mirror of
https://github.com/BobTheBlinker/android_kernel_motorola_sm6375.git
synced 2026-10-28 18:51:56 -04:00
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
ALSA: usb-audio: add Yamaha MOX6/MOX8 support
ALSA: virtuoso: add S/PDIF input support for all Xonars
ALSA: ice1724 - Support for ooAoo SQ210a
ALSA: ice1724 - Allow card info based on model only
ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
ALSA: hdspm - Provide unique driver id based on card serial
ASoC: Dynamically allocate the rtd device for a non-empty release()
ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
ALSA: hda/cirrus - support for iMac12,2 model
ASoC: cx20442: add bias control over a platform provided regulator
ALSA: usb-audio - Avoid flood of frame-active debug messages
ALSA: snd-usb-us122l: Delete calls to preempt_disable
mfd: Put WM8994 into cache only mode when suspending
...
Fix up trivial conflicts in:
- arch/arm/mach-s3c64xx/mach-crag6410.c:
renamed speyside_wm8962 to tobermory, added littlemill right
next to it
- drivers/base/regmap/{regcache.c,regmap.c}:
duplicate diff that had already come in with other changes in
the regmap tree
This commit is contained in:
Linus Torvalds
commit
a429638cac
510 changed files with 17132 additions and 16566 deletions
|
|
@ -6,3 +6,5 @@ header-y += hdsp.h
|
|||
header-y += hdspm.h
|
||||
header-y += sb16_csp.h
|
||||
header-y += sfnt_info.h
|
||||
header-y += compress_params.h
|
||||
header-y += compress_offload.h
|
||||
|
|
|
|||
167
include/sound/compress_driver.h
Normal file
167
include/sound/compress_driver.h
Normal file
|
|
@ -0,0 +1,167 @@
|
|||
/*
|
||||
* compress_driver.h - compress offload driver definations
|
||||
*
|
||||
* Copyright (C) 2011 Intel Corporation
|
||||
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
|
||||
* Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.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 of the License.
|
||||
*
|
||||
* 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 __COMPRESS_DRIVER_H
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||||
#define __COMPRESS_DRIVER_H
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||||
|
||||
#include <linux/types.h>
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||||
#include <linux/sched.h>
|
||||
#include <sound/compress_offload.h>
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||||
#include <sound/asound.h>
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#include <sound/pcm.h>
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||||
|
||||
struct snd_compr_ops;
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||||
|
||||
/**
|
||||
* struct snd_compr_runtime: runtime stream description
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||||
* @state: stream state
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||||
* @ops: pointer to DSP callbacks
|
||||
* @buffer: pointer to kernel buffer, valid only when not in mmap mode or
|
||||
* DSP doesn't implement copy
|
||||
* @buffer_size: size of the above buffer
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||||
* @fragment_size: size of buffer fragment in bytes
|
||||
* @fragments: number of such fragments
|
||||
* @hw_pointer: offset of last location in buffer where DSP copied data
|
||||
* @app_pointer: offset of last location in buffer where app wrote data
|
||||
* @total_bytes_available: cumulative number of bytes made available in
|
||||
* the ring buffer
|
||||
* @total_bytes_transferred: cumulative bytes transferred by offload DSP
|
||||
* @sleep: poll sleep
|
||||
*/
|
||||
struct snd_compr_runtime {
|
||||
snd_pcm_state_t state;
|
||||
struct snd_compr_ops *ops;
|
||||
void *buffer;
|
||||
u64 buffer_size;
|
||||
u32 fragment_size;
|
||||
u32 fragments;
|
||||
u64 hw_pointer;
|
||||
u64 app_pointer;
|
||||
u64 total_bytes_available;
|
||||
u64 total_bytes_transferred;
|
||||
wait_queue_head_t sleep;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_stream: compressed stream
|
||||
* @name: device name
|
||||
* @ops: pointer to DSP callbacks
|
||||
* @runtime: pointer to runtime structure
|
||||
* @device: device pointer
|
||||
* @direction: stream direction, playback/recording
|
||||
* @private_data: pointer to DSP private data
|
||||
*/
|
||||
struct snd_compr_stream {
|
||||
const char *name;
|
||||
struct snd_compr_ops *ops;
|
||||
struct snd_compr_runtime *runtime;
|
||||
struct snd_compr *device;
|
||||
enum snd_compr_direction direction;
|
||||
void *private_data;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_ops: compressed path DSP operations
|
||||
* @open: Open the compressed stream
|
||||
* This callback is mandatory and shall keep dsp ready to receive the stream
|
||||
* parameter
|
||||
* @free: Close the compressed stream, mandatory
|
||||
* @set_params: Sets the compressed stream parameters, mandatory
|
||||
* This can be called in during stream creation only to set codec params
|
||||
* and the stream properties
|
||||
* @get_params: retrieve the codec parameters, mandatory
|
||||
* @trigger: Trigger operations like start, pause, resume, drain, stop.
