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android_kernel_samsung_msm8226/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c

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/* Copyright (c) 2012-2013, The Linux Foundation. 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 and
* only version 2 as published by the Free Software Foundation.
*
* 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.
*/
#include <linux/init.h>
#include <linux/err.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/time.h>
#include <linux/wait.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
#include <sound/initval.h>
#include <sound/control.h>
#include <sound/q6asm-v2.h>
#include <sound/pcm_params.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/msm_audio_ion.h>
#include <sound/timer.h>
#include "msm-compr-q6-v2.h"
#include "msm-pcm-routing-v2.h"
#include "audio_ocmem.h"
#include <sound/tlv.h>
#define COMPRE_CAPTURE_NUM_PERIODS 16
/* Allocate the worst case frame size for compressed audio */
#define COMPRE_CAPTURE_HEADER_SIZE (sizeof(struct snd_compr_audio_info))
/* Changing period size to 4032. 4032 will make sure COMPRE_CAPTURE_PERIOD_SIZE
* is 4096 with meta data size of 64 and MAX_NUM_FRAMES_PER_BUFFER 1
*/
#define COMPRE_CAPTURE_MAX_FRAME_SIZE (4032)
#define COMPRE_CAPTURE_PERIOD_SIZE ((COMPRE_CAPTURE_MAX_FRAME_SIZE + \
COMPRE_CAPTURE_HEADER_SIZE) * \
MAX_NUM_FRAMES_PER_BUFFER)
#define COMPRE_OUTPUT_METADATA_SIZE (sizeof(struct output_meta_data_st))
#define COMPRESSED_LR_VOL_MAX_STEPS 0x20002000
#define MAX_AC3_PARAM_SIZE (18*2*sizeof(int))
#define AMR_WB_BAND_MODE 8
#define AMR_WB_DTX_MODE 0
const DECLARE_TLV_DB_LINEAR(compr_rx_vol_gain, 0,
COMPRESSED_LR_VOL_MAX_STEPS);
struct snd_msm {
atomic_t audio_ocmem_req;
};
static struct snd_msm compressed_audio;
static struct audio_locks the_locks;
static struct snd_pcm_hardware msm_compr_hardware_capture = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 8,
.buffer_bytes_max =
COMPRE_CAPTURE_PERIOD_SIZE * COMPRE_CAPTURE_NUM_PERIODS ,
.period_bytes_min = COMPRE_CAPTURE_PERIOD_SIZE,
.period_bytes_max = COMPRE_CAPTURE_PERIOD_SIZE,
.periods_min = COMPRE_CAPTURE_NUM_PERIODS,
.periods_max = COMPRE_CAPTURE_NUM_PERIODS,
.fifo_size = 0,
};
static struct snd_pcm_hardware msm_compr_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 8,
.buffer_bytes_max = 1024 * 1024,
.period_bytes_min = 128 * 1024,
.period_bytes_max = 256 * 1024,
.periods_min = 4,
.periods_max = 8,
.fifo_size = 0,
};
/* Conventional and unconventional sample rate supported */
static unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
};
/* Add supported codecs for compress capture path */
static uint32_t supported_compr_capture_codecs[] = {
SND_AUDIOCODEC_AMRWB
};
static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.count = ARRAY_SIZE(supported_sample_rates),
.list = supported_sample_rates,
.mask = 0,
};
static bool msm_compr_capture_codecs(uint32_t req_codec)
{
int i;
pr_debug("%s req_codec:%d\n", __func__, req_codec);
if (req_codec == 0)
return false;
for (i = 0; i < ARRAY_SIZE(supported_compr_capture_codecs); i++) {
if (req_codec == supported_compr_capture_codecs[i])
return true;
}
return false;
}
static void compr_event_handler(uint32_t opcode,
uint32_t token, uint32_t *payload, void *priv)
{
struct compr_audio *compr = priv;
struct msm_audio *prtd = &compr->prtd;
struct snd_pcm_substream *substream = prtd->substream;
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_aio_write_param param;
struct audio_aio_read_param read_param;
struct audio_buffer *buf = NULL;
struct output_meta_data_st output_meta_data;
uint32_t *ptrmem = (uint32_t *)payload;
int i = 0;
int time_stamp_flag = 0;
int buffer_length = 0;
int stop_playback = 0;
pr_debug("%s opcode =%08x\n", __func__, opcode);
switch (opcode) {
case ASM_DATA_EVENT_WRITE_DONE_V2: {
uint32_t *ptrmem = (uint32_t *)&param;
pr_debug("ASM_DATA_EVENT_WRITE_DONE\n");
pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
prtd->pcm_irq_pos += prtd->pcm_count;
if (atomic_read(&prtd->start))
snd_pcm_period_elapsed(substream);
else
if (substream->timer_running)
snd_timer_interrupt(substream->timer, 1);
atomic_inc(&prtd->out_count);
wake_up(&the_locks.write_wait);
if (!atomic_read(&prtd->start)) {
atomic_set(&prtd->pending_buffer, 1);
break;
} else
atomic_set(&prtd->pending_buffer, 0);
/*
* check for underrun
*/
snd_pcm_stream_lock_irq(substream);
if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
