DL-Mirrors/kernel-tenderloin-3.0
0
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Merge "ASoC: msm: Add driver to support compressed audio." into msm-3.0

Browse Source
This commit is contained in:
Linux Build Service Account
2011-12-08 10:28:10 -08:00
committed by QuIC Gerrit Code Review
parent a3d4e56017 305d1750ef
commit 0a1f5b898f
7 changed files with 673 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
sound/soc/msm/Makefile
View File

@@ -56,7 +56,7 @@ obj-$(CONFIG_SND_SOC_MSM8X60) += snd-soc-lpass-dma.o
obj-$(CONFIG_SND_SOC_MSM_QDSP6_INTF) += qdsp6/
snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o
snd-soc-qdsp6-objs := msm-dai-q6.o msm-pcm-q6.o msm-pcm-routing.o msm-dai-fe.o msm-compr-q6.o
obj-$(CONFIG_SND_SOC_VOICE) += msm-pcm-voice.o msm-pcm-voip.o
snd-soc-qdsp6-objs += msm-pcm-lpa.o msm-pcm-afe.o
obj-$(CONFIG_SND_SOC_QDSP6) += snd-soc-qdsp6.o

567
sound/soc/msm/msm-compr-q6.c Normal file
View File

@@ -0,0 +1,567 @@
/* Copyright (c) 2011, Code Aurora Forum. 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 <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/android_pmem.h>
#include "msm-compr-q6.h"
#include "msm-pcm-routing.h"
static struct audio_locks the_locks;
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,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 1,
.channels_max = 2,
.buffer_bytes_max = 1200 * 1024 * 2,
.period_bytes_min = 60 * 1024,
.period_bytes_max = 1200 * 1024,
.periods_min = 2,
.periods_max = 40,
.fifo_size = 0,
};
/* Conventional and unconventional sample rate supported */
static unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
};
static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
.count = ARRAY_SIZE(supported_sample_rates),
.list = supported_sample_rates,
.mask = 0,
};
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_buffer *buf = NULL;
int i = 0;
pr_debug("%s opcode =%08x\n", __func__, opcode);
switch (opcode) {
case ASM_DATA_EVENT_WRITE_DONE: {
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);
atomic_inc(&prtd->out_count);
wake_up(&the_locks.write_wait);
if (!atomic_read(&prtd->start)) {
prtd->pending_buffer = 1;
break;
} else
prtd->pending_buffer = 0;
if (runtime->status->hw_ptr >= runtime->control->appl_ptr)
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)));
param.paddr = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count);
param.len = prtd->pcm_count;
param.msw_ts = 0;
param.lsw_ts = 0;
param.flags = NO_TIMESTAMP;
param.uid = (unsigned long)buf[0].phys
+ (prtd->out_head * prtd->pcm_count);
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_CMDRSP_EOS:
pr_debug("ASM_DATA_CMDRSP_EOS\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.eos_wait);
break;
case APR_BASIC_RSP_RESULT: {
switch (payload[0]) {
case ASM_SESSION_CMD_RUN: {
if (!prtd->pending_buffer &&
!atomic_read(&prtd->start))
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)));
param.paddr = (unsigned long)buf[prtd->out_head].phys;
param.len = prtd->pcm_count;
param.msw_ts = 0;
param.lsw_ts = 0;
param.flags = NO_TIMESTAMP;
param.uid = (unsigned long)buf[prtd->out_head].phys;
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_STREAM_CMD_FLUSH:
pr_debug("ASM_STREAM_CMD_FLUSH\n");
prtd->cmd_ack = 1;
wake_up(&the_locks.eos_wait);
break;
default:
break;
}
break;
}
default:
pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
break;
}
}
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;
int ret;
pr_debug("%s\n", __func__);
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;
if (prtd->enabled)
return 0;
ret = q6asm_media_format_block(prtd->audio_client, compr->codec);
if (ret < 0)
pr_info("%s: CMD Format block failed\n", __func__);
atomic_set(&prtd->out_count, runtime->periods);
prtd->enabled = 1;
prtd->cmd_ack = 0;
return 0;
}
static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
{
int ret = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
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;
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");
atomic_set(&prtd->start, 0);
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);
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 = 1;
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;
/* Add new codecs here */
}
static int msm_compr_open(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;
struct msm_audio *prtd;
int ret = 0;
/* Capture path */
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
return -EINVAL;
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;
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;
}
runtime->hw = msm_compr_hardware_playback;
pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
prtd->session_id = prtd->audio_client->session;
msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->be_id,
prtd->session_id, substream->stream);
prtd->cmd_ack = 1;
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;
prtd->pending_buffer = 1;
compr->codec = FORMAT_MP3;
populate_codec_list(compr, runtime);
runtime->private_data = compr;
return 0;
}
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;
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_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 = EINVAL;
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 = EINVAL;
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("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
}
static int msm_compr_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma)
{
int result = 0;
struct snd_pcm_runtime *runtime = substream->runtime;
struct compr_audio *compr = runtime->private_data;
struct msm_audio *prtd = &compr->prtd;
pr_debug("%s\n", __func__);
prtd->mmap_flag = 1;
if (runtime->dma_addr && runtime->dma_bytes) {
vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
result = remap_pfn_range(vma, vma->vm_start,
runtime->dma_addr >> PAGE_SHIFT,
runtime->dma_bytes,
vma->vm_page_prot);
} else {
pr_err("Physical address or size of buf is NULL");
return -EINVAL;
}
return result;
}
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 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;
pr_debug("%s\n", __func__);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dir = IN;
else
return -EINVAL;
ret = q6asm_open_write(prtd->audio_client, compr->codec);
if (ret < 0) {
pr_err("%s: Session out open failed\n", __func__);
return -ENOMEM;
}
ret = q6asm_set_io_mode(prtd->audio_client, ASYNC_IO_MODE);
if (ret < 0) {
pr_err("%s: Set IO mode failed\n", __func__);
return -ENOMEM;
}
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;
pr_debug("%s:buf = %p\n", __func__, 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;
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;
switch (cmd) {
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: ");
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;
default:
pr_debug("FORMAT_LINEAR_PCM\n");
compr->codec = FORMAT_LINEAR_PCM;
break;
}
return 0;
case SNDRV_PCM_IOCTL1_RESET:
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);
rc = wait_event_timeout(the_locks.eos_wait,
prtd->cmd_ack, 5 * HZ);
if (rc < 0)
pr_err("Flush cmd timeout\n");
prtd->pcm_irq_pos = 0;
break;
default:
break;
}
return snd_pcm_lib_ioctl(substream, cmd, arg);
}
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,
};
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);
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)
{
pr_info("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
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 struct platform_driver msm_compr_driver = {
.driver = {
.name = "msm-compr-dsp",
.owner = THIS_MODULE,
},
.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);
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");