|
||||
* This callback is mandatory
|
||||
* @pointer: Retrieve current h/w pointer information. Mandatory
|
||||
* @copy: Copy the compressed data to/from userspace, Optional
|
||||
* Can't be implemented if DSP supports mmap
|
||||
* @mmap: DSP mmap method to mmap DSP memory
|
||||
* @ack: Ack for DSP when data is written to audio buffer, Optional
|
||||
* Not valid if copy is implemented
|
||||
* @get_caps: Retrieve DSP capabilities, mandatory
|
||||
* @get_codec_caps: Retrieve capabilities for a specific codec, mandatory
|
||||
*/
|
||||
struct snd_compr_ops {
|
||||
int (*open)(struct snd_compr_stream *stream);
|
||||
int (*free)(struct snd_compr_stream *stream);
|
||||
int (*set_params)(struct snd_compr_stream *stream,
|
||||
struct snd_compr_params *params);
|
||||
int (*get_params)(struct snd_compr_stream *stream,
|
||||
struct snd_codec *params);
|
||||
int (*trigger)(struct snd_compr_stream *stream, int cmd);
|
||||
int (*pointer)(struct snd_compr_stream *stream,
|
||||
struct snd_compr_tstamp *tstamp);
|
||||
int (*copy)(struct snd_compr_stream *stream, const char __user *buf,
|
||||
size_t count);
|
||||
int (*mmap)(struct snd_compr_stream *stream,
|
||||
struct vm_area_struct *vma);
|
||||
int (*ack)(struct snd_compr_stream *stream, size_t bytes);
|
||||
int (*get_caps) (struct snd_compr_stream *stream,
|
||||
struct snd_compr_caps *caps);
|
||||
int (*get_codec_caps) (struct snd_compr_stream *stream,
|
||||
struct snd_compr_codec_caps *codec);
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr: Compressed device
|
||||
* @name: DSP device name
|
||||
* @dev: Device pointer
|
||||
* @ops: pointer to DSP callbacks
|
||||
* @private_data: pointer to DSP pvt data
|
||||
* @card: sound card pointer
|
||||
* @direction: Playback or capture direction
|
||||
* @lock: device lock
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||||
* @device: device id
|
||||
*/
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||||
struct snd_compr {
|
||||
const char *name;
|
||||
struct device *dev;
|
||||
struct snd_compr_ops *ops;
|
||||
void *private_data;
|
||||
struct snd_card *card;
|
||||
unsigned int direction;
|
||||
struct mutex lock;
|
||||
int device;
|
||||
};
|
||||
|
||||
/* compress device register APIs */
|
||||
int snd_compress_register(struct snd_compr *device);
|
||||
int snd_compress_deregister(struct snd_compr *device);
|
||||
int snd_compress_new(struct snd_card *card, int device,
|
||||
int type, struct snd_compr *compr);
|
||||
|
||||
/* dsp driver callback apis
|
||||
* For playback: driver should call snd_compress_fragment_elapsed() to let the
|
||||
* framework know that a fragment has been consumed from the ring buffer
|
||||
*
|
||||
* For recording: we want to know when a frame is available or when
|
||||
* at least one frame is available so snd_compress_frame_elapsed()
|
||||
* callback should be called when a encodeded frame is available
|
||||
*/
|
||||
static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream)
|
||||
{
|
||||
wake_up(&stream->runtime->sleep);
|
||||
}
|
||||
|
||||
#endif
|
||||
161
include/sound/compress_offload.h
Normal file
161
include/sound/compress_offload.h
Normal file
|
|
@ -0,0 +1,161 @@
|
|||
/*
|
||||
* compress_offload.h - compress offload header definations
|
||||
*
|
||||
* Copyright (C) 2011 Intel Corporation
|
||||
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
|
||||
* Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.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 of the License.
|
||||
*
|
||||
* 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 __COMPRESS_OFFLOAD_H
|
||||
#define __COMPRESS_OFFLOAD_H
|
||||
|
||||
#include <linux/types.h>
|
||||
#include <sound/asound.h>
|
||||
#include <sound/compress_params.h>
|
||||
|
||||
|
||||
#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
|
||||
/**
|
||||
* struct snd_compressed_buffer: compressed buffer
|
||||
* @fragment_size: size of buffer fragment in bytes
|
||||
* @fragments: number of such fragments
|
||||
*/
|
||||
struct snd_compressed_buffer {
|
||||
__u32 fragment_size;
|
||||
__u32 fragments;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_params: compressed stream params
|
||||
* @buffer: buffer description
|
||||
* @codec: codec parameters
|
||||
* @no_wake_mode: dont wake on fragment elapsed
|
||||
*/
|
||||
struct snd_compr_params {
|
||||
struct snd_compressed_buffer buffer;
|
||||
struct snd_codec codec;
|
||||
__u8 no_wake_mode;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_tstamp: timestamp descriptor
|
||||
* @byte_offset: Byte offset in ring buffer to DSP
|
||||
* @copied_total: Total number of bytes copied from/to ring buffer to/by DSP
|
||||
* @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by
|
||||
* large steps and should only be used to monitor encoding/decoding
|
||||
* progress. It shall not be used for timing estimates.
|
||||
* @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio
|
||||
* output/input. This field should be used for A/V sync or time estimates.
|
||||
* @sampling_rate: sampling rate of audio
|
||||
*/
|
||||
struct snd_compr_tstamp {
|
||||
__u32 byte_offset;
|
||||
__u32 copied_total;
|
||||
snd_pcm_uframes_t pcm_frames;
|
||||
snd_pcm_uframes_t pcm_io_frames;
|
||||
__u32 sampling_rate;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_avail: avail descriptor
|
||||
* @avail: Number of bytes available in ring buffer for writing/reading
|
||||
* @tstamp: timestamp infomation
|
||||
*/
|
||||
struct snd_compr_avail {
|
||||
__u64 avail;
|
||||
struct snd_compr_tstamp tstamp;
|
||||
};
|
||||
|
||||
enum snd_compr_direction {
|
||||
SND_COMPRESS_PLAYBACK = 0,
|
||||
SND_COMPRESS_CAPTURE
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_caps: caps descriptor
|
||||
* @codecs: pointer to array of codecs
|
||||
* @direction: direction supported. Of type snd_compr_direction
|
||||
* @min_fragment_size: minimum fragment supported by DSP
|
||||
* @max_fragment_size: maximum fragment supported by DSP
|
||||
* @min_fragments: min fragments supported by DSP
|
||||
* @max_fragments: max fragments supported by DSP
|
||||
* @num_codecs: number of codecs supported
|
||||
* @reserved: reserved field
|
||||
*/
|
||||
struct snd_compr_caps {
|
||||
__u32 num_codecs;
|
||||
__u32 direction;
|
||||
__u32 min_fragment_size;
|
||||
__u32 max_fragment_size;
|
||||
__u32 min_fragments;
|
||||
__u32 max_fragments;
|
||||
__u32 codecs[MAX_NUM_CODECS];
|
||||
__u32 reserved[11];
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_compr_codec_caps: query capability of codec
|
||||
* @codec: codec for which capability is queried
|
||||
* @num_descriptors: number of codec descriptors
|
||||
* @descriptor: array of codec capability descriptor
|
||||
*/
|
||||
struct snd_compr_codec_caps {
|
||||
__u32 codec;
|
||||
__u32 num_descriptors;
|
||||
struct snd_codec_desc descriptor[MAX_NUM_CODEC_DESCRIPTORS];
|
||||
};
|
||||
|
||||
/**
|
||||
* compress path ioctl definitions
|
||||
* SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
|
||||
* SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
|
||||
* SNDRV_COMPRESS_SET_PARAMS: Set codec and stream parameters
|
||||
* Note: only codec params can be changed runtime and stream params cant be
|
||||
* SNDRV_COMPRESS_GET_PARAMS: Query codec params
|
||||
* SNDRV_COMPRESS_TSTAMP: get the current timestamp value
|
||||
* SNDRV_COMPRESS_AVAIL: get the current buffer avail value.
