOR("Rudolf Koenig, Brent Baccala and Martin Habets"); @@ -297,7 +298,7 @@ struct dbri_streaminfo { /* This structure holds the information for both chips (DBRI & CS4215) */ struct snd_dbri { int regs_size, irq; /* Needed for unload */ - struct sbus_dev *sdev; /* SBUS device info */ + struct of_device *op; /* OF device info */ spinlock_t lock; struct dbri_dma *dma; /* Pointer to our DMA block */ @@ -2093,14 +2094,15 @@ static int snd_dbri_hw_params(struct snd_pcm_substream *substream, */ if (info->dvma_buffer == 0) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; - info->dvma_buffer = sbus_map_single(dbri->sdev, - runtime->dma_area, - params_buffer_bytes(hw_params), - direction); + info->dvma_buffer = + dma_map_single(&dbri->op->dev, + runtime->dma_area, + params_buffer_bytes(hw_params), + direction); } direction = params_buffer_bytes(hw_params); @@ -2121,12 +2123,12 @@ static int snd_dbri_hw_free(struct snd_pcm_substream *substream) */ if (info->dvma_buffer) { if (DBRI_STREAMNO(substream) == DBRI_PLAY) - direction = SBUS_DMA_TODEVICE; + direction = DMA_TO_DEVICE; else - direction = SBUS_DMA_FROMDEVICE; + direction = DMA_FROM_DEVICE; - sbus_unmap_single(dbri->sdev, info->dvma_buffer, - substream->runtime->buffer_size, direction); + dma_unmap_single(&dbri->op->dev, info->dvma_buffer, + substream->runtime->buffer_size, direction); info->dvma_buffer = 0; } if (info->pipe != -1) { @@ -2519,31 +2521,34 @@ static void __devinit snd_dbri_proc(struct snd_card *card) static void snd_dbri_free(struct snd_dbri *dbri); static int __devinit snd_dbri_create(struct snd_card *card, - struct sbus_dev *sdev, - int irq, int dev) + struct of_device *op, + int irq, int dev) { struct snd_dbri *dbri = card->private_data; int err; spin_lock_init(&dbri->lock); - dbri->sdev = sdev; + dbri->op = op; dbri->irq = irq; - dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma), - &dbri->dma_dvma); + dbri->dma = dma_alloc_coherent(&op->dev, + sizeof(struct dbri_dma), + &dbri->dma_dvma, GFP_ATOMIC); + if (!dbri->dma) + return -ENOMEM; memset((void *)dbri->dma, 0, sizeof(struct dbri_dma)); dprintk(D_GEN, "DMA Cmd Block 0x%p (0x%08x)\n", dbri->dma, dbri->dma_dvma); /* Map the registers into memory. */ - dbri->regs_size = sdev->reg_addrs[0].reg_size; - dbri->regs = sbus_ioremap(&sdev->resource[0], 0, - dbri->regs_size, "DBRI Registers"); + dbri->regs_size = resource_size(&op->resource[0]); + dbri->regs = of_ioremap(&op->resource[0], 0, + dbri->regs_size, "DBRI Registers"); if (!dbri->regs) { printk(KERN_ERR "DBRI: could not allocate registers\n"); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return -EIO; } @@ -2551,9 +2556,9 @@ static int __devinit snd_dbri_create(struct snd_card *card, "DBRI audio", dbri); if (err) { printk(KERN_ERR "DBRI: Can't get irq %d\n", dbri->irq); - sbus_iounmap(dbri->regs, dbri->regs_size); - sbus_free_consistent(sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + of_iounmap(&op->resource[0], dbri->regs, dbri->regs_size); + dma_free_coherent(&op->dev, sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); return err; } @@ -2577,27 +2582,23 @@ static void snd_dbri_free(struct snd_dbri *dbri) free_irq(dbri->irq, dbri); if (dbri->regs) - sbus_iounmap(dbri->regs, dbri->regs_size); + of_iounmap(&dbri->op->resource[0], dbri->regs, dbri->regs_size); if (dbri->dma) - sbus_free_consistent(dbri->sdev, sizeof(struct dbri_dma), - (void *)dbri->dma, dbri->dma_dvma); + dma_free_coherent(&dbri->op->dev, + sizeof(struct dbri_dma), + (void *)dbri->dma, dbri->dma_dvma); } -static int __devinit dbri_probe(struct of_device *of_dev, - const struct of_device_id *match) +static int __devinit dbri_probe(struct of_device *op, const struct of_device_id *match) { - struct sbus_dev *sdev = to_sbus_device(&of_dev->dev); struct snd_dbri *dbri; - int irq; struct resource *rp; struct snd_card *card; static int dev = 0; + int irq; int err; - dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", - sdev->prom_name, sdev->slot); - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2605,7 +2606,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, return -ENOENT; } - irq = sdev->irqs[0]; + irq = op->irqs[0]; if (irq <= 0) { printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev); return -ENODEV; @@ -2618,12 +2619,12 @@ static int __devinit dbri_probe(struct of_device *of_dev, strcpy(card->driver, "DBRI"); strcpy(card->shortname, "Sun DBRI"); - rp = &sdev->resource[0]; + rp = &op->resource[0]; sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d", card->shortname, rp->flags & 0xffL, (unsigned long long)rp->start, irq); - err = snd_dbri_create(card, sdev, irq, dev); + err = snd_dbri_create(card, op, irq, dev); if (err < 0) { snd_card_free(card); return err; @@ -2640,7 +2641,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, /* /proc file handling */ snd_dbri_proc(card); - dev_set_drvdata(&of_dev->dev, card); + dev_set_drvdata(&op->dev, card); err = snd_card_register(card); if (err < 0) @@ -2648,7 +2649,7 @@ static