runtime->render_flag |= SNDRV_RENDER_STOPPED;
stop_playback = 1;
}
snd_pcm_stream_unlock_irq(substream);
if (stop_playback) {
pr_err("underrun! render stopped\n");
break;
}
buf = prtd->audio_client->port[IN].buf;
pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
__func__, prtd->pcm_count, prtd->out_head);
pr_debug("%s:writing buffer[%d] from 0x%08x\n",
__func__, prtd->out_head,
((unsigned int)buf[0].phys
+ (prtd->out_head * prtd->pcm_count)));
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
time_stamp_flag = SET_TIMESTAMP;
else
time_stamp_flag = NO_TIMESTAMP;
memcpy(&output_meta_data, (char *)(buf->data +
prtd->out_head * prtd->pcm_count),
COMPRE_OUTPUT_METADATA_SIZE);
buffer_length = output_meta_data.frame_size;
pr_debug("meta_data_length: %d, frame_length: %d\n",
output_meta_data.meta_data_length,
output_meta_data.frame_size);
pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
output_meta_data.timestamp_msw,
output_meta_data.timestamp_lsw);
if (buffer_length == 0) {
pr_debug("Recieved a zero length buffer-break out");
break;
}
param.paddr = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count)
+ output_meta_data.meta_data_length;
param.len = buffer_length;
param.msw_ts = output_meta_data.timestamp_msw;
param.lsw_ts = output_meta_data.timestamp_lsw;
param.flags = time_stamp_flag;
param.uid = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count
+ output_meta_data.meta_data_length);
for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
i++, ++ptrmem)
pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
if (q6asm_async_write(prtd->audio_client,
&param) < 0)
pr_err("%s:q6asm_async_write failed\n",
__func__);
else
prtd->out_head =
(prtd->out_head + 1) & (runtime->periods - 1);
break;
}
case ASM_DATA_EVENT_RENDERED_EOS:
pr_debug("ASM_DATA_CMDRSP_EOS\n");
if (atomic_read(&prtd->eos)) {
pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.eos_wait);
atomic_set(&prtd->eos, 0);
}
break;
case ASM_DATA_EVENT_READ_DONE_V2: {
pr_debug("ASM_DATA_EVENT_READ_DONE\n");
pr_debug("buf = %p, data = 0x%X, *data = %p,\n"
"prtd->pcm_irq_pos = %d\n",
prtd->audio_client->port[OUT].buf,
*(uint32_t *)prtd->audio_client->port[OUT].buf->data,
prtd->audio_client->port[OUT].buf->data,
prtd->pcm_irq_pos);
memcpy(prtd->audio_client->port[OUT].buf->data +
prtd->pcm_irq_pos, (ptrmem + READDONE_IDX_SIZE),
COMPRE_CAPTURE_HEADER_SIZE);
pr_debug("buf = %p, updated data = 0x%X, *data = %p\n",
prtd->audio_client->port[OUT].buf,
*(uint32_t *)(prtd->audio_client->port[OUT].buf->data +
prtd->pcm_irq_pos),
prtd->audio_client->port[OUT].buf->data);
if (!atomic_read(&prtd->start))
break;
pr_debug("frame size=%d, buffer = 0x%X\n",
ptrmem[READDONE_IDX_SIZE],
ptrmem[READDONE_IDX_BUFADD_LSW]);
if (ptrmem[READDONE_IDX_SIZE] > COMPRE_CAPTURE_MAX_FRAME_SIZE) {
pr_err("Frame length exceeded the max length");
break;
}
buf = prtd->audio_client->port[OUT].buf;
pr_debug("pcm_irq_pos=%d, buf[0].phys = 0x%X\n",
prtd->pcm_irq_pos, (uint32_t)buf[0].phys);
read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
read_param.paddr = (unsigned long)(buf[0].phys) +
prtd->pcm_irq_pos + COMPRE_CAPTURE_HEADER_SIZE;
prtd->pcm_irq_pos += prtd->pcm_count;
if (atomic_read(&prtd->start))
snd_pcm_period_elapsed(substream);
q6asm_async_read(prtd->audio_client, &read_param);
break;
}
case APR_BASIC_RSP_RESULT: {
switch (payload[0]) {
case ASM_SESSION_CMD_RUN_V2: {
if (substream->stream
!= SNDRV_PCM_STREAM_PLAYBACK) {
atomic_set(&prtd->start, 1);
break;
}
if (!atomic_read(&prtd->pending_buffer))
break;
pr_debug("%s:writing %d bytes of buffer[%d] to dsp\n",
__func__, prtd->pcm_count, prtd->out_head);
buf = prtd->audio_client->port[IN].buf;
pr_debug("%s:writing buffer[%d] from 0x%08x\n",
__func__, prtd->out_head,
((unsigned int)buf[0].phys
+ (prtd->out_head * prtd->pcm_count)));
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
time_stamp_flag = SET_TIMESTAMP;
else
time_stamp_flag = NO_TIMESTAMP;
memcpy(&output_meta_data, (char *)(buf->data +
prtd->out_head * prtd->pcm_count),
COMPRE_OUTPUT_METADATA_SIZE);
buffer_length = output_meta_data.frame_size;
pr_debug("meta_data_length: %d, frame_length: %d\n",
output_meta_data.meta_data_length,
output_meta_data.frame_size);
pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
output_meta_data.timestamp_msw,
output_meta_data.timestamp_lsw);
param.paddr = (unsigned long)buf[prtd->out_head].phys
+ output_meta_data.meta_data_length;
param.len = buffer_length;
