diff options
Diffstat (limited to 'include')
| -rw-r--r-- | include/linux/mfd/wm8994/pdata.h | 4 | ||||
| -rw-r--r-- | include/linux/platform_data/davinci_asp.h | 104 | ||||
| -rw-r--r-- | include/sound/compress_driver.h | 1 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 3 | ||||
| -rw-r--r-- | include/sound/soc.h | 16 | ||||
| -rw-r--r-- | include/sound/wm0010.h | 27 | ||||
| -rw-r--r-- | include/sound/wm8993.h | 4 |
7 files changed, 158 insertions, 1 deletions
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index f0361c0..fc87be4 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -164,6 +164,10 @@ struct wm8994_pdata { int num_micd_rates; struct wm8958_micd_rate *micd_rates; + /* Power up delays to add after microphone bias power up (ms) */ + int micb1_delay; + int micb2_delay; + /* LINEOUT can be differential or single ended */ unsigned int lineout1_diff:1; unsigned int lineout2_diff:1; diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h new file mode 100644 index 0000000..79c26aa --- /dev/null +++ b/include/linux/platform_data/davinci_asp.h @@ -0,0 +1,104 @@ +/* + * TI DaVinci Audio Serial Port support + * + * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.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 version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any + * kind, whether express or implied; without even the implied warranty + * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __DAVINCI_ASP_H +#define __DAVINCI_ASP_H + +struct snd_platform_data { + u32 tx_dma_offset; + u32 rx_dma_offset; + int asp_chan_q; /* event queue number for ASP channel */ + int ram_chan_q; /* event queue number for RAM channel */ + unsigned int codec_fmt; + /* + * Allowing this is more efficient and eliminates left and right swaps + * caused by underruns, but will swap the left and right channels + * when compared to previous behavior. + */ + unsigned enable_channel_combine:1; + unsigned sram_size_playback; + unsigned sram_size_capture; + + /* + * If McBSP peripheral gets the clock from an external pin, + * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR + * and MCBSP_CLKS. + * Depending on different hardware connections it is possible + * to use this setting to change the behaviour of McBSP + * driver. + */ + int clk_input_pin; + + /* + * This flag works when both clock and FS are outputs for the cpu + * and makes clock more accurate (FS is not symmetrical and the + * clock is very fast. + * The clock becoming faster is named + * i2s continuous serial clock (I2S_SCK) and it is an externally + * visible bit clock. + * + * first line : WordSelect + * second line : ContinuousSerialClock + * third line: SerialData + * + * SYMMETRICAL APPROACH: + * _______________________ LEFT + * _| RIGHT |______________________| + * _ _ _ _ _ _ _ _ + * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_ + * _ _ _ _ _ _ _ _ + * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_ + * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/ + * + * ACCURATE CLOCK APPROACH: + * ______________ LEFT + * _| RIGHT |_______________________________| + * _ _ _ _ _ _ _ _ _ + * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| | + * _ _ _ _ dummy cycles + * _/ \_ ... _/ \_/ \_ ... _/ \__________________ + * \_/ \_/ \_/ \_/ + * + */ + bool i2s_accurate_sck; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ +}; + +enum mcbsp_clk_input_pin { + MCBSP_CLKR = 0, /* as in DM365 */ + MCBSP_CLKS, +}; + +#define INACTIVE_MODE 0 +#define TX_MODE 1 +#define RX_MODE 2 + +#define DAVINCI_MCASP_IIS_MODE 0 +#define DAVINCI_MCASP_DIT_MODE 1 + +#endif diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 48f2a1f..f2912ab 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -61,6 +61,7 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; + void *private_data; }; /** diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1f69e0a..628db7b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -18,6 +18,7 @@ struct snd_pcm_substream; struct snd_soc_dapm_widget; +struct snd_compr_stream; /* * DAI hardware audio formats. @@ -205,6 +206,8 @@ struct snd_soc_dai_driver { int (*remove)(struct snd_soc_dai *dai); int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); + /* compress dai */ + bool compress_dai; /* ops */ const struct snd_soc_dai_ops *ops; diff --git a/include/sound/soc.h b/include/sound/soc.h index e063380..313b766 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -22,6 +22,7 @@ #include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> +#include <sound/compress_driver.h> #include <sound/control.h> #include <sound/ac97_codec.h> @@ -399,6 +400,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); @@ -632,6 +634,13 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; +struct snd_soc_compr_ops { + int (*startup)(struct snd_compr_stream *); + void (*shutdown)(struct snd_compr_stream *); + int (*set_params)(struct snd_compr_stream *); + int (*trigger)(struct snd_compr_stream *); +}; + /* SoC cache ops */ struct snd_soc_cache_ops { const char *name; @@ -787,9 +796,12 @@ struct snd_soc_platform_driver { snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *); - /* platform stream ops */ + /* platform stream pcm ops */ struct snd_pcm_ops *ops; + /* platform stream compress ops */ + struct snd_compr_ops *compr_ops; + /* platform stream completion event */ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); @@ -891,6 +903,7 @@ struct snd_soc_dai_link { /* machine stream operations */ struct snd_soc_ops *ops; + struct snd_soc_compr_ops *compr_ops; }; struct snd_soc_codec_conf { @@ -1027,6 +1040,7 @@ struct snd_soc_pcm_runtime { /* runtime devices */ struct snd_pcm *pcm; + struct snd_compr *compr; struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai; diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h new file mode 100644 index 0000000..3261e90 --- /dev/null +++ b/include/sound/wm0010.h @@ -0,0 +1,27 @@ +/* + * wm0010.h -- Platform data for WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM0010_PDATA_H +#define WM0010_PDATA_H + +struct wm0010_pdata { + int gpio_reset; + + /* Set if there is an inverter between the GPIO controlling + * the reset signal and the device. + */ + int reset_active_high; + int irq_flags; +}; + +#endif diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h index eee19f6..8016fd8 100644 --- a/include/sound/wm8993.h +++ b/include/sound/wm8993.h @@ -32,6 +32,10 @@ struct wm8993_platform_data { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* Delay to add for microphones to stabalise after power up */ + int micbias1_delay; + int micbias2_delay; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; |
