diff options
author | Stephen Rothwell <sfr@canb.auug.org.au> | 2010-03-24 12:14:32 +1100 |
---|---|---|
committer | Stephen Rothwell <sfr@canb.auug.org.au> | 2010-03-24 12:14:32 +1100 |
commit | b60da9856d20d95d1407441ee0b1cdd8d62812e3 (patch) | |
tree | 1cc66df8268d9554ac74b05f78cd0c1dce2308ba | |
parent | 2c8c046ba34bd34c39e593230e47605e8bd6a3d8 (diff) | |
parent | a48d926d0e1c0ca480171ef73061c6dc0a6116b4 (diff) |
Merge remote branch 'sound/for-next'
138 files changed, 12063 insertions, 6675 deletions
diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index d15beceb632e..512174a874a1 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -604,7 +604,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#ifdef CONFIG_SND_DM365_AIC3X_CODEC dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + dm365_init_vc(&dm365_evm_snd_data); +#endif dm365_init_rtc(); dm365_init_ks(&dm365evm_ks_data); diff --git a/arch/arm/plat-omap/include/plat/mcbsp.h b/arch/arm/plat-omap/include/plat/mcbsp.h index 39748354ce45..1bd7021336c2 100644 --- a/arch/arm/plat-omap/include/plat/mcbsp.h +++ b/arch/arm/plat-omap/include/plat/mcbsp.h @@ -149,6 +149,8 @@ #define OMAP_MCBSP_REG_WAKEUPEN 0xA8 #define OMAP_MCBSP_REG_XCCR 0xAC #define OMAP_MCBSP_REG_RCCR 0xB0 +#define OMAP_MCBSP_REG_XBUFFSTAT 0xB4 +#define OMAP_MCBSP_REG_RBUFFSTAT 0xB8 #define OMAP_MCBSP_REG_SSELCR 0xBC #define OMAP_ST_REG_REV 0x00 @@ -471,6 +473,8 @@ void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold); void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold); u16 omap_mcbsp_get_max_tx_threshold(unsigned int id); u16 omap_mcbsp_get_max_rx_threshold(unsigned int id); +u16 omap_mcbsp_get_tx_delay(unsigned int id); +u16 omap_mcbsp_get_rx_delay(unsigned int id); int omap_mcbsp_get_dma_op_mode(unsigned int id); #else static inline void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) @@ -479,6 +483,8 @@ static inline void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) { } static inline u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) { return 0; } static inline u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) { return 0; } +static inline u16 omap_mcbsp_get_tx_delay(unsigned int id) { return 0; } +static inline u16 omap_mcbsp_get_rx_delay(unsigned int id) { return 0; } static inline int omap_mcbsp_get_dma_op_mode(unsigned int id) { return 0; } #endif int omap_mcbsp_request(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index 52dfcc81511e..4ff7a11b1838 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -559,6 +559,61 @@ u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) } EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold); +#define MCBSP2_FIFO_SIZE 0x500 /* 1024 + 256 locations */ +#define MCBSP1345_FIFO_SIZE 0x80 /* 128 locations */ +/* + * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO + */ +u16 omap_mcbsp_get_tx_delay(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + u16 buffstat; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + /* Returns the number of free locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); + + /* Number of slots are different in McBSP ports */ + if (mcbsp->id == 2) + return MCBSP2_FIFO_SIZE - buffstat; + else + return MCBSP1345_FIFO_SIZE - buffstat; +} +EXPORT_SYMBOL(omap_mcbsp_get_tx_delay); + +/* + * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO + * to reach the threshold value (when the DMA will be triggered to read it) + */ +u16 omap_mcbsp_get_rx_delay(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + u16 buffstat, threshold; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + /* Returns the number of used locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); + /* RX threshold */ + threshold = MCBSP_READ(mcbsp, THRSH1); + + /* Return the number of location till we reach the threshold limit */ + if (threshold <= buffstat) + return 0; + else + return threshold - buffstat; +} +EXPORT_SYMBOL(omap_mcbsp_get_rx_delay); + /* * omap_mcbsp_get_dma_op_mode just return the current configured * operating mode for the mcbsp channel diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 2a5a0b78f84e..b2073e0266f5 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -53,6 +53,10 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + tristate + select MFD_CORE + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index 22715add99a7..4fbf8f89a49b 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o obj-$(CONFIG_HTC_I2CPLD) += htc-i2cpld.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o tmio_core.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 000000000000..9886aa8de250 --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,189 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + struct mfd_cell *cell = NULL; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + davinci_vc->pdev = pdev; + + /* Voice codec interface client */ + cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; + cell->name = "davinci_vcif"; + cell->driver_data = davinci_vc; + + /* Voice codec CQ93VC client */ + cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; + cell->name = "cq93vc"; + cell->driver_data = davinci_vc; + + ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, + DAVINCI_VC_CELLS, NULL, 0); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client devices\n"); + goto fail4; + } + + return 0; + +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + mfd_remove_devices(&pdev->dev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 562cd4935e17..720e099e506d 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -109,7 +109,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) + defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -708,7 +708,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* Phoenix*/ if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; - child = add_child(sub_chip_id, "twl6030_codec", + child = add_child(sub_chip_id, "twl6040_codec", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index fb6784e86d5f..ebd90ce58ca2 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -569,9 +569,9 @@ struct twl4030_codec_data { struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; - /* twl6030 */ - int audpwron_gpio; /* audio power-on gpio */ - int naudint_irq; /* audio interrupt */ + /* twl6040 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 000000000000..0ab61320ffa8 --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,126 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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 __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/mfd/core.h> + +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +#define DAVINCI_VC_CELLS 2 + +enum davinci_vc_cells { + DAVINCI_VC_VCIF_CELL, + DAVINCI_VC_CQ93VC_CELL, +}; + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct platform_device *pdev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* MFD cells */ + struct mfd_cell cells[DAVINCI_VC_CELLS]; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index d899dc0223ba..a95141eafce3 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -492,6 +492,8 @@ */ #define WM8350_JACK_L_LVL 0x0800 #define WM8350_JACK_R_LVL 0x0400 +#define WM8350_JACK_MICSCD_LVL 0x0200 +#define WM8350_JACK_MICSD_LVL 0x0100 /* * WM8350 Platform setup diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h new file mode 100644 index 000000000000..0952231e6c3f --- /dev/null +++ b/include/linux/usb/audio-v2.h @@ -0,0 +1,366 @@ +/* + * Copyright (c) 2010 Daniel Mack <daniel@caiaq.de> + * + * This software is distributed under the terms of the GNU General Public + * License ("GPL") version 2, as published by the Free Software Foundation. + * + * This file holds USB constants and structures defined + * by the USB Device Class Definition for Audio Devices in version 2.0. + * Comments below reference relevant sections of the documents contained + * in http://www.usb.org/developers/devclass_docs/Audio2.0_final.zip + */ + +#ifndef __LINUX_USB_AUDIO_V2_H +#define __LINUX_USB_AUDIO_V2_H + +#include <linux/types.h> + +/* v1.0 and v2.0 of this standard have many things in common. For the rest + * of the definitions, please refer to audio.h */ + +/* 4.7.2.1 Clock Source Descriptor */ + +struct uac_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __u8 bmControls; + __u8 bAssocTerminal; + __u8 iClockSource; +} __attribute__((packed)); + +/* 4.7.2.2 Clock Source Descriptor */ + +struct uac_clock_selector_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bNrInPins; + __u8 bmControls; + __u8 baCSourceID[]; +} __attribute__((packed)); + +/* 4.7.2.4 Input terminal descriptor */ + +struct uac2_input_terminal_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalID; + __u16 wTerminalType; + __u8 bAssocTerminal; + __u8 bCSourceID; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; + __u16 bmControls; + __u8 iTerminal; +} __attribute__((packed)); + +/* 4.7.2.5 Output terminal descriptor */ + +struct uac2_output_terminal_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalID; + __u16 wTerminalType; + __u8 bAssocTerminal; + __u8 bSourceID; + __u8 bCSourceID; + __u16 bmControls; + __u8 iTerminal; +} __attribute__((packed)); + + + +/* 4.7.2.8 Feature Unit Descriptor */ + +struct uac2_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + /* bmaControls is actually u32, + * but u8 is needed for the hybrid parser */ + __u8 bmaControls[0]; /* variable length */ +} __attribute__((packed)); + +/* 4.9.2 Class-Specific AS Interface Descriptor */ + +struct uac_as_header_descriptor_v2 { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __u8 bmControls; + __u8 bFormatType; + __u32 bmFormats; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; +} __attribute__((packed)); + + +/* A.7 Audio Function Category Codes */ +#define UAC2_FUNCTION_SUBCLASS_UNDEFINED 0x00 +#define UAC2_FUNCTION_DESKTOP_SPEAKER 0x01 +#define UAC2_FUNCTION_HOME_THEATER 0x02 +#define UAC2_FUNCTION_MICROPHONE 0x03 +#define UAC2_FUNCTION_HEADSET 0x04 +#define UAC2_FUNCTION_TELEPHONE 0x05 +#define UAC2_FUNCTION_CONVERTER 0x06 +#define UAC2_FUNCTION_SOUND_RECORDER 0x07 +#define UAC2_FUNCTION_IO_BOX 0x08 +#define UAC2_FUNCTION_MUSICAL_INSTRUMENT 0x09 +#define UAC2_FUNCTION_PRO_AUDIO 0x0a +#define UAC2_FUNCTION_AUDIO_VIDEO 0x0b +#define UAC2_FUNCTION_CONTROL_PANEL 0x0c +#define UAC2_FUNCTION_OTHER 0xff + +/* A.9 Audio Class-Specific AC Interface Descriptor Subtypes */ +/* see audio.h for the rest, which is identical to v1 */ +#define UAC2_EFFECT_UNIT 0x07 +#define UAC2_PROCESSING_UNIT_V2 0x08 +#define UAC2_EXTENSION_UNIT_V2 0x09 +#define UAC2_CLOCK_SOURCE 0x0a +#define UAC2_CLOCK_SELECTOR 0x0b +#define UAC2_CLOCK_MULTIPLIER 0x0c +#define UAC2_SAMPLE_RATE_CONVERTER 0x0d + +/* A.10 Audio Class-Specific AS Interface Descriptor Subtypes */ +/* see audio.h for the rest, which is identical to v1 */ +#define UAC2_ENCODER 0x03 +#define UAC2_DECODER 0x04 + +/* A.11 Effect Unit Effect Types */ +#define UAC2_EFFECT_UNDEFINED 0x00 +#define UAC2_EFFECT_PARAM_EQ 0x01 +#define UAC2_EFFECT_REVERB 0x02 +#define UAC2_EFFECT_MOD_DELAY 0x03 +#define UAC2_EFFECT_DYN_RANGE_COMP 0x04 + +/* A.12 Processing Unit Process Types */ +#define UAC2_PROCESS_UNDEFINED 0x00 +#define UAC2_PROCESS_UP_DOWNMIX 0x01 +#define UAC2_PROCESS_DOLBY_PROLOCIC 0x02 +#define UAC2_PROCESS_STEREO_EXTENDER 0x03 + +/* A.14 Audio Class-Specific Request Codes */ +#define UAC2_CS_CUR 0x01 +#define UAC2_CS_RANGE 0x02 + +/* A.15 Encoder Type Codes */ +#define UAC2_ENCODER_UNDEFINED 0x00 +#define UAC2_ENCODER_OTHER 0x01 +#define UAC2_ENCODER_MPEG 0x02 +#define UAC2_ENCODER_AC3 0x03 +#define UAC2_ENCODER_WMA 0x04 +#define UAC2_ENCODER_DTS 0x05 + +/* A.16 Decoder Type Codes */ +#define UAC2_DECODER_UNDEFINED 0x00 +#define UAC2_DECODER_OTHER 0x01 +#define UAC2_DECODER_MPEG 0x02 +#define UAC2_DECODER_AC3 0x03 +#define UAC2_DECODER_WMA 0x04 +#define UAC2_DECODER_DTS 0x05 + +/* A.17.1 Clock Source Control Selectors */ +#define UAC2_CS_UNDEFINED 0x00 +#define UAC2_CS_CONTROL_SAM_FREQ 0x01 +#define UAC2_CS_CONTROL_CLOCK_VALID 0x02 + +/* A.17.2 Clock Selector Control Selectors */ +#define UAC2_CX_UNDEFINED 0x00 +#define UAC2_CX_CLOCK_SELECTOR 0x01 + +/* A.17.3 Clock Multiplier Control Selectors */ +#define UAC2_CM_UNDEFINED 0x00 +#define UAC2_CM_NUMERATOR 0x01 +#define UAC2_CM_DENOMINTATOR 0x02 + +/* A.17.4 Terminal Control Selectors */ +#define UAC2_TE_UNDEFINED 0x00 +#define UAC2_TE_COPY_PROTECT 0x01 +#define UAC2_TE_CONNECTOR 0x02 +#define UAC2_TE_OVERLOAD 0x03 +#define UAC2_TE_CLUSTER 0x04 +#define UAC2_TE_UNDERFLOW 0x05 +#define UAC2_TE_OVERFLOW 0x06 +#define UAC2_TE_LATENCY 0x07 + +/* A.17.5 Mixer Control Selectors */ +#define UAC2_MU_UNDEFINED 0x00 +#define UAC2_MU_MIXER 0x01 +#define UAC2_MU_CLUSTER 0x02 +#define UAC2_MU_UNDERFLOW 0x03 +#define UAC2_MU_OVERFLOW 0x04 +#define UAC2_MU_LATENCY 0x05 + +/* A.17.6 Selector Control Selectors */ +#define UAC2_SU_UNDEFINED 0x00 +#define UAC2_SU_SELECTOR 0x01 +#define UAC2_SU_LATENCY 0x02 + +/* A.17.7 Feature Unit Control Selectors */ +/* see audio.h for the rest, which is identical to v1 */ +#define UAC2_FU_INPUT_GAIN 0x0b +#define UAC2_FU_INPUT_GAIN_PAD 0x0c +#define UAC2_FU_PHASE_INVERTER 0x0d +#define UAC2_FU_UNDERFLOW 0x0e +#define UAC2_FU_OVERFLOW 0x0f +#define UAC2_FU_LATENCY 0x10 + +/* A.17.8.1 Parametric Equalizer Section Effect Unit Control Selectors */ +#define UAC2_PE_UNDEFINED 0x00 +#define UAC2_PE_ENABLE 0x01 +#define UAC2_PE_CENTERFREQ 0x02 +#define UAC2_PE_QFACTOR 0x03 +#define UAC2_PE_GAIN 0x04 +#define UAC2_PE_UNDERFLOW 0x05 +#define UAC2_PE_OVERFLOW 0x06 +#define UAC2_PE_LATENCY 0x07 + +/* A.17.8.2 Reverberation Effect Unit Control Selectors */ +#define UAC2_RV_UNDEFINED 0x00 +#define UAC2_RV_ENABLE 0x01 +#define UAC2_RV_TYPE 0x02 +#define UAC2_RV_LEVEL 0x03 +#define UAC2_RV_TIME 0x04 +#define UAC2_RV_FEEDBACK 0x05 +#define UAC2_RV_PREDELAY 0x06 +#define UAC2_RV_DENSITY 0x07 +#define UAC2_RV_HIFREQ_ROLLOFF 0x08 +#define UAC2_RV_UNDERFLOW 0x09 +#define UAC2_RV_OVERFLOW 0x0a +#define UAC2_RV_LATENCY 0x0b + +/* A.17.8.3 Modulation Delay Effect Control Selectors */ +#define UAC2_MD_UNDEFINED 0x00 +#define UAC2_MD_ENABLE 0x01 +#define UAC2_MD_BALANCE 0x02 +#define UAC2_MD_RATE 0x03 +#define UAC2_MD_DEPTH 0x04 +#define UAC2_MD_TIME 0x05 +#define UAC2_MD_FEEDBACK 0x06 +#define UAC2_MD_UNDERFLOW 0x07 +#define UAC2_MD_OVERFLOW 0x08 +#define UAC2_MD_LATENCY 0x09 + +/* A.17.8.4 Dynamic Range Compressor Effect Unit Control Selectors */ +#define UAC2_DR_UNDEFINED 0x00 +#define UAC2_DR_ENABLE 0x01 +#define UAC2_DR_COMPRESSION_RATE 0x02 +#define UAC2_DR_MAXAMPL 0x03 +#define UAC2_DR_THRESHOLD 0x04 +#define UAC2_DR_ATTACK_TIME 0x05 +#define UAC2_DR_RELEASE_TIME 0x06 +#define UAC2_DR_UNDEFLOW 0x07 +#define UAC2_DR_OVERFLOW 0x08 +#define UAC2_DR_LATENCY 0x09 + +/* A.17.9.1 Up/Down-mix Processing Unit Control Selectors */ +#define UAC2_UD_UNDEFINED 0x00 +#define UAC2_UD_ENABLE 0x01 +#define UAC2_UD_MODE_SELECT 0x02 +#define UAC2_UD_CLUSTER 0x03 +#define UAC2_UD_UNDERFLOW 0x04 +#define UAC2_UD_OVERFLOW 0x05 +#define UAC2_UD_LATENCY 0x06 + +/* A.17.9.2 Dolby Prologic[tm] Processing Unit Control Selectors */ +#define UAC2_DP_UNDEFINED 0x00 +#define UAC2_DP_ENABLE 0x01 +#define UAC2_DP_MODE_SELECT 0x02 +#define UAC2_DP_CLUSTER 0x03 +#define UAC2_DP_UNDERFFLOW 0x04 +#define UAC2_DP_OVERFLOW 0x05 +#define UAC2_DP_LATENCY 0x06 + +/* A.17.9.3 Stereo Expander Processing Unit Control Selectors */ +#define UAC2_ST_EXT_UNDEFINED 0x00 +#define UAC2_ST_EXT_ENABLE 0x01 +#define UAC2_ST_EXT_WIDTH 0x02 +#define UAC2_ST_EXT_UNDEFLOW 0x03 +#define UAC2_ST_EXT_OVERFLOW 0x04 +#define UAC2_ST_EXT_LATENCY 0x05 + +/* A.17.10 Extension Unit Control Selectors */ +#define UAC2_XU_UNDEFINED 0x00 +#define UAC2_XU_ENABLE 0x01 +#define UAC2_XU_CLUSTER 0x02 +#define UAC2_XU_UNDERFLOW 0x03 +#define UAC2_XU_OVERFLOW 0x04 +#define UAC2_XU_LATENCY 0x05 + +/* A.17.11 AudioStreaming Interface Control Selectors */ +#define UAC2_AS_UNDEFINED 0x00 +#define UAC2_AS_ACT_ALT_SETTING 0x01 +#define UAC2_AS_VAL_ALT_SETTINGS 0x02 +#define UAC2_AS_AUDIO_DATA_FORMAT 0x03 + +/* A.17.12 Encoder Control Selectors */ +#define UAC2_EN_UNDEFINED 0x00 +#define UAC2_EN_BIT_RATE 0x01 +#define UAC2_EN_QUALITY 0x02 +#define UAC2_EN_VBR 0x03 +#define UAC2_EN_TYPE 0x04 +#define UAC2_EN_UNDERFLOW 0x05 +#define UAC2_EN_OVERFLOW 0x06 +#define UAC2_EN_ENCODER_ERROR 0x07 +#define UAC2_EN_PARAM1 0x08 +#define UAC2_EN_PARAM2 0x09 +#define UAC2_EN_PARAM3 0x0a +#define UAC2_EN_PARAM4 0x0b +#define UAC2_EN_PARAM5 0x0c +#define UAC2_EN_PARAM6 0x0d +#define UAC2_EN_PARAM7 0x0e +#define UAC2_EN_PARAM8 0x0f + +/* A.17.13.1 MPEG Decoder Control Selectors */ +#define UAC2_MPEG_UNDEFINED 0x00 +#define UAC2_MPEG_DUAL_CHANNEL 0x01 +#define UAC2_MPEG_SECOND_STEREO 0x02 +#define UAC2_MPEG_MULTILINGUAL 0x03 +#define UAC2_MPEG_DYN_RANGE 0x04 +#define UAC2_MPEG_SCALING 0x05 +#define UAC2_MPEG_HILO_SCALING 0x06 +#define UAC2_MPEG_UNDERFLOW 0x07 +#define UAC2_MPEG_OVERFLOW 0x08 +#define UAC2_MPEG_DECODER_ERROR 0x09 + +/* A17.13.2 AC3 Decoder Control Selectors */ +#define UAC2_AC3_UNDEFINED 0x00 +#define UAC2_AC3_MODE 0x01 +#define UAC2_AC3_DYN_RANGE 0x02 +#define UAC2_AC3_SCALING 0x03 +#define UAC2_AC3_HILO_SCALING 0x04 +#define UAC2_AC3_UNDERFLOW 0x05 +#define UAC2_AC3_OVERFLOW 0x06 +#define UAC2_AC3_DECODER_ERROR 0x07 + +/* A17.13.3 WMA Decoder Control Selectors */ +#define UAC2_WMA_UNDEFINED 0x00 +#define UAC2_WMA_UNDERFLOW 0x01 +#define UAC2_WMA_OVERFLOW 0x02 +#define UAC2_WMA_DECODER_ERROR 0x03 + +/* A17.13.4 DTS Decoder Control Selectors */ +#define UAC2_DTS_UNDEFINED 0x00 +#define UAC2_DTS_UNDERFLOW 0x01 +#define UAC2_DTS_OVERFLOW 0x02 +#define UAC2_DTS_DECODER_ERROR 0x03 + +/* A17.14 Endpoint Control Selectors */ +#define UAC2_EP_CS_UNDEFINED 0x00 +#define UAC2_EP_CS_PITCH 0x01 +#define UAC2_EP_CS_DATA_OVERRUN 0x02 +#define UAC2_EP_CS_DATA_UNDERRUN 0x03 + +#endif /* __LINUX_USB_AUDIO_V2_H */ + diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 4d3e450e2b03..905a87caf3fb 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -13,6 +13,9 @@ * Comments below reference relevant sections of that document: * * http://www.usb.org/developers/devclass_docs/audio10.pdf + * + * Types and defines in this file are either specific to version 1.0 of + * this standard or common for newer versions. */ #ifndef __LINUX_USB_AUDIO_H @@ -20,14 +23,15 @@ #include <linux/types.h> +/* bInterfaceProtocol values to denote the version of the standard used */ +#define UAC_VERSION_1 0x00 +#define UAC_VERSION_2 0x20 + /* A.2 Audio Interface Subclass Codes */ #define USB_SUBCLASS_AUDIOCONTROL 0x01 #define USB_SUBCLASS_AUDIOSTREAMING 0x02 #define USB_SUBCLASS_MIDISTREAMING 0x03 -#define UAC_VERSION_1 0x00 -#define UAC_VERSION_2 0x20 - /* A.5 Audio Class-Specific AC Interface Descriptor Subtypes */ #define UAC_HEADER 0x01 #define UAC_INPUT_TERMINAL 0x02 @@ -38,15 +42,6 @@ #define UAC_PROCESSING_UNIT_V1 0x07 #define UAC_EXTENSION_UNIT_V1 0x08 -/* UAC v2.0 types */ -#define UAC_EFFECT_UNIT 0x07 -#define UAC_PROCESSING_UNIT_V2 0x08 -#define UAC_EXTENSION_UNIT_V2 0x09 -#define UAC_CLOCK_SOURCE 0x0a -#define UAC_CLOCK_SELECTOR 0x0b -#define UAC_CLOCK_MULTIPLIER 0x0c -#define UAC_SAMPLE_RATE_CONVERTER 0x0d - /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 #define UAC_FORMAT_TYPE 0x02 @@ -78,10 +73,6 @@ #define UAC_GET_STAT 0xff -/* Audio class v2.0 handles all the parameter calls differently */ -#define UAC2_CS_CUR 0x01 -#define UAC2_CS_RANGE 0x02 - /* MIDI - A.1 MS Class-Specific Interface Descriptor Subtypes */ #define UAC_MS_HEADER 0x01 #define UAC_MIDI_IN_JACK 0x02 @@ -190,6 +181,156 @@ struct uac_feature_unit_descriptor_##ch { \ __u8 iFeature; \ } __attribute__ ((packed)) +/* 4.3.2.3 Mixer Unit Descriptor */ +struct uac_mixer_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bNrInPins; + __u8 baSourceID[]; +} __attribute__ ((packed)); + +static inline __u8 uac_mixer_unit_bNrChannels(struct uac_mixer_unit_descriptor *desc) +{ + return desc->baSourceID[desc->bNrInPins]; +} + +static inline __u32 uac_mixer_unit_wChannelConfig(struct uac_mixer_unit_descriptor *desc, + int protocol) +{ + if (protocol == UAC_VERSION_1) + return (desc->baSourceID[desc->bNrInPins + 2] << 8) | + desc->baSourceID[desc->bNrInPins + 1]; + else + return (desc->baSourceID[desc->bNrInPins + 4] << 24) | + (desc->baSourceID[desc->bNrInPins + 3] << 16) | + (desc->baSourceID[desc->bNrInPins + 2] << 8) | + (desc->baSourceID[desc->bNrInPins + 1]); +} + +static inline __u8 uac_mixer_unit_iChannelNames(struct uac_mixer_unit_descriptor *desc, + int protocol) +{ + return (protocol == UAC_VERSION_1) ? + desc->baSourceID[desc->bNrInPins + 3] : + desc->baSourceID[desc->bNrInPins + 5]; +} + +static inline __u8 *uac_mixer_unit_bmControls(struct uac_mixer_unit_descriptor *desc, + int protocol) +{ + return (protocol == UAC_VERSION_1) ? + &desc->baSourceID[desc->bNrInPins + 4] : + &desc->baSourceID[desc->bNrInPins + 6]; +} + +static inline __u8 uac_mixer_unit_iMixer(struct uac_mixer_unit_descriptor *desc) +{ + __u8 *raw = (__u8 *) desc; + return raw[desc->bLength - 1]; +} + +/* 4.3.2.4 Selector Unit Descriptor */ +struct uac_selector_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUintID; + __u8 bNrInPins; + __u8 baSourceID[]; +} __attribute__ ((packed)); + +static inline __u8 uac_selector_unit_iSelector(struct uac_selector_unit_descriptor *desc) +{ + __u8 *raw = (__u8 *) desc; + return raw[desc->bLength - 1]; +} + +/* 4.3.2.5 Feature Unit Descriptor */ +struct uac_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + __u8 bControlSize; + __u8 bmaControls[0]; /* variable length */ +} __attribute__((packed)); + +static inline __u8 uac_feature_unit_iFeature(struct uac_feature_unit_descriptor *desc) +{ + __u8 *raw = (__u8 *) desc; + return raw[desc->bLength - 1]; +} + +/* 4.3.2.6 Processing Unit Descriptors */ +struct uac_processing_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u16 wProcessType; + __u8 bNrInPins; + __u8 baSourceID[]; +} __attribute__ ((packed)); + +static inline __u8 uac_processing_unit_bNrChannels(struct uac_processing_unit_descriptor *desc) +{ + return desc->baSourceID[desc->bNrInPins]; +} + +static inline __u32 uac_processing_unit_wChannelConfig(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + if (protocol == UAC_VERSION_1) + return (desc->baSourceID[desc->bNrInPins + 2] << 8) | + desc->baSourceID[desc->bNrInPins + 1]; + else + return (desc->baSourceID[desc->bNrInPins + 4] << 24) | + (desc->baSourceID[desc->bNrInPins + 3] << 16) | + (desc->baSourceID[desc->bNrInPins + 2] << 8) | + (desc->baSourceID[desc->bNrInPins + 1]); +} + +static inline __u8 uac_processing_unit_iChannelNames(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + return (protocol == UAC_VERSION_1) ? + desc->baSourceID[desc->bNrInPins + 3] : + desc->baSourceID[desc->bNrInPins + 5]; +} + +static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + return (protocol == UAC_VERSION_1) ? + desc->baSourceID[desc->bNrInPins + 4] : + desc->baSourceID[desc->bNrInPins + 6]; +} + +static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + return (protocol == UAC_VERSION_1) ? + &desc->baSourceID[desc->bNrInPins + 5] : + &desc->baSourceID[desc->bNrInPins + 7]; +} + +static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); + return desc->baSourceID[desc->bNrInPins + control_size]; +} + +static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc, + int protocol) +{ + __u8 control_size = uac_processing_unit_bControlSize(desc, protocol); + return &desc->baSourceID[desc->bNrInPins + control_size + 1]; +} + /* 4.5.2 Class-Specific AS Interface Descriptor */ struct uac_as_header_descriptor_v1 { __u8 bLength; /* in bytes: 7 */ @@ -200,19 +341,6 @@ struct uac_as_header_descriptor_v1 { __le16 wFormatTag; /* The Audio Data Format */ } __attribute__ ((packed)); -struct uac_as_header_descriptor_v2 { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bTerminalLink; - __u8 bmControls; - __u8 bFormatType; - __u32 bmFormats; - __u8 bNrChannels; - __u32 bmChannelConfig; - __u8 iChannelNames; -} __attribute__((packed)); - #define UAC_DT_AS_HEADER_SIZE 7 /* Formats - A.1.1 Audio Data Format Type I Codes */ @@ -277,7 +405,6 @@ struct uac_format_type_i_ext_descriptor { __u8 bSideBandProtocol; } __attribute__((packed)); - /* Formats - Audio Data Format Type I Codes */ #define UAC_FORMAT_TYPE_II_MPEG 0x1001 @@ -336,31 +463,8 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_PITCH_CONTROL 0x02 #define UAC_EP_CS_ATTR_FILL_MAX 0x80 -/* Audio class v2.0: CLOCK_SOURCE descriptor */ - -struct uac_clock_source_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bClockID; - __u8 bmAttributes; - __u8 bmControls; - __u8 bAssocTerminal; - __u8 iClockSource; -} __attribute__((packed)); - /* A.10.2 Feature Unit Control Selectors */ -struct uac_feature_unit_descriptor { - __u8 bLength; - __u8 bDescriptorType; - __u8 bDescriptorSubtype; - __u8 bUnitID; - __u8 bSourceID; - __u8 bControlSize; - __u8 controls[0]; /* variable length */ -} __attribute__((packed)); - #define UAC_FU_CONTROL_UNDEFINED 0x00 #define UAC_MUTE_CONTROL 0x01 #define UAC_VOLUME_CONTROL 0x02 diff --git a/include/sound/jack.h b/include/sound/jack.h index f236e426a706..d90b9fa32707 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -42,6 +42,11 @@ enum snd_jack_types { SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ SND_JACK_VIDEOOUT = 0x0010, SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT, + + /* Kept separate from switches to facilitate implementation */ + SND_JACK_BTN_0 = 0x4000, + SND_JACK_BTN_1 = 0x2000, + SND_JACK_BTN_2 = 0x1000, }; struct snd_jack { @@ -50,6 +55,7 @@ struct snd_jack { int type; const char *id; char name[100]; + unsigned int key[3]; /* Keep in sync with definitions above */ void *private_data; void (*private_free)(struct snd_jack *); }; @@ -59,6 +65,8 @@ struct snd_jack { int snd_jack_new(struct snd_card *card, const char *id, int type, struct snd_jack **jack); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent); +int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type, + int keytype); void snd_jack_report(struct snd_jack *jack, int status); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 061f16d4c878..377693a14385 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -182,6 +182,12 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); + /* + * For hardware based FIFO caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); }; /* @@ -215,11 +221,9 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; /* DAI runtime info */ - struct snd_pcm_runtime *runtime; struct snd_soc_codec *codec; unsigned int active; unsigned char pop_wait:1; - void *dma_data; /* DAI private data */ void *private_data; @@ -230,4 +234,21 @@ struct snd_soc_dai { struct list_head list; }; +static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, + const struct snd_pcm_substream *ss) +{ + return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dai->playback.dma_data : dai->capture.dma_data; +} + +static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, + const struct snd_pcm_substream *ss, + void *data) +{ + if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback.dma_data = data; + else + dai->capture.dma_data = data; +} + #endif diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c0922a034223..d5d6ba862dfe 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -339,6 +339,8 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin); int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin); int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin); int snd_soc_dapm_sync(struct snd_soc_codec *codec); +int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, + const char *pin); /* dapm widget types */ enum snd_soc_dapm_type { @@ -425,9 +427,8 @@ struct snd_soc_dapm_widget { unsigned char connected:1; /* connected codec pin */ unsigned char new:1; /* cnew complete */ unsigned char ext:1; /* has external widgets */ - unsigned char muted:1; /* muted for pop reduction */ unsigned char suspend:1; /* was active before suspend */ - unsigned char pmdown:1; /* waiting for timeout */ + unsigned char force:1; /* force state */ int (*power_check)(struct snd_soc_dapm_widget *w); diff --git a/include/sound/soc.h b/include/sound/soc.h index 5d234a8c2506..80dfac162723 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -15,6 +15,7 @@ #include <linux/platform_device.h> #include <linux/types.h> +#include <linux/notifier.h> #include <linux/workqueue.h> #include <linux/interrupt.h> #include <linux/kernel.h> @@ -212,6 +213,7 @@ struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; struct snd_soc_platform; +struct snd_soc_dai_link; struct snd_soc_codec; struct soc_enum; struct snd_soc_ac97_ops; @@ -260,6 +262,10 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); +void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, + struct notifier_block *nb); +void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, + struct notifier_block *nb); #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); @@ -363,6 +369,7 @@ struct snd_soc_jack { struct snd_soc_card *card; struct list_head pins; int status; + struct blocking_notifier_head notifier; }; /* SoC PCM stream information */ @@ -374,7 +381,8 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active:1; /* stream is in use */ + unsigned int active; /* num of active users of the stream */ + void *dma_data; /* used by platform code */ }; /* SoC audio ops */ @@ -461,14 +469,21 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); - int (*suspend)(struct snd_soc_dai *dai); - int (*resume)(struct snd_soc_dai *dai); + int (*suspend)(struct snd_soc_dai_link *dai_link); + int (*resume)(struct snd_soc_dai_link *dai_link); /* pcm creation and destruction */ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); + /* + * For platform caused delay reporting. + * Optional. + */ + snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, + struct snd_soc_dai *); + /* platform stream ops */ struct snd_pcm_ops *pcm_ops; }; diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h index ac0665264bdf..3f428d53195b 100644 --- a/include/sound/tlv320dac33-plat.h +++ b/include/sound/tlv320dac33-plat.h @@ -15,6 +15,7 @@ struct tlv320dac33_platform_data { int power_gpio; + int keep_bclk; /* Keep the BCLK running in FIFO modes */ u8 burst_bclkdiv; }; diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h new file mode 100644 index 000000000000..b4a0db2307ef --- /dev/null +++ b/include/sound/wm8903.h @@ -0,0 +1,249 @@ +/* + * linux/sound/wm8903.h -- Platform data for WM8903 + * + * Copyright 2010 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 version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM8903_H +#define __LINUX_SND_WM8903_H + +/* Used to enable configuration of a GPIO to all zeros */ +#define WM8903_GPIO_NO_CONFIG 0x8000 + +/* + * R6 (0x06) - Mic Bias Control 0 + */ +#define WM8903_MICDET_HYST_ENA 0x0080 /* MICDET_HYST_ENA */ +#define WM8903_MICDET_HYST_ENA_MASK 0x0080 /* MICDET_HYST_ENA */ +#define WM8903_MICDET_HYST_ENA_SHIFT 7 /* MICDET_HYST_ENA */ +#define WM8903_MICDET_HYST_ENA_WIDTH 1 /* MICDET_HYST_ENA */ +#define WM8903_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ +#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ +#define WM8903_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ +#define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ +#define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ +#define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ +#define WM8903_MICDET_ENA 0x0002 /* MICDET_ENA */ +#define WM8903_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ +#define WM8903_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ +#define WM8903_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ +#define WM8903_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ +#define WM8903_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ +#define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ +#define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ + +/* + * R116 (0x74) - GPIO Control 1 + */ +#define WM8903_GP1_FN_MASK 0x1F00 /* GP1_FN - [12:8] */ +#define WM8903_GP1_FN_SHIFT 8 /* GP1_FN - [12:8] */ +#define WM8903_GP1_FN_WIDTH 5 /* GP1_FN - [12:8] */ +#define WM8903_GP1_DIR 0x0080 /* GP1_DIR */ +#define WM8903_GP1_DIR_MASK 0x0080 /* GP1_DIR */ +#define WM8903_GP1_DIR_SHIFT 7 /* GP1_DIR */ +#define WM8903_GP1_DIR_WIDTH 1 /* GP1_DIR */ +#define WM8903_GP1_OP_CFG 0x0040 /* GP1_OP_CFG */ +#define WM8903_GP1_OP_CFG_MASK 0x0040 /* GP1_OP_CFG */ +#define WM8903_GP1_OP_CFG_SHIFT 6 /* GP1_OP_CFG */ +#define WM8903_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ +#define WM8903_GP1_IP_CFG 0x0020 /* GP1_IP_CFG */ +#define WM8903_GP1_IP_CFG_MASK 0x0020 /* GP1_IP_CFG */ +#define WM8903_GP1_IP_CFG_SHIFT 5 /* GP1_IP_CFG */ +#define WM8903_GP1_IP_CFG_WIDTH 1 /* GP1_IP_CFG */ +#define WM8903_GP1_LVL 0x0010 /* GP1_LVL */ +#define WM8903_GP1_LVL_MASK 0x0010 /* GP1_LVL */ +#define WM8903_GP1_LVL_SHIFT 4 /* GP1_LVL */ +#define WM8903_GP1_LVL_WIDTH 1 /* GP1_LVL */ +#define WM8903_GP1_PD 0x0008 /* GP1_PD */ +#define WM8903_GP1_PD_MASK 0x0008 /* GP1_PD */ +#define WM8903_GP1_PD_SHIFT 3 /* GP1_PD */ +#define WM8903_GP1_PD_WIDTH 1 /* GP1_PD */ +#define WM8903_GP1_PU 0x0004 /* GP1_PU */ +#define WM8903_GP1_PU_MASK 0x0004 /* GP1_PU */ +#define WM8903_GP1_PU_SHIFT 2 /* GP1_PU */ +#define WM8903_GP1_PU_WIDTH 1 /* GP1_PU */ +#define WM8903_GP1_INTMODE 0x0002 /* GP1_INTMODE */ +#define WM8903_GP1_INTMODE_MASK 0x0002 /* GP1_INTMODE */ +#define WM8903_GP1_INTMODE_SHIFT 1 /* GP1_INTMODE */ +#define WM8903_GP1_INTMODE_WIDTH 1 /* GP1_INTMODE */ +#define WM8903_GP1_DB 0x0001 /* GP1_DB */ +#define WM8903_GP1_DB_MASK 0x0001 /* GP1_DB */ +#define WM8903_GP1_DB_SHIFT 0 /* GP1_DB */ +#define WM8903_GP1_DB_WIDTH 1 /* GP1_DB */ + +/* + * R117 (0x75) - GPIO Control 2 + */ +#define WM8903_GP2_FN_MASK 0x1F00 /* GP2_FN - [12:8] */ +#define WM8903_GP2_FN_SHIFT 8 /* GP2_FN - [12:8] */ +#define WM8903_GP2_FN_WIDTH 5 /* GP2_FN - [12:8] */ +#define WM8903_GP2_DIR 0x0080 /* GP2_DIR */ +#define WM8903_GP2_DIR_MASK 0x0080 /* GP2_DIR */ +#define WM8903_GP2_DIR_SHIFT 7 /* GP2_DIR */ +#define WM8903_GP2_DIR_WIDTH 1 /* GP2_DIR */ +#define WM8903_GP2_OP_CFG 0x0040 /* GP2_OP_CFG */ +#define WM8903_GP2_OP_CFG_MASK 0x0040 /* GP2_OP_CFG */ +#define WM8903_GP2_OP_CFG_SHIFT 6 /* GP2_OP_CFG */ +#define WM8903_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ +#define WM8903_GP2_IP_CFG 0x0020 /* GP2_IP_CFG */ +#define WM8903_GP2_IP_CFG_MASK 0x0020 /* GP2_IP_CFG */ +#define WM8903_GP2_IP_CFG_SHIFT 5 /* GP2_IP_CFG */ +#define WM8903_GP2_IP_CFG_WIDTH 1 /* GP2_IP_CFG */ +#define WM8903_GP2_LVL 0x0010 /* GP2_LVL */ +#define WM8903_GP2_LVL_MASK 0x0010 /* GP2_LVL */ +#define WM8903_GP2_LVL_SHIFT 4 /* GP2_LVL */ +#define WM8903_GP2_LVL_WIDTH 1 /* GP2_LVL */ +#define WM8903_GP2_PD 0x0008 /* GP2_PD */ +#define WM8903_GP2_PD_MASK 0x0008 /* GP2_PD */ +#define WM8903_GP2_PD_SHIFT 3 /* GP2_PD */ +#define WM8903_GP2_PD_WIDTH 1 /* GP2_PD */ +#define WM8903_GP2_PU 0x0004 /* GP2_PU */ +#define WM8903_GP2_PU_MASK 0x0004 /* GP2_PU */ +#define WM8903_GP2_PU_SHIFT 2 /* GP2_PU */ +#define WM8903_GP2_PU_WIDTH 1 /* GP2_PU */ +#define WM8903_GP2_INTMODE 0x0002 /* GP2_INTMODE */ +#define WM8903_GP2_INTMODE_MASK 0x0002 /* GP2_INTMODE */ +#define WM8903_GP2_INTMODE_SHIFT 1 /* GP2_INTMODE */ +#define WM8903_GP2_INTMODE_WIDTH 1 /* GP2_INTMODE */ +#define WM8903_GP2_DB 0x0001 /* GP2_DB */ +#define WM8903_GP2_DB_MASK 0x0001 /* GP2_DB */ +#define WM8903_GP2_DB_SHIFT 0 /* GP2_DB */ +#define WM8903_GP2_DB_WIDTH 1 /* GP2_DB */ + +/* + * R118 (0x76) - GPIO Control 3 + */ +#define WM8903_GP3_FN_MASK 0x1F00 /* GP3_FN - [12:8] */ +#define WM8903_GP3_FN_SHIFT 8 /* GP3_FN - [12:8] */ +#define WM8903_GP3_FN_WIDTH 5 /* GP3_FN - [12:8] */ +#define WM8903_GP3_DIR 0x0080 /* GP3_DIR */ +#define WM8903_GP3_DIR_MASK 0x0080 /* GP3_DIR */ +#define WM8903_GP3_DIR_SHIFT 7 /* GP3_DIR */ +#define WM8903_GP3_DIR_WIDTH 1 /* GP3_DIR */ +#define WM8903_GP3_OP_CFG 0x0040 /* GP3_OP_CFG */ +#define WM8903_GP3_OP_CFG_MASK 0x0040 /* GP3_OP_CFG */ +#define WM8903_GP3_OP_CFG_SHIFT 6 /* GP3_OP_CFG */ +#define WM8903_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ +#define WM8903_GP3_IP_CFG 0x0020 /* GP3_IP_CFG */ +#define WM8903_GP3_IP_CFG_MASK 0x0020 /* GP3_IP_CFG */ +#define WM8903_GP3_IP_CFG_SHIFT 5 /* GP3_IP_CFG */ +#define WM8903_GP3_IP_CFG_WIDTH 1 /* GP3_IP_CFG */ +#define WM8903_GP3_LVL 0x0010 /* GP3_LVL */ +#define WM8903_GP3_LVL_MASK 0x0010 /* GP3_LVL */ +#define WM8903_GP3_LVL_SHIFT 4 /* GP3_LVL */ +#define WM8903_GP3_LVL_WIDTH 1 /* GP3_LVL */ +#define WM8903_GP3_PD 0x0008 /* GP3_PD */ +#define WM8903_GP3_PD_MASK 0x0008 /* GP3_PD */ +#define WM8903_GP3_PD_SHIFT 3 /* GP3_PD */ +#define WM8903_GP3_PD_WIDTH 1 /* GP3_PD */ +#define WM8903_GP3_PU 0x0004 /* GP3_PU */ +#define WM8903_GP3_PU_MASK 0x0004 /* GP3_PU */ +#define WM8903_GP3_PU_SHIFT 2 /* GP3_PU */ +#define WM8903_GP3_PU_WIDTH 1 /* GP3_PU */ +#define WM8903_GP3_INTMODE 0x0002 /* GP3_INTMODE */ +#define WM8903_GP3_INTMODE_MASK 0x0002 /* GP3_INTMODE */ +#define WM8903_GP3_INTMODE_SHIFT 1 /* GP3_INTMODE */ +#define WM8903_GP3_INTMODE_WIDTH 1 /* GP3_INTMODE */ +#define WM8903_GP3_DB 0x0001 /* GP3_DB */ +#define WM8903_GP3_DB_MASK 0x0001 /* GP3_DB */ +#define WM8903_GP3_DB_SHIFT 0 /* GP3_DB */ +#define WM8903_GP3_DB_WIDTH 1 /* GP3_DB */ + +/* + * R119 (0x77) - GPIO Control 4 + */ +#define WM8903_GP4_FN_MASK 0x1F00 /* GP4_FN - [12:8] */ +#define WM8903_GP4_FN_SHIFT 8 /* GP4_FN - [12:8] */ +#define WM8903_GP4_FN_WIDTH 5 /* GP4_FN - [12:8] */ +#define WM8903_GP4_DIR 0x0080 /* GP4_DIR */ +#define WM8903_GP4_DIR_MASK 0x0080 /* GP4_DIR */ +#define WM8903_GP4_DIR_SHIFT 7 /* GP4_DIR */ +#define WM8903_GP4_DIR_WIDTH 1 /* GP4_DIR */ +#define WM8903_GP4_OP_CFG 0x0040 /* GP4_OP_CFG */ +#define WM8903_GP4_OP_CFG_MASK 0x0040 /* GP4_OP_CFG */ +#define WM8903_GP4_OP_CFG_SHIFT 6 /* GP4_OP_CFG */ +#define WM8903_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ +#define WM8903_GP4_IP_CFG 0x0020 /* GP4_IP_CFG */ +#define WM8903_GP4_IP_CFG_MASK 0x0020 /* GP4_IP_CFG */ +#define WM8903_GP4_IP_CFG_SHIFT 5 /* GP4_IP_CFG */ +#define WM8903_GP4_IP_CFG_WIDTH 1 /* GP4_IP_CFG */ +#define WM8903_GP4_LVL 0x0010 /* GP4_LVL */ +#define WM8903_GP4_LVL_MASK 0x0010 /* GP4_LVL */ +#define WM8903_GP4_LVL_SHIFT 4 /* GP4_LVL */ +#define WM8903_GP4_LVL_WIDTH 1 /* GP4_LVL */ +#define WM8903_GP4_PD 0x0008 /* GP4_PD */ +#define WM8903_GP4_PD_MASK 0x0008 /* GP4_PD */ +#define WM8903_GP4_PD_SHIFT 3 /* GP4_PD */ +#define WM8903_GP4_PD_WIDTH 1 /* GP4_PD */ +#define WM8903_GP4_PU 0x0004 /* GP4_PU */ +#define WM8903_GP4_PU_MASK 0x0004 /* GP4_PU */ +#define WM8903_GP4_PU_SHIFT 2 /* GP4_PU */ +#define WM8903_GP4_PU_WIDTH 1 /* GP4_PU */ +#define WM8903_GP4_INTMODE 0x0002 /* GP4_INTMODE */ +#define WM8903_GP4_INTMODE_MASK 0x0002 /* GP4_INTMODE */ +#define WM8903_GP4_INTMODE_SHIFT 1 /* GP4_INTMODE */ +#define WM8903_GP4_INTMODE_WIDTH 1 /* GP4_INTMODE */ +#define WM8903_GP4_DB 0x0001 /* GP4_DB */ +#define WM8903_GP4_DB_MASK 0x0001 /* GP4_DB */ +#define WM8903_GP4_DB_SHIFT 0 /* GP4_DB */ +#define WM8903_GP4_DB_WIDTH 1 /* GP4_DB */ + +/* + * R120 (0x78) - GPIO Control 5 + */ +#define WM8903_GP5_FN_MASK 0x1F00 /* GP5_FN - [12:8] */ +#define WM8903_GP5_FN_SHIFT 8 /* GP5_FN - [12:8] */ +#define WM8903_GP5_FN_WIDTH 5 /* GP5_FN - [12:8] */ +#define WM8903_GP5_DIR 0x0080 /* GP5_DIR */ +#define WM8903_GP5_DIR_MASK 0x0080 /* GP5_DIR */ +#define WM8903_GP5_DIR_SHIFT 7 /* GP5_DIR */ +#define WM8903_GP5_DIR_WIDTH 1 /* GP5_DIR */ +#define WM8903_GP5_OP_CFG 0x0040 /* GP5_OP_CFG */ +#define WM8903_GP5_OP_CFG_MASK 0x0040 /* GP5_OP_CFG */ +#define WM8903_GP5_OP_CFG_SHIFT 6 /* GP5_OP_CFG */ +#define WM8903_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */ +#define WM8903_GP5_IP_CFG 0x0020 /* GP5_IP_CFG */ +#define WM8903_GP5_IP_CFG_MASK 0x0020 /* GP5_IP_CFG */ +#define WM8903_GP5_IP_CFG_SHIFT 5 /* GP5_IP_CFG */ +#define WM8903_GP5_IP_CFG_WIDTH 1 /* GP5_IP_CFG */ +#define WM8903_GP5_LVL 0x0010 /* GP5_LVL */ +#define WM8903_GP5_LVL_MASK 0x0010 /* GP5_LVL */ +#define WM8903_GP5_LVL_SHIFT 4 /* GP5_LVL */ +#define WM8903_GP5_LVL_WIDTH 1 /* GP5_LVL */ +#define WM8903_GP5_PD 0x0008 /* GP5_PD */ +#define WM8903_GP5_PD_MASK 0x0008 /* GP5_PD */ +#define WM8903_GP5_PD_SHIFT 3 /* GP5_PD */ +#define WM8903_GP5_PD_WIDTH 1 /* GP5_PD */ +#define WM8903_GP5_PU 0x0004 /* GP5_PU */ +#define WM8903_GP5_PU_MASK 0x0004 /* GP5_PU */ +#define WM8903_GP5_PU_SHIFT 2 /* GP5_PU */ +#define WM8903_GP5_PU_WIDTH 1 /* GP5_PU */ +#define WM8903_GP5_INTMODE 0x0002 /* GP5_INTMODE */ +#define WM8903_GP5_INTMODE_MASK 0x0002 /* GP5_INTMODE */ +#define WM8903_GP5_INTMODE_SHIFT 1 /* GP5_INTMODE */ +#define WM8903_GP5_INTMODE_WIDTH 1 /* GP5_INTMODE */ +#define WM8903_GP5_DB 0x0001 /* GP5_DB */ +#define WM8903_GP5_DB_MASK 0x0001 /* GP5_DB */ +#define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ +#define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ + +struct wm8903_platform_data { + bool irq_active_low; /* Set if IRQ active low, default high */ + + /* Default register value for R6 (Mic bias), used to configure + * microphone detection. In conjunction with gpio_cfg this + * can be used to route the microphone status signals out onto + * the GPIOs for use with snd_soc_jack_add_gpios(). + */ + u16 micdet_cfg; + + int micdet_delay; /* Delay after microphone detection (ms) */ + + u32 gpio_cfg[5]; /* Default register values for GPIO pin mux */ +}; + +#endif diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index d66575a601be..898be3a8db9a 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -15,8 +15,111 @@ #ifndef __MFD_WM8994_PDATA_H__ #define __MFD_WM8994_PDATA_H__ -#define WM8904_DRC_REGS 4 -#define WM8904_EQ_REGS 25 +/* Used to enable configuration of a GPIO to all zeros */ +#define WM8904_GPIO_NO_CONFIG 0x8000 + +/* + * R6 (0x06) - Mic Bias Control 0 + */ +#define WM8904_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ +#define WM8904_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ +#define WM8904_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ +#define WM8904_MICDET_ENA 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ +#define WM8904_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ +#define WM8904_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ +#define WM8904_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ + +/* + * R7 (0x07) - Mic Bias Control 1 + */ +#define WM8904_MIC_DET_FILTER_ENA 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_MASK 0x8000 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_SHIFT 15 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_DET_FILTER_ENA_WIDTH 1 /* MIC_DET_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_MASK 0x4000 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT 14 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH 1 /* MIC_SHORT_FILTER_ENA */ +#define WM8904_MICBIAS_SEL_MASK 0x0007 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_SHIFT 0 /* MICBIAS_SEL - [2:0] */ +#define WM8904_MICBIAS_SEL_WIDTH 3 /* MICBIAS_SEL - [2:0] */ + + +/* + * R121 (0x79) - GPIO Control 1 + */ +#define WM8904_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_SHIFT 5 /* GPIO1_PU */ +#define WM8904_GPIO1_PU_WIDTH 1 /* GPIO1_PU */ +#define WM8904_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_SHIFT 4 /* GPIO1_PD */ +#define WM8904_GPIO1_PD_WIDTH 1 /* GPIO1_PD */ +#define WM8904_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */ +#define WM8904_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */ + +/* + * R122 (0x7A) - GPIO Control 2 + */ +#define WM8904_GPIO2_PU 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_MASK 0x0020 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_SHIFT 5 /* GPIO2_PU */ +#define WM8904_GPIO2_PU_WIDTH 1 /* GPIO2_PU */ +#define WM8904_GPIO2_PD 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_MASK 0x0010 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_SHIFT 4 /* GPIO2_PD */ +#define WM8904_GPIO2_PD_WIDTH 1 /* GPIO2_PD */ +#define WM8904_GPIO2_SEL_MASK 0x000F /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_SHIFT 0 /* GPIO2_SEL - [3:0] */ +#define WM8904_GPIO2_SEL_WIDTH 4 /* GPIO2_SEL - [3:0] */ + +/* + * R123 (0x7B) - GPIO Control 3 + */ +#define WM8904_GPIO3_PU 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_MASK 0x0020 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_SHIFT 5 /* GPIO3_PU */ +#define WM8904_GPIO3_PU_WIDTH 1 /* GPIO3_PU */ +#define WM8904_GPIO3_PD 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_MASK 0x0010 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_SHIFT 4 /* GPIO3_PD */ +#define WM8904_GPIO3_PD_WIDTH 1 /* GPIO3_PD */ +#define WM8904_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_SHIFT 0 /* GPIO3_SEL - [3:0] */ +#define WM8904_GPIO3_SEL_WIDTH 4 /* GPIO3_SEL - [3:0] */ + +/* + * R124 (0x7C) - GPIO Control 4 + */ +#define WM8904_GPI7_ENA 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_MASK 0x0200 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_SHIFT 9 /* GPI7_ENA */ +#define WM8904_GPI7_ENA_WIDTH 1 /* GPI7_ENA */ +#define WM8904_GPI8_ENA 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_MASK 0x0100 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_SHIFT 8 /* GPI8_ENA */ +#define WM8904_GPI8_ENA_WIDTH 1 /* GPI8_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_MASK 0x0080 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT 7 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH 1 /* GPIO_BCLK_MODE_ENA */ +#define WM8904_GPIO_BCLK_SEL_MASK 0x000F /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_SHIFT 0 /* GPIO_BCLK_SEL - [3:0] */ +#define WM8904_GPIO_BCLK_SEL_WIDTH 4 /* GPIO_BCLK_SEL - [3:0] */ + +#define WM8904_MIC_REGS 2 +#define WM8904_GPIO_REGS 4 +#define WM8904_DRC_REGS 4 +#define WM8904_EQ_REGS 25 /** * DRC configurations are specified with a label and a set of register @@ -52,6 +155,9 @@ struct wm8904_pdata { int num_retune_mobile_cfgs; struct wm8904_retune_mobile_cfg *retune_mobile_cfgs; + + u32 gpio_cfg[WM8904_GPIO_REGS]; + u32 mic_cfg[WM8904_MIC_REGS]; }; #endif diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h new file mode 100644 index 000000000000..74e9a95529c5 --- /dev/null +++ b/include/sound/wm8960.h @@ -0,0 +1,24 @@ +/* + * wm8960.h -- WM8960 Soc Audio driver platform data + * + * 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 _WM8960_PDATA_H +#define _WM8960_PDATA_H + +#define WM8960_DRES_400R 0 +#define WM8960_DRES_200R 1 +#define WM8960_DRES_600R 2 +#define WM8960_DRES_150R 3 +#define WM8960_DRES_MAX 3 + +struct wm8960_data { + bool capless; /* Headphone outputs configured in capless mode */ + + int dres; /* Discharge resistance for headphone outputs */ +}; + +#endif diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 743ac6a29065..fd51fa8b06a1 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -205,6 +205,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (!rtd->dma_desc_array) goto err1; + rtd->dma_ch = -1; runtime->private_data = rtd; return 0; diff --git a/sound/core/jack.c b/sound/core/jack.c index f705eec7372a..3813e7b04d05 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -23,7 +23,7 @@ #include <sound/jack.h> #include <sound/core.h> -static int jack_types[] = { +static int jack_switch_types[] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -55,7 +55,7 @@ static int snd_jack_dev_register(struct snd_device *device) { struct snd_jack *jack = device->device_data; struct snd_card *card = device->card; - int err; + int err, i; snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); @@ -65,6 +65,19 @@ static int snd_jack_dev_register(struct snd_device *device) if (!jack->input_dev->dev.parent) jack->input_dev->dev.parent = snd_card_get_device_link(card); + /* Add capabilities for any keys that are enabled */ + for (i = 0; i < ARRAY_SIZE(jack->key); i++) { + int testbit = SND_JACK_BTN_0 >> i; + + if (!(jack->type & testbit)) + continue; + + if (!jack->key[i]) + jack->key[i] = BTN_0 + i; + + input_set_capability(jack->input_dev, EV_KEY, jack->key[i]); + } + err = input_register_device(jack->input_dev); if (err == 0) jack->registered = 1; @@ -112,10 +125,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < ARRAY_SIZE(jack_types); i++) + for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, - jack_types[i]); + jack_switch_types[i]); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -151,6 +164,43 @@ void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) EXPORT_SYMBOL(snd_jack_set_parent); /** + * snd_jack_set_key - Set a key mapping on a jack + * + * @jack: The jack to configure + * @type: Jack report type for this key + * @keytype: Input layer key type to be reported + * + * Map a SND_JACK_BTN_ button type to an input layer key, allowing + * reporting of keys on accessories via the jack abstraction. If no + * mapping is provided but keys are enabled in the jack type then + * BTN_n numeric buttons will be reported. + * + * Note that this is intended to be use by simple devices with small + * numbers of keys that can be reported. It is also possible to + * access the input device directly - devices with complex input + * capabilities on accessories should consider doing this rather than + * using this abstraction. + * + * This function may only be called prior to registration of the jack. + */ +int snd_jack_set_key(struct snd_jack *jack, enum snd_jack_types type, + int keytype) +{ + int key = fls(SND_JACK_BTN_0) - fls(type); + + WARN_ON(jack->registered); + + if (!keytype || key >= ARRAY_SIZE(jack->key)) + return -EINVAL; + + jack->type |= type; + jack->key[key] = keytype; + + return 0; +} +EXPORT_SYMBOL(snd_jack_set_key); + +/** * snd_jack_report - Report the current status of a jack * * @jack: The jack to report status for @@ -163,10 +213,19 @@ void snd_jack_report(struct snd_jack *jack, int status) if (!jack) return; - for (i = 0; i < ARRAY_SIZE(jack_types); i++) { + for (i = 0; i < ARRAY_SIZE(jack->key); i++) { + int testbit = SND_JACK_BTN_0 >> i; + + if (jack->type & testbit) + input_report_key(jack->input_dev, jack->key[i], + status & testbit); + } + + for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) { int testbit = 1 << i; if (jack->type & testbit) - input_report_switch(jack->input_dev, jack_types[i], + input_report_switch(jack->input_dev, + jack_switch_types[i], status & testbit); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 872887624030..9681518aa392 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3434,14 +3434,28 @@ out: #endif /* CONFIG_SND_SUPPORT_OLD_API */ #ifndef CONFIG_MMU -unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, - unsigned long len, unsigned long pgoff, - unsigned long flags) -{ - return 0; +static unsigned long snd_pcm_get_unmapped_area(struct file *file, + unsigned long addr, + unsigned long len, + unsigned long pgoff, + unsigned long flags) +{ + struct snd_pcm_file *pcm_file = file->private_data; + struct snd_pcm_substream *substream = pcm_file->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long offset = pgoff << PAGE_SHIFT; + + switch (offset) { + case SNDRV_PCM_MMAP_OFFSET_STATUS: + return (unsigned long)runtime->status; + case SNDRV_PCM_MMAP_OFFSET_CONTROL: + return (unsigned long)runtime->control; + default: + return (unsigned long)runtime->dma_area + offset; + } } #else -# define dummy_get_unmapped_area NULL +# define snd_pcm_get_unmapped_area NULL #endif /* @@ -3460,7 +3474,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, }, { .owner = THIS_MODULE, @@ -3473,6 +3487,6 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, } }; diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 725fef0f59a3..a4127bab9231 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -363,13 +363,13 @@ static void vidc_audio_trigger(int dev, int enable_bits) struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { - if (!(adev->flags & DMA_ACTIVE)) { + if (!(adev->dmap_out->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ - adev->flags |= DMA_ACTIVE; + adev->dmap_out->flags |= DMA_ACTIVE; dma_interrupt = vidc_audio_dma_interrupt; vidc_sound_dma_irq(0, NULL); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0e76ac2b2ace..d72bcff8d271 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4218,7 +4218,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, break; case AC_JACK_MIC_IN: { int preferred, alt; - if (loc == AC_JACK_LOC_FRONT) { + if (loc == AC_JACK_LOC_FRONT || + (loc & 0x30) == AC_JACK_LOC_INTERNAL) { preferred = AUTO_PIN_FRONT_MIC; alt = AUTO_PIN_MIC; } else { diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 194a28c54992..61682e1d09da 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1591,6 +1591,21 @@ static int patch_cxt5047(struct hda_codec *codec) #endif } spec->vmaster_nid = 0x13; + + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptops have really bad sound over 0 dB on NID 0x10. + * Fix max PCM level to 0 dB (originally it has 0x1e steps + * with 0 dB offset 0x17) + */ + snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); + break; + } + return 0; } diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 70669a246902..3c10c0b149f4 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -538,8 +538,6 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0003, .name = "MCP77/78 HDMI", @@ -550,12 +548,16 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000c, .name = "MCP89 HDMI", + { .id = 0x10de000a, .name = "GT220 HDMI", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000b, .name = "GT21x HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000c, .name = "MCP89 HDMI", + .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; @@ -564,11 +566,12 @@ MODULE_ALIAS("snd-hda-codec-id:10de0003"); MODULE_ALIAS("snd-hda-codec-id:10de0005"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); -MODULE_ALIAS("snd-hda-codec-id:10de0067"); -MODULE_ALIAS("snd-hda-codec-id:10de8001"); -MODULE_ALIAS("snd-hda-codec-id:10de000c"); +MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ec57633af88..06b2b50dad55 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -275,6 +275,18 @@ struct alc_mic_route { #define MUX_IDX_UNDEF ((unsigned char)-1) +struct alc_customize_define { + unsigned int sku_cfg; + unsigned char port_connectivity; + unsigned char check_sum; + unsigned char customization; + unsigned char external_amp; + unsigned int enable_pcbeep:1; + unsigned int platform_type:1; + unsigned int swap:1; + unsigned int override:1; +}; + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -332,6 +344,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; + struct alc_customize_define cdefine; struct snd_array kctls; struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; @@ -1247,6 +1260,62 @@ static void alc_init_auto_mic(struct hda_codec *codec) spec->unsol_event = alc_sku_unsol_event; } +static int alc_auto_parse_customize_define(struct hda_codec *codec) +{ + unsigned int ass, tmp, i; + unsigned nid = 0; + struct alc_spec *spec = codec->spec; + + ass = codec->subsystem_id & 0xffff; + if (ass != codec->bus->pci->subsystem_device && (ass & 1)) + goto do_sku; + + nid = 0x1d; + if (codec->vendor_id == 0x10ec0260) + nid = 0x17; + ass = snd_hda_codec_get_pincfg(codec, nid); + + if (!