|
||||
* This also queries the tstamp properties
|
||||
* SNDRV_COMPRESS_PAUSE: Pause the running stream
|
||||
* SNDRV_COMPRESS_RESUME: resume a paused stream
|
||||
* SNDRV_COMPRESS_START: Start a stream
|
||||
* SNDRV_COMPRESS_STOP: stop a running stream, discarding ring buffer content
|
||||
* and the buffers currently with DSP
|
||||
* SNDRV_COMPRESS_DRAIN: Play till end of buffers and stop after that
|
||||
* SNDRV_COMPRESS_IOCTL_VERSION: Query the API version
|
||||
*/
|
||||
#define SNDRV_COMPRESS_IOCTL_VERSION _IOR('C', 0x00, int)
|
||||
#define SNDRV_COMPRESS_GET_CAPS _IOWR('C', 0x10, struct snd_compr_caps)
|
||||
#define SNDRV_COMPRESS_GET_CODEC_CAPS _IOWR('C', 0x11,\
|
||||
struct snd_compr_codec_caps)
|
||||
#define SNDRV_COMPRESS_SET_PARAMS _IOW('C', 0x12, struct snd_compr_params)
|
||||
#define SNDRV_COMPRESS_GET_PARAMS _IOR('C', 0x13, struct snd_codec)
|
||||
#define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp)
|
||||
#define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail)
|
||||
#define SNDRV_COMPRESS_PAUSE _IO('C', 0x30)
|
||||
#define SNDRV_COMPRESS_RESUME _IO('C', 0x31)
|
||||
#define SNDRV_COMPRESS_START _IO('C', 0x32)
|
||||
#define SNDRV_COMPRESS_STOP _IO('C', 0x33)
|
||||
#define SNDRV_COMPRESS_DRAIN _IO('C', 0x34)
|
||||
/*
|
||||
* TODO
|
||||
* 1. add mmap support
|
||||
*
|
||||
*/
|
||||
#define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
|
||||
#endif
|
||||
397
include/sound/compress_params.h
Normal file
397
include/sound/compress_params.h
Normal file
|
|
@ -0,0 +1,397 @@
|
|||
/*
|
||||
* compress_params.h - codec types and parameters for compressed data
|
||||
* streaming interface
|
||||
*
|
||||
* Copyright (C) 2011 Intel Corporation
|
||||
* Authors: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
|
||||
* Vinod Koul <vinod.koul@linux.intel.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 of the License.
|
||||
*
|
||||
* 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.
|
||||
*
|
||||
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
|
||||
*
|
||||
* The definitions in this file are derived from the OpenMAX AL version 1.1
|
||||
* and OpenMAX IL v 1.1.2 header files which contain the copyright notice below.
|
||||
*
|
||||
* Copyright (c) 2007-2010 The Khronos Group Inc.
|
||||
*
|
||||
* Permission is hereby granted, free of charge, to any person obtaining
|
||||
* a copy of this software and/or associated documentation files (the
|
||||
* "Materials "), to deal in the Materials without restriction, including
|
||||
* without limitation the rights to use, copy, modify, merge, publish,
|
||||
* distribute, sublicense, and/or sell copies of the Materials, and to
|
||||
* permit persons to whom the Materials are furnished to do so, subject to
|
||||
* the following conditions:
|
||||
*
|
||||
* The above copyright notice and this permission notice shall be included
|
||||
* in all copies or substantial portions of the Materials.
|
||||
*
|
||||
* THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS
|
||||
* OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
|
||||
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
|
||||
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
|
||||
* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
|
||||
* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
|
||||
* MATERIALS OR THE USE OR OTHER DEALINGS IN THE MATERIALS.
