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DivestOS/Patches/Linux_CVEs/CVE-2017-9677/0.patch

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From b62291edb424281ed31a4e15140b16972ce9eef1 Mon Sep 17 00:00:00 2001
From: Xiaojun Sang <xsang@codeaurora.org>
Date: Thu, 27 Apr 2017 14:44:25 +0800
Subject: ASoC: msm: remove unused msm-compr-q6-v2
msm-compr-q6-v2.c and msm-compr-q6-v2.h are no longer used.
CRs-Fixed: 2022953
Bug: 62379475
Change-Id: I856d90a212a3e123a2c8b80092aff003f7c608c7
Signed-off-by: Xiaojun Sang <xsang@codeaurora.org>
---
sound/soc/msm/apq8084-i2s.c | 2 +-
sound/soc/msm/apq8084.c | 2 +-
sound/soc/msm/msm8226.c | 2 +-
sound/soc/msm/msm8974.c | 2 +-
sound/soc/msm/msm8994.c | 2 +-
sound/soc/msm/qdsp6v2/Makefile | 2 +-
sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c | 1707 -------------------------------
sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h | 36 -
8 files changed, 6 insertions(+), 1749 deletions(-)
delete mode 100644 sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
delete mode 100644 sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
diff --git a/sound/soc/msm/apq8084-i2s.c b/sound/soc/msm/apq8084-i2s.c
index 794aa25..5897e9c 100644
--- a/sound/soc/msm/apq8084-i2s.c
+++ b/sound/soc/msm/apq8084-i2s.c
@@ -1826,7 +1826,7 @@ static struct snd_soc_dai_link apq8084_dai_links[] = {
.name = "APQ8084 Compr8",
.stream_name = "COMPR8",
.cpu_dai_name = "MultiMedia8",
- .platform_name = "msm-compr-dsp",
+ .platform_name = "msm-compress-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/msm/apq8084.c b/sound/soc/msm/apq8084.c
index aa2e25f..2b02e5d 100644
--- a/sound/soc/msm/apq8084.c
+++ b/sound/soc/msm/apq8084.c
@@ -3046,7 +3046,7 @@ static struct snd_soc_dai_link apq8084_common_dai_links[] = {
.name = "APQ8084 Compr8",
.stream_name = "COMPR8",
.cpu_dai_name = "MultiMedia8",
- .platform_name = "msm-compr-dsp",
+ .platform_name = "msm-compress-dsp",
.dynamic = 1,
.async_ops = ASYNC_DPCM_SND_SOC_PREPARE
| ASYNC_DPCM_SND_SOC_HW_PARAMS,
diff --git a/sound/soc/msm/msm8226.c b/sound/soc/msm/msm8226.c
index 4095c12..113d77b 100644
--- a/sound/soc/msm/msm8226.c
+++ b/sound/soc/msm/msm8226.c
@@ -1495,7 +1495,7 @@ static struct snd_soc_dai_link msm8226_common_dai[] = {
.name = "MSM8226 Compr8",
.stream_name = "COMPR8",
.cpu_dai_name = "MultiMedia8",
- .platform_name = "msm-compr-dsp",
+ .platform_name = "msm-compress-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/msm/msm8974.c b/sound/soc/msm/msm8974.c
index fd69611..4cfd7c3 100644
--- a/sound/soc/msm/msm8974.c
+++ b/sound/soc/msm/msm8974.c
@@ -2164,7 +2164,7 @@ static struct snd_soc_dai_link msm8974_common_dai_links[] = {
.name = "MSM8974 Compr8",
.stream_name = "COMPR8",
.cpu_dai_name = "MultiMedia8",
- .platform_name = "msm-compr-dsp",
+ .platform_name = "msm-compress-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/msm/msm8994.c b/sound/soc/msm/msm8994.c
index 1285c59..8678fb1 100644
--- a/sound/soc/msm/msm8994.c
+++ b/sound/soc/msm/msm8994.c
@@ -2684,7 +2684,7 @@ static struct snd_soc_dai_link msm8994_common_dai_links[] = {
.name = "MSM8994 Compr8",
.stream_name = "COMPR8",
.cpu_dai_name = "MultiMedia8",
- .platform_name = "msm-compr-dsp",
+ .platform_name = "msm-compress-dsp",
.dynamic = 1,
.trigger = {SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/msm/qdsp6v2/Makefile b/sound/soc/msm/qdsp6v2/Makefile
index 5865eb9..41f3984 100644
--- a/sound/soc/msm/qdsp6v2/Makefile
+++ b/sound/soc/msm/qdsp6v2/Makefile
@@ -1,5 +1,5 @@
snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \
- msm-compress-q6-v2.o msm-compr-q6-v2.o \
+ msm-compress-q6-v2.o \
msm-pcm-lpa-v2.o \
msm-pcm-afe-v2.o msm-pcm-voip-v2.o \
msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
deleted file mode 100644
index 5fe5f24..0000000
--- a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
+++ /dev/null
@@ -1,1707 +0,0 @@
-/* Copyright (c) 2012-2014, 2016 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;
- phys_addr_t temp;
- 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);
- temp = buf[0].phys + (prtd->out_head * prtd->pcm_count);
- pr_debug("%s:writing buffer[%d] from 0x%pa\n",
- __func__, prtd->out_head, &temp);
-
- 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 = temp + 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 = prtd->session_id;