(ass & 1)) { + printk(KERN_INFO "hda_codec: %s: SKU not ready 0x%08x\n", + codec->chip_name, ass); + return -1; + } + + /* check sum */ + tmp = 0; + for (i = 1; i < 16; i++) { + if ((ass >> i) & 1) + tmp++; + } + if (((ass >> 16) & 0xf) != tmp) + return -1; + + spec->cdefine.port_connectivity = ass >> 30; + spec->cdefine.enable_pcbeep = (ass & 0x100000) >> 20; + spec->cdefine.check_sum = (ass >> 16) & 0xf; + spec->cdefine.customization = ass >> 8; +do_sku: + spec->cdefine.sku_cfg = ass; + spec->cdefine.external_amp = (ass & 0x38) >> 3; + spec->cdefine.platform_type = (ass & 0x4) >> 2; + spec->cdefine.swap = (ass & 0x2) >> 1; + spec->cdefine.override = ass & 0x1; + + snd_printd("SKU: Nid=0x%x sku_cfg=0x%08x\n", + nid, spec->cdefine.sku_cfg); + snd_printd("SKU: port_connectivity=0x%x\n", + spec->cdefine.port_connectivity); + snd_printd("SKU: enable_pcbeep=0x%x\n", spec->cdefine.enable_pcbeep); + snd_printd("SKU: check_sum=0x%08x\n", spec->cdefine.check_sum); + snd_printd("SKU: customization=0x%08x\n", spec->cdefine.customization); + snd_printd("SKU: external_amp=0x%x\n", spec->cdefine.external_amp); + snd_printd("SKU: platform_type=0x%x\n", spec->cdefine.platform_type); + snd_printd("SKU: swap=0x%x\n", spec->cdefine.swap); + snd_printd("SKU: override=0x%x\n", spec->cdefine.override); + + return 0; +} + /* check subsystem ID and set up device-specific initialization; * return 1 if initialized, 0 if invalid SSID */ @@ -2532,8 +2601,6 @@ static int alc_build_controls(struct hda_codec *codec) return err; } - alc_free_kctls(codec); /* no longer needed */ - /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); if (!kctl) @@ -2602,6 +2669,9 @@ static int alc_build_controls(struct hda_codec *codec) } } } + + alc_free_kctls(codec); /* no longer needed */ + return 0; } @@ -3779,6 +3849,17 @@ static struct hda_codec_ops alc_patch_ops = { .reboot_notify = alc_shutup, }; +/* replace the codec chip_name with the given string */ +static int alc_codec_rename(struct hda_codec *codec, const char *name) +{ + kfree(codec->chip_name); + codec->chip_name = kstrdup(name, GFP_KERNEL); + if (!codec->chip_name) { + alc_free(codec); + return -ENOMEM; + } + return 0; +} /* * Test configuration for debugging @@ -10267,6 +10348,8 @@ static int patch_alc882(struct hda_codec *codec) codec->spec = spec; + alc_auto_parse_customize_define(codec); + switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: @@ -10362,7 +10445,9 @@ static int patch_alc882(struct hda_codec *codec) } set_capture_mixer(codec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + + if (spec->cdefine.enable_pcbeep) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -12146,6 +12231,7 @@ static int patch_alc262(struct hda_codec *codec) snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PROC_COEF, tmp | 0x80); } #endif + alc_auto_parse_customize_define(codec); alc_fix_pll_init(codec, 0x20, 0x0a, 10); @@ -12224,7 +12310,7 @@ static int patch_alc262(struct hda_codec *codec) } if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(codec); - if (!spec->no_analog) + if (!spec->no_analog && spec->cdefine.enable_pcbeep) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -14094,17 +14180,17 @@ static int patch_alc269(struct hda_codec *codec) codec->spec = spec; - alc_fix_pll_init(codec, 0x20, 0x04, 15); + alc_auto_parse_customize_define(codec); if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){ - kfree(codec->chip_name); - codec->chip_name = kstrdup("ALC259", GFP_KERNEL); - if (!codec->chip_name) { - alc_free(codec); - return -ENOMEM; - } + if (codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) + alc_codec_rename(codec, "ALC271X"); + else + alc_codec_rename(codec, "ALC259"); is_alc269vb = 1; - } + } else + alc_fix_pll_init(codec, 0x20, 0x04, 15); board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, @@ -14164,7 +14250,8 @@ static int patch_alc269(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (spec->cdefine.enable_pcbeep) + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -18314,16 +18401,16 @@ static int patch_alc662(struct hda_codec *codec) codec->spec = spec; + alc_auto_parse_customize_define(codec); + alc_fix_pll_init(codec, 0x20, 0x04, 15); - if (alc_read_coef_idx(codec, 0)==0x8020){ - kfree(codec->chip_name); - codec->chip_name = kstrdup("ALC661", GFP_KERNEL); - if (!codec->chip_name) { - alc_free(codec); - return -ENOMEM; - } - } + if (alc_read_coef_idx(codec, 0) == 0x8020) + alc_codec_rename(codec, "ALC661"); + else if ((alc_read_coef_idx(codec, 0) & (1 << 14)) && + codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) + alc_codec_rename(codec, "ALC272X"); board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, @@ -18373,18 +18460,20 @@ static int patch_alc662(struct hda_codec *codec) if (!spec->cap_mixer) set_capture_mixer(codec); - switch (codec->vendor_id) { - case 0x10ec0662: - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - break; - case 0x10ec0272: - case 0x10ec0663: - case 0x10ec0665: - set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); - break; - case 0x10ec0273: - set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); - break; + if (spec->cdefine.enable_pcbeep) { + switch (codec->vendor_id) { + case 0x10ec0662: + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + break; + case 0x10ec0272: + case 0x10ec0663: + case 0x10ec0665: + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + break; + case 0x10ec0273: + set_beep_amp(spec, 0x0b, 0x03, HDA_INPUT); + break; + } } spec->vmaster_nid = 0x02; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c416bb18a57..c4be3fab94e5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 9e66f6d306f8..2f6252266a02 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1956,11 +1956,10 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) return 0; } - /* - * initialize the chip + * reset the chip */ -static int __devinit aureon_init(struct snd_ice1712 *ice) +static int aureon_reset(struct snd_ice1712 *ice) { static const unsigned short wm_inits_aureon[] = { /* These come first to reduce init pop noise */ @@ -2047,30 +2046,10 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) 0x0605, /* slave, 24bit, MSB on second OSCLK, SDOUT for right channel when OLRCK is high */ (unsigned short)-1 }; - struct aureon_spec *spec; unsigned int tmp; const unsigned short *p; - int err, i; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - ice->spec = spec; - - if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { - ice->num_total_dacs = 6; - ice->num_total_adcs = 2; - } else { - /* aureon 7.1 and prodigy 7.1 */ - ice->num_total_dacs = 8; - ice->num_total_adcs = 2; - } - - /* to remeber the register values of CS8415 */ - ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (!ice->akm) - return -ENOMEM; - ice->akm_codecs = 1; + int err; + struct aureon_spec *spec = ice->spec; err = aureon_ac97_init(ice); if (err != 0) @@ -2118,6 +2097,61 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* initialize PCA9554 pin directions & set default input */ aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ + return 0; +} + +/* + * suspend/resume + */ +#ifdef CONFIG_PM +static int aureon_resume(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec = ice->spec; + int err, i; + + err = aureon_reset(ice); + if (err != 0) + return err; + + /* workaround for poking volume with alsamixer after resume: + * just set stored volume again */ + for (i = 0; i < ice->num_total_dacs; i++) + wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); + return 0; +} +#endif + +/* + * initialize the chip + */ +static int __devinit aureon_init(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec; + int i, err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { + ice->num_total_dacs = 6; + ice->num_total_adcs = 2; + } else { + /* aureon 7.1 and prodigy 7.1 */ + ice->num_total_dacs = 8; + ice->num_total_adcs = 2; + } + + /* to remeber the register values of CS8415 */ + ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (!ice->akm) + return -ENOMEM; + ice->akm_codecs = 1; + + err = aureon_reset(ice); + if (err != 0) + return err; spec->master[0] = WM_VOL_MUTE; spec->master[1] = WM_VOL_MUTE; @@ -2126,6 +2160,11 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } +#ifdef CONFIG_PM + ice->pm_resume = aureon_resume; + ice->pm_suspend_enabled = 1; +#endif + return 0; } diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 9ef6b96373f5..f6b3cc04b34b 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, 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 = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; @@ -415,9 +415,12 @@ static void atmel_pcm_free_dma_buffers(struct snd_pcm *pcm) } #ifdef CONFIG_PM -static int atmel_pcm_suspend(struct snd_soc_dai *dai) +static int atmel_pcm_suspend(struct snd_soc_dai_link *dai_link) { - struct snd_pcm_runtime *runtime = dai->runtime; + struct snd_pcm *pcm = dai_link->pcm; + struct snd_pcm_str *stream = &pcm->streams[0]; + struct snd_pcm_substream *substream = stream->substream; + struct snd_pcm_runtime *runtime = substream->runtime; struct atmel_runtime_data *prtd; struct atmel_pcm_dma_params *params; @@ -439,9 +442,12 @@ static int atmel_pcm_suspend(struct snd_soc_dai *dai) return 0; } -static int atmel_pcm_resume(struct snd_soc_dai *dai) +static int atmel_pcm_resume(struct snd_soc_dai_link *dai_link) { - struct snd_pcm_runtime *runtime = dai->runtime; + struct snd_pcm *pcm = dai_link->pcm; + struct snd_pcm_str *stream = &pcm->streams[0]; + struct snd_pcm_substream *substream = stream->substream; + struct snd_pcm_runtime *runtime = substream->runtime; struct atmel_runtime_data *prtd; struct atmel_pcm_dma_params *params; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e588e63f18d2..0b59806905d1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_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() + * The snd_soc_pcm_stream->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; + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params); channels = params_channels(params); diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 97f1a251e446..8ef25025f3dc 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -49,13 +49,14 @@ config SND_BF5XX_SOC_AD1836 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" +config SND_BF5XX_SOC_AD193X + tristate "SoC AD193X Audio support for Blackfin" depends on SND_BF5XX_TDM select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 + select SND_SOC_AD193X help - Say Y if you want to add support for AD1938 codec on Blackfin. + Say Y if you want to add support for AD193X codec on Blackfin. + This driver supports AD1936, AD1937, AD1938 and AD1939. config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 87e30423912f..49af3f32aec8 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -20,10 +20,10 @@ snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o -snd-ad1938-objs := bf5xx-ad1938.o +snd-ad193x-objs := bf5xx-ad193x.o obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(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 -obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad193x.c index 2ef1e5013b8c..b8c9060cfd8e 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -1,9 +1,9 @@ /* - * File: sound/soc/blackfin/bf5xx-ad1938.c + * File: sound/soc/blackfin/bf5xx-ad193x.c * Author: Barry Song <Barry.Song@analog.com> * * Created: Thur June 4 2009 - * Description: Board driver for ad1938 sound chip + * Description: Board driver for ad193x sound chip * * Bugs: Enter bugs at http://blackfin.uclinux.org/ * @@ -38,15 +38,15 @@ #include <asm/dma.h> #include <asm/portmux.h> -#include "../codecs/ad1938.h" +#include "../codecs/ad193x.h" #include "bf5xx-sport.h" #include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" -static struct snd_soc_card bf5xx_ad1938; +static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) +static int bf5xx_ad193x_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; @@ -55,7 +55,7 @@ static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) return 0; } -static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, +static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -89,61 +89,61 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops bf5xx_ad1938_ops = { - .startup = bf5xx_ad1938_startup, - .hw_params = bf5xx_ad1938_hw_params, +static struct snd_soc_ops bf5xx_ad193x_ops = { + .startup = bf5xx_ad193x_startup, + .hw_params = bf5xx_ad193x_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad1938_dai = { - .name = "ad1938", - .stream_name = "AD1938", +static struct snd_soc_dai_link bf5xx_ad193x_dai = { + .name = "ad193x", + .stream_name = "AD193X", .cpu_dai = &bf5xx_tdm_dai, - .codec_dai = &ad1938_dai, - .ops = &bf5xx_ad1938_ops, + .codec_dai = &ad193x_dai, + .ops = &bf5xx_ad193x_ops, }; -static struct snd_soc_card bf5xx_ad1938 = { - .name = "bf5xx_ad1938", +static struct snd_soc_card bf5xx_ad193x = { + .name = "bf5xx_ad193x", .platform = &bf5xx_tdm_soc_platform, - .dai_link = &bf5xx_ad1938_dai, + .dai_link = &bf5xx_ad193x_dai, .num_links = 1, }; -static struct snd_soc_device bf5xx_ad1938_snd_devdata = { - .card = &bf5xx_ad1938, - .codec_dev = &soc_codec_dev_ad1938, +static struct snd_soc_device bf5xx_ad193x_snd_devdata = { + .card = &bf5xx_ad193x, + .codec_dev = &soc_codec_dev_ad193x, }; -static struct platform_device *bfxx_ad1938_snd_device; +static struct platform_device *bfxx_ad193x_snd_device; -static int __init bf5xx_ad1938_init(void) +static int __init bf5xx_ad193x_init(void) { int ret; - bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1938_snd_device) + bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad193x_snd_device) return -ENOMEM; - platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata); - bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev; - ret = platform_device_add(bfxx_ad1938_snd_device); + platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x_snd_devdata); + bf5xx_ad193x_snd_devdata.dev = &bfxx_ad193x_snd_device->dev; + ret = platform_device_add(bfxx_ad193x_snd_device); if (ret) - platform_device_put(bfxx_ad1938_snd_device); + platform_device_put(bfxx_ad193x_snd_device); return ret; } -static void __exit bf5xx_ad1938_exit(void) +static void __exit bf5xx_ad193x_exit(void) { - platform_device_unregister(bfxx_ad1938_snd_device); + platform_device_unregister(bfxx_ad193x_snd_device); } -module_init(bf5xx_ad1938_init); -module_exit(bf5xx_ad1938_exit); +module_init(bf5xx_ad193x_init); +module_exit(bf5xx_ad193x_exit); /* Module information */ MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD1938 board driver"); +MODULE_DESCRIPTION("ALSA SoC AD193X board driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 2e63dea73e9c..a86e8cc0b2d3 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -34,33 +34,7 @@ #include <linux/wait.h> #include <linux/workqueue.h> #include <asm/dma.h> - -struct sport_register { - u16 tcr1; u16 reserved0; - u16 tcr2; u16 reserved1; - u16 tclkdiv; u16 reserved2; - u16 tfsdiv; u16 reserved3; - u32 tx; - u32 reserved_l0; - u32 rx; - u32 reserved_l1; - u16 rcr1; u16 reserved4; - u16 rcr2; u16 reserved5; - u16 rclkdiv; u16 reserved6; - u16 rfsdiv; u16 reserved7; - u16 stat; u16 reserved8; - u16 chnl; u16 reserved9; - u16 mcmc1; u16 reserved10; - u16 mcmc2; u16 reserved11; - u32 mtcs0; - u32 mtcs1; - u32 mtcs2; - u32 mtcs3; - u32 mrcs0; - u32 mrcs1; - u32 mrcs2; - u32 mrcs3; -}; +#include <asm/bfin_sport.h> #define DESC_ELEMENT_COUNT 9 diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1743d565e996..bc0ab47e156b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -13,7 +13,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER - select SND_SOC_AD1938 if SPI_MASTER + select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_DA7210 if I2C @@ -34,6 +35,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -90,7 +92,7 @@ config SND_SOC_AC97_CODEC config SND_SOC_AD1836 tristate -config SND_SOC_AD1938 +config SND_SOC_AD193X tristate config SND_SOC_AD1980 @@ -114,6 +116,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate @@ -164,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dd5ce6df6292..337904167358 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,6 @@ snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o -snd-soc-ad1938-objs := ad1938.o +snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ads117x-objs := ads117x.o @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o @@ -21,6 +22,7 @@ snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -62,7 +64,7 @@ snd-soc-wm2000-objs := wm2000.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o -obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o +obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o @@ -70,6 +72,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o @@ -83,6 +86,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c deleted file mode 100644 index c233810d463d..000000000000 --- a/sound/soc/codecs/ad1938.c +++ /dev/null @@ -1,521 +0,0 @@ -/* - * File: sound/soc/codecs/ad1938.c - * Author: Barry Song <Barry.Song@analog.com> - * - * Created: June 04 2009 - * Description: Driver for AD1938 sound chip - * - * Modified: - * Copyright 2009 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/init.h> -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <sound/tlv.h> -#include <sound/soc-dapm.h> -#include <linux/spi/spi.h> -#include "ad1938.h" - -/* codec private data */ -struct ad1938_priv { - struct snd_soc_codec codec; - u8 reg_cache[AD1938_NUM_REGS]; -}; - -/* ad1938 register cache & default register settings */ -static const u8 ad1938_reg[AD1938_NUM_REGS] = { - 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, -}; - -static struct snd_soc_codec *ad1938_codec; -struct snd_soc_codec_device soc_codec_dev_ad1938; -static int ad1938_register(struct ad1938_priv *ad1938); -static void ad1938_unregister(struct ad1938_priv *ad1938); - -/* - * AD1938 volume/mute/de-emphasis etc. controls - */ -static const char *ad1938_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; - -static const struct soc_enum ad1938_deemp_enum = - SOC_ENUM_SINGLE(AD1938_DAC_CTRL2, 1, 4, ad1938_deemp); - -static const struct snd_kcontrol_new ad1938_snd_controls[] = { - /* DAC volume control */ - SOC_DOUBLE_R("DAC1 Volume", AD1938_DAC_L1_VOL, - AD1938_DAC_R1_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC2 Volume", AD1938_DAC_L2_VOL, - AD1938_DAC_R2_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC3 Volume", AD1938_DAC_L3_VOL, - AD1938_DAC_R3_VOL, 0, 0xFF, 1), - SOC_DOUBLE_R("DAC4 Volume", AD1938_DAC_L4_VOL, - AD1938_DAC_R4_VOL, 0, 0xFF, 1), - - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, - AD1938_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, - AD1938_ADCR2_MUTE, 1, 1), - - /* DAC switch control */ - SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, - AD1938_DACR1_MUTE, 1, 1), - SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, - AD1938_DACR2_MUTE, 1, 1), - SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, - AD1938_DACR3_MUTE, 1, 1), - SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, - AD1938_DACR4_MUTE, 1, 1), - - /* ADC high-pass filter */ - SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0, - AD1938_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad1938_deemp_enum), -}; - -static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { - SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_SUPPLY("PLL_PWR", AD1938_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_OUTPUT("DAC1OUT"), - SND_SOC_DAPM_OUTPUT("DAC2OUT"), - SND_SOC_DAPM_OUTPUT("DAC3OUT"), - SND_SOC_DAPM_OUTPUT("DAC4OUT"), - SND_SOC_DAPM_INPUT("ADC1IN"), - SND_SOC_DAPM_INPUT("ADC2IN"), -}; - -static const struct snd_soc_dapm_route audio_paths[] = { - { "DAC", NULL, "PLL_PWR" }, - { "ADC", NULL, "PLL_PWR" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, - { "DAC1OUT", "DAC1 Switch", "DAC" }, - { "DAC2OUT", "DAC2 Switch", "DAC" }, - { "DAC3OUT", "DAC3 Switch", "DAC" }, - { "DAC4OUT", "DAC4 Switch", "DAC" }, - { "ADC", "ADC1 Switch", "ADC1IN" }, - { "ADC", "ADC2 Switch", "ADC2IN" }, -}; - -/* - * DAI ops entries - */ - -static int ad1938_mute(struct snd_soc_dai *dai, int mute) -{ - struct snd_soc_codec *codec = dai->codec; - int reg; - - reg = snd_soc_read(codec, AD1938_DAC_CTRL2); - reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & - (~AD1938_DAC_MASTER_MUTE); - snd_soc_write(codec, AD1938_DAC_CTRL2, reg); - - return 0; -} - -static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, - unsigned int rx_mask, int slots, int width) -{ - struct snd_soc_codec *codec = dai->codec; - int dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); - int adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); - - dac_reg &= ~AD1938_DAC_CHAN_MASK; - adc_reg &= ~AD1938_ADC_CHAN_MASK; - - switch (slots) { - case 2: - dac_reg |= AD1938_DAC_2_CHANNELS << AD1938_DAC_CHAN_SHFT; - adc_reg |= AD1938_ADC_2_CHANNELS << AD1938_ADC_CHAN_SHFT; - break; - case 4: - dac_reg |= AD1938_DAC_4_CHANNELS << AD1938_DAC_CHAN_SHFT; - adc_reg |= AD1938_ADC_4_CHANNELS << AD1938_ADC_CHAN_SHFT; - break; - case 8: - dac_reg |= AD1938_DAC_8_CHANNELS << AD1938_DAC_CHAN_SHFT; - adc_reg |= AD1938_ADC_8_CHANNELS << AD1938_ADC_CHAN_SHFT; - break; - case 16: - dac_reg |= AD1938_DAC_16_CHANNELS << AD1938_DAC_CHAN_SHFT; - adc_reg |= AD1938_ADC_16_CHANNELS << AD1938_ADC_CHAN_SHFT; - break; - default: - return -EINVAL; - } - - snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); - snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); - - return 0; -} - -static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) -{ - struct snd_soc_codec *codec = codec_dai->codec; - int adc_reg, dac_reg; - - adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2); - dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1); - - /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S - * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) - */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - adc_reg &= ~AD1938_ADC_SERFMT_MASK; - adc_reg |= AD1938_ADC_SERFMT_TDM; - break; - case SND_SOC_DAIFMT_DSP_A: - adc_reg &= ~AD1938_ADC_SERFMT_MASK; - adc_reg |= AD1938_ADC_SERFMT_AUX; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ - adc_reg &= ~AD1938_ADC_LEFT_HIGH; - adc_reg &= ~AD1938_ADC_BCLK_INV; - dac_reg &= ~AD1938_DAC_LEFT_HIGH; - dac_reg &= ~AD1938_DAC_BCLK_INV; - break; - case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */ - adc_reg |= AD1938_ADC_LEFT_HIGH; - adc_reg &= ~AD1938_ADC_BCLK_INV; - dac_reg |= AD1938_DAC_LEFT_HIGH; - dac_reg &= ~AD1938_DAC_BCLK_INV; - break; - case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */ - adc_reg &= ~AD1938_ADC_LEFT_HIGH; - adc_reg |= AD1938_ADC_BCLK_INV; - dac_reg &= ~AD1938_DAC_LEFT_HIGH; - dac_reg |= AD1938_DAC_BCLK_INV; - break; - - case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */ - adc_reg |= AD1938_ADC_LEFT_HIGH; - adc_reg |= AD1938_ADC_BCLK_INV; - dac_reg |= AD1938_DAC_LEFT_HIGH; - dac_reg |= AD1938_DAC_BCLK_INV; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ - adc_reg |= AD1938_ADC_LCR_MASTER; - adc_reg |= AD1938_ADC_BCLK_MASTER; - dac_reg |= AD1938_DAC_LCR_MASTER; - dac_reg |= AD1938_DAC_BCLK_MASTER; - break; - case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ - adc_reg |= AD1938_ADC_LCR_MASTER; - adc_reg &= ~AD1938_ADC_BCLK_MASTER; - dac_reg |= AD1938_DAC_LCR_MASTER; - dac_reg &= ~AD1938_DAC_BCLK_MASTER; - break; - case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ - adc_reg &= ~AD1938_ADC_LCR_MASTER; - adc_reg |= AD1938_ADC_BCLK_MASTER; - dac_reg &= ~AD1938_DAC_LCR_MASTER; - dac_reg |= AD1938_DAC_BCLK_MASTER; - break; - case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ - adc_reg &= ~AD1938_ADC_LCR_MASTER; - adc_reg &= ~AD1938_ADC_BCLK_MASTER; - dac_reg &= ~AD1938_DAC_LCR_MASTER; - dac_reg &= ~AD1938_DAC_BCLK_MASTER; - break; - default: - return -EINVAL; - } - - snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg); - snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg); - - return 0; -} - -static int ad1938_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - int word_len = 0, reg = 0; - - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->card->codec; - - /* bit size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; - break; - case SNDRV_PCM_FORMAT_S24_LE: - case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; - break; - } - - reg = snd_soc_read(codec, AD1938_DAC_CTRL2); - reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len; - snd_soc_write(codec, AD1938_DAC_CTRL2, reg); - - reg = snd_soc_read(codec, AD1938_ADC_CTRL1); - reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len; - snd_soc_write(codec, AD1938_ADC_CTRL1, reg); - - return 0; -} - -static int __devinit ad1938_spi_probe(struct spi_device *spi) -{ - struct snd_soc_codec *codec; - struct ad1938_priv *ad1938; - - ad1938 = kzalloc(sizeof(struct ad1938_priv), GFP_KERNEL); - if (ad1938 == NULL) - return -ENOMEM; - - codec = &ad1938->codec; - codec->control_data = spi; - codec->dev = &spi->dev; - - dev_set_drvdata(&spi->dev, ad1938); - - return ad1938_register(ad1938); -} - -static int __devexit ad1938_spi_remove(struct spi_device *spi) -{ - struct ad1938_priv *ad1938 = dev_get_drvdata(&spi->dev); - - ad1938_unregister(ad1938); - return 0; -} - -static struct spi_driver ad1938_spi_driver = { - .driver = { - .name = "ad1938", - .owner = THIS_MODULE, - }, - .probe = ad1938_spi_probe, - .remove = __devexit_p(ad1938_spi_remove), -}; - -static struct snd_soc_dai_ops ad1938_dai_ops = { - .hw_params = ad1938_hw_params, - .digital_mute = ad1938_mute, - .set_tdm_slot = ad1938_set_tdm_slot, - .set_fmt = ad1938_set_dai_fmt, -}; - -/* codec DAI instance */ -struct snd_soc_dai ad1938_dai = { - .name = "AD1938", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad1938_dai_ops, -}; -EXPORT_SYMBOL_GPL(ad1938_dai); - -static int ad1938_register(struct ad1938_priv *ad1938) -{ - int ret; - struct snd_soc_codec *codec = &ad1938->codec; - - if (ad1938_codec) { - dev_err(codec->dev, "Another ad1938 is registered\n"); - return -EINVAL; - } - - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - codec->private_data = ad1938; - codec->reg_cache = ad1938->reg_cache; - codec->reg_cache_size = AD1938_NUM_REGS; - codec->name = "AD1938"; - codec->owner = THIS_MODULE; - codec->dai = &ad1938_dai; - codec->num_dai = 1; - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - ad1938_dai.dev = codec->dev; - ad1938_codec = codec; - - memcpy(codec->reg_cache, ad1938_reg, AD1938_NUM_REGS); - - ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_SPI); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", - ret); - kfree(ad1938); - return ret; - } - - /* default setting for ad1938 */ - - /* unmute dac channels */ - snd_soc_write(codec, AD1938_DAC_CHNL_MUTE, 0x0); - /* de-emphasis: 48kHz, powedown dac */ - snd_soc_write(codec, AD1938_DAC_CTRL2, 0x1A); - /* powerdown dac, dac in tdm mode */ - snd_soc_write(codec, AD1938_DAC_CTRL0, 0x41); - /* high-pass filter enable */ - snd_soc_write(codec, AD1938_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - snd_soc_write(codec, AD1938_ADC_CTRL1, 0x43); - /* pll input: mclki/xi */ - snd_soc_write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); - snd_soc_write(codec, AD1938_PLL_CLK_CTRL1, 0x04); - - ret = snd_soc_register_codec(codec); - if (ret != 0) { - dev_err(codec->dev, "Failed to register codec: %d\n", ret); - kfree(ad1938); - return ret; - } - - ret = snd_soc_register_dai(&ad1938_dai); - if (ret != 0) { - dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - kfree(ad1938); - return ret; - } - - return 0; -} - -static void ad1938_unregister(struct ad1938_priv *ad1938) -{ - snd_soc_unregister_dai(&ad1938_dai); - snd_soc_unregister_codec(&ad1938->codec); - kfree(ad1938); - ad1938_codec = NULL; -} - -static int ad1938_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec; - int ret = 0; - - if (ad1938_codec == NULL) { - dev_err(&pdev->dev, "Codec device not registered\n"); - return -ENODEV; - } - - socdev->card->codec = ad1938_codec; - codec = ad1938_codec; - - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - dev_err(codec->dev, "failed to create pcms: %d\n", ret); - goto pcm_err; - } - - snd_soc_add_controls(codec, ad1938_snd_controls, - ARRAY_SIZE(ad1938_snd_controls)); - snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets, - ARRAY_SIZE(ad1938_dapm_widgets)); - snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - - -pcm_err: - return ret; -} - -/* power down chip */ -static int ad1938_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_ad1938 = { - .probe = ad1938_probe, - .remove = ad1938_remove, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938); - -static int __init ad1938_init(void) -{ - int ret; - - ret = spi_register_driver(&ad1938_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register ad1938 SPI driver: %d\n", - ret); - } - - return ret; -} -module_init(ad1938_init); - -static void __exit ad1938_exit(void) -{ - spi_unregister_driver(&ad1938_spi_driver); -} -module_exit(ad1938_exit); - -MODULE_DESCRIPTION("ASoC ad1938 driver"); -MODULE_AUTHOR("Barry Song "); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1938.h b/sound/soc/codecs/ad1938.h deleted file mode 100644 index fe3c48cd2d5b..000000000000 --- a/sound/soc/codecs/ad1938.h +++ /dev/null @@ -1,100 +0,0 @@ -/* - * File: sound/soc/codecs/ad1836.h - * Based on: - * Author: Barry Song <Barry.Song@analog.com> - * - * Created: May 25, 2009 - * Description: definitions for AD1938 registers - * - * Modified: - * - * 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 __AD1938_H__ -#define __AD1938_H__ - -#define AD1938_PLL_CLK_CTRL0 0 -#define AD1938_PLL_POWERDOWN 0x01 -#define AD1938_PLL_CLK_CTRL1 1 -#define AD1938_DAC_CTRL0 2 -#define AD1938_DAC_POWERDOWN 0x01 -#define AD1938_DAC_SERFMT_MASK 0xC0 -#define AD1938_DAC_SERFMT_STEREO (0 << 6) -#define AD1938_DAC_SERFMT_TDM (1 << 6) -#define AD1938_DAC_CTRL1 3 -#define AD1938_DAC_2_CHANNELS 0 -#define AD1938_DAC_4_CHANNELS 1 -#define AD1938_DAC_8_CHANNELS 2 -#define AD1938_DAC_16_CHANNELS 3 -#define AD1938_DAC_CHAN_SHFT 1 -#define AD1938_DAC_CHAN_MASK (3 << AD1938_DAC_CHAN_SHFT) -#define AD1938_DAC_LCR_MASTER (1 << 4) -#define AD1938_DAC_BCLK_MASTER (1 << 5) -#define AD1938_DAC_LEFT_HIGH (1 << 3) -#define AD1938_DAC_BCLK_INV (1 << 7) -#define AD1938_DAC_CTRL2 4 -#define AD1938_DAC_WORD_LEN_MASK 0xC -#define AD1938_DAC_MASTER_MUTE 1 -#define AD1938_DAC_CHNL_MUTE 5 -#define AD1938_DACL1_MUTE 0 -#define AD1938_DACR1_MUTE 1 -#define AD1938_DACL2_MUTE 2 -#define AD1938_DACR2_MUTE 3 -#define AD1938_DACL3_MUTE 4 -#define AD1938_DACR3_MUTE 5 -#define AD1938_DACL4_MUTE 6 -#define AD1938_DACR4_MUTE 7 -#define AD1938_DAC_L1_VOL 6 -#define AD1938_DAC_R1_VOL 7 -#define AD1938_DAC_L2_VOL 8 -#define AD1938_DAC_R2_VOL 9 -#define AD1938_DAC_L3_VOL 10 -#define AD1938_DAC_R3_VOL 11 -#define AD1938_DAC_L4_VOL 12 -#define AD1938_DAC_R4_VOL 13 -#define AD1938_ADC_CTRL0 14 -#define AD1938_ADC_POWERDOWN 0x01 -#define AD1938_ADC_HIGHPASS_FILTER 1 -#define AD1938_ADCL1_MUTE 2 -#define AD1938_ADCR1_MUTE 3 -#define AD1938_ADCL2_MUTE 4 -#define AD1938_ADCR2_MUTE 5 -#define AD1938_ADC_CTRL1 15 -#define AD1938_ADC_SERFMT_MASK 0x60 -#define AD1938_ADC_SERFMT_STEREO (0 << 5) -#define AD1938_ADC_SERFMT_TDM (1 << 2) -#define AD1938_ADC_SERFMT_AUX (2 << 5) -#define AD1938_ADC_WORD_LEN_MASK 0x3 -#define AD1938_ADC_CTRL2 16 -#define AD1938_ADC_2_CHANNELS 0 -#define AD1938_ADC_4_CHANNELS 1 -#define AD1938_ADC_8_CHANNELS 2 -#define AD1938_ADC_16_CHANNELS 3 -#define AD1938_ADC_CHAN_SHFT 4 -#define AD1938_ADC_CHAN_MASK (3 << AD1938_ADC_CHAN_SHFT) -#define AD1938_ADC_LCR_MASTER (1 << 3) -#define AD1938_ADC_BCLK_MASTER (1 << 6) -#define AD1938_ADC_LEFT_HIGH (1 << 2) -#define AD1938_ADC_BCLK_INV (1 << 1) - -#define AD1938_NUM_REGS 17 - -extern struct snd_soc_dai ad1938_dai; -extern struct snd_soc_codec_device soc_codec_dev_ad1938; -#endif diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c new file mode 100644 index 000000000000..d034464e26ff --- /dev/null +++ b/sound/soc/codecs/ad193x.c @@ -0,0 +1,558 @@ +/* + * AD193X Audio Codec driver supporting AD1936/7/8/9 + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-dapm.h> +#include "ad193x.h" + +/* codec private data */ +struct ad193x_priv { + struct snd_soc_codec codec; + u8 reg_cache[AD193X_NUM_REGS]; +}; + +/* ad193x register cache & default register settings */ +static const u8 ad193x_reg[AD193X_NUM_REGS] = { + 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, +}; + +static struct snd_soc_codec *ad193x_codec; +struct snd_soc_codec_device soc_codec_dev_ad193x; +static int ad193x_register(struct ad193x_priv *ad193x, int bus_type); +static void ad193x_unregister(struct ad193x_priv *ad193x); + +/* + * AD193X volume/mute/de-emphasis etc. controls + */ +static const char *ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; + +static const struct soc_enum ad193x_deemp_enum = + SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp); + +static const struct snd_kcontrol_new ad193x_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD193X_DAC_L1_VOL, + AD193X_DAC_R1_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC2 Volume", AD193X_DAC_L2_VOL, + AD193X_DAC_R2_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC3 Volume", AD193X_DAC_L3_VOL, + AD193X_DAC_R3_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC4 Volume", AD193X_DAC_L4_VOL, + AD193X_DAC_R4_VOL, 0, 0xFF, 1), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, + AD193X_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL2_MUTE, + AD193X_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL3_MUTE, + AD193X_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, + AD193X_DACR4_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, + AD193X_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD193X_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "PLL_PWR" }, + { "ADC", NULL, "PLL_PWR" }, + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad193x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = snd_soc_read(codec, AD193X_DAC_CTRL2); + reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg & + (~AD193X_DAC_MASTER_MUTE); + snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + + return 0; +} + +static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct snd_soc_codec *codec = dai->codec; + int dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1); + int adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2); + + dac_reg &= ~AD193X_DAC_CHAN_MASK; + adc_reg &= ~AD193X_ADC_CHAN_MASK; + + switch (slots) { + case 2: + dac_reg |= AD193X_DAC_2_CHANNELS << AD193X_DAC_CHAN_SHFT; + adc_reg |= AD193X_ADC_2_CHANNELS << AD193X_ADC_CHAN_SHFT; + break; + case 4: + dac_reg |= AD193X_DAC_4_CHANNELS << AD193X_DAC_CHAN_SHFT; + adc_reg |= AD193X_ADC_4_CHANNELS << AD193X_ADC_CHAN_SHFT; + break; + case 8: + dac_reg |= AD193X_DAC_8_CHANNELS << AD193X_DAC_CHAN_SHFT; + adc_reg |= AD193X_ADC_8_CHANNELS << AD193X_ADC_CHAN_SHFT; + break; + case 16: + dac_reg |= AD193X_DAC_16_CHANNELS << AD193X_DAC_CHAN_SHFT; + adc_reg |= AD193X_ADC_16_CHANNELS << AD193X_ADC_CHAN_SHFT; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg); + snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg); + + return 0; +} + +static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int adc_reg, dac_reg; + + adc_reg = snd_soc_read(codec, AD193X_ADC_CTRL2); + dac_reg = snd_soc_read(codec, AD193X_DAC_CTRL1); + + /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S + * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adc_reg &= ~AD193X_ADC_SERFMT_MASK; + adc_reg |= AD193X_ADC_SERFMT_TDM; + break; + case SND_SOC_DAIFMT_DSP_A: + adc_reg &= ~AD193X_ADC_SERFMT_MASK; + adc_reg |= AD193X_ADC_SERFMT_AUX; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + adc_reg &= ~AD193X_ADC_LEFT_HIGH; + adc_reg &= ~AD193X_ADC_BCLK_INV; + dac_reg &= ~AD193X_DAC_LEFT_HIGH; + dac_reg &= ~AD193X_DAC_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */ + adc_reg |= AD193X_ADC_LEFT_HIGH; + adc_reg &= ~AD193X_ADC_BCLK_INV; + dac_reg |= AD193X_DAC_LEFT_HIGH; + dac_reg &= ~AD193X_DAC_BCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */ + adc_reg &= ~AD193X_ADC_LEFT_HIGH; + adc_reg |= AD193X_ADC_BCLK_INV; + dac_reg &= ~AD193X_DAC_LEFT_HIGH; + dac_reg |= AD193X_DAC_BCLK_INV; + break; + + case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */ + adc_reg |= AD193X_ADC_LEFT_HIGH; + adc_reg |= AD193X_ADC_BCLK_INV; + dac_reg |= AD193X_DAC_LEFT_HIGH; + dac_reg |= AD193X_DAC_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ + adc_reg |= AD193X_ADC_LCR_MASTER; + adc_reg |= AD193X_ADC_BCLK_MASTER; + dac_reg |= AD193X_DAC_LCR_MASTER; + dac_reg |= AD193X_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ + adc_reg |= AD193X_ADC_LCR_MASTER; + adc_reg &= ~AD193X_ADC_BCLK_MASTER; + dac_reg |= AD193X_DAC_LCR_MASTER; + dac_reg &= ~AD193X_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + adc_reg &= ~AD193X_ADC_LCR_MASTER; + adc_reg |= AD193X_ADC_BCLK_MASTER; + dac_reg &= ~AD193X_DAC_LCR_MASTER; + dac_reg |= AD193X_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ + adc_reg &= ~AD193X_ADC_LCR_MASTER; + adc_reg &= ~AD193X_ADC_BCLK_MASTER; + dac_reg &= ~AD193X_DAC_LCR_MASTER; + dac_reg &= ~AD193X_DAC_BCLK_MASTER; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AD193X_ADC_CTRL2, adc_reg); + snd_soc_write(codec, AD193X_DAC_CTRL1, dac_reg); + + return 0; +} + +static int ad193x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0, reg = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + reg = snd_soc_read(codec, AD193X_DAC_CTRL2); + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + snd_soc_write(codec, AD193X_DAC_CTRL2, reg); + + reg = snd_soc_read(codec, AD193X_ADC_CTRL1); + reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len; + snd_soc_write(codec, AD193X_ADC_CTRL1, reg); + + return 0; +} + +static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) +{ + struct snd_soc_codec *codec; + struct ad193x_priv *ad193x; + + ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); + if (ad193x == NULL) + return -ENOMEM; + + codec = &ad193x->codec; + codec->control_data = ctrl_data; + codec->dev = dev; + + dev_set_drvdata(dev, ad193x); + + return ad193x_register(ad193x, bus_type); +} + +static int ad193x_bus_remove(struct device *dev) +{ + struct ad193x_priv *ad193x = dev_get_drvdata(dev); + + ad193x_unregister(ad193x); + return 0; +} + +static struct snd_soc_dai_ops ad193x_dai_ops = { + .hw_params = ad193x_hw_params, + .digital_mute = ad193x_mute, + .set_tdm_slot = ad193x_set_tdm_slot, + .set_fmt = ad193x_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad193x_dai = { + .name = "AD193X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad193x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad193x_dai); + +static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) +{ + int ret; + struct snd_soc_codec *codec = &ad193x->codec; + + if (ad193x_codec) { + dev_err(codec->dev, "Another ad193x is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + codec->private_data = ad193x; + codec->reg_cache = ad193x->reg_cache; + codec->reg_cache_size = AD193X_NUM_REGS; + codec->name = "AD193X"; + codec->owner = THIS_MODULE; + codec->dai = &ad193x_dai; + codec->num_dai = 1; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad193x_dai.dev = codec->dev; + ad193x_codec = codec; + + memcpy(codec->reg_cache, ad193x_reg, AD193X_NUM_REGS); + + if (bus_type == SND_SOC_I2C) + ret = snd_soc_codec_set_cache_io(codec, 8, 8, bus_type); + else + ret = snd_soc_codec_set_cache_io(codec, 16, 8, bus_type); + if (ret < 0) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", + ret); + kfree(ad193x); + return ret; + } + + /* default setting for ad193x */ + + /* unmute dac channels */ + snd_soc_write(codec, AD193X_DAC_CHNL_MUTE, 0x0); + /* de-emphasis: 48kHz, powedown dac */ + snd_soc_write(codec, AD193X_DAC_CTRL2, 0x1A); + /* powerdown dac, dac in tdm mode */ + snd_soc_write(codec, AD193X_DAC_CTRL0, 0x41); + /* high-pass filter enable */ + snd_soc_write(codec, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + snd_soc_write(codec, AD193X_ADC_CTRL1, 0x43); + /* pll input: mclki/xi */ + snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ + snd_soc_write(codec, AD193X_PLL_CLK_CTRL1, 0x04); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad193x); + return ret; + } + + ret = snd_soc_register_dai(&ad193x_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + kfree(ad193x); + return ret; + } + + return 0; +} + +static void ad193x_unregister(struct ad193x_priv *ad193x) +{ + snd_soc_unregister_dai(&ad193x_dai); + snd_soc_unregister_codec(&ad193x->codec); + kfree(ad193x); + ad193x_codec = NULL; +} + +static int ad193x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ad193x_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ad193x_codec; + codec = ad193x_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ad193x_snd_controls, + ARRAY_SIZE(ad193x_snd_controls)); + snd_soc_dapm_new_controls(codec, ad193x_dapm_widgets, + ARRAY_SIZE(ad193x_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + +pcm_err: + return ret; +} + +/* power down chip */ +static int ad193x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad193x = { + .probe = ad193x_probe, + .remove = ad193x_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad193x); + +#if defined(CONFIG_SPI_MASTER) +static int __devinit ad193x_spi_probe(struct spi_device *spi) +{ + return ad193x_bus_probe(&spi->dev, spi, SND_SOC_SPI); +} + +static int __devexit ad193x_spi_remove(struct spi_device *spi) +{ + return ad193x_bus_remove(&spi->dev); +} + +static struct spi_driver ad193x_spi_driver = { + .driver = { + .name = "ad193x", + .owner = THIS_MODULE, + }, + .probe = ad193x_spi_probe, + .remove = __devexit_p(ad193x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id ad193x_id[] = { + { "ad1936", 0 }, + { "ad1937", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ad193x_id); + +static int __devinit ad193x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return ad193x_bus_probe(&client->dev, client, SND_SOC_I2C); +} + +static int __devexit ad193x_i2c_remove(struct i2c_client *client) +{ + return ad193x_bus_remove(&client->dev); +} + +static struct i2c_driver ad193x_i2c_driver = { + .driver = { + .name = "ad193x", + }, + .probe = ad193x_i2c_probe, + .remove = __devexit_p(ad193x_i2c_remove), + .id_table = ad193x_id, +}; +#endif + +static int __init ad193x_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ad193x_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n", + ret); + } +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&ad193x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(ad193x_modinit); + +static void __exit ad193x_modexit(void) +{ +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&ad193x_spi_driver); +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ad193x_i2c_driver); +#endif +} +module_exit(ad193x_modexit); + +MODULE_DESCRIPTION("ASoC ad193x driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h new file mode 100644 index 000000000000..a03c880d52f9 --- /dev/null +++ b/sound/soc/codecs/ad193x.h @@ -0,0 +1,81 @@ +/* + * AD193X Audio Codec driver + * + * Copyright 2010 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __AD193X_H__ +#define __AD193X_H__ + +#define AD193X_PLL_CLK_CTRL0 0x800 +#define AD193X_PLL_POWERDOWN 0x01 +#define AD193X_PLL_CLK_CTRL1 0x801 +#define AD193X_DAC_CTRL0 0x802 +#define AD193X_DAC_POWERDOWN 0x01 +#define AD193X_DAC_SERFMT_MASK 0xC0 +#define AD193X_DAC_SERFMT_STEREO (0 << 6) +#define AD193X_DAC_SERFMT_TDM (1 << 6) +#define AD193X_DAC_CTRL1 0x803 +#define AD193X_DAC_2_CHANNELS 0 +#define AD193X_DAC_4_CHANNELS 1 +#define AD193X_DAC_8_CHANNELS 2 +#define AD193X_DAC_16_CHANNELS 3 +#define AD193X_DAC_CHAN_SHFT 1 +#define AD193X_DAC_CHAN_MASK (3 << AD193X_DAC_CHAN_SHFT) +#define AD193X_DAC_LCR_MASTER (1 << 4) +#define AD193X_DAC_BCLK_MASTER (1 << 5) +#define AD193X_DAC_LEFT_HIGH (1 << 3) +#define AD193X_DAC_BCLK_INV (1 << 7) +#define AD193X_DAC_CTRL2 0x804 +#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_MASTER_MUTE 1 +#define AD193X_DAC_CHNL_MUTE 0x805 +#define AD193X_DACL1_MUTE 0 +#define AD193X_DACR1_MUTE 1 +#define AD193X_DACL2_MUTE 2 +#define AD193X_DACR2_MUTE 3 +#define AD193X_DACL3_MUTE 4 +#define AD193X_DACR3_MUTE 5 +#define AD193X_DACL4_MUTE 6 +#define AD193X_DACR4_MUTE 7 +#define AD193X_DAC_L1_VOL 0x806 +#define AD193X_DAC_R1_VOL 0x807 +#define AD193X_DAC_L2_VOL 0x808 +#define AD193X_DAC_R2_VOL 0x809 +#define AD193X_DAC_L3_VOL 0x80a +#define AD193X_DAC_R3_VOL 0x80b +#define AD193X_DAC_L4_VOL 0x80c +#define AD193X_DAC_R4_VOL 0x80d +#define AD193X_ADC_CTRL0 0x80e +#define AD193X_ADC_POWERDOWN 0x01 +#define AD193X_ADC_HIGHPASS_FILTER 1 +#define AD193X_ADCL1_MUTE 2 +#define AD193X_ADCR1_MUTE 3 +#define AD193X_ADCL2_MUTE 4 +#define AD193X_ADCR2_MUTE 5 +#define AD193X_ADC_CTRL1 0x80f +#define AD193X_ADC_SERFMT_MASK 0x60 +#define AD193X_ADC_SERFMT_STEREO (0 << 5) +#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_AUX (2 << 5) +#define AD193X_ADC_WORD_LEN_MASK 0x3 +#define AD193X_ADC_CTRL2 0x810 +#define AD193X_ADC_2_CHANNELS 0 +#define AD193X_ADC_4_CHANNELS 1 +#define AD193X_ADC_8_CHANNELS 2 +#define AD193X_ADC_16_CHANNELS 3 +#define AD193X_ADC_CHAN_SHFT 4 +#define AD193X_ADC_CHAN_MASK (3 << AD193X_ADC_CHAN_SHFT) +#define AD193X_ADC_LCR_MASTER (1 << 3) +#define AD193X_ADC_BCLK_MASTER (1 << 6) +#define AD193X_ADC_LEFT_HIGH (1 << 2) +#define AD193X_ADC_BCLK_INV (1 << 1) + +#define AD193X_NUM_REGS 17 + +extern struct snd_soc_dai ad193x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad193x; + +#endif diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ef16bbc8c83..de1809dc8d91 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -80,12 +80,39 @@ #define AK4642_CACHEREGNUM 0x25 +/* PW_MGMT2 */ +#define HPMTN (1 << 6) +#define PMHPL (1 << 5) +#define PMHPR (1 << 4) +#define MS (1 << 3) /* master/slave select */ +#define MCKO (1 << 1) +#define PMPLL (1 << 0) + +#define PMHP_MASK (PMHPL | PMHPR) +#define PMHP PMHP_MASK + +/* MD_CTL1 */ +#define PLL3 (1 << 7) +#define PLL2 (1 << 6) +#define PLL1 (1 << 5) +#define PLL0 (1 << 4) +#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) + +#define BCKO_MASK (1 << 3) +#define BCKO_64 BCKO_MASK + +/* MD_CTL2 */ +#define FS0 (1 << 0) +#define FS1 (1 << 1) +#define FS2 (1 << 2) +#define FS3 (1 << 5) +#define FS_MASK (FS0 | FS1 | FS2 | FS3) + struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; - unsigned int sysclk; }; static struct snd_soc_codec *ak4642_codec; @@ -176,17 +203,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Sampling Frequency: 44.1kHz - * Digital Volume: −8dB + * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. - * - * Example code use 0x39, 0x79 value for 0x01 address, - * But we need MCKO (0x02) bit now */ - ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x0f, 0x09); ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); @@ -194,15 +216,14 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0a, 0x28); ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); - ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ - ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); + snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input * * PLL Master Mode * Audio I/F Format:MSB justified (ADC & DAC) - * Sampling Frequency:44.1kHz * Pre MIC AMP:+20dB * MIC Power On * ALC setting:Refer to Table 35 @@ -211,7 +232,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x02, 0x05); ak4642_write(codec, 0x06, 0x3c); ak4642_write(codec, 0x08, 0xe1); @@ -232,8 +252,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { /* stop headphone output */ - ak4642_write(codec, 0x01, 0x3b); - ak4642_write(codec, 0x01, 0x0b); + snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); + snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x0e, 0x11); ak4642_write(codec, 0x0f, 0x08); @@ -249,9 +269,111 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct ak4642_priv *ak4642 = codec->private_data; + u8 pll; + + switch (freq) { + case 11289600: + pll = PLL2; + break; + case 12288000: + pll = PLL2 | PLL0; + break; + case 12000000: + pll = PLL2 | PLL1; + break; + case 24000000: + pll = PLL2 | PLL1 | PLL0; + break; + case 13500000: + pll = PLL3 | PLL2; + break; + case 27000000: + pll = PLL3 | PLL2 | PLL0; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll); + + return 0; +} + +static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 data; + u8 bcko; + + data = MCKO | PMPLL; /* use MCKO */ + bcko = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + data |= MS; + bcko = BCKO_64; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + + return 0; +} + +static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 rate; + + switch (params_rate(params)) { + case 7350: + rate = FS2; + break; + case 8000: + rate = 0; + break; + case 11025: + rate = FS2 | FS0; + break; + case 12000: + rate = FS0; + break; + case 14700: + rate = FS2 | FS1; + break; + case 16000: + rate = FS1; + break; + case 22050: + rate = FS2 | FS1 | FS0; + break; + case 24000: + rate = FS1 | FS0; + break; + case 29400: + rate = FS3 | FS2 | FS1; + break; + case 32000: + rate = FS3 | FS1; + break; + case 44100: + rate = FS3 | FS2 | FS1 | FS0; + break; + case 48000: + rate = FS3 | FS1 | FS0; + break; + default: + return -EINVAL; + break; + } + snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); - ak4642->sysclk = freq; return 0; } @@ -259,6 +381,8 @@ static struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, .set_sysclk = ak4642_dai_set_sysclk, + .set_fmt = ak4642_dai_set_fmt, + .hw_params = ak4642_dai_hw_params, }; struct snd_soc_dai ak4642_dai = { @@ -276,6 +400,7 @@ struct snd_soc_dai ak4642_dai = { .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .ops = &ak4642_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(ak4642_dai); @@ -337,26 +462,6 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } - /* - * clock setting - * - * Audio I/F Format: MSB justified (ADC & DAC) - * BICK frequency at Master Mode: 64fs - * Input Master Clock Select at PLL Mode: 11.2896MHz - * MCKO: Enable - * Sampling Frequency: 44.1kHz - * - * This operation came from example code of - * "ASAHI KASEI AK4642" (japanese) manual p89. - * - * please fix-me - */ - ak4642_write(codec, 0x01, 0x08); - ak4642_write(codec, 0x04, 0x4a); - ak4642_write(codec, 0x05, 0x27); - ak4642_write(codec, 0x00, 0x40); - ak4642_write(codec, 0x01, 0x0b); - return ret; reg_cache_err: diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 000000000000..513297441369 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,298 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_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 davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(davinci_vc->dev, "failed to register codec\n"); + goto fail1; + } + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail2; + } + return 0; + +fail2: + snd_soc_unregister_codec(codec); + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + snd_soc_unregister_codec(&codec); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 000000000000..845b1968ef9c --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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 _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf2975a7294a..35b881638658 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -55,8 +55,14 @@ #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV1 0x29 +#define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_A_HID_UNLOCK 0x8A +#define DA7210_A_TEST_UNLOCK 0x8B +#define DA7210_A_PLL1 0x90 +#define DA7210_A_CP_MODE 0xA7 /* STARTUP1 bit fields */ #define DA7210_SC_MST_EN (1 << 0) @@ -74,15 +80,14 @@ /* INMIX_R bit fields */ #define DA7210_IN_R_EN (1 << 7) -/* ADC_HPF bit fields */ -#define DA7210_ADC_VOICE_EN (1 << 7) - /* ADC bit fields */ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) -/* DAC_HPF fields */ -#define DA7210_DAC_VOICE_EN (1 << 7) +/* DAC/ADC HPF fields */ +#define DA7210_VOICE_F0_MASK (0x7 << 4) +#define DA7210_VOICE_F0_25 (1 << 4) +#define DA7210_VOICE_EN (1 << 7) /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) @@ -123,7 +128,19 @@ #define DA7210_PLL_BYP (1 << 6) /* PLL bit fields */ -#define DA7210_PLL_FS_48000 (11 << 0) +#define DA7210_PLL_FS_MASK (0xF << 0) +#define DA7210_PLL_FS_8000 (0x1 << 0) +#define DA7210_PLL_FS_11025 (0x2 << 0) +#define DA7210_PLL_FS_12000 (0x3 << 0) +#define DA7210_PLL_FS_16000 (0x5 << 0) +#define DA7210_PLL_FS_22050 (0x6 << 0) +#define DA7210_PLL_FS_24000 (0x7 << 0) +#define DA7210_PLL_FS_32000 (0x9 << 0) +#define DA7210_PLL_FS_44100 (0xA << 0) +#define DA7210_PLL_FS_48000 (0xB << 0) +#define DA7210_PLL_FS_88200 (0xE << 0) +#define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_PLL_EN (0x1 << 7) #define DA7210_VERSION "0.0.1" @@ -241,7 +258,8 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; u32 dai_cfg1; - u32 reg, mask; + u32 hpf_reg, hpf_mask, hpf_value; + u32 fs, bypass; /* set DAI source to Left and Right ADC */ da7210_write(codec, DA7210_DAI_SRC_SEL, @@ -265,25 +283,84 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - /* FIXME - * - * It support 48K only now - */ + hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? + DA7210_DAC_HPF : DA7210_ADC_HPF; + switch (params_rate(params)) { + case 8000: + fs = DA7210_PLL_FS_8000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 11025: + fs = DA7210_PLL_FS_11025; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = 0; + break; + case 12000: + fs = DA7210_PLL_FS_12000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 16000: + fs = DA7210_PLL_FS_16000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 22050: + fs = DA7210_PLL_FS_22050; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; + break; + case 32000: + fs = DA7210_PLL_FS_32000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 44100: + fs = DA7210_PLL_FS_44100; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; + break; case 48000: - if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { - reg = DA7210_DAC_HPF; - mask = DA7210_DAC_VOICE_EN; - } else { - reg = DA7210_ADC_HPF; - mask = DA7210_ADC_VOICE_EN; - } + fs = DA7210_PLL_FS_48000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 88200: + fs = DA7210_PLL_FS_88200; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; + break; + case 96000: + fs = DA7210_PLL_FS_96000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = DA7210_PLL_BYP; break; default: return -EINVAL; } - snd_soc_update_bits(codec, reg, mask, 0); + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + + snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, + DA7210_SC_MST_EN, DA7210_SC_MST_EN); return 0; } @@ -361,6 +438,7 @@ struct snd_soc_dai da7210_dai = { .formats = DA7210_FORMATS, }, .ops = &da7210_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(da7210_dai); @@ -415,9 +493,23 @@ static int da7210_init(struct da7210_priv *da7210) /* FIXME * * This driver use fixed value here + * And below settings expects MCLK = 12.288MHz + * + * When you select different MCLK, please check... + * DA7210_PLL_DIV1 val + * DA7210_PLL_DIV2 val + * DA7210_PLL_DIV3 val + * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx */ /* + * make sure that DA7210 use bypass mode before start up + */ + da7210_write(codec, DA7210_STARTUP1, 0); + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* * ADC settings */ @@ -453,9 +545,28 @@ static int da7210_init(struct da7210_priv *da7210) /* Diable PLL and bypass it */ da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* Bypass PLL and set MCLK freq rang to 10-20MHz */ - da7210_write(codec, DA7210_PLL_DIV3, + /* + * If 48kHz sound came, it use bypass mode, + * and when it is 44.1kHz, it use PLL. + * + * This time, this driver sets PLL always ON + * and controls bypass/PLL mode by switching + * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. + * see da7210_hw_params + */ + da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + da7210_write(codec, DA7210_PLL_DIV2, 0x99); + da7210_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* As suggested by Dialog */ + da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + da7210_write(codec, DA7210_A_PLL1, 0x01); + da7210_write(codec, DA7210_A_CP_MODE, 0x7C); + da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d2ff1cde6883..942f5dc30801 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -139,6 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0), SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), @@ -604,8 +605,7 @@ static int ssm2602_init(struct snd_soc_device *socdev) reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); /*select Line in as default input*/ - ssm2602_write(codec, SSM2602_APANA, - APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC | + ssm2602_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f9f367d29a90..ee2e1e36d995 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -93,6 +93,8 @@ struct tlv320dac33_priv { unsigned int nsample; /* burst read amount from host */ u8 burst_bclkdiv; /* BCLK divider value in burst mode */ + int keep_bclk; /* Keep the BCLK continuously running + * in FIFO modes */ enum dac33_state state; }; @@ -310,7 +312,8 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) if (power) reg |= DAC33_PDNALLB; else - reg &= ~DAC33_PDNALLB; + reg &= ~(DAC33_PDNALLB | DAC33_OSCPDNB | + DAC33_DACRPDNB | DAC33_DACLPDNB); dac33_write(codec, DAC33_PWR_CTRL, reg); } @@ -634,26 +637,6 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev) return IRQ_HANDLED; } -static void dac33_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->card->codec; - struct tlv320dac33_priv *dac33 = codec->private_data; - unsigned int pwr_ctrl; - - /* Stop pending workqueue */ - if (dac33->fifo_mode) - cancel_work_sync(&dac33->work); - - mutex_lock(&dac33->mutex); - pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); - pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); - dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); - mutex_unlock(&dac33->mutex); -} - static void dac33_oscwait(struct snd_soc_codec *codec) { int timeout = 20; @@ -751,6 +734,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) } mutex_lock(&dac33->mutex); + dac33_soft_power(codec, 0); dac33_soft_power(codec, 1); reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); @@ -778,7 +762,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) if (dac33->fifo_mode) { /* Generic for all FIFO modes */ /* 50-51 : ASRC Control registers */ - dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCLKDIV(1)); dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ /* Write registers 0x34 and 0x35 (MSB, LSB) */ @@ -821,7 +805,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) */ fifoctrl_a &= ~DAC33_FBYPAS; fifoctrl_a &= ~DAC33_FAUTO; - aictrl_b &= ~DAC33_BCLKON; + if (dac33->keep_bclk) + aictrl_b |= DAC33_BCLKON; + else + aictrl_b &= ~DAC33_BCLKON; break; case DAC33_FIFO_MODE7: /* @@ -832,7 +819,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) */ fifoctrl_a &= ~DAC33_FBYPAS; fifoctrl_a |= DAC33_FAUTO; - aictrl_b &= ~DAC33_BCLKON; + if (dac33->keep_bclk) + aictrl_b |= DAC33_BCLKON; + else + aictrl_b &= ~DAC33_BCLKON; break; default: /* @@ -1038,11 +1028,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_DSP_A: aictrl_a |= DAC33_AFMT_DSP; aictrl_b &= ~DAC33_DATA_DELAY_MASK; - aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ - break; - case SND_SOC_DAIFMT_DSP_B: - aictrl_a |= DAC33_AFMT_DSP; - aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + aictrl_b |= DAC33_DATA_DELAY(0); break; case SND_SOC_DAIFMT_RIGHT_J: aictrl_a |= DAC33_AFMT_RIGHT_J; @@ -1066,7 +1052,7 @@ static void dac33_init_chip(struct snd_soc_codec *codec) { /* 44-46: DAC Control Registers */ /* A : DAC sample rate Fsref/1.5 */ - dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); /* B : DAC src=normal, not muted */ dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | DAC33_DACSRCL_LEFT); @@ -1185,7 +1171,6 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); #define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE static struct snd_soc_dai_ops dac33_dai_ops = { - .shutdown = dac33_shutdown, .hw_params = dac33_hw_params, .prepare = dac33_pcm_prepare, .trigger = dac33_pcm_trigger, @@ -1253,6 +1238,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, dac33->power_gpio = pdata->power_gpio; dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; /* Disable FIFO use by default */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af20052..bf59b8a4d1d7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -135,9 +135,11 @@ struct twl4030_priv { unsigned int sysclk; - /* Headset output state handling */ - unsigned int hsl_enabled; - unsigned int hsr_enabled; + /* Output (with associated amp) states */ + u8 hsl_enabled, hsr_enabled; + u8 earpiece_enabled; + u8 predrivel_enabled, predriver_enabled; + u8 carkitl_enabled, carkitr_enabled; }; /* @@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct twl4030_priv *twl4030 = codec->private_data; + int write_to_reg = 0; + twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, - reg); - else - return 0; + if (likely(reg < TWL4030_REG_SW_SHADOW)) { + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; + } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + } + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) @@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w, * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the * amplifier associated with the output pin) - * On mute bypass the reg_cache and mute the volume - * On unmute: restore the register content + * On mute bypass the reg_cache and write 0 to the register + * On unmute: restore the register content from the reg_cache * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - u8 reg_val; \ + struct twl4030_priv *twl4030 = w->codec->private_data; \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ + twl4030->pin_name##_enabled = 1; \ twl4030_write(w->codec, reg, \ twl4030_read_reg_cache(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ - reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - reg_val & (~mask), \ - reg); \ + twl4030->pin_name##_enabled = 0; \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + 0, reg); \ break; \ } \ return 0; \ @@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + /* Actually write to the register */ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100755 index 000000000000..0e4dce7de052 --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1227 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729@ti.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/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x60, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x4A, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ + 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x00, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); +} + +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + + /* sync registers updated during power-down sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 88200, + 96000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 96000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6040_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->card->codec; + struct twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6040_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->card->codec; + struct twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +#ifdef CONFIG_PM +static int twl6040_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} +#else +#define twl6040_suspend NULL +#define twl6040_resume NULL +#endif + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_add_widgets(codec); + + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 000000000000..c472070a1da2 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 codec driver + * + * Author: Misael Lopez Cruz <x0052729@ti.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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index df2c6d9617fb..8ac92f5e4f91 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -54,6 +54,7 @@ struct wm8350_output { struct wm8350_jack_data { struct snd_soc_jack *jack; int report; + int short_report; }; struct wm8350_data { @@ -62,6 +63,7 @@ struct wm8350_data { struct wm8350_output out2; struct wm8350_jack_data hpl; struct wm8350_jack_data hpr; + struct wm8350_jack_data mic; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; int fll_freq_out; int fll_freq_in; @@ -1391,7 +1393,8 @@ static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) * @jack: jack to report detection events on * @report: value to report * - * Enables the headphone jack detection of the WM8350. + * Enables the headphone jack detection of the WM8350. If no report + * is specified then detection is disabled. */ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, struct snd_soc_jack *jack, int report) @@ -1420,8 +1423,12 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, return -EINVAL; } - wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); - wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); + if (report) { + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); + } else { + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, ena); + } /* Sync status */ wm8350_hp_jack_handler(irq + wm8350->irq_base, priv); @@ -1430,6 +1437,60 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, } EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); +static irqreturn_t wm8350_mic_handler(int irq, void *data) +{ + struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->codec.control_data; + u16 reg; + int report = 0; + + reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS); + if (reg & WM8350_JACK_MICSCD_LVL) + report |= priv->mic.short_report; + if (reg & WM8350_JACK_MICSD_LVL) + report |= priv->mic.report; + + snd_soc_jack_report(priv->mic.jack, report, + priv->mic.report | priv->mic.short_report); + + return IRQ_HANDLED; +} + +/** + * wm8350_mic_jack_detect - Enable microphone jack detection. + * + * @codec: WM8350 codec + * @jack: jack to report detection events on + * @detect_report: value to report when presence detected + * @short_report: value to report when microphone short detected + * + * Enables the microphone jack detection of the WM8350. If both reports + * are specified as zero then detection is disabled. + */ +int wm8350_mic_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, + int detect_report, int short_report) +{ + struct wm8350_data *priv = codec->private_data; + struct wm8350 *wm8350 = codec->control_data; + + priv->mic.jack = jack; + priv->mic.report = detect_report; + priv->mic.short_report = short_report; + + if (detect_report || short_report) { + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1, + WM8350_MIC_DET_ENA); + } else { + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_1, + WM8350_MIC_DET_ENA); + } + + return 0; +} +EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect); + static struct snd_soc_codec *wm8350_codec; static int wm8350_probe(struct platform_device *pdev) @@ -1493,6 +1554,10 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hp_jack_handler, 0, "Right jack detect", priv); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + wm8350_mic_handler, 0, "Microphone short", priv); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + wm8350_mic_handler, 0, "Microphone detect", priv); ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -1521,11 +1586,14 @@ static int wm8350_remove(struct platform_device *pdev) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICD, priv); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); priv->hpl.jack = NULL; priv->hpr.jack = NULL; + priv->mic.jack = NULL; /* cancel any work waiting to be queued. */ ret = cancel_delayed_work(&codec->delayed_work); diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index d088eb4b88bb..9ed0467c71db 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -25,5 +25,8 @@ enum wm8350_jack { int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, struct snd_soc_jack *jack, int report); +int wm8350_mic_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, + int detect_report, int short_report); #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 475c67ac7818..ee084083a49d 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,8 +29,6 @@ #include "wm8750.h" -#define WM8750_VERSION "0.12" - /* codec private data */ struct wm8750_priv { unsigned int sysclk; @@ -613,10 +611,16 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_PREPARE: - /* set vmid to 5k for quick power up */ - snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Set VMID to 5k */ + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); + + /* ...and ramp */ + msleep(1000); + } + /* mute dac and set vmid to 500k, enable VREF */ snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; @@ -660,13 +664,6 @@ struct snd_soc_dai wm8750_dai = { }; EXPORT_SYMBOL_GPL(wm8750_dai); -static void wm8750_work(struct work_struct *work) -{ - struct snd_soc_codec *codec = - container_of(work, struct snd_soc_codec, delayed_work.work); - wm8750_set_bias_level(codec, codec->bias_level); -} - static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); @@ -695,14 +692,6 @@ static int wm8750_resume(struct platform_device *pdev) wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* charge wm8750 caps */ - if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { - wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->bias_level = SND_SOC_BIAS_ON; - schedule_delayed_work(&codec->delayed_work, - msecs_to_jiffies(1000)); - } - return 0; } @@ -746,9 +735,7 @@ static int wm8750_init(struct snd_soc_device *socdev, } /* charge output caps */ - wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->bias_level = SND_SOC_BIAS_STANDBY; - schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); + wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set the update bits */ reg = snd_soc_read(codec, WM8750_LDAC); @@ -912,7 +899,6 @@ static int wm8750_probe(struct platform_device *pdev) struct wm8750_priv *wm8750; int ret; - pr_info("WM8750 Audio Codec %s", WM8750_VERSION); codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; @@ -929,7 +915,6 @@ static int wm8750_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); wm8750_socdev = socdev; - INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); ret = -ENODEV; @@ -953,25 +938,6 @@ static int wm8750_probe(struct platform_device *pdev) return ret; } -/* - * This function forces any delayed work to be queued and run. - */ -static int run_delayed_work(struct delayed_work *dwork) -{ - int ret; - - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(dwork); - - /* if there was any work waiting then we run it now and - * wait for it's completion */ - if (ret) { - schedule_delayed_work(dwork, 0); - flush_scheduled_work(); - } - return ret; -} - /* power down chip */ static int wm8750_remove(struct platform_device *pdev) { @@ -980,7 +946,6 @@ static int wm8750_remove(struct platform_device *pdev) if (codec->control_data) wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3595bd57c4eb..134b17532f23 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -11,25 +11,26 @@ * * TODO: * - TDM mode configuration. - * - Mic detect. * - Digital microphone support. - * - Interrupt support (mic detect and sequencer). */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/completion.