|
||||
*
|
||||
*/
|
||||
#ifndef __SND_COMPRESS_PARAMS_H
|
||||
#define __SND_COMPRESS_PARAMS_H
|
||||
|
||||
/* AUDIO CODECS SUPPORTED */
|
||||
#define MAX_NUM_CODECS 32
|
||||
#define MAX_NUM_CODEC_DESCRIPTORS 32
|
||||
#define MAX_NUM_BITRATES 32
|
||||
|
||||
/* Codecs are listed linearly to allow for extensibility */
|
||||
#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001)
|
||||
#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOCODEC_AMR ((__u32) 0x00000003)
|
||||
#define SND_AUDIOCODEC_AMRWB ((__u32) 0x00000004)
|
||||
#define SND_AUDIOCODEC_AMRWBPLUS ((__u32) 0x00000005)
|
||||
#define SND_AUDIOCODEC_AAC ((__u32) 0x00000006)
|
||||
#define SND_AUDIOCODEC_WMA ((__u32) 0x00000007)
|
||||
#define SND_AUDIOCODEC_REAL ((__u32) 0x00000008)
|
||||
#define SND_AUDIOCODEC_VORBIS ((__u32) 0x00000009)
|
||||
#define SND_AUDIOCODEC_FLAC ((__u32) 0x0000000A)
|
||||
#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B)
|
||||
#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C)
|
||||
#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D)
|
||||
|
||||
/*
|
||||
* Profile and modes are listed with bit masks. This allows for a
|
||||
* more compact representation of fields that will not evolve
|
||||
* (in contrast to the list of codecs)
|
||||
*/
|
||||
|
||||
#define SND_AUDIOPROFILE_PCM ((__u32) 0x00000001)
|
||||
|
||||
/* MP3 modes are only useful for encoders */
|
||||
#define SND_AUDIOCHANMODE_MP3_MONO ((__u32) 0x00000001)
|
||||
#define SND_AUDIOCHANMODE_MP3_STEREO ((__u32) 0x00000002)
|
||||
#define SND_AUDIOCHANMODE_MP3_JOINTSTEREO ((__u32) 0x00000004)
|
||||
#define SND_AUDIOCHANMODE_MP3_DUAL ((__u32) 0x00000008)
|
||||
|
||||
#define SND_AUDIOPROFILE_AMR ((__u32) 0x00000001)
|
||||
|
||||
/* AMR modes are only useful for encoders */
|
||||
#define SND_AUDIOMODE_AMR_DTX_OFF ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_AMR_VAD1 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_AMR_VAD2 ((__u32) 0x00000004)
|
||||
|
||||
#define SND_AUDIOSTREAMFORMAT_UNDEFINED ((__u32) 0x00000000)
|
||||
#define SND_AUDIOSTREAMFORMAT_CONFORMANCE ((__u32) 0x00000001)
|
||||
#define SND_AUDIOSTREAMFORMAT_IF1 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOSTREAMFORMAT_IF2 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOSTREAMFORMAT_FSF ((__u32) 0x00000008)
|
||||
#define SND_AUDIOSTREAMFORMAT_RTPPAYLOAD ((__u32) 0x00000010)
|
||||
#define SND_AUDIOSTREAMFORMAT_ITU ((__u32) 0x00000020)
|
||||
|
||||
#define SND_AUDIOPROFILE_AMRWB ((__u32) 0x00000001)
|
||||
|
||||
/* AMRWB modes are only useful for encoders */
|
||||
#define SND_AUDIOMODE_AMRWB_DTX_OFF ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_AMRWB_VAD1 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_AMRWB_VAD2 ((__u32) 0x00000004)
|
||||
|
||||
#define SND_AUDIOPROFILE_AMRWBPLUS ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOPROFILE_AAC ((__u32) 0x00000001)
|
||||
|
||||
/* AAC modes are required for encoders and decoders */
|
||||
#define SND_AUDIOMODE_AAC_MAIN ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_AAC_LC ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_AAC_SSR ((__u32) 0x00000004)
|
||||
#define SND_AUDIOMODE_AAC_LTP ((__u32) 0x00000008)
|
||||
#define SND_AUDIOMODE_AAC_HE ((__u32) 0x00000010)
|
||||
#define SND_AUDIOMODE_AAC_SCALABLE ((__u32) 0x00000020)
|
||||
#define SND_AUDIOMODE_AAC_ERLC ((__u32) 0x00000040)
|
||||
#define SND_AUDIOMODE_AAC_LD ((__u32) 0x00000080)
|
||||
#define SND_AUDIOMODE_AAC_HE_PS ((__u32) 0x00000100)
|
||||
#define SND_AUDIOMODE_AAC_HE_MPS ((__u32) 0x00000200)
|
||||
|
||||
/* AAC formats are required for encoders and decoders */
|
||||
#define SND_AUDIOSTREAMFORMAT_MP2ADTS ((__u32) 0x00000001)
|
||||
#define SND_AUDIOSTREAMFORMAT_MP4ADTS ((__u32) 0x00000002)
|
||||
#define SND_AUDIOSTREAMFORMAT_MP4LOAS ((__u32) 0x00000004)
|
||||
#define SND_AUDIOSTREAMFORMAT_MP4LATM ((__u32) 0x00000008)
|
||||
#define SND_AUDIOSTREAMFORMAT_ADIF ((__u32) 0x00000010)
|
||||
#define SND_AUDIOSTREAMFORMAT_MP4FF ((__u32) 0x00000020)
|
||||
#define SND_AUDIOSTREAMFORMAT_RAW ((__u32) 0x00000040)
|
||||
|
||||
#define SND_AUDIOPROFILE_WMA7 ((__u32) 0x00000001)
|
||||
#define SND_AUDIOPROFILE_WMA8 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOPROFILE_WMA9 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOPROFILE_WMA10 ((__u32) 0x00000008)
|
||||
|
||||
#define SND_AUDIOMODE_WMA_LEVEL1 ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_WMA_LEVEL2 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_WMA_LEVEL3 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOMODE_WMA_LEVEL4 ((__u32) 0x00000008)
|
||||
#define SND_AUDIOMODE_WMAPRO_LEVELM0 ((__u32) 0x00000010)
|
||||
