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-rw-r--r--include/sound/Kbuild2
-rw-r--r--include/sound/adau1373.h34
-rw-r--r--include/sound/asound.h4
-rw-r--r--include/sound/compress_driver.h167
-rw-r--r--include/sound/compress_offload.h161
-rw-r--r--include/sound/compress_params.h397
-rw-r--r--include/sound/control.h8
-rw-r--r--include/sound/core.h7
-rw-r--r--include/sound/info.h6
-rw-r--r--include/sound/initval.h2
-rw-r--r--include/sound/jack.h1
-rw-r--r--include/sound/minors.h4
-rw-r--r--include/sound/mpu401.h7
-rw-r--r--include/sound/pcm.h8
-rw-r--r--include/sound/saif.h16
-rw-r--r--include/sound/seq_kernel.h4
-rw-r--r--include/sound/sh_fsi.h12
-rw-r--r--include/sound/soc-dai.h37
-rw-r--r--include/sound/soc-dapm.h21
-rw-r--r--include/sound/soc.h134
-rw-r--r--include/sound/sta32x.h35
-rw-r--r--include/sound/tpa6130a2-plat.h6
-rw-r--r--include/sound/wm1250-ev1.h27
-rw-r--r--include/sound/wm5100.h59
-rw-r--r--include/sound/wm8903.h7
25 files changed, 1074 insertions, 92 deletions
diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index 802947f6091..6df30ed1581 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -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
diff --git a/include/sound/adau1373.h b/include/sound/adau1373.h
new file mode 100644
index 00000000000..1b19c766657
--- /dev/null
+++ b/include/sound/adau1373.h
@@ -0,0 +1,34 @@
+/*
+ * Analog Devices ADAU1373 Audio Codec drive
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef __SOUND_ADAU1373_H__
+#define __SOUND_ADAU1373_H__
+
+enum adau1373_micbias_voltage {
+ ADAU1373_MICBIAS_2_9V = 0,
+ ADAU1373_MICBIAS_2_2V = 1,
+ ADAU1373_MICBIAS_2_6V = 2,
+ ADAU1373_MICBIAS_1_8V = 3,
+};
+
+#define ADAU1373_DRC_SIZE 13
+
+struct adau1373_platform_data {
+ bool input_differential[4];
+ bool lineout_differential;
+ bool lineout_ground_sense;
+
+ unsigned int num_drc;
+ uint8_t drc_setting[3][ADAU1373_DRC_SIZE];
+
+ enum adau1373_micbias_voltage micbias1;
+ enum adau1373_micbias_voltage micbias2;
+};
+
+#endif
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 5d6074faa27..a2e4ff5ba9e 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -706,7 +706,7 @@ struct snd_timer_tread {
* *
****************************************************************************/
-#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6)
+#define SNDRV_CTL_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7)
struct snd_ctl_card_info {
int card; /* card number */
@@ -803,6 +803,8 @@ struct snd_ctl_elem_info {
unsigned int items; /* R: number of items */
unsigned int item; /* W: item number */
char name[64]; /* R: value name */
+ __u64 names_ptr; /* W: names list (ELEM_ADD only) */
+ unsigned int names_length;
} enumerated;
unsigned char reserved[128];
} value;
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
new file mode 100644
index 00000000000..48f2a1ff2bb
--- /dev/null
+++ b/include/sound/compress_driver.h
@@ -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
+#define __COMPRESS_DRIVER_H
+
+#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/compress_offload.h>
+#include <sound/asound.h>
+#include <sound/pcm.h>
+
+struct snd_compr_ops;
+
+/**
+ * struct snd_compr_runtime: runtime stream description
+ * @state: stream state
+ * @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
+ * @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
+ * @device: device id
+ */
+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
diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h
new file mode 100644
index 00000000000..05341a43fed
--- /dev/null
+++ b/include/sound/compress_offload.h
@@ -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
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
new file mode 100644
index 00000000000..d97d69f81a7
--- /dev/null
+++ b/include/sound/compress_params.h
@@ -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
diff --git a/include/sound/control.h b/include/sound/control.h
index 1a94a216ed9..b2796e83c7a 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -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 */
diff --git a/include/sound/core.h b/include/sound/core.h
index 1fa2407c966..5ab255f196c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -22,7 +22,6 @@
*
*/
-#include <linux/module.h>
#include <linux/sched.h> /* wake_up() */
#include <linux/mutex.h> /* struct mutex */
#include <linux/rwsem.h> /* struct rw_semaphore */
@@ -43,6 +42,7 @@
#ifdef CONFIG_PCI
struct pci_dev;
#endif
+struct module;
/* device allocation stuff */
@@ -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;
@@ -326,9 +327,9 @@ void release_and_free_resource(struct resource *res);
/* --- */
#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+__printf(4, 5)
void __snd_printk(unsigned int level, const char *file, int line,
- const char *format, ...)
