ASoC: mid-x86 - add support for compressed streams

Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This commit is contained in:
Vinod Koul 2012-08-16 17:10:42 +05:30 committed by Mark Brown
parent 1245b7005d
commit c514a9119a
4 changed files with 371 additions and 2 deletions

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@ -318,6 +318,15 @@ static struct snd_soc_dai_link mfld_msic_dailink[] = {
.platform_name = "sst-platform",
.init = NULL,
},
{
.name = "Medfield Compress",
.stream_name = "Speaker",
.cpu_dai_name = "Compress-cpu-dai",
.codec_dai_name = "SN95031 Speaker",
.codec_name = "sn95031",
.platform_name = "sst-platform",
.init = NULL,
},
};
/* SoC card */

134
sound/soc/mid-x86/sst_dsp.h Normal file
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@ -0,0 +1,134 @@
#ifndef __SST_DSP_H__
#define __SST_DSP_H__
/*
* sst_dsp.h - Intel SST Driver for audio engine
*
* Copyright (C) 2008-12 Intel Corporation
* Authors: 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.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
enum sst_codec_types {
/* AUDIO/MUSIC CODEC Type Definitions */
SST_CODEC_TYPE_UNKNOWN = 0,
SST_CODEC_TYPE_PCM, /* Pass through Audio codec */
SST_CODEC_TYPE_MP3,
SST_CODEC_TYPE_MP24,
SST_CODEC_TYPE_AAC,
SST_CODEC_TYPE_AACP,
SST_CODEC_TYPE_eAACP,
};
enum stream_type {
SST_STREAM_TYPE_NONE = 0,
SST_STREAM_TYPE_MUSIC = 1,
};
struct snd_pcm_params {
u16 codec; /* codec type */
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u32 reserved; /* Bitrate in bits per second */
u32 sfreq; /* Sampling rate in Hz */
u8 use_offload_path;
u8 reserved2;
u16 reserved3;
u8 channel_map[8];
} __packed;
/* MP3 Music Parameters Message */
struct snd_mp3_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u8 crc_check; /* crc_check - disable (0) or enable (1) */
u8 reserved1; /* unused*/
u16 reserved2; /* Unused */
} __packed;
#define AAC_BIT_STREAM_ADTS 0
#define AAC_BIT_STREAM_ADIF 1
#define AAC_BIT_STREAM_RAW 2
/* AAC Music Parameters Message */
struct snd_aac_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo*/
u8 pcm_wd_sz; /* 16/24 - bit*/
u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
u16 reser2;
u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
u8 reser1;
u16 reser3;
} __packed;
/* WMA Music Parameters Message */
struct snd_wma_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u32 brate; /* Use the hard coded value. */
u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
u32 channel_mask; /* Channel Mask */
u16 format_tag; /* Format Tag */
u16 block_align; /* packet size */
u16 wma_encode_opt;/* Encoder option */
u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */
u8 reserved; /* reserved */
} __packed;
/* Codec params struture */
union snd_sst_codec_params {
struct snd_pcm_params pcm_params;
struct snd_mp3_params mp3_params;
struct snd_aac_params aac_params;
struct snd_wma_params wma_params;
} __packed;
/* Address and size info of a frame buffer */
struct sst_address_info {
u32 addr; /* Address at IA */
u32 size; /* Size of the buffer */
};
struct snd_sst_alloc_params_ext {
struct sst_address_info ring_buf_info[8];
u8 sg_count;
u8 reserved;
u16 reserved2;
u32 frag_size; /*Number of samples after which period elapsed
message is sent valid only if path = 0*/
} __packed;
struct snd_sst_stream_params {
union snd_sst_codec_params uc;
} __packed;
struct snd_sst_params {
u32 stream_id;
u8 codec;
u8 ops;
u8 stream_type;
u8 device_type;
struct snd_sst_stream_params sparams;
struct snd_sst_alloc_params_ext aparams;
};
#endif /* __SST_DSP_H__ */