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/tlv.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/wm8903.h> #include "wm8903.h" @@ -221,6 +222,14 @@ struct wm8903_priv { int playback_active; int capture_active; + struct completion wseq; + + struct snd_soc_jack *mic_jack; + int mic_det; + int mic_short; + int mic_last_report; + int mic_delay; + struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; }; @@ -243,13 +252,14 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) { u16 reg[5]; struct i2c_client *i2c = codec->control_data; + struct wm8903_priv *wm8903 = codec->private_data; BUG_ON(start > 48); - /* Enable the sequencer */ + /* Enable the sequencer if it's not already on */ reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0); - reg[0] |= WM8903_WSEQ_ENA; - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, + reg[0] | WM8903_WSEQ_ENA); dev_dbg(&i2c->dev, "Starting sequence at %d\n", start); @@ -257,20 +267,19 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) start | WM8903_WSEQ_START); /* Wait for it to complete. If we have the interrupt wired up then - * we could block waiting for an interrupt, though polling may still - * be desirable for diagnostic purposes. + * that will break us out of the poll early. */ do { - msleep(10); + wait_for_completion_timeout(&wm8903->wseq, + msecs_to_jiffies(10)); reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4); } while (reg[4] & WM8903_WSEQ_BUSY); dev_dbg(&i2c->dev, "Sequence complete\n"); - /* Disable the sequencer again */ - snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, - reg[0] & ~WM8903_WSEQ_ENA); + /* Disable the sequencer again if we enabled it */ + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); return 0; } @@ -1435,6 +1444,116 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, return 0; } +/** + * wm8903_mic_detect - Enable microphone detection via the WM8903 IRQ + * + * @codec: WM8903 codec + * @jack: jack to report detection events on + * @det: value to report for presence detection + * @shrt: value to report for short detection + * + * Enable microphone detection via IRQ on the WM8903. If GPIOs are + * being used to bring out signals to the processor then only platform + * data configuration is needed for WM8903 and processor GPIOs should + * be configured using snd_soc_jack_add_gpios() instead. + * + * The current threasholds for detection should be configured using + * micdet_cfg in the platform data. Using this function will force on + * the microphone bias for the device. + */ +int wm8903_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, + int det, int shrt) +{ + struct wm8903_priv *wm8903 = codec->private_data; + int irq_mask = WM8903_MICDET_EINT | WM8903_MICSHRT_EINT; + + dev_dbg(codec->dev, "Enabling microphone detection: %x %x\n", + det, shrt); + + /* Store the configuration */ + wm8903->mic_jack = jack; + wm8903->mic_det = det; + wm8903->mic_short = shrt; + + /* Enable interrupts we've got a report configured for */ + if (det) + irq_mask &= ~WM8903_MICDET_EINT; + if (shrt) + irq_mask &= ~WM8903_MICSHRT_EINT; + + snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_MICDET_EINT | WM8903_MICSHRT_EINT, + irq_mask); + + if (det && shrt) { + /* Enable mic detection, this may not have been set through + * platform data (eg, if the defaults are OK). */ + snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0, + WM8903_MICDET_ENA, WM8903_MICDET_ENA); + } else { + snd_soc_update_bits(codec, WM8903_MIC_BIAS_CONTROL_0, + WM8903_MICDET_ENA, 0); + } + + return 0; +} +EXPORT_SYMBOL_GPL(wm8903_mic_detect); + +static irqreturn_t wm8903_irq(int irq, void *data) +{ + struct wm8903_priv *wm8903 = data; + struct snd_soc_codec *codec = &wm8903->codec; + int mic_report; + int int_pol; + int int_val = 0; + int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK); + + int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; + + if (int_val & WM8903_WSEQ_BUSY_EINT) { + dev_dbg(codec->dev, "Write sequencer done\n"); + complete(&wm8903->wseq); + } + + /* + * The rest is microphone jack detection. We need to manually + * invert the polarity of the interrupt after each event - to + * simplify the code keep track of the last state we reported + * and just invert the relevant bits in both the report and + * the polarity register. + */ + mic_report = wm8903->mic_last_report; + int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1); + + if (int_val & WM8903_MICSHRT_EINT) { + dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol); + + mic_report ^= wm8903->mic_short; + int_pol ^= WM8903_MICSHRT_INV; + } + + if (int_val & WM8903_MICDET_EINT) { + dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol); + + mic_report ^= wm8903->mic_det; + int_pol ^= WM8903_MICDET_INV; + + msleep(wm8903->mic_delay); + } + + snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1, + WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); + + snd_soc_jack_report(wm8903->mic_jack, mic_report, + wm8903->mic_short | wm8903->mic_det); + + wm8903->mic_last_report = mic_report; + + return IRQ_HANDLED; +} + #define WM8903_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ SNDRV_PCM_RATE_11025 | \ SNDRV_PCM_RATE_16000 | \ @@ -1529,9 +1648,11 @@ static struct snd_soc_codec *wm8903_codec; static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; struct snd_soc_codec *codec; - int ret; + int ret, i; + int trigger, irq_pol; u16 val; wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); @@ -1555,6 +1676,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, codec->reg_cache = &wm8903->reg_cache[0]; codec->private_data = wm8903; codec->volatile_register = wm8903_volatile_register; + init_completion(&wm8903->wseq); i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1578,6 +1700,53 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_reset(codec); + /* Set up GPIOs and microphone detection */ + if (pdata) { + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if (!pdata->gpio_cfg[i]) + continue; + + snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0xffff); + } + + snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + wm8903->mic_delay = pdata->micdet_delay; + } + + if (i2c->irq) { + if (pdata && pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8903_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL, + WM8903_IRQ_POL, irq_pol); + + ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq, + trigger | IRQF_ONESHOT, + "wm8903", wm8903); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request IRQ: %d\n", + ret); + goto err; + } + + /* Enable write sequencer interrupts */ + snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_IM_WSEQ_BUSY_EINT, 0); + } + /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1618,7 +1787,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_irq; } ret = snd_soc_register_dai(&wm8903_dai); @@ -1631,6 +1800,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, err_codec: snd_soc_unregister_codec(codec); +err_irq: + if (i2c->irq) + free_irq(i2c->irq, wm8903); err: wm8903_codec = NULL; kfree(wm8903); @@ -1640,12 +1812,16 @@ err: static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); + struct wm8903_priv *priv = codec->private_data; snd_soc_unregister_dai(&wm8903_dai); snd_soc_unregister_codec(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); + if (client->irq) + free_irq(client->irq, priv); + kfree(codec->private_data); wm8903_codec = NULL; diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index 0ea27e2b9963..ce384a2ad820 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -18,6 +18,10 @@ extern struct snd_soc_dai wm8903_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8903; +extern int wm8903_mic_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, + int det, int shrt); + #define WM8903_MCLK_DIV_2 1 #define WM8903_CLK_SYS 2 #define WM8903_BCLK 3 @@ -173,28 +177,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8903; #define WM8903_VMID_RES_5K 4 /* - * R6 (0x06) - Mic Bias Control 0 - */ -#define WM8903_MICDET_HYST_ENA 0x0080 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_MASK 0x0080 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_SHIFT 7 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_HYST_ENA_WIDTH 1 /* MICDET_HYST_ENA */ -#define WM8903_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ -#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ -#define WM8903_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ -#define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ -#define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ -#define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ -#define WM8903_MICDET_ENA 0x0002 /* MICDET_ENA */ -#define WM8903_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ -#define WM8903_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ -#define WM8903_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ -#define WM8903_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ -#define WM8903_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ -#define WM8903_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ -#define WM8903_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ - -/* * R8 (0x08) - Analogue DAC 0 */ #define WM8903_DACBIAS_SEL_MASK 0x0018 /* DACBIAS_SEL - [4:3] */ @@ -1135,201 +1117,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8903; #define WM8903_MASK_WRITE_ENA_WIDTH 1 /* MASK_WRITE_ENA */ /* - * R116 (0x74) - GPIO Control 1 - */ -#define WM8903_GP1_FN_MASK 0x1F00 /* GP1_FN - [12:8] */ -#define WM8903_GP1_FN_SHIFT 8 /* GP1_FN - [12:8] */ -#define WM8903_GP1_FN_WIDTH 5 /* GP1_FN - [12:8] */ -#define WM8903_GP1_DIR 0x0080 /* GP1_DIR */ -#define WM8903_GP1_DIR_MASK 0x0080 /* GP1_DIR */ -#define WM8903_GP1_DIR_SHIFT 7 /* GP1_DIR */ -#define WM8903_GP1_DIR_WIDTH 1 /* GP1_DIR */ -#define WM8903_GP1_OP_CFG 0x0040 /* GP1_OP_CFG */ -#define WM8903_GP1_OP_CFG_MASK 0x0040 /* GP1_OP_CFG */ -#define WM8903_GP1_OP_CFG_SHIFT 6 /* GP1_OP_CFG */ -#define WM8903_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ -#define WM8903_GP1_IP_CFG 0x0020 /* GP1_IP_CFG */ -#define WM8903_GP1_IP_CFG_MASK 0x0020 /* GP1_IP_CFG */ -#define WM8903_GP1_IP_CFG_SHIFT 5 /* GP1_IP_CFG */ -#define WM8903_GP1_IP_CFG_WIDTH 1 /* GP1_IP_CFG */ -#define WM8903_GP1_LVL 0x0010 /* GP1_LVL */ -#define WM8903_GP1_LVL_MASK 0x0010 /* GP1_LVL */ -#define WM8903_GP1_LVL_SHIFT 4 /* GP1_LVL */ -#define WM8903_GP1_LVL_WIDTH 1 /* GP1_LVL */ -#define WM8903_GP1_PD 0x0008 /* GP1_PD */ -#define WM8903_GP1_PD_MASK 0x0008 /* GP1_PD */ -#define WM8903_GP1_PD_SHIFT 3 /* GP1_PD */ -#define WM8903_GP1_PD_WIDTH 1 /* GP1_PD */ -#define WM8903_GP1_PU 0x0004 /* GP1_PU */ -#define WM8903_GP1_PU_MASK 0x0004 /* GP1_PU */ -#define WM8903_GP1_PU_SHIFT 2 /* GP1_PU */ -#define WM8903_GP1_PU_WIDTH 1 /* GP1_PU */ -#define WM8903_GP1_INTMODE 0x0002 /* GP1_INTMODE */ -#define WM8903_GP1_INTMODE_MASK 0x0002 /* GP1_INTMODE */ -#define WM8903_GP1_INTMODE_SHIFT 1 /* GP1_INTMODE */ -#define WM8903_GP1_INTMODE_WIDTH 1 /* GP1_INTMODE */ -#define WM8903_GP1_DB 0x0001 /* GP1_DB */ -#define WM8903_GP1_DB_MASK 0x0001 /* GP1_DB */ -#define WM8903_GP1_DB_SHIFT 0 /* GP1_DB */ -#define WM8903_GP1_DB_WIDTH 1 /* GP1_DB */ - -/* - * R117 (0x75) - GPIO Control 2 - */ -#define WM8903_GP2_FN_MASK 0x1F00 /* GP2_FN - [12:8] */ -#define WM8903_GP2_FN_SHIFT 8 /* GP2_FN - [12:8] */ -#define WM8903_GP2_FN_WIDTH 5 /* GP2_FN - [12:8] */ -#define WM8903_GP2_DIR 0x0080 /* GP2_DIR */ -#define WM8903_GP2_DIR_MASK 0x0080 /* GP2_DIR */ -#define WM8903_GP2_DIR_SHIFT 7 /* GP2_DIR */ -#define WM8903_GP2_DIR_WIDTH 1 /* GP2_DIR */ -#define WM8903_GP2_OP_CFG 0x0040 /* GP2_OP_CFG */ -#define WM8903_GP2_OP_CFG_MASK 0x0040 /* GP2_OP_CFG */ -#define WM8903_GP2_OP_CFG_SHIFT 6 /* GP2_OP_CFG */ -#define WM8903_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ -#define WM8903_GP2_IP_CFG 0x0020 /* GP2_IP_CFG */ -#define WM8903_GP2_IP_CFG_MASK 0x0020 /* GP2_IP_CFG */ -#define WM8903_GP2_IP_CFG_SHIFT 5 /* GP2_IP_CFG */ -#define WM8903_GP2_IP_CFG_WIDTH 1 /* GP2_IP_CFG */ -#define WM8903_GP2_LVL 0x0010 /* GP2_LVL */ -#define WM8903_GP2_LVL_MASK 0x0010 /* GP2_LVL */ -#define WM8903_GP2_LVL_SHIFT 4 /* GP2_LVL */ -#define WM8903_GP2_LVL_WIDTH 1 /* GP2_LVL */ -#define WM8903_GP2_PD 0x0008 /* GP2_PD */ -#define WM8903_GP2_PD_MASK 0x0008 /* GP2_PD */ -#define WM8903_GP2_PD_SHIFT 3 /* GP2_PD */ -#define WM8903_GP2_PD_WIDTH 1 /* GP2_PD */ -#define WM8903_GP2_PU 0x0004 /* GP2_PU */ -#define WM8903_GP2_PU_MASK 0x0004 /* GP2_PU */ -#define WM8903_GP2_PU_SHIFT 2 /* GP2_PU */ -#define WM8903_GP2_PU_WIDTH 1 /* GP2_PU */ -#define WM8903_GP2_INTMODE 0x0002 /* GP2_INTMODE */ -#define WM8903_GP2_INTMODE_MASK 0x0002 /* GP2_INTMODE */ -#define WM8903_GP2_INTMODE_SHIFT 1 /* GP2_INTMODE */ -#define WM8903_GP2_INTMODE_WIDTH 1 /* GP2_INTMODE */ -#define WM8903_GP2_DB 0x0001 /* GP2_DB */ -#define WM8903_GP2_DB_MASK 0x0001 /* GP2_DB */ -#define WM8903_GP2_DB_SHIFT 0 /* GP2_DB */ -#define WM8903_GP2_DB_WIDTH 1 /* GP2_DB */ - -/* - * R118 (0x76) - GPIO Control 3 - */ -#define WM8903_GP3_FN_MASK 0x1F00 /* GP3_FN - [12:8] */ -#define WM8903_GP3_FN_SHIFT 8 /* GP3_FN - [12:8] */ -#define WM8903_GP3_FN_WIDTH 5 /* GP3_FN - [12:8] */ -#define WM8903_GP3_DIR 0x0080 /* GP3_DIR */ -#define WM8903_GP3_DIR_MASK 0x0080 /* GP3_DIR */ -#define WM8903_GP3_DIR_SHIFT 7 /* GP3_DIR */ -#define WM8903_GP3_DIR_WIDTH 1 /* GP3_DIR */ -#define WM8903_GP3_OP_CFG 0x0040 /* GP3_OP_CFG */ -#define WM8903_GP3_OP_CFG_MASK 0x0040 /* GP3_OP_CFG */ -#define WM8903_GP3_OP_CFG_SHIFT 6 /* GP3_OP_CFG */ -#define WM8903_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ -#define WM8903_GP3_IP_CFG 0x0020 /* GP3_IP_CFG */ -#define WM8903_GP3_IP_CFG_MASK 0x0020 /* GP3_IP_CFG */ -#define WM8903_GP3_IP_CFG_SHIFT 5 /* GP3_IP_CFG */ -#define WM8903_GP3_IP_CFG_WIDTH 1 /* GP3_IP_CFG */ -#define WM8903_GP3_LVL 0x0010 /* GP3_LVL */ -#define WM8903_GP3_LVL_MASK 0x0010 /* GP3_LVL */ -#define WM8903_GP3_LVL_SHIFT 4 /* GP3_LVL */ -#define WM8903_GP3_LVL_WIDTH 1 /* GP3_LVL */ -#define WM8903_GP3_PD 0x0008 /* GP3_PD */ -#define WM8903_GP3_PD_MASK 0x0008 /* GP3_PD */ -#define WM8903_GP3_PD_SHIFT 3 /* GP3_PD */ -#define WM8903_GP3_PD_WIDTH 1 /* GP3_PD */ -#define WM8903_GP3_PU 0x0004 /* GP3_PU */ -#define WM8903_GP3_PU_MASK 0x0004 /* GP3_PU */ -#define WM8903_GP3_PU_SHIFT 2 /* GP3_PU */ -#define WM8903_GP3_PU_WIDTH 1 /* GP3_PU */ -#define WM8903_GP3_INTMODE 0x0002 /* GP3_INTMODE */ -#define WM8903_GP3_INTMODE_MASK 0x0002 /* GP3_INTMODE */ -#define WM8903_GP3_INTMODE_SHIFT 1 /* GP3_INTMODE */ -#define WM8903_GP3_INTMODE_WIDTH 1 /* GP3_INTMODE */ -#define WM8903_GP3_DB 0x0001 /* GP3_DB */ -#define WM8903_GP3_DB_MASK 0x0001 /* GP3_DB */ -#define WM8903_GP3_DB_SHIFT 0 /* GP3_DB */ -#define WM8903_GP3_DB_WIDTH 1 /* GP3_DB */ - -/* - * R119 (0x77) - GPIO Control 4 - */ -#define WM8903_GP4_FN_MASK 0x1F00 /* GP4_FN - [12:8] */ -#define WM8903_GP4_FN_SHIFT 8 /* GP4_FN - [12:8] */ -#define WM8903_GP4_FN_WIDTH 5 /* GP4_FN - [12:8] */ -#define WM8903_GP4_DIR 0x0080 /* GP4_DIR */ -#define WM8903_GP4_DIR_MASK 0x0080 /* GP4_DIR */ -#define WM8903_GP4_DIR_SHIFT 7 /* GP4_DIR */ -#define WM8903_GP4_DIR_WIDTH 1 /* GP4_DIR */ -#define WM8903_GP4_OP_CFG 0x0040 /* GP4_OP_CFG */ -#define WM8903_GP4_OP_CFG_MASK 0x0040 /* GP4_OP_CFG */ -#define WM8903_GP4_OP_CFG_SHIFT 6 /* GP4_OP_CFG */ -#define WM8903_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ -#define WM8903_GP4_IP_CFG 0x0020 /* GP4_IP_CFG */ -#define WM8903_GP4_IP_CFG_MASK 0x0020 /* GP4_IP_CFG */ -#define WM8903_GP4_IP_CFG_SHIFT 5 /* GP4_IP_CFG */ -#define WM8903_GP4_IP_CFG_WIDTH 1 /* GP4_IP_CFG */ -#define WM8903_GP4_LVL 0x0010 /* GP4_LVL */ -#define WM8903_GP4_LVL_MASK 0x0010 /* GP4_LVL */ -#define WM8903_GP4_LVL_SHIFT 4 /* GP4_LVL */ -#define WM8903_GP4_LVL_WIDTH 1 /* GP4_LVL */ -#define WM8903_GP4_PD 0x0008 /* GP4_PD */ -#define WM8903_GP4_PD_MASK 0x0008 /* GP4_PD */ -#define WM8903_GP4_PD_SHIFT 3 /* GP4_PD */ -#define WM8903_GP4_PD_WIDTH 1 /* GP4_PD */ -#define WM8903_GP4_PU 0x0004 /* GP4_PU */ -#define WM8903_GP4_PU_MASK 0x0004 /* GP4_PU */ -#define WM8903_GP4_PU_SHIFT 2 /* GP4_PU */ -#define WM8903_GP4_PU_WIDTH 1 /* GP4_PU */ -#define WM8903_GP4_INTMODE 0x0002 /* GP4_INTMODE */ -#define WM8903_GP4_INTMODE_MASK 0x0002 /* GP4_INTMODE */ -#define WM8903_GP4_INTMODE_SHIFT 1 /* GP4_INTMODE */ -#define WM8903_GP4_INTMODE_WIDTH 1 /* GP4_INTMODE */ -#define WM8903_GP4_DB 0x0001 /* GP4_DB */ -#define WM8903_GP4_DB_MASK 0x0001 /* GP4_DB */ -#define WM8903_GP4_DB_SHIFT 0 /* GP4_DB */ -#define WM8903_GP4_DB_WIDTH 1 /* GP4_DB */ - -/* - * R120 (0x78) - GPIO Control 5 - */ -#define WM8903_GP5_FN_MASK 0x1F00 /* GP5_FN - [12:8] */ -#define WM8903_GP5_FN_SHIFT 8 /* GP5_FN - [12:8] */ -#define WM8903_GP5_FN_WIDTH 5 /* GP5_FN - [12:8] */ -#define WM8903_GP5_DIR 0x0080 /* GP5_DIR */ -#define WM8903_GP5_DIR_MASK 0x0080 /* GP5_DIR */ -#define WM8903_GP5_DIR_SHIFT 7 /* GP5_DIR */ -#define WM8903_GP5_DIR_WIDTH 1 /* GP5_DIR */ -#define WM8903_GP5_OP_CFG 0x0040 /* GP5_OP_CFG */ -#define WM8903_GP5_OP_CFG_MASK 0x0040 /* GP5_OP_CFG */ -#define WM8903_GP5_OP_CFG_SHIFT 6 /* GP5_OP_CFG */ -#define WM8903_GP5_OP_CFG_WIDTH 1 /* GP5_OP_CFG */ -#define WM8903_GP5_IP_CFG 0x0020 /* GP5_IP_CFG */ -#define WM8903_GP5_IP_CFG_MASK 0x0020 /* GP5_IP_CFG */ -#define WM8903_GP5_IP_CFG_SHIFT 5 /* GP5_IP_CFG */ -#define WM8903_GP5_IP_CFG_WIDTH 1 /* GP5_IP_CFG */ -#define WM8903_GP5_LVL 0x0010 /* GP5_LVL */ -#define WM8903_GP5_LVL_MASK 0x0010 /* GP5_LVL */ -#define WM8903_GP5_LVL_SHIFT 4 /* GP5_LVL */ -#define WM8903_GP5_LVL_WIDTH 1 /* GP5_LVL */ -#define WM8903_GP5_PD 0x0008 /* GP5_PD */ -#define WM8903_GP5_PD_MASK 0x0008 /* GP5_PD */ -#define WM8903_GP5_PD_SHIFT 3 /* GP5_PD */ -#define WM8903_GP5_PD_WIDTH 1 /* GP5_PD */ -#define WM8903_GP5_PU 0x0004 /* GP5_PU */ -#define WM8903_GP5_PU_MASK 0x0004 /* GP5_PU */ -#define WM8903_GP5_PU_SHIFT 2 /* GP5_PU */ -#define WM8903_GP5_PU_WIDTH 1 /* GP5_PU */ -#define WM8903_GP5_INTMODE 0x0002 /* GP5_INTMODE */ -#define WM8903_GP5_INTMODE_MASK 0x0002 /* GP5_INTMODE */ -#define WM8903_GP5_INTMODE_SHIFT 1 /* GP5_INTMODE */ -#define WM8903_GP5_INTMODE_WIDTH 1 /* GP5_INTMODE */ -#define WM8903_GP5_DB 0x0001 /* GP5_DB */ -#define WM8903_GP5_DB_MASK 0x0001 /* GP5_DB */ -#define WM8903_GP5_DB_SHIFT 0 /* GP5_DB */ -#define WM8903_GP5_DB_WIDTH 1 /* GP5_DB */ - -/* * R121 (0x79) - Interrupt Status 1 */ #define WM8903_MICSHRT_EINT 0x8000 /* MICSHRT_EINT */ diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 593e47d0e0eb..c5b50d73f2d5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2425,6 +2425,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8904); static int wm8904_register(struct wm8904_priv *wm8904, enum snd_soc_control_type control) { + struct wm8904_pdata *pdata = wm8904->pdata; int ret; struct snd_soc_codec *codec = &wm8904->codec; int i; @@ -2530,6 +2531,22 @@ static int wm8904_register(struct wm8904_priv *wm8904, WM8904_LINEOUTRZC; wm8904->reg_cache[WM8904_CLOCK_RATES_0] &= ~WM8904_SR_MODE; + /* Apply configuration from the platform data. */ + if (wm8904->pdata) { + for (i = 0; i < WM8904_GPIO_REGS; i++) { + if (!pdata->gpio_cfg[i]) + continue; + + wm8904->reg_cache[WM8904_GPIO_CONTROL_1 + i] + = pdata->gpio_cfg[i] & 0xffff; + } + + /* Zero is the default value for these anyway */ + for (i = 0; i < WM8904_MIC_REGS; i++) + wm8904->reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] + = pdata->mic_cfg[i]; + } + /* Set Class W by default - this will be managed by the Class * G widget at runtime where bypass paths are available. */ diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h index b68886df34e4..abe5059b3004 100644 --- a/sound/soc/codecs/wm8904.h +++ b/sound/soc/codecs/wm8904.h @@ -186,39 +186,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8904; #define WM8904_VMID_ENA_WIDTH 1 /* VMID_ENA */ /* - * R6 (0x06) - Mic Bias Control 0 - */ -#define WM8904_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */ -#define WM8904_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */ -#define WM8904_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */ -#define WM8904_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */ -#define WM8904_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */ -#define WM8904_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */ -#define WM8904_MICDET_ENA 0x0002 /* MICDET_ENA */ -#define WM8904_MICDET_ENA_MASK 0x0002 /* MICDET_ENA */ -#define WM8904_MICDET_ENA_SHIFT 1 /* MICDET_ENA */ -#define WM8904_MICDET_ENA_WIDTH 1 /* MICDET_ENA */ -#define WM8904_MICBIAS_ENA 0x0001 /* MICBIAS_ENA */ -#define WM8904_MICBIAS_ENA_MASK 0x0001 /* MICBIAS_ENA */ -#define WM8904_MICBIAS_ENA_SHIFT 0 /* MICBIAS_ENA */ -#define WM8904_MICBIAS_ENA_WIDTH 1 /* MICBIAS_ENA */ - -/* - * R7 (0x07) - Mic Bias Control 1 - */ -#define WM8904_MIC_DET_FILTER_ENA 0x8000 /* MIC_DET_FILTER_ENA */ -#define WM8904_MIC_DET_FILTER_ENA_MASK 0x8000 /* MIC_DET_FILTER_ENA */ -#define WM8904_MIC_DET_FILTER_ENA_SHIFT 15 /* MIC_DET_FILTER_ENA */ -#define WM8904_MIC_DET_FILTER_ENA_WIDTH 1 /* MIC_DET_FILTER_ENA */ -#define WM8904_MIC_SHORT_FILTER_ENA 0x4000 /* MIC_SHORT_FILTER_ENA */ -#define WM8904_MIC_SHORT_FILTER_ENA_MASK 0x4000 /* MIC_SHORT_FILTER_ENA */ -#define WM8904_MIC_SHORT_FILTER_ENA_SHIFT 14 /* MIC_SHORT_FILTER_ENA */ -#define WM8904_MIC_SHORT_FILTER_ENA_WIDTH 1 /* MIC_SHORT_FILTER_ENA */ -#define WM8904_MICBIAS_SEL_MASK 0x0007 /* MICBIAS_SEL - [2:0] */ -#define WM8904_MICBIAS_SEL_SHIFT 0 /* MICBIAS_SEL - [2:0] */ -#define WM8904_MICBIAS_SEL_WIDTH 3 /* MICBIAS_SEL - [2:0] */ - -/* * R8 (0x08) - Analogue DAC 0 */ #define WM8904_DAC_BIAS_SEL_MASK 0x0018 /* DAC_BIAS_SEL - [4:3] */ @@ -1200,70 +1167,6 @@ extern struct snd_soc_codec_device soc_codec_dev_wm8904; #define WM8904_FLL_CLK_REF_SRC_WIDTH 2 /* FLL_CLK_REF_SRC - [1:0] */ /* - * R121 (0x79) - GPIO Control 1 - */ -#define WM8904_GPIO1_PU 0x0020 /* GPIO1_PU */ -#define WM8904_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */ -#define WM8904_GPIO1_PU_SHIFT 5 /* GPIO1_PU */ -#define WM8904_GPIO1_PU_WIDTH 1 /* GPIO1_PU */ -#define WM8904_GPIO1_PD 0x0010 /* GPIO1_PD */ -#define WM8904_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */ -#define WM8904_GPIO1_PD_SHIFT 4 /* GPIO1_PD */ -#define WM8904_GPIO1_PD_WIDTH 1 /* GPIO1_PD */ -#define WM8904_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ -#define WM8904_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */ -#define WM8904_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */ - -/* - * R122 (0x7A) - GPIO Control 2 - */ -#define WM8904_GPIO2_PU 0x0020 /* GPIO2_PU */ -#define WM8904_GPIO2_PU_MASK 0x0020 /* GPIO2_PU */ -#define WM8904_GPIO2_PU_SHIFT 5 /* GPIO2_PU */ -#define WM8904_GPIO2_PU_WIDTH 1 /* GPIO2_PU */ -#define WM8904_GPIO2_PD 0x0010 /* GPIO2_PD */ -#define WM8904_GPIO2_PD_MASK 0x0010 /* GPIO2_PD */ -#define WM8904_GPIO2_PD_SHIFT 4 /* GPIO2_PD */ -#define WM8904_GPIO2_PD_WIDTH 1 /* GPIO2_PD */ -#define WM8904_GPIO2_SEL_MASK 0x000F /* GPIO2_SEL - [3:0] */ -#define WM8904_GPIO2_SEL_SHIFT 0 /* GPIO2_SEL - [3:0] */ -#define WM8904_GPIO2_SEL_WIDTH 4 /* GPIO2_SEL - [3:0] */ - -/* - * R123 (0x7B) - GPIO Control 3 - */ -#define WM8904_GPIO3_PU 0x0020 /* GPIO3_PU */ -#define WM8904_GPIO3_PU_MASK 0x0020 /* GPIO3_PU */ -#define WM8904_GPIO3_PU_SHIFT 5 /* GPIO3_PU */ -#define WM8904_GPIO3_PU_WIDTH 1 /* GPIO3_PU */ -#define WM8904_GPIO3_PD 0x0010 /* GPIO3_PD */ -#define WM8904_GPIO3_PD_MASK 0x0010 /* GPIO3_PD */ -#define WM8904_GPIO3_PD_SHIFT 4 /* GPIO3_PD */ -#define WM8904_GPIO3_PD_WIDTH 1 /* GPIO3_PD */ -#define WM8904_GPIO3_SEL_MASK 0x000F /* GPIO3_SEL - [3:0] */ -#define WM8904_GPIO3_SEL_SHIFT 0 /* GPIO3_SEL - [3:0] */ -#define WM8904_GPIO3_SEL_WIDTH 4 /* GPIO3_SEL - [3:0] */ - -/* - * R124 (0x7C) - GPIO Control 4 - */ -#define WM8904_GPI7_ENA 0x0200 /* GPI7_ENA */ -#define WM8904_GPI7_ENA_MASK 0x0200 /* GPI7_ENA */ -#define WM8904_GPI7_ENA_SHIFT 9 /* GPI7_ENA */ -#define WM8904_GPI7_ENA_WIDTH 1 /* GPI7_ENA */ -#define WM8904_GPI8_ENA 0x0100 /* GPI8_ENA */ -#define WM8904_GPI8_ENA_MASK 0x0100 /* GPI8_ENA */ -#define WM8904_GPI8_ENA_SHIFT 8 /* GPI8_ENA */ -#define WM8904_GPI8_ENA_WIDTH 1 /* GPI8_ENA */ -#define WM8904_GPIO_BCLK_MODE_ENA 0x0080 /* GPIO_BCLK_MODE_ENA */ -#define WM8904_GPIO_BCLK_MODE_ENA_MASK 0x0080 /* GPIO_BCLK_MODE_ENA */ -#define WM8904_GPIO_BCLK_MODE_ENA_SHIFT 7 /* GPIO_BCLK_MODE_ENA */ -#define WM8904_GPIO_BCLK_MODE_ENA_WIDTH 1 /* GPIO_BCLK_MODE_ENA */ -#define WM8904_GPIO_BCLK_SEL_MASK 0x000F /* GPIO_BCLK_SEL - [3:0] */ -#define WM8904_GPIO_BCLK_SEL_SHIFT 0 /* GPIO_BCLK_SEL - [3:0] */ -#define WM8904_GPIO_BCLK_SEL_WIDTH 4 /* GPIO_BCLK_SEL - [3:0] */ - -/* * R126 (0x7E) - Digital Pulls */ #define WM8904_MCLK_PU 0x0080 /* MCLK_PU */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d07bcc1e1c60..c2960d3ec6df 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -22,6 +22,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/wm8960.h> #include "wm8960.h" @@ -30,8 +31,14 @@ struct snd_soc_codec_device soc_codec_dev_wm8960; /* R25 - Power 1 */ +#define WM8960_VMID_MASK 0x180 #define WM8960_VREF 0x40 +/* R26 - Power 2 */ +#define WM8960_PWR2_LOUT1 0x40 +#define WM8960_PWR2_ROUT1 0x20 +#define WM8960_PWR2_OUT3 0x02 + /* R28 - Anti-pop 1 */ #define WM8960_POBCTRL 0x80 #define WM8960_BUFDCOPEN 0x10 @@ -41,6 +48,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8960; /* R29 - Anti-pop 2 */ #define WM8960_DISOP 0x40 +#define WM8960_DRES_MASK 0x30 /* * wm8960 register cache @@ -67,6 +75,9 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { struct wm8960_priv { u16 reg_cache[WM8960_CACHEREGNUM]; struct snd_soc_codec codec; + struct snd_soc_dapm_widget *lout1; + struct snd_soc_dapm_widget *rout1; + struct snd_soc_dapm_widget *out3; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -225,10 +236,6 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0, &wm8960_routput_mixer[0], ARRAY_SIZE(wm8960_routput_mixer)), -SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0, - &wm8960_mono_out[0], - ARRAY_SIZE(wm8960_mono_out)), - SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0), @@ -247,6 +254,17 @@ SND_SOC_DAPM_OUTPUT("SPK_RN"), SND_SOC_DAPM_OUTPUT("OUT3"), }; +static const struct snd_soc_dapm_widget wm8960_dapm_widgets_out3[] = { +SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0, + &wm8960_mono_out[0], + ARRAY_SIZE(wm8960_mono_out)), +}; + +/* Represent OUT3 as a PGA so that it gets turned on with LOUT1/ROUT1 */ +static const struct snd_soc_dapm_widget wm8960_dapm_widgets_capless[] = { +SND_SOC_DAPM_PGA("OUT3 VMID", WM8960_POWER2, 1, 0, NULL, 0), +}; + static const struct snd_soc_dapm_route audio_paths[] = { { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" }, { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" }, @@ -277,9 +295,6 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } , { "Right Output Mixer", "PCM Playback Switch", "Right DAC" }, - { "Mono Output Mixer", "Left Switch", "Left Output Mixer" }, - { "Mono Output Mixer", "Right Switch", "Right Output Mixer" }, - { "LOUT1 PGA", NULL, "Left Output Mixer" }, { "ROUT1 PGA", NULL, "Right Output Mixer" }, @@ -296,17 +311,65 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "SPK_LP", NULL, "Left Speaker Output" }, { "SPK_RN", NULL, "Right Speaker Output" }, { "SPK_RP", NULL, "Right Speaker Output" }, +}; + +static const struct snd_soc_dapm_route audio_paths_out3[] = { + { "Mono Output Mixer", "Left Switch", "Left Output Mixer" }, + { "Mono Output Mixer", "Right Switch", "Right Output Mixer" }, { "OUT3", NULL, "Mono Output Mixer", } }; +static const struct snd_soc_dapm_route audio_paths_capless[] = { + { "HP_L", NULL, "OUT3 VMID" }, + { "HP_R", NULL, "OUT3 VMID" }, + + { "OUT3 VMID", NULL, "Left Output Mixer" }, + { "OUT3 VMID", NULL, "Right Output Mixer" }, +}; + static int wm8960_add_widgets(struct snd_soc_codec *codec) { + struct wm8960_data *pdata = codec->dev->platform_data; + struct wm8960_priv *wm8960 = codec->private_data; + struct snd_soc_dapm_widget *w; + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets, ARRAY_SIZE(wm8960_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + /* In capless mode OUT3 is used to provide VMID for the + * headphone outputs, otherwise it is used as a mono mixer. + */ + if (pdata && pdata->capless) { + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_capless, + ARRAY_SIZE(wm8960_dapm_widgets_capless)); + + snd_soc_dapm_add_routes(codec, audio_paths_capless, + ARRAY_SIZE(audio_paths_capless)); + } else { + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets_out3, + ARRAY_SIZE(wm8960_dapm_widgets_out3)); + + snd_soc_dapm_add_routes(codec, audio_paths_out3, + ARRAY_SIZE(audio_paths_out3)); + } + + /* We need to power up the headphone output stage out of + * sequence for capless mode. To save scanning the widget + * list each time to find the desired power state do so now + * and save the result. + */ + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (strcmp(w->name, "LOUT1 PGA") == 0) + wm8960->lout1 = w; + if (strcmp(w->name, "ROUT1 PGA") == 0) + wm8960->rout1 = w; + if (strcmp(w->name, "OUT3 VMID") == 0) + wm8960->out3 = w; + } + return 0; } @@ -407,10 +470,9 @@ static int wm8960_mute(struct snd_soc_dai *dai, int mute) return 0; } -static int wm8960_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) +static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) { - struct wm8960_data *pdata = codec->dev->platform_data; u16 reg; switch (level) { @@ -429,18 +491,8 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, - WM8960_POBCTRL | WM8960_SOFT_ST | - WM8960_BUFDCOPEN | WM8960_BUFIOEN); - - /* Discharge HP output */ - reg = WM8960_DISOP; - if (pdata) - reg |= pdata->dres << 4; - snd_soc_write(codec, WM8960_APOP2, reg); - - msleep(400); - - snd_soc_write(codec, WM8960_APOP2, 0); + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Enable & ramp VMID at 2x50k */ reg = snd_soc_read(codec, WM8960_POWER1); @@ -471,8 +523,101 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID and VREF, let them discharge */ snd_soc_write(codec, WM8960_POWER1, 0); msleep(600); + break; + } + + codec->bias_level = level; + + return 0; +} + +static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8960_priv *wm8960 = codec->private_data; + int reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + /* Enable anti pop mode */ + snd_soc_update_bits(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN); + + /* Enable LOUT1, ROUT1 and OUT3 if they're enabled */ + reg = 0; + if (wm8960->lout1 && wm8960->lout1->power) + reg |= WM8960_PWR2_LOUT1; + if (wm8960->rout1 && wm8960->rout1->power) + reg |= WM8960_PWR2_ROUT1; + if (wm8960->out3 && wm8960->out3->power) + reg |= WM8960_PWR2_OUT3; + snd_soc_update_bits(codec, WM8960_POWER2, + WM8960_PWR2_LOUT1 | + WM8960_PWR2_ROUT1 | + WM8960_PWR2_OUT3, reg); + + /* Enable VMID at 2*50k */ + snd_soc_update_bits(codec, WM8960_POWER1, + WM8960_VMID_MASK, 0x80); + + /* Ramp */ + msleep(100); + + /* Enable VREF */ + snd_soc_update_bits(codec, WM8960_POWER1, + WM8960_VREF, WM8960_VREF); + + msleep(100); + break; + + case SND_SOC_BIAS_ON: + /* Enable anti-pop mode */ + snd_soc_update_bits(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN); + + /* Disable VMID and VREF */ + snd_soc_update_bits(codec, WM8960_POWER1, + WM8960_VREF | WM8960_VMID_MASK, 0); + break; + + default: + break; + } + break; + + case SND_SOC_BIAS_STANDBY: + switch (codec->bias_level) { + case SND_SOC_BIAS_PREPARE: + /* Disable HP discharge */ + snd_soc_update_bits(codec, WM8960_APOP2, + WM8960_DISOP | WM8960_DRES_MASK, + 0); + + /* Disable anti-pop features */ + snd_soc_update_bits(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN); + break; + + default: + break; + } + break; - snd_soc_write(codec, WM8960_APOP1, 0); + case SND_SOC_BIAS_OFF: break; } @@ -662,7 +807,7 @@ static int wm8960_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF); + codec->set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -681,8 +826,8 @@ static int wm8960_resume(struct platform_device *pdev) codec->hw_write(codec->control_data, data, 2); } - wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8960_set_bias_level(codec, codec->suspend_bias_level); + codec->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->set_bias_level(codec, codec->suspend_bias_level); return 0; } @@ -752,6 +897,8 @@ static int wm8960_register(struct wm8960_priv *wm8960, goto err; } + codec->set_bias_level = wm8960_set_bias_level_out3; + if (!pdata) { dev_warn(codec->dev, "No platform data supplied\n"); } else { @@ -759,6 +906,9 @@ static int wm8960_register(struct wm8960_priv *wm8960, dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); pdata->dres = 0; } + + if (pdata->capless) + codec->set_bias_level = wm8960_set_bias_level_capless; } mutex_init(&codec->mutex); @@ -769,7 +919,6 @@ static int wm8960_register(struct wm8960_priv *wm8960, codec->name = "WM8960"; codec->owner = THIS_MODULE; codec->bias_level = SND_SOC_BIAS_OFF; - codec->set_bias_level = wm8960_set_bias_level; codec->dai = &wm8960_dai; codec->num_dai = 1; codec->reg_cache_size = WM8960_CACHEREGNUM; @@ -791,7 +940,7 @@ static int wm8960_register(struct wm8960_priv *wm8960, wm8960_dai.dev = codec->dev; - wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ reg = snd_soc_read(codec, WM8960_LINVOL); @@ -840,7 +989,7 @@ err: static void wm8960_unregister(struct wm8960_priv *wm8960) { - wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF); + wm8960->codec.set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8960_dai); snd_soc_unregister_codec(&wm8960->codec); kfree(wm8960); @@ -882,7 +1031,7 @@ MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); static struct i2c_driver wm8960_i2c_driver = { .driver = { - .name = "WM8960 I2C Codec", + .name = "wm8960", .owner = THIS_MODULE, }, .probe = wm8960_i2c_probe, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index c9af56c9d9d4..d67bfe1300da 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -114,14 +114,4 @@ extern struct snd_soc_dai wm8960_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8960; -#define WM8960_DRES_400R 0 -#define WM8960_DRES_200R 1 -#define WM8960_DRES_600R 2 -#define WM8960_DRES_150R 3 -#define WM8960_DRES_MAX 3 - -struct wm8960_data { - int dres; -}; - #endif diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771c33a4..d01d3091fe81 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3332,6 +3332,36 @@ static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int reg, val, mask; + + switch (codec_dai->id) { + case 1: + reg = WM8994_AIF1_MASTER_SLAVE; + mask = WM8994_AIF1_TRI; + break; + case 2: + reg = WM8994_AIF2_MASTER_SLAVE; + mask = WM8994_AIF2_TRI; + break; + case 3: + reg = WM8994_POWER_MANAGEMENT_6; + mask = WM8994_AIF3_TRI; + break; + default: + return -EINVAL; + } + + if (tristate) + val = mask; + else + val = 0; + + return snd_soc_update_bits(codec, reg, mask, reg); +} + #define WM8994_RATES SNDRV_PCM_RATE_8000_96000 #define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ @@ -3343,6 +3373,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .hw_params = wm8994_hw_params, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, + .set_tristate = wm8994_set_tristate, }; static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { @@ -3351,6 +3382,11 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .hw_params = wm8994_hw_params, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, + .set_tristate = wm8994_set_tristate, +}; + +static struct snd_soc_dai_ops wm8994_aif3_dai_ops = { + .set_tristate = wm8994_set_tristate, }; struct snd_soc_dai wm8994_dai[] = { @@ -3394,6 +3430,7 @@ struct snd_soc_dai wm8994_dai[] = { }, { .name = "WM8994 AIF3", + .id = 3, .playback = { .stream_name = "AIF3 Playback", .channels_min = 2, @@ -3408,6 +3445,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, + .ops = &wm8994_aif3_dai_ops, } }; EXPORT_SYMBOL_GPL(wm8994_dai); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 0ad9f5d536c6..486bdd21a98a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -74,7 +74,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) msleep(1); reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY); + } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); if (reg & WM8993_DCS_DATAPATH_BUSY) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39418c0..6bbf001f6591 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,15 +12,38 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d71b988..a93679d618cd 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe6684fc2..97f74d6a33e6 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,10 +28,12 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" +#include "davinci-vcif.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -81,10 +83,24 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int evm_spdif_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; + + /* set cpu DAI configuration */ + return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); +} + static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; +static struct snd_soc_ops evm_spdif_ops = { + .hw_params = evm_spdif_hw_params, +}; + /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -151,6 +167,22 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai = { +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +#endif +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -165,7 +197,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .stream_name = "spdif", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], .codec_dai = &dit_stub_dai, - .ops = &evm_ops, + .ops = &evm_spdif_ops, }, }; static struct snd_soc_dai_link da8xx_evm_dai = { @@ -177,7 +209,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +217,15 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = 1, +}; + + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +258,17 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +296,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 6362ca05506e..4aad7ecc90a2 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -585,7 +585,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; - davinci_i2s_dai.dma_data = dev->dma_params; + davinci_i2s_dai.capture.dma_data = dev->dma_params; + davinci_i2s_dai.playback.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ab6518d86f18..c056bfbe0340 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -917,7 +917,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; - davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 80c7fdf2f521..2dc406f42fe7 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *pa; struct davinci_pcm_dma_params *params; + + pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); if (!pa) return -ENODEV; params = &pa[substream->stream]; diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 000000000000..54b91e1768c4 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,273 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.capture.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.playback.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 000000000000..571c9948724f --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar@ridgerun.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 _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index c7d0fd9b7de8..eba9b9d257a1 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC && BROKEN + depends on ARCH_MXC select SND_PCM select FIQ select SND_SOC_AC97_BUS @@ -11,3 +11,11 @@ config SND_IMX_SOC config SND_MXC_SOC_SSI tristate +config SND_MXC_SOC_WM1133_EV1 + tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" + depends on SND_IMX_SOC && EXPERIMENTAL + select SND_SOC_WM8350 + select SND_MXC_SOC_SSI + help + Enable support for audio on the i.MX31ADS with the WM1133-EV1 + PMIC board with WM8835x fitted. diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 9f8bb92ddfcc..2d203635ac11 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -9,4 +9,7 @@ obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-wm1133-ev1-objs := wm1133-ev1.o + obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 19452e44afdc..c78c000e2afe 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -83,11 +83,13 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); if (iprtd->dma < 0) { pr_err("Failed to claim the audio DMA\n"); @@ -192,10 +194,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int err; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->substream = substream; iprtd->buf = (unsigned int *)substream->dma_buffer.area; iprtd->period_cnt = 0; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 56f46a75d297..28e55c7b14b4 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -234,17 +234,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct imx_ssi *ssi = cpu_dai->private_data; + struct imx_pcm_dma_params *dma_data; u32 reg, sccr; /* Tx/Rx config */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg = SSI_STCCR; - cpu_dai->dma_data = &ssi->dma_params_tx; + dma_data = &ssi->dma_params_tx; } else { reg = SSI_SRCCR; - cpu_dai->dma_data = &ssi->dma_params_rx; + dma_data = &ssi->dma_params_rx; } + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; /* DAI data (word) size */ diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c new file mode 100644 index 000000000000..a6e7d9497639 --- /dev/null +++ b/sound/soc/imx/wm1133-ev1.c @@ -0,0 +1,308 @@ +/* + * wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS + * + * Copyright (c) 2010 Wolfson Microelectronics plc + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * Based on an earlier driver for the same hardware by Liam Girdwood. + * + * 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/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/audmux.h> + +#include "imx-ssi.h" +#include "../codecs/wm8350.h" + +/* There is a silicon mic on the board optionally connected via a solder pad + * SP1. Define this to enable it. + */ +#undef USE_SIMIC + +struct _wm8350_audio { + unsigned int channels; + snd_pcm_format_t format; + unsigned int rate; + unsigned int sysclk; + unsigned int bclkdiv; + unsigned int clkdiv; + unsigned int lr_rate; +}; + +/* in order of power consumption per rate (lowest first) */ +static const struct _wm8350_audio wm8350_audio[] = { + /* 16bit mono modes */ + {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1, + WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,}, + + /* 16 bit stereo modes */ + {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000, + WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000, + WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000, + WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600, + WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600, + WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200, + WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, + + /* 24bit stereo modes */ + {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, + {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200, + WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, +}; + +static int wm1133_ev1_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 i, found = 0; + snd_pcm_format_t format = params_format(params); + unsigned int rate = params_rate(params); + unsigned int channels = params_channels(params); + u32 dai_format; + + /* find the correct audio parameters */ + for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { + if (rate == wm8350_audio[i].rate && + format == wm8350_audio[i].format && + channels == wm8350_audio[i].channels) { + found = 1; + break; + } + } + if (!found) + return -EINVAL; + + /* codec FLL input is 14.75 MHz from MCLK */ + snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); + + dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + /* TODO: The SSI driver should figure this out for us */ + switch (channels) { + case 2: + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + break; + case 1: + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0); + break; + default: + return -EINVAL; + } + + /* set MCLK as the codec system clock for DAC and ADC */ + snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK, + wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN); + + /* set codec BCLK division for sample rate */ + snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV, + wm8350_audio[i].bclkdiv); + + /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */ + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate); + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate); + + /* now configure DAC and ADC clocks */ + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv); + + snd_soc_dai_set_clkdiv(codec_dai, + WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv); + + return 0; +} + +static struct snd_soc_ops wm1133_ev1_ops = { + .hw_params = wm1133_ev1_hw_params, +}; + +static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = { +#ifdef USE_SIMIC + SND_SOC_DAPM_MIC("SiMIC", NULL), +#endif + SND_SOC_DAPM_MIC("Mic1 Jack", NULL), + SND_SOC_DAPM_MIC("Mic2 Jack", NULL), + SND_SOC_DAPM_LINE("Line In Jack", NULL), + SND_SOC_DAPM_LINE("Line Out Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +/* imx32ads soc_card audio map */ +static const struct snd_soc_dapm_route wm1133_ev1_map[] = { + +#ifdef USE_SIMIC + /* SiMIC --> IN1LN (with automatic bias) via SP1 */ + { "IN1LN", NULL, "Mic Bias" }, + { "Mic Bias", NULL, "SiMIC" }, +#endif + + /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */ + { "IN1LN", NULL, "Mic Bias" }, + { "IN1LP", NULL, "Mic1 Jack" }, + { "Mic Bias", NULL, "Mic1 Jack" }, + + /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */ + { "IN1RN", NULL, "Mic Bias" }, + { "IN1RP", NULL, "Mic2 Jack" }, + { "Mic Bias", NULL, "Mic2 Jack" }, + + /* Line in Jack --> AUX (L+R) */ + { "IN3R", NULL, "Line In Jack" }, + { "IN3L", NULL, "Line In Jack" }, + + /* Out1 --> Headphone Jack */ + { "Headphone Jack", NULL, "OUT1R" }, + { "Headphone Jack", NULL, "OUT1L" }, + + /* Out1 --> Line Out Jack */ + { "Line Out Jack", NULL, "OUT2R" }, + { "Line Out Jack", NULL, "OUT2L" }, +}; + +static struct snd_soc_jack hp_jack; + +static struct snd_soc_jack_pin hp_jack_pins[] = { + { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE }, +}; + +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE }, + { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE }, +}; + +static int wm1133_ev1_init(struct snd_soc_codec *codec) +{ + struct snd_soc_card *card = codec->socdev->card; + + snd_soc_dapm_new_controls(codec, wm1133_ev1_widgets, + ARRAY_SIZE(wm1133_ev1_widgets)); + + snd_soc_dapm_add_routes(codec, wm1133_ev1_map, + ARRAY_SIZE(wm1133_ev1_map)); + + /* Headphone jack detection */ + snd_soc_jack_new(card, "Headphone", SND_JACK_HEADPHONE, &hp_jack); + snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), + hp_jack_pins); + wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); + + /* Microphone jack detection */ + snd_soc_jack_new(card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, + SND_JACK_BTN_0); + + snd_soc_dapm_force_enable_pin(codec, "Mic Bias"); + + return 0; +} + + +static struct snd_soc_dai_link wm1133_ev1_dai = { + .name = "WM1133-EV1", + .stream_name = "Audio", + .cpu_dai = &imx_ssi_pcm_dai[0], + .codec_dai = &wm8350_dai, + .init = wm1133_ev1_init, + .ops = &wm1133_ev1_ops, + .symmetric_rates = 1, +}; + +static struct snd_soc_card wm1133_ev1 = { + .name = "WM1133-EV1", + .platform = &imx_soc_platform, + .dai_link = &wm1133_ev1_dai, + .num_links = 1, +}; + +static struct snd_soc_device wm1133_ev1_snd_devdata = { + .card = &wm1133_ev1, + .codec_dev = &soc_codec_dev_wm8350, +}; + +static struct platform_device *wm1133_ev1_snd_device; + +static int __init wm1133_ev1_audio_init(void) +{ + int ret; + unsigned int ptcr, pdcr; + + /* SSI0 mastered by port 5 */ + ptcr = MXC_AUDMUX_V2_PTCR_SYN | + MXC_AUDMUX_V2_PTCR_TFSDIR | + MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | + MXC_AUDMUX_V2_PTCR_TCLKDIR | + MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); + + ptcr = MXC_AUDMUX_V2_PTCR_SYN; + pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); + mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); + + wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); + if (!wm1133_ev1_snd_device) + return -ENOMEM; + + platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1_snd_devdata); + wm1133_ev1_snd_devdata.dev = &wm1133_ev1_snd_device->dev; + ret = platform_device_add(wm1133_ev1_snd_device); + + if (ret) + platform_device_put(wm1133_ev1_snd_device); + + return ret; +} +module_init(wm1133_ev1_audio_init); + +static void __exit wm1133_ev1_audio_exit(void) +{ + platform_device_unregister(wm1133_ev1_snd_device); +} +module_exit(wm1133_ev1_audio_exit); + +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index ad8df6cfae88..479d3b6306d7 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -1,5 +1,5 @@ /* - * mcpdm.c -- McPDM interface driver + * mcpdm.c -- McPDM interface driver * * Author: Jorge Eduardo Candelaria <x0107209@ti.com> * Copyright (C) 2009 - Texas Instruments, Inc. @@ -38,46 +38,46 @@ static struct omap_mcpdm *mcpdm; static inline void omap_mcpdm_write(u16 reg, u32 val) { - __raw_writel(val, mcpdm->io_base + reg); + __raw_writel(val, mcpdm->io_base + reg); } static inline int omap_mcpdm_read(u16 reg) { - return __raw_readl(mcpdm->io_base + reg); + return __raw_readl(mcpdm->io_base + reg); } static void omap_mcpdm_reg_dump(void) { - dev_dbg(mcpdm->dev, "***********************\n"); - dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); - dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQSTATUS)); - dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_SET)); - dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); - dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", - omap_mcpdm_read(MCPDM_IRQWAKE_EN)); - dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_SET)); - dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); - dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", - omap_mcpdm_read(MCPDM_DMAWAKEEN)); - dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", - omap_mcpdm_read(MCPDM_CTRL)); - dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_DATA)); - dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", - omap_mcpdm_read(MCPDM_UP_DATA)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); - dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", - omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); - dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", - omap_mcpdm_read(MCPDM_DN_OFFSET)); - dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); } /* @@ -86,26 +86,26 @@ static void omap_mcpdm_reg_dump(void) */ static void omap_mcpdm_reset_capture(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } static void omap_mcpdm_reset_playback(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -114,14 +114,14 @@ static void omap_mcpdm_reset_playback(int reset) */ void omap_mcpdm_start(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -130,14 +130,14 @@ void omap_mcpdm_start(int stream) */ void omap_mcpdm_stop(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -146,38 +146,38 @@ void omap_mcpdm_stop(int stream) */ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!uplink) - return -EINVAL; + if (!uplink) + return -EINVAL; - mcpdm->uplink = uplink; + mcpdm->uplink = uplink; - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - /* Configure uplink threshold */ - if (uplink->threshold > UP_THRES_MAX) - uplink->threshold = UP_THRES_MAX; + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= uplink->format & PDMOUTFORMAT; + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; - /* Uplink channels */ - mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -186,38 +186,38 @@ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!downlink) - return -EINVAL; + if (!downlink) + return -EINVAL; - mcpdm->downlink = downlink; + mcpdm->downlink = downlink; - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); - /* Configure uplink threshold */ - if (downlink->threshold > DN_THRES_MAX) - downlink->threshold = DN_THRES_MAX; + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); - /* Set pdm out format */ - ctrl = omap_mcpdm_read(MCPDM_CTRL); - ctrl &= ~PDMOUTFORMAT; - ctrl |= downlink->format & PDMOUTFORMAT; + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; - /* Downlink channels */ - mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -226,24 +226,24 @@ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) */ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; + int irq_mask = 0; - if (!uplink) - return -EINVAL; + if (!uplink) + return -EINVAL; - /* Disable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - /* Clear Downlink channels */ - mcpdm->up_channels = 0; + /* Clear Downlink channels */ + mcpdm->up_channels = 0; - mcpdm->uplink = NULL; + mcpdm->uplink = NULL; - return 0; + return 0; } /* @@ -252,124 +252,124 @@ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; + int irq_mask = 0; - if (!downlink) - return -EINVAL; + if (!downlink) + return -EINVAL; - /* Disable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); - /* Disable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - /* clear Downlink channels */ - mcpdm->dn_channels = 0; + /* clear Downlink channels */ + mcpdm->dn_channels = 0; - mcpdm->downlink = NULL; + mcpdm->downlink = NULL; - return 0; + return 0; } static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) { - struct omap_mcpdm *mcpdm_irq = dev_id; - int irq_status; - - irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); - - /* Acknowledge irq event */ - omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); - - if (irq & MCPDM_DN_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); - omap_mcpdm_reset_playback(1); - omap_mcpdm_playback_open(mcpdm_irq->downlink); - omap_mcpdm_reset_playback(0); - } - - if (irq & MCPDM_DN_IRQ) { - dev_dbg(mcpdm_irq->dev, "DN write request\n"); - } - - if (irq & MCPDM_UP_IRQ_FULL) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ_EMPTY) { - dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); - omap_mcpdm_reset_capture(1); - omap_mcpdm_capture_open(mcpdm_irq->uplink); - omap_mcpdm_reset_capture(0); - } - - if (irq & MCPDM_UP_IRQ) { - dev_dbg(mcpdm_irq->dev, "UP write request\n"); - } - - return IRQ_HANDLED; + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; } int omap_mcpdm_request(void) { - int ret; + int ret; - clk_enable(mcpdm->clk); + clk_enable(mcpdm->clk); - spin_lock(&mcpdm->lock); + spin_lock(&mcpdm->lock); - if (!mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is in use\n"); - spin_unlock(&mcpdm->lock); - ret = -EBUSY; - goto err; - } - mcpdm->free = 0; + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; - spin_unlock(&mcpdm->lock); + spin_unlock(&mcpdm->lock); - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); - ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); - if (ret) { - dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); - goto err; - } + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } - return 0; + return 0; err: - clk_disable(mcpdm->clk); - return ret; + clk_disable(mcpdm->clk); + return ret; } void omap_mcpdm_free(void) { - spin_lock(&mcpdm->lock); - if (mcpdm->free) { - dev_err(mcpdm->dev, "McPDM interface is already free\n"); - spin_unlock(&mcpdm->lock); - return; - } - mcpdm->free = 1; - spin_unlock(&mcpdm->lock); - - clk_disable(mcpdm->clk); - - free_irq(mcpdm->irq, (void *)mcpdm); + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); } /* Enable/disable DC offset cancelation for the analog @@ -377,108 +377,108 @@ void omap_mcpdm_free(void) */ int omap_mcpdm_set_offset(int offset1, int offset2) { - int offset; + int offset; - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); - /* offset cancellation for channel 1 */ - if (offset1) - offset |= DN_OFST_RX1_EN; - else - offset &= ~DN_OFST_RX1_EN; + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; - /* offset cancellation for channel 2 */ - if (offset2) - offset |= DN_OFST_RX2_EN; - else - offset &= ~DN_OFST_RX2_EN; + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; - omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - return 0; + return 0; } static int __devinit omap_mcpdm_probe(struct platform_device *pdev) { - struct resource *res; - int ret = 0; - - mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); - if (!mcpdm) { - ret = -ENOMEM; - goto exit; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no resource\n"); - goto err_resource; - } - - spin_lock_init(&mcpdm->lock); - mcpdm->free = 1; - mcpdm->io_base = ioremap(res->start, resource_size(res)); - if (!mcpdm->io_base) { - ret = -ENOMEM; - goto err_resource; - } - - mcpdm->irq = platform_get_irq(pdev, 0); - - mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); - if (IS_ERR(mcpdm->clk)) { - ret = PTR_ERR(mcpdm->clk); - dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); - goto err_clk; - } - - mcpdm->dev = &pdev->dev; - platform_set_drvdata(pdev, mcpdm); - - return 0; + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; err_clk: - iounmap(mcpdm->io_base); + iounmap(mcpdm->io_base); err_resource: - kfree(mcpdm); + kfree(mcpdm); exit: - return ret; + return ret; } static int __devexit omap_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - platform_set_drvdata(pdev, NULL); + platform_set_drvdata(pdev, NULL); - clk_put(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); - iounmap(mcpdm_ptr->io_base); + iounmap(mcpdm_ptr->io_base); - mcpdm_ptr->clk = NULL; - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; - kfree(mcpdm_ptr); + kfree(mcpdm_ptr); - return 0; + return 0; } static struct platform_driver omap_mcpdm_driver = { - .probe = omap_mcpdm_probe, - .remove = __devexit_p(omap_mcpdm_remove), - .driver = { - .name = "omap-mcpdm", - }, + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, }; static