param.msw_ts = output_meta_data.timestamp_msw;
param.lsw_ts = output_meta_data.timestamp_lsw;
param.flags = time_stamp_flag;
param.uid = (unsigned long)buf[prtd->out_head].phys
+ output_meta_data.meta_data_length;
param.metadata_len = COMPRE_OUTPUT_METADATA_SIZE;
if (q6asm_async_write(prtd->audio_client,
&param) < 0)
pr_err("%s:q6asm_async_write failed\n",
__func__);
else
prtd->out_head =
(prtd->out_head + 1)
& (runtime->periods - 1);
atomic_set(&prtd->pending_buffer, 0);
}
break;
case ASM_STREAM_CMD_FLUSH:
pr_debug("ASM_STREAM_CMD_FLUSH\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.flush_wait);
break;
default:
break;
}
break;
}
default:
pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
break;
}
}
static int msm_compr_send_ddp_cfg(struct audio_client *ac,
struct snd_dec_ddp *ddp)
{
int i, rc;
pr_debug("%s\n", __func__);
for (i = 0; i < ddp->params_length/2; i++) {
rc = q6asm_ds1_set_endp_params(ac, ddp->params_id[i],
ddp->params_value[i]);
if (rc) {
pr_err("sending params_id: %d failed\n",
ddp->params_id[i]);
return rc;
}
}
return 0;
}
static int msm_compr_playback_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
struct asm_aac_cfg aac_cfg;
int ret;
pr_debug("compressed stream prepare\n");
prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
prtd->pcm_irq_pos = 0;
/* rate and channels are sent to audio driver */
prtd->samp_rate = runtime->rate;
prtd->channel_mode = runtime->channels;
prtd->out_head = 0;
atomic_set(&prtd->out_count, runtime->periods);
if (prtd->enabled)
return 0;
switch (compr->info.codec_param.codec.id) {
case SND_AUDIOCODEC_MP3:
/* No media format block for mp3 */
break;
case SND_AUDIOCODEC_AAC:
pr_debug("%s: SND_AUDIOCODEC_AAC\n", __func__);
memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
aac_cfg.format = 0x03;
aac_cfg.ch_cfg = runtime->channels;
aac_cfg.sample_rate = runtime->rate;
ret = q6asm_media_format_block_aac(prtd->audio_client,
&aac_cfg);
if (ret < 0)
pr_err("%s: CMD Format block failed\n", __func__);
break;
case SND_AUDIOCODEC_AC3: {
struct snd_dec_ddp *ddp =
&compr->info.codec_param.codec.options.ddp;
pr_debug("%s: SND_AUDIOCODEC_AC3\n", __func__);
ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
if (ret < 0)
pr_err("%s: DDP CMD CFG failed\n", __func__);
break;
}
case SND_AUDIOCODEC_EAC3: {
struct snd_dec_ddp *ddp =
&compr->info.codec_param.codec.options.ddp;
pr_debug("%s: SND_AUDIOCODEC_EAC3\n", __func__);
ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
if (ret < 0)
pr_err("%s: DDP CMD CFG failed\n", __func__);
break;
}
default:
return -EINVAL;
}
prtd->enabled = 1;
prtd->cmd_ack = 0;
prtd->cmd_interrupt = 0;
return 0;
}
static int msm_compr_capture_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
struct audio_buffer *buf = prtd->audio_client->port[OUT].buf;
struct snd_codec *codec = &compr->info.codec_param.codec;
struct audio_aio_read_param read_param;
int ret = 0;
int i;
prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
prtd->pcm_irq_pos = 0;
if (!msm_compr_capture_codecs(codec->id)) {
/*request codec invalid or not supported,
use default compress format*/
codec->id = SND_AUDIOCODEC_AMRWB;
}
/* rate and channels are sent to audio driver */
prtd->samp_rate = runtime->rate;
prtd->channel_mode = runtime->channels;
if (prtd->enabled)
return ret;
read_param.len = prtd->pcm_count;
switch (codec->id) {
case SND_AUDIOCODEC_AMRWB:
pr_debug("SND_AUDIOCODEC_AMRWB\n");
ret = q6asm_enc_cfg_blk_amrwb(prtd->audio_client,
MAX_NUM_FRAMES_PER_BUFFER,
/* use fixed band mode and dtx mode
* band mode - 23.85 kbps
*/
AMR_WB_BAND_MODE,
/* dtx mode - disable */
AMR_WB_DTX_MODE);
if (ret < 0)
pr_err("%s: CMD Format block" \
"failed: %d\n", __func__, ret);
break;
default:
pr_debug("No config for codec %d\n", codec->id);
}
pr_debug("%s: Samp_rate = %d, Channel = %d, pcm_size = %d,\n"
"pcm_count = %d, periods = %d\n",
__func__, prtd->samp_rate, prtd->channel_mode,
prtd->pcm_size, prtd->pcm_count, runtime->periods);
for (i = 0; i < runtime->periods; i++) {
read_param.uid = i;
switch (codec->id) {
case SND_AUDIOCODEC_AMRWB:
read_param.len = prtd->pcm_count
- COMPRE_CAPTURE_HEADER_SIZE;
read_param.paddr = (unsigned long)(buf[i].phys)
+ COMPRE_CAPTURE_HEADER_SIZE;
pr_debug("Push buffer [%d] to DSP, "\
"paddr: %p, vaddr: %p\n",
i, (void *) read_param.paddr,
buf[i].data);
q6asm_async_read(prtd->audio_client, &read_param);
break;
default:
read_param.paddr = (unsigned long)(buf[i].phys);
/*q6asm_async_read_compressed(prtd->audio_client,
&read_param);*/