36
sound/soc/msm/msm-compr-q6.h Normal file
View File

@@ -0,0 +1,36 @@
/*
* Copyright (c) 2011, Code Aurora Forum. 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.
*/
#ifndef _MSM_COMPR_H
#define _MSM_COMPR_H
#include <sound/apr_audio.h>
#include <sound/q6asm.h>
#include <sound/snd_compress_params.h>
#include <sound/compress_offload.h>
#include <sound/compress_driver.h>
#include "msm-pcm-q6.h"
struct compr_info {
struct snd_compr_caps compr_cap;
struct snd_compr_codec_caps codec_caps;
struct snd_compr_params codec_param;
};
struct compr_audio {
struct msm_audio prtd;
struct compr_info info;
uint32_t codec;
};
#endif /*_MSM_COMPR_H*/

14
sound/soc/msm/msm-dai-fe.c
View File

@@ -151,6 +151,20 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
.ops = &msm_fe_Multimedia_dai_ops,
.name = "MultiMedia3",
},
{
.playback = {
.stream_name = "MultiMedia4 Playback",
.rates = (SNDRV_PCM_RATE_8000_48000 |
SNDRV_PCM_RATE_KNOT),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
.channels_max = 2,
.rate_min = 8000,
.rate_max = 48000,
},
.ops = &msm_fe_Multimedia_dai_ops,
.name = "MultiMedia4",
},
/* FE DAIs created for hostless operation purpose */
{
.playback = {