#define SND_AUDIOMODE_WMAPRO_LEVELM1 ((__u32) 0x00000020)
|
||||
#define SND_AUDIOMODE_WMAPRO_LEVELM2 ((__u32) 0x00000040)
|
||||
#define SND_AUDIOMODE_WMAPRO_LEVELM3 ((__u32) 0x00000080)
|
||||
|
||||
#define SND_AUDIOSTREAMFORMAT_WMA_ASF ((__u32) 0x00000001)
|
||||
/*
|
||||
* Some implementations strip the ASF header and only send ASF packets
|
||||
* to the DSP
|
||||
*/
|
||||
#define SND_AUDIOSTREAMFORMAT_WMA_NOASF_HDR ((__u32) 0x00000002)
|
||||
|
||||
#define SND_AUDIOPROFILE_REALAUDIO ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOMODE_REALAUDIO_G2 ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_REALAUDIO_8 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_REALAUDIO_10 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOMODE_REALAUDIO_SURROUND ((__u32) 0x00000008)
|
||||
|
||||
#define SND_AUDIOPROFILE_VORBIS ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOMODE_VORBIS ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOPROFILE_FLAC ((__u32) 0x00000001)
|
||||
|
||||
/*
|
||||
* Define quality levels for FLAC encoders, from LEVEL0 (fast)
|
||||
* to LEVEL8 (best)
|
||||
*/
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL0 ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL1 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL2 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL3 ((__u32) 0x00000008)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL4 ((__u32) 0x00000010)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL5 ((__u32) 0x00000020)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL6 ((__u32) 0x00000040)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL7 ((__u32) 0x00000080)
|
||||
#define SND_AUDIOMODE_FLAC_LEVEL8 ((__u32) 0x00000100)
|
||||
|
||||
#define SND_AUDIOSTREAMFORMAT_FLAC ((__u32) 0x00000001)
|
||||
#define SND_AUDIOSTREAMFORMAT_FLAC_OGG ((__u32) 0x00000002)
|
||||
|
||||
/* IEC61937 payloads without CUVP and preambles */
|
||||
#define SND_AUDIOPROFILE_IEC61937 ((__u32) 0x00000001)
|
||||
/* IEC61937 with S/PDIF preambles+CUVP bits in 32-bit containers */
|
||||
#define SND_AUDIOPROFILE_IEC61937_SPDIF ((__u32) 0x00000002)
|
||||
|
||||
/*
|
||||
* IEC modes are mandatory for decoders. Format autodetection
|
||||
* will only happen on the DSP side with mode 0. The PCM mode should
|
||||
* not be used, the PCM codec should be used instead.
|
||||
*/
|
||||
#define SND_AUDIOMODE_IEC_REF_STREAM_HEADER ((__u32) 0x00000000)
|
||||
#define SND_AUDIOMODE_IEC_LPCM ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_IEC_AC3 ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_IEC_MPEG1 ((__u32) 0x00000004)
|
||||
#define SND_AUDIOMODE_IEC_MP3 ((__u32) 0x00000008)
|
||||
#define SND_AUDIOMODE_IEC_MPEG2 ((__u32) 0x00000010)
|
||||
#define SND_AUDIOMODE_IEC_AACLC ((__u32) 0x00000020)
|
||||
#define SND_AUDIOMODE_IEC_DTS ((__u32) 0x00000040)
|
||||
#define SND_AUDIOMODE_IEC_ATRAC ((__u32) 0x00000080)
|
||||
#define SND_AUDIOMODE_IEC_SACD ((__u32) 0x00000100)
|
||||
#define SND_AUDIOMODE_IEC_EAC3 ((__u32) 0x00000200)
|
||||
#define SND_AUDIOMODE_IEC_DTS_HD ((__u32) 0x00000400)
|
||||
#define SND_AUDIOMODE_IEC_MLP ((__u32) 0x00000800)
|
||||
#define SND_AUDIOMODE_IEC_DST ((__u32) 0x00001000)
|
||||
#define SND_AUDIOMODE_IEC_WMAPRO ((__u32) 0x00002000)
|
||||
#define SND_AUDIOMODE_IEC_REF_CXT ((__u32) 0x00004000)
|
||||
#define SND_AUDIOMODE_IEC_HE_AAC ((__u32) 0x00008000)
|
||||
#define SND_AUDIOMODE_IEC_HE_AAC2 ((__u32) 0x00010000)
|
||||
#define SND_AUDIOMODE_IEC_MPEG_SURROUND ((__u32) 0x00020000)
|
||||
|
||||
#define SND_AUDIOPROFILE_G723_1 ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOMODE_G723_1_ANNEX_A ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_G723_1_ANNEX_B ((__u32) 0x00000002)
|
||||
#define SND_AUDIOMODE_G723_1_ANNEX_C ((__u32) 0x00000004)
|
||||
|
||||
#define SND_AUDIOPROFILE_G729 ((__u32) 0x00000001)
|
||||
|
||||
#define SND_AUDIOMODE_G729_ANNEX_A ((__u32) 0x00000001)
|
||||
#define SND_AUDIOMODE_G729_ANNEX_B ((__u32) 0x00000002)
|
||||
|
||||
/* <FIXME: multichannel encoders aren't supported for now. Would need
|
||||
an additional definition of channel arrangement> */
|
||||
|
||||
/* VBR/CBR definitions */
|
||||
#define SND_RATECONTROLMODE_CONSTANTBITRATE ((__u32) 0x00000001)
|
||||
#define SND_RATECONTROLMODE_VARIABLEBITRATE ((__u32) 0x00000002)
|
||||
|
||||
/* Encoder options */
|
||||
|
||||
struct snd_enc_wma {
|
||||
__u32 super_block_align; /* WMA Type-specific data */
|
||||
};
|
||||
|
||||
|
||||
/**
|
||||
* struct snd_enc_vorbis
|
||||
* @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
|
||||
* In the default mode of operation, the quality level is 3.
|
||||
* Normal quality range is 0 - 10.