pr_debug("%s: To add support for read compressed\n",
__func__);
ret = -EINVAL;
break;
}
}
prtd->periods = runtime->periods;
prtd->enabled = 1;
return ret;
}
static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
pr_debug("%s\n", __func__);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
prtd->pcm_irq_pos = 0;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
if (!msm_compr_capture_codecs(
compr->info.codec_param.codec.id)) {
/*request codec invalid or not supported,
use default compress format*/
compr->info.codec_param.codec.id =
SND_AUDIOCODEC_AMRWB;
}
switch (compr->info.codec_param.codec.id) {
case SND_AUDIOCODEC_AMRWB:
break;
default:
msm_pcm_routing_reg_psthr_stream(
soc_prtd->dai_link->be_id,
prtd->session_id, substream->stream);
break;
}
}
atomic_set(&prtd->pending_buffer, 1);
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
pr_debug("%s: Trigger start\n", __func__);
q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
atomic_set(&prtd->start, 1);
break;
case SNDRV_PCM_TRIGGER_STOP:
pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
switch (compr->info.codec_param.codec.id) {
case SND_AUDIOCODEC_AMRWB:
break;
default:
msm_pcm_routing_reg_psthr_stream(
soc_prtd->dai_link->be_id,
prtd->session_id, substream->stream);
break;
}
}
atomic_set(&prtd->start, 0);
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
atomic_set(&prtd->start, 0);
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static void populate_codec_list(struct compr_audio *compr,
struct snd_pcm_runtime *runtime)
{
pr_debug("%s\n", __func__);
/* MP3 Block */
compr->info.compr_cap.num_codecs = 5;
compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
compr->info.compr_cap.codecs[2] = SND_AUDIOCODEC_AC3;
compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3;
compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_AMRWB;
/* Add new codecs here */
}
static int msm_compr_open(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr;
struct msm_audio *prtd;
int ret = 0;
pr_debug("%s\n", __func__);
compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
if (compr == NULL) {
pr_err("Failed to allocate memory for msm_audio\n");
return -ENOMEM;
}
prtd = &compr->prtd;
prtd->substream = substream;
runtime->render_flag = SNDRV_DMA_MODE;
prtd->audio_client = q6asm_audio_client_alloc(
(app_cb)compr_event_handler, compr);
if (!prtd->audio_client) {
pr_info("%s: Could not allocate memory\n", __func__);
kfree(prtd);
return -ENOMEM;
}
prtd->audio_client->perf_mode = false;
pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
runtime->hw = msm_compr_hardware_playback;
prtd->cmd_ack = 1;
} else {
runtime->hw = msm_compr_hardware_capture;
}
ret = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_sample_rates);
if (ret < 0)
pr_info("snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
pr_info("snd_pcm_hw_constraint_integer failed\n");
prtd->dsp_cnt = 0;
atomic_set(&prtd->pending_buffer, 1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
compr->codec = FORMAT_MP3;
populate_codec_list(compr, runtime);
runtime->private_data = compr;
atomic_set(&prtd->eos, 0);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (!atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 0, 1))
audio_ocmem_process_req(AUDIO, true);
else
atomic_inc(&compressed_audio.audio_ocmem_req);
pr_debug("%s: req: %d\n", __func__,
atomic_read(&compressed_audio.audio_ocmem_req));
}
return 0;
}
static int compressed_set_volume(struct msm_audio *prtd, uint32_t volume)
{
int rc = 0;
int avg_vol = 0;
int lgain = (volume >> 16) & 0xFFFF;
int rgain = volume & 0xFFFF;
if (prtd && prtd->audio_client) {
pr_debug("%s: channels %d volume 0x%x\n", __func__,
prtd->channel_mode, volume);
if ((prtd->channel_mode == 2) &&
(lgain != rgain)) {
pr_debug("%s: call q6asm_set_lrgain\n", __func__);
rc = q6asm_set_lrgain(prtd->audio_client, lgain, rgain);
} else {
avg_vol = (lgain + rgain)/2;
pr_debug("%s: call q6asm_set_volume\n", __func__);
rc = q6asm_set_volume(prtd->audio_client, avg_vol);
}
if (rc < 0) {
pr_err("%s: Send Volume command failed rc=%d\n",
__func__, rc);
}
}
return rc;
}
static int msm_compr_playback_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
int dir = 0;
pr_debug("%s\n", __func__);
dir = IN;
atomic_set(&prtd->pending_buffer, 0);
if (atomic_read(&compressed_audio.audio_ocmem_req) > 1)
atomic_dec(&compressed_audio.audio_ocmem_req);
else if (atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 1, 0))