- __attribute__ ((format (printf, 4, 5)));
+ const char *format, ...);
#else
#define __snd_printk(level, file, line, format, args...) \
printk(format, ##args)
diff --git a/include/sound/info.h b/include/sound/info.h
index 4e94cf1ff76..9ca1a493d37 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -72,7 +72,7 @@ struct snd_info_entry_ops {
struct snd_info_entry {
const char *name;
- mode_t mode;
+ umode_t mode;
long size;
unsigned short content;
union {
@@ -110,8 +110,8 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
-int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) \
- __attribute__ ((format (printf, 2, 3)));
+__printf(2, 3)
+int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...);
int snd_info_init(void);
int snd_info_done(void);
diff --git a/include/sound/initval.h b/include/sound/initval.h
index 1daa6dff829..f99a0d2ddfe 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -62,7 +62,7 @@ static int snd_legacy_find_free_irq(int *irq_table)
{
while (*irq_table != -1) {
if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
- IRQF_DISABLED | IRQF_PROBE_SHARED, "ALSA Test IRQ",
+ IRQF_PROBE_SHARED, "ALSA Test IRQ",
(void *) irq_table)) {
free_irq(*irq_table, (void *) irq_table);
return *irq_table;
diff --git a/include/sound/jack.h b/include/sound/jack.h
index c140fc7cbd3..63c790742db 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -42,6 +42,7 @@ enum snd_jack_types {
SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
SND_JACK_VIDEOOUT = 0x0010,
SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
+ SND_JACK_LINEIN = 0x0020,
/* Kept separate from switches to facilitate implementation */
SND_JACK_BTN_0 = 0x4000,
diff --git a/include/sound/minors.h b/include/sound/minors.h
index 8f764204a85..5978f9a8c8b 100644
--- a/include/sound/minors.h
+++ b/include/sound/minors.h
@@ -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 */
diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h
index 1f1d53f8830..20230db00ef 100644
--- a/include/sound/mpu401.h
+++ b/include/sound/mpu401.h
@@ -50,7 +50,10 @@
#define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */
#define MPU401_INFO_MMIO (1 << 3) /* MMIO access */
#define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */
+#define MPU401_INFO_IRQ_HOOK (1 << 5) /* mpu401 irq handler is called
+ from driver irq handler */
#define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */
+#define MPU401_INFO_USE_TIMER (1 << 15) /* internal */
#define MPU401_MODE_BIT_INPUT 0
#define MPU401_MODE_BIT_OUTPUT 1
@@ -73,8 +76,7 @@ struct snd_mpu401 {
unsigned long port; /* base port of MPU-401 chip */
unsigned long cport; /* port + 1 (usually) */
struct resource *res; /* port resource */
- int irq; /* IRQ number of MPU-401 chip (-1 = poll) */
- int irq_flags;
+ int irq; /* IRQ number of MPU-401 chip */
unsigned long mode; /* MPU401_MODE_XXXX */
int timer_invoked;
@@ -131,7 +133,6 @@ int snd_mpu401_uart_new(struct snd_card *card,
unsigned long port,
unsigned int info_flags,
int irq,
- int irq_flags,
struct snd_rawmidi ** rrawmidi);
#endif /* __SOUND_MPU401_H */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 57e71fa33f7..0cf91b2f08c 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -29,7 +29,7 @@
#include <linux/poll.h>
#include <linux/mm.h>
#include <linux/bitops.h>
-#include <linux/pm_qos_params.h>
+#include <linux/pm_qos.h>
#define snd_pcm_substream_chip(substream) ((substream)->private_data)
#define snd_pcm_chip(pcm) ((pcm)->private_data)
@@ -373,7 +373,7 @@ struct snd_pcm_substream {