|
||||
* @managed: Boolean. Set bitrate management mode. This turns off the
|
||||
* normal VBR encoding, but allows hard or soft bitrate constraints to be
|
||||
* enforced by the encoder. This mode can be slower, and may also be
|
||||
* lower quality. It is primarily useful for streaming.
|
||||
* @max_bit_rate: Enabled only if managed is TRUE
|
||||
* @min_bit_rate: Enabled only if managed is TRUE
|
||||
* @downmix: Boolean. Downmix input from stereo to mono (has no effect on
|
||||
* non-stereo streams). Useful for lower-bitrate encoding.
|
||||
*
|
||||
* These options were extracted from the OpenMAX IL spec and Gstreamer vorbisenc
|
||||
* properties
|
||||
*
|
||||
* For best quality users should specify VBR mode and set quality levels.
|
||||
*/
|
||||
|
||||
struct snd_enc_vorbis {
|
||||
__s32 quality;
|
||||
__u32 managed;
|
||||
__u32 max_bit_rate;
|
||||
__u32 min_bit_rate;
|
||||
__u32 downmix;
|
||||
};
|
||||
|
||||
|
||||
/**
|
||||
* struct snd_enc_real
|
||||
* @quant_bits: number of coupling quantization bits in the stream
|
||||
* @start_region: coupling start region in the stream
|
||||
* @num_regions: number of regions value
|
||||
*
|
||||
* These options were extracted from the OpenMAX IL spec
|
||||
*/
|
||||
|
||||
struct snd_enc_real {
|
||||
__u32 quant_bits;
|
||||
__u32 start_region;
|
||||
__u32 num_regions;
|
||||
};
|
||||
|
||||
/**
|
||||
* struct snd_enc_flac
|
||||
* @num: serial number, valid only for OGG formats
|
||||
* needs to be set by application
|
||||
* @gain: Add replay gain tags
|
||||
*
|
||||
* These options were extracted from the FLAC online documentation
|
||||
* at http://flac.sourceforge.net/documentation_tools_flac.html
|
||||
*
|
||||
* To make the API simpler, it is assumed that the user will select quality
|
||||
* profiles. Additional options that affect encoding quality and speed can
|
||||
* be added at a later stage if needed.
|
||||
*
|
||||
* By default the Subset format is used by encoders.
|
||||
*
|
||||
* TAGS such as pictures, etc, cannot be handled by an offloaded encoder and are
|
||||
* not supported in this API.
|
||||
*/
|
||||
|
||||
struct snd_enc_flac {
|
||||
__u32 num;
|
||||
__u32 gain;
|
||||
};
|
||||
|
||||
struct snd_enc_generic {
|
||||
__u32 bw; /* encoder bandwidth */
|
||||
__s32 reserved[15];
|
||||
};
|
||||
|
||||
union snd_codec_options {
|
||||
struct snd_enc_wma wma;
|
||||
struct snd_enc_vorbis vorbis;
|
||||
struct snd_enc_real real;
|
||||
struct snd_enc_flac flac;
|
||||
struct snd_enc_generic generic;
|
||||
};
|
||||
|
||||
/** struct snd_codec_desc - description of codec capabilities
|
||||
* @max_ch: Maximum number of audio channels
|
||||
* @sample_rates: Sampling rates in Hz, use SNDRV_PCM_RATE_xxx for this
|
||||
* @bit_rate: Indexed array containing supported bit rates
|
||||
* @num_bitrates: Number of valid values in bit_rate array
|
||||
* @rate_control: value is specified by SND_RATECONTROLMODE defines.
|
||||
* @profiles: Supported profiles. See SND_AUDIOPROFILE defines.
|
||||
* @modes: Supported modes. See SND_AUDIOMODE defines
|
||||
* @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines
|
||||
* @min_buffer: Minimum buffer size handled by codec implementation
|
||||
* @reserved: reserved for future use
|
||||
*
|
||||
* This structure provides a scalar value for profiles, modes and stream
|
||||
* format fields.
|
||||
* If an implementation supports multiple combinations, they will be listed as
|
||||
* codecs with different descriptors, for example there would be 2 descriptors
|
||||
* for AAC-RAW and AAC-ADTS.
|
||||
* This entails some redundancy but makes it easier to avoid invalid
|
||||
* configurations.
|
||||
*
|
||||
*/
|
||||
|
||||
struct snd_codec_desc {
|
||||
__u32 max_ch;
|
||||
__u32 sample_rates;
|
||||
__u32 bit_rate[MAX_NUM_BITRATES];
|
||||
__u32 num_bitrates;
|
||||
__u32 rate_control;
|
||||
__u32 profiles;
|
||||
__u32 modes;
|
||||
__u32 formats;
|
||||
__u32 min_buffer;
|
||||
__u32 reserved[15];
|
||||
};
|
||||
|
||||
/** struct snd_codec
|
||||
* @id: Identifies the supported audio encoder/decoder.
|
||||
* See SND_AUDIOCODEC macros.
|
||||
* @ch_in: Number of input audio channels
|
||||
* @ch_out: Number of output channels. In case of contradiction between
|
||||
* this field and the channelMode field, the channelMode field
|
||||
* overrides.
|
||||
* @sample_rate: Audio sample rate of input data
|
||||
* @bit_rate: Bitrate of encoded data. May be ignored by decoders
|
||||
* @rate_control: Encoding rate control. See SND_RATECONTROLMODE defines.
|
||||
* Encoders may rely on profiles for quality levels.
|
||||
* May be ignored by decoders.
|
||||
* @profile: Mandatory for encoders, can be mandatory for specific
|
||||
* decoders as well. See SND_AUDIOPROFILE defines.
|
||||
* @level: Supported level (Only used by WMA at the moment)
|
||||
* @ch_mode: Channel mode for encoder. See SND_AUDIOCHANMODE defines
|
||||
* @format: Format of encoded bistream. Mandatory when defined.