audio_ocmem_process_req(AUDIO, false);
pr_debug("%s: req: %d\n", __func__,
atomic_read(&compressed_audio.audio_ocmem_req));
prtd->pcm_irq_pos = 0;
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
q6asm_audio_client_buf_free_contiguous(dir,
prtd->audio_client);
msm_pcm_routing_dereg_phy_stream(
soc_prtd->dai_link->be_id,
SNDRV_PCM_STREAM_PLAYBACK);
q6asm_audio_client_free(prtd->audio_client);
kfree(prtd);
return 0;
}
static int msm_compr_capture_close(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
int dir = OUT;
pr_debug("%s\n", __func__);
atomic_set(&prtd->pending_buffer, 0);
q6asm_cmd(prtd->audio_client, CMD_CLOSE);
q6asm_audio_client_buf_free_contiguous(dir,
prtd->audio_client);
msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
SNDRV_PCM_STREAM_CAPTURE);
q6asm_audio_client_free(prtd->audio_client);
kfree(prtd);
return 0;
}
static int msm_compr_close(struct snd_pcm_substream *substream)
{
int ret = 0;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ret = msm_compr_playback_close(substream);
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
ret = msm_compr_capture_close(substream);
return ret;
}
static int msm_compr_prepare(struct snd_pcm_substream *substream)
{
int ret = 0;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
ret = msm_compr_playback_prepare(substream);
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
ret = msm_compr_capture_prepare(substream);
return ret;
}
static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
if (prtd->pcm_irq_pos >= prtd->pcm_size)
prtd->pcm_irq_pos = 0;
pr_debug("%s: pcm_irq_pos = %d, pcm_size = %d, sample_bits = %d,\n"
"frame_bits = %d\n", __func__, prtd->pcm_irq_pos,
prtd->pcm_size, runtime->sample_bits,
runtime->frame_bits);
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
static int msm_compr_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct msm_audio *prtd = runtime->private_data;
struct audio_client *ac = prtd->audio_client;
struct audio_port_data *apd = ac->port;
struct audio_buffer *ab;
int dir = -1;
prtd->mmap_flag = 1;
runtime->render_flag = SNDRV_NON_DMA_MODE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dir = IN;
else
dir = OUT;
ab = &(apd[dir].buf[0]);
return msm_audio_ion_mmap(ab, vma);
}
static int msm_compr_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
struct audio_buffer *buf;
int dir, ret;
uint16_t bits_per_sample = 16;
struct asm_softpause_params softpause = {
.enable = SOFT_PAUSE_ENABLE,
.period = SOFT_PAUSE_PERIOD,
.step = SOFT_PAUSE_STEP,
.rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
};
struct asm_softvolume_params softvol = {
.period = SOFT_VOLUME_PERIOD,
.step = SOFT_VOLUME_STEP,
.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
};
pr_debug("%s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dir = IN;
else
dir = OUT;
if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
bits_per_sample = 24;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
ret = q6asm_open_write_v2(prtd->audio_client,
compr->codec, bits_per_sample);
if (ret < 0) {
pr_err("%s: Session out open failed\n",
__func__);
return -ENOMEM;
}
msm_pcm_routing_reg_phy_stream(
soc_prtd->dai_link->be_id,
prtd->audio_client->perf_mode,
prtd->session_id,
substream->stream);
/* the number of channels are required to call volume api
accoridngly. So, get channels from hw params */
if ((params_channels(params) > 0) &&
(params_periods(params) <= runtime->hw.channels_max))
prtd->channel_mode = params_channels(params);
ret = compressed_set_volume(prtd, 0);
if (ret < 0)
pr_err("%s : Set Volume failed : %d", __func__, ret);
ret = q6asm_set_softpause(prtd->audio_client, &softpause);
if (ret < 0)
pr_err("%s: Send SoftPause Param failed ret=%d\n",
__func__, ret);
ret = q6asm_set_softvolume(prtd->audio_client, &softvol);
if (ret < 0)
pr_err("%s: Send SoftVolume Param failed ret=%d\n",
__func__, ret);
} else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
if (!msm_compr_capture_codecs(
compr->info.codec_param.codec.id)) {
/*request codec invalid or not supported,
use default compress format*/
compr->info.codec_param.codec.id =
SND_AUDIOCODEC_AMRWB;
}
switch (compr->info.codec_param.codec.id) {
case SND_AUDIOCODEC_AMRWB:
pr_debug("q6asm_open_read(FORMAT_AMRWB)\n");
ret = q6asm_open_read(prtd->audio_client,
FORMAT_AMRWB);
if (ret < 0) {
pr_err("%s: compressed Session out open failed\n",
__func__);
return -ENOMEM;
}
pr_debug("msm_pcm_routing_reg_phy_stream\n");
msm_pcm_routing_reg_phy_stream(