struct platform_device *omap_mcpdm_device; static int __init omap_mcpdm_init(void) { - return platform_driver_register(&omap_mcpdm_driver); + return platform_driver_register(&omap_mcpdm_driver); } arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e814a9591f78..2d33a89f147a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -256,6 +256,31 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, return err; } +static snd_pcm_sframes_t omap_mcbsp_dai_delay( + 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 omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + u16 fifo_use; + snd_pcm_sframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_use = omap_mcbsp_get_tx_delay(mcbsp_data->bus_id); + else + fifo_use = omap_mcbsp_get_rx_delay(mcbsp_data->bus_id); + + /* + * Divide the used locations with the channel count to get the + * FIFO usage in samples (don't care about partial samples in the + * buffer). + */ + delay = fifo_use / substream->runtime->channels; + + return delay; +} + static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -297,7 +322,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; omap_mcbsp_dai_dma_params[id][substream->stream].data_type = OMAP_DMA_DATA_TYPE_S16; - cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcbsp_dai_dma_params[id][substream->stream]); if (mcbsp_data->configured) { /* McBSP already configured by another stream */ @@ -306,7 +333,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || + format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ regs->rcr2 |= RPHASE; regs->xcr2 |= XPHASE; @@ -351,6 +379,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID((framesize >> 1) - 1); break; @@ -402,6 +431,14 @@ 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_LEFT_J: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + regs->spcr1 |= RJUST(2); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); @@ -607,6 +644,7 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .startup = omap_mcbsp_dai_startup, .shutdown = omap_mcbsp_dai_shutdown, .trigger = omap_mcbsp_dai_trigger, + .delay = omap_mcbsp_dai_delay, .hw_params = omap_mcbsp_dai_hw_params, .set_fmt = omap_mcbsp_dai_set_dai_fmt, .set_clkdiv = omap_mcbsp_dai_set_clkdiv, diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 25f19e4728bf..b7f4f7e015f3 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; int channels, err, link_mask = 0; - cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcpdm_dai_dma_params[stream]); channels = params_channels(params); switch (channels) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 825db385f01f..39538c0f81f0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -100,9 +100,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_runtime_data *prtd = runtime->private_data; - struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; + struct omap_pcm_dma_data *dma_data; int err = 0; + dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9e95e5117c88..6959c5199160 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -121,10 +121,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, ssp_disable(ssp); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + return ret; } @@ -141,10 +140,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable(ssp->clk); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } #ifdef CONFIG_PM @@ -569,19 +566,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + struct pxa2xx_pcm_dma_params *dma_data; + + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); + kfree(dma_data); /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + dma_data = ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_dma_data(dai, substream, dma_data); + /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e9ae7b3a7e00..d314115e3dd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; + dma_data = &pxa2xx_ac97_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; + dma_data = &pxa2xx_ac97_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; + dma_data = &pxa2xx_ac97_pcm_aux_mono_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b8f655d1ad8..c1a5275721e4 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); @@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; + dma_data = &pxa2xx_i2s_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; + dma_data = &pxa2xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d38e39575f51..adc7e6f15f93 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct pxa2xx_pcm_dma_params *dma; int ret; + dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma) diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index ee8ed9d7e703..ecf4fd04ae96 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c_ac97_pcm_out; + dma_data = &s3c_ac97_pcm_out; else - cpu_dai->dma_data = &s3c_ac97_pcm_in; + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } @@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &s3c_ac97_mic_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); return 0; } @@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 7725e26d6c91..1b61c23ff300 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); int ret = 0; + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d8374fe6..865f93143bf1 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -16,18 +16,12 @@ * option) any later version. */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> #include <linux/delay.h> #include <linux/clk.h> -#include <linux/kernel.h> #include <linux/io.h> -#include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/initval.h> #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> @@ -332,54 +326,41 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, return 0; } -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, +static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *socdai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = i2s->dma_playback; + dma_data = i2s->dma_playback; else - dai->cpu_dai->dma_data = i2s->dma_capture; + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } -#endif - -#ifdef CONFIG_PLAT_S3C64XX - iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK); + iismod &= ~S3C64XX_IISMOD_BLC_MASK; /* Sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: - /* 8 bit sample, 16fs BCLK */ - iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS); + iismod |= S3C64XX_IISMOD_BLC_8BIT; break; case SNDRV_PCM_FORMAT_S16_LE: - /* 16 bit sample, 32fs BCLK */ break; case SNDRV_PCM_FORMAT_S24_LE: - /* 24 bit sample, 48fs BCLK */ - iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS); + iismod |= S3C64XX_IISMOD_BLC_24BIT; break; } -#endif writel(iismod, i2s->regs + S3C2412_IISMOD); pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); @@ -394,8 +375,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -431,7 +412,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, * of the auto reload mechanism of S3C24XX. * This call won't bother S3C64XX. */ - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; @@ -469,29 +450,25 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case S3C_I2SV2_DIV_BCLK: - if (div > 3) { - /* convert value to bit field */ - - switch (div) { - case 16: - div = S3C2412_IISMOD_BCLK_16FS; - break; + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; - case 32: - div = S3C2412_IISMOD_BCLK_32FS; - break; + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; - case 24: - div = S3C2412_IISMOD_BCLK_24FS; - break; + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; - case 48: - div = S3C2412_IISMOD_BCLK_48FS; - break; + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; - default: - return -EINVAL; - } + default: + return -EINVAL; } reg = readl(i2s->regs + S3C2412_IISMOD); @@ -502,29 +479,25 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, break; case S3C_I2SV2_DIV_RCLK: - if (div > 3) { - /* convert value to bit field */ - - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; - default: - return -EINVAL; - } + default: + return -EINVAL; } reg = readl(i2s->regs + S3C2412_IISMOD); @@ -550,6 +523,21 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + snd_pcm_sframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; @@ -732,10 +720,15 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) struct snd_soc_dai_ops *ops = dai->ops; ops->trigger = s3c2412_i2s_trigger; - ops->hw_params = s3c2412_i2s_hw_params; + if (!ops->hw_params) + ops->hw_params = s3c_i2sv2_hw_params; ops->set_fmt = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay; + dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index ecf8eaaed1db..b094d3c23cbe 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -25,6 +25,10 @@ #define S3C_I2SV2_DIV_RCLK (2) #define S3C_I2SV2_DIV_PRESCALER (3) +#define S3C_I2SV2_CLKSRC_PCLK 0 +#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 +#define S3C_I2SV2_CLKSRC_CDCLK 2 + /** * struct s3c_i2sv2_info - S3C I2S-V2 information * @dev: The parent device passed to use from the probe. diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index a98f40c3cd29..326f0a9e7e30 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = pcm->dma_playback; + dma_data = pcm->dma_playback; else - dai->cpu_dai->dma_data = pcm->dma_capture; + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Strictly check for sample size */ switch (params_format(params)) { diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 359e59346ba2..f3148f98b419 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -103,6 +103,10 @@ struct clk *s3c2412_get_iisclk(void) } EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} static int s3c2412_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) @@ -142,6 +146,41 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, return 0; } +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + #define S3C2412_I2S_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 | \ @@ -149,6 +188,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { .set_sysclk = s3c2412_i2s_set_sysclk, + .hw_params = s3c2412_i2s_hw_params, }; struct snd_soc_dai s3c2412_i2s_dai = { diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 92848e54be16..60cac002a830 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -21,8 +21,8 @@ #define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK #define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER -#define S3C2412_CLKSRC_PCLK (0) -#define S3C2412_CLKSRC_I2SCLK (1) +#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS extern struct clk *s3c2412_get_iisclk(void); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0bc5950b9f02..c3ac890a3986 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; + dma_data = &s3c24xx_i2s_pcm_stereo_out; else - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -258,13 +261,11 @@ 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 s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 1; + dma_data->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 2; + dma_data->dma_size = 2; break; default: return -EINVAL; @@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index a72c251401ac..ab1fa159d3ae 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -12,9 +12,6 @@ * published by the Free Software Foundation. */ -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> #include <linux/clk.h> #include <linux/gpio.h> #include <linux/io.h> @@ -130,15 +127,6 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, } -#define S3C64XX_I2S_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 S3C64XX_I2S_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) - static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { .set_sysclk = s3c64xx_i2s_set_sysclk, }; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index abe7253b55fc..53d2a0a0df36 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -23,9 +23,18 @@ struct clk; #define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK #define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER -#define S3C64XX_CLKSRC_PCLK (0) -#define S3C64XX_CLKSRC_MUX (1) -#define S3C64XX_CLKSRC_CDCLK (2) +#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS +#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK + +#define S3C64XX_I2S_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 S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) extern struct snd_soc_dai s3c64xx_i2s_dai[]; diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index c5cda187ecab..fa23854c5f3a 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -518,7 +518,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) s6000_i2s_dai.dev = &pdev->dev; s6000_i2s_dai.private_data = dev; - s6000_i2s_dai.dma_data = &dev->dma_params; + s6000_i2s_dai.capture.dma_data = &dev->dma_params; + s6000_i2s_dai.playback.dma_data = &dev->dma_params; dev->sifbase = sifmem->start; dev->scbbase = mmio; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1d61109e09fa..9c7f7f00cebb 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -58,13 +58,15 @@ static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int channel; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; @@ -101,7 +103,8 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) { struct snd_pcm *pcm = data; struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); struct s6000_runtime_data *prtd; unsigned int has_xrun; int i, ret = IRQ_NONE; @@ -172,11 +175,13 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; int srcinc; u32 dma; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -212,10 +217,12 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) { struct s6000_runtime_data *prtd = substream->runtime->private_data; struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; unsigned long flags; u32 channel; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) channel = par->dma_out; else @@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + ret = par->trigger(substream, cmd, 0); if (ret < 0) return ret; @@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream) static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd = runtime->private_data; unsigned long flags; unsigned int offset; dma_addr_t count; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) static int s6000_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); ret = snd_pcm_hw_constraint_step(runtime, 0, @@ -364,7 +376,7 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par; int ret; ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (par->same_rate) { spin_lock(&par->lock); if (par->rate == -1 || @@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; - struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *par = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); spin_lock(&par->lock); par->in_use &= ~(1 << substream->stream); @@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = { static void s6000_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); free_irq(params->irq, pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -429,9 +445,11 @@ static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; - struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_pcm_dma_params *params; int res; + params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 106674979b53..f07f6d8b93e1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU select DMA_ENGINE select DMADEVICES select SH_DMAE + select FW_LOADER ## ## Boards diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index 5263ab18f827..be018542314e 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -22,11 +22,25 @@ #include <sound/sh_fsi.h> #include <../sound/soc/codecs/ak4642.h> +static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0); + + return ret; +} + static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", .cpu_dai = &fsi_soc_dai[0], /* fsi */ .codec_dai = &ak4642_dai, + .init = fsi_ak4642_dai_init, .ops = NULL, }; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 5869dc3be781..9dfe9a58a314 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -159,7 +159,8 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, BUG_ON(codec->volatile_register); - data[0] = reg & 0xff; + reg &= 0xff; + data[0] = reg; data[1] = value & 0xff; if (reg < codec->reg_cache_size) @@ -180,6 +181,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, unsigned int reg) { u8 *cache = codec->reg_cache; + reg &= 0xff; if (reg >= codec->reg_cache_size) return -1; return cache[reg]; @@ -226,6 +228,40 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, } #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg xfer[2]; + u8 reg = r; + u8 data; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 1; + xfer[1].buf = &data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return data; +} +#else +#define snd_soc_8_8_read_i2c NULL +#endif + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, unsigned int r) { @@ -366,6 +402,84 @@ static int snd_soc_16_8_spi_write(void *control_data, const char *data, #define snd_soc_16_8_spi_write NULL #endif +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg xfer[2]; + u16 reg = cpu_to_be16(r); + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 2; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return be16_to_cpu(data); +} +#else +#define snd_soc_16_16_read_i2c NULL +#endif + +static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg)) { + if (codec->cache_only) + return -EINVAL; + + return codec->hw_read(codec, reg); + } + + return cache[reg]; +} + +static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[4]; + int ret; + + data[0] = (reg >> 8) & 0xff; + data[1] = reg & 0xff; + data[2] = (value >> 8) & 0xff; + data[3] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->cache_only) { + codec->cache_sync = 1; + return 0; + } + + ret = codec->hw_write(codec->control_data, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} static struct { int addr_bits; @@ -388,6 +502,7 @@ static struct { { .addr_bits = 8, .data_bits = 8, .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, + .i2c_read = snd_soc_8_8_read_i2c, }, { .addr_bits = 8, .data_bits = 16, @@ -400,6 +515,11 @@ static struct { .i2c_read = snd_soc_16_8_read_i2c, .spi_write = snd_soc_16_8_spi_write, }, + { + .addr_bits = 16, .data_bits = 16, + .write = snd_soc_16_16_write, .read = snd_soc_16_16_read, + .i2c_read = snd_soc_16_16_read_i2c, + }, }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556ef431..1ff553bbc063 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -315,7 +315,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || machine->symmetric_rates) { - dev_dbg(card->dev, "Symmetry forces %dHz rate\n", + dev_dbg(card->dev, "Symmetry forces %dHz rate\n", machine->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, @@ -404,6 +404,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.formats & cpu_dai->playback.formats; runtime->hw.rates = codec_dai->playback.rates & cpu_dai->playback.rates; + if (codec_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->playback.rates; + if (cpu_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->playback.rates; } else { runtime->hw.rate_min = max(codec_dai->capture.rate_min, @@ -421,6 +427,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->capture.formats & cpu_dai->capture.formats; runtime->hw.rates = codec_dai->capture.rates & cpu_dai->capture.rates; + if (codec_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->capture.rates; + if (cpu_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->capture.rates; } snd_pcm_limit_hw_rates(runtime); @@ -454,12 +466,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) 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; - else - cpu_dai->capture.active = codec_dai->capture.active = 1; - cpu_dai->active = codec_dai->active = 1; - cpu_dai->runtime = runtime; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active++; + codec_dai->playback.active++; + } else { + cpu_dai->capture.active++; + codec_dai->capture.active++; + } + cpu_dai->active++; + codec_dai->active++; card->codec->active++; mutex_unlock(&pcm_mutex); return 0; @@ -535,15 +550,16 @@ static int soc_codec_close(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->playback.active = codec_dai->playback.active = 0; - else - cpu_dai->capture.active = codec_dai->capture.active = 0; - - if (codec_dai->playback.active == 0 && - codec_dai->capture.active == 0) { - cpu_dai->active = codec_dai->active = 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active--; + codec_dai->playback.active--; + } else { + cpu_dai->capture.active--; + codec_dai->capture.active--; } + + cpu_dai->active--; + codec_dai->active--; codec->active--; /* Muting the DAC suppresses artifacts caused during digital @@ -563,7 +579,6 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (platform->pcm_ops->close) platform->pcm_ops->close(substream); - cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* start delayed pop wq here for playback streams */ @@ -801,6 +816,41 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +/* + * soc level wrapper for pointer callback + * If cpu_dai, codec_dai, platform driver has the delay callback, than + * the runtime->delay will be updated accordingly. + */ +static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; + struct snd_soc_platform *platform = card->platform; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t offset = 0; + snd_pcm_sframes_t delay = 0; + + if (platform->pcm_ops->pointer) + offset = platform->pcm_ops->pointer(substream); + + if (cpu_dai->ops->delay) + delay += cpu_dai->ops->delay(substream, cpu_dai); + + if (codec_dai->ops->delay) + delay += codec_dai->ops->delay(substream, codec_dai); + + if (platform->delay) + delay += platform->delay(substream, codec_dai); + + runtime->delay = delay; + + return offset; +} + /* ASoC PCM operations */ static struct snd_pcm_ops soc_pcm_ops = { .open = soc_pcm_open, @@ -809,6 +859,7 @@ static struct snd_pcm_ops soc_pcm_ops = { .hw_free = soc_pcm_hw_free, .prepare = soc_pcm_prepare, .trigger = soc_pcm_trigger, + .pointer = soc_pcm_pointer, }; #ifdef CONFIG_PM @@ -858,7 +909,7 @@ static int soc_suspend(struct device *dev) if (cpu_dai->suspend && !cpu_dai->ac97_control) cpu_dai->suspend(cpu_dai); if (platform->suspend) - platform->suspend(cpu_dai); + platform->suspend(&card->dai_link[i]); } /* close any waiting streams and save state */ @@ -947,7 +998,7 @@ static void soc_resume_deferred(struct work_struct *work) if (cpu_dai->resume && !cpu_dai->ac97_control) cpu_dai->resume(cpu_dai); if (platform->resume) - platform->resume(cpu_dai); + platform->resume(&card->dai_link[i]); } if (card->resume_post) @@ -1232,26 +1283,25 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (!card->instantiated) - return 0; + if (card->instantiated) { + run_delayed_work(&card->delayed_work); - run_delayed_work(&card->delayed_work); + if (platform->remove) + platform->remove(pdev); - if (platform->remove) - platform->remove(pdev); + if (codec_dev->remove) + codec_dev->remove(pdev); - if (codec_dev->remove) - codec_dev->remove(pdev); + 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); + } - 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 (card->remove) + card->remove(pdev); } - if (card->remove) - card->remove(pdev); - snd_soc_unregister_card(card); return 0; @@ -1335,7 +1385,6 @@ static int soc_new_pcm(struct snd_soc_device *socdev, dai_link->pcm = pcm; pcm->private_data = rtd; soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.pointer = platform->pcm_ops->pointer; soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; soc_pcm_ops.copy = platform->pcm_ops->copy; soc_pcm_ops.silence = platform->pcm_ops->silence; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6c3351095786..476dbe6dabad 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -97,7 +97,6 @@ static void pop_dbg(u32 pop_time, const char *fmt, ...) if (pop_time) { vprintk(fmt, args); - pop_wait(pop_time); } va_end(args); @@ -314,62 +313,14 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) 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(codec->pop_time); + snd_soc_write(codec, widget->reg, new); } pr_debug("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } -/* ramps the volume up or down to minimise pops before or after a - * DAPM power event */ -static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) -{ - const struct snd_kcontrol_new *k = widget->kcontrols; - - if (widget->muted && !power) - return 0; - if (!widget->muted && power) - return 0; - - if (widget->num_kcontrols && k) { - struct soc_mixer_control *mc = - (struct soc_mixer_control *)k->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - - if (power) { - int i; - /* power up has happended, increase volume to last level */ - if (invert) { - for (i = max; i > widget->saved_value; i--) - snd_soc_update_bits(widget->codec, reg, mask, i); - } else { - for (i = 0; i < widget->saved_value; i++) - snd_soc_update_bits(widget->codec, reg, mask, i); - } - widget->muted = 0; - } else { - /* power down is about to occur, decrease volume to mute */ - int val = snd_soc_read(widget->codec, reg); - int i = widget->saved_value = (val >> shift) & mask; - if (invert) { - for (; i < mask; i++) - snd_soc_update_bits(widget->codec, reg, mask, i); - } else { - for (; i > 0; i--) - snd_soc_update_bits(widget->codec, reg, mask, i); - } - widget->muted = 1; - } - } - return 0; -} - /* create new dapm mixer control */ static int dapm_new_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *w) @@ -464,20 +415,10 @@ err: static int dapm_new_pga(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *w) { - struct snd_kcontrol *kcontrol; - int ret = 0; - - if (!w->num_kcontrols) - return -EINVAL; + if (w->num_kcontrols) + pr_err("asoc: PGA controls not supported: '%s'\n", w->name); - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); - ret = snd_ctl_add(codec->card, kcontrol); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); - return ret; - } - - return ret; + return 0; } /* reset 'walked' bit for each dapm path */ @@ -633,16 +574,8 @@ static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) return ret; } - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !w->power) - dapm_set_pga(w, w->power); - dapm_update_bits(w); - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && w->power) - dapm_set_pga(w, w->power); - /* power up post event */ if (w->power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { @@ -809,10 +742,6 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, pr_err("%s: pre event failed: %d\n", w->name, ret); } - - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !w->power) - dapm_set_pga(w, w->power); } if (reg >= 0) { @@ -824,10 +753,6 @@ static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, } list_for_each_entry(w, pending, power_list) { - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && w->power) - dapm_set_pga(w, w->power); - /* power up post event */ if (w->power && w->event && (w->event_flags & SND_SOC_DAPM_POST_PMU)) { @@ -980,7 +905,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) break; default: - power = w->power_check(w); + if (!w->force) + power = w->power_check(w); + else + power = 1; if (power) sys_power = 1; break; @@ -1075,6 +1003,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n", codec->pop_time); + pop_wait(codec->pop_time); return 0; } @@ -1337,6 +1266,9 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, if (!strcmp(w->name, pin)) { pr_debug("dapm: %s: pin %s\n", codec->name, pin); w->connected = status; + /* Allow disabling of forced pins */ + if (status == 0) + w->force = 0; return 0; } } @@ -1593,12 +1525,6 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, unsigned int invert = mc->invert; unsigned int mask = (1 << fls(max)) - 1; - /* return the saved value if we are powered down */ - if (widget->id == snd_soc_dapm_pga && !widget->power) { - ucontrol->value.integer.value[0] = widget->saved_value; - return 0; - } - ucontrol->value.integer.value[0] = (snd_soc_read(widget->codec, reg) >> shift) & mask; if (shift != rshift) @@ -1658,13 +1584,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - /* save volume value if the widget is powered down */ - if (widget->id == snd_soc_dapm_pga && !widget->power) { - widget->saved_value = val; - mutex_unlock(&widget->codec->mutex); - return 1; - } - if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { if (val) /* new connection */ @@ -2134,6 +2053,36 @@ int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin) EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** + * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * @codec: SoC codec + * @pin: pin name + * + * Enables input/output pin regardless of any other state. This is + * intended for use with microphone bias supplies used in microphone + * jack detection. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_force_enable_pin(struct snd_soc_codec *codec, const char *pin) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + pr_debug("dapm: %s: pin %s\n", codec->name, pin); + w->connected = 1; + w->force = 1; + return 0; + } + } + + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); + +/** * snd_soc_dapm_disable_pin - disable pin. * @codec: SoC codec * @pin: pin name diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 3c07a94c2e30..f8fd22cc70bc 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, { jack->card = card; INIT_LIST_HEAD(&jack->pins); + BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(card->codec->card, id, type, &jack->jack); } @@ -93,6 +94,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_disable_pin(codec, pin->pin); } + /* Report before the DAPM sync to help users updating micbias status */ + blocking_notifier_call_chain(&jack->notifier, status, NULL); + snd_soc_dapm_sync(codec); snd_jack_report(jack->jack, status); @@ -143,6 +147,40 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, } EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); +/** + * snd_soc_jack_notifier_register - Register a notifier for jack status + * + * @jack: ASoC jack + * @nb: Notifier block to register + * + * Register for notification of the current status of the jack. Note + * that it is not possible to report additional jack events in the + * callback from the notifier, this is intended to support + * applications such as enabling electrical detection only when a + * mechanical detection event has occurred. + */ +void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, + struct notifier_block *nb) +{ + blocking_notifier_chain_register(&jack->notifier, nb); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_register); + +/** + * snd_soc_jack_notifier_unregister - Unregister a notifier for jack status + * + * @jack: ASoC jack + * @nb: Notifier block to unregister + * + * Stop notifying for status changes. + */ +void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, + struct notifier_block *nb) +{ + blocking_notifier_chain_unregister(&jack->notifier, nb); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); + #ifdef CONFIG_GPIOLIB /* gpio detect */ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index c570ae3e6d55..44d6d2ec964f 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -22,8 +22,7 @@ config SND_USB_AUDIO will be called snd-usb-audio. config SND_USB_UA101 - tristate "Edirol UA-101/UA-1000 driver (EXPERIMENTAL)" - depends on EXPERIMENTAL + tristate "Edirol UA-101/UA-1000 driver" select SND_PCM select SND_RAWMIDI help @@ -65,6 +64,7 @@ config SND_USB_CAIAQ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O * Native Instruments Guitar Rig mobile + * Native Instruments Traktor Kontrol X1 To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 5bf64aef9558..e7ac7f493a8f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -2,14 +2,24 @@ # Makefile for ALSA # -snd-usb-audio-objs := usbaudio.o usbmixer.o -snd-usb-lib-objs := usbmidi.o -snd-ua101-objs := ua101.o +snd-usb-audio-objs := card.o \ + mixer.o \ + mixer_quirks.o \ + proc.o \ + quirks.o \ + format.o \ + endpoint.o \ + urb.o \ + pcm.o \ + helper.o + +snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency -obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usb-lib.o -obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o snd-usb-lib.o -obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-lib.o -obj-$(CONFIG_SND_USB_US122L) += snd-usb-lib.o +obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usbmidi-lib.o + +obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o +obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o +obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += usx2y/ caiaq/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 537102ba6b9d..36ed703a7416 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -35,33 +35,41 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; - unsigned int id = dev->chip.usb_id; + int maxval = 63; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) - && (pos == 0)) { - /* current input mode of A8DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; - return 0; - } + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): + if (pos == 0) { + /* current input mode of A8DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 2; + return 0; + } + break; - if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ) - && (pos == 0)) { - /* current input mode of A4DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + if (pos == 0) { + /* current input mode of A4DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + maxval = 127; + break; } if (is_intval) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 64; + uinfo->value.integer.max = maxval; } else { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->value.integer.min = 0; @@ -102,9 +110,10 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + unsigned char cmd = EP1_CMD_WRITE_IO; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): { /* A4DJ has only one control */ /* do not expose hardware input mode 0 */ dev->control_state[0] = ucontrol->value.integer.value[0] + 1; @@ -113,10 +122,15 @@ static int control_put(struct snd_kcontrol *kcontrol, return 1; } + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + cmd = EP1_CMD_DIMM_LEDS; + break; + } + if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + snd_usb_caiaq_send_command(dev, cmd, dev->control_state, sizeof(dev->control_state)); } else { if (ucontrol->value.integer.value[0]) @@ -124,7 +138,7 @@ static int control_put(struct snd_kcontrol *kcontrol, else dev->control_state[pos / 8] &= ~(1 << (pos % 8)); - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + snd_usb_caiaq_send_command(dev, cmd, dev->control_state, sizeof(dev->control_state)); } @@ -273,6 +287,43 @@ static struct caiaq_controller a4dj_controller[] = { { "Current input mode", 0 | CNT_INTVAL } }; +static struct caiaq_controller kontrolx1_controller[] = { + { "LED FX A: ON", 7 | CNT_INTVAL }, + { "LED FX A: 1", 6 | CNT_INTVAL }, + { "LED FX A: 2", 5 | CNT_INTVAL }, + { "LED FX A: 3", 4 | CNT_INTVAL }, + { "LED FX B: ON", 3 | CNT_INTVAL }, + { "LED FX B: 1", 2 | CNT_INTVAL }, + { "LED FX B: 2", 1 | CNT_INTVAL }, + { "LED FX B: 3", 0 | CNT_INTVAL }, + + { "LED Hotcue", 28 | CNT_INTVAL }, + { "LED Shift (white)", 29 | CNT_INTVAL }, + { "LED Shift (green)", 30 | CNT_INTVAL }, + + { "LED Deck A: FX1", 24 | CNT_INTVAL }, + { "LED Deck A: FX2", 25 | CNT_INTVAL }, + { "LED Deck A: IN", 17 | CNT_INTVAL }, + { "LED Deck A: OUT", 16 | CNT_INTVAL }, + { "LED Deck A: < BEAT", 19 | CNT_INTVAL }, + { "LED Deck A: BEAT >", 18 | CNT_INTVAL }, + { "LED Deck A: CUE/ABS", 21 | CNT_INTVAL }, + { "LED Deck A: CUP/REL", 20 | CNT_INTVAL }, + { "LED Deck A: PLAY", 23 | CNT_INTVAL }, + { "LED Deck A: SYNC", 22 | CNT_INTVAL }, + + { "LED Deck B: FX1", 26 | CNT_INTVAL }, + { "LED Deck B: FX2", 27 | CNT_INTVAL }, + { "LED Deck B: IN", 15 | CNT_INTVAL }, + { "LED Deck B: OUT", 14 | CNT_INTVAL }, + { "LED Deck B: < BEAT", 13 | CNT_INTVAL }, + { "LED Deck B: BEAT >", 12 | CNT_INTVAL }, + { "LED Deck B: CUE/ABS", 11 | CNT_INTVAL }, + { "LED Deck B: CUP/REL", 10 | CNT_INTVAL }, + { "LED Deck B: PLAY", 9 | CNT_INTVAL }, + { "LED Deck B: SYNC", 8 | CNT_INTVAL }, +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -321,10 +372,16 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): ret = add_controls(a4dj_controller, ARRAY_SIZE(a4dj_controller), dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + ret = add_controls(kontrolx1_controller, + ARRAY_SIZE(kontrolx1_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index a3f02dd97440..08ee2545830e 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -46,7 +46,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," - "{Native Instruments, GuitarRig mobile}"); + "{Native Instruments, GuitarRig mobile}" + "{Native Instruments, Traktor Kontrol X1}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -127,6 +128,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO2DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_TRAKTORKONTROLX1 + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 44e3edf88bef..f1117ecc84fd 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -5,18 +5,20 @@ #define USB_VID_NATIVEINSTRUMENTS 0x17cc -#define USB_PID_RIGKONTROL2 0x1969 -#define USB_PID_RIGKONTROL3 0x1940 -#define USB_PID_KORECONTROLLER 0x4711 -#define USB_PID_KORECONTROLLER2 0x4712 -#define USB_PID_AK1 0x0815 -#define USB_PID_AUDIO2DJ 0x041c -#define USB_PID_AUDIO4DJ 0x0839 -#define USB_PID_AUDIO8DJ 0x1978 -#define USB_PID_SESSIONIO 0x1915 -#define USB_PID_GUITARRIGMOBILE 0x0d8d +#define USB_PID_RIGKONTROL2 0x1969 +#define USB_PID_RIGKONTROL3 0x1940 +#define USB_PID_KORECONTROLLER 0x4711 +#define USB_PID_KORECONTROLLER2 0x4712 +#define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c +#define USB_PID_AUDIO4DJ 0x0839 +#define USB_PID_AUDIO8DJ 0x1978 +#define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d +#define USB_PID_TRAKTORKONTROLX1 0x2305 #define EP1_BUFSIZE 64 +#define EP4_BUFSIZE 512 #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 @@ -104,6 +106,8 @@ struct snd_usb_caiaqdev { struct input_dev *input_dev; char phys[64]; /* physical device path */ unsigned short keycode[64]; + struct urb *ep4_in_urb; + unsigned char ep4_in_buf[EP4_BUFSIZE]; #endif /* ALSA */ diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a48d309bd94c..27ed0bc651ae 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/usb.h> #include <linux/usb/input.h> +#include <sound/core.h> #include <sound/pcm.h> #include "device.h" @@ -65,6 +66,8 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; +#define KONTROLX1_INPUTS 40 + #define DEG90 (range / 2) #define DEG180 (range) #define DEG270 (DEG90 + DEG180) @@ -162,6 +165,17 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); input_sync(input_dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + input_report_abs(input_dev, ABS_HAT0X, (buf[8] << 8) | buf[9]); + input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); + input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); + input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); + input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]); + input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); + input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); + input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); + input_sync(input_dev); + break; } } @@ -201,7 +215,7 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, } static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, - char *buf, unsigned int len) + unsigned char *buf, unsigned int len) { struct input_dev *input_dev = dev->input_dev; unsigned short *keycode = input_dev->keycode; @@ -218,15 +232,84 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, input_report_key(input_dev, keycode[i], buf[i / 8] & (1 << (i % 8))); - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER) || - dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2)) + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]); + break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + /* rotary encoders */ + input_report_abs(dev->input_dev, ABS_X, buf[5] & 0xf); + input_report_abs(dev->input_dev, ABS_Y, buf[5] >> 4); + input_report_abs(dev->input_dev, ABS_Z, buf[6] & 0xf); + input_report_abs(dev->input_dev, ABS_MISC, buf[6] >> 4); + break; + } input_sync(input_dev); } +static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) +{ + struct snd_usb_caiaqdev *dev = urb->context; + unsigned char *buf = urb->transfer_buffer; + int ret; + + if (urb->status || !dev || urb != dev->ep4_in_urb) + return; + + if (urb->actual_length < 24) + goto requeue; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (buf[0] & 0x3) + snd_caiaq_input_read_io(dev, buf + 1, 7); + + if (buf[0] & 0x4) + snd_caiaq_input_read_analog(dev, buf + 8, 16); + + break; + } + +requeue: + dev->ep4_in_urb->actual_length = 0; + ret = usb_submit_urb(dev->ep4_in_urb, GFP_ATOMIC); + if (ret < 0) + log("unable to submit urb. OOM!?\n"); +} + +static int snd_usb_caiaq_input_open(struct input_dev *idev) +{ + struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + + if (!dev) + return -EINVAL; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) + return -EIO; + break; + } + + return 0; +} + +static void snd_usb_caiaq_input_close(struct input_dev *idev) +{ + struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + + if (!dev) + return; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + usb_kill_urb(dev->ep4_in_urb); + break; + } +} + void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, char *buf, unsigned int len) @@ -251,7 +334,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) { struct usb_device *usb_dev = dev->chip.dev; struct input_dev *input; - int i, ret; + int i, ret = 0; input = input_allocate_device(); if (!input) @@ -265,7 +348,9 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) usb_to_input_id(usb_dev, &input->id); input->dev.parent = &usb_dev->dev; - switch (dev->chip.usb_id) { + input_set_drvdata(input, dev); + + switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | @@ -326,25 +411,72 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_MISC, 0, 255, 0, 1); snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | + BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | + BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) | + BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) | + BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | + BIT_MASK(ABS_Z); + input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC); + BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLX1_INPUTS); + for (i = 0; i < KONTROLX1_INPUTS; i++) + dev->keycode[i] = BTN_MISC + i; + input->keycodemax = KONTROLX1_INPUTS; + + /* analog potentiometers */ + input_set_abs_params(input, ABS_HAT0X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT0Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT1X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT1Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT2X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT2Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT3X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT3Y, 0, 4096, 0, 10); + + /* rotary encoders */ + input_set_abs_params(input, ABS_X, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_Y, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_Z, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_MISC, 0, 0xf, 0, 1); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + + break; default: /* no input methods supported on this device */ - input_free_device(input); - return 0; + goto exit_free_idev; } + input->open = snd_usb_caiaq_input_open; + input->close = snd_usb_caiaq_input_close; input->keycode = dev->keycode; input->keycodesize = sizeof(unsigned short); for (i = 0; i < input->keycodemax; i++) __set_bit(dev->keycode[i], input->keybit); ret = input_register_device(input); - if (ret < 0) { - input_free_device(input); - return ret; - } + if (ret < 0) + goto exit_free_idev; dev->input_dev = input; return 0; + +exit_free_idev: + input_free_device(input); + return ret; } void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) @@ -352,6 +484,10 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) if (!dev || !dev->input_dev) return; + usb_kill_urb(dev->ep4_in_urb); + usb_free_urb(dev->ep4_in_urb); + dev->ep4_in_urb = NULL; + input_unregister_device(dev->input_dev); dev->input_dev = NULL; } diff --git a/sound/usb/card.c b/sound/usb/card.c new file mode 100644 index 000000000000..da1346bd4856 --- /dev/null +++ b/sound/usb/card.c @@ -0,0 +1,652 @@ +/* + * (Tentative) USB Audio Driver for ALSA + * + * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan@lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer@ife.ee.ethz.ch) + * + * + * 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 + * + * + * NOTES: + * + * - async unlink should be used for avoiding the sleep inside lock. + * 2.4.22 usb-uhci seems buggy for async unlinking and results in + * oops. in such a cse, pass async_unlink=0 option. + * - the linked URBs would be preferred but not used so far because of + * the instability of unlinking. + * - type II is not supported properly. there is no device which supports + * this type *correctly*. SB extigy looks as if it supports, but it's + * indeed an AC3 stream packed in SPDIF frames (i.e. no real AC3 stream). + */ + + +#include <linux/bitops.h> +#include <linux/init.h> +#include <linux/list.h> +#include <linux/slab.h> +#include <linux/string.h> +#include <linux/usb.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#include "usbaudio.h" +#include "card.h" +#include "midi.h" +#include "mixer.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "helper.h" +#include "debug.h" +#include "pcm.h" +#include "urb.h" +#include "format.h" + +MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); +MODULE_DESCRIPTION("USB Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}"); + + +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 */ +/* Vendor/product IDs for this card */ +static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; +static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; +static int nrpacks = 8; /* max. number of packets per urb */ +static int async_unlink = 1; +static int device_setup[SNDRV_CARDS]; /* device parameter for this card */ +static int ignore_ctl_error; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for the USB audio adapter."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable USB audio adapter."); +module_param_array(vid, int, NULL, 0444); +MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); +module_param_array(pid, int, NULL, 0444); +MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); +module_param(nrpacks, int, 0644); +MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); +module_param(async_unlink, bool, 0444); +MODULE_PARM_DESC(async_unlink, "Use async unlink mode."); +module_param_array(device_setup, int, NULL, 0444); +MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); +module_param(ignore_ctl_error, bool, 0444); +MODULE_PARM_DESC(ignore_ctl_error, + "Ignore errors from USB controller for mixer interfaces."); + +/* + * we keep the snd_usb_audio_t instances by ourselves for merging + * the all interfaces on the same card as one sound device. + */ + +static DEFINE_MUTEX(register_mutex); +static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; +static struct usb_driver usb_audio_driver; + +/* + * disconnect streams + * called from snd_usb_audio_disconnect() + */ +static void snd_usb_stream_disconnect(struct list_head *head) +{ + int idx; + struct snd_usb_stream *as; + struct snd_usb_substream *subs; + + as = list_entry(head, struct snd_usb_stream, list); + for (idx = 0; idx < 2; idx++) { + subs = &as->substream[idx]; + if (!subs->num_formats) + return; + snd_usb_release_substream_urbs(subs, 1); + subs->interface = -1; + } +} + +static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface = usb_ifnum_to_if(dev, interface); + + if (!iface) { + snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + if (usb_interface_claimed(iface)) { + snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { + snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + + return 0; + } + + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { + snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", + dev->devnum, ctrlif, interface, altsd->bInterfaceClass); + /* skip non-supported classes */ + return -EINVAL; + } + + if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { + snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + return -EINVAL; + } + + if (! snd_usb_parse_audio_endpoints(chip, interface)) { + usb_set_interface(dev, interface, 0); /* reset the current interface */ + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return -EINVAL; + } + + return 0; +} + +/* + * parse audio control descriptor and create pcm/midi streams + */ +static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *host_iface; + struct usb_interface_descriptor *altsd; + void *control_header; + int i, protocol; + + /* find audiocontrol interface */ + host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; + control_header = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, UAC_HEADER); + altsd = get_iface_desc(host_iface); + protocol = altsd->bInterfaceProtocol; + + if (!control_header) { + snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); + return -EINVAL; + } + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_ac_header_descriptor_v1 *h1 = control_header; + + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); + return -EINVAL; + } + + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); + return -EINVAL; + } + + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + + break; + } + + case UAC_VERSION_2: { + struct uac_clock_source_descriptor *cs; + struct usb_interface_assoc_descriptor *assoc = + usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + + if (!assoc) { + snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); + return -EINVAL; + } + + /* FIXME: for now, we expect there is at least one clock source + * descriptor and we always take the first one. + * We should properly support devices with multiple clock sources, + * clock selectors and sample rate conversion units. */ + + cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, + NULL, UAC2_CLOCK_SOURCE); + + if (!cs) { + snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); + return -EINVAL; + } + + chip->clock_id = cs->bClockID; + + for (i = 0; i < assoc->bInterfaceCount; i++) { + int intf = assoc->bFirstInterface + i; + + if (intf != ctrlif) + snd_usb_create_stream(chip, ctrlif, intf); + } + + break; + } + + default: + snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); + return -EINVAL; + } + + return 0; +} + +/* + * free the chip instance + * + * here we have to do not much, since pcm and controls are already freed + * + */ + +static int snd_usb_audio_free(struct snd_usb_audio *chip) +{ + kfree(chip); + return 0; +} + +static int snd_usb_audio_dev_free(struct snd_device *device) +{ + struct snd_usb_audio *chip = device->device_data; + return snd_usb_audio_free(chip); +} + + +/* + * create a chip instance and set its names. + */ +static int snd_usb_audio_create(struct usb_device *dev, int idx, + const struct snd_usb_audio_quirk *quirk, + struct snd_usb_audio **rchip) +{ + struct snd_card *card; + struct snd_usb_audio *chip; + int err, len; + char component[14]; + static struct snd_device_ops ops = { + .dev_free = snd_usb_audio_dev_free, + }; + + *rchip = NULL; + + if (snd_usb_get_speed(dev) != USB_SPEED_LOW && + snd_usb_get_speed(dev) != USB_SPEED_FULL && + snd_usb_get_speed(dev) != USB_SPEED_HIGH) { + snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev)); + return -ENXIO; + } + + err = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "cannot create card instance %d\n", idx); + return err; + } + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (! chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->index = idx; + chip->dev = dev; + chip->card = card; + chip->setup = device_setup[idx]; + chip->nrpacks = nrpacks; + chip->async_unlink = async_unlink; + + chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)); + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->midi_list); + INIT_LIST_HEAD(&chip->mixer_list); + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + snd_usb_audio_free(chip); + snd_card_free(card); + return err; + } + + strcpy(card->driver, "USB-Audio"); + sprintf(component, "USB%04x:%04x", + USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); + snd_component_add(card, component); + + /* retrieve the device string as shortname */ + if (quirk && quirk->product_name) { + strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname)); + } else { + if (!dev->descriptor.iProduct || + usb_string(dev, dev->descriptor.iProduct, + card->shortname, sizeof(card->shortname)) <= 0) { + /* no name available from anywhere, so use ID */ + sprintf(card->shortname, "USB Device %#04x:%#04x", + USB_ID_VENDOR(chip->usb_id), + USB_ID_PRODUCT(chip->usb_id)); + } + } + + /* retrieve the vendor and device strings as longname */ + if (quirk && quirk->vendor_name) { + len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname)); + } else { + if (dev->descriptor.iManufacturer) + len = usb_string(dev, dev->descriptor.iManufacturer, + card->longname, sizeof(card->longname)); + else + len = 0; + /* we don't really care if there isn't any vendor string */ + } + if (len > 0) + strlcat(card->longname, " ", sizeof(card->longname)); + + strlcat(card->longname, card->shortname, sizeof(card->longname)); + + len = strlcat(card->longname, " at ", sizeof(card->longname)); + + if (len < sizeof(card->longname)) + usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); + + strlcat(card->longname, + snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" : + snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" : + ", high speed", + sizeof(card->longname)); + + snd_usb_audio_create_proc(chip); + + *rchip = chip; + return 0; +} + +/* + * probe the active usb device + * + * note that this can be called multiple times per a device, when it + * includes multiple audio control interfaces. + * + * thus we check the usb device pointer and creates the card instance + * only at the first time. the successive calls of this function will + * append the pcm interface to the corresponding card. + */ +static void *snd_usb_audio_probe(struct usb_device *dev, + struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct snd_usb_audio_quirk *quirk = (const struct snd_usb_audio_quirk *)usb_id->driver_info; + int i, err; + struct snd_usb_audio *chip; + struct usb_host_interface *alts; + int ifnum; + u32 id; + + alts = &intf->altsetting[0]; + ifnum = get_iface_desc(alts)->bInterfaceNumber; + id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)); + if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) + goto __err_val; + + if (snd_usb_apply_boot_quirk(dev, intf, quirk) < 0) + goto __err_val; + + /* + * found a config. now register to ALSA + */ + + /* check whether it's already registered */ + chip = NULL; + mutex_lock(®ister_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (usb_chip[i] && usb_chip[i]->dev == dev) { + if (usb_chip[i]->shutdown) { + snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); + goto __error; + } + chip = usb_chip[i]; + break; + } + } + if (! chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && ! usb_chip[i] && + (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && + (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { + if (snd_usb_audio_create(dev, i, quirk, &chip) < 0) { + goto __error; + } + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + printk(KERN_ERR "no available usb audio device\n"); + goto __error; + } + } + + chip->txfr_quirk = 0; + err = 1; /* continue */ + if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { + /* need some special handlings */ + if ((err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk)) < 0) + goto __error; + } + + if (err > 0) { + /* create normal USB audio interfaces */ + if (snd_usb_create_streams(chip, ifnum) < 0 || + snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { + goto __error; + } + } + + /* we are allowed to call snd_card_register() many times */ + if (snd_card_register(chip->card) < 0) { + goto __error; + } + + usb_chip[chip->index] = chip; + chip->num_interfaces++; + mutex_unlock(®ister_mutex); + return chip; + + __error: + if (chip && !chip->num_interfaces) + snd_card_free(chip->card); + mutex_unlock(®ister_mutex); + __err_val: + return NULL; +} + +/* + * we need to take care of counter, since disconnection can be called also + * many times as well as usb_audio_probe(). + */ +static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) +{ + struct snd_usb_audio *chip; + struct snd_card *card; + struct list_head *p; + + if (ptr == (void *)-1L) + return; + + chip = ptr; + card = chip->card; + mutex_lock(®ister_mutex); + chip->shutdown = 1; + chip->num_interfaces--; + if (chip->num_interfaces <= 0) { + snd_card_disconnect(card); + /* release the pcm resources */ + list_for_each(p, &chip->pcm_list) { + snd_usb_stream_disconnect(p); + } + /* release the midi resources */ + list_for_each(p, &chip->midi_list) { + snd_usbmidi_disconnect(p); + } + /* release mixer resources */ + list_for_each(p, &chip->mixer_list) { + snd_usb_mixer_disconnect(p); + } + usb_chip[chip->index] = NULL; + mutex_unlock(®ister_mutex); + snd_card_free_when_closed(card); + } else { + mutex_unlock(®ister_mutex); + } +} + +/* + * new 2.5 USB kernel API + */ +static int usb_audio_probe(struct usb_interface *intf, + const struct usb_device_id *id) +{ + void *chip; + chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id); + if (chip) { + usb_set_intfdata(intf, chip); + return 0; + } else + return -EIO; +} + +static void usb_audio_disconnect(struct usb_interface *intf) +{ + snd_usb_audio_disconnect(interface_to_usbdev(intf), + usb_get_intfdata(intf)); +} + +#ifdef CONFIG_PM +static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) +{ + struct snd_usb_audio *chip = usb_get_intfdata(intf); + struct list_head *p; + struct snd_usb_stream *as; + + if (chip == (void *)-1L) + return 0; + + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + if (!chip->num_suspended_intf++) { + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + snd_pcm_suspend_all(as->pcm); + } + } + + return 0; +} + +static int usb_audio_resume(struct usb_interface *intf) +{ + struct snd_usb_audio *chip = usb_get_intfdata(intf); + + if (chip == (void *)-1L) + return 0; + if (--chip->num_suspended_intf) + return 0; + /* + * ALSA leaves material resumption to user space + * we just notify + */ + + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + + return 0; +} +#else +#define usb_audio_suspend NULL +#define usb_audio_resume NULL +#endif /* CONFIG_PM */ + +static struct usb_device_id usb_audio_ids [] = { +#include "quirks-table.h" + { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, + { } /* Terminating entry */ +}; + +MODULE_DEVICE_TABLE (usb, usb_audio_ids); + +/* + * entry point for linux usb interface + */ + +static struct usb_driver usb_audio_driver = { + .name = "snd-usb-audio", + .probe = usb_audio_probe, + .disconnect = usb_audio_disconnect, + .suspend = usb_audio_suspend, + .resume = usb_audio_resume, + .id_table = usb_audio_ids, +}; + +static int __init snd_usb_audio_init(void) +{ + if (nrpacks < 1 || nrpacks > MAX_PACKS) { + printk(KERN_WARNING "invalid nrpacks value.