31
sound/soc/msm/msm-pcm-routing.c
View File

@@ -125,6 +125,8 @@ static int fe_dai_map[MSM_FRONTEND_DAI_MAX][2] = {
{INVALID_SESSION, INVALID_SESSION},
/* MULTIMEDIA3 */
{INVALID_SESSION, INVALID_SESSION},
/* MULTIMEDIA4 */
{INVALID_SESSION, INVALID_SESSION},
};
static void msm_pcm_routing_build_matrix(int fedai_id, int dspst_id,
@@ -615,6 +617,9 @@ static const struct snd_kcontrol_new pri_i2s_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_PRI_I2S_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_PRI_I2S_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
@@ -627,6 +632,9 @@ static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
@@ -639,6 +647,9 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_HDMI_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_HDMI_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = {
@@ -651,6 +662,9 @@ static const struct snd_kcontrol_new int_bt_sco_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_BT_SCO_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_BT_SCO_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
@@ -663,6 +677,9 @@ static const struct snd_kcontrol_new int_fm_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_INT_FM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_INT_FM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
@@ -675,6 +692,9 @@ static const struct snd_kcontrol_new afe_pcm_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AFE_PCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AFE_PCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
@@ -687,6 +707,9 @@ static const struct snd_kcontrol_new auxpcm_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia3", MSM_BACKEND_DAI_AUXPCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
SOC_SINGLE_EXT("MultiMedia4", MSM_BACKEND_DAI_AUXPCM_RX,
MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
msm_routing_put_audio_mixer),
};
static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
@@ -1014,6 +1037,7 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_IN("VOIP_DL", "VoIP Playback", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
@@ -1111,16 +1135,19 @@ static const struct snd_soc_dapm_route intercon[] = {
{"PRI_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"PRI_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"PRI_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"PRI_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"PRI_I2S_RX", NULL, "PRI_RX Audio Mixer"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
{"HDMI Mixer", "MultiMedia1", "MM_DL1"},
{"HDMI Mixer", "MultiMedia2", "MM_DL2"},
{"HDMI Mixer", "MultiMedia3", "MM_DL3"},
{"HDMI Mixer", "MultiMedia4", "MM_DL4"},
{"HDMI", NULL, "HDMI Mixer"},
{"MultiMedia1 Mixer", "PRI_TX", "PRI_I2S_TX"},
@@ -1130,16 +1157,19 @@ static const struct snd_soc_dapm_route intercon[] = {
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"INTERNAL_BT_SCO_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"INT_BT_SCO_RX", NULL, "INTERNAL_BT_SCO_RX Audio Mixer"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"INTERNAL_FM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"INT_FM_RX", NULL, "INTERNAL_FM_RX Audio Mixer"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"AFE_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"PCM_RX", NULL, "AFE_PCM_RX Audio Mixer"},
{"MultiMedia1 Mixer", "INTERNAL_BT_SCO_TX", "INT_BT_SCO_TX"},
@@ -1153,6 +1183,7 @@ static const struct snd_soc_dapm_route intercon[] = {
{"AUX_PCM_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
{"AUX_PCM_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
{"AUX_PCM_RX", NULL, "AUX_PCM_RX Audio Mixer"},
{"PRI_RX_Voice Mixer", "CSVoice", "CS-VOICE_DL1"},

1
sound/soc/msm/msm-pcm-routing.h
View File

@@ -31,6 +31,7 @@ enum {
MSM_FRONTEND_DAI_MULTIMEDIA1 = 0,
MSM_FRONTEND_DAI_MULTIMEDIA2,
MSM_FRONTEND_DAI_MULTIMEDIA3,
MSM_FRONTEND_DAI_MULTIMEDIA4,
MSM_FRONTEND_DAI_CS_VOICE,
MSM_FRONTEND_DAI_VOIP,
MSM_FRONTEND_DAI_AFE_RX,

24
sound/soc/msm/qdsp6/q6asm.c
View File

@@ -1202,6 +1202,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format)
case FORMAT_WMA_V10PRO:
open.format = WMA_V10PRO;
break;
case FORMAT_MP3:
open.format = MP3;
break;
default:
pr_err("%s: Invalid format[%d]\n", __func__, format);
goto fail_cmd;
@@ -1865,7 +1868,26 @@ int q6asm_media_format_block(struct audio_client *ac, uint32_t format)
q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FORMAT_UPDATE;
fmt.format = format;
switch (format) {
case FORMAT_V13K:
fmt.format = V13K_FS;
break;
case FORMAT_EVRC:
fmt.format = EVRC_FS;
break;
case FORMAT_AMRWB:
fmt.format = AMRWB_FS;
break;
case FORMAT_AMRNB:
fmt.format = AMRNB_FS;
break;
case FORMAT_MP3:
fmt.format = MP3;
break;
default:
pr_err("Invalid format[%d]\n", format);
goto fail_cmd;
}
fmt.cfg_size = 0;
rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);