soc_prtd->dai_link->be_id,
prtd->audio_client->perf_mode,
prtd->session_id, substream->stream);
break;
default:
pr_debug("q6asm_open_read_compressed(COMPRESSED_META_DATA_MODE)\n");
/*
ret = q6asm_open_read_compressed(prtd->audio_client,
MAX_NUM_FRAMES_PER_BUFFER,
COMPRESSED_META_DATA_MODE);
*/
ret = -EINVAL;
break;
}
if (ret < 0) {
pr_err("%s: compressed Session out open failed\n",
__func__);
return -ENOMEM;
}
}
ret = q6asm_set_io_mode(prtd->audio_client,
(COMPRESSED_IO | ASYNC_IO_MODE));
if (ret < 0) {
pr_err("%s: Set IO mode failed\n", __func__);
return -ENOMEM;
}
/* Modifying kernel hardware params based on userspace config */
if (params_periods(params) > 0 &&
(params_periods(params) != runtime->hw.periods_max)) {
runtime->hw.periods_max = params_periods(params);
}
if (params_period_bytes(params) > 0 &&
(params_period_bytes(params) != runtime->hw.period_bytes_min)) {
runtime->hw.period_bytes_min = params_period_bytes(params);
}
runtime->hw.buffer_bytes_max =
runtime->hw.period_bytes_min * runtime->hw.periods_max;
pr_debug("allocate %d buffers each of size %d\n",
runtime->hw.period_bytes_min,
runtime->hw.periods_max);
ret = q6asm_audio_client_buf_alloc_contiguous(dir,
prtd->audio_client,
runtime->hw.period_bytes_min,
runtime->hw.periods_max);
if (ret < 0) {
pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
ret);
return -ENOMEM;
}
buf = prtd->audio_client->port[dir].buf;
dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
dma_buf->dev.dev = substream->pcm->card->dev;
dma_buf->private_data = NULL;
dma_buf->area = buf[0].data;
dma_buf->addr = buf[0].phys;
dma_buf->bytes = runtime->hw.buffer_bytes_max;
pr_debug("%s: buf[%p]dma_buf->area[%p]dma_buf->addr[%p]\n"
"dma_buf->bytes[%d]\n", __func__,
(void *)buf, (void *)dma_buf->area,
(void *)dma_buf->addr, dma_buf->bytes);
if (!dma_buf->area)
return -ENOMEM;
snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
}
static int msm_compr_ioctl(struct snd_pcm_substream *substream,
unsigned int cmd, void *arg)
{
int rc = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
uint64_t timestamp;
uint64_t temp;
switch (cmd) {
case SNDRV_COMPRESS_TSTAMP: {
struct snd_compr_tstamp tstamp;
pr_debug("SNDRV_COMPRESS_TSTAMP\n");
memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
rc = q6asm_get_session_time(prtd->audio_client, &timestamp);
if (rc < 0) {
pr_err("%s: Get Session Time return value =%lld\n",
__func__, timestamp);
return -EAGAIN;
}
temp = (timestamp * 2 * runtime->channels);
temp = temp * (runtime->rate/1000);
temp = div_u64(temp, 1000);
tstamp.sampling_rate = runtime->rate;
tstamp.timestamp = timestamp;
pr_debug("%s: bytes_consumed:,timestamp = %lld,\n",
__func__,
tstamp.timestamp);
if (copy_to_user((void *) arg, &tstamp,
sizeof(struct snd_compr_tstamp)))
return -EFAULT;
return 0;
}
case SNDRV_COMPRESS_GET_CAPS:
pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
if (copy_to_user((void *) arg, &compr->info.compr_cap,
sizeof(struct snd_compr_caps))) {
rc = -EFAULT;
pr_err("%s: ERROR: copy to user\n", __func__);
return rc;
}
return 0;
case SNDRV_COMPRESS_SET_PARAMS:
pr_debug("SNDRV_COMPRESS_SET_PARAMS:\n");
if (copy_from_user(&compr->info.codec_param, (void *) arg,
sizeof(struct snd_compr_params))) {
rc = -EFAULT;
pr_err("%s: ERROR: copy from user\n", __func__);
return rc;
}
switch (compr->info.codec_param.codec.id) {
case SND_AUDIOCODEC_MP3:
/* For MP3 we dont need any other parameter */
pr_debug("SND_AUDIOCODEC_MP3\n");
compr->codec = FORMAT_MP3;
break;
case SND_AUDIOCODEC_AAC:
pr_debug("SND_AUDIOCODEC_AAC\n");
compr->codec = FORMAT_MPEG4_AAC;
break;
case SND_AUDIOCODEC_AC3: {
char params_value[MAX_AC3_PARAM_SIZE];
int *params_value_data = (int *)params_value;
/* 36 is the max param length for ddp */
int i;
struct snd_dec_ddp *ddp =
&compr->info.codec_param.codec.options.ddp;
uint32_t params_length = 0;
memset(params_value, 0, MAX_AC3_PARAM_SIZE);
/* check integer overflow */
if (ddp->params_length > UINT_MAX/sizeof(int)) {
pr_err("%s: Integer overflow ddp->params_length %d\n",
__func__, ddp->params_length);
return -EINVAL;
}
params_length = ddp->params_length*sizeof(int);
if (params_length > MAX_AC3_PARAM_SIZE) {
/*MAX is 36*sizeof(int) this should not happen*/
pr_err("%s: params_length(%d) is greater than %zd\n",
__func__, params_length, MAX_AC3_PARAM_SIZE);
return -EINVAL;
}
pr_debug("SND_AUDIOCODEC_AC3\n");
compr->codec = FORMAT_AC3;
if (copy_from_user(params_value, (void *)ddp->params,
params_length))