- 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 = %pK, data = 0x%X, *data = %pK,\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 = %pK, updated data = 0x%X, *data = %pK\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%pa\n",
- prtd->pcm_irq_pos, &buf[0].phys);
- read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
- read_param.paddr = 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%pa head %d count %d\n",
- __func__, prtd->out_head, &buf[0].phys,
- prtd->pcm_count, prtd->out_head);
- 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 = 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 = prtd->session_id;
- 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 snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct msm_audio *prtd = &compr->prtd;
- struct snd_pcm_hw_params *params;
- struct asm_aac_cfg aac_cfg;
- uint16_t bits_per_sample = 16;
- int ret;
-
- 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__);
-
- params = &soc_prtd->dpcm[substream->stream].hw_params;
- if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
- bits_per_sample = 24;
-
- 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 = 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);
-
- 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;
- }
-
- 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 snd_soc_pcm_runtime *soc_prtd = substream->private_data;
- struct audio_aio_read_param read_param;
- uint16_t bits_per_sample = 16;
- 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 (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
- bits_per_sample = 24;
-
- 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;
- }
-
- 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 = buf[i].phys
- + COMPRE_CAPTURE_HEADER_SIZE;
- pr_debug("Push buffer [%d] to DSP, paddr: %pa, vaddr: %pK\n",
- i, &read_param.paddr,
- buf[i].data);
- q6asm_async_read(prtd->audio_client, &read_param);
- break;
- default:
- read_param.paddr = 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 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;
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dir = IN;
- else
- dir = OUT;
- /* 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 %zd 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[%pK]dma_buf->area[%pK]dma_buf->addr[%pa]\n"
- "dma_buf->bytes[%zd]\n", __func__,
- (void *)buf, (void *)dma_buf->area,
- &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_shared(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");
- tstamp = arg;
- memset(tstamp, 0x0, sizeof(*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);
- return 0;
- }
- case SNDRV_COMPRESS_GET_CAPS: {
- struct snd_compr_caps *caps;
- caps = arg;
- memset(caps, 0, sizeof(*caps));
- pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
- memcpy(caps, &compr->info.compr_cap, sizeof(*caps));
- return 0;
- }
- case SNDRV_COMPRESS_SET_PARAMS:
- pr_debug("SNDRV_COMPRESS_SET_PARAMS:\n");
- memcpy(&compr->info.codec_param, (void *) arg,
- sizeof(struct snd_compr_params));
- 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;
- pr_debug("params_length: %d\n", ddp->params_length);
- for (i = 0; i < params_length/sizeof(int); 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;
- }
- 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_EAC3\n");
- compr->codec = FORMAT_EAC3;
- 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);
-}
-#ifdef CONFIG_COMPAT
-struct snd_enc_wma32 {
- u32 super_block_align; /* WMA Type-specific data */
- u32 encodeopt1;
- u32 encodeopt2;
-};
-
-struct snd_enc_vorbis32 {
- s32 quality;
- u32 managed;
- u32 max_bit_rate;
- u32 min_bit_rate;
- u32 downmix;
-};
-
-struct snd_enc_real32 {
- u32 quant_bits;
- u32 start_region;
- u32 num_regions;
-};
-
-struct snd_enc_flac32 {
- u32 num;
- u32 gain;
-};
-
-struct snd_enc_generic32 {
- u32 bw; /* encoder bandwidth */
- s32 reserved[15];
-};
-struct snd_dec_ddp32 {
- u32 params_length;
- u32 params_id[18];
- u32 params_value[18];
-};
-
-union snd_codec_options32 {
- struct snd_enc_wma32 wma;
- struct snd_enc_vorbis32 vorbis;
- struct snd_enc_real32 real;
- struct snd_enc_flac32 flac;
- struct snd_enc_generic32 generic;
- struct snd_dec_ddp32 ddp;
-};
-
-struct snd_codec32 {