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@ -1,7 +1,7 @@
/*
* sst_platform.c - Intel MID Platform driver
*
* Copyright (C) 2010 Intel Corp
* Copyright (C) 2010-2012 Intel Corp
* Author: Vinod Koul <vinod.koul@intel.com>
* Author: Harsha Priya <priya.harsha@intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
@ -32,6 +32,7 @@
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/compress_driver.h>
#include "sst_platform.h"
static struct sst_device *sst;
@ -152,6 +153,16 @@ static struct snd_soc_dai_driver sst_platform_dai[] = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
},
},
{
.name = "Compress-cpu-dai",
.compress_dai = 1,
.playback = {
.channels_min = SST_STEREO,
.channels_max = SST_STEREO,
.rates = SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
},
};
/* helper functions */
@ -463,8 +474,199 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd)
}
return retval;
}
/* compress stream operations */
static void sst_compr_fragment_elapsed(void *arg)
{
struct snd_compr_stream *cstream = (struct snd_compr_stream *)arg;
pr_debug("fragment elapsed by driver\n");
if (cstream)
snd_compr_fragment_elapsed(cstream);
}
static int sst_platform_compr_open(struct snd_compr_stream *cstream)
{
int ret_val = 0;
struct snd_compr_runtime *runtime = cstream->runtime;
struct sst_runtime_stream *stream;
stream = kzalloc(sizeof(*stream), GFP_KERNEL);
if (!stream)
return -ENOMEM;
spin_lock_init(&stream->status_lock);
/* get the sst ops */
if (!sst || !try_module_get(sst->dev->driver->owner)) {
pr_err("no device available to run\n");
ret_val = -ENODEV;
goto out_ops;
}
stream->compr_ops = sst->compr_ops;
stream->id = 0;
sst_set_stream_status(stream, SST_PLATFORM_INIT);
runtime->private_data = stream;
return 0;
out_ops:
kfree(stream);
return ret_val;
}
static int sst_platform_compr_free(struct snd_compr_stream *cstream)
{
struct sst_runtime_stream *stream;
int ret_val = 0, str_id;
stream = cstream->runtime->private_data;
/*need to check*/
str_id = stream->id;
if (str_id)
ret_val = stream->compr_ops->close(str_id);
module_put(sst->dev->driver->owner);
kfree(stream);
pr_debug("%s: %d\n", __func__, ret_val);
return 0;
}
static int sst_platform_compr_set_params(struct snd_compr_stream *cstream,
struct snd_compr_params *params)
{
struct sst_runtime_stream *stream;
int retval;
struct snd_sst_params str_params;
struct sst_compress_cb cb;
stream = cstream->runtime->private_data;
/* construct fw structure for this*/
memset(&str_params, 0, sizeof(str_params));
str_params.ops = STREAM_OPS_PLAYBACK;
str_params.stream_type = SST_STREAM_TYPE_MUSIC;
str_params.device_type = SND_SST_DEVICE_COMPRESS;
switch (params->codec.id) {
case SND_AUDIOCODEC_MP3: {
str_params.codec = SST_CODEC_TYPE_MP3;
str_params.sparams.uc.mp3_params.codec = SST_CODEC_TYPE_MP3;
str_params.sparams.uc.mp3_params.num_chan = params->codec.ch_in;
str_params.sparams.uc.mp3_params.pcm_wd_sz = 16;
break;
}
case SND_AUDIOCODEC_AAC: {
str_params.codec = SST_CODEC_TYPE_AAC;
str_params.sparams.uc.aac_params.codec = SST_CODEC_TYPE_AAC;
str_params.sparams.uc.aac_params.num_chan = params->codec.ch_in;
str_params.sparams.uc.aac_params.pcm_wd_sz = 16;
if (params->codec.format == SND_AUDIOSTREAMFORMAT_MP4ADTS)
str_params.sparams.uc.aac_params.bs_format =
AAC_BIT_STREAM_ADTS;
else if (params->codec.format == SND_AUDIOSTREAMFORMAT_RAW)
str_params.sparams.uc.aac_params.bs_format =
AAC_BIT_STREAM_RAW;
else {
pr_err("Undefined format%d\n", params->codec.format);
return -EINVAL;
}
str_params.sparams.uc.aac_params.externalsr =
params->codec.sample_rate;
break;
}
default:
pr_err("codec not supported, id =%d\n", params->codec.id);
return -EINVAL;
}
str_params.aparams.ring_buf_info[0].addr =
virt_to_phys(cstream->runtime->buffer);
str_params.aparams.ring_buf_info[0].size =
cstream->runtime->buffer_size;
str_params.aparams.sg_count = 1;
str_params.aparams.frag_size = cstream->runtime->fragment_size;
cb.param = cstream;
cb.compr_cb = sst_compr_fragment_elapsed;
retval = stream->compr_ops->open(&str_params, &cb);
if (retval < 0) {
pr_err("stream allocation failed %d\n", retval);
return retval;
}
stream->id = retval;
return 0;
}
static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd)
{
struct sst_runtime_stream *stream =
cstream->runtime->private_data;
return stream->compr_ops->control(cmd, stream->id);
}
static int sst_platform_compr_pointer(struct snd_compr_stream *cstream,
struct snd_compr_tstamp *tstamp)
{
struct sst_runtime_stream *stream;
stream = cstream->runtime->private_data;
stream->compr_ops->tstamp(stream->id, tstamp);
tstamp->byte_offset = tstamp->copied_total %
(u32)cstream->runtime->buffer_size;
pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset);
return 0;
}
static int sst_platform_compr_ack(struct snd_compr_stream *cstream,
size_t bytes)
{
struct sst_runtime_stream *stream;
stream = cstream->runtime->private_data;
stream->compr_ops->ack(stream->id, (unsigned long)bytes);
stream->bytes_written += bytes;
return 0;
}
static int sst_platform_compr_get_caps(struct snd_compr_stream *cstream,
struct snd_compr_caps *caps)
{
struct sst_runtime_stream *stream =
cstream->runtime->private_data;
return stream->compr_ops->get_caps(caps);
}
static int sst_platform_compr_get_codec_caps(struct snd_compr_stream *cstream,
struct snd_compr_codec_caps *codec)
{
struct sst_runtime_stream *stream =
cstream->runtime->private_data;
return stream->compr_ops->get_codec_caps(codec);
}
static struct snd_compr_ops sst_platform_compr_ops = {
.open = sst_platform_compr_open,
.free = sst_platform_compr_free,
.set_params = sst_platform_compr_set_params,
.trigger = sst_platform_compr_trigger,
.pointer = sst_platform_compr_pointer,
.ack = sst_platform_compr_ack,
.get_caps = sst_platform_compr_get_caps,
.get_codec_caps = sst_platform_compr_get_codec_caps,
};
static struct snd_soc_platform_driver sst_soc_platform_drv = {
.ops = &sst_platform_ops,
.compr_ops = &sst_platform_compr_ops,
.pcm_new = sst_pcm_new,
.pcm_free = sst_pcm_free,
};