int number;
char name[32]; /* substream name */
int stream; /* stream (direction) */
- struct pm_qos_request_list latency_pm_qos_req; /* pm_qos request */
+ struct pm_qos_request latency_pm_qos_req; /* pm_qos request */
size_t buffer_bytes_max; /* limit ring buffer size */
struct snd_dma_buffer dma_buffer;
unsigned int dma_buf_id;
@@ -825,6 +825,8 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime,
int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var);
+int snd_pcm_hw_rule_noresample(struct snd_pcm_runtime *runtime,
+ unsigned int base_rate);
int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime,
unsigned int cond,
int var,
@@ -1035,6 +1037,8 @@ static inline void snd_pcm_mmap_data_close(struct vm_area_struct *area)
atomic_dec(&substream->mmap_count);
}
+int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *area);
/* mmap for io-memory area */
#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA)
#define SNDRV_PCM_INFO_MMAP_IOMEM SNDRV_PCM_INFO_MMAP
diff --git a/include/sound/saif.h b/include/sound/saif.h
new file mode 100644
index 00000000000..f22f3e16edf
--- /dev/null
+++ b/include/sound/saif.h
@@ -0,0 +1,16 @@
+/*
+ * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __SOUND_SAIF_H__
+#define __SOUND_SAIF_H__
+
+struct mxs_saif_platform_data {
+ bool master_mode; /* if true use master mode */
+ int master_id; /* id of the master if in slave mode */
+};
+#endif
diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h
index 3d9afb6a8c9..f352a98ce4f 100644
--- a/include/sound/seq_kernel.h
+++ b/include/sound/seq_kernel.h
@@ -75,9 +75,9 @@ struct snd_seq_port_callback {
};
/* interface for kernel client */
+__printf(3, 4)
int snd_seq_create_kernel_client(struct snd_card *card, int client_index,
- const char *name_fmt, ...)
- __attribute__ ((format (printf, 3, 4)));
+ const char *name_fmt, ...);
int snd_seq_delete_kernel_client(int client);
int snd_seq_kernel_client_enqueue(int client, struct snd_seq_event *ev, int atomic, int hop);
int snd_seq_kernel_client_dispatch(int client, struct snd_seq_event *ev, int atomic, int hop);
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index 9a155f9d0a1..9b1aacaa82f 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -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 */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 5ad5f3a50c6..2413acc5488 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -24,13 +24,13 @@ struct snd_pcm_substream;
* Describes the physical PCM data formating and clocking. Add new formats
* to the end.
*/
-#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_AC97 5 /* AC97 */
-#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
+#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */
+#define SND_SOC_DAIFMT_AC97 6 /* AC97 */
+#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */
/* left and right justified also known as MSB and LSB respectively */
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
@@ -42,8 +42,8 @@ struct snd_pcm_substream;
* 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 */
-#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
+#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED (2 << 4) /* clock is gated */
/*
* DAI hardware signal inversions.
@@ -51,10 +51,10 @@ struct snd_pcm_substream;
* Specifies whether the DAI can also support inverted clocks for the specified
* 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_NF (1 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
/*
* DAI hardware clock masters.