|
||||
* See SND_AUDIOSTREAMFORMAT defines.
|
||||
* @align: Block alignment in bytes of an audio sample.
|
||||
* Only required for PCM or IEC formats.
|
||||
* @options: encoder-specific settings
|
||||
* @reserved: reserved for future use
|
||||
*/
|
||||
|
||||
struct snd_codec {
|
||||
__u32 id;
|
||||
__u32 ch_in;
|
||||
__u32 ch_out;
|
||||
__u32 sample_rate;
|
||||
__u32 bit_rate;
|
||||
__u32 rate_control;
|
||||
__u32 profile;
|
||||
__u32 level;
|
||||
__u32 ch_mode;
|
||||
__u32 format;
|
||||
__u32 align;
|
||||
union snd_codec_options options;
|
||||
__u32 reserved[3];
|
||||
};
|
||||
|
||||
#endif
|
||||
|
|
@ -227,4 +227,12 @@ snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
|
|||
return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
|
||||
}
|
||||
|
||||
/*
|
||||
* Helper functions for jack-detection controls
|
||||
*/
|
||||
struct snd_kcontrol *
|
||||
snd_kctl_jack_new(const char *name, int idx, void *private_data);
|
||||
void snd_kctl_jack_report(struct snd_card *card,
|
||||
struct snd_kcontrol *kctl, bool status);
|
||||
|
||||
#endif /* __SOUND_CONTROL_H */
|
||||
|
|
|
|||
|
|
@ -62,6 +62,7 @@ typedef int __bitwise snd_device_type_t;
|
|||
#define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007)
|
||||
#define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008)
|
||||
#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009)
|
||||
#define SNDRV_DEV_COMPRESS ((__force snd_device_type_t) 0x100A)
|
||||
#define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000)
|
||||
|
||||
typedef int __bitwise snd_device_state_t;
|
||||
|
|
|
|||
|
|
@ -35,7 +35,7 @@
|
|||
#define SNDRV_MINOR_TIMER 33 /* SNDRV_MINOR_GLOBAL + 1 * 32 */
|
||||
|
||||
#ifndef CONFIG_SND_DYNAMIC_MINORS
|
||||
/* 2 - 3 (reserved) */
|
||||
#define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */
|
||||
#define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */
|
||||
#define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */
|
||||
#define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */
|
||||
|
|
@ -49,6 +49,7 @@
|
|||
#define SNDRV_DEVICE_TYPE_PCM_CAPTURE SNDRV_MINOR_PCM_CAPTURE
|
||||
#define SNDRV_DEVICE_TYPE_SEQUENCER SNDRV_MINOR_SEQUENCER
|
||||
#define SNDRV_DEVICE_TYPE_TIMER SNDRV_MINOR_TIMER
|
||||
#define SNDRV_DEVICE_TYPE_COMPRESS SNDRV_MINOR_COMPRESS
|
||||
|
||||
#else /* CONFIG_SND_DYNAMIC_MINORS */
|
||||
|
||||
|
|
@ -60,6 +61,7 @@ enum {
|
|||
SNDRV_DEVICE_TYPE_RAWMIDI,
|
||||
SNDRV_DEVICE_TYPE_PCM_PLAYBACK,
|
||||
SNDRV_DEVICE_TYPE_PCM_CAPTURE,
|
||||
SNDRV_DEVICE_TYPE_COMPRESS,
|
||||
};
|
||||
|
||||
#endif /* CONFIG_SND_DYNAMIC_MINORS */
|
||||
|
|
|
|||
|
|
@ -78,4 +78,16 @@ struct sh_fsi_platform_info {
|
|||
int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
|
||||
};
|
||||
|
||||
/*
|
||||
* for fsi-ak4642
|
||||
*/
|
||||
struct fsi_ak4642_info {
|
||||
const char *name;
|
||||
const char *card;
|
||||
const char *cpu_dai;
|
||||
const char *codec;
|
||||
const char *platform;
|
||||
int id;
|
||||
};
|
||||
|
||||
#endif /* __SOUND_FSI_H */
|
||||
|
|
|
|||
|
|
@ -43,6 +43,9 @@
|
|||
.num_kcontrols = 0}
|
||||
|
||||
/* platform domain */
|
||||
#define SND_SOC_DAPM_SIGGEN(wname) \
|
||||
{ .id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
|
||||
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
|
||||
#define SND_SOC_DAPM_INPUT(wname) \
|
||||
{ .id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
|
||||
.num_kcontrols = 0, .reg = SND_SOC_NOPM }
|
||||
|
|
@ -380,6 +383,7 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
|
|||
const char *pin);
|
||||
int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
|
||||
const char *pin);
|
||||
void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
|
||||
|
||||
/* Mostly internal - should not normally be used */
|
||||
void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
|
||||
|
|
@ -409,6 +413,7 @@ enum snd_soc_dapm_type {
|
|||
snd_soc_dapm_supply, /* power/clock supply */
|
||||
snd_soc_dapm_aif_in, /* audio interface input */
|
||||
snd_soc_dapm_aif_out, /* audio interface output */
|
||||
snd_soc_dapm_siggen, /* signal generator */
|
||||
};
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -231,6 +231,7 @@ enum snd_soc_bias_level {
|
|||
SND_SOC_BIAS_ON = 3,
|
||||
};
|
||||
|
||||
struct device_node;
|
||||
struct snd_jack;
|
||||
struct snd_soc_card;
|
||||
struct snd_soc_pcm_stream;
|
||||
|
|
@ -266,8 +267,6 @@ enum snd_soc_control_type {
|
|||
|
||||
enum snd_soc_compress_type {
|
||||
SND_SOC_FLAT_COMPRESSION = 1,
|
||||
SND_SOC_LZO_COMPRESSION,
|
||||
SND_SOC_RBTREE_COMPRESSION
|