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@ -27,6 +27,8 @@
#ifndef __SST_PLATFORMDRV_H__
#define __SST_PLATFORMDRV_H__
#include "sst_dsp.h"
#define SST_MONO 1
#define SST_STEREO 2
#define SST_MAX_CAP 5
@ -42,7 +44,6 @@
#define SST_MIN_PERIODS 2
#define SST_MAX_PERIODS (1024*2)
#define SST_FIFO_SIZE 0
#define SST_CODEC_TYPE_PCM 1
struct pcm_stream_info {
int str_id;
@ -83,6 +84,7 @@ enum sst_audio_device_type {
SND_SST_DEVICE_VIBRA,
SND_SST_DEVICE_HAPTIC,
SND_SST_DEVICE_CAPTURE,
SND_SST_DEVICE_COMPRESS,
};
/* PCM Parameters */
@ -107,6 +109,24 @@ struct sst_stream_params {
struct sst_pcm_params sparams;
};
struct sst_compress_cb {
void *param;
void (*compr_cb)(void *param);
};
struct compress_sst_ops {
const char *name;
int (*open) (struct snd_sst_params *str_params,
struct sst_compress_cb *cb);
int (*control) (unsigned int cmd, unsigned int str_id);
int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp);
int (*ack) (unsigned int str_id, unsigned long bytes);
int (*close) (unsigned int str_id);
int (*get_caps) (struct snd_compr_caps *caps);
int (*get_codec_caps) (struct snd_compr_codec_caps *codec);
};
struct sst_ops {
int (*open) (struct sst_stream_params *str_param);
int (*device_control) (int cmd, void *arg);
@ -115,8 +135,11 @@ struct sst_ops {
struct sst_runtime_stream {
int stream_status;
unsigned int id;
size_t bytes_written;
struct pcm_stream_info stream_info;
struct sst_ops *ops;
struct compress_sst_ops *compr_ops;
spinlock_t status_lock;
};
@ -124,6 +147,7 @@ struct sst_device {
char *name;
struct device *dev;
struct sst_ops *ops;
struct compress_sst_ops *compr_ops;
};
int sst_register_dsp(struct sst_device *sst);