@@ -63,10 +63,10 @@ struct snd_pcm_substream;
* 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_CFS (2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
+#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
@@ -242,6 +242,9 @@ struct snd_soc_dai {
void *playback_dma_data;
void *capture_dma_data;
+ /* Symmetry data - only valid if symmetry is being enforced */
+ unsigned int rate;
+
/* parent platform/codec */
union {
struct snd_soc_platform *platform;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e0583b7769c..d26a9b78477 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.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 +383,10 @@ 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);
/* dapm widget types */
enum snd_soc_dapm_type {
@@ -406,6 +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 */
};
/*
@@ -473,6 +481,8 @@ struct snd_soc_dapm_widget {
unsigned char ext:1; /* has external widgets */
unsigned char force:1; /* force state */
unsigned char ignore_suspend:1; /* kept enabled over suspend */
+ unsigned char new_power:1; /* power from this run */
+ unsigned char power_checked:1; /* power checked this run */
int subseq; /* sort within widget type */
int (*power_check)(struct snd_soc_dapm_widget *w);
@@ -492,6 +502,9 @@ struct snd_soc_dapm_widget {
/* used during DAPM updates */
struct list_head power_list;
+ struct list_head dirty;
+ int inputs;
+ int outputs;
};
struct snd_soc_dapm_update {
@@ -524,6 +537,8 @@ struct snd_soc_dapm_context {
enum snd_soc_bias_level target_bias_level;
struct list_head list;
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_dapm;
#endif
@@ -535,4 +550,10 @@ struct snd_soc_dapm_widget_list {
struct snd_soc_dapm_widget *widgets[0];
};
+struct snd_soc_dapm_stats {
+ int power_checks;
+ int path_checks;
+ int neighbour_checks;
+};
+
#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index aa19f5a32ba..0992dff5595 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -19,6 +19,7 @@
#include <linux/workqueue.h>
#include <linux/interrupt.h>
#include <linux/kernel.h>
+#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/control.h>
@@ -27,13 +28,20 @@
/*
* Convenience kcontrol builders
*/
-#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
+#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \
((unsigned long)&(struct soc_mixer_control) \
- {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
- .platform_max = xmax, .invert = xinvert})
+ {.reg = xreg, .rreg = xreg, .shift = shift_left, \
+ .rshift = shift_right, .max = xmax, .platform_max = xmax, \
+ .invert = xinvert})
+#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
+ SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert)
#define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
((unsigned long)&(struct soc_mixer_control) \
{.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_VALUE(xlreg, xrreg, xshift, xmax, xinvert) \
+ ((unsigned long)&(struct soc_mixer_control) \
+ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+ .max = xmax, .platform_max = xmax, .invert = xinvert})
#define SOC_SINGLE(xname, reg, shift, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
@@ -47,40 +55,36 @@
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
-#define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) \
+#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
- .private_value = (unsigned long)&(struct soc_mixer_control) \
- {.reg = xreg, .shift = shift_left, .rshift = shift_right, \
- .max = xmax, .platform_max = xmax, .invert = xinvert} }
+ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
+ max, invert) }
#define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
- .info = snd_soc_info_volsw_2r, \
- .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
- .private_value = (unsigned long)&(struct soc_mixer_control) \
- {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
- .max = xmax, .platform_max = xmax, .invert = xinvert} }
-#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
+#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
.put = snd_soc_put_volsw, \
- .private_value = (unsigned long)&(struct soc_mixer_control) \
- {.reg = xreg, .shift = shift_left, .rshift = shift_right,\
- .max = xmax, .platform_max = xmax, .invert = xinvert} }