\n"); + return -EINVAL; + } + return usb_register(&usb_audio_driver); +} + +static void __exit snd_usb_audio_cleanup(void) +{ + usb_deregister(&usb_audio_driver); +} + +module_init(snd_usb_audio_init); +module_exit(snd_usb_audio_cleanup); diff --git a/sound/usb/card.h b/sound/usb/card.h new file mode 100644 index 000000000000..ed92420c1095 --- /dev/null +++ b/sound/usb/card.h @@ -0,0 +1,105 @@ +#ifndef __USBAUDIO_CARD_H +#define __USBAUDIO_CARD_H + +#define MAX_PACKS 20 +#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ +#define MAX_URBS 8 +#define SYNC_URBS 4 /* always four urbs for sync */ +#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ + +struct audioformat { + struct list_head list; + u64 formats; /* ALSA format bits */ + unsigned int channels; /* # channels */ + unsigned int fmt_type; /* USB audio format type (1-3) */ + unsigned int frame_size; /* samples per frame for non-audio */ + int iface; /* interface number */ + unsigned char altsetting; /* corresponding alternate setting */ + unsigned char altset_idx; /* array index of altenate setting */ + unsigned char attributes; /* corresponding attributes of cs endpoint */ + unsigned char endpoint; /* endpoint */ + unsigned char ep_attr; /* endpoint attributes */ + unsigned char datainterval; /* log_2 of data packet interval */ + unsigned int maxpacksize; /* max. packet size */ + unsigned int rates; /* rate bitmasks */ + unsigned int rate_min, rate_max; /* min/max rates */ + unsigned int nr_rates; /* number of rate table entries */ + unsigned int *rate_table; /* rate table */ +}; + +struct snd_usb_substream; + +struct snd_urb_ctx { + struct urb *urb; + unsigned int buffer_size; /* size of data buffer, if data URB */ + struct snd_usb_substream *subs; + int index; /* index for urb array */ + int packets; /* number of packets per urb */ +}; + +struct snd_urb_ops { + int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); + int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); + int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); + int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); +}; + +struct snd_usb_substream { + struct snd_usb_stream *stream; + struct usb_device *dev; + struct snd_pcm_substream *pcm_substream; + int direction; /* playback or capture */ + int interface; /* current interface */ + int endpoint; /* assigned endpoint */ + struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ + unsigned int cur_rate; /* current rate (for hw_params callback) */ + unsigned int period_bytes; /* current period bytes (for hw_params callback) */ + unsigned int altset_idx; /* USB data format: index of alternate setting */ + unsigned int datapipe; /* the data i/o pipe */ + unsigned int syncpipe; /* 1 - async out or adaptive in */ + unsigned int datainterval; /* log_2 of data packet interval */ + unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ + unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ + unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + unsigned int freqmax; /* maximum sampling rate, used for buffer management */ + unsigned int phase; /* phase accumulator */ + unsigned int maxpacksize; /* max packet size in bytes */ + unsigned int maxframesize; /* max packet size in frames */ + unsigned int curpacksize; /* current packet size in bytes (for capture) */ + unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int fill_max: 1; /* fill max packet size always */ + unsigned int txfr_quirk:1; /* allow sub-frame alignment */ + unsigned int fmt_type; /* USB audio format type (1-3) */ + + unsigned int running: 1; /* running status */ + + unsigned int hwptr_done; /* processed byte position in the buffer */ + unsigned int transfer_done; /* processed frames since last period update */ + unsigned long active_mask; /* bitmask of active urbs */ + unsigned long unlink_mask; /* bitmask of unlinked urbs */ + + unsigned int nurbs; /* # urbs */ + struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */ + struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ + char *syncbuf; /* sync buffer for all sync URBs */ + dma_addr_t sync_dma; /* DMA address of syncbuf */ + + u64 formats; /* format bitmasks (all or'ed) */ + unsigned int num_formats; /* number of supported audio formats (list) */ + struct list_head fmt_list; /* format list */ + struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ + spinlock_t lock; + + struct snd_urb_ops ops; /* callbacks (must be filled at init) */ +}; + +struct snd_usb_stream { + struct snd_usb_audio *chip; + struct snd_pcm *pcm; + int pcm_index; + unsigned int fmt_type; /* USB audio format type (1-3) */ + struct snd_usb_substream substream[2]; + struct list_head list; +}; + +#endif /* __USBAUDIO_CARD_H */ diff --git a/sound/usb/debug.h b/sound/usb/debug.h new file mode 100644 index 000000000000..343ec2d9ee66 --- /dev/null +++ b/sound/usb/debug.h @@ -0,0 +1,15 @@ +#ifndef __USBAUDIO_DEBUG_H +#define __USBAUDIO_DEBUG_H + +/* + * h/w constraints + */ + +#ifdef HW_CONST_DEBUG +#define hwc_debug(fmt, args...) printk(KERN_DEBUG fmt, ##args) +#else +#define hwc_debug(fmt, args...) /**/ +#endif + +#endif /* __USBAUDIO_DEBUG_H */ + diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c new file mode 100644 index 000000000000..b1309cdc1ac0 --- /dev/null +++ b/sound/usb/endpoint.c @@ -0,0 +1,359 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "card.h" +#include "proc.h" +#include "quirks.h" +#include "endpoint.h" +#include "urb.h" +#include "pcm.h" +#include "helper.h" +#include "format.h" + +/* + * free a substream + */ +static void free_substream(struct snd_usb_substream *subs) +{ + struct list_head *p, *n; + + if (!subs->num_formats) + return; /* not initialized */ + list_for_each_safe(p, n, &subs->fmt_list) { + struct audioformat *fp = list_entry(p, struct audioformat, list); + kfree(fp->rate_table); + kfree(fp); + } + kfree(subs->rate_list.list); +} + + +/* + * free a usb stream instance + */ +static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) +{ + free_substream(&stream->substream[0]); + free_substream(&stream->substream[1]); + list_del(&stream->list); + kfree(stream); +} + +static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) +{ + struct snd_usb_stream *stream = pcm->private_data; + if (stream) { + stream->pcm = NULL; + snd_usb_audio_stream_free(stream); + } +} + + +/* + * add this endpoint to the chip instance. + * if a stream with the same endpoint already exists, append to it. + * if not, create a new pcm stream. + */ +int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp) +{ + struct list_head *p; + struct snd_usb_stream *as; + struct snd_usb_substream *subs; + struct snd_pcm *pcm; + int err; + + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (!subs->endpoint) + continue; + if (subs->endpoint == fp->endpoint) { + list_add_tail(&fp->list, &subs->fmt_list); + subs->num_formats++; + subs->formats |= fp->formats; + return 0; + } + } + /* look for an empty stream */ + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + if (as->fmt_type != fp->fmt_type) + continue; + subs = &as->substream[stream]; + if (subs->endpoint) + continue; + err = snd_pcm_new_stream(as->pcm, stream, 1); + if (err < 0) + return err; + snd_usb_init_substream(as, stream, fp); + return 0; + } + + /* create a new pcm */ + as = kzalloc(sizeof(*as), GFP_KERNEL); + if (!as) + return -ENOMEM; + as->pcm_index = chip->pcm_devs; + as->chip = chip; + as->fmt_type = fp->fmt_type; + err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, + stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, + &pcm); + if (err < 0) { + kfree(as); + return err; + } + as->pcm = pcm; + pcm->private_data = as; + pcm->private_free = snd_usb_audio_pcm_free; + pcm->info_flags = 0; + if (chip->pcm_devs > 0) + sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); + else + strcpy(pcm->name, "USB Audio"); + + snd_usb_init_substream(as, stream, fp); + + list_add(&as->list, &chip->pcm_list); + chip->pcm_devs++; + + snd_usb_proc_pcm_format_add(as); + + return 0; +} + +int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) +{ + struct usb_device *dev; + struct usb_interface *iface; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + int i, altno, err, stream; + int format = 0, num_channels = 0; + struct audioformat *fp = NULL; + unsigned char *fmt, *csep; + int num, protocol; + + dev = chip->dev; + + /* parse the interface's altsettings */ + iface = usb_ifnum_to_if(dev, iface_no); + + num = iface->num_altsetting; + + /* + * Dallas DS4201 workaround: It presents 5 altsettings, but the last + * one misses syncpipe, and does not produce any sound. + */ + if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + num = 4; + + for (i = 0; i < num; i++) { + alts = &iface->altsetting[i]; + altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; + /* skip invalid one */ + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && + altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || + altsd->bNumEndpoints < 1 || + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) + continue; + /* must be isochronous */ + if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != + USB_ENDPOINT_XFER_ISOC) + continue; + /* check direction */ + stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + altno = altsd->bAlternateSetting; + + if (snd_usb_apply_interface_quirk(chip, iface_no, altno)) + continue; + + /* get audio formats */ + switch (protocol) { + case UAC_VERSION_1: { + struct uac_as_header_descriptor_v1 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; + } + + case UAC_VERSION_2: { + struct uac_as_header_descriptor_v2 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + break; + } + + default: + snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", + dev->devnum, iface_no, altno, protocol); + continue; + } + + /* get format type */ + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); + if (!fmt) { + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || + ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", + dev->devnum, iface_no, altno); + continue; + } + + /* + * Blue Microphones workaround: The last altsetting is identical + * with the previous one, except for a larger packet size, but + * is actually a mislabeled two-channel setting; ignore it. + */ + if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && + fp && fp->altsetting == 1 && fp->channels == 1 && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && + protocol == UAC_VERSION_1 && + le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == + fp->maxpacksize * 2) + continue; + + csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); + /* Creamware Noah has this descriptor after the 2nd endpoint */ + if (!csep && altsd->bNumEndpoints >= 2) + csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); + if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { + snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" + " class specific endpoint descriptor\n", + dev->devnum, iface_no, altno); + csep = NULL; + } + + fp = kzalloc(sizeof(*fp), GFP_KERNEL); + if (! fp) { + snd_printk(KERN_ERR "cannot malloc\n"); + return -ENOMEM; + } + + fp->iface = iface_no; + fp->altsetting = altno; + fp->altset_idx = i; + fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; + fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; + if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) + fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) + * (fp->maxpacksize & 0x7ff); + fp->attributes = csep ? csep[3] : 0; + + /* some quirks for attributes here */ + + switch (chip->usb_id) { + case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ + /* Optoplay sets the sample rate attribute although + * it seems not supporting it in fact. + */ + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + /* doesn't set the sample rate attribute, but supports it */ + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; + break; + case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ + case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is + an older model 77d:223) */ + /* + * plantronics headset and Griffin iMic have set adaptive-in + * although it's really not... + */ + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; + else + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; + break; + } + + /* ok, let's parse further... */ + if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { + kfree(fp->rate_table); + kfree(fp); + continue; + } + + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); + err = snd_usb_add_audio_endpoint(chip, stream, fp); + if (err < 0) { + kfree(fp->rate_table); + kfree(fp); + return err; + } + /* try to set the interface... */ + usb_set_interface(chip->dev, iface_no, altno); + snd_usb_init_pitch(chip, iface_no, alts, fp); + snd_usb_init_sample_rate(chip, iface_no, alts, fp, fp->rate_max); + } + return 0; +} + diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h new file mode 100644 index 000000000000..64dd0db023b2 --- /dev/null +++ b/sound/usb/endpoint.h @@ -0,0 +1,11 @@ +#ifndef __USBAUDIO_ENDPOINT_H +#define __USBAUDIO_ENDPOINT_H + +int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, + int iface_no); + +int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, + int stream, + struct audioformat *fp); + +#endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c new file mode 100644 index 000000000000..fcadedd94543 --- /dev/null +++ b/sound/usb/format.c @@ -0,0 +1,431 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "card.h" +#include "quirks.h" +#include "helper.h" +#include "debug.h" + +/* + * parse the audio format type I descriptor + * and returns the corresponding pcm format + * + * @dev: usb device + * @fp: audioformat record + * @format: the format tag (wFormatTag) + * @fmt: the format type descriptor + */ +static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + int protocol) +{ + int sample_width, sample_bytes; + u64 pcm_formats; + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + format = 1 << format; + break; + } + + case UAC_VERSION_2: { + struct uac_format_type_i_ext_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubslotSize; + format <<= 1; + break; + } + + default: + return -EINVAL; + } + + pcm_formats = 0; + + if (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED)) { + /* some devices don't define this correctly... */ + snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", + chip->dev->devnum, fp->iface, fp->altsetting); + format = 1 << UAC_FORMAT_TYPE_I_PCM; + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { + if (sample_width > sample_bytes * 8) { + snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); + } + /* check the format byte size */ + switch (sample_bytes) { + case 1: + pcm_formats |= SNDRV_PCM_FMTBIT_S8; + break; + case 2: + if (snd_usb_is_big_endian_format(chip, fp)) + pcm_formats |= SNDRV_PCM_FMTBIT_S16_BE; /* grrr, big endian!! */ + else + pcm_formats |= SNDRV_PCM_FMTBIT_S16_LE; + break; + case 3: + if (snd_usb_is_big_endian_format(chip, fp)) + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3BE; /* grrr, big endian!! */ + else + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3LE; + break; + case 4: + pcm_formats |= SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); + break; + } + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM8)) { + /* Dallas DS4201 workaround: it advertises U8 format, but really + supports S8. */ + if (chip->usb_id == USB_ID(0x04fa, 0x4201)) + pcm_formats |= SNDRV_PCM_FMTBIT_S8; + else + pcm_formats |= SNDRV_PCM_FMTBIT_U8; + } + if (format & (1 << UAC_FORMAT_TYPE_I_IEEE_FLOAT)) { + pcm_formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; + } + if (format & (1 << UAC_FORMAT_TYPE_I_ALAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_A_LAW; + } + if (format & (1 << UAC_FORMAT_TYPE_I_MULAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_MU_LAW; + } + if (format & ~0x3f) { + snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n", + chip->dev->devnum, fp->iface, fp->altsetting, format); + } + return pcm_formats; +} + + +/* + * parse the format descriptor and stores the possible sample rates + * on the audioformat table (audio class v1). + * + * @dev: usb device + * @fp: audioformat record + * @fmt: the format descriptor + * @offset: the start offset of descriptor pointing the rate type + * (7 for type I and II, 8 for type II) + */ +static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp, + unsigned char *fmt, int offset) +{ + int nr_rates = fmt[offset]; + + if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", + chip->dev->devnum, fp->iface, fp->altsetting); + return -1; + } + + if (nr_rates) { + /* + * build the rate table and bitmap flags + */ + int r, idx; + + fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + if (fp->rate_table == NULL) { + snd_printk(KERN_ERR "cannot malloc\n"); + return -1; + } + + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; + for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { + unsigned int rate = combine_triple(&fmt[idx]); + if (!rate) + continue; + /* C-Media CM6501 mislabels its 96 kHz altsetting */ + if (rate == 48000 && nr_rates == 1 && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && + fp->altsetting == 5 && fp->maxpacksize == 392) + rate = 96000; + /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ + if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) + rate = 8000; + + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; + } + if (!fp->nr_rates) { + hwc_debug("All rates were zero. Skipping format!\n"); + return -1; + } + } else { + /* continuous rates */ + fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + fp->rate_min = combine_triple(&fmt[offset + 1]); + fp->rate_max = combine_triple(&fmt[offset + 4]); + } + return 0; +} + +/* + * parse the format descriptor and stores the possible sample rates + * on the audioformat table (audio class v2). + */ +static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, + struct audioformat *fp, + struct usb_host_interface *iface) +{ + struct usb_device *dev = chip->dev; + unsigned char tmp[2], *data; + int i, nr_rates, data_size, ret = 0; + + /* get the number of sample rates first by only fetching 2 bytes */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + tmp, sizeof(tmp), 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + goto err; + } + + nr_rates = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_rates; + data = kzalloc(data_size, GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto err; + } + + /* now get the full information */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + data, data_size, 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + ret = -EINVAL; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_rates; i++) { + int rate = combine_quad(&data[2 + 12 * i]); + + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; + } + +err_free: + kfree(data); +err: + return ret; +} + +/* + * parse the format type I and III descriptors + */ +static int parse_audio_format_i(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + int protocol = altsd->bInterfaceProtocol; + int pcm_format, ret; + + if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { + /* FIXME: the format type is really IECxxx + * but we give normal PCM format to get the existing + * apps working... + */ + switch (chip->usb_id) { + + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + if (chip->setup == 0x00 && + fp->altsetting == 6) + pcm_format = SNDRV_PCM_FORMAT_S16_BE; + else + pcm_format = SNDRV_PCM_FORMAT_S16_LE; + break; + default: + pcm_format = SNDRV_PCM_FORMAT_S16_LE; + } + fp->formats = 1uLL << pcm_format; + } else { + fp->formats = parse_audio_format_i_type(chip, fp, format, + fmt, protocol); + if (!fp->formats) + return -1; + } + + /* gather possible sample rates */ + /* audio class v1 reports possible sample rates as part of the + * proprietary class specific descriptor. + * audio class v2 uses class specific EP0 range requests for that. + */ + switch (protocol) { + case UAC_VERSION_1: + fp->channels = fmt->bNrChannels; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + break; + case UAC_VERSION_2: + /* fp->channels is already set in this case */ + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } + + if (fp->channels < 1) { + snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", + chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); + return -1; + } + + return ret; +} + +/* + * parse the format type II descriptor + */ +static int parse_audio_format_ii(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) +{ + int brate, framesize, ret; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + int protocol = altsd->bInterfaceProtocol; + + switch (format) { + case UAC_FORMAT_TYPE_II_AC3: + /* FIXME: there is no AC3 format defined yet */ + // fp->formats = SNDRV_PCM_FMTBIT_AC3; + fp->formats = SNDRV_PCM_FMTBIT_U8; /* temporary hack to receive byte streams */ + break; + case UAC_FORMAT_TYPE_II_MPEG: + fp->formats = SNDRV_PCM_FMTBIT_MPEG; + break; + default: + snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", + chip->dev->devnum, fp->iface, fp->altsetting, format); + fp->formats = SNDRV_PCM_FMTBIT_MPEG; + break; + } + + fp->channels = 1; + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */ + break; + } + case UAC_VERSION_2: { + struct uac_format_type_ii_ext_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } + } + + return ret; +} + +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, + int format, unsigned char *fmt, int stream, + struct usb_host_interface *iface) +{ + int err; + + switch (fmt[3]) { + case UAC_FORMAT_TYPE_I: + case UAC_FORMAT_TYPE_III: + err = parse_audio_format_i(chip, fp, format, fmt, iface); + break; + case UAC_FORMAT_TYPE_II: + err = parse_audio_format_ii(chip, fp, format, fmt, iface); + break; + default: + snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", + chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + return -1; + } + fp->fmt_type = fmt[3]; + if (err < 0) + return err; +#if 1 + /* FIXME: temporary hack for extigy/audigy 2 nx/zs */ + /* extigy apparently supports sample rates other than 48k + * but not in ordinary way. so we enable only 48k atm. + */ + if (chip->usb_id == USB_ID(0x041e, 0x3000) || + chip->usb_id == USB_ID(0x041e, 0x3020) || + chip->usb_id == USB_ID(0x041e, 0x3061)) { + if (fmt[3] == UAC_FORMAT_TYPE_I && + fp->rates != SNDRV_PCM_RATE_48000 && + fp->rates != SNDRV_PCM_RATE_96000) + return -1; + } +#endif + return 0; +} + diff --git a/sound/usb/format.h b/sound/usb/format.h new file mode 100644 index 000000000000..8298c4e8ddfa --- /dev/null +++ b/sound/usb/format.h @@ -0,0 +1,8 @@ +#ifndef __USBAUDIO_FORMAT_H +#define __USBAUDIO_FORMAT_H + +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, + int format, unsigned char *fmt, int stream, + struct usb_host_interface *iface); + +#endif /* __USBAUDIO_FORMAT_H */ diff --git a/sound/usb/helper.c b/sound/usb/helper.c new file mode 100644 index 000000000000..ba7dba4d5cf2 --- /dev/null +++ b/sound/usb/helper.c @@ -0,0 +1,112 @@ +/* + * 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/usb.h> + +#include "usbaudio.h" +#include "helper.h" + +/* + * combine bytes and get an integer value + */ +unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size) +{ + switch (size) { + case 1: return *bytes; + case 2: return combine_word(bytes); + case 3: return combine_triple(bytes); + case 4: return combine_quad(bytes); + default: return 0; + } +} + +/* + * parse descriptor buffer and return the pointer starting the given + * descriptor type. + */ +void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype) +{ + u8 *p, *end, *next; + + p = descstart; + end = p + desclen; + for (; p < end;) { + if (p[0] < 2) + return NULL; + next = p + p[0]; + if (next > end) + return NULL; + if (p[1] == dtype && (!after || (void *)p > after)) { + return p; + } + p = next; + } + return NULL; +} + +/* + * find a class-specified interface descriptor with the given subtype. + */ +void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype) +{ + unsigned char *p = after; + + while ((p = snd_usb_find_desc(buffer, buflen, p, + USB_DT_CS_INTERFACE)) != NULL) { + if (p[0] >= 3 && p[2] == dsubtype) + return p; + } + return NULL; +} + +/* + * Wrapper for usb_control_msg(). + * Allocates a temp buffer to prevent dmaing from/to the stack. + */ +int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, + __u8 requesttype, __u16 value, __u16 index, void *data, + __u16 size, int timeout) +{ + int err; + void *buf = NULL; + + if (size > 0) { + buf = kmemdup(data, size, GFP_KERNEL); + if (!buf) + return -ENOMEM; + } + err = usb_control_msg(dev, pipe, request, requesttype, + value, index, buf, size, timeout); + if (size > 0) { + memcpy(data, buf, size); + kfree(buf); + } + return err; +} + +unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, + struct usb_host_interface *alts) +{ + if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && + get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + else + return 0; +} + diff --git a/sound/usb/helper.h b/sound/usb/helper.h new file mode 100644 index 000000000000..a6b0e51b3a9a --- /dev/null +++ b/sound/usb/helper.h @@ -0,0 +1,32 @@ +#ifndef __USBAUDIO_HELPER_H +#define __USBAUDIO_HELPER_H + +unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); + +void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); +void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); + +int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, + __u8 request, __u8 requesttype, __u16 value, __u16 index, + void *data, __u16 size, int timeout); + +unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, + struct usb_host_interface *alts); + +/* + * retrieve usb_interface descriptor from the host interface + * (conditional for compatibility with the older API) + */ +#ifndef get_iface_desc +#define get_iface_desc(iface) (&(iface)->desc) +#define get_endpoint(alt,ep) (&(alt)->endpoint[ep].desc) +#define get_ep_desc(ep) (&(ep)->desc) +#define get_cfg_desc(cfg) (&(cfg)->desc) +#endif + +#ifndef snd_usb_get_speed +#define snd_usb_get_speed(dev) ((dev)->speed) +#endif + + +#endif /* __USBAUDIO_HELPER_H */ diff --git a/sound/usb/usbmidi.c b/sound/usb/midi.c index 2c59afd99611..c6ee4a18e513 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/midi.c @@ -53,7 +53,8 @@ #include <sound/rawmidi.h> #include <sound/asequencer.h> #include "usbaudio.h" - +#include "midi.h" +#include "helper.h" /* * define this to log all USB packets diff --git a/sound/usb/midi.h b/sound/usb/midi.h new file mode 100644 index 000000000000..2089ec987c66 --- /dev/null +++ b/sound/usb/midi.h @@ -0,0 +1,48 @@ +#ifndef __USBMIDI_H +#define __USBMIDI_H + +/* maximum number of endpoints per interface */ +#define MIDI_MAX_ENDPOINTS 2 + +/* data for QUIRK_MIDI_FIXED_ENDPOINT */ +struct snd_usb_midi_endpoint_info { + int8_t out_ep; /* ep number, 0 autodetect */ + uint8_t out_interval; /* interval for interrupt endpoints */ + int8_t in_ep; + uint8_t in_interval; + uint16_t out_cables; /* bitmask */ + uint16_t in_cables; /* bitmask */ +}; + +/* for QUIRK_MIDI_YAMAHA, data is NULL */ + +/* for QUIRK_MIDI_MIDIMAN, data points to a snd_usb_midi_endpoint_info + * structure (out_cables and in_cables only) */ + +/* for QUIRK_COMPOSITE, data points to an array of snd_usb_audio_quirk + * structures, terminated with .ifnum = -1 */ + +/* for QUIRK_AUDIO_FIXED_ENDPOINT, data points to an audioformat structure */ + +/* for QUIRK_AUDIO/MIDI_STANDARD_INTERFACE, data is NULL */ + +/* for QUIRK_AUDIO_EDIROL_UA700_UA25/UA1000, data is NULL */ + +/* for QUIRK_IGNORE_INTERFACE, data is NULL */ + +/* for QUIRK_MIDI_NOVATION and _RAW, data is NULL */ + +/* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info + * structure (out_cables and in_cables only) */ + +/* for QUIRK_MIDI_CME, data is NULL */ + +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk); +void snd_usbmidi_input_stop(struct list_head* p); +void snd_usbmidi_input_start(struct list_head* p); +void snd_usbmidi_disconnect(struct list_head *p); + +#endif /* __USBMIDI_H */ diff --git a/sound/usb/misc/Makefile b/sound/usb/misc/Makefile new file mode 100644 index 000000000000..ccefd8158936 --- /dev/null +++ b/sound/usb/misc/Makefile @@ -0,0 +1,2 @@ +snd-ua101-objs := ua101.o +obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o diff --git a/sound/usb/ua101.c b/sound/usb/misc/ua101.c index 3d458d3b9962..796d8b25ee89 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/misc/ua101.c @@ -23,7 +23,8 @@ #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -#include "usbaudio.h" +#include "../usbaudio.h" +#include "../midi.h" MODULE_DESCRIPTION("Edirol UA-101/1000 driver"); MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); diff --git a/sound/usb/usbmixer.c b/sound/usb/mixer.c index 8e8f871b74ca..1deef623c081 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/mixer.c @@ -33,6 +33,7 @@ #include <linux/string.h> #include <linux/usb.h> #include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> #include <sound/core.h> #include <sound/control.h> @@ -41,60 +42,12 @@ #include <sound/tlv.h> #include "usbaudio.h" - -/* - */ - -/* ignore error from controls - for debugging */ -/* #define IGNORE_CTL_ERROR */ - -/* - * Sound Blaster remote control configuration - * - * format of remote control data: - * Extigy: xx 00 - * Audigy 2 NX: 06 80 xx 00 00 00 - * Live! 24-bit: 06 80 xx yy 22 83 - */ -static const struct rc_config { - u32 usb_id; - u8 offset; - u8 length; - u8 packet_length; - u8 min_packet_length; /* minimum accepted length of the URB result */ - u8 mute_mixer_id; - u32 mute_code; -} rc_configs[] = { - { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ - { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ - { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ - { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ -}; +#include "mixer.h" +#include "helper.h" +#include "mixer_quirks.h" #define MAX_ID_ELEMS 256 -struct usb_mixer_interface { - struct snd_usb_audio *chip; - unsigned int ctrlif; - struct list_head list; - unsigned int ignore_ctl_error; - struct urb *urb; - /* array[MAX_ID_ELEMS], indexed by unit id */ - struct usb_mixer_elem_info **id_elems; - - /* Sound Blaster remote control stuff */ - const struct rc_config *rc_cfg; - u32 rc_code; - wait_queue_head_t rc_waitq; - struct urb *rc_urb; - struct usb_ctrlrequest *rc_setup_packet; - u8 rc_buffer[6]; - - u8 audigy2nx_leds[3]; - u8 xonar_u1_status; -}; - - struct usb_audio_term { int id; int type; @@ -116,39 +69,6 @@ struct mixer_build { const struct usbmix_selector_map *selector_map; }; -#define MAX_CHANNELS 10 /* max logical channels */ - -struct usb_mixer_elem_info { - struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ - struct snd_ctl_elem_id *elem_id; - unsigned int id; - unsigned int control; /* CS or ICN (high byte) */ - unsigned int cmask; /* channel mask bitmap: 0 = master */ - int channels; - int val_type; - int min, max, res; - int dBmin, dBmax; - int cached; - int cache_val[MAX_CHANNELS]; - u8 initialized; -}; - - -enum { - USB_FEATURE_NONE = 0, - USB_FEATURE_MUTE = 1, - USB_FEATURE_VOLUME, - USB_FEATURE_BASS, - USB_FEATURE_MID, - USB_FEATURE_TREBLE, - USB_FEATURE_GEQ, - USB_FEATURE_AGC, - USB_FEATURE_DELAY, - USB_FEATURE_BASSBOOST, - USB_FEATURE_LOUDNESS -}; - enum { USB_MIXER_BOOLEAN, USB_MIXER_INV_BOOLEAN, @@ -213,7 +133,7 @@ enum { * if the mixer topology is too complicated and the parsed names are * ambiguous, add the entries in usbmixer_maps.c. */ -#include "usbmixer_maps.c" +#include "mixer_maps.c" static const struct usbmix_name_map * find_map(struct mixer_build *state, int unitid, int control) @@ -278,6 +198,7 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid, /* * find an audio control unit with the given unit id + * this doesn't return any clock related units, so they need to be handled elsewhere */ static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit) { @@ -286,7 +207,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit) + if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC2_EXTENSION_UNIT_V2 && p[3] == unit) return p; } return NULL; @@ -383,7 +304,7 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) * retrieve a mixer value */ -static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { unsigned char buf[2]; int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; @@ -405,6 +326,58 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali return -EINVAL; } +static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +{ + unsigned char buf[14]; /* enough space for one range of 4 bytes */ + unsigned char *val; + int ret; + __u8 bRequest; + + bRequest = (request == UAC_GET_CUR) ? + UAC2_CS_CUR : UAC2_CS_RANGE; + + ret = snd_usb_ctl_msg(cval->mixer->chip->dev, + usb_rcvctrlpipe(cval->mixer->chip->dev, 0), + bRequest, + USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, + validx, cval->mixer->ctrlif | (cval->id << 8), + buf, sizeof(buf), 1000); + + if (ret < 0) { + snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", + request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type); + return ret; + } + + switch (request) { + case UAC_GET_CUR: + val = buf; + break; + case UAC_GET_MIN: + val = buf + sizeof(__u16); + break; + case UAC_GET_MAX: + val = buf + sizeof(__u16) * 2; + break; + case UAC_GET_RES: + val = buf + sizeof(__u16) * 3; + break; + default: + return -EINVAL; + } + + *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(val, sizeof(__u16))); + + return 0; +} + +static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) +{ + return (cval->mixer->protocol == UAC_VERSION_1) ? + get_ctl_value_v1(cval, request, validx, value_ret) : + get_ctl_value_v2(cval, request, validx, value_ret); +} + static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) { return get_ctl_value(cval, UAC_GET_CUR, validx, value); @@ -429,8 +402,7 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, err = get_cur_mix_raw(cval, channel, value); if (err < 0) { if (!cval->mixer->ignore_ctl_error) - snd_printd(KERN_ERR "cannot get current value for " - "control %d ch %d: err = %d\n", + snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, channel, err); return err; } @@ -444,11 +416,26 @@ static int get_cur_mix_value(struct usb_mixer_elem_info *cval, * set a mixer value */ -static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) +int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, + int request, int validx, int value_set) { unsigned char buf[2]; - int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; - int timeout = 10; + int val_len, timeout = 10; + + if (cval->mixer->protocol == UAC_VERSION_1) { + val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; + } else { /* UAC_VERSION_2 */ + /* audio class v2 controls are always 2 bytes in size */ + val_len = sizeof(__u16); + + /* FIXME */ + if (request != UAC_SET_CUR) { + snd_printdd(KERN_WARNING "RANGE setting not yet supported\n"); + return -EINVAL; + } + + request = UAC2_CS_CUR; + } value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; @@ -468,14 +455,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) { - return set_ctl_value(cval, UAC_SET_CUR, validx, value); + return snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, validx, value); } static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; - err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, + err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) return err; @@ -644,46 +631,65 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm */ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term) { - unsigned char *p1; + void *p1; memset(term, 0, sizeof(*term)); while ((p1 = find_audio_control_unit(state, id)) != NULL) { + unsigned char *hdr = p1; term->id = id; - switch (p1[2]) { + switch (hdr[2]) { case UAC_INPUT_TERMINAL: - term->type = combine_word(p1 + 4); - term->channels = p1[7]; - term->chconfig = combine_word(p1 + 8); - term->name = p1[11]; + if (state->mixer->protocol == UAC_VERSION_1) { + struct uac_input_terminal_descriptor *d = p1; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le16_to_cpu(d->wChannelConfig); + term->name = d->iTerminal; + } else { /* UAC_VERSION_2 */ + struct uac2_input_terminal_descriptor *d = p1; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; + } return 0; - case UAC_FEATURE_UNIT: - id = p1[4]; + case UAC_FEATURE_UNIT: { + /* the header is the same for v1 and v2 */ + struct uac_feature_unit_descriptor *d = p1; + id = d->bUnitID; break; /* continue to parse */ - case UAC_MIXER_UNIT: - term->type = p1[2] << 16; /* virtual type */ - term->channels = p1[5 + p1[4]]; - term->chconfig = combine_word(p1 + 6 + p1[4]); - term->name = p1[p1[0] - 1]; + } + case UAC_MIXER_UNIT: { + struct uac_mixer_unit_descriptor *d = p1; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->channels = uac_mixer_unit_bNrChannels(d); + term->chconfig = uac_mixer_unit_wChannelConfig(d, state->mixer->protocol); + term->name = uac_mixer_unit_iMixer(d); return 0; - case UAC_SELECTOR_UNIT: + } + case UAC_SELECTOR_UNIT: { + struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, p1[5], term) < 0) + if (check_input_term(state, d->baSourceID[0], term) < 0) return -ENODEV; - term->type = p1[2] << 16; /* virtual type */ + term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; - term->name = p1[9 + p1[0] - 1]; + term->name = uac_selector_unit_iSelector(d); return 0; + } case UAC_PROCESSING_UNIT_V1: - case UAC_EXTENSION_UNIT_V1: - if (p1[6] == 1) { - id = p1[7]; + case UAC_EXTENSION_UNIT_V1: { + struct uac_processing_unit_descriptor *d = p1; + if (d->bNrInPins) { + id = d->baSourceID[0]; break; /* continue to parse */ } - term->type = p1[2] << 16; /* virtual type */ - term->channels = p1[7 + p1[6]]; - term->chconfig = combine_word(p1 + 8 + p1[6]); - term->name = p1[12 + p1[6] + p1[11 + p1[6]]]; + term->type = d->bDescriptorSubtype << 16; /* virtual type */ + term->channels = uac_processing_unit_bNrChannels(d); + term->chconfig = uac_processing_unit_wChannelConfig(d, state->mixer->protocol); + term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol); return 0; + } default: return -ENODEV; } @@ -764,7 +770,8 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) int last_valid_res = cval->res; while (cval->res > 1) { - if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) + if (snd_usb_mixer_set_ctl_value(cval, UAC_SET_RES, + (cval->control << 8) | minchn, cval->res / 2) < 0) break; cval->res /= 2; } @@ -929,6 +936,15 @@ static struct snd_kcontrol_new usb_feature_unit_ctl = { .put = mixer_ctl_feature_put, }; +/* the read-only variant */ +static struct snd_kcontrol_new usb_feature_unit_ctl_ro = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", /* will be filled later manually */ + .info = mixer_ctl_feature_info, + .get = mixer_ctl_feature_get, + .put = NULL, +}; + /* * build a feature control @@ -939,20 +955,22 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) return strlcat(kctl->id.name, str, sizeof(kctl->id.name)); } -static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, +static void build_feature_ctl(struct mixer_build *state, void *raw_desc, unsigned int ctl_mask, int control, - struct usb_audio_term *iterm, int unitid) + struct usb_audio_term *iterm, int unitid, + int read_only) { + struct uac_feature_unit_descriptor *desc = raw_desc; unsigned int len = 0; int mapped_name = 0; - int nameid = desc[desc[0] - 1]; + int nameid = uac_feature_unit_iFeature(desc); struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; const struct usbmix_name_map *map; control++; /* change from zero-based to 1-based value */ - if (control == USB_FEATURE_GEQ) { + if (control == UAC_GRAPHIC_EQUALIZER_CONTROL) { /* FIXME: not supported yet */ return; } @@ -984,7 +1002,11 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, /* get min/max values */ get_min_max(cval, 0); - kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + if (read_only) + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + else + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + if (! kctl) { snd_printk(KERN_ERR "cannot malloc kcontrol\n"); kfree(cval); @@ -999,8 +1021,8 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, kctl->id.name, sizeof(kctl->id.name)); switch (control) { - case USB_FEATURE_MUTE: - case USB_FEATURE_VOLUME: + case UAC_MUTE_CONTROL: + case UAC_VOLUME_CONTROL: /* determine the control name. the rule is: * - if a name id is given in descriptor, use it. * - if the connected input can be determined, then use the name @@ -1027,9 +1049,9 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, len = append_ctl_name(kctl, " Playback"); } } - append_ctl_name(kctl, control == USB_FEATURE_MUTE ? + append_ctl_name(kctl, control == UAC_MUTE_CONTROL ? " Switch" : " Volume"); - if (control == USB_FEATURE_VOLUME) { + if (control == UAC_VOLUME_CONTROL) { kctl->tlv.c = mixer_vol_tlv; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | @@ -1094,49 +1116,92 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void struct usb_audio_term iterm; unsigned int master_bits, first_ch_bits; int err, csize; - struct uac_feature_unit_descriptor *ftr = _ftr; + struct uac_feature_unit_descriptor *hdr = _ftr; + __u8 *bmaControls; + + if (state->mixer->protocol == UAC_VERSION_1) { + csize = hdr->bControlSize; + channels = (hdr->bLength - 7) / csize - 1; + bmaControls = hdr->bmaControls; + } else { + struct uac2_feature_unit_descriptor *ftr = _ftr; + csize = 4; + channels = (hdr->bLength - 6) / 4; + bmaControls = ftr->bmaControls; + } - if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { + if (hdr->bLength < 7 || !csize || hdr->bLength < 7 + csize) { snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } /* parse the source unit */ - if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0) + if ((err = parse_audio_unit(state, hdr->bSourceID)) < 0) return err; /* determine the input source type and name */ - if (check_input_term(state, ftr->bSourceID, &iterm) < 0) + if (check_input_term(state, hdr->bSourceID, &iterm) < 0) return -EINVAL; - channels = (ftr->bLength - 7) / csize - 1; - - master_bits = snd_usb_combine_bytes(ftr->controls, csize); + master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ switch (state->chip->usb_id) { case USB_ID(0x08bb, 0x2702): snd_printk(KERN_INFO "usbmixer: master volume quirk for PCM2702 chip\n"); /* disable non-functional volume control */ - master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1)); + master_bits &= ~UAC_FU_VOLUME; break; } if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize); + first_ch_bits = snd_usb_combine_bytes(bmaControls + csize, csize); else first_ch_bits = 0; - /* check all control types */ - for (i = 0; i < 10; i++) { - unsigned int ch_bits = 0; - for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize); - if (mask & (1 << i)) - ch_bits |= (1 << j); + + if (state->mixer->protocol == UAC_VERSION_1) { + /* check all control types */ + for (i = 0; i < 10; i++) { + unsigned int ch_bits = 0; + for (j = 0; j < channels; j++) { + unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + if (mask & (1 << i)) + ch_bits |= (1 << j); + } + /* audio class v1 controls are never read-only */ + if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, 0); + if (master_bits & (1 << i)) + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, 0); + } + } else { /* UAC_VERSION_2 */ + for (i = 0; i < 30/2; i++) { + /* From the USB Audio spec v2.0: + bmaControls() is a (ch+1)-element array of 4-byte bitmaps, + each containing a set of bit pairs. If a Control is present, + it must be Host readable. If a certain Control is not + present then the bit pair must be set to 0b00. + If a Control is present but read-only, the bit pair must be + set to 0b01. If a Control is also Host programmable, the bit + pair must be set to 0b11. The value 0b10 is not allowed. */ + unsigned int ch_bits = 0; + unsigned int ch_read_only = 0; + + for (j = 0; j < channels; j++) { + unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); + if (mask & (1 << (i * 2))) { + ch_bits |= (1 << j); + if (~mask & (1 << ((i * 2) + 1))) + ch_read_only |= (1 << j); + } + } + + /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ + if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); + if (master_bits & (1 << i * 2)) + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, + ~master_bits & (1 << ((i * 2) + 1))); } - if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid); - if (master_bits & (1 << i)) - build_feature_ctl(state, _ftr, 0, i, &iterm, unitid); } return 0; @@ -1154,13 +1219,13 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void * input channel number (zero based) is given in control field instead. */ -static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, +static void build_mixer_unit_ctl(struct mixer_build *state, + struct uac_mixer_unit_descriptor *desc, int in_pin, int in_ch, int unitid, struct usb_audio_term *iterm) { struct usb_mixer_elem_info *cval; - unsigned int input_pins = desc[4]; - unsigned int num_outs = desc[5 + input_pins]; + unsigned int num_outs = uac_mixer_unit_bNrChannels(desc); unsigned int i, len; struct snd_kcontrol *kctl; const struct usbmix_name_map *map; @@ -1178,7 +1243,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, cval->control = in_ch + 1; /* based on 1 */ cval->val_type = USB_MIXER_S16; for (i = 0; i < num_outs; i++) { - if (check_matrix_bitmap(desc + 9 + input_pins, in_ch, i, num_outs)) { + if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), in_ch, i, num_outs)) { cval->cmask |= (1 << i); cval->channels++; } @@ -1211,18 +1276,19 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, /* * parse a mixer unit */ -static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, unsigned char *desc) +static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, void *raw_desc) { + struct uac_mixer_unit_descriptor *desc = raw_desc; struct usb_audio_term iterm; int input_pins, num_ins, num_outs; int pin, ich, err; - if (desc[0] < 11 || ! (input_pins = desc[4]) || ! (num_outs = desc[5 + input_pins])) { + if (desc->bLength < 11 || ! (input_pins = desc->bNrInPins) || ! (num_outs = uac_mixer_unit_bNrChannels(desc))) { snd_printk(KERN_ERR "invalid MIXER UNIT descriptor %d\n", unitid); return -EINVAL; } /* no bmControls field (e.g. Maya44) -> ignore */ - if (desc[0] <= 10 + input_pins) { + if (desc->bLength <= 10 + input_pins) { snd_printdd(KERN_INFO "MU %d has no bmControls field\n", unitid); return 0; } @@ -1230,10 +1296,10 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, unsigne num_ins = 0; ich = 0; for (pin = 0; pin < input_pins; pin++) { - err = parse_audio_unit(state, desc[5 + pin]); + err = parse_audio_unit(state, desc->baSourceID[pin]); if (err < 0) return err; - err = check_input_term(state, desc[5 + pin], &iterm); + err = check_input_term(state, desc->baSourceID[pin], &iterm); if (err < 0) return err; num_ins += iterm.channels; @@ -1241,7 +1307,7 @@ static int parse_audio_mixer_unit(struct mixer_build *state, int unitid, unsigne int och, ich_has_controls = 0; for (och = 0; och < num_outs; ++och) { - if (check_matrix_bitmap(desc + 9 + input_pins, + if (check_matrix_bitmap(uac_mixer_unit_bmControls(desc, state->mixer->protocol), ich, och, num_outs)) { ich_has_controls = 1; break; @@ -1402,9 +1468,10 @@ static struct procunit_info extunits[] = { /* * build a processing/extension unit */ -static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned char *dsc, struct procunit_info *list, char *name) +static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw_desc, struct procunit_info *list, char *name) { - int num_ins = dsc[6]; + struct uac_processing_unit_descriptor *desc = raw_desc; + int num_ins = desc->bNrInPins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -1419,17 +1486,18 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned 0, NULL, default_value_info }; - if (dsc[0] < 13 || dsc[0] < 13 + num_ins || dsc[0] < num_ins + dsc[11 + num_ins]) { + if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { snd_printk(KERN_ERR "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; } for (i = 0; i < num_ins; i++) { - if ((err = parse_audio_unit(state, dsc[7 + i])) < 0) + if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) return err; } - type = combine_word(&dsc[4]); + type = le16_to_cpu(desc->wProcessType); for (info = list; info && info->type; info++) if (info->type == type) break; @@ -1437,8 +1505,9 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned info = &default_info; for (valinfo = info->values; valinfo->control; valinfo++) { - /* FIXME: bitmap might be longer than 8bit */ - if (! (dsc[12 + num_ins] & (1 << (valinfo->control - 1)))) + __u8 *controls = uac_processing_unit_bmControls(desc, state->mixer->protocol); + + if (! (controls[valinfo->control / 8] & (1 << ((valinfo->control % 8) - 1)))) continue; map = find_map(state, unitid, valinfo->control); if (check_ignored_ctl(map)) @@ -1456,9 +1525,10 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned /* get min/max values */ if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) { + __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol); /* FIXME: hard-coded */ cval->min = 1; - cval->max = dsc[15]; + cval->max = control_spec[0]; cval->res = 1; cval->initialized = 1; } else { @@ -1488,7 +1558,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned else if (info->name) strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); else { - nameid = dsc[12 + num_ins + dsc[11 + num_ins]]; + nameid = uac_processing_unit_iProcessing(desc, state->mixer->protocol); len = 0; if (nameid) len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); @@ -1507,14 +1577,16 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned } -static int parse_audio_processing_unit(struct mixer_build *state, int unitid, unsigned char *desc) +static int parse_audio_processing_unit(struct mixer_build *state, int unitid, void *raw_desc) { - return build_audio_procunit(state, unitid, desc, procunits, "Processing Unit"); + return build_audio_procunit(state, unitid, raw_desc, procunits, "Processing Unit"); } -static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc) +static int parse_audio_extension_unit(struct mixer_build *state, int unitid, void *raw_desc) { - return build_audio_procunit(state, unitid, desc, extunits, "Extension Unit"); + /* Note that we parse extension units with processing unit descriptors. + * That's ok as the layout is the same */ + return build_audio_procunit(state, unitid, raw_desc, extunits, "Extension Unit"); } @@ -1616,9 +1688,9 @@ static void usb_mixer_selector_elem_free(struct snd_kcontrol *kctl) /* * parse a selector unit */ -static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsigned char *desc) +static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void *raw_desc) { - unsigned int num_ins = desc[4]; + struct uac_selector_unit_descriptor *desc = raw_desc; unsigned int i, nameid, len; int err; struct usb_mixer_elem_info *cval; @@ -1626,17 +1698,17 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi const struct usbmix_name_map *map; char **namelist; - if (! num_ins || desc[0] < 5 + num_ins) { + if (!desc->bNrInPins || desc->bLength < 5 + desc->bNrInPins) { snd_printk(KERN_ERR "invalid SELECTOR UNIT descriptor %d\n", unitid); return -EINVAL; } - for (i = 0; i < num_ins; i++) { - if ((err = parse_audio_unit(state, desc[5 + i])) < 0) + for (i = 0; i < desc->bNrInPins; i++) { + if ((err = parse_audio_unit(state, desc->baSourceID[i])) < 0) return err; } - if (num_ins == 1) /* only one ? nonsense! */ + if (desc->bNrInPins == 1) /* only one ? nonsense! */ return 0; map = find_map(state, unitid, 0); @@ -1653,18 +1725,18 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi cval->val_type = USB_MIXER_U8; cval->channels = 1; cval->min = 1; - cval->max = num_ins; + cval->max = desc->bNrInPins; cval->res = 1; cval->initialized = 1; - namelist = kmalloc(sizeof(char *) * num_ins, GFP_KERNEL); + namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (! namelist) { snd_printk(KERN_ERR "cannot malloc\n"); kfree(cval); return -ENOMEM; } #define MAX_ITEM_NAME_LEN 64 - for (i = 0; i < num_ins; i++) { + for (i = 0; i < desc->bNrInPins; i++) { struct usb_audio_term iterm; len = 0; namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL); @@ -1678,7 +1750,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi } len = check_mapped_selector_name(state, unitid, i, namelist[i], MAX_ITEM_NAME_LEN); - if (! len && check_input_term(state, desc[5 + i], &iterm) >= 0) + if (! len && check_input_term(state, desc->baSourceID[i], &iterm) >= 0) len = get_term_name(state, &iterm, namelist[i], MAX_ITEM_NAME_LEN, 0); if (! len) sprintf(namelist[i], "Input %d", i); @@ -1694,7 +1766,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi kctl->private_value = (unsigned long)namelist; kctl->private_free = usb_mixer_selector_elem_free; - nameid = desc[desc[0] - 1]; + nameid = uac_selector_unit_iSelector(desc); len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (len) ; @@ -1713,7 +1785,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi } snd_printdd(KERN_INFO "[%d] SU [%s] items = %d\n", - cval->id, kctl->id.name, num_ins); + cval->id, kctl->id.name, desc->bNrInPins); if ((err = add_control_to_empty(state, kctl)) < 0) return err; @@ -1748,9 +1820,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); case UAC_PROCESSING_UNIT_V1: - return parse_audio_processing_unit(state, unitid, p1); + /* UAC2_EFFECT_UNIT has the same value */ + if (state->mixer->protocol == UAC_VERSION_1) + return parse_audio_processing_unit(state, unitid, p1); + else + return 0; /* FIXME - effect units not implemented yet */ case UAC_EXTENSION_UNIT_V1: - return parse_audio_extension_unit(state, unitid, p1); + /* UAC2_PROCESSING_UNIT_V2 has the same value */ + if (state->mixer->protocol == UAC_VERSION_1) + return parse_audio_extension_unit(state, unitid, p1); + else /* UAC_VERSION_2 */ + return parse_audio_processing_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; @@ -1783,11 +1863,11 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { - struct uac_output_terminal_descriptor_v1 *desc; struct mixer_build state; int err; const struct usbmix_ctl_map *map; struct usb_host_interface *hostif; + void *p; hostif = &usb_ifnum_to_if(mixer->chip->dev, mixer->ctrlif)->altsetting[0]; memset(&state, 0, sizeof(state)); @@ -1806,23 +1886,39 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } } - desc = NULL; - while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) { - if (desc->bLength < 9) - continue; /* invalid descriptor? */ - set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ - state.oterm.id = desc->bTerminalID; - state.oterm.type = le16_to_cpu(desc->wTerminalType); - state.oterm.name = desc->iTerminal; - err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) - return err; + p = NULL; + while ((p = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, p, UAC_OUTPUT_TERMINAL)) != NULL) { + if (mixer->protocol == UAC_VERSION_1) { + struct uac_output_terminal_descriptor_v1 *desc = p; + + if (desc->bLength < sizeof(*desc)) + continue; /* invalid descriptor? */ + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); + if (err < 0) + return err; + } else { /* UAC_VERSION_2 */ + struct uac2_output_terminal_descriptor *desc = p; + + if (desc->bLength < sizeof(*desc)) + continue; /* invalid descriptor? */ + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); + if (err < 0) + return err; + } } + return 0; } -static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, - int unitid) +void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid) { struct usb_mixer_elem_info *info; @@ -1871,34 +1967,6 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, } } -static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, - int unitid) -{ - if (!mixer->rc_cfg) - return; - /* unit ids specific to Extigy/Audigy 2 NX: */ - switch (unitid) { - case 0: /* remote control */ - mixer->rc_urb->dev = mixer->chip->dev; - usb_submit_urb(mixer->rc_urb, GFP_ATOMIC); - break; - case 4: /* digital in jack */ - case 7: /* line in jacks */ - case 19: /* speaker out jacks */ - case 20: /* headphones out jack */ - break; - /* live24ext: 4 = line-in jack */ - case 3: /* hp-out jack (may actuate Mute) */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) - snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); - break; - default: - snd_printd(KERN_DEBUG "memory change in unknown unit %d\n", unitid); - break; - } -} - static void snd_usb_mixer_status_complete(struct urb *urb) { struct usb_mixer_interface *mixer = urb->context; @@ -1916,7 +1984,7 @@ static void snd_usb_mixer_status_complete(struct urb *urb) if (!(buf[0] & 0x40)) snd_usb_mixer_notify_id(mixer, buf[1]); else - snd_usb_mixer_memory_change(mixer, buf[1]); + snd_usb_mixer_rc_memory_change(mixer, buf[1]); } } if (urb->status != -ENOENT && urb->status != -ECONNRESET) { @@ -1960,296 +2028,6 @@ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) return 0; } -static void snd_usb_soundblaster_remote_complete(struct urb *urb) -{ - struct usb_mixer_interface *mixer = urb->context; - const struct rc_config *rc = mixer->rc_cfg; - u32 code; - - if (urb->status < 0 || urb->actual_length < rc->min_packet_length) - return; - - code = mixer->rc_buffer[rc->offset]; - if (rc->length == 2) - code |= mixer->rc_buffer[rc->offset + 1] << 8; - - /* the Mute button actually changes the mixer control */ - if (code == rc->mute_code) - snd_usb_mixer_notify_id(mixer, rc->mute_mixer_id); - mixer->rc_code = code; - wmb(); - wake_up(&mixer->rc_waitq); -} - -static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf, - long count, loff_t *offset) -{ - struct usb_mixer_interface *mixer = hw->private_data; - int err; - u32 rc_code; - - if (count != 1 && count != 4) - return -EINVAL; - err = wait_event_interruptible(mixer->rc_waitq, - (rc_code = xchg(&mixer->rc_code, 0)) != 0); - if (err == 0) { - if (count == 1) - err = put_user(rc_code, buf); - else - err = put_user(rc_code, (u32 __user *)buf); - } - return err < 0 ? err : count; -} - -static unsigned int snd_usb_sbrc_hwdep_poll(struct snd_hwdep *hw, struct file *file, - poll_table *wait) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - poll_wait(file, &mixer->rc_waitq, wait); - return mixer->rc_code ? POLLIN | POLLRDNORM : 0; -} - -static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) -{ - struct snd_hwdep *hwdep; - int err, len, i; - - for (i = 0; i < ARRAY_SIZE(rc_configs); ++i) - if (rc_configs[i].usb_id == mixer->chip->usb_id) - break; - if (i >= ARRAY_SIZE(rc_configs)) - return 0; - mixer->rc_cfg = &rc_configs[i]; - - len = mixer->rc_cfg->packet_length; - - init_waitqueue_head(&mixer->rc_waitq); - err = snd_hwdep_new(mixer->chip->card, "SB remote control", 0, &hwdep); - if (err < 0) - return err; - snprintf(hwdep->name, sizeof(hwdep->name), - "%s remote control", mixer->chip->card->shortname); - hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC; - hwdep->private_data = mixer; - hwdep->ops.read = snd_usb_sbrc_hwdep_read; - hwdep->ops.poll = snd_usb_sbrc_hwdep_poll; - hwdep->exclusive = 1; - - mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL); - if (!mixer->rc_urb) - return -ENOMEM; - mixer->rc_setup_packet = kmalloc(sizeof(*mixer->rc_setup_packet), GFP_KERNEL); - if (!mixer->rc_setup_packet) { - usb_free_urb(mixer->rc_urb); - mixer->rc_urb = NULL; - return -ENOMEM; - } - mixer->rc_setup_packet->bRequestType = - USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; - mixer->rc_setup_packet->bRequest = UAC_GET_MEM; - mixer->rc_setup_packet->wValue = cpu_to_le16(0); - mixer->rc_setup_packet->wIndex = cpu_to_le16(0); - mixer->rc_setup_packet->wLength = cpu_to_le16(len); - usb_fill_control_urb(mixer->rc_urb, mixer->chip->dev, - usb_rcvctrlpipe(mixer->chip->dev, 0), - (u8*)mixer->rc_setup_packet, mixer->rc_buffer, len, - snd_usb_soundblaster_remote_complete, mixer); - return 0; -} - -#define snd_audigy2nx_led_info snd_ctl_boolean_mono_info - -static int snd_audigy2nx_led_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - int index = kcontrol->private_value; - - ucontrol->value.integer.value[0] = mixer->audigy2nx_leds[index]; - return 0; -} - -static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - int index = kcontrol->private_value; - int value = ucontrol->value.integer.value[0]; - int err, changed; - - if (value > 1) - return -EINVAL; - changed = value != mixer->audigy2nx_leds[index]; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - value, index + 2, NULL, 0, 100); - if (err < 0) - return err; - mixer->audigy2nx_leds[index] = value; - return changed; -} - -static struct snd_kcontrol_new snd_audigy2nx_controls[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "CMSS LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 0, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Power LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 1, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Dolby Digital LED Switch", - .info = snd_audigy2nx_led_info, - .get = snd_audigy2nx_led_get, - .put = snd_audigy2nx_led_put, - .private_value = 2, - }, -}; - -static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) -{ - int i, err; - - for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { - if (i > 1 && /* Live24ext has 2 LEDs only */ - (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) - break; - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); - if (err < 0) - return err; - } - mixer->audigy2nx_leds[1] = 1; /* Power LED is on by default */ - return 0; -} - -static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - static const struct sb_jack { - int unitid; - const char *name; - } jacks_audigy2nx[] = { - {4, "dig in "}, - {7, "line in"}, - {19, "spk out"}, - {20, "hph out"}, - {-1, NULL} - }, jacks_live24ext[] = { - {4, "line in"}, /* &1=Line, &2=Mic*/ - {3, "hph out"}, /* headphones */ - {0, "RC "}, /* last command, 6 bytes see rc_config above */ - {-1, NULL} - }; - const struct sb_jack *jacks; - struct usb_mixer_interface *mixer = entry->private_data; - int i, err; - u8 buf[3]; - - snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) - jacks = jacks_audigy2nx; - else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) - jacks = jacks_live24ext; - else - return; - - for (i = 0; jacks[i].name; ++i) { - snd_iprintf(buffer, "%s: ", jacks[i].name); - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_rcvctrlpipe(mixer->chip->dev, 0), - UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | - USB_RECIP_INTERFACE, 0, - jacks[i].unitid << 8, buf, 3, 100); - if (err == 3 && (buf[0] == 3 || buf[0] == 6)) - snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); - else - snd_iprintf(buffer, "?\n"); - } -} - -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); - return 0; -} - -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - u8 old_status, new_status; - int err, changed; - - old_status = mixer->xonar_u1_status; - if (ucontrol->value.integer.value[0]) - new_status = old_status | 0x02; - else - new_status = old_status & ~0x02; - changed = new_status != old_status; - err = snd_usb_ctl_msg(mixer->chip->dev, - usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 50, 0, &new_status, 1, 100); - if (err < 0) - return err; - mixer->xonar_u1_status = new_status; - return changed; -} - -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Playback Switch", - .