pr_err("%s: copy ddp params value, size=%d\n",
__func__, params_length);
pr_debug("params_length: %d\n", ddp->params_length);
for (i = 0; i < params_length; i++)
pr_debug("params_value[%d]: %x\n", i,
params_value_data[i]);
for (i = 0; i < ddp->params_length/2; i++) {
ddp->params_id[i] = params_value_data[2*i];
ddp->params_value[i] = params_value_data[2*i+1];
}
if (atomic_read(&prtd->start)) {
rc = msm_compr_send_ddp_cfg(prtd->audio_client,
ddp);
if (rc < 0)
pr_err("%s: DDP CMD CFG failed\n",
__func__);
}
break;
}
case SND_AUDIOCODEC_EAC3: {
char params_value[MAX_AC3_PARAM_SIZE];
int *params_value_data = (int *)params_value;
/* 36 is the max param length for ddp */
int i;
struct snd_dec_ddp *ddp =
&compr->info.codec_param.codec.options.ddp;
uint32_t params_length = 0;
memset(params_value, 0, MAX_AC3_PARAM_SIZE);
/* check integer overflow */
if (ddp->params_length > UINT_MAX/sizeof(int)) {
pr_err("%s: Integer overflow ddp->params_length %d\n",
__func__, ddp->params_length);
return -EINVAL;
}
if (params_length > MAX_AC3_PARAM_SIZE) {
/*MAX is 36*sizeof(int) this should not happen*/
pr_err("%s: params_length(%d) is greater than %d\n",
__func__, params_length, MAX_AC3_PARAM_SIZE);
return -EINVAL;
}
pr_debug("SND_AUDIOCODEC_EAC3\n");
compr->codec = FORMAT_EAC3;
if (copy_from_user(params_value, (void *)ddp->params,
params_length))
pr_err("%s: copy ddp params value, size=%d\n",
__func__, params_length);
pr_debug("params_length: %d\n", ddp->params_length);
for (i = 0; i < ddp->params_length; i++)
pr_debug("params_value[%d]: %x\n", i,
params_value_data[i]);
for (i = 0; i < ddp->params_length/2; i++) {
ddp->params_id[i] = params_value_data[2*i];
ddp->params_value[i] = params_value_data[2*i+1];
}
if (atomic_read(&prtd->start)) {
rc = msm_compr_send_ddp_cfg(prtd->audio_client,
ddp);
if (rc < 0)
pr_err("%s: DDP CMD CFG failed\n",
__func__);
}
break;
}
default:
pr_debug("FORMAT_LINEAR_PCM\n");
compr->codec = FORMAT_LINEAR_PCM;
break;
}
return 0;
case SNDRV_PCM_IOCTL1_RESET:
pr_debug("SNDRV_PCM_IOCTL1_RESET\n");
/* Flush only when session is started during CAPTURE,
while PLAYBACK has no such restriction. */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
(substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
atomic_read(&prtd->start))) {
if (atomic_read(&prtd->eos)) {
prtd->cmd_interrupt = 1;
wake_up(&the_locks.eos_wait);
atomic_set(&prtd->eos, 0);
}
/* A unlikely race condition possible with FLUSH
DRAIN if ack is set by flush and reset by drain */
prtd->cmd_ack = 0;
rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
if (rc < 0) {
pr_err("%s: flush cmd failed rc=%d\n",
__func__, rc);
return rc;
}
rc = wait_event_timeout(the_locks.flush_wait,
prtd->cmd_ack, 5 * HZ);
if (!rc)
pr_err("Flush cmd timeout\n");
prtd->pcm_irq_pos = 0;
}
break;
case SNDRV_COMPRESS_DRAIN:
pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
if (atomic_read(&prtd->pending_buffer)) {
pr_debug("%s: no pending writes, drain would block\n",
__func__);
return -EWOULDBLOCK;
}
atomic_set(&prtd->eos, 1);
atomic_set(&prtd->pending_buffer, 0);
prtd->cmd_ack = 0;
q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
/* Wait indefinitely for DRAIN. Flush can also signal this*/
rc = wait_event_interruptible(the_locks.eos_wait,
(prtd->cmd_ack || prtd->cmd_interrupt));
if (rc < 0)
pr_err("EOS cmd interrupted\n");
pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait\n", __func__);
if (prtd->cmd_interrupt)
rc = -EINTR;
prtd->cmd_interrupt = 0;
return rc;
default:
break;
}
return snd_pcm_lib_ioctl(substream, cmd, arg);
}
static int msm_compr_restart(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
struct audio_aio_write_param param;
struct audio_buffer *buf = NULL;
struct output_meta_data_st output_meta_data;
int time_stamp_flag = 0;
int buffer_length = 0;
pr_debug("%s, trigger restart\n", __func__);
if (runtime->render_flag & SNDRV_RENDER_STOPPED) {
buf = prtd->audio_client->port[IN].buf;
pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
__func__, prtd->pcm_count, prtd->out_head);
pr_debug("%s:writing buffer[%d] from 0x%08x\n",
__func__, prtd->out_head,
((unsigned int)buf[0].phys
+ (prtd->out_head * prtd->pcm_count)));
if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
time_stamp_flag = SET_TIMESTAMP;
else
time_stamp_flag = NO_TIMESTAMP;
memcpy(&output_meta_data, (char *)(buf->data +
prtd->out_head * prtd->pcm_count),
COMPRE_OUTPUT_METADATA_SIZE);
buffer_length = output_meta_data.frame_size;
pr_debug("meta_data_length: %d, frame_length: %d\n",