+ .private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
+ max, invert) }
#define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
- .info = snd_soc_info_volsw_2r, \
- .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
- .private_value = (unsigned long)&(struct soc_mixer_control) \
- {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
- .max = xmax, .platform_max = xmax, .invert = xinvert} }
+ .info = snd_soc_info_volsw, \
+ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, xinvert) }
#define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -120,14 +124,13 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
-#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+#define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\
xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.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, .platform_max = xmax, .invert = xinvert} }
+ .private_value = \
+ SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -145,20 +148,18 @@
.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, .platform_max = xmax, .invert = xinvert} }
+ .private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \
+ xmax, 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, \
+ .info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
- .private_value = (unsigned long)&(struct soc_mixer_control) \
- {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
- .max = xmax, .platform_max = xmax, .invert = xinvert} }
+ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
+ xmax, 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, \
@@ -230,6 +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;
@@ -260,12 +262,11 @@ extern struct snd_ac97_bus_ops soc_ac97_ops;
enum snd_soc_control_type {
SND_SOC_I2C = 1,
SND_SOC_SPI,
+ SND_SOC_REGMAP,
};
enum snd_soc_compress_type {
SND_SOC_FLAT_COMPRESSION = 1,
- SND_SOC_LZO_COMPRESSION,
- SND_SOC_RBTREE_COMPRESSION
};
enum snd_soc_pcm_subclass {
@@ -274,7 +275,7 @@ enum snd_soc_pcm_subclass {
};
int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
- unsigned int freq, int dir);
+ int source, unsigned int freq, int dir);
int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
unsigned int freq_in, unsigned int freq_out);
@@ -316,6 +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);
@@ -391,12 +393,8 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
-int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo);
-int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
+#define snd_soc_get_volsw_2r snd_soc_get_volsw
+#define snd_soc_put_volsw_2r snd_soc_put_volsw
int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
@@ -576,9 +574,11 @@ struct snd_soc_codec {
const void *reg_def_copy;
const struct snd_soc_cache_ops *cache_ops;
struct mutex cache_rw_mutex;
+ int val_bytes;
/* dapm */
struct snd_soc_dapm_context dapm;
+ unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_codec_root;
@@ -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 */
@@ -607,7 +606,7 @@ struct snd_soc_codec_driver {
/* codec wide operations */
int (*set_sysclk)(struct snd_soc_codec *codec,
- int clk_id, unsigned int freq, int dir);
+ int clk_id, int source, unsigned int freq, int dir);
int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source,
unsigned int freq_in, unsigned int freq_out);
@@ -619,7 +618,7 @@ struct snd_soc_codec_driver {
int (*volatile_register)(struct snd_soc_codec *, unsigned int);
int (*readable_register)(struct snd_soc_codec *, unsigned int);
int (*writable_register)(struct snd_soc_codec *, unsigned int);
- short reg_cache_size;
+ unsigned int reg_cache_size;
short reg_cache_step;
short reg_word_size;
const void *reg_cache_default;
@@ -630,10 +629,14 @@ struct snd_soc_codec_driver {
/* codec bias level */
int (*set_bias_level)(struct snd_soc_codec *,
enum snd_soc_bias_level level);
+ bool idle_bias_off;
void (*seq_notifier)(struct snd_soc_dapm_context *,
enum snd_soc_dapm_type, int);
+ /* codec stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -669,6 +672,9 @@ struct snd_soc_platform_driver {
/* platform stream ops */
struct snd_pcm_ops *ops;
+ /* platform stream completion event */
+ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
+