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, -}; - -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) -{ - int err; - - err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); - if (err < 0) - return err; - mixer->xonar_u1_status = 0x05; - return 0; -} - -void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, - unsigned char samplerate_id) -{ - struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ - - list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer->id_elems[unitid]; - if (cval) { - set_cur_ctl_value(cval, cval->control << 8, - samplerate_id); - snd_usb_mixer_notify_id(mixer, unitid); - } - break; - } -} - int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { @@ -2259,7 +2037,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, struct usb_mixer_interface *mixer; struct snd_info_entry *entry; struct usb_host_interface *host_iface; - int err, protocol; + int err; strcpy(chip->card->mixername, "USB Mixer"); @@ -2277,38 +2055,13 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - protocol = host_iface->desc.bInterfaceProtocol; - - /* FIXME! */ - if (protocol != UAC_VERSION_1) { - snd_printk(KERN_WARNING "mixer interface protocol 0x%02x not yet supported\n", - protocol); - return 0; - } + mixer->protocol = host_iface->desc.bInterfaceProtocol; if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; - if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0) - goto _error; - - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { - if ((err = snd_audigy2nx_controls_create(mixer)) < 0) - goto _error; - if (!snd_card_proc_new(chip->card, "audigy2nx", &entry)) - snd_info_set_text_ops(entry, mixer, - snd_audigy2nx_proc_read); - } - - if (mixer->chip->usb_id == USB_ID(0x0b05, 0x1739) || - mixer->chip->usb_id == USB_ID(0x0b05, 0x1743)) { - err = snd_xonar_u1_controls_create(mixer); - if (err < 0) - goto _error; - } + snd_usb_mixer_apply_create_quirk(mixer); err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops); if (err < 0) @@ -2329,7 +2082,7 @@ _error: void snd_usb_mixer_disconnect(struct list_head *p) { struct usb_mixer_interface *mixer; - + mixer = list_entry(p, struct usb_mixer_interface, list); usb_kill_urb(mixer->urb); usb_kill_urb(mixer->rc_urb); diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h new file mode 100644 index 000000000000..130123854a6c --- /dev/null +++ b/sound/usb/mixer.h @@ -0,0 +1,55 @@ +#ifndef __USBMIXER_H +#define __USBMIXER_H + +struct usb_mixer_interface { + struct snd_usb_audio *chip; + unsigned int ctrlif; + struct list_head list; + unsigned int ignore_ctl_error; + struct urb *urb; + /* array[MAX_ID_ELEMS], indexed by unit id */ + struct usb_mixer_elem_info **id_elems; + + /* the usb audio specification version this interface complies to */ + int protocol; + + /* Sound Blaster remote control stuff */ + const struct rc_config *rc_cfg; + u32 rc_code; + wait_queue_head_t rc_waitq; + struct urb *rc_urb; + struct usb_ctrlrequest *rc_setup_packet; + u8 rc_buffer[6]; + + u8 audigy2nx_leds[3]; + u8 xonar_u1_status; +}; + +#define MAX_CHANNELS 10 /* max logical channels */ + +struct usb_mixer_elem_info { + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ + struct snd_ctl_elem_id *elem_id; + unsigned int id; + unsigned int control; /* CS or ICN (high byte) */ + unsigned int cmask; /* channel mask bitmap: 0 = master */ + int channels; + int val_type; + int min, max, res; + int dBmin, dBmax; + int cached; + int cache_val[MAX_CHANNELS]; + u8 initialized; +}; + +int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, + int ignore_error); +void snd_usb_mixer_disconnect(struct list_head *p); + +void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); + +int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, + int request, int validx, int value_set); + +#endif /* __USBMIXER_H */ diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/mixer_maps.c index 79e903a60862..d93fc89beba8 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -85,8 +85,8 @@ static struct usbmix_name_map extigy_map[] = { /* 16: MU (w/o controls) */ { 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */ { 17, "Channel Routing", 2 }, /* PU: mode select */ - { 18, "Tone Control - Bass", USB_FEATURE_BASS }, /* FU */ - { 18, "Tone Control - Treble", USB_FEATURE_TREBLE }, /* FU */ + { 18, "Tone Control - Bass", UAC_BASS_CONTROL }, /* FU */ + { 18, "Tone Control - Treble", UAC_TREBLE_CONTROL }, /* FU */ { 18, "Master Playback" }, /* FU; others */ /* 19: OT speaker */ /* 20: OT headphone */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c new file mode 100644 index 000000000000..56b665938b1b --- /dev/null +++ b/sound/usb/mixer_quirks.c @@ -0,0 +1,411 @@ +/* + * USB Audio Driver for ALSA + * + * Quirks and vendor-specific extensions for mixer interfaces + * + * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de> + * + * Many codes borrowed from audio.c by + * Alan Cox (alan@lxorguk.ukuu.org.uk) + * Thomas Sailer (sailer@ife.ee.ethz.ch) + * + * + * 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/usb.h> +#include <linux/usb/audio.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/hwdep.h> +#include <sound/info.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" + +/* + * Sound Blaster remote control configuration + * + * format of remote control data: + * Extigy: xx 00 + * Audigy 2 NX: 06 80 xx 00 00 00 + * Live! 24-bit: 06 80 xx yy 22 83 + */ +static const struct rc_config { + u32 usb_id; + u8 offset; + u8 length; + u8 packet_length; + u8 min_packet_length; /* minimum accepted length of the URB result */ + u8 mute_mixer_id; + u32 mute_code; +} rc_configs[] = { + { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ + { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ + { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ +}; + +static void snd_usb_soundblaster_remote_complete(struct urb *urb) +{ + struct usb_mixer_interface *mixer = urb->context; + const struct rc_config *rc = mixer->rc_cfg; + u32 code; + + if (urb->status < 0 || urb->actual_length < rc->min_packet_length) + return; + + code = mixer->rc_buffer[rc->offset]; + if (rc->length == 2) + code |= mixer->rc_buffer[rc->offset + 1] << 8; + + /* the Mute button actually changes the mixer control */ + if (code == rc->mute_code) + snd_usb_mixer_notify_id(mixer, rc->mute_mixer_id); + mixer->rc_code = code; + wmb(); + wake_up(&mixer->rc_waitq); +} + +static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf, + long count, loff_t *offset) +{ + struct usb_mixer_interface *mixer = hw->private_data; + int err; + u32 rc_code; + + if (count != 1 && count != 4) + return -EINVAL; + err = wait_event_interruptible(mixer->rc_waitq, + (rc_code = xchg(&mixer->rc_code, 0)) != 0); + if (err == 0) { + if (count == 1) + err = put_user(rc_code, buf); + else + err = put_user(rc_code, (u32 __user *)buf); + } + return err < 0 ? err : count; +} + +static unsigned int snd_usb_sbrc_hwdep_poll(struct snd_hwdep *hw, struct file *file, + poll_table *wait) +{ + struct usb_mixer_interface *mixer = hw->private_data; + + poll_wait(file, &mixer->rc_waitq, wait); + return mixer->rc_code ? POLLIN | POLLRDNORM : 0; +} + +static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) +{ + struct snd_hwdep *hwdep; + int err, len, i; + + for (i = 0; i < ARRAY_SIZE(rc_configs); ++i) + if (rc_configs[i].usb_id == mixer->chip->usb_id) + break; + if (i >= ARRAY_SIZE(rc_configs)) + return 0; + mixer->rc_cfg = &rc_configs[i]; + + len = mixer->rc_cfg->packet_length; + + init_waitqueue_head(&mixer->rc_waitq); + err = snd_hwdep_new(mixer->chip->card, "SB remote control", 0, &hwdep); + if (err < 0) + return err; + snprintf(hwdep->name, sizeof(hwdep->name), + "%s remote control", mixer->chip->card->shortname); + hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC; + hwdep->private_data = mixer; + hwdep->ops.read = snd_usb_sbrc_hwdep_read; + hwdep->ops.poll = snd_usb_sbrc_hwdep_poll; + hwdep->exclusive = 1; + + mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!mixer->rc_urb) + return -ENOMEM; + mixer->rc_setup_packet = kmalloc(sizeof(*mixer->rc_setup_packet), GFP_KERNEL); + if (!mixer->rc_setup_packet) { + usb_free_urb(mixer->rc_urb); + mixer->rc_urb = NULL; + return -ENOMEM; + } + mixer->rc_setup_packet->bRequestType = + USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; + mixer->rc_setup_packet->bRequest = UAC_GET_MEM; + mixer->rc_setup_packet->wValue = cpu_to_le16(0); + mixer->rc_setup_packet->wIndex = cpu_to_le16(0); + mixer->rc_setup_packet->wLength = cpu_to_le16(len); + usb_fill_control_urb(mixer->rc_urb, mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + (u8*)mixer->rc_setup_packet, mixer->rc_buffer, len, + snd_usb_soundblaster_remote_complete, mixer); + return 0; +} + +#define snd_audigy2nx_led_info snd_ctl_boolean_mono_info + +static int snd_audigy2nx_led_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int index = kcontrol->private_value; + + ucontrol->value.integer.value[0] = mixer->audigy2nx_leds[index]; + return 0; +} + +static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int index = kcontrol->private_value; + int value = ucontrol->value.integer.value[0]; + int err, changed; + + if (value > 1) + return -EINVAL; + changed = value != mixer->audigy2nx_leds[index]; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + value, index + 2, NULL, 0, 100); + if (err < 0) + return err; + mixer->audigy2nx_leds[index] = value; + return changed; +} + +static struct snd_kcontrol_new snd_audigy2nx_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CMSS LED Switch", + .info = snd_audigy2nx_led_info, + .get = snd_audigy2nx_led_get, + .put = snd_audigy2nx_led_put, + .private_value = 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Power LED Switch", + .info = snd_audigy2nx_led_info, + .get = snd_audigy2nx_led_get, + .put = snd_audigy2nx_led_put, + .private_value = 1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Dolby Digital LED Switch", + .info = snd_audigy2nx_led_info, + .get = snd_audigy2nx_led_get, + .put = snd_audigy2nx_led_put, + .private_value = 2, + }, +}; + +static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + if (i > 1 && /* Live24ext has 2 LEDs only */ + (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) + break; + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->audigy2nx_leds[1] = 1; /* Power LED is on by default */ + return 0; +} + +static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + static const struct sb_jack { + int unitid; + const char *name; + } jacks_audigy2nx[] = { + {4, "dig in "}, + {7, "line in"}, + {19, "spk out"}, + {20, "hph out"}, + {-1, NULL} + }, jacks_live24ext[] = { + {4, "line in"}, /* &1=Line, &2=Mic*/ + {3, "hph out"}, /* headphones */ + {0, "RC "}, /* last command, 6 bytes see rc_config above */ + {-1, NULL} + }; + const struct sb_jack *jacks; + struct usb_mixer_interface *mixer = entry->private_data; + int i, err; + u8 buf[3]; + + snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) + jacks = jacks_audigy2nx; + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) + jacks = jacks_live24ext; + else + return; + + for (i = 0; jacks[i].name; ++i) { + snd_iprintf(buffer, "%s: ", jacks[i].name); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0, + jacks[i].unitid << 8, buf, 3, 100); + if (err == 3 && (buf[0] == 3 || buf[0] == 6)) + snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); + else + snd_iprintf(buffer, "?\n"); + } +} + +static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + return 0; +} + +static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + u8 old_status, new_status; + int err, changed; + + old_status = mixer->xonar_u1_status; + if (ucontrol->value.integer.value[0]) + new_status = old_status | 0x02; + else + new_status = old_status & ~0x02; + changed = new_status != old_status; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + 50, 0, &new_status, 1, 100); + if (err < 0) + return err; + mixer->xonar_u1_status = new_status; + return changed; +} + +static struct snd_kcontrol_new snd_xonar_u1_output_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_xonar_u1_switch_get, + .put = snd_xonar_u1_switch_put, +}; + +static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +{ + int err; + + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + if (err < 0) + return err; + mixer->xonar_u1_status = 0x05; + return 0; +} + +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id) +{ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ + + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, + cval->control << 8, + samplerate_id); + snd_usb_mixer_notify_id(mixer, unitid); + } + break; + } +} + +int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) +{ + int err; + struct snd_info_entry *entry; + + if ((err = snd_usb_soundblaster_remote_init(mixer)) < 0) + return err; + + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { + if ((err = snd_audigy2nx_controls_create(mixer)) < 0) + return err; + if (!snd_card_proc_new(mixer->chip->card, "audigy2nx", &entry)) + snd_info_set_text_ops(entry, mixer, + snd_audigy2nx_proc_read); + } + + if (mixer->chip->usb_id == USB_ID(0x0b05, 0x1739) || + mixer->chip->usb_id == USB_ID(0x0b05, 0x1743)) { + err = snd_xonar_u1_controls_create(mixer); + if (err < 0) + return err; + } + + return 0; +} + +void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, + int unitid) +{ + if (!mixer->rc_cfg) + return; + /* unit ids specific to Extigy/Audigy 2 NX: */ + switch (unitid) { + case 0: /* remote control */ + mixer->rc_urb->dev = mixer->chip->dev; + usb_submit_urb(mixer->rc_urb, GFP_ATOMIC); + break; + case 4: /* digital in jack */ + case 7: /* line in jacks */ + case 19: /* speaker out jacks */ + case 20: /* headphones out jack */ + break; + /* live24ext: 4 = line-in jack */ + case 3: /* hp-out jack (may actuate Mute) */ + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) + snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); + break; + default: + snd_printd(KERN_DEBUG "memory change in unknown unit %d\n", unitid); + break; + } +} + diff --git a/sound/usb/mixer_quirks.h b/sound/usb/mixer_quirks.h new file mode 100644 index 000000000000..bdbfab093816 --- /dev/null +++ b/sound/usb/mixer_quirks.h @@ -0,0 +1,13 @@ +#ifndef SND_USB_MIXER_QUIRKS_H +#define SND_USB_MIXER_QUIRKS_H + +int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer); + +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id); + +void snd_usb_mixer_rc_memory_change(struct usb_mixer_interface *mixer, + int unitid); + +#endif /* SND_USB_MIXER_QUIRKS_H */ + diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c new file mode 100644 index 000000000000..630e2203b344 --- /dev/null +++ b/sound/usb/pcm.c @@ -0,0 +1,934 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> +#include <linux/usb/audio-v2.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "usbaudio.h" +#include "card.h" +#include "quirks.h" +#include "debug.h" +#include "urb.h" +#include "helper.h" +#include "pcm.h" + +/* + * return the current pcm pointer. just based on the hwptr_done value. + */ +static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_usb_substream *subs; + unsigned int hwptr_done; + + subs = (struct snd_usb_substream *)substream->runtime->private_data; + spin_lock(&subs->lock); + hwptr_done = subs->hwptr_done; + spin_unlock(&subs->lock); + return hwptr_done / (substream->runtime->frame_bits >> 3); +} + +/* + * find a matching audio format + */ +static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned int format, + unsigned int rate, unsigned int channels) +{ + struct list_head *p; + struct audioformat *found = NULL; + int cur_attr = 0, attr; + + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + fp = list_entry(p, struct audioformat, list); + if (!(fp->formats & (1uLL << format))) + continue; + if (fp->channels != channels) + continue; + if (rate < fp->rate_min || rate > fp->rate_max) + continue; + if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) { + unsigned int i; + for (i = 0; i < fp->nr_rates; i++) + if (fp->rate_table[i] == rate) + break; + if (i >= fp->nr_rates) + continue; + } + attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (! found) { + found = fp; + cur_attr = attr; + continue; + } + /* avoid async out and adaptive in if the other method + * supports the same format. + * this is a workaround for the case like + * M-audio audiophile USB. + */ + if (attr != cur_attr) { + if ((attr == USB_ENDPOINT_SYNC_ASYNC && + subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || + (attr == USB_ENDPOINT_SYNC_ADAPTIVE && + subs->direction == SNDRV_PCM_STREAM_CAPTURE)) + continue; + if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && + subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || + (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && + subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { + found = fp; + cur_attr = attr; + continue; + } + } + /* find the format with the largest max. packet size */ + if (fp->maxpacksize > found->maxpacksize) { + found = fp; + cur_attr = attr; + } + } + return found; +} + +static int init_pitch_v1(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt) +{ + struct usb_device *dev = chip->dev; + unsigned int ep; + unsigned char data[1]; + int err; + + ep = get_endpoint(alts, 0)->bEndpointAddress; + + /* if endpoint doesn't have pitch control, bail out */ + if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) + return 0; + + data[0] = 1; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", + dev->devnum, iface, ep); + return err; + } + + return 0; +} + +/* + * initialize the picth control and sample rate + */ +int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + + switch (altsd->bInterfaceProtocol) { + case UAC_VERSION_1: + return init_pitch_v1(chip, iface, alts, fmt); + + case UAC_VERSION_2: + /* not implemented yet */ + return 0; + } + + return -EINVAL; +} + +static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned int ep; + unsigned char data[3]; + int err, crate; + + ep = get_endpoint(alts, 0)->bEndpointAddress; + /* if endpoint doesn't have sampling rate control, bail out */ + if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) { + snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n", + dev->devnum, iface, fmt->altsetting); + return 0; + } + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", + dev->devnum, iface, fmt->altsetting, rate, ep); + return err; + } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", + dev->devnum, iface, fmt->altsetting, ep); + return 0; /* some devices don't support reading */ + } + crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (crate != rate) { + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + // runtime->rate = crate; + } + + return 0; +} + +static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_device *dev = chip->dev; + unsigned char data[4]; + int err, crate; + + data[0] = rate; + data[1] = rate >> 8; + data[2] = rate >> 16; + data[3] = rate >> 24; + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n", + dev->devnum, iface, fmt->altsetting, rate); + return err; + } + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8, + data, sizeof(data), 1000)) < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + return err; + } + crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + if (crate != rate) + snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + + return 0; +} + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) +{ + struct usb_interface_descriptor *altsd = get_iface_desc(alts); + + switch (altsd->bInterfaceProtocol) { + case UAC_VERSION_1: + return set_sample_rate_v1(chip, iface, alts, fmt, rate); + + case UAC_VERSION_2: + return set_sample_rate_v2(chip, iface, alts, fmt, rate); + } + + return -EINVAL; +} + +/* + * find a matching format and set up the interface + */ +static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) +{ + struct usb_device *dev = subs->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface; + unsigned int ep, attr; + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + int err; + + iface = usb_ifnum_to_if(dev, fmt->iface); + if (WARN_ON(!iface)) + return -EINVAL; + alts = &iface->altsetting[fmt->altset_idx]; + altsd = get_iface_desc(alts); + if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) + return -EINVAL; + + if (fmt == subs->cur_audiofmt) + return 0; + + /* close the old interface */ + if (subs->interface >= 0 && subs->interface != fmt->iface) { + if (usb_set_interface(subs->dev, subs->interface, 0) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n", + dev->devnum, fmt->iface, fmt->altsetting); + return -EIO; + } + subs->interface = -1; + subs->altset_idx = 0; + } + + /* set interface */ + if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { + if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { + snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", + dev->devnum, fmt->iface, fmt->altsetting); + return -EIO; + } + snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + } + + /* create a data pipe */ + ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK; + if (is_playback) + subs->datapipe = usb_sndisocpipe(dev, ep); + else + subs->datapipe = usb_rcvisocpipe(dev, ep); + subs->datainterval = fmt->datainterval; + subs->syncpipe = subs->syncinterval = 0; + subs->maxpacksize = fmt->maxpacksize; + subs->fill_max = 0; + + /* we need a sync pipe in async OUT or adaptive IN mode */ + /* check the number of EP, since some devices have broken + * descriptors which fool us. if it has only one EP, + * assume it as adaptive-out or sync-in. + */ + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && + altsd->bNumEndpoints >= 2) { + /* check sync-pipe endpoint */ + /* ... and check descriptor size before accessing bSynchAddress + because there is a version of the SB Audigy 2 NX firmware lacking + the audio fields in the endpoint descriptors */ + if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || + (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bSynchAddress != 0)) { + snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", + dev->devnum, fmt->iface, fmt->altsetting); + return -EINVAL; + } + ep = get_endpoint(alts, 1)->bEndpointAddress; + if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", + dev->devnum, fmt->iface, fmt->altsetting); + return -EINVAL; + } + ep &= USB_ENDPOINT_NUMBER_MASK; + if (is_playback) + subs->syncpipe = usb_rcvisocpipe(dev, ep); + else + subs->syncpipe = usb_sndisocpipe(dev, ep); + if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bRefresh >= 1 && + get_endpoint(alts, 1)->bRefresh <= 9) + subs->syncinterval = get_endpoint(alts, 1)->bRefresh; + else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + subs->syncinterval = 1; + else if (get_endpoint(alts, 1)->bInterval >= 1 && + get_endpoint(alts, 1)->bInterval <= 16) + subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; + else + subs->syncinterval = 3; + } + + /* always fill max packet size */ + if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) + subs->fill_max = 1; + + if ((err = snd_usb_init_pitch(subs->stream->chip, subs->interface, alts, fmt)) < 0) + return err; + + subs->cur_audiofmt = fmt; + + snd_usb_set_format_quirk(subs, fmt); + +#if 0 + printk(KERN_DEBUG + "setting done: format = %d, rate = %d..%d, channels = %d\n", + fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); + printk(KERN_DEBUG + " datapipe = 0x%0x, syncpipe = 0x%0x\n", + subs->datapipe, subs->syncpipe); +#endif + + return 0; +} + +/* + * hw_params callback + * + * allocate a buffer and set the given audio format. + * + * so far we use a physically linear buffer although packetize transfer + * doesn't need a continuous area. + * if sg buffer is supported on the later version of alsa, we'll follow + * that. + */ +static int snd_usb_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + struct audioformat *fmt; + unsigned int channels, rate, format; + int ret, changed; + + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + format = params_format(hw_params); + rate = params_rate(hw_params); + channels = params_channels(hw_params); + fmt = find_format(subs, format, rate, channels); + if (!fmt) { + snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", + format, rate, channels); + return -EINVAL; + } + + changed = subs->cur_audiofmt != fmt || + subs->period_bytes != params_period_bytes(hw_params) || + subs->cur_rate != rate; + if ((ret = set_format(subs, fmt)) < 0) + return ret; + + if (subs->cur_rate != rate) { + struct usb_host_interface *alts; + struct usb_interface *iface; + iface = usb_ifnum_to_if(subs->dev, fmt->iface); + alts = &iface->altsetting[fmt->altset_idx]; + ret = snd_usb_init_sample_rate(subs->stream->chip, subs->interface, alts, fmt, rate); + if (ret < 0) + return ret; + subs->cur_rate = rate; + } + + if (changed) { + /* format changed */ + snd_usb_release_substream_urbs(subs, 0); + /* influenced: period_bytes, channels, rate, format, */ + ret = snd_usb_init_substream_urbs(subs, params_period_bytes(hw_params), + params_rate(hw_params), + snd_pcm_format_physical_width(params_format(hw_params)) * + params_channels(hw_params)); + } + + return ret; +} + +/* + * hw_free callback + * + * reset the audio format and release the buffer + */ +static int snd_usb_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + subs->cur_audiofmt = NULL; + subs->cur_rate = 0; + subs->period_bytes = 0; + if (!subs->stream->chip->shutdown) + snd_usb_release_substream_urbs(subs, 0); + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +/* + * prepare callback + * + * only a few subtle things... + */ +static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_usb_substream *subs = runtime->private_data; + + if (! subs->cur_audiofmt) { + snd_printk(KERN_ERR "usbaudio: no format is specified!\n"); + return -ENXIO; + } + + /* some unit conversions in runtime */ + subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize); + subs->curframesize = bytes_to_frames(runtime, subs->curpacksize); + + /* reset the pointer */ + subs->hwptr_done = 0; + subs->transfer_done = 0; + subs->phase = 0; + runtime->delay = 0; + + return snd_usb_substream_prepare(subs, runtime); +} + +static struct snd_pcm_hardware snd_usb_hardware = +{ + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE, + .buffer_bytes_max = 1024 * 1024, + .period_bytes_min = 64, + .period_bytes_max = 512 * 1024, + .periods_min = 2, + .periods_max = 1024, +}; + +static int hw_check_valid_format(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *params, + struct audioformat *fp) +{ + struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + struct snd_mask check_fmts; + unsigned int ptime; + + /* check the format */ + snd_mask_none(&check_fmts); + check_fmts.bits[0] = (u32)fp->formats; + check_fmts.bits[1] = (u32)(fp->formats >> 32); + snd_mask_intersect(&check_fmts, fmts); + if (snd_mask_empty(&check_fmts)) { + hwc_debug(" > check: no supported format %d\n", fp->format); + return 0; + } + /* check the channels */ + if (fp->channels < ct->min || fp->channels > ct->max) { + hwc_debug(" > check: no valid channels %d (%d/%d)\n", fp->channels, ct->min, ct->max); + return 0; + } + /* check the rate is within the range */ + if (fp->rate_min > it->max || (fp->rate_min == it->max && it->openmax)) { + hwc_debug(" > check: rate_min %d > max %d\n", fp->rate_min, it->max); + return 0; + } + if (fp->rate_max < it->min || (fp->rate_max == it->min && it->openmin)) { + hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min); + return 0; + } + /* check whether the period time is >= the data packet interval */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + ptime = 125 * (1 << fp->datainterval); + if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { + hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); + return 0; + } + } + return 1; +} + +static int hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + struct list_head *p; + struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + unsigned int rmin, rmax; + int changed; + + hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); + changed = 0; + rmin = rmax = 0; + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + fp = list_entry(p, struct audioformat, list); + if (!hw_check_valid_format(subs, params, fp)) + continue; + if (changed++) { + if (rmin > fp->rate_min) + rmin = fp->rate_min; + if (rmax < fp->rate_max) + rmax = fp->rate_max; + } else { + rmin = fp->rate_min; + rmax = fp->rate_max; + } + } + + if (!changed) { + hwc_debug(" --> get empty\n"); + it->empty = 1; + return -EINVAL; + } + + changed = 0; + if (it->min < rmin) { + it->min = rmin; + it->openmin = 0; + changed = 1; + } + if (it->max > rmax) { + it->max = rmax; + it->openmax = 0; + changed = 1; + } + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed); + return changed; +} + + +static int hw_rule_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + struct list_head *p; + struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + unsigned int rmin, rmax; + int changed; + + hwc_debug("hw_rule_channels: (%d,%d)\n", it->min, it->max); + changed = 0; + rmin = rmax = 0; + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + fp = list_entry(p, struct audioformat, list); + if (!hw_check_valid_format(subs, params, fp)) + continue; + if (changed++) { + if (rmin > fp->channels) + rmin = fp->channels; + if (rmax < fp->channels) + rmax = fp->channels; + } else { + rmin = fp->channels; + rmax = fp->channels; + } + } + + if (!changed) { + hwc_debug(" --> get empty\n"); + it->empty = 1; + return -EINVAL; + } + + changed = 0; + if (it->min < rmin) { + it->min = rmin; + it->openmin = 0; + changed = 1; + } + if (it->max > rmax) { + it->max = rmax; + it->openmax = 0; + changed = 1; + } + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed); + return changed; +} + +static int hw_rule_format(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + struct list_head *p; + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u64 fbits; + u32 oldbits[2]; + int changed; + + hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); + fbits = 0; + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + fp = list_entry(p, struct audioformat, list); + if (!hw_check_valid_format(subs, params, fp)) + continue; + fbits |= fp->formats; + } + + oldbits[0] = fmt->bits[0]; + oldbits[1] = fmt->bits[1]; + fmt->bits[0] &= (u32)fbits; + fmt->bits[1] &= (u32)(fbits >> 32); + if (!fmt->bits[0] && !fmt->bits[1]) { + hwc_debug(" --> get empty\n"); + return -EINVAL; + } + changed = (oldbits[0] != fmt->bits[0] || oldbits[1] != fmt->bits[1]); + hwc_debug(" --> %x:%x (changed = %d)\n", fmt->bits[0], fmt->bits[1], changed); + return changed; +} + +static int hw_rule_period_time(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_usb_substream *subs = rule->private; + struct audioformat *fp; + struct snd_interval *it; + unsigned char min_datainterval; + unsigned int pmin; + int changed; + + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max); + min_datainterval = 0xff; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) + continue; + min_datainterval = min(min_datainterval, fp->datainterval); + } + if (min_datainterval == 0xff) { + hwc_debug(" --> get emtpy\n"); + it->empty = 1; + return -EINVAL; + } + pmin = 125 * (1 << min_datainterval); + changed = 0; + if (it->min < pmin) { + it->min = pmin; + it->openmin = 0; + changed = 1; + } + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed); + return changed; +} + +/* + * If the device supports unusual bit rates, does the request meet these? + */ +static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, + struct snd_usb_substream *subs) +{ + struct audioformat *fp; + int count = 0, needs_knot = 0; + int err; + + list_for_each_entry(fp, &subs->fmt_list, list) { + if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) + return 0; + count += fp->nr_rates; + if (fp->rates & SNDRV_PCM_RATE_KNOT) + needs_knot = 1; + } + if (!needs_knot) + return 0; + + subs->rate_list.count = count; + subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); + subs->rate_list.mask = 0; + count = 0; + list_for_each_entry(fp, &subs->fmt_list, list) { + int i; + for (i = 0; i < fp->nr_rates; i++) + subs->rate_list.list[count++] = fp->rate_table[i]; + } + err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &subs->rate_list); + if (err < 0) + return err; + + return 0; +} + + +/* + * set up the runtime hardware information. + */ + +static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) +{ + struct list_head *p; + unsigned int pt, ptmin; + int param_period_time_if_needed; + int err; + + runtime->hw.formats = subs->formats; + + runtime->hw.rate_min = 0x7fffffff; + runtime->hw.rate_max = 0; + runtime->hw.channels_min = 256; + runtime->hw.channels_max = 0; + runtime->hw.rates = 0; + ptmin = UINT_MAX; + /* check min/max rates and channels */ + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + fp = list_entry(p, struct audioformat, list); + runtime->hw.rates |= fp->rates; + if (runtime->hw.rate_min > fp->rate_min) + runtime->hw.rate_min = fp->rate_min; + if (runtime->hw.rate_max < fp->rate_max) + runtime->hw.rate_max = fp->rate_max; + if (runtime->hw.channels_min > fp->channels) + runtime->hw.channels_min = fp->channels; + if (runtime->hw.channels_max < fp->channels) + runtime->hw.channels_max = fp->channels; + if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { + /* FIXME: there might be more than one audio formats... */ + runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = + fp->frame_size; + } + pt = 125 * (1 << fp->datainterval); + ptmin = min(ptmin, pt); + } + + param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; + if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + /* full speed devices have fixed data packet interval */ + ptmin = 1000; + if (ptmin == 1000) + /* if period time doesn't go below 1 ms, no rules needed */ + param_period_time_if_needed = -1; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, + ptmin, UINT_MAX); + + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) + return err; + if (param_period_time_if_needed >= 0) { + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + hw_rule_period_time, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, + -1); + if (err < 0) + return err; + } + if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) + return err; + return 0; +} + +static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) +{ + struct snd_usb_stream *as = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_usb_substream *subs = &as->substream[direction]; + + subs->interface = -1; + subs->altset_idx = 0; + runtime->hw = snd_usb_hardware; + runtime->private_data = subs; + subs->pcm_substream = substream; + return setup_hw_info(runtime, subs); +} + +static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) +{ + struct snd_usb_stream *as = snd_pcm_substream_chip(substream); + struct snd_usb_substream *subs = &as->substream[direction]; + + if (!as->chip->shutdown && subs->interface >= 0) { + usb_set_interface(subs->dev, subs->interface, 0); + subs->interface = -1; + } + subs->pcm_substream = NULL; + return 0; +} + +static int snd_usb_playback_open(struct snd_pcm_substream *substream) +{ + return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); +} + +static int snd_usb_playback_close(struct snd_pcm_substream *substream) +{ + return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); +} + +static int snd_usb_capture_open(struct snd_pcm_substream *substream) +{ + return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); +} + +static int snd_usb_capture_close(struct snd_pcm_substream *substream) +{ + return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); +} + +static struct snd_pcm_ops snd_usb_playback_ops = { + .open = snd_usb_playback_open, + .close = snd_usb_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_playback_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +static struct snd_pcm_ops snd_usb_capture_ops = { + .open = snd_usb_capture_open, + .close = snd_usb_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_usb_hw_params, + .hw_free = snd_usb_hw_free, + .prepare = snd_usb_pcm_prepare, + .trigger = snd_usb_substream_capture_trigger, + .pointer = snd_usb_pcm_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, +}; + +void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream) +{ + snd_pcm_set_ops(pcm, stream, + stream == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_usb_playback_ops : &snd_usb_capture_ops); +} diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h new file mode 100644 index 000000000000..1c931b68f3b5 --- /dev/null +++ b/sound/usb/pcm.h @@ -0,0 +1,14 @@ +#ifndef __USBAUDIO_PCM_H +#define __USBAUDIO_PCM_H + +void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream); + +int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt); + +int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, + struct usb_host_interface *alts, + struct audioformat *fmt, int rate); + +#endif /* __USBAUDIO_PCM_H */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c new file mode 100644 index 000000000000..f5e3f356b95f --- /dev/null +++ b/sound/usb/proc.c @@ -0,0 +1,168 @@ +/* + * 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/usb.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "helper.h" +#include "card.h" +#include "proc.h" + +/* convert our full speed USB rate into sampling rate in Hz */ +static inline unsigned get_full_speed_hz(unsigned int usb_rate) +{ + return (usb_rate * 125 + (1 << 12)) >> 13; +} + +/* convert our high speed USB rate into sampling rate in Hz */ +static inline unsigned get_high_speed_hz(unsigned int usb_rate) +{ + return (usb_rate * 125 + (1 << 9)) >> 10; +} + +/* + * common proc files to show the usb device info + */ +static void proc_audio_usbbus_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +{ + struct snd_usb_audio *chip = entry->private_data; + if (!chip->shutdown) + snd_iprintf(buffer, "%03d/%03d\n", chip->dev->bus->busnum, chip->dev->devnum); +} + +static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +{ + struct snd_usb_audio *chip = entry->private_data; + if (!chip->shutdown) + snd_iprintf(buffer, "%04x:%04x\n", + USB_ID_VENDOR(chip->usb_id), + USB_ID_PRODUCT(chip->usb_id)); +} + +void snd_usb_audio_create_proc(struct snd_usb_audio *chip) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(chip->card, "usbbus", &entry)) + snd_info_set_text_ops(entry, chip, proc_audio_usbbus_read); + if (!snd_card_proc_new(chip->card, "usbid", &entry)) + snd_info_set_text_ops(entry, chip, proc_audio_usbid_read); +} + +/* + * proc interface for list the supported pcm formats + */ +static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) +{ + struct list_head *p; + static char *sync_types[4] = { + "NONE", "ASYNC", "ADAPTIVE", "SYNC" + }; + + list_for_each(p, &subs->fmt_list) { + struct audioformat *fp; + snd_pcm_format_t fmt; + fp = list_entry(p, struct audioformat, list); + snd_iprintf(buffer, " Interface %d\n", fp->iface); + snd_iprintf(buffer, " Altset %d\n", fp->altsetting); + snd_iprintf(buffer, " Format:"); + for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + if (fp->formats & (1uLL << fmt)) + snd_iprintf(buffer, " %s", + snd_pcm_format_name(fmt)); + snd_iprintf(buffer, "\n"); + snd_iprintf(buffer, " Channels: %d\n", fp->channels); + snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", + fp->endpoint & USB_ENDPOINT_NUMBER_MASK, + fp->endpoint & USB_DIR_IN ? "IN" : "OUT", + sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); + if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { + snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", + fp->rate_min, fp->rate_max); + } else { + unsigned int i; + snd_iprintf(buffer, " Rates: "); + for (i = 0; i < fp->nr_rates; i++) { + if (i > 0) + snd_iprintf(buffer, ", "); + snd_iprintf(buffer, "%d", fp->rate_table[i]); + } + snd_iprintf(buffer, "\n"); + } + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + snd_iprintf(buffer, " Data packet interval: %d us\n", + 125 * (1 << fp->datainterval)); + // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); + // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); + } +} + +static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) +{ + if (subs->running) { + unsigned int i; + snd_iprintf(buffer, " Status: Running\n"); + snd_iprintf(buffer, " Interface = %d\n", subs->interface); + snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); + snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); + for (i = 0; i < subs->nurbs; i++) + snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); + snd_iprintf(buffer, "]\n"); + snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize); + snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", + snd_usb_get_speed(subs->dev) == USB_SPEED_FULL + ? get_full_speed_hz(subs->freqm) + : get_high_speed_hz(subs->freqm), + subs->freqm >> 16, subs->freqm & 0xffff); + } else { + snd_iprintf(buffer, " Status: Stop\n"); + } +} + +static void proc_pcm_format_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) +{ + struct snd_usb_stream *stream = entry->private_data; + + snd_iprintf(buffer, "%s : %s\n", stream->chip->card->longname, stream->pcm->name); + + if (stream->substream[SNDRV_PCM_STREAM_PLAYBACK].num_formats) { + snd_iprintf(buffer, "\nPlayback:\n"); + proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer); + proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer); + } + if (stream->substream[SNDRV_PCM_STREAM_CAPTURE].num_formats) { + snd_iprintf(buffer, "\nCapture:\n"); + proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer); + proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer); + } +} + +void snd_usb_proc_pcm_format_add(struct snd_usb_stream *stream) +{ + struct snd_info_entry *entry; + char name[32]; + struct snd_card *card = stream->chip->card; + + sprintf(name, "stream%d", stream->pcm_index); + if (!snd_card_proc_new(card, name, &entry)) + snd_info_set_text_ops(entry, stream, proc_pcm_format_read); +} + diff --git a/sound/usb/proc.h b/sound/usb/proc.h new file mode 100644 index 000000000000..a45b765e4cf1 --- /dev/null +++ b/sound/usb/proc.h @@ -0,0 +1,8 @@ +#ifndef __USBAUDIO_PROC_H +#define __USBAUDIO_PROC_H + +void snd_usb_audio_create_proc(struct snd_usb_audio *chip); +void snd_usb_proc_pcm_format_add(struct snd_usb_stream *stream); + +#endif /* __USBAUDIO_PROC_H */ + diff --git a/sound/usb/usbquirks.h b/sound/usb/quirks-table.h index 2b426c1fd0e8..81c5f8a312ce 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/quirks-table.h @@ -279,7 +279,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 4, .iface = 0, .altsetting = 1, @@ -296,7 +296,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -580,7 +580,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, @@ -597,7 +597,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -793,7 +793,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -810,7 +810,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 2, .altsetting = 1, @@ -2203,7 +2203,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3BE, + .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, .altsetting = 1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c new file mode 100644 index 000000000000..d4ced64e465b --- /dev/null +++ b/sound/usb/quirks.c @@ -0,0 +1,593 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> + +#include <sound/core.h> +#include <sound/info.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "midi.h" +#include "quirks.h" +#include "helper.h" +#include "endpoint.h" +#include "pcm.h" + +/* + * handle the quirks for the contained interfaces + */ +static int create_composite_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber; + int err; + + for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) { + iface = usb_ifnum_to_if(chip->dev, quirk->ifnum); + if (!iface) + continue; + if (quirk->ifnum != probed_ifnum && + usb_interface_claimed(iface)) + continue; + err = snd_usb_create_quirk(chip, iface, driver, quirk); + if (err < 0) + return err; + if (quirk->ifnum != probed_ifnum) + usb_driver_claim_interface(driver, iface, (void *)-1L); + } + return 0; +} + +static int ignore_interface_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + return 0; +} + + +/* + * Allow alignment on audio sub-slot (channel samples) rather than + * on audio slots (audio frames) + */ +static int create_align_transfer_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + chip->txfr_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + +static int create_any_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *intf, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + return snd_usbmidi_create(chip->card, intf, &chip->midi_list, quirk); +} + +/* + * create a stream for an interface with proper descriptors + */ +static int create_standard_audio_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + int err; + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + err = snd_usb_parse_audio_endpoints(chip, altsd->bInterfaceNumber); + if (err < 0) { + snd_printk(KERN_ERR "cannot setup if %d: error %d\n", + altsd->bInterfaceNumber, err); + return err; + } + /* reset the current interface */ + usb_set_interface(chip->dev, altsd->bInterfaceNumber, 0); + return 0; +} + +/* + * create a stream for an endpoint/altsetting without proper descriptors + */ +static int create_fixed_stream_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + struct audioformat *fp; + struct usb_host_interface *alts; + int stream, err; + unsigned *rate_table = NULL; + + fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); + if (! fp) { + snd_printk(KERN_ERR "cannot memdup\n"); + return -ENOMEM; + } + if (fp->nr_rates > 0) { + rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); + if (!rate_table) { + kfree(fp); + return -ENOMEM; + } + memcpy(rate_table, fp->rate_table, sizeof(int) * fp->nr_rates); + fp->rate_table = rate_table; + } + + stream = (fp->endpoint & USB_DIR_IN) + ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + err = snd_usb_add_audio_endpoint(chip, stream, fp); + if (err < 0) { + kfree(fp); + kfree(rate_table); + return err; + } + if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || + fp->altset_idx >= iface->num_altsetting) { + kfree(fp); + kfree(rate_table); + return -EINVAL; + } + alts = &iface->altsetting[fp->altset_idx]; + fp->datainterval = snd_usb_parse_datainterval(chip, alts); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + usb_set_interface(chip->dev, fp->iface, 0); + snd_usb_init_pitch(chip, fp->iface, alts, fp); + snd_usb_init_sample_rate(chip, fp->iface, alts, fp, fp->rate_max); + return 0; +} + +/* + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * The only way to detect the sample rate is by looking at wMaxPacketSize. + */ +static int create_uaxx_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + static const struct audioformat ua_format = { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .fmt_type = UAC_FORMAT_TYPE_I, + .altsetting = 1, + .altset_idx = 1, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + }; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct audioformat *fp; + int stream, err; + + /* both PCM and MIDI interfaces have 2 or more altsettings */ + if (iface->num_altsetting < 2) + return -ENXIO; + alts = &iface->altsetting[1]; + altsd = get_iface_desc(alts); + + if (altsd->bNumEndpoints == 2) { + static const struct snd_usb_midi_endpoint_info ua700_ep = { + .out_cables = 0x0003, + .in_cables = 0x0003 + }; + static const struct snd_usb_audio_quirk ua700_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &ua700_ep + }; + static const struct snd_usb_midi_endpoint_info uaxx_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk uaxx_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &uaxx_ep + }; + const struct snd_usb_audio_quirk *quirk = + chip->usb_id == USB_ID(0x0582, 0x002b) + ? &ua700_quirk : &uaxx_quirk; + return snd_usbmidi_create(chip->card, iface, + &chip->midi_list, quirk); + } + + if (altsd->bNumEndpoints != 1) + return -ENXIO; + + fp = kmalloc(sizeof(*fp), GFP_KERNEL); + if (!fp) + return -ENOMEM; + memcpy(fp, &ua_format, sizeof(*fp)); + + fp->iface = altsd->bInterfaceNumber; + fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; + fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = 0; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + + switch (fp->maxpacksize) { + case 0x120: + fp->rate_max = fp->rate_min = 44100; + break; + case 0x138: + case 0x140: + fp->rate_max = fp->rate_min = 48000; + break; + case 0x258: + case 0x260: + fp->rate_max = fp->rate_min = 96000; + break; + default: + snd_printk(KERN_ERR "unknown sample rate\n"); + kfree(fp); + return -ENXIO; + } + + stream = (fp->endpoint & USB_DIR_IN) + ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + err = snd_usb_add_audio_endpoint(chip, stream, fp); + if (err < 0) { + kfree(fp); + return err; + } + usb_set_interface(chip->dev, fp->iface, 0); + return 0; +} + +/* + * audio-interface quirks + * + * returns zero if no standard audio/MIDI parsing is needed. + * returns a postive value if standard audio/midi interfaces are parsed + * after this. + * returns a negative value at error. + */ +int snd_usb_create_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + typedef int (*quirk_func_t)(struct snd_usb_audio *, + struct usb_interface *, + struct usb_driver *, + const struct snd_usb_audio_quirk *); + static const quirk_func_t quirk_funcs[] = { + [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, + [QUIRK_COMPOSITE] = create_composite_quirk, + [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, + [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, + [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, + [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, + [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, + [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, + [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, + [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, + [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk + }; + + if (quirk->type < QUIRK_TYPE_COUNT) { + return quirk_funcs[quirk->type](chip, iface, driver, quirk); + } else { + snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); + return -ENXIO; + } +} + +/* + * boot quirks + */ + +#define EXTIGY_FIRMWARE_SIZE_OLD 794 +#define EXTIGY_FIRMWARE_SIZE_NEW 483 + +static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interface *intf) +{ + struct usb_host_config *config = dev->actconfig; + int err; + + if (le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_OLD || + le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) { + snd_printdd("sending Extigy boot sequence...\n"); + /* Send message to force it to reconnect with full interface. */ + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), + 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000); + if (err < 0) snd_printdd("error sending boot message: %d\n", err); + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); + err = usb_reset_configuration(dev); + if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); + snd_printdd("extigy_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + return -ENODEV; /* quit this anyway */ + } + return 0; +} + +static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) +{ + u8 buf = 1; + + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER, + 0, 0, &buf, 1, 1000); + if (buf == 0) { + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + 1, 2000, NULL, 0, 1000); + return -ENODEV; + } + return 0; +} + +/* + * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely + * documented in the device's data sheet. + */ +static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) +{ + u8 buf[4]; + buf[0] = 0x20; + buf[1] = value & 0xff; + buf[2] = (value >> 8) & 0xff; + buf[3] = reg; + return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, + USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, + 0, 0, &buf, 4, 1000); +} + +static int snd_usb_cm106_boot_quirk(struct usb_device *dev) +{ + /* + * Enable line-out driver mode, set headphone source to front + * channels, enable stereo mic. + */ + return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); +} + +/* + * C-Media CM6206 is based on CM106 with two additional + * registers that are not documented in the data sheet. + * Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) +{ + int err, reg; + int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; + + for (reg = 0; reg < ARRAY_SIZE(val); reg++) { + err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); + if (err < 0) + return err; + } + + return err; +} + +/* + * This call will put the synth in "USB send" mode, i.e it will send MIDI + * messages through USB (this is disabled at startup). The synth will + * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB + * sign on its LCD. Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) +{ + int err, actual_length; + + /* "midi send" enable */ + static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + + void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (!buf) + return -ENOMEM; + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, + ARRAY_SIZE(seq), &actual_length, 1000); + kfree(buf); + if (err < 0) + return err; + + return 0; +} + +/* + * Setup quirks + */ +#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ +#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */ +#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */ +#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */ +#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */ +#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */ +#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */ +#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */ +#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */ +#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */ + +static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, + int altno) +{ + /* Reset ALL ifaces to 0 altsetting. + * Call it for every possible altsetting of every interface. + */ + usb_set_interface(chip->dev, iface, 0); + + if (chip->setup & AUDIOPHILE_SET) { + if ((chip->setup & AUDIOPHILE_SET_DTS) + && altno != 6) + return 1; /* skip this altsetting */ + if ((chip->setup & AUDIOPHILE_SET_96K) + && altno != 1) + return 1; /* skip this altsetting */ + if ((chip->setup & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_24B_48K_DI && altno != 2) + return 1; /* skip this altsetting */ + if ((chip->setup & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3) + return 1; /* skip this altsetting */ + if ((chip->setup & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_16B_48K_DI && altno != 4) + return 1; /* skip this altsetting */ + if ((chip->setup & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) + return 1; /* skip this altsetting */ + } + + return 0; /* keep this altsetting */ +} + +int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, + int iface, + int altno) +{ + /* audiophile usb: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0763, 0x2003)) + return audiophile_skip_setting_quirk(chip, iface, altno); + + return 0; +} + +int snd_usb_apply_boot_quirk(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk) +{ + u32 id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)); + + /* SB Extigy needs special boot-up sequence */ + /* if more models come, this will go to the quirk list. */ + if (id == USB_ID(0x041e, 0x3000)) + return snd_usb_extigy_boot_quirk(dev, intf); + + /* SB Audigy 2 NX needs its own boot-up magic, too */ + if (id == USB_ID(0x041e, 0x3020)) + return snd_usb_audigy2nx_boot_quirk(dev); + + /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */ + if (id == USB_ID(0x10f5, 0x0200)) + return snd_usb_cm106_boot_quirk(dev); + + /* C-Media CM6206 / CM106-Like Sound Device */ + if (id == USB_ID(0x0d8c, 0x0102)) + return snd_usb_cm6206_boot_quirk(dev); + + /* Access Music VirusTI Desktop */ + if (id == USB_ID(0x133e, 0x0815)) + return snd_usb_accessmusic_boot_quirk(dev); + + return 0; +} + +/* + * check if the device uses big-endian samples + */ +int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) +{ + switch (chip->usb_id) { + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ + if (fp->endpoint & USB_DIR_IN) + return 1; + break; + case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + if (chip->setup == 0x00 || + fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3) + return 1; + } + return 0; +} + +/* + * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * not for interface. + */ + +enum { + EMU_QUIRK_SR_44100HZ = 0, + EMU_QUIRK_SR_48000HZ, + EMU_QUIRK_SR_88200HZ, + EMU_QUIRK_SR_96000HZ, + EMU_QUIRK_SR_176400HZ, + EMU_QUIRK_SR_192000HZ +}; + +static void set_format_emu_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + unsigned char emu_samplerate_id = 0; + + /* When capture is active + * sample rate shouldn't be changed + * by playback substream + */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) + return; + } + + switch (fmt->rate_min) { + case 48000: + emu_samplerate_id = EMU_QUIRK_SR_48000HZ; + break; + case 88200: + emu_samplerate_id = EMU_QUIRK_SR_88200HZ; + break; + case 96000: + emu_samplerate_id = EMU_QUIRK_SR_96000HZ; + break; + case 176400: + emu_samplerate_id = EMU_QUIRK_SR_176400HZ; + break; + case 192000: + emu_samplerate_id = EMU_QUIRK_SR_192000HZ; + break; + default: + emu_samplerate_id = EMU_QUIRK_SR_44100HZ; + break; + } + snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); +} + +void snd_usb_set_format_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + switch (subs->stream->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + set_format_emu_quirk(subs, fmt); + break; + } +} + diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h new file mode 100644 index 000000000000..03e5e94098cd --- /dev/null +++ b/sound/usb/quirks.h @@ -0,0 +1,23 @@ +#ifndef __USBAUDIO_QUIRKS_H +#define __USBAUDIO_QUIRKS_H + +int snd_usb_create_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk); + +int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, + int iface, + int altno); + +int snd_usb_apply_boot_quirk(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk); + +void snd_usb_set_format_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt); + +int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, + struct audioformat *fp); + +#endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/urb.c b/sound/usb/urb.c new file mode 100644 index 000000000000..ad50d4398921 --- /dev/null +++ b/sound/usb/urb.c @@ -0,0 +1,989 @@ +/* + * 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/usb.h> +#include <linux/usb/audio.h> + +#include <sound/core.h> +#include <sound/pcm.h> + +#include "usbaudio.h" +#include "helper.h" +#include "card.h" +#include "urb.h" +#include "pcm.h" + +/* + * convert a sampling rate into our full speed format (fs/1000 in Q16.16) + * this will overflow at approx 524 kHz + */ +static inline unsigned get_usb_full_speed_rate(unsigned int rate) +{ + return ((rate << 13) + 62) / 125; +} + +/* + * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) + * this will overflow at approx 4 MHz + */ +static inline unsigned get_usb_high_speed_rate(unsigned int rate) +{ + return ((rate << 10) + 62) / 125; +} + +/* + * unlink active urbs. + */ +static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) +{ + struct snd_usb_audio *chip = subs->stream->chip; + unsigned int i; + int async; + + subs->running = 0; + + if (!force && subs->stream->chip->shutdown) /* to be sure... */ + return -EBADFD; + + async = !can_sleep && chip->async_unlink; + + if (!async && in_interrupt()) + return 0; + + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) { + if (!test_and_set_bit(i, &subs->unlink_mask)) { + struct urb *u = subs->dataurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i+16, &subs->active_mask)) { + if (!test_and_set_bit(i+16, &subs->unlink_mask)) { + struct urb *u = subs->syncurb[i].urb; + if (async) + usb_unlink_urb(u); + else + usb_kill_urb(u); + } + } + } + } + return 0; +} + + +/* + * release a urb data + */ +static void release_urb_ctx(struct snd_urb_ctx *u) +{ + if (u->urb) { + if (u->buffer_size) + usb_buffer_free(u->subs->dev, u->buffer_size, + u->urb->transfer_buffer, + u->urb->transfer_dma); + usb_free_urb(u->urb); + u->urb = NULL; + } +} + +/* + * wait until all urbs are processed. + */ +static int wait_clear_urbs(struct snd_usb_substream *subs) +{ + unsigned long end_time = jiffies + msecs_to_jiffies(1000); + unsigned int i; + int alive; + + do { + alive = 0; + for (i = 0; i < subs->nurbs; i++) { + if (test_bit(i, &subs->active_mask)) + alive++; + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (test_bit(i + 16, &subs->active_mask)) + alive++; + } + } + if (! alive) + break; + schedule_timeout_uninterruptible(1); + } while (time_before(jiffies, end_time)); + if (alive) + snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); + return 0; +} + +/* + * release a substream + */ +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force) +{ + int i; + + /* stop urbs (to be sure) */ + deactivate_urbs(subs, force, 1); + wait_clear_urbs(subs); + + for (i = 0; i < MAX_URBS; i++) + release_urb_ctx(&subs->dataurb[i]); + for (i = 0; i < SYNC_URBS; i++) + release_urb_ctx(&subs->syncurb[i]); + usb_buffer_free(subs->dev, SYNC_URBS * 4, + subs->syncbuf, subs->sync_dma); + subs->syncbuf = NULL; + subs->nurbs = 0; +} + +/* + * complete callback from data urb + */ +static void snd_complete_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } + } +} + + +/* + * complete callback from sync urb + */ +static void snd_complete_sync_urb(struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + struct snd_usb_substream *subs = ctx->subs; + struct snd_pcm_substream *substream = ctx->subs->pcm_substream; + int err = 0; + + if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || + !subs->running || /* can be stopped during retire callback */ + (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || + (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { + clear_bit(ctx->index + 16, &subs->active_mask); + if (err < 0) { + snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + } + } +} + + +/* + * initialize a substream for plaback/capture + */ +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits) +{ + unsigned int maxsize, i; + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int urb_packs, total_packs, packs_per_ms; + struct snd_usb_audio *chip = subs->stream->chip; + + /* calculate the frequency in 16.16 format */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) + subs->freqn = get_usb_full_speed_rate(rate); + else + subs->freqn = get_usb_high_speed_rate(rate); + subs->freqm = subs->freqn; + /* calculate max. frequency */ + if (subs->maxpacksize) { + /* whatever fits into a max. size packet */ + maxsize = subs->maxpacksize; + subs->freqmax = (maxsize / (frame_bits >> 3)) + << (16 - subs->datainterval); + } else { + /* no max. packet size: just take 25% higher than nominal */ + subs->freqmax = subs->freqn + (subs->freqn >> 2); + maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - subs->datainterval); + } + subs->phase = 0; + + if (subs->fill_max) + subs->curpacksize = subs->maxpacksize; + else + subs->curpacksize = maxsize; + + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + packs_per_ms = 8 >> subs->datainterval; + else + packs_per_ms = 1; + + if (is_playback) { + urb_packs = max(chip->nrpacks, 1); + urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); + } else + urb_packs = 1; + urb_packs *= packs_per_ms; + if (subs->syncpipe) + urb_packs = min(urb_packs, 1U << subs->syncinterval); + + /* decide how many packets to be used */ + if (is_playback) { + unsigned int minsize, maxpacks; + /* determine how small a packet can be */ + minsize = (subs->freqn >> (16 - subs->datainterval)) + * (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (subs->syncpipe) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + total_packs = (period_bytes + minsize - 1) / minsize; + /* we need at least two URBs for queueing */ + if (total_packs < 2) { + total_packs = 2; + } else { + /* and we don't want too long a queue either */ + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); + } + } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + total_packs = MAX_URBS * urb_packs; + } + subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; + if (subs->nurbs > MAX_URBS) { + /* too much... */ + subs->nurbs = MAX_URBS; + total_packs = MAX_URBS * urb_packs; + } else if (subs->nurbs < 2) { + /* too little - we need at least two packets + * to ensure contiguous playback/capture + */ + subs->nurbs = 2; + } + + /* allocate and initialize data urbs */ + for (i = 0; i < subs->nurbs; i++) { + struct snd_urb_ctx *u = &subs->dataurb[i]; + u->index = i; + u->subs = subs; + u->packets = (i + 1) * total_packs / subs->nurbs + - i * total_packs / subs->nurbs; + u->buffer_size = maxsize * u->packets; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) + u->packets++; /* for transfer delimiter */ + u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = + usb_buffer_alloc(subs->dev, u->buffer_size, GFP_KERNEL, + &u->urb->transfer_dma); + if (!u->urb->transfer_buffer) + goto out_of_memory; + u->urb->pipe = subs->datapipe; + u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; + u->urb->interval = 1 << subs->datainterval; + u->urb->context = u; + u->urb->complete = snd_complete_urb; + } + + if (subs->syncpipe) { + /* allocate and initialize sync urbs */ + subs->syncbuf = usb_buffer_alloc(subs->dev, SYNC_URBS * 4, + GFP_KERNEL, &subs->sync_dma); + if (!subs->syncbuf) + goto out_of_memory; + for (i = 0; i < SYNC_URBS; i++) { + struct snd_urb_ctx *u = &subs->syncurb[i]; + u->index = i; + u->subs = subs; + u->packets = 1; + u->urb = usb_alloc_urb(1, GFP_KERNEL); + if (!u->urb) + goto out_of_memory; + u->urb->transfer_buffer = subs->syncbuf + i * 4; + u->urb->transfer_dma = subs->sync_dma + i * 4; + u->urb->transfer_buffer_length = 4; + u->urb->pipe = subs->syncpipe; + u->urb->transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + u->urb->number_of_packets = 1; + u->urb->interval = 1 << subs->syncinterval; + u->urb->context = u; + u->urb->complete = snd_complete_sync_urb; + } + } + return 0; + +out_of_memory: + snd_usb_release_substream_urbs(subs, 0); + return -ENOMEM; +} + +/* + * prepare urb for full speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 10.14 frequency is passed through the pipe. + */ +static int prepare_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn >> 2; + cp[1] = subs->freqn >> 10; + cp[2] = subs->freqn >> 18; + return 0; +} + +/* + * prepare urb for high speed capture sync pipe + * + * fill the length and offset of each urb descriptor. + * the fixed 12.13 frequency is passed as 16.16 through the pipe. + */ +static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned char *cp = urb->transfer_buffer; + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + cp[0] = subs->freqn; + cp[1] = subs->freqn >> 8; + cp[2] = subs->freqn >> 16; + cp[3] = subs->freqn >> 24; + return 0; +} + +/* + * process after capture sync complete + * - nothing to do + */ +static int retire_capture_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; +} + +/* + * prepare urb for capture data pipe + * + * fill the offset and length of each descriptor. + * + * we use a temporary buffer to write the captured data. + * since the length of written data is determined by host, we cannot + * write onto the pcm buffer directly... the data is thus copied + * later at complete callback to the global buffer. + */ +static int prepare_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + int i, offs; + struct snd_urb_ctx *ctx = urb->context; + + offs = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + for (i = 0; i < ctx->packets; i++) { + urb->iso_frame_desc[i].offset = offs; + urb->iso_frame_desc[i].length = subs->curpacksize; + offs += subs->curpacksize; + } + urb->transfer_buffer_length = offs; + urb->number_of_packets = ctx->packets; + return 0; +} + +/* + * process after capture complete + * + * copy the data from each desctiptor to the pcm buffer, and + * update the current position. + */ +static int retire_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + unsigned char *cp; + int i; + unsigned int stride, frames, bytes, oldptr; + int period_elapsed = 0; + + stride = runtime->frame_bits >> 3; + + for (i = 0; i < urb->number_of_packets; i++) { + cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; + if (urb->iso_frame_desc[i].status) { + snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); + // continue; + } + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } + /* update the current pointer */ + spin_lock_irqsave(&subs->lock, flags); + oldptr = subs->hwptr_done; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; + subs->transfer_done += frames; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + } + spin_unlock_irqrestore(&subs->lock, flags); + /* copy a data chunk */ + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); + } else { + memcpy(runtime->dma_area + oldptr, cp, bytes); + } + } + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} + +/* + * Process after capture complete when paused. Nothing to do. + */ +static int retire_paused_capture_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + return 0; +} + + +/* + * prepare urb for full speed playback sync pipe + * + * set up the offset and length to receive the current frequency. + */ + +static int prepare_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].offset = 0; + return 0; +} + +/* + * prepare urb for high speed playback sync pipe + * + * set up the offset and length to receive the current frequency. + */ + +static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + struct snd_urb_ctx *ctx = urb->context; + + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->iso_frame_desc[0].length = 4; + urb->iso_frame_desc[0].offset = 0; + return 0; +} + +/* + * process after full speed playback sync complete + * + * retrieve the current 10.14 frequency from pipe, and set it. + * the value is referred in prepare_playback_urb(). + */ +static int retire_playback_sync_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + unsigned long flags; + + if (urb->iso_frame_desc[0].status == 0 && + urb->iso_frame_desc[0].actual_length == 3) { + f = combine_triple((u8*)urb->transfer_buffer) << 2; + if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } + } + + return 0; +} + +/* + * process after high speed playback sync complete + * + * retrieve the current 12.13 frequency from pipe, and set it. + * the value is referred in prepare_playback_urb(). + */ +static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + unsigned long flags; + + if (urb->iso_frame_desc[0].status == 0 && + urb->iso_frame_desc[0].actual_length == 4) { + f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; + if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } + } + + return 0; +} + +/* + * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete + * + * These devices return the number of samples per packet instead of the number + * of samples per microframe. + */ +static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int f; + unsigned long flags; + + if (urb->iso_frame_desc[0].status == 0 && + urb->iso_frame_desc[0].actual_length == 4) { + f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; + f >>= subs->datainterval; + if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } + } + + return 0; +} + +/* determine the number of frames in the next packet */ +static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) +{ + if (subs->fill_max) + return subs->maxframesize; + else { + subs->phase = (subs->phase & 0xffff) + + (subs->freqm << subs->datainterval); + return min(subs->phase >> 16, subs->maxframesize); + } +} + +/* + * Prepare urb for streaming before playback starts or when paused. + * + * We don't have any data, so we send silence. + */ +static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned int i, offs, counts; + struct snd_urb_ctx *ctx = urb->context; + int stride = runtime->frame_bits >> 3; + + offs = 0; + urb->dev = ctx->subs->dev; + for (i = 0; i < ctx->packets; ++i) { + counts = snd_usb_audio_next_packet_size(subs); + urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].length = counts * stride; + offs += counts; + } + urb->number_of_packets = ctx->packets; + urb->transfer_buffer_length = offs * stride; + memset(urb->transfer_buffer, + runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, + offs * stride); + return 0; +} + +/* + * prepare urb for playback data pipe + * + * Since a URB can handle only a single linear buffer, we must use double + * buffering when the data to be transferred overflows the buffer boundary. + * To avoid inconsistencies when updating hwptr_done, we use double buffering + * for all URBs. + */ +static int prepare_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + int i, stride; + unsigned int counts, frames, bytes; + unsigned long flags; + int period_elapsed = 0; + struct snd_urb_ctx *ctx = urb->context; + + stride = runtime->frame_bits >> 3; + + frames = 0; + urb->dev = ctx->subs->dev; /* we need to set this at each time */ + urb->number_of_packets = 0; + spin_lock_irqsave(&subs->lock, flags); + for (i = 0; i < ctx->packets; i++) { + counts = snd_usb_audio_next_packet_size(subs); + /* set up descriptor */ + urb->iso_frame_desc[i].offset = frames * stride; + urb->iso_frame_desc[i].length = counts * stride; + frames += counts; + urb->number_of_packets++; + subs->transfer_done += counts; + if (subs->transfer_done >= runtime->period_size) { + subs->transfer_done -= runtime->period_size; + period_elapsed = 1; + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { + if (subs->transfer_done > 0) { + /* FIXME: fill-max mode is not + * supported yet */ + frames -= subs->transfer_done; + counts -= subs->transfer_done; + urb->iso_frame_desc[i].length = + counts * stride; + subs->transfer_done = 0; + } + i++; + if (i < ctx->packets) { + /* add a transfer delimiter */ + urb->iso_frame_desc[i].offset = + frames * stride; + urb->iso_frame_desc[i].length = 0; + urb->number_of_packets++; + } + break; + } + } + if (period_elapsed) /* finish at the period boundary */ + break; + } + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { + /* err, the transferred area goes over buffer boundary. */ + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); + } else { + memcpy(urb->transfer_buffer, + runtime->dma_area + subs->hwptr_done, bytes); + } + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; + spin_unlock_irqrestore(&subs->lock, flags); + urb->transfer_buffer_length = bytes; + if (period_elapsed) + snd_pcm_period_elapsed(subs->pcm_substream); + return 0; +} + +/* + * process after playback data complete + * - decrease the delay count again + */ +static int retire_playback_urb(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime, + struct urb *urb) +{ + unsigned long flags; + int stride = runtime->frame_bits >> 3; + int processed = urb->transfer_buffer_length / stride; + + spin_lock_irqsave(&subs->lock, flags); + if (processed > runtime->delay) + runtime->delay = 0; + else + runtime->delay -= processed; + spin_unlock_irqrestore(&subs->lock, flags); + return 0; +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +/* + * set up and start data/sync urbs + */ +static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) +{ + unsigned int i; + int err; + + if (subs->stream->chip->shutdown) + return -EBADFD; + + for (i = 0; i < subs->nurbs; i++) { + if (snd_BUG_ON(!subs->dataurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); + goto __error; + } + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + if (snd_BUG_ON(!subs->syncurb[i].urb)) + return -EINVAL; + if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { + snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); + goto __error; + } + } + } + + subs->active_mask = 0; + subs->unlink_mask = 0; + subs->running = 1; + for (i = 0; i < subs->nurbs; i++) { + err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit datapipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i, &subs->active_mask); + } + if (subs->syncpipe) { + for (i = 0; i < SYNC_URBS; i++) { + err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit syncpipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); + goto __error; + } + set_bit(i + 16, &subs->active_mask); + } + } + return 0; + + __error: + // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + deactivate_urbs(subs, 0, 0); + return -EPIPE; +} + + +/* + */ +static struct snd_urb_ops audio_urb_ops[2] = { + { + .prepare = prepare_nodata_playback_urb, + .retire = retire_playback_urb, + .prepare_sync = prepare_playback_sync_urb, + .retire_sync = retire_playback_sync_urb, + }, + { + .prepare = prepare_capture_urb, + .retire = retire_capture_urb, + .prepare_sync = prepare_capture_sync_urb, + .retire_sync = retire_capture_sync_urb, + }, +}; + +static struct snd_urb_ops audio_urb_ops_high_speed[2] = { + { + .prepare = prepare_nodata_playback_urb, + .retire = retire_playback_urb, + .prepare_sync = prepare_playback_sync_urb_hs, + .retire_sync = retire_playback_sync_urb_hs, + }, + { + .prepare = prepare_capture_urb, + .retire = retire_capture_urb, + .prepare_sync = prepare_capture_sync_urb_hs, + .retire_sync = retire_capture_sync_urb, + }, +}; + +/* + * initialize the substream instance. + */ + +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, struct audioformat *fp) +{ + struct snd_usb_substream *subs = &as->substream[stream]; + + INIT_LIST_HEAD(&subs->fmt_list); + spin_lock_init(&subs->lock); + + subs->stream = as; + subs->direction = stream; + subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; + if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { + subs->ops = audio_urb_ops[stream]; + } else { + subs->ops = audio_urb_ops_high_speed[stream]; + switch (as->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; + break; + } + } + + snd_usb_set_pcm_ops(as->pcm, stream); + + list_add_tail(&fp->list, &subs->fmt_list); + subs->formats |= fp->formats; + subs->endpoint = fp->endpoint; + subs->num_formats++; + subs->fmt_type = fp->fmt_type; +} + +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.prepare = prepare_playback_urb; + return 0; + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.prepare = prepare_nodata_playback_urb; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + subs->ops.retire = retire_capture_urb; + return start_urbs(subs, substream->runtime); + case SNDRV_PCM_TRIGGER_STOP: + return deactivate_urbs(subs, 0, 0); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + subs->ops.retire = retire_paused_capture_urb; + return 0; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + subs->ops.retire = retire_capture_urb; + return 0; + } + + return -EINVAL; +} + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime) +{ + /* clear urbs (to be sure) */ + deactivate_urbs(subs, 0, 1); + wait_clear_urbs(subs); + + /* for playback, submit the URBs now; otherwise, the first hwptr_done + * updates for all URBs would happen at the same time when starting */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + subs->ops.prepare = prepare_nodata_playback_urb; + return start_urbs(subs, runtime); + } + + return 0; +} + diff --git a/sound/usb/urb.h b/sound/usb/urb.h new file mode 100644 index 000000000000..888da38079cf --- /dev/null +++ b/sound/usb/urb.h @@ -0,0 +1,21 @@ +#ifndef __USBAUDIO_URB_H +#define __USBAUDIO_URB_H + +void snd_usb_init_substream(struct snd_usb_stream *as, + int stream, + struct audioformat *fp); + +int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, + unsigned int period_bytes, + unsigned int rate, + unsigned int frame_bits); + +void snd_usb_release_substream_urbs(struct snd_usb_substream *subs, int force); + +int snd_usb_substream_prepare(struct snd_usb_substream *subs, + struct snd_pcm_runtime *runtime); + +int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd); +int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd); + +#endif /* __USBAUDIO_URB_H */ diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c deleted file mode 100644 index 11b0826b8fe6..000000000000 --- a/sound/usb/usbaudio.c +++ /dev/null @@ -1,4050 +0,0 @@ -/* - * (Tentative) USB Audio Driver for ALSA - * - * Main and PCM part - * - * Copyright (c) 2002 by Takashi Iwai <tiwai@suse.de> - * - * Many codes borrowed from audio.c by - * Alan Cox (alan@lxorguk.ukuu.org.uk) - * Thomas Sailer (sailer@ife.ee.ethz.ch) - * - * - * 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 - * - * - * NOTES: - * - * - async unlink should be used for avoiding the sleep inside lock. - * 2.4.22 usb-uhci seems buggy for async unlinking and results in - * oops. in such a cse, pass async_unlink=0 option. - * - the linked URBs would be preferred but not used so far because of - * the instability of unlinking. - * - type II is not supported properly. there is no device which supports - * this type *correctly*. SB extigy looks as if it supports, but it's - * indeed an AC3 stream packed in SPDIF frames (i.e. no real AC3 stream). - */ - - -#include <linux/bitops.h> -#include <linux/init.h> -#include <linux/list.h> -#include <linux/slab.h> -#include <linux/string.h> -#include <linux/usb.h> -#include <linux/moduleparam.h> -#include <linux/mutex.h> -#include <linux/usb/audio.h> -#include <linux/usb/ch9.h> - -#include <sound/core.h> -#include <sound/info.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> - -#include "usbaudio.h" - - -MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); -MODULE_DESCRIPTION("USB Audio"); -MODULE_LICENSE("GPL"); -MODULE_SUPPORTED_DEVICE("{{Generic,USB Audio}}"); - - -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 */ -/* Vendor/product IDs for this card */ -static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; -static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; -static int nrpacks = 8; /* max. number of packets per urb */ -static int async_unlink = 1; -static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/ -static int ignore_ctl_error; - -module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); -module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for the USB audio adapter."); -module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable USB audio adapter."); -module_param_array(vid, int, NULL, 0444); -MODULE_PARM_DESC(vid, "Vendor ID for the USB audio device."); -module_param_array(pid, int, NULL, 0444); -MODULE_PARM_DESC(pid, "Product ID for the USB audio device."); -module_param(nrpacks, int, 0644); -MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); -module_param(async_unlink, bool, 0444); -MODULE_PARM_DESC(async_unlink, "Use async unlink mode."); -module_param_array(device_setup, int, NULL, 0444); -MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); -module_param(ignore_ctl_error, bool, 0444); -MODULE_PARM_DESC(ignore_ctl_error, - "Ignore errors from USB controller for mixer interfaces."); - -/* - * debug the h/w constraints - */ -/* #define HW_CONST_DEBUG */ - - -/* - * - */ - -#define MAX_PACKS 20 -#define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ -#define MAX_URBS 8 -#define SYNC_URBS 4 /* always four urbs for sync */ -#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ - -struct audioformat { - struct list_head list; - snd_pcm_format_t format; /* format type */ - unsigned int channels; /* # channels */ - unsigned int fmt_type; /* USB audio format type (1-3) */ - unsigned int frame_size; /* samples per frame for non-audio */ - int iface; /* interface number */ - unsigned char altsetting; /* corresponding alternate setting */ - unsigned char altset_idx; /* array index of altenate setting */ - unsigned char attributes; /* corresponding attributes of cs endpoint */ - unsigned char endpoint; /* endpoint */ - unsigned char ep_attr; /* endpoint attributes */ - unsigned char datainterval; /* log_2 of data packet interval */ - unsigned int maxpacksize; /* max. packet size */ - unsigned int rates; /* rate bitmasks */ - unsigned int rate_min, rate_max; /* min/max rates */ - unsigned int nr_rates; /* number of rate table entries */ - unsigned int *rate_table; /* rate table */ -}; - -struct snd_usb_substream; - -struct snd_urb_ctx { - struct urb *urb; - unsigned int buffer_size; /* size of data buffer, if data URB */ - struct snd_usb_substream *subs; - int index; /* index for urb array */ - int packets; /* number of packets per urb */ -}; - -struct snd_urb_ops { - int (*prepare)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*prepare_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); - int (*retire_sync)(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *u); -}; - -struct snd_usb_substream { - struct snd_usb_stream *stream; - struct usb_device *dev; - struct snd_pcm_substream *pcm_substream; - int direction; /* playback or capture */ - int interface; /* current interface */ - int endpoint; /* assigned endpoint */ - struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ - unsigned int cur_rate; /* current rate (for hw_params callback) */ - unsigned int period_bytes; /* current period bytes (for hw_params callback) */ - unsigned int format; /* USB data format */ - unsigned int datapipe; /* the data i/o pipe */ - unsigned int syncpipe; /* 1 - async out or adaptive in */ - unsigned int datainterval; /* log_2 of data packet interval */ - unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ - unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ - unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ - unsigned int freqmax; /* maximum sampling rate, used for buffer management */ - unsigned int phase; /* phase accumulator */ - unsigned int maxpacksize; /* max packet size in bytes */ - unsigned int maxframesize; /* max packet size in frames */ - unsigned int curpacksize; /* current packet size in bytes (for capture) */ - unsigned int curframesize; /* current packet size in frames (for capture) */ - unsigned int fill_max: 1; /* fill max packet size always */ - unsigned int txfr_quirk:1; /* allow sub-frame alignment */ - unsigned int fmt_type; /* USB audio format type (1-3) */ - - unsigned int running: 1; /* running status */ - - unsigned int hwptr_done; /* processed byte position in the buffer */ - unsigned int transfer_done; /* processed frames since last period update */ - unsigned long active_mask; /* bitmask of active urbs */ - unsigned long unlink_mask; /* bitmask of unlinked urbs */ - - unsigned int nurbs; /* # urbs */ - struct snd_urb_ctx dataurb[MAX_URBS]; /* data urb table */ - struct snd_urb_ctx syncurb[SYNC_URBS]; /* sync urb table */ - char *syncbuf; /* sync buffer for all sync URBs */ - dma_addr_t sync_dma; /* DMA address of syncbuf */ - - u64 formats; /* format bitmasks (all or'ed) */ - unsigned int num_formats; /* number of supported audio formats (list) */ - struct list_head fmt_list; /* format list */ - struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ - spinlock_t lock; - - struct snd_urb_ops ops; /* callbacks (must be filled at init) */ -}; - - -struct snd_usb_stream { - struct snd_usb_audio *chip; - struct snd_pcm *pcm; - int pcm_index; - unsigned int fmt_type; /* USB audio format type (1-3) */ - struct snd_usb_substream substream[2]; - struct list_head list; -}; - - -/* - * we keep the snd_usb_audio_t instances by ourselves for merging - * the all interfaces on the same card as one sound device. - */ - -static DEFINE_MUTEX(register_mutex); -static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; - - -/* - * convert a sampling rate into our full speed format (fs/1000 in Q16.16) - * this will overflow at approx 524 kHz - */ -static inline unsigned get_usb_full_speed_rate(unsigned int rate) -{ - return ((rate << 13) + 62) / 125; -} - -/* - * convert a sampling rate into USB high speed format (fs/8000 in Q16.16) - * this will overflow at approx 4 MHz - */ -static inline unsigned get_usb_high_speed_rate(unsigned int rate) -{ - return ((rate << 10) + 62) / 125; -} - -/* convert our full speed USB rate into sampling rate in Hz */ -static inline unsigned get_full_speed_hz(unsigned int usb_rate) -{ - return (usb_rate * 125 + (1 << 12)) >> 13; -} - -/* convert our high speed USB rate into sampling rate in Hz */ -static inline unsigned get_high_speed_hz(unsigned int usb_rate) -{ - return (usb_rate * 125 + (1 << 9)) >> 10; -} - - -/* - * prepare urb for full speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 10.14 frequency is passed through the pipe. - */ -static int prepare_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn >> 2; - cp[1] = subs->freqn >> 10; - cp[2] = subs->freqn >> 18; - return 0; -} - -/* - * prepare urb for high speed capture sync pipe - * - * fill the length and offset of each urb descriptor. - * the fixed 12.13 frequency is passed as 16.16 through the pipe. - */ -static int prepare_capture_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned char *cp = urb->transfer_buffer; - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - cp[0] = subs->freqn; - cp[1] = subs->freqn >> 8; - cp[2] = subs->freqn >> 16; - cp[3] = subs->freqn >> 24; - return 0; -} - -/* - * process after capture sync complete - * - nothing to do - */ -static int retire_capture_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - -/* - * prepare urb for capture data pipe - * - * fill the offset and length of each descriptor. - * - * we use a temporary buffer to write the captured data. - * since the length of written data is determined by host, we cannot - * write onto the pcm buffer directly... the data is thus copied - * later at complete callback to the global buffer. - */ -static int prepare_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, offs; - struct snd_urb_ctx *ctx = urb->context; - - offs = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - for (i = 0; i < ctx->packets; i++) { - urb->iso_frame_desc[i].offset = offs; - urb->iso_frame_desc[i].length = subs->curpacksize; - offs += subs->curpacksize; - } - urb->transfer_buffer_length = offs; - urb->number_of_packets = ctx->packets; - return 0; -} - -/* - * process after capture complete - * - * copy the data from each desctiptor to the pcm buffer, and - * update the current position. - */ -static int retire_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - unsigned char *cp; - int i; - unsigned int stride, frames, bytes, oldptr; - int period_elapsed = 0; - - stride = runtime->frame_bits >> 3; - - for (i = 0; i < urb->number_of_packets; i++) { - cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; - if (urb->iso_frame_desc[i].status) { - snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); - // continue; - } - bytes = urb->iso_frame_desc[i].actual_length; - frames = bytes / stride; - if (!subs->txfr_quirk) - bytes = frames * stride; - if (bytes % (runtime->sample_bits >> 3) != 0) { -#ifdef CONFIG_SND_DEBUG_VERBOSE - int oldbytes = bytes; -#endif - bytes = frames * stride; - snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", - oldbytes, bytes); - } - /* update the current pointer */ - spin_lock_irqsave(&subs->lock, flags); - oldptr = subs->hwptr_done; - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - frames = (bytes + (oldptr % stride)) / stride; - subs->transfer_done += frames; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - } - spin_unlock_irqrestore(&subs->lock, flags); - /* copy a data chunk */ - if (oldptr + bytes > runtime->buffer_size * stride) { - unsigned int bytes1 = - runtime->buffer_size * stride - oldptr; - memcpy(runtime->dma_area + oldptr, cp, bytes1); - memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); - } else { - memcpy(runtime->dma_area + oldptr, cp, bytes); - } - } - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * Process after capture complete when paused. Nothing to do. - */ -static int retire_paused_capture_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - return 0; -} - - -/* - * prepare urb for full speed playback sync pipe - * - * set up the offset and length to receive the current frequency. - */ - -static int prepare_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * prepare urb for high speed playback sync pipe - * - * set up the offset and length to receive the current frequency. - */ - -static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after full speed playback sync complete - * - * retrieve the current 10.14 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). - */ -static int retire_playback_sync_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 3) { - f = combine_triple((u8*)urb->transfer_buffer) << 2; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} - -/* - * process after high speed playback sync complete - * - * retrieve the current 12.13 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). - */ -static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} - -/* - * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete - * - * These devices return the number of samples per packet instead of the number - * of samples per microframe. - */ -static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - f >>= subs->datainterval; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} - -/* determine the number of frames in the next packet */ -static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) -{ - if (subs->fill_max) - return subs->maxframesize; - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - return min(subs->phase >> 16, subs->maxframesize); - } -} - -/* - * Prepare urb for streaming before playback starts or when paused. - * - * We don't have any data, so we send silence. - */ -static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int i, offs, counts; - struct snd_urb_ctx *ctx = urb->context; - int stride = runtime->frame_bits >> 3; - - offs = 0; - urb->dev = ctx->subs->dev; - for (i = 0; i < ctx->packets; ++i) { - counts = snd_usb_audio_next_packet_size(subs); - urb->iso_frame_desc[i].offset = offs * stride; - urb->iso_frame_desc[i].length = counts * stride; - offs += counts; - } - urb->number_of_packets = ctx->packets; - urb->transfer_buffer_length = offs * stride; - memset(urb->transfer_buffer, - subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, - offs * stride); - return 0; -} - -/* - * prepare urb for playback data pipe - * - * Since a URB can handle only a single linear buffer, we must use double - * buffering when the data to be transferred overflows the buffer boundary. - * To avoid inconsistencies when updating hwptr_done, we use double buffering - * for all URBs. - */ -static int prepare_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - int i, stride; - unsigned int counts, frames, bytes; - unsigned long flags; - int period_elapsed = 0; - struct snd_urb_ctx *ctx = urb->context; - - stride = runtime->frame_bits >> 3; - - frames = 0; - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->number_of_packets = 0; - spin_lock_irqsave(&subs->lock, flags); - for (i = 0; i < ctx->packets; i++) { - counts = snd_usb_audio_next_packet_size(subs); - /* set up descriptor */ - urb->iso_frame_desc[i].offset = frames * stride; - urb->iso_frame_desc[i].length = counts * stride; - frames += counts; - urb->number_of_packets++; - subs->transfer_done += counts; - if (subs->transfer_done >= runtime->period_size) { - subs->transfer_done -= runtime->period_size; - period_elapsed = 1; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) { - if (subs->transfer_done > 0) { - /* FIXME: fill-max mode is not - * supported yet */ - frames -= subs->transfer_done; - counts -= subs->transfer_done; - urb->iso_frame_desc[i].length = - counts * stride; - subs->transfer_done = 0; - } - i++; - if (i < ctx->packets) { - /* add a transfer delimiter */ - urb->iso_frame_desc[i].offset = - frames * stride; - urb->iso_frame_desc[i].length = 0; - urb->number_of_packets++; - } - break; - } - } - if (period_elapsed) /* finish at the period boundary */ - break; - } - bytes = frames * stride; - if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { - /* err, the transferred area goes over buffer boundary. */ - unsigned int bytes1 = - runtime->buffer_size * stride - subs->hwptr_done; - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes1); - memcpy(urb->transfer_buffer + bytes1, - runtime->dma_area, bytes - bytes1); - } else { - memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done, bytes); - } - subs->hwptr_done += bytes; - if (subs->hwptr_done >= runtime->buffer_size * stride) - subs->hwptr_done -= runtime->buffer_size * stride; - runtime->delay += frames; - spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = bytes; - if (period_elapsed) - snd_pcm_period_elapsed(subs->pcm_substream); - return 0; -} - -/* - * process after playback data complete - * - decrease the delay count again - */ -static int retire_playback_urb(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned long flags; - int stride = runtime->frame_bits >> 3; - int processed = urb->transfer_buffer_length / stride; - - spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; - else - runtime->delay -= processed; - spin_unlock_irqrestore(&subs->lock, flags); - return 0; -} - - -/* - */ -static struct snd_urb_ops audio_urb_ops[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb, - .retire_sync = retire_playback_sync_urb, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb, - .retire_sync = retire_capture_sync_urb, - }, -}; - -static struct snd_urb_ops audio_urb_ops_high_speed[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb_hs, - .retire_sync = retire_playback_sync_urb_hs, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb_hs, - .retire_sync = retire_capture_sync_urb, - }, -}; - -/* - * complete callback from data urb - */ -static void snd_complete_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * complete callback from sync urb - */ -static void snd_complete_sync_urb(struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - struct snd_usb_substream *subs = ctx->subs; - struct snd_pcm_substream *substream = ctx->subs->pcm_substream; - int err = 0; - - if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || - !subs->running || /* can be stopped during retire callback */ - (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || - (err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { - clear_bit(ctx->index + 16, &subs->active_mask); - if (err < 0) { - snd_printd(KERN_ERR "cannot submit sync urb (err = %d)\n", err); - snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - } - } -} - - -/* - * unlink active urbs. - */ -static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sleep) -{ - unsigned int i; - int async; - - subs->running = 0; - - if (!force && subs->stream->chip->shutdown) /* to be sure... */ - return -EBADFD; - - async = !can_sleep && async_unlink; - - if (!async && in_interrupt()) - return 0; - - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) { - if (!test_and_set_bit(i, &subs->unlink_mask)) { - struct urb *u = subs->dataurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i+16, &subs->active_mask)) { - if (!test_and_set_bit(i+16, &subs->unlink_mask)) { - struct urb *u = subs->syncurb[i].urb; - if (async) - usb_unlink_urb(u); - else - usb_kill_urb(u); - } - } - } - } - return 0; -} - - -static const char *usb_error_string(int err) -{ - switch (err) { - case -ENODEV: - return "no device"; - case -ENOENT: - return "endpoint not enabled"; - case -EPIPE: - return "endpoint stalled"; - case -ENOSPC: - return "not enough bandwidth"; - case -ESHUTDOWN: - return "device disabled"; - case -EHOSTUNREACH: - return "device suspended"; - case -EINVAL: - case -EAGAIN: - case -EFBIG: - case -EMSGSIZE: - return "internal error"; - default: - return "unknown error"; - } -} - -/* - * set up and start data/sync urbs - */ -static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime) -{ - unsigned int i; - int err; - - if (subs->stream->chip->shutdown) - return -EBADFD; - - for (i = 0; i < subs->nurbs; i++) { - if (snd_BUG_ON(!subs->dataurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare(subs, runtime, subs->dataurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare datapipe for urb %d\n", i); - goto __error; - } - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (snd_BUG_ON(!subs->syncurb[i].urb)) - return -EINVAL; - if (subs->ops.prepare_sync(subs, runtime, subs->syncurb[i].urb) < 0) { - snd_printk(KERN_ERR "cannot prepare syncpipe for urb %d\n", i); - goto __error; - } - } - } - - subs->active_mask = 0; - subs->unlink_mask = 0; - subs->running = 1; - for (i = 0; i < subs->nurbs; i++) { - err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i, &subs->active_mask); - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); - if (err < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe " - "for urb %d, error %d: %s\n", - i, err, usb_error_string(err)); - goto __error; - } - set_bit(i + 16, &subs->active_mask); - } - } - return 0; - - __error: - // snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); - deactivate_urbs(subs, 0, 0); - return -EPIPE; -} - - -/* - * wait until all urbs are processed. - */ -static int wait_clear_urbs(struct snd_usb_substream *subs) -{ - unsigned long end_time = jiffies + msecs_to_jiffies(1000); - unsigned int i; - int alive; - - do { - alive = 0; - for (i = 0; i < subs->nurbs; i++) { - if (test_bit(i, &subs->active_mask)) - alive++; - } - if (subs->syncpipe) { - for (i = 0; i < SYNC_URBS; i++) { - if (test_bit(i + 16, &subs->active_mask)) - alive++; - } - } - if (! alive) - break; - schedule_timeout_uninterruptible(1); - } while (time_before(jiffies, end_time)); - if (alive) - snd_printk(KERN_ERR "timeout: still %d active urbs..\n", alive); - return 0; -} - - -/* - * return the current pcm pointer. just based on the hwptr_done value. - */ -static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_usb_substream *subs; - unsigned int hwptr_done; - - subs = (struct snd_usb_substream *)substream->runtime->private_data; - spin_lock(&subs->lock); - hwptr_done = subs->hwptr_done; - spin_unlock(&subs->lock); - return hwptr_done / (substream->runtime->frame_bits >> 3); -} - - -/* - * start/stop playback substream - */ -static int snd_usb_pcm_playback_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.prepare = prepare_playback_urb; - return 0; - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.prepare = prepare_nodata_playback_urb; - return 0; - default: - return -EINVAL; - } -} - -/* - * start/stop capture substream - */ -static int snd_usb_pcm_capture_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - subs->ops.retire = retire_capture_urb; - return start_urbs(subs, substream->runtime); - case SNDRV_PCM_TRIGGER_STOP: - return deactivate_urbs(subs, 0, 0); - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - subs->ops.retire = retire_paused_capture_urb; - return 0; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - subs->ops.retire = retire_capture_urb; - return 0; - default: - return -EINVAL; - } -} - - -/* - * release a urb data - */ -static void release_urb_ctx(struct snd_urb_ctx *u) -{ - if (u->urb) { - if (u->buffer_size) - usb_buffer_free(u->subs->dev, u->buffer_size, - u->urb->transfer_buffer, - u->urb->transfer_dma); - usb_free_urb(u->urb); - u->urb = NULL; - } -} - -/* - * release a substream - */ -static void release_substream_urbs(struct snd_usb_substream *subs, int force) -{ - int i; - - /* stop urbs (to be sure) */ - deactivate_urbs(subs, force, 1); - wait_clear_urbs(subs); - - for (i = 0; i < MAX_URBS; i++) - release_urb_ctx(&subs->dataurb[i]); - for (i = 0; i < SYNC_URBS; i++) - release_urb_ctx(&subs->syncurb[i]); - usb_buffer_free(subs->dev, SYNC_URBS * 4, - subs->syncbuf, subs->sync_dma); - subs->syncbuf = NULL; - subs->nurbs = 0; -} - -/* - * initialize a substream for plaback/capture - */ -static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int period_bytes, - unsigned int rate, unsigned int frame_bits) -{ - unsigned int maxsize, i; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int urb_packs, total_packs, packs_per_ms; - - /* calculate the frequency in 16.16 format */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->freqn = get_usb_full_speed_rate(rate); - else - subs->freqn = get_usb_high_speed_rate(rate); - subs->freqm = subs->freqn; - /* calculate max. frequency */ - if (subs->maxpacksize) { - /* whatever fits into a max. size packet */ - maxsize = subs->maxpacksize; - subs->freqmax = (maxsize / (frame_bits >> 3)) - << (16 - subs->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - subs->freqmax = subs->freqn + (subs->freqn >> 2); - maxsize = ((subs->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - subs->datainterval); - } - subs->phase = 0; - - if (subs->fill_max) - subs->curpacksize = subs->maxpacksize; - else - subs->curpacksize = maxsize; - - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) - packs_per_ms = 8 >> subs->datainterval; - else - packs_per_ms = 1; - - if (is_playback) { - urb_packs = max(nrpacks, 1); - urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); - } else - urb_packs = 1; - urb_packs *= packs_per_ms; - if (subs->syncpipe) - urb_packs = min(urb_packs, 1U << subs->syncinterval); - - /* decide how many packets to be used */ - if (is_playback) { - unsigned int minsize, maxpacks; - /* determine how small a packet can be */ - minsize = (subs->freqn >> (16 - subs->datainterval)) - * (frame_bits >> 3); - /* with sync from device, assume it can be 12% lower */ - if (subs->syncpipe) - minsize -= minsize >> 3; - minsize = max(minsize, 1u); - total_packs = (period_bytes + minsize - 1) / minsize; - /* we need at least two URBs for queueing */ - if (total_packs < 2) { - total_packs = 2; - } else { - /* and we don't want too long a queue either */ - maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); - total_packs = min(total_packs, maxpacks); - } - } else { - while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) - urb_packs >>= 1; - total_packs = MAX_URBS * urb_packs; - } - subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; - if (subs->nurbs > MAX_URBS) { - /* too much... */ - subs->nurbs = MAX_URBS; - total_packs = MAX_URBS * urb_packs; - } else if (subs->nurbs < 2) { - /* too little - we need at least two packets - * to ensure contiguous playback/capture - */ - subs->nurbs = 2; - } - - /* allocate and initialize data urbs */ - for (i = 0; i < subs->nurbs; i++) { - struct snd_urb_ctx *u = &subs->dataurb[i]; - u->index = i; - u->subs = subs; - u->packets = (i + 1) * total_packs / subs->nurbs - - i * total_packs / subs->nurbs; - u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == UAC_FORMAT_TYPE_II) - u->packets++; /* for transfer delimiter */ - u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = - usb_buffer_alloc(subs->dev, u->buffer_size, GFP_KERNEL, - &u->urb->transfer_dma); - if (!u->urb->transfer_buffer) - goto out_of_memory; - u->urb->pipe = subs->datapipe; - u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP; - u->urb->interval = 1 << subs->datainterval; - u->urb->context = u; - u->urb->complete = snd_complete_urb; - } - - if (subs->syncpipe) { - /* allocate and initialize sync urbs */ - subs->syncbuf = usb_buffer_alloc(subs->dev, SYNC_URBS * 4, - GFP_KERNEL, &subs->sync_dma); - if (!subs->syncbuf) - goto out_of_memory; - for (i = 0; i < SYNC_URBS; i++) { - struct snd_urb_ctx *u = &subs->syncurb[i]; - u->index = i; - u->subs = subs; - u->packets = 1; - u->urb = usb_alloc_urb(1, GFP_KERNEL); - if (!u->urb) - goto out_of_memory; - u->urb->transfer_buffer = subs->syncbuf + i * 4; - u->urb->transfer_dma = subs->sync_dma + i * 4; - u->urb->transfer_buffer_length = 4; - u->urb->pipe = subs->syncpipe; - u->urb->transfer_flags = URB_ISO_ASAP | - URB_NO_TRANSFER_DMA_MAP; - u->urb->number_of_packets = 1; - u->urb->interval = 1 << subs->syncinterval; - u->urb->context = u; - u->urb->complete = snd_complete_sync_urb; - } - } - return 0; - -out_of_memory: - release_substream_urbs(subs, 0); - return -ENOMEM; -} - - -/* - * find a matching audio format - */ -static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned int format, - unsigned int rate, unsigned int channels) -{ - struct list_head *p; - struct audioformat *found = NULL; - int cur_attr = 0, attr; - - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - if (fp->format != format || fp->channels != channels) - continue; - if (rate < fp->rate_min || rate > fp->rate_max) - continue; - if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) { - unsigned int i; - for (i = 0; i < fp->nr_rates; i++) - if (fp->rate_table[i] == rate) - break; - if (i >= fp->nr_rates) - continue; - } - attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (! found) { - found = fp; - cur_attr = attr; - continue; - } - /* avoid async out and adaptive in if the other method - * supports the same format. - * this is a workaround for the case like - * M-audio audiophile USB. - */ - if (attr != cur_attr) { - if ((attr == USB_ENDPOINT_SYNC_ASYNC && - subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (attr == USB_ENDPOINT_SYNC_ADAPTIVE && - subs->direction == SNDRV_PCM_STREAM_CAPTURE)) - continue; - if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && - subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && - subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { - found = fp; - cur_attr = attr; - continue; - } - } - /* find the format with the largest max. packet size */ - if (fp->maxpacksize > found->maxpacksize) { - found = fp; - cur_attr = attr; - } - } - return found; -} - - -/* - * initialize the picth control and sample rate - */ -static int init_usb_pitch(struct usb_device *dev, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt) -{ - unsigned int ep; - unsigned char data[1]; - int err; - - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint has pitch control, enable it */ - if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) { - data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", - dev->devnum, iface, ep); - return err; - } - } - return 0; -} - -static int init_usb_sample_rate(struct usb_device *dev, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) -{ - unsigned int ep; - unsigned char data[3]; - int err; - - ep = get_endpoint(alts, 0)->bEndpointAddress; - /* if endpoint has sampling rate control, set it */ - if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) { - int crate; - data[0] = rate; - data[1] = rate >> 8; - data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", - dev->devnum, iface, fmt->altsetting, rate, ep); - return err; - } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, - USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", - dev->devnum, iface, fmt->altsetting, ep); - return 0; /* some devices don't support reading */ - } - crate = data[0] | (data[1] << 8) | (data[2] << 16); - if (crate != rate) { - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); - // runtime->rate = crate; - } - } - return 0; -} - -/* - * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, - * not for interface. - */ -static void set_format_emu_quirk(struct snd_usb_substream *subs, - struct audioformat *fmt) -{ - unsigned char emu_samplerate_id = 0; - - /* When capture is active - * sample rate shouldn't be changed - * by playback substream - */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) - return; - } - - switch (fmt->rate_min) { - case 48000: - emu_samplerate_id = EMU_QUIRK_SR_48000HZ; - break; - case 88200: - emu_samplerate_id = EMU_QUIRK_SR_88200HZ; - break; - case 96000: - emu_samplerate_id = EMU_QUIRK_SR_96000HZ; - break; - case 176400: - emu_samplerate_id = EMU_QUIRK_SR_176400HZ; - break; - case 192000: - emu_samplerate_id = EMU_QUIRK_SR_192000HZ; - break; - default: - emu_samplerate_id = EMU_QUIRK_SR_44100HZ; - break; - } - snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); -} - -/* - * find a matching format and set up the interface - */ -static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) -{ - struct usb_device *dev = subs->dev; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct usb_interface *iface; - unsigned int ep, attr; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err; - - iface = usb_ifnum_to_if(dev, fmt->iface); - if (WARN_ON(!iface)) - return -EINVAL; - alts = &iface->altsetting[fmt->altset_idx]; - altsd = get_iface_desc(alts); - if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) - return -EINVAL; - - if (fmt == subs->cur_audiofmt) - return 0; - - /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { - if (usb_set_interface(subs->dev, subs->interface, 0) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - subs->interface = -1; - subs->format = 0; - } - - /* set interface */ - if (subs->interface != fmt->iface || subs->format != fmt->altset_idx) { - if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EIO; - } - snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); - subs->interface = fmt->iface; - subs->format = fmt->altset_idx; - } - - /* create a data pipe */ - ep = fmt->endpoint & USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->datapipe = usb_sndisocpipe(dev, ep); - else - subs->datapipe = usb_rcvisocpipe(dev, ep); - subs->datainterval = fmt->datainterval; - subs->syncpipe = subs->syncinterval = 0; - subs->maxpacksize = fmt->maxpacksize; - subs->fill_max = 0; - - /* we need a sync pipe in async OUT or adaptive IN mode */ - /* check the number of EP, since some devices have broken - * descriptors which fool us. if it has only one EP, - * assume it as adaptive-out or sync-in. - */ - attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && - altsd->bNumEndpoints >= 2) { - /* check sync-pipe endpoint */ - /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ - if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != 0x01 || - (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EINVAL; - } - ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); - return -EINVAL; - } - ep &= USB_ENDPOINT_NUMBER_MASK; - if (is_playback) - subs->syncpipe = usb_rcvisocpipe(dev, ep); - else - subs->syncpipe = usb_sndisocpipe(dev, ep); - if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bRefresh >= 1 && - get_endpoint(alts, 1)->bRefresh <= 9) - subs->syncinterval = get_endpoint(alts, 1)->bRefresh; - else if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) - subs->syncinterval = 1; - else if (get_endpoint(alts, 1)->bInterval >= 1 && - get_endpoint(alts, 1)->bInterval <= 16) - subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; - else - subs->syncinterval = 3; - } - - /* always fill max packet size */ - if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) - subs->fill_max = 1; - - if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0) - return err; - - subs->cur_audiofmt = fmt; - - switch (subs->stream->chip->usb_id) { - case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ - case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ - case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ - set_format_emu_quirk(subs, fmt); - break; - } - -#if 0 - printk(KERN_DEBUG - "setting done: format = %d, rate = %d..%d, channels = %d\n", - fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); - printk(KERN_DEBUG - " datapipe = 0x%0x, syncpipe = 0x%0x\n", - subs->datapipe, subs->syncpipe); -#endif - - return 0; -} - -/* - * hw_params callback - * - * allocate a buffer and set the given audio format. - * - * so far we use a physically linear buffer although packetize transfer - * doesn't need a continuous area. - * if sg buffer is supported on the later version of alsa, we'll follow - * that. - */ -static int snd_usb_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - struct audioformat *fmt; - unsigned int channels, rate, format; - int ret, changed; - - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); - if (ret < 0) - return ret; - - format = params_format(hw_params); - rate = params_rate(hw_params); - channels = params_channels(hw_params); - fmt = find_format(subs, format, rate, channels); - if (!fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", - format, rate, channels); - return -EINVAL; - } - - changed = subs->cur_audiofmt != fmt || - subs->period_bytes != params_period_bytes(hw_params) || - subs->cur_rate != rate; - if ((ret = set_format(subs, fmt)) < 0) - return ret; - - if (subs->cur_rate != rate) { - struct usb_host_interface *alts; - struct usb_interface *iface; - iface = usb_ifnum_to_if(subs->dev, fmt->iface); - alts = &iface->altsetting[fmt->altset_idx]; - ret = init_usb_sample_rate(subs->dev, subs->interface, alts, fmt, rate); - if (ret < 0) - return ret; - subs->cur_rate = rate; - } - - if (changed) { - /* format changed */ - release_substream_urbs(subs, 0); - /* influenced: period_bytes, channels, rate, format, */ - ret = init_substream_urbs(subs, params_period_bytes(hw_params), - params_rate(hw_params), - snd_pcm_format_physical_width(params_format(hw_params)) * params_channels(hw_params)); - } - - return ret; -} - -/* - * hw_free callback - * - * reset the audio format and release the buffer - */ -static int snd_usb_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_usb_substream *subs = substream->runtime->private_data; - - subs->cur_audiofmt = NULL; - subs->cur_rate = 0; - subs->period_bytes = 0; - if (!subs->stream->chip->shutdown) - release_substream_urbs(subs, 0); - return snd_pcm_lib_free_vmalloc_buffer(substream); -} - -/* - * prepare callback - * - * only a few subtle things... - */ -static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_usb_substream *subs = runtime->private_data; - - if (! subs->cur_audiofmt) { - snd_printk(KERN_ERR "usbaudio: no format is specified!\n"); - return -ENXIO; - } - - /* some unit conversions in runtime */ - subs->maxframesize = bytes_to_frames(runtime, subs->maxpacksize); - subs->curframesize = bytes_to_frames(runtime, subs->curpacksize); - - /* reset the pointer */ - subs->hwptr_done = 0; - subs->transfer_done = 0; - subs->phase = 0; - runtime->delay = 0; - - /* clear urbs (to be sure) */ - deactivate_urbs(subs, 0, 1); - wait_clear_urbs(subs); - - /* for playback, submit the URBs now; otherwise, the first hwptr_done - * updates for all URBs would happen at the same time when starting */ - if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { - subs->ops.prepare = prepare_nodata_playback_urb; - return start_urbs(subs, runtime); - } else - return 0; -} - -static struct snd_pcm_hardware snd_usb_hardware = -{ - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE, - .buffer_bytes_max = 1024 * 1024, - .period_bytes_min = 64, - .period_bytes_max = 512 * 1024, - .periods_min = 2, - .periods_max = 1024, -}; - -/* - * h/w constraints - */ - -#ifdef HW_CONST_DEBUG -#define hwc_debug(fmt, args...) printk(KERN_DEBUG fmt, ##args) -#else -#define hwc_debug(fmt, args...) /**/ -#endif - -static int hw_check_valid_format(struct snd_usb_substream *subs, - struct snd_pcm_hw_params *params, - struct audioformat *fp) -{ - struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); - unsigned int ptime; - - /* check the format */ - if (!snd_mask_test(fmts, fp->format)) { - hwc_debug(" > check: no supported format %d\n", fp->format); - return 0; - } - /* check the channels */ - if (fp->channels < ct->min || fp->channels > ct->max) { - hwc_debug(" > check: no valid channels %d (%d/%d)\n", fp->channels, ct->min, ct->max); - return 0; - } - /* check the rate is within the range */ - if (fp->rate_min > it->max || (fp->rate_min == it->max && it->openmax)) { - hwc_debug(" > check: rate_min %d > max %d\n", fp->rate_min, it->max); - return 0; - } - if (fp->rate_max < it->min || (fp->rate_max == it->min && it->openmin)) { - hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min); - return 0; - } - /* check whether the period time is >= the data packet interval */ - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { - ptime = 125 * (1 << fp->datainterval); - if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { - hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); - return 0; - } - } - return 1; -} - -static int hw_rule_rate(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - struct list_head *p; - struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - unsigned int rmin, rmax; - int changed; - - hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); - changed = 0; - rmin = rmax = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(subs, params, fp)) - continue; - if (changed++) { - if (rmin > fp->rate_min) - rmin = fp->rate_min; - if (rmax < fp->rate_max) - rmax = fp->rate_max; - } else { - rmin = fp->rate_min; - rmax = fp->rate_max; - } - } - - if (!changed) { - hwc_debug(" --> get empty\n"); - it->empty = 1; - return -EINVAL; - } - - changed = 0; - if (it->min < rmin) { - it->min = rmin; - it->openmin = 0; - changed = 1; - } - if (it->max > rmax) { - it->max = rmax; - it->openmax = 0; - changed = 1; - } - if (snd_interval_checkempty(it)) { - it->empty = 1; - return -EINVAL; - } - hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed); - return changed; -} - - -static int hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - struct list_head *p; - struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - unsigned int rmin, rmax; - int changed; - - hwc_debug("hw_rule_channels: (%d,%d)\n", it->min, it->max); - changed = 0; - rmin = rmax = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(subs, params, fp)) - continue; - if (changed++) { - if (rmin > fp->channels) - rmin = fp->channels; - if (rmax < fp->channels) - rmax = fp->channels; - } else { - rmin = fp->channels; - rmax = fp->channels; - } - } - - if (!changed) { - hwc_debug(" --> get empty\n"); - it->empty = 1; - return -EINVAL; - } - - changed = 0; - if (it->min < rmin) { - it->min = rmin; - it->openmin = 0; - changed = 1; - } - if (it->max > rmax) { - it->max = rmax; - it->openmax = 0; - changed = 1; - } - if (snd_interval_checkempty(it)) { - it->empty = 1; - return -EINVAL; - } - hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed); - return changed; -} - -static int hw_rule_format(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - struct list_head *p; - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - u64 fbits; - u32 oldbits[2]; - int changed; - - hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]); - fbits = 0; - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(subs, params, fp)) - continue; - fbits |= (1ULL << fp->format); - } - - oldbits[0] = fmt->bits[0]; - oldbits[1] = fmt->bits[1]; - fmt->bits[0] &= (u32)fbits; - fmt->bits[1] &= (u32)(fbits >> 32); - if (!fmt->bits[0] && !fmt->bits[1]) { - hwc_debug(" --> get empty\n"); - return -EINVAL; - } - changed = (oldbits[0] != fmt->bits[0] || oldbits[1] != fmt->bits[1]); - hwc_debug(" --> %x:%x (changed = %d)\n", fmt->bits[0], fmt->bits[1], changed); - return changed; -} - -static int hw_rule_period_time(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct snd_usb_substream *subs = rule->private; - struct audioformat *fp; - struct snd_interval *it; - unsigned char min_datainterval; - unsigned int pmin; - int changed; - - it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); - hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max); - min_datainterval = 0xff; - list_for_each_entry(fp, &subs->fmt_list, list) { - if (!hw_check_valid_format(subs, params, fp)) - continue; - min_datainterval = min(min_datainterval, fp->datainterval); - } - if (min_datainterval == 0xff) { - hwc_debug(" --> get emtpy\n"); - it->empty = 1; - return -EINVAL; - } - pmin = 125 * (1 << min_datainterval); - changed = 0; - if (it->min < pmin) { - it->min = pmin; - it->openmin = 0; - changed = 1; - } - if (snd_interval_checkempty(it)) { - it->empty = 1; - return -EINVAL; - } - hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed); - return changed; -} - -/* - * If the device supports unusual bit rates, does the request meet these? - */ -static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) -{ - struct audioformat *fp; - int count = 0, needs_knot = 0; - int err; - - list_for_each_entry(fp, &subs->fmt_list, list) { - if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) - return 0; - count += fp->nr_rates; - if (fp->rates & SNDRV_PCM_RATE_KNOT) - needs_knot = 1; - } - if (!needs_knot) - return 0; - - subs->rate_list.count = count; - subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); - subs->rate_list.mask = 0; - count = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - int i; - for (i = 0; i < fp->nr_rates; i++) - subs->rate_list.list[count++] = fp->rate_table[i]; - } - err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &subs->rate_list); - if (err < 0) - return err; - - return 0; -} - - -/* - * set up the runtime hardware information. - */ - -static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) -{ - struct list_head *p; - unsigned int pt, ptmin; - int param_period_time_if_needed; - int err; - - runtime->hw.formats = subs->formats; - - runtime->hw.rate_min = 0x7fffffff; - runtime->hw.rate_max = 0; - runtime->hw.channels_min = 256; - runtime->hw.channels_max = 0; - runtime->hw.rates = 0; - ptmin = UINT_MAX; - /* check min/max rates and channels */ - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - runtime->hw.rates |= fp->rates; - if (runtime->hw.rate_min > fp->rate_min) - runtime->hw.rate_min = fp->rate_min; - if (runtime->hw.rate_max < fp->rate_max) - runtime->hw.rate_max = fp->rate_max; - if (runtime->hw.channels_min > fp->channels) - runtime->hw.channels_min = fp->channels; - if (runtime->hw.channels_max < fp->channels) - runtime->hw.channels_max = fp->channels; - if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { - /* FIXME: there might be more than one audio formats... */ - runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = - fp->frame_size; - } - pt = 125 * (1 << fp->datainterval); - ptmin = min(ptmin, pt); - } - - param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; - if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) - /* full speed devices have fixed data packet interval */ - ptmin = 1000; - if (ptmin == 1000) - /* if period time doesn't go below 1 ms, no rules needed */ - param_period_time_if_needed = -1; - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - ptmin, UINT_MAX); - - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - param_period_time_if_needed, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - param_period_time_if_needed, - -1)) < 0) - return err; - if (param_period_time_if_needed >= 0) { - err = snd_pcm_hw_rule_add(runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_TIME, - hw_rule_period_time, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - SNDRV_PCM_HW_PARAM_RATE, - -1); - if (err < 0) - return err; - } - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) - return err; - return 0; -} - -static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) -{ - struct snd_usb_stream *as = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_usb_substream *subs = &as->substream[direction]; - - subs->interface = -1; - subs->format = 0; - runtime->hw = snd_usb_hardware; - runtime->private_data = subs; - subs->pcm_substream = substream; - return setup_hw_info(runtime, subs); -} - -static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) -{ - struct snd_usb_stream *as = snd_pcm_substream_chip(substream); - struct snd_usb_substream *subs = &as->substream[direction]; - - if (!as->chip->shutdown && subs->interface >= 0) { - usb_set_interface(subs->dev, subs->interface, 0); - subs->interface = -1; - } - subs->pcm_substream = NULL; - return 0; -} - -static int snd_usb_playback_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_playback_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_capture_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static int snd_usb_capture_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static struct snd_pcm_ops snd_usb_playback_ops = { - .open = snd_usb_playback_open, - .close = snd_usb_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, - .prepare = snd_usb_pcm_prepare, - .trigger = snd_usb_pcm_playback_trigger, - .