||||
};
|
||||
|
||||
enum snd_soc_pcm_subclass {
|
||||
|
|
@ -318,6 +317,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform,
|
|||
unsigned int reg);
|
||||
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);
|
||||
|
||||
/* Utility functions to get clock rates from various things */
|
||||
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
|
||||
|
|
@ -593,8 +593,7 @@ struct snd_soc_codec_driver {
|
|||
/* driver ops */
|
||||
int (*probe)(struct snd_soc_codec *);
|
||||
int (*remove)(struct snd_soc_codec *);
|
||||
int (*suspend)(struct snd_soc_codec *,
|
||||
pm_message_t state);
|
||||
int (*suspend)(struct snd_soc_codec *);
|
||||
int (*resume)(struct snd_soc_codec *);
|
||||
|
||||
/* Default control and setup, added after probe() is run */
|
||||
|
|
@ -706,8 +705,11 @@ struct snd_soc_dai_link {
|
|||
const char *name; /* Codec name */
|
||||
const char *stream_name; /* Stream name */
|
||||
const char *codec_name; /* for multi-codec */
|
||||
const struct device_node *codec_of_node;
|
||||
const char *platform_name; /* for multi-platform */
|
||||
const struct device_node *platform_of_node;
|
||||
const char *cpu_dai_name;
|
||||
const struct device_node *cpu_dai_of_node;
|
||||
const char *codec_dai_name;
|
||||
|
||||
unsigned int dai_fmt; /* format to set on init */
|
||||
|
|
@ -718,6 +720,9 @@ struct snd_soc_dai_link {
|
|||
/* Symmetry requirements */
|
||||
unsigned int symmetric_rates:1;
|
||||
|
||||
/* pmdown_time is ignored at stop */
|
||||
unsigned int ignore_pmdown_time:1;
|
||||
|
||||
/* codec/machine specific init - e.g. add machine controls */
|
||||
int (*init)(struct snd_soc_pcm_runtime *rtd);
|
||||
|
||||
|
|
@ -813,6 +818,7 @@ struct snd_soc_card {
|
|||
int num_dapm_widgets;
|
||||
const struct snd_soc_dapm_route *dapm_routes;
|
||||
int num_dapm_routes;
|
||||
bool fully_routed;
|
||||
|
||||
struct work_struct deferred_resume_work;
|
||||
|
||||
|
|
@ -840,8 +846,8 @@ struct snd_soc_card {
|
|||
};
|
||||
|
||||
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
|
||||
struct snd_soc_pcm_runtime {
|
||||
struct device dev;
|
||||
struct snd_soc_pcm_runtime {
|
||||
struct device *dev;
|
||||
struct snd_soc_card *card;
|
||||
struct snd_soc_dai_link *dai_link;
|
||||
struct mutex pcm_mutex;
|
||||
|
|
@ -927,12 +933,12 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo
|
|||
static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd,
|
||||
void *data)
|
||||
{
|
||||
dev_set_drvdata(&rtd->dev, data);
|
||||
dev_set_drvdata(rtd->dev, data);
|
||||
}
|
||||
|
||||
static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd)
|
||||
{
|
||||
return dev_get_drvdata(&rtd->dev);
|
||||
return dev_get_drvdata(rtd->dev);
|
||||
}
|
||||
|
||||
static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
|
||||
|
|
@ -960,6 +966,11 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
|
|||
int snd_soc_util_init(void);
|
||||
void snd_soc_util_exit(void);
|
||||
|
||||
int snd_soc_of_parse_card_name(struct snd_soc_card *card,
|
||||
const char *propname);
|
||||
int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
|
||||
const char *propname);
|
||||
|
||||
#include <sound/soc-dai.h>
|
||||
|
||||
#ifdef CONFIG_DEBUG_FS
|
||||
|
|
|
|||
35
include/sound/sta32x.h
Normal file
35
include/sound/sta32x.h
Normal file
|
|
@ -0,0 +1,35 @@
|
|||
/*
|
||||
* Platform data for ST STA32x ASoC codec driver.
|
||||
*
|
||||
* Copyright: 2011 Raumfeld GmbH
|
||||
* Author: Johannes Stezenbach <js@sig21.net>
|
||||
*
|
||||
* 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 __LINUX_SND__STA32X_H
|
||||
#define __LINUX_SND__STA32X_H
|
||||
|
||||
#define STA32X_OCFG_2CH 0
|
||||
#define STA32X_OCFG_2_1CH 1
|
||||
#define STA32X_OCFG_1CH 3
|
||||
|
||||
#define STA32X_OM_CH1 0
|
||||
#define STA32X_OM_CH2 1
|
||||
#define STA32X_OM_CH3 2
|
||||
|
||||
#define STA32X_THERMAL_ADJUSTMENT_ENABLE 1
|
||||
#define STA32X_THERMAL_RECOVERY_ENABLE 2
|
||||
|
||||
struct sta32x_platform_data {
|
||||
int output_conf;
|
||||
int ch1_output_mapping;
|
||||
int ch2_output_mapping;
|
||||
int ch3_output_mapping;
|
||||
int thermal_conf;
|
||||
int needs_esd_watchdog;
|
||||
};
|
||||
|
||||
#endif /* __LINUX_SND__STA32X_H */
|
||||
|
|
@ -11,8 +11,11 @@
|
|||
#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
|
||||
/*
|
||||
* Used to enable configuration of a GPIO to all zeros; a gpio_cfg value of
|
||||
* zero in platform data means "don't touch this pin".
|
||||
*/
|
||||
#define WM8903_GPIO_CONFIG_ZERO 0x8000
|
||||
|
||||
/*
|
||||
* R6 (0x06) - Mic Bias Control 0
|
||||
|
|
|
|||
Loading…
Reference in a new issue