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -699,16 +705,24 @@ 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 */
+
/* Keep DAI active over suspend */
unsigned int ignore_suspend:1;
/* 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);
@@ -804,6 +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;
@@ -815,9 +830,11 @@ struct snd_soc_card {
struct list_head widgets;
struct list_head paths;
struct list_head dapm_list;
+ struct list_head dapm_dirty;
/* Generic DAPM context for the card */
struct snd_soc_dapm_context dapm;
+ struct snd_soc_dapm_stats dapm_stats;
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_card_root;
@@ -829,8 +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;
@@ -840,8 +857,6 @@ struct snd_soc_pcm_runtime {
unsigned int complete:1;
unsigned int dev_registered:1;
- /* Symmetry data - only valid if symmetry is being enforced */
- unsigned int rate;
long pmdown_time;
/* runtime devices */
@@ -918,12 +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)
@@ -936,9 +951,26 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
INIT_LIST_HEAD(&card->dapm_list);
}
+static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
+{
+ if (mc->reg == mc->rreg && mc->shift == mc->rshift)
+ return 0;
+ /*
+ * mc->reg == mc->rreg && mc->shift != mc->rshift, or
+ * mc->reg != mc->rreg means that the control is
+ * stereo (bits in one register or in two registers)
+ */
+ return 1;
+}
+
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
diff --git a/include/sound/sta32x.h b/include/sound/sta32x.h
new file mode 100644
index 00000000000..8d93b0357a1
--- /dev/null
+++ b/include/sound/sta32x.h
@@ -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 */
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
index 89beccb57ed..4cc1093844c 100644
--- a/include/sound/tpa6130a2-plat.h
+++ b/include/sound/tpa6130a2-plat.h
@@ -23,13 +23,7 @@
#ifndef TPA6130A2_PLAT_H
#define TPA6130A2_PLAT_H
-enum tpa_model {
- TPA6130A2,
- TPA6140A2,
-};
-
struct tpa6130a2_platform_data {
- enum tpa_model id;
int power_gpio;
};
diff --git a/include/sound/wm1250-ev1.h b/include/sound/wm1250-ev1.h
new file mode 100644
index 00000000000..7dff8283412
--- /dev/null
+++ b/include/sound/wm1250-ev1.h
@@ -0,0 +1,27 @@
+/*
+ * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1
+ *
+ * Copyright 2011 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_WM1250_EV1_H
+#define __LINUX_SND_WM1250_EV1_H
+
+#define WM1250_EV1_NUM_GPIOS 5
+
+#define WM1250_EV1_GPIO_CLK_ENA 0
+#define WM1250_EV1_GPIO_CLK_SEL0 1
+#define WM1250_EV1_GPIO_CLK_SEL1 2
+#define WM1250_EV1_GPIO_OSR 3
+#define WM1250_EV1_GPIO_MASTER 4
+
+
+struct wm1250_ev1_pdata {
+ int gpios[WM1250_EV1_NUM_GPIOS];
+};
+
+#endif
diff --git a/include/sound/wm5100.h b/include/sound/wm5100.h
new file mode 100644
index 00000000000..617d0c4a159
--- /dev/null
+++ b/include/sound/wm5100.h
@@ -0,0 +1,59 @@
+/*
+ * linux/sound/wm5100.h -- Platform data for WM5100
+ *
+ * Copyright 2011 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_WM5100_H
+#define __LINUX_SND_WM5100_H
+
+enum wm5100_in_mode {
+ WM5100_IN_SE = 0,
+ WM5100_IN_DIFF = 1,
+ WM5100_IN_DMIC = 2,
+};
+
+enum wm5100_dmic_sup {
+ WM5100_DMIC_SUP_MICVDD = 0,
+ WM5100_DMIC_SUP_MICBIAS1 = 1,
+ WM5100_DMIC_SUP_MICBIAS2 = 2,
+ WM5100_DMIC_SUP_MICBIAS3 = 3,
+};
+
+enum wm5100_micdet_bias {
+ WM5100_MICDET_MICBIAS1 = 0,
+ WM5100_MICDET_MICBIAS2 = 1,
+ WM5100_MICDET_MICBIAS3 = 2,
+};
+
+struct wm5100_jack_mode {
+ enum wm5100_micdet_bias bias;
+ int hp_pol;
+ int micd_src;
+};
+
+#define WM5100_GPIO_SET 0x10000
+
+struct wm5100_pdata {
+ int reset; /** GPIO controlling /RESET, if any */
+ int ldo_ena; /** GPIO controlling LODENA, if any */
+ int hp_pol; /** GPIO controlling headset polarity, if any */
+ int irq_flags;
+ int gpio_base;
+
+ struct wm5100_jack_mode jack_modes[2];
+
+ /* Input pin mode selection */
+ enum wm5100_in_mode in_mode[4];
+
+ /* DMIC supply selection */
+ enum wm5100_dmic_sup dmic_sup[4];
+
+ int gpio_defaults[6];
+};
+
+#endif
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index cf7ccb76a8d..b310c5a3a95 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.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
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 22:35:32 +0000