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - -static struct snd_pcm_ops snd_usb_capture_ops = { - .open = snd_usb_capture_open, - .close = snd_usb_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, - .prepare = snd_usb_pcm_prepare, - .trigger = snd_usb_pcm_capture_trigger, - .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - - - -/* - * helper functions - */ - -/* - * combine bytes and get an integer value - */ -unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size) -{ - switch (size) { - case 1: return *bytes; - case 2: return combine_word(bytes); - case 3: return combine_triple(bytes); - case 4: return combine_quad(bytes); - default: return 0; - } -} - -/* - * parse descriptor buffer and return the pointer starting the given - * descriptor type. - */ -void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype) -{ - u8 *p, *end, *next; - - p = descstart; - end = p + desclen; - for (; p < end;) { - if (p[0] < 2) - return NULL; - next = p + p[0]; - if (next > end) - return NULL; - if (p[1] == dtype && (!after || (void *)p > after)) { - return p; - } - p = next; - } - return NULL; -} - -/* - * find a class-specified interface descriptor with the given subtype. - */ -void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype) -{ - unsigned char *p = after; - - while ((p = snd_usb_find_desc(buffer, buflen, p, - USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 3 && p[2] == dsubtype) - return p; - } - return NULL; -} - -/* - * Wrapper for usb_control_msg(). - * Allocates a temp buffer to prevent dmaing from/to the stack. - */ -int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, - __u8 requesttype, __u16 value, __u16 index, void *data, - __u16 size, int timeout) -{ - int err; - void *buf = NULL; - - if (size > 0) { - buf = kmemdup(data, size, GFP_KERNEL); - if (!buf) - return -ENOMEM; - } - err = usb_control_msg(dev, pipe, request, requesttype, - value, index, buf, size, timeout); - if (size > 0) { - memcpy(data, buf, size); - kfree(buf); - } - return err; -} - - -/* - * entry point for linux usb interface - */ - -static int usb_audio_probe(struct usb_interface *intf, - const struct usb_device_id *id); -static void usb_audio_disconnect(struct usb_interface *intf); - -#ifdef CONFIG_PM -static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message); -static int usb_audio_resume(struct usb_interface *intf); -#else -#define usb_audio_suspend NULL -#define usb_audio_resume NULL -#endif - -static struct usb_device_id usb_audio_ids [] = { -#include "usbquirks.h" - { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), - .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, - { } /* Terminating entry */ -}; - -MODULE_DEVICE_TABLE (usb, usb_audio_ids); - -static struct usb_driver usb_audio_driver = { - .name = "snd-usb-audio", - .probe = usb_audio_probe, - .disconnect = usb_audio_disconnect, - .suspend = usb_audio_suspend, - .resume = usb_audio_resume, - .id_table = usb_audio_ids, -}; - - -#if defined(CONFIG_PROC_FS) && defined(CONFIG_SND_VERBOSE_PROCFS) - -/* - * proc interface for list the supported pcm formats - */ -static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) -{ - struct list_head *p; - static char *sync_types[4] = { - "NONE", "ASYNC", "ADAPTIVE", "SYNC" - }; - - list_for_each(p, &subs->fmt_list) { - struct audioformat *fp; - fp = list_entry(p, struct audioformat, list); - snd_iprintf(buffer, " Interface %d\n", fp->iface); - snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %s\n", - snd_pcm_format_name(fp->format)); - snd_iprintf(buffer, " Channels: %d\n", fp->channels); - snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", - fp->endpoint & USB_ENDPOINT_NUMBER_MASK, - fp->endpoint & USB_DIR_IN ? "IN" : "OUT", - sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); - if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { - snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", - fp->rate_min, fp->rate_max); - } else { - unsigned int i; - snd_iprintf(buffer, " Rates: "); - for (i = 0; i < fp->nr_rates; i++) { - if (i > 0) - snd_iprintf(buffer, ", "); - snd_iprintf(buffer, "%d", fp->rate_table[i]); - } - snd_iprintf(buffer, "\n"); - } - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) - snd_iprintf(buffer, " Data packet interval: %d us\n", - 125 * (1 << fp->datainterval)); - // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); - // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); - } -} - -static void proc_dump_substream_status(struct snd_usb_substream *subs, struct snd_info_buffer *buffer) -{ - if (subs->running) { - unsigned int i; - snd_iprintf(buffer, " Status: Running\n"); - snd_iprintf(buffer, " Interface = %d\n", subs->interface); - snd_iprintf(buffer, " Altset = %d\n", subs->format); - snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); - for (i = 0; i < subs->nurbs; i++) - snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); - snd_iprintf(buffer, "]\n"); - snd_iprintf(buffer, " Packet Size = %d\n", subs->curpacksize); - snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", - snd_usb_get_speed(subs->dev) == USB_SPEED_FULL - ? get_full_speed_hz(subs->freqm) - : get_high_speed_hz(subs->freqm), - subs->freqm >> 16, subs->freqm & 0xffff); - } else { - snd_iprintf(buffer, " Status: Stop\n"); - } -} - -static void proc_pcm_format_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) -{ - struct snd_usb_stream *stream = entry->private_data; - - snd_iprintf(buffer, "%s : %s\n", stream->chip->card->longname, stream->pcm->name); - - if (stream->substream[SNDRV_PCM_STREAM_PLAYBACK].num_formats) { - snd_iprintf(buffer, "\nPlayback:\n"); - proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer); - proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_PLAYBACK], buffer); - } - if (stream->substream[SNDRV_PCM_STREAM_CAPTURE].num_formats) { - snd_iprintf(buffer, "\nCapture:\n"); - proc_dump_substream_status(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer); - proc_dump_substream_formats(&stream->substream[SNDRV_PCM_STREAM_CAPTURE], buffer); - } -} - -static void proc_pcm_format_add(struct snd_usb_stream *stream) -{ - struct snd_info_entry *entry; - char name[32]; - struct snd_card *card = stream->chip->card; - - sprintf(name, "stream%d", stream->pcm_index); - if (!snd_card_proc_new(card, name, &entry)) - snd_info_set_text_ops(entry, stream, proc_pcm_format_read); -} - -#else - -static inline void proc_pcm_format_add(struct snd_usb_stream *stream) -{ -} - -#endif - -/* - * initialize the substream instance. - */ - -static void init_substream(struct snd_usb_stream *as, int stream, struct audioformat *fp) -{ - struct snd_usb_substream *subs = &as->substream[stream]; - - INIT_LIST_HEAD(&subs->fmt_list); - spin_lock_init(&subs->lock); - - subs->stream = as; - subs->direction = stream; - subs->dev = as->chip->dev; - subs->txfr_quirk = as->chip->txfr_quirk; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { - subs->ops = audio_urb_ops[stream]; - } else { - subs->ops = audio_urb_ops_high_speed[stream]; - switch (as->chip->usb_id) { - case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ - case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ - case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ - subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; - break; - } - } - snd_pcm_set_ops(as->pcm, stream, - stream == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_usb_playback_ops : &snd_usb_capture_ops); - - list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= 1ULL << fp->format; - subs->endpoint = fp->endpoint; - subs->num_formats++; - subs->fmt_type = fp->fmt_type; -} - - -/* - * free a substream - */ -static void free_substream(struct snd_usb_substream *subs) -{ - struct list_head *p, *n; - - if (!subs->num_formats) - return; /* not initialized */ - list_for_each_safe(p, n, &subs->fmt_list) { - struct audioformat *fp = list_entry(p, struct audioformat, list); - kfree(fp->rate_table); - kfree(fp); - } - kfree(subs->rate_list.list); -} - - -/* - * free a usb stream instance - */ -static void snd_usb_audio_stream_free(struct snd_usb_stream *stream) -{ - free_substream(&stream->substream[0]); - free_substream(&stream->substream[1]); - list_del(&stream->list); - kfree(stream); -} - -static void snd_usb_audio_pcm_free(struct snd_pcm *pcm) -{ - struct snd_usb_stream *stream = pcm->private_data; - if (stream) { - stream->pcm = NULL; - snd_usb_audio_stream_free(stream); - } -} - - -/* - * add this endpoint to the chip instance. - * if a stream with the same endpoint already exists, append to it. - * if not, create a new pcm stream. - */ -static int add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct audioformat *fp) -{ - struct list_head *p; - struct snd_usb_stream *as; - struct snd_usb_substream *subs; - struct snd_pcm *pcm; - int err; - - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (!subs->endpoint) - continue; - if (subs->endpoint == fp->endpoint) { - list_add_tail(&fp->list, &subs->fmt_list); - subs->num_formats++; - subs->formats |= 1ULL << fp->format; - return 0; - } - } - /* look for an empty stream */ - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - if (as->fmt_type != fp->fmt_type) - continue; - subs = &as->substream[stream]; - if (subs->endpoint) - continue; - err = snd_pcm_new_stream(as->pcm, stream, 1); - if (err < 0) - return err; - init_substream(as, stream, fp); - return 0; - } - - /* create a new pcm */ - as = kzalloc(sizeof(*as), GFP_KERNEL); - if (!as) - return -ENOMEM; - as->pcm_index = chip->pcm_devs; - as->chip = chip; - as->fmt_type = fp->fmt_type; - err = snd_pcm_new(chip->card, "USB Audio", chip->pcm_devs, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 1 : 0, - stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1, - &pcm); - if (err < 0) { - kfree(as); - return err; - } - as->pcm = pcm; - pcm->private_data = as; - pcm->private_free = snd_usb_audio_pcm_free; - pcm->info_flags = 0; - if (chip->pcm_devs > 0) - sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); - else - strcpy(pcm->name, "USB Audio"); - - init_substream(as, stream, fp); - - list_add(&as->list, &chip->pcm_list); - chip->pcm_devs++; - - proc_pcm_format_add(as); - - return 0; -} - - -/* - * check if the device uses big-endian samples - */ -static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) -{ - switch (chip->usb_id) { - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ - if (fp->endpoint & USB_DIR_IN) - return 1; - break; - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - if (device_setup[chip->index] == 0x00 || - fp->altsetting==1 || fp->altsetting==2 || fp->altsetting==3) - return 1; - } - return 0; -} - -/* - * parse the audio format type I descriptor - * and returns the corresponding pcm format - * - * @dev: usb device - * @fp: audioformat record - * @format: the format tag (wFormatTag) - * @fmt: the format type descriptor - */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, - int protocol) -{ - int pcm_format, i; - int sample_width, sample_bytes; - - switch (protocol) { - case UAC_VERSION_1: { - struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - sample_width = fmt->bBitResolution; - sample_bytes = fmt->bSubframeSize; - break; - } - - case UAC_VERSION_2: { - struct uac_format_type_i_ext_descriptor *fmt = _fmt; - sample_width = fmt->bBitResolution; - sample_bytes = fmt->bSubslotSize; - - /* - * FIXME - * USB audio class v2 devices specify a bitmap of possible - * audio formats rather than one fix value. For now, we just - * pick one of them and report that as the only possible - * value for this setting. - * The bit allocation map is in fact compatible to the - * wFormatTag of the v1 AS streaming descriptors, which is why - * we can simply map the matrix. - */ - - for (i = 0; i < 5; i++) - if (format & (1UL << i)) { - format = i + 1; - break; - } - - break; - } - - default: - return -EINVAL; - } - - /* FIXME: correct endianess and sign? */ - pcm_format = -1; - - switch (format) { - case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ - snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", - chip->dev->devnum, fp->iface, fp->altsetting); - /* fall-through */ - case UAC_FORMAT_TYPE_I_PCM: - if (sample_width > sample_bytes * 8) { - snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, - sample_width, sample_bytes); - } - /* check the format byte size */ - switch (sample_bytes) { - case 1: - pcm_format = SNDRV_PCM_FORMAT_S8; - break; - case 2: - if (is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */ - else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; - break; - case 3: - if (is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */ - else - pcm_format = SNDRV_PCM_FORMAT_S24_3LE; - break; - case 4: - pcm_format = SNDRV_PCM_FORMAT_S32_LE; - break; - default: - snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", - chip->dev->devnum, fp->iface, fp->altsetting, - sample_width, sample_bytes); - break; - } - break; - case UAC_FORMAT_TYPE_I_PCM8: - pcm_format = SNDRV_PCM_FORMAT_U8; - - /* Dallas DS4201 workaround: it advertises U8 format, but really - supports S8. */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - pcm_format = SNDRV_PCM_FORMAT_S8; - break; - case UAC_FORMAT_TYPE_I_IEEE_FLOAT: - pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; - break; - case UAC_FORMAT_TYPE_I_ALAW: - pcm_format = SNDRV_PCM_FORMAT_A_LAW; - break; - case UAC_FORMAT_TYPE_I_MULAW: - pcm_format = SNDRV_PCM_FORMAT_MU_LAW; - break; - default: - snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, format); - break; - } - return pcm_format; -} - - -/* - * parse the format descriptor and stores the possible sample rates - * on the audioformat table (audio class v1). - * - * @dev: usb device - * @fp: audioformat record - * @fmt: the format descriptor - * @offset: the start offset of descriptor pointing the rate type - * (7 for type I and II, 8 for type II) - */ -static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned char *fmt, int offset) -{ - int nr_rates = fmt[offset]; - - if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", - chip->dev->devnum, fp->iface, fp->altsetting); - return -1; - } - - if (nr_rates) { - /* - * build the rate table and bitmap flags - */ - int r, idx; - - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); - if (fp->rate_table == NULL) { - snd_printk(KERN_ERR "cannot malloc\n"); - return -1; - } - - fp->nr_rates = 0; - fp->rate_min = fp->rate_max = 0; - for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { - unsigned int rate = combine_triple(&fmt[idx]); - if (!rate) - continue; - /* C-Media CM6501 mislabels its 96 kHz altsetting */ - if (rate == 48000 && nr_rates == 1 && - (chip->usb_id == USB_ID(0x0d8c, 0x0201) || - chip->usb_id == USB_ID(0x0d8c, 0x0102)) && - fp->altsetting == 5 && fp->maxpacksize == 392) - rate = 96000; - /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ - if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) - rate = 8000; - fp->rate_table[fp->nr_rates] = rate; - if (!fp->rate_min || rate < fp->rate_min) - fp->rate_min = rate; - if (!fp->rate_max || rate > fp->rate_max) - fp->rate_max = rate; - fp->rates |= snd_pcm_rate_to_rate_bit(rate); - fp->nr_rates++; - } - if (!fp->nr_rates) { - hwc_debug("All rates were zero. Skipping format!\n"); - return -1; - } - } else { - /* continuous rates */ - fp->rates = SNDRV_PCM_RATE_CONTINUOUS; - fp->rate_min = combine_triple(&fmt[offset + 1]); - fp->rate_max = combine_triple(&fmt[offset + 4]); - } - return 0; -} - -/* - * parse the format descriptor and stores the possible sample rates - * on the audioformat table (audio class v2). - */ -static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, - struct audioformat *fp, - struct usb_host_interface *iface) -{ - struct usb_device *dev = chip->dev; - unsigned char tmp[2], *data; - int i, nr_rates, data_size, ret = 0; - - /* get the number of sample rates first by only fetching 2 bytes */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); - - if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); - goto err; - } - - nr_rates = (tmp[1] << 8) | tmp[0]; - data_size = 2 + 12 * nr_rates; - data = kzalloc(data_size, GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto err; - } - - /* now get the full information */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - 0x0100, chip->clock_id << 8, data, data_size, 1000); - - if (ret < 0) { - snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); - ret = -EINVAL; - goto err_free; - } - - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); - if (!fp->rate_table) { - ret = -ENOMEM; - goto err_free; - } - - fp->nr_rates = 0; - fp->rate_min = fp->rate_max = 0; - - for (i = 0; i < nr_rates; i++) { - int rate = combine_quad(&data[2 + 12 * i]); - - fp->rate_table[fp->nr_rates] = rate; - if (!fp->rate_min || rate < fp->rate_min) - fp->rate_min = rate; - if (!fp->rate_max || rate > fp->rate_max) - fp->rate_max = rate; - fp->rates |= snd_pcm_rate_to_rate_bit(rate); - fp->nr_rates++; - } - -err_free: - kfree(data); -err: - return ret; -} - -/* - * parse the format type I and III descriptors - */ -static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, - struct usb_host_interface *iface) -{ - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - int protocol = altsd->bInterfaceProtocol; - int pcm_format, ret; - - if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { - /* FIXME: the format type is really IECxxx - * but we give normal PCM format to get the existing - * apps working... - */ - switch (chip->usb_id) { - - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - if (device_setup[chip->index] == 0x00 && - fp->altsetting == 6) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; - else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; - break; - default: - pcm_format = SNDRV_PCM_FORMAT_S16_LE; - } - } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); - if (pcm_format < 0) - return -1; - } - - fp->format = pcm_format; - - /* gather possible sample rates */ - /* audio class v1 reports possible sample rates as part of the - * proprietary class specific descriptor. - * audio class v2 uses class specific EP0 range requests for that. - */ - switch (protocol) { - case UAC_VERSION_1: - fp->channels = fmt->bNrChannels; - ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); - break; - case UAC_VERSION_2: - /* fp->channels is already set in this case */ - ret = parse_audio_format_rates_v2(chip, fp, iface); - break; - } - - if (fp->channels < 1) { - snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); - return -1; - } - - return ret; -} - -/* - * parse the format type II descriptor - */ -static int parse_audio_format_ii(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, - struct usb_host_interface *iface) -{ - int brate, framesize, ret; - struct usb_interface_descriptor *altsd = get_iface_desc(iface); - int protocol = altsd->bInterfaceProtocol; - - switch (format) { - case UAC_FORMAT_TYPE_II_AC3: - /* FIXME: there is no AC3 format defined yet */ - // fp->format = SNDRV_PCM_FORMAT_AC3; - fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ - break; - case UAC_FORMAT_TYPE_II_MPEG: - fp->format = SNDRV_PCM_FORMAT_MPEG; - break; - default: - snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", - chip->dev->devnum, fp->iface, fp->altsetting, format); - fp->format = SNDRV_PCM_FORMAT_MPEG; - break; - } - - fp->channels = 1; - - switch (protocol) { - case UAC_VERSION_1: { - struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; - brate = le16_to_cpu(fmt->wMaxBitRate); - framesize = le16_to_cpu(fmt->wSamplesPerFrame); - snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); - fp->frame_size = framesize; - ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */ - break; - } - case UAC_VERSION_2: { - struct uac_format_type_ii_ext_descriptor *fmt = _fmt; - brate = le16_to_cpu(fmt->wMaxBitRate); - framesize = le16_to_cpu(fmt->wSamplesPerFrame); - snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); - fp->frame_size = framesize; - ret = parse_audio_format_rates_v2(chip, fp, iface); - break; - } - } - - return ret; -} - -static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface) -{ - int err; - - switch (fmt[3]) { - case UAC_FORMAT_TYPE_I: - case UAC_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt, iface); - break; - case UAC_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt, iface); - break; - default: - snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); - return -1; - } - fp->fmt_type = fmt[3]; - if (err < 0) - return err; -#if 1 - /* FIXME: temporary hack for extigy/audigy 2 nx/zs */ - /* extigy apparently supports sample rates other than 48k - * but not in ordinary way. so we enable only 48k atm. - */ - if (chip->usb_id == USB_ID(0x041e, 0x3000) || - chip->usb_id == USB_ID(0x041e, 0x3020) || - chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == UAC_FORMAT_TYPE_I && - fp->rates != SNDRV_PCM_RATE_48000 && - fp->rates != SNDRV_PCM_RATE_96000) - return -1; - } -#endif - return 0; -} - -static unsigned char parse_datainterval(struct snd_usb_audio *chip, - struct usb_host_interface *alts) -{ - if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - return get_endpoint(alts, 0)->bInterval - 1; - else - return 0; -} - -static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, - int iface, int altno); -static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) -{ - struct usb_device *dev; - struct usb_interface *iface; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - int i, altno, err, stream; - int format = 0, num_channels = 0; - struct audioformat *fp = NULL; - unsigned char *fmt, *csep; - int num, protocol; - - dev = chip->dev; - - /* parse the interface's altsettings */ - iface = usb_ifnum_to_if(dev, iface_no); - - num = iface->num_altsetting; - - /* - * Dallas DS4201 workaround: It presents 5 altsettings, but the last - * one misses syncpipe, and does not produce any sound. - */ - if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - num = 4; - - for (i = 0; i < num; i++) { - alts = &iface->altsetting[i]; - altsd = get_iface_desc(alts); - protocol = altsd->bInterfaceProtocol; - /* skip invalid one */ - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && - altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || - altsd->bNumEndpoints < 1 || - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) - continue; - /* must be isochronous */ - if ((get_endpoint(alts, 0)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != - USB_ENDPOINT_XFER_ISOC) - continue; - /* check direction */ - stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? - SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - altno = altsd->bAlternateSetting; - - /* audiophile usb: skip altsets incompatible with device_setup - */ - if (chip->usb_id == USB_ID(0x0763, 0x2003) && - audiophile_skip_setting_quirk(chip, iface_no, altno)) - continue; - - /* get audio formats */ - switch (protocol) { - case UAC_VERSION_1: { - struct uac_as_header_descriptor_v1 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; - } - - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ - break; - } - - case UAC_VERSION_2: { - struct uac_as_header_descriptor_v2 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); - - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; - } - - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; - } - - num_channels = as->bNrChannels; - format = le32_to_cpu(as->bmFormats); - - break; - } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; - } - - /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); - if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } - if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || - ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", - dev->devnum, iface_no, altno); - continue; - } - - /* - * Blue Microphones workaround: The last altsetting is identical - * with the previous one, except for a larger packet size, but - * is actually a mislabeled two-channel setting; ignore it. - */ - if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && - fp && fp->altsetting == 1 && fp->channels == 1 && - fp->format == SNDRV_PCM_FORMAT_S16_LE && - protocol == UAC_VERSION_1 && - le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == - fp->maxpacksize * 2) - continue; - - csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT); - /* Creamware Noah has this descriptor after the 2nd endpoint */ - if (!csep && altsd->bNumEndpoints >= 2) - csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { - snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" - " class specific endpoint descriptor\n", - dev->devnum, iface_no, altno); - csep = NULL; - } - - fp = kzalloc(sizeof(*fp), GFP_KERNEL); - if (! fp) { - snd_printk(KERN_ERR "cannot malloc\n"); - return -ENOMEM; - } - - fp->iface = iface_no; - fp->altsetting = altno; - fp->altset_idx = i; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - /* num_channels is only set for v2 interfaces */ - fp->channels = num_channels; - if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) - fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) - * (fp->maxpacksize & 0x7ff); - fp->attributes = csep ? csep[3] : 0; - - /* some quirks for attributes here */ - - switch (chip->usb_id) { - case USB_ID(0x0a92, 0x0053): /* AudioTrak Optoplay */ - /* Optoplay sets the sample rate attribute although - * it seems not supporting it in fact. - */ - fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ - case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ - /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; - break; - case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ - case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is - an older model 77d:223) */ - /* - * plantronics headset and Griffin iMic have set adaptive-in - * although it's really not... - */ - fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; - else - fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; - break; - } - - /* ok, let's parse further... */ - if (parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { - kfree(fp->rate_table); - kfree(fp); - continue; - } - - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp->rate_table); - kfree(fp); - return err; - } - /* try to set the interface... */ - usb_set_interface(chip->dev, iface_no, altno); - init_usb_pitch(chip->dev, iface_no, alts, fp); - init_usb_sample_rate(chip->dev, iface_no, alts, fp, fp->rate_max); - } - return 0; -} - - -/* - * disconnect streams - * called from snd_usb_audio_disconnect() - */ -static void snd_usb_stream_disconnect(struct list_head *head) -{ - int idx; - struct snd_usb_stream *as; - struct snd_usb_substream *subs; - - as = list_entry(head, struct snd_usb_stream, list); - for (idx = 0; idx < 2; idx++) { - subs = &as->substream[idx]; - if (!subs->num_formats) - return; - release_substream_urbs(subs, 1); - subs->interface = -1; - } -} - -static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) -{ - struct usb_device *dev = chip->dev; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct usb_interface *iface = usb_ifnum_to_if(dev, interface); - - if (!iface) { - snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", - dev->devnum, ctrlif, interface); - return -EINVAL; - } - - if (usb_interface_claimed(iface)) { - snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", - dev->devnum, ctrlif, interface); - return -EINVAL; - } - - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || - altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); - if (err < 0) { - snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", - dev->devnum, ctrlif, interface); - return -EINVAL; - } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - - return 0; - } - - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { - snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", - dev->devnum, ctrlif, interface, altsd->bInterfaceClass); - /* skip non-supported classes */ - return -EINVAL; - } - - if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); - return -EINVAL; - } - - if (! parse_audio_endpoints(chip, interface)) { - usb_set_interface(dev, interface, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - return -EINVAL; - } - - return 0; -} - -/* - * parse audio control descriptor and create pcm/midi streams - */ -static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) -{ - struct usb_device *dev = chip->dev; - struct usb_host_interface *host_iface; - struct usb_interface_descriptor *altsd; - void *control_header; - int i, protocol; - - /* find audiocontrol interface */ - host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - control_header = snd_usb_find_csint_desc(host_iface->extra, - host_iface->extralen, - NULL, UAC_HEADER); - altsd = get_iface_desc(host_iface); - protocol = altsd->bInterfaceProtocol; - - if (!control_header) { - snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); - return -EINVAL; - } - - switch (protocol) { - case UAC_VERSION_1: { - struct uac_ac_header_descriptor_v1 *h1 = control_header; - - if (!h1->bInCollection) { - snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); - return -EINVAL; - } - - if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); - return -EINVAL; - } - - for (i = 0; i < h1->bInCollection; i++) - snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); - - break; - } - - case UAC_VERSION_2: { - struct uac_clock_source_descriptor *cs; - struct usb_interface_assoc_descriptor *assoc = - usb_ifnum_to_if(dev, ctrlif)->intf_assoc; - - if (!assoc) { - snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); - return -EINVAL; - } - - /* FIXME: for now, we expect there is at least one clock source - * descriptor and we always take the first one. - * We should properly support devices with multiple clock sources, - * clock selectors and sample rate conversion units. */ - - cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, UAC_CLOCK_SOURCE); - - if (!cs) { - snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); - return -EINVAL; - } - - chip->clock_id = cs->bClockID; - - for (i = 0; i < assoc->bInterfaceCount; i++) { - int intf = assoc->bFirstInterface + i; - - if (intf != ctrlif) - snd_usb_create_stream(chip, ctrlif, intf); - } - - break; - } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; - } - - return 0; -} - -/* - * create a stream for an endpoint/altsetting without proper descriptors - */ -static int create_fixed_stream_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - struct audioformat *fp; - struct usb_host_interface *alts; - int stream, err; - unsigned *rate_table = NULL; - - fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL); - if (! fp) { - snd_printk(KERN_ERR "cannot memdup\n"); - return -ENOMEM; - } - if (fp->nr_rates > 0) { - rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); - if (!rate_table) { - kfree(fp); - return -ENOMEM; - } - memcpy(rate_table, fp->rate_table, sizeof(int) * fp->nr_rates); - fp->rate_table = rate_table; - } - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - kfree(rate_table); - return err; - } - if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber || - fp->altset_idx >= iface->num_altsetting) { - kfree(fp); - kfree(rate_table); - return -EINVAL; - } - alts = &iface->altsetting[fp->altset_idx]; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - usb_set_interface(chip->dev, fp->iface, 0); - init_usb_pitch(chip->dev, fp->iface, alts, fp); - init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); - return 0; -} - -/* - * create a stream for an interface with proper descriptors - */ -static int create_standard_audio_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - int err; - - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - err = parse_audio_endpoints(chip, altsd->bInterfaceNumber); - if (err < 0) { - snd_printk(KERN_ERR "cannot setup if %d: error %d\n", - altsd->bInterfaceNumber, err); - return err; - } - /* reset the current interface */ - usb_set_interface(chip->dev, altsd->bInterfaceNumber, 0); - return 0; -} - -/* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. - * The only way to detect the sample rate is by looking at wMaxPacketSize. - */ -static int create_uaxx_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua_format = { - .format = SNDRV_PCM_FORMAT_S24_3LE, - .channels = 2, - .fmt_type = UAC_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - /* both PCM and MIDI interfaces have 2 or more altsettings */ - if (iface->num_altsetting < 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - - if (altsd->bNumEndpoints == 2) { - static const struct snd_usb_midi_endpoint_info ua700_ep = { - .out_cables = 0x0003, - .in_cables = 0x0003 - }; - static const struct snd_usb_audio_quirk ua700_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &ua700_ep - }; - static const struct snd_usb_midi_endpoint_info uaxx_ep = { - .out_cables = 0x0001, - .in_cables = 0x0001 - }; - static const struct snd_usb_audio_quirk uaxx_quirk = { - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &uaxx_ep - }; - const struct snd_usb_audio_quirk *quirk = - chip->usb_id == USB_ID(0x0582, 0x002b) - ? &ua700_quirk : &uaxx_quirk; - return snd_usbmidi_create(chip->card, iface, - &chip->midi_list, quirk); - } - - if (altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmalloc(sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - memcpy(fp, &ua_format, sizeof(*fp)); - - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = 0; - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - - switch (fp->maxpacksize) { - case 0x120: - fp->rate_max = fp->rate_min = 44100; - break; - case 0x138: - case 0x140: - fp->rate_max = fp->rate_min = 48000; - break; - case 0x258: - case 0x260: - fp->rate_max = fp->rate_min = 96000; - break; - default: - snd_printk(KERN_ERR "unknown sample rate\n"); - kfree(fp); - return -ENXIO; - } - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - -static int snd_usb_create_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk); - -/* - * handle the quirks for the contained interfaces - */ -static int create_composite_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - int probed_ifnum = get_iface_desc(iface->altsetting)->bInterfaceNumber; - int err; - - for (quirk = quirk->data; quirk->ifnum >= 0; ++quirk) { - iface = usb_ifnum_to_if(chip->dev, quirk->ifnum); - if (!iface) - continue; - if (quirk->ifnum != probed_ifnum && - usb_interface_claimed(iface)) - continue; - err = snd_usb_create_quirk(chip, iface, quirk); - if (err < 0) - return err; - if (quirk->ifnum != probed_ifnum) - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - } - return 0; -} - -static int ignore_interface_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - return 0; -} - -/* - * Allow alignment on audio sub-slot (channel samples) rather than - * on audio slots (audio frames) - */ -static int create_align_transfer_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - chip->txfr_quirk = 1; - return 1; /* Continue with creating streams and mixer */ -} - - -/* - * boot quirks - */ - -#define EXTIGY_FIRMWARE_SIZE_OLD 794 -#define EXTIGY_FIRMWARE_SIZE_NEW 483 - -static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interface *intf) -{ - struct usb_host_config *config = dev->actconfig; - int err; - - if (le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_OLD || - le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) { - snd_printdd("sending Extigy boot sequence...\n"); - /* Send message to force it to reconnect with full interface. */ - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), - 0x10, 0x43, 0x0001, 0x000a, NULL, 0, 1000); - if (err < 0) snd_printdd("error sending boot message: %d\n", err); - err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, - &dev->descriptor, sizeof(dev->descriptor)); - config = dev->actconfig; - if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); - err = usb_reset_configuration(dev); - if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); - snd_printdd("extigy_boot: new boot length = %d\n", - le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - return -ENODEV; /* quit this anyway */ - } - return 0; -} - -static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) -{ - u8 buf = 1; - - snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x2a, - USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 0, 0, &buf, 1, 1000); - if (buf == 0) { - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x29, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, - 1, 2000, NULL, 0, 1000); - return -ENODEV; - } - return 0; -} - -/* - * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely - * documented in the device's data sheet. - */ -static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value) -{ - u8 buf[4]; - buf[0] = 0x20; - buf[1] = value & 0xff; - buf[2] = (value >> 8) & 0xff; - buf[3] = reg; - return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION, - USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT, - 0, 0, &buf, 4, 1000); -} - -static int snd_usb_cm106_boot_quirk(struct usb_device *dev) -{ - /* - * Enable line-out driver mode, set headphone source to front - * channels, enable stereo mic. - */ - return snd_usb_cm106_write_int_reg(dev, 2, 0x8004); -} - -/* - * C-Media CM6206 is based on CM106 with two additional - * registers that are not documented in the data sheet. - * Values here are chosen based on sniffing USB traffic - * under Windows. - */ -static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) -{ - int err, reg; - int val[] = {0x200c, 0x3000, 0xf800, 0x143f, 0x0000, 0x3000}; - - for (reg = 0; reg < ARRAY_SIZE(val); reg++) { - err = snd_usb_cm106_write_int_reg(dev, reg, val[reg]); - if (err < 0) - return err; - } - - return err; -} - -/* - * This call will put the synth in "USB send" mode, i.e it will send MIDI - * messages through USB (this is disabled at startup). The synth will - * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB - * sign on its LCD. Values here are chosen based on sniffing USB traffic - * under Windows. - */ -static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) -{ - int err, actual_length; - - /* "midi send" enable */ - static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; - - void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); - if (!buf) - return -ENOMEM; - err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, - ARRAY_SIZE(seq), &actual_length, 1000); - kfree(buf); - if (err < 0) - return err; - - return 0; -} - -/* - * Setup quirks - */ -#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ -#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */ -#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */ -#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */ -#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */ -#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */ -#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */ -#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */ -#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */ -#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */ - -static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, - int iface, int altno) -{ - /* Reset ALL ifaces to 0 altsetting. - * Call it for every possible altsetting of every interface. - */ - usb_set_interface(chip->dev, iface, 0); - - if (device_setup[chip->index] & AUDIOPHILE_SET) { - if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS) - && altno != 6) - return 1; /* skip this altsetting */ - if ((device_setup[chip->index] & AUDIOPHILE_SET_96K) - && altno != 1) - return 1; /* skip this altsetting */ - if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == - AUDIOPHILE_SET_24B_48K_DI && altno != 2) - return 1; /* skip this altsetting */ - if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == - AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3) - return 1; /* skip this altsetting */ - if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == - AUDIOPHILE_SET_16B_48K_DI && altno != 4) - return 1; /* skip this altsetting */ - if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == - AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) - return 1; /* skip this altsetting */ - } - return 0; /* keep this altsetting */ -} - -static int create_any_midi_quirk(struct snd_usb_audio *chip, - struct usb_interface *intf, - const struct snd_usb_audio_quirk *quirk) -{ - return snd_usbmidi_create(chip->card, intf, &chip->midi_list, quirk); -} - -/* - * audio-interface quirks - * - * returns zero if no standard audio/MIDI parsing is needed. - * returns a postive value if standard audio/midi interfaces are parsed - * after this. - * returns a negative value at error. - */ -static int snd_usb_create_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - typedef int (*quirk_func_t)(struct snd_usb_audio *, struct usb_interface *, - const struct snd_usb_audio_quirk *); - static const quirk_func_t quirk_funcs[] = { - [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, - [QUIRK_COMPOSITE] = create_composite_quirk, - [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, - [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, - [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, - [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, - [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, - [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, - [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, - [QUIRK_MIDI_CME] = create_any_midi_quirk, - [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, - [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, - [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk - }; - - if (quirk->type < QUIRK_TYPE_COUNT) { - return quirk_funcs[quirk->type](chip, iface, quirk); - } else { - snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); - return -ENXIO; - } -} - - -/* - * common proc files to show the usb device info - */ -static void proc_audio_usbbus_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) -{ - struct snd_usb_audio *chip = entry->private_data; - if (!chip->shutdown) - snd_iprintf(buffer, "%03d/%03d\n", chip->dev->bus->busnum, chip->dev->devnum); -} - -static void proc_audio_usbid_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) -{ - struct snd_usb_audio *chip = entry->private_data; - if (!chip->shutdown) - snd_iprintf(buffer, "%04x:%04x\n", - USB_ID_VENDOR(chip->usb_id), - USB_ID_PRODUCT(chip->usb_id)); -} - -static void snd_usb_audio_create_proc(struct snd_usb_audio *chip) -{ - struct snd_info_entry *entry; - if (!snd_card_proc_new(chip->card, "usbbus", &entry)) - snd_info_set_text_ops(entry, chip, proc_audio_usbbus_read); - if (!snd_card_proc_new(chip->card, "usbid", &entry)) - snd_info_set_text_ops(entry, chip, proc_audio_usbid_read); -} - -/* - * free the chip instance - * - * here we have to do not much, since pcm and controls are already freed - * - */ - -static int snd_usb_audio_free(struct snd_usb_audio *chip) -{ - kfree(chip); - return 0; -} - -static int snd_usb_audio_dev_free(struct snd_device *device) -{ - struct snd_usb_audio *chip = device->device_data; - return snd_usb_audio_free(chip); -} - - -/* - * create a chip instance and set its names. - */ -static int snd_usb_audio_create(struct usb_device *dev, int idx, - const struct snd_usb_audio_quirk *quirk, - struct snd_usb_audio **rchip) -{ - struct snd_card *card; - struct snd_usb_audio *chip; - int err, len; - char component[14]; - static struct snd_device_ops ops = { - .dev_free = snd_usb_audio_dev_free, - }; - - *rchip = NULL; - - if (snd_usb_get_speed(dev) != USB_SPEED_LOW && - snd_usb_get_speed(dev) != USB_SPEED_FULL && - snd_usb_get_speed(dev) != USB_SPEED_HIGH) { - snd_printk(KERN_ERR "unknown device speed %d\n", snd_usb_get_speed(dev)); - return -ENXIO; - } - - err = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); - if (err < 0) { - snd_printk(KERN_ERR "cannot create card instance %d\n", idx); - return err; - } - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (! chip) { - snd_card_free(card); - return -ENOMEM; - } - - chip->index = idx; - chip->dev = dev; - chip->card = card; - chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct)); - INIT_LIST_HEAD(&chip->pcm_list); - INIT_LIST_HEAD(&chip->midi_list); - INIT_LIST_HEAD(&chip->mixer_list); - - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_usb_audio_free(chip); - snd_card_free(card); - return err; - } - - strcpy(card->driver, "USB-Audio"); - sprintf(component, "USB%04x:%04x", - USB_ID_VENDOR(chip->usb_id), USB_ID_PRODUCT(chip->usb_id)); - snd_component_add(card, component); - - /* retrieve the device string as shortname */ - if (quirk && quirk->product_name) { - strlcpy(card->shortname, quirk->product_name, sizeof(card->shortname)); - } else { - if (!dev->descriptor.iProduct || - usb_string(dev, dev->descriptor.iProduct, - card->shortname, sizeof(card->shortname)) <= 0) { - /* no name available from anywhere, so use ID */ - sprintf(card->shortname, "USB Device %#04x:%#04x", - USB_ID_VENDOR(chip->usb_id), - USB_ID_PRODUCT(chip->usb_id)); - } - } - - /* retrieve the vendor and device strings as longname */ - if (quirk && quirk->vendor_name) { - len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname)); - } else { - if (dev->descriptor.iManufacturer) - len = usb_string(dev, dev->descriptor.iManufacturer, - card->longname, sizeof(card->longname)); - else - len = 0; - /* we don't really care if there isn't any vendor string */ - } - if (len > 0) - strlcat(card->longname, " ", sizeof(card->longname)); - - strlcat(card->longname, card->shortname, sizeof(card->longname)); - - len = strlcat(card->longname, " at ", sizeof(card->longname)); - - if (len < sizeof(card->longname)) - usb_make_path(dev, card->longname + len, sizeof(card->longname) - len); - - strlcat(card->longname, - snd_usb_get_speed(dev) == USB_SPEED_LOW ? ", low speed" : - snd_usb_get_speed(dev) == USB_SPEED_FULL ? ", full speed" : - ", high speed", - sizeof(card->longname)); - - snd_usb_audio_create_proc(chip); - - *rchip = chip; - return 0; -} - - -/* - * probe the active usb device - * - * note that this can be called multiple times per a device, when it - * includes multiple audio control interfaces. - * - * thus we check the usb device pointer and creates the card instance - * only at the first time. the successive calls of this function will - * append the pcm interface to the corresponding card. - */ -static void *snd_usb_audio_probe(struct usb_device *dev, - struct usb_interface *intf, - const struct usb_device_id *usb_id) -{ - const struct snd_usb_audio_quirk *quirk = (const struct snd_usb_audio_quirk *)usb_id->driver_info; - int i, err; - struct snd_usb_audio *chip; - struct usb_host_interface *alts; - int ifnum; - u32 id; - - alts = &intf->altsetting[0]; - ifnum = get_iface_desc(alts)->bInterfaceNumber; - id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct)); - if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) - goto __err_val; - - /* SB Extigy needs special boot-up sequence */ - /* if more models come, this will go to the quirk list. */ - if (id == USB_ID(0x041e, 0x3000)) { - if (snd_usb_extigy_boot_quirk(dev, intf) < 0) - goto __err_val; - } - /* SB Audigy 2 NX needs its own boot-up magic, too */ - if (id == USB_ID(0x041e, 0x3020)) { - if (snd_usb_audigy2nx_boot_quirk(dev) < 0) - goto __err_val; - } - - /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */ - if (id == USB_ID(0x10f5, 0x0200)) { - if (snd_usb_cm106_boot_quirk(dev) < 0) - goto __err_val; - } - - /* C-Media CM6206 / CM106-Like Sound Device */ - if (id == USB_ID(0x0d8c, 0x0102)) { - if (snd_usb_cm6206_boot_quirk(dev) < 0) - goto __err_val; - } - - /* Access Music VirusTI Desktop */ - if (id == USB_ID(0x133e, 0x0815)) { - if (snd_usb_accessmusic_boot_quirk(dev) < 0) - goto __err_val; - } - - /* - * found a config. now register to ALSA - */ - - /* check whether it's already registered */ - chip = NULL; - mutex_lock(®ister_mutex); - for (i = 0; i < SNDRV_CARDS; i++) { - if (usb_chip[i] && usb_chip[i]->dev == dev) { - if (usb_chip[i]->shutdown) { - snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); - goto __error; - } - chip = usb_chip[i]; - break; - } - } - if (! chip) { - /* it's a fresh one. - * now look for an empty slot and create a new card instance - */ - for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && - (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && - (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - if (snd_usb_audio_create(dev, i, quirk, &chip) < 0) { - goto __error; - } - snd_card_set_dev(chip->card, &intf->dev); - break; - } - if (!chip) { - printk(KERN_ERR "no available usb audio device\n"); - goto __error; - } - } - - chip->txfr_quirk = 0; - err = 1; /* continue */ - if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { - /* need some special handlings */ - if ((err = snd_usb_create_quirk(chip, intf, quirk)) < 0) - goto __error; - } - - if (err > 0) { - /* create normal USB audio interfaces */ - if (snd_usb_create_streams(chip, ifnum) < 0 || - snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { - goto __error; - } - } - - /* we are allowed to call snd_card_register() many times */ - if (snd_card_register(chip->card) < 0) { - goto __error; - } - - usb_chip[chip->index] = chip; - chip->num_interfaces++; - mutex_unlock(®ister_mutex); - return chip; - - __error: - if (chip && !chip->num_interfaces) - snd_card_free(chip->card); - mutex_unlock(®ister_mutex); - __err_val: - return NULL; -} - -/* - * we need to take care of counter, since disconnection can be called also - * many times as well as usb_audio_probe(). - */ -static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) -{ - struct snd_usb_audio *chip; - struct snd_card *card; - struct list_head *p; - - if (ptr == (void *)-1L) - return; - - chip = ptr; - card = chip->card; - mutex_lock(®ister_mutex); - chip->shutdown = 1; - chip->num_interfaces--; - if (chip->num_interfaces <= 0) { - snd_card_disconnect(card); - /* release the pcm resources */ - list_for_each(p, &chip->pcm_list) { - snd_usb_stream_disconnect(p); - } - /* release the midi resources */ - list_for_each(p, &chip->midi_list) { - snd_usbmidi_disconnect(p); - } - /* release mixer resources */ - list_for_each(p, &chip->mixer_list) { - snd_usb_mixer_disconnect(p); - } - usb_chip[chip->index] = NULL; - mutex_unlock(®ister_mutex); - snd_card_free_when_closed(card); - } else { - mutex_unlock(®ister_mutex); - } -} - -/* - * new 2.5 USB kernel API - */ -static int usb_audio_probe(struct usb_interface *intf, - const struct usb_device_id *id) -{ - void *chip; - chip = snd_usb_audio_probe(interface_to_usbdev(intf), intf, id); - if (chip) { - usb_set_intfdata(intf, chip); - return 0; - } else - return -EIO; -} - -static void usb_audio_disconnect(struct usb_interface *intf) -{ - snd_usb_audio_disconnect(interface_to_usbdev(intf), - usb_get_intfdata(intf)); -} - -#ifdef CONFIG_PM -static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) -{ - struct snd_usb_audio *chip = usb_get_intfdata(intf); - struct list_head *p; - struct snd_usb_stream *as; - - if (chip == (void *)-1L) - return 0; - - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); - if (!chip->num_suspended_intf++) { - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - snd_pcm_suspend_all(as->pcm); - } - } - - return 0; -} - -static int usb_audio_resume(struct usb_interface *intf) -{ - struct snd_usb_audio *chip = usb_get_intfdata(intf); - - if (chip == (void *)-1L) - return 0; - if (--chip->num_suspended_intf) - return 0; - /* - * ALSA leaves material resumption to user space - * we just notify - */ - - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - - return 0; -} -#endif /* CONFIG_PM */ - -static int __init snd_usb_audio_init(void) -{ - if (nrpacks < 1 || nrpacks > MAX_PACKS) { - printk(KERN_WARNING "invalid nrpacks value.\n"); - return -EINVAL; - } - return usb_register(&usb_audio_driver); -} - - -static void __exit snd_usb_audio_cleanup(void) -{ - usb_deregister(&usb_audio_driver); -} - -module_init(snd_usb_audio_init); -module_exit(snd_usb_audio_cleanup); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 42c299cbf63a..d679e72a3e5c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,15 +21,13 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -/* maximum number of endpoints per interface */ -#define MIDI_MAX_ENDPOINTS 2 - /* handling of USB vendor/product ID pairs as 32-bit numbers */ #define USB_ID(vendor, product) (((vendor) << 16) | (product)) #define USB_ID_VENDOR(id) ((id) >> 16) #define USB_ID_PRODUCT(id) ((u16)(id)) /* + * */ struct snd_usb_audio { @@ -51,6 +49,10 @@ struct snd_usb_audio { struct list_head midi_list; /* list of midi interfaces */ struct list_head mixer_list; /* list of mixer interfaces */ + + int setup; /* from the 'device_setup' module param */ + int nrpacks; /* from the 'nrpacks' module param */ + int async_unlink; /* from the 'async_unlink' module param */ }; /* @@ -89,93 +91,8 @@ struct snd_usb_audio_quirk { const void *data; }; -/* data for QUIRK_MIDI_FIXED_ENDPOINT */ -struct snd_usb_midi_endpoint_info { - int8_t out_ep; /* ep number, 0 autodetect */ - uint8_t out_interval; /* interval for interrupt endpoints */ - int8_t in_ep; - uint8_t in_interval; - uint16_t out_cables; /* bitmask */ - uint16_t in_cables; /* bitmask */ -}; - -/* for QUIRK_MIDI_YAMAHA, data is NULL */ - -/* for QUIRK_MIDI_MIDIMAN, data points to a snd_usb_midi_endpoint_info - * structure (out_cables and in_cables only) */ - -/* for QUIRK_COMPOSITE, data points to an array of snd_usb_audio_quirk - * structures, terminated with .ifnum = -1 */ - -/* for QUIRK_AUDIO_FIXED_ENDPOINT, data points to an audioformat structure */ - -/* for QUIRK_AUDIO/MIDI_STANDARD_INTERFACE, data is NULL */ - -/* for QUIRK_AUDIO_EDIROL_UAXX, data is NULL */ - -/* for QUIRK_IGNORE_INTERFACE, data is NULL */ - -/* for QUIRK_MIDI_NOVATION and _RAW, data is NULL */ - -/* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info - * structure (out_cables and in_cables only) */ - -/* for QUIRK_MIDI_CME, data is NULL */ - -/* - */ - -/*E-mu USB samplerate control quirk*/ -enum { - EMU_QUIRK_SR_44100HZ = 0, - EMU_QUIRK_SR_48000HZ, - EMU_QUIRK_SR_88200HZ, - EMU_QUIRK_SR_96000HZ, - EMU_QUIRK_SR_176400HZ, - EMU_QUIRK_SR_192000HZ -}; - #define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) -unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); - -void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); -void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); - -int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, - __u8 request, __u8 requesttype, __u16 value, __u16 index, - void *data, __u16 size, int timeout); - -int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, - int ignore_error); -void snd_usb_mixer_disconnect(struct list_head *p); - -int snd_usbmidi_create(struct snd_card *card, - struct usb_interface *iface, - struct list_head *midi_list, - const struct snd_usb_audio_quirk *quirk); -void snd_usbmidi_input_stop(struct list_head* p); -void snd_usbmidi_input_start(struct list_head* p); -void snd_usbmidi_disconnect(struct list_head *p); - -void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, - unsigned char samplerate_id); - -/* - * retrieve usb_interface descriptor from the host interface - * (conditional for compatibility with the older API) - */ -#ifndef get_iface_desc -#define get_iface_desc(iface) (&(iface)->desc) -#define get_endpoint(alt,ep) (&(alt)->endpoint[ep].desc) -#define get_ep_desc(ep) (&(ep)->desc) -#define get_cfg_desc(cfg) (&(cfg)->desc) -#endif - -#ifndef snd_usb_get_speed -#define snd_usb_get_speed(dev) ((dev)->speed) -#endif - #endif /* __USBAUDIO_H */ diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 44deb21b1777..5f7b942ff577 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -25,6 +25,7 @@ #define MODNAME "US122L" #include "usb_stream.c" #include "../usbaudio.h" +#include "../midi.h" #include "us122l.h" MODULE_AUTHOR("Karsten Wiese <fzu@wemgehoertderstaat.de>"); diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 1d174cea352b..e43c0a86441a 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -1,6 +1,7 @@ #ifndef USBUSX2Y_H #define USBUSX2Y_H #include "../usbaudio.h" +#include "../midi.h" #include "usbus428ctldefs.h" #define NRURBS 2 |