output_meta_data.meta_data_length,
output_meta_data.frame_size);
pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
output_meta_data.timestamp_msw,
output_meta_data.timestamp_lsw);
param.paddr = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count)
+ output_meta_data.meta_data_length;
param.len = buffer_length;
param.msw_ts = output_meta_data.timestamp_msw;
param.lsw_ts = output_meta_data.timestamp_lsw;
param.flags = time_stamp_flag;
param.uid = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count
+ output_meta_data.meta_data_length);
if (q6asm_async_write(prtd->audio_client,
&param) < 0)
pr_err("%s:q6asm_async_write failed\n",
__func__);
else
prtd->out_head =
(prtd->out_head + 1) & (runtime->periods - 1);
runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
return 0;
}
return 0;
}
static int msm_compr_volume_ctl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int rc = 0;
struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
struct snd_pcm_substream *substream =
vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
struct msm_audio *prtd;
int volume = ucontrol->value.integer.value[0];
pr_debug("%s: volume : %x\n", __func__, volume);
if (!substream)
return -ENODEV;
if (!substream->runtime)
return 0;
prtd = substream->runtime->private_data;
if (prtd)
rc = compressed_set_volume(prtd, volume);
return rc;
}
static int msm_compr_volume_ctl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
struct snd_pcm_substream *substream =
vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
struct msm_audio *prtd;
pr_debug("%s\n", __func__);
if (!substream)
return -ENODEV;
if (!substream->runtime)
return 0;
prtd = substream->runtime->private_data;
if (prtd)
ucontrol->value.integer.value[0] = prtd->volume;
return 0;
}
static int msm_compr_add_controls(struct snd_soc_pcm_runtime *rtd)
{
int ret = 0;
struct snd_pcm *pcm = rtd->pcm;
struct snd_pcm_volume *volume_info;
struct snd_kcontrol *kctl;
dev_dbg(rtd->dev, "%s, Volume cntrl add\n", __func__);
ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
NULL, 1, rtd->dai_link->be_id,
&volume_info);
if (ret < 0)
return ret;
kctl = volume_info->kctl;
kctl->put = msm_compr_volume_ctl_put;
kctl->get = msm_compr_volume_ctl_get;
kctl->tlv.p = compr_rx_vol_gain;
return 0;
}
static struct snd_pcm_ops msm_compr_ops = {
.open = msm_compr_open,
.hw_params = msm_compr_hw_params,
.close = msm_compr_close,
.ioctl = msm_compr_ioctl,
.prepare = msm_compr_prepare,
.trigger = msm_compr_trigger,
.pointer = msm_compr_pointer,
.mmap = msm_compr_mmap,
.restart = msm_compr_restart,
};
static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
struct snd_card *card = rtd->card->snd_card;
int ret = 0;
if (!card->dev->coherent_dma_mask)
card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
ret = msm_compr_add_controls(rtd);
if (ret)
pr_err("%s, kctl add failed\n", __func__);
return ret;
}
static struct snd_soc_platform_driver msm_soc_platform = {
.ops = &msm_compr_ops,
.pcm_new = msm_asoc_pcm_new,
};
static __devinit int msm_compr_probe(struct platform_device *pdev)
{
if (pdev->dev.of_node)
dev_set_name(&pdev->dev, "%s", "msm-compr-dsp");
dev_info(&pdev->dev, "%s: dev name %s\n",
__func__, dev_name(&pdev->dev));
atomic_set(&compressed_audio.audio_ocmem_req, 0);
return snd_soc_register_platform(&pdev->dev,
&msm_soc_platform);
}
static int msm_compr_remove(struct platform_device *pdev)
{
snd_soc_unregister_platform(&pdev->dev);
return 0;
}
static const struct of_device_id msm_compr_dt_match[] = {
{.compatible = "qcom,msm-compr-dsp"},
{}
};
MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
static struct platform_driver msm_compr_driver = {
.driver = {
.name = "msm-compr-dsp",
.owner = THIS_MODULE,
.of_match_table = msm_compr_dt_match,
},
.probe = msm_compr_probe,
.remove = __devexit_p(msm_compr_remove),
};
static int __init msm_soc_platform_init(void)
{
init_waitqueue_head(&the_locks.enable_wait);
init_waitqueue_head(&the_locks.eos_wait);
init_waitqueue_head(&the_locks.write_wait);
init_waitqueue_head(&the_locks.read_wait);
init_waitqueue_head(&the_locks.flush_wait);
return platform_driver_register(&msm_compr_driver);
}
module_init(msm_soc_platform_init);
static void __exit msm_soc_platform_exit(void)
{
platform_driver_unregister(&msm_compr_driver);
}
module_exit(msm_soc_platform_exit);
MODULE_DESCRIPTION("PCM module platform driver");
MODULE_LICENSE("GPL v2");
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