int __devinit dbri_probe(struct of_device *of_dev, printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n", dev, dbri->regs, - dbri->irq, sdev->prom_name[9], dbri->mm.version); + dbri->irq, op->node->name[9], dbri->mm.version); dev++; return 0; @@ -2659,19 +2660,19 @@ _err: return err; } -static int __devexit dbri_remove(struct of_device *dev) +static int __devexit dbri_remove(struct of_device *op) { - struct snd_card *card = dev_get_drvdata(&dev->dev); + struct snd_card *card = dev_get_drvdata(&op->dev); snd_dbri_free(card->private_data); snd_card_free(card); - dev_set_drvdata(&dev->dev, NULL); + dev_set_drvdata(&op->dev, NULL); return 0; } -static struct of_device_id dbri_match[] = { +static const struct of_device_id dbri_match[] = { { .name = "SUNW,DBRIe", }, @@ -2693,7 +2694,7 @@ static struct of_platform_driver dbri_sbus_driver = { /* Probe for the dbri chip and then attach the driver. */ static int __init dbri_init(void) { - return of_register_driver(&dbri_sbus_driver, &sbus_bus_type); + return of_register_driver(&dbri_sbus_driver, &of_bus_type); } static void __exit dbri_exit(void) diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index 798ca124da58..ccd763dd7167 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -247,69 +247,56 @@ static struct caiaq_controller a8dj_controller[] = { { "Software lock", 40 } }; -int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +static int __devinit add_controls(struct caiaq_controller *c, int num, + struct snd_usb_caiaqdev *dev) { - int i; + int i, ret; struct snd_kcontrol *kc; + for (i = 0; i < num; i++, c++) { + kcontrol_template.name = c->name; + kcontrol_template.private_value = c->index; + kc = snd_ctl_new1(&kcontrol_template, dev); + ret = snd_ctl_add(dev->chip.card, kc); + if (ret < 0) + return ret; + } + + return 0; +} + +int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) +{ + int ret = 0; + switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): - for (i = 0; i < ARRAY_SIZE(ak1_controller); i++) { - struct caiaq_controller *c = ak1_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(ak1_controller, + ARRAY_SIZE(ak1_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): - for (i = 0; i < ARRAY_SIZE(rk2_controller); i++) { - struct caiaq_controller *c = rk2_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(rk2_controller, + ARRAY_SIZE(rk2_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): - for (i = 0; i < ARRAY_SIZE(rk3_controller); i++) { - struct caiaq_controller *c = rk3_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(rk3_controller, + ARRAY_SIZE(rk3_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): - for (i = 0; i < ARRAY_SIZE(kore_controller); i++) { - struct caiaq_controller *c = kore_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(kore_controller, + ARRAY_SIZE(kore_controller), dev); break; case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): - for (i = 0; i < ARRAY_SIZE(a8dj_controller); i++) { - struct caiaq_controller *c = a8dj_controller + i; - kcontrol_template.name = c->name; - kcontrol_template.private_value = c->index; - kc = snd_ctl_new1(&kcontrol_template, dev); - snd_ctl_add(dev->chip.card, kc); - } - + ret = add_controls(a8dj_controller, + ARRAY_SIZE(a8dj_controller), dev); break; } - return 0; + return ret; } diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 83175083e50f..b143ef7152f7 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.8"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.9"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 69689e79bf79..92115755d98e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1480,6 +1480,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* Advanced modes of the Edirol UA-25EX. + * For the standard mode, UA-25EX has ID 0582:00e7, which + * offers only 16-bit PCM at 44.1 kHz and no MIDI. + */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UA-25EX", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_EDIROL_UAXX + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index b441fe2cd190..c2515b680f9f 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -118,12 +118,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, void *vaddr; struct us122l *us122l = area->vm_private_data; struct usb_stream *s; - int vm_f = VM_FAULT_SIGBUS; mutex_lock(&us122l->mutex); s = us122l->sk.s; if (!s) - goto out; + goto unlock; offset = vmf->pgoff << PAGE_SHIFT; if (offset < PAGE_ALIGN(s->read_size)) @@ -131,7 +130,7 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, else { offset -= PAGE_ALIGN(s->read_size); if (offset >= PAGE_ALIGN(s->write_size)) - goto out; + goto unlock; vaddr = us122l->sk.write_page + offset; } @@ -141,9 +140,11 @@ static int usb_stream_hwdep_vm_fault(struct vm_area_struct *area, mutex_unlock(&us122l->mutex); vmf->page = page; - vm_f = 0; -out: - return vm_f; + + return 0; +unlock: + mutex_unlock(&us122l->mutex); + return VM_FAULT_SIGBUS; } static void usb_stream_hwdep_vm_close(struct vm_area_struct *area) |