diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/linux/tty.h | 4 | ||||
| -rw-r--r-- | include/sound/ac97_codec.h | 6 | ||||
| -rw-r--r-- | include/sound/memalloc.h | 6 | ||||
| -rw-r--r-- | include/sound/pcm.h | 23 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 27 | ||||
| -rw-r--r-- | include/sound/soc-dapm.h | 1 | ||||
| -rw-r--r-- | include/sound/soc.h | 47 | ||||
| -rw-r--r-- | include/sound/tlv.h | 14 | ||||
| -rw-r--r-- | include/sound/uda1380.h | 22 | ||||
| -rw-r--r-- | include/sound/wm8993.h | 44 |
10 files changed, 175 insertions, 19 deletions
diff --git a/include/linux/tty.h b/include/linux/tty.h index e8c6c9136c97..0d3974f59c53 100644 --- a/include/linux/tty.h +++ b/include/linux/tty.h @@ -23,7 +23,7 @@ */ #define NR_UNIX98_PTY_DEFAULT 4096 /* Default maximum for Unix98 ptys */ #define NR_UNIX98_PTY_MAX (1 << MINORBITS) /* Absolute limit */ -#define NR_LDISCS 19 +#define NR_LDISCS 20 /* line disciplines */ #define N_TTY 0 @@ -47,6 +47,8 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ +#define N_V253 19 /* Codec control over voice modem */ + /* * This character is the same as _POSIX_VDISABLE: it cannot be used as * a c_cc[] character, but indicates that a particular special character diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 251fc1cd5002..9b1c0985480c 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -642,4 +642,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime); /* ad hoc AC97 device driver access */ extern struct bus_type ac97_bus_type; +/* AC97 platform_data adding function */ +static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data) +{ + ac97->dev.platform_data = data; +} + #endif /* __SOUND_AC97_CODEC_H */ diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 7ccce94a5255..c42506212649 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -47,7 +47,11 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */ #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ +#ifdef CONFIG_SND_DMA_SGBUF #define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#endif /* * info for buffer allocation @@ -60,6 +64,7 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +#ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages */ @@ -107,6 +112,7 @@ static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset) { return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE; } +#endif /* CONFIG_SND_DMA_SGBUF */ /* allocate/release a buffer */ int snd_dma_alloc_pages(int type, struct device *dev, size_t size, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 23893523dc8c..1691c7fe35af 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -902,6 +902,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +#ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling */ @@ -927,6 +928,28 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, unsigned int ofs, unsigned int size); +#else /* !SND_DMA_SGBUF */ +/* + * fake using a continuous buffer + */ +static inline dma_addr_t +snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_addr + ofs; +} + +static inline void * +snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_area + ofs; +} + +#define snd_pcm_sgbuf_ops_page NULL + +#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size) + +#endif /* SND_DMA_SGBUF */ + /* handle mmap counter - PCM mmap callback should handle this counter properly */ static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area) { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7eee9b6d..25d62ac53fc5 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -27,8 +27,8 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ #define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ #define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ +#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ /* left and right justified also known as MSB and LSB respectively */ @@ -38,7 +38,7 @@ struct snd_pcm_substream; /* * DAI Clock gating. * - * DAI bit clocks can be be gated (disabled) when not the DAI is not + * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ @@ -51,21 +51,21 @@ struct snd_pcm_substream; * format. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ +#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. * * This is wrt the codec, the inverse is true for the interface - * i.e. if the codec is clk and frm master then the interface is + * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ +#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ +#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 @@ -116,12 +116,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); /* * Digital Audio Interface. * - * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 - * operations an capabilities. Codec and platfom drivers will register a this + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this * structure for every DAI they have. * * This structure covers the clocking, formating and ALSA operations for each - * interface a + * interface. */ struct snd_soc_dai_ops { /* @@ -179,6 +179,7 @@ struct snd_soc_dai { int ac97_control; struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ /* DAI callbacks */ int (*probe)(struct platform_device *pdev, diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f9a069..35814ced2d22 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -279,6 +279,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..dbb1702688cd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -135,6 +135,28 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -187,10 +209,25 @@ typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum snd_soc_control_type { + SND_SOC_CUSTOM, + SND_SOC_I2C, + SND_SOC_SPI, +}; + int snd_soc_register_platform(struct snd_soc_platform *platform); void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control); + +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); @@ -216,9 +253,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); @@ -356,8 +393,10 @@ struct snd_soc_codec { int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); + int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); hw_write_t hw_write; - hw_read_t hw_read; + unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; short reg_cache_size; short reg_cache_step; @@ -369,8 +408,6 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - struct list_head up_list; - struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2181ed..9fd5b19ccf5c 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h new file mode 100644 index 000000000000..381319c7000c --- /dev/null +++ b/include/sound/uda1380.h @@ -0,0 +1,22 @@ +/* + * UDA1380 ALSA SoC Codec driver + * + * Copyright 2009 Philipp Zabel + * + * 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 __UDA1380_H +#define __UDA1380_H + +struct uda1380_platform_data { + int gpio_power; + int gpio_reset; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#endif /* __UDA1380_H */ diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h new file mode 100644 index 000000000000..9c661f2f8cda --- /dev/null +++ b/include/sound/wm8993.h @@ -0,0 +1,44 @@ +/* + * linux/sound/wm8993.h -- Platform data for WM8993 + * + * Copyright 2009 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_WM8993_H +#define __LINUX_SND_WM8993_H + +/* Note that EQ1 only contains the enable/disable bit so will be + ignored but is included for simplicity. + */ +struct wm8993_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[24]; +}; + +struct wm8993_platform_data { + struct wm8993_retune_mobile_setting *retune_configs; + int num_retune_configs; + + /* LINEOUT can be differential or single ended */ + unsigned int lineout1_diff:1; + unsigned int lineout2_diff:1; + + /* Common mode feedback */ + unsigned int lineout1fb:1; + unsigned int lineout2fb:1; + + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + unsigned int micbias1_lvl:1; + unsigned int micbias2_lvl:1; + + /* Jack detect threashold levels, see datasheet for values */ + unsigned int jd_scthr:2; + unsigned int jd_thr:2; +}; + +#endif |