- 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_options32 options;
- u32 reserved[3];
-};
-
-struct snd_compressed_buffer32 {
- u32 fragment_size;
- u32 fragments;
-};
-
-struct snd_compr_params32 {
- struct snd_compressed_buffer32 buffer;
- struct snd_codec32 codec;
- u8 no_wake_mode;
-};
-
-struct snd_compr_caps32 {
- 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_tstamp32 {
- u32 byte_offset;
- u32 copied_total;
- compat_ulong_t pcm_frames;
- compat_ulong_t pcm_io_frames;
- u32 sampling_rate;
- compat_u64 timestamp;
-};
-enum {
- SNDRV_COMPRESS_TSTAMP32 = _IOR('C', 0x20, struct snd_compr_tstamp32),
- SNDRV_COMPRESS_GET_CAPS32 = _IOWR('C', 0x10, struct snd_compr_caps32),
- SNDRV_COMPRESS_SET_PARAMS32 =
- _IOW('C', 0x12, struct snd_compr_params32),
-};
-static int msm_compr_compat_ioctl(struct snd_pcm_substream *substream,
- unsigned int cmd, void *arg)
-{
- int err = 0;
- switch (cmd) {
- case SNDRV_COMPRESS_TSTAMP32: {
- struct snd_compr_tstamp tstamp;
- struct snd_compr_tstamp32 tstamp32;
- memset(&tstamp, 0, sizeof(tstamp));
- memset(&tstamp32, 0, sizeof(tstamp32));
- cmd = SNDRV_COMPRESS_TSTAMP;
- err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
- if (err) {
- pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
- __func__, err);
- goto bail_out;
- }
- tstamp32.byte_offset = tstamp.byte_offset;
- tstamp32.copied_total = tstamp.copied_total;
- tstamp32.pcm_frames = tstamp.pcm_frames;
- tstamp32.pcm_io_frames = tstamp.pcm_io_frames;
- tstamp32.sampling_rate = tstamp.sampling_rate;
- tstamp32.timestamp = tstamp.timestamp;
- if (copy_to_user(arg, &tstamp32, sizeof(tstamp32))) {
- pr_err("%s: copytouser failed COMPRESS_TSTAMP32\n",
- __func__);
- err = -EFAULT;
- }
- break;
- }
- case SNDRV_COMPRESS_GET_CAPS32: {
- struct snd_compr_caps caps;
- struct snd_compr_caps32 caps32;
- u32 i;
- memset(&caps, 0, sizeof(caps));
- memset(&caps32, 0, sizeof(caps32));
- cmd = SNDRV_COMPRESS_GET_CAPS;
- err = msm_compr_ioctl_shared(substream, cmd, &caps);
- if (err) {
- pr_err("%s: GET_CAPS failed rc %d\n",
- __func__, err);
- goto bail_out;
- }
- pr_debug("SNDRV_COMPRESS_GET_CAPS_32\n");
- if (!err && caps.num_codecs >= MAX_NUM_CODECS) {
- pr_err("%s: Invalid number of codecs\n", __func__);
- err = -EINVAL;
- goto bail_out;
- }
- caps32.direction = caps.direction;
- caps32.max_fragment_size = caps.max_fragment_size;
- caps32.max_fragments = caps.max_fragments;
- caps32.min_fragment_size = caps.min_fragment_size;
- caps32.num_codecs = caps.num_codecs;
- for (i = 0; i < caps.num_codecs; i++)
- caps32.codecs[i] = caps.codecs[i];
- if (copy_to_user(arg, &caps32, sizeof(caps32))) {
- pr_err("%s: copytouser failed COMPRESS_GETCAPS32\n",
- __func__);
- err = -EFAULT;
- }
- break;
- }
- case SNDRV_COMPRESS_SET_PARAMS32: {
- struct snd_compr_params32 params32;
- struct snd_compr_params params;
- memset(&params32, 0 , sizeof(params32));
- memset(&params, 0 , sizeof(params));
- cmd = SNDRV_COMPRESS_SET_PARAMS;
- if (copy_from_user(&params32, arg, sizeof(params32))) {
- pr_err("%s: copyfromuser failed SET_PARAMS32\n",
- __func__);
- err = -EFAULT;
- goto bail_out;
- }
- params.no_wake_mode = params32.no_wake_mode;
- params.codec.id = params32.codec.id;
- params.codec.ch_in = params32.codec.ch_in;
- params.codec.ch_out = params32.codec.ch_out;
- params.codec.sample_rate = params32.codec.sample_rate;
- params.codec.bit_rate = params32.codec.bit_rate;
- params.codec.rate_control = params32.codec.rate_control;
- params.codec.profile = params32.codec.profile;
- params.codec.level = params32.codec.level;
- params.codec.ch_mode = params32.codec.ch_mode;
- params.codec.format = params32.codec.format;
- params.codec.align = params32.codec.align;
-
- switch (params.codec.id) {
- case SND_AUDIOCODEC_WMA:
- case SND_AUDIOCODEC_WMA_PRO:
- params.codec.options.wma.encodeopt1 =
- params32.codec.options.wma.encodeopt1;
- params.codec.options.wma.encodeopt2 =
- params32.codec.options.wma.encodeopt2;
- params.codec.options.wma.super_block_align =
- params32.codec.options.wma.super_block_align;
- break;
- case SND_AUDIOCODEC_VORBIS:
- params.codec.options.vorbis.downmix =
- params32.codec.options.vorbis.downmix;
- params.codec.options.vorbis.managed =
- params32.codec.options.vorbis.managed;
- params.codec.options.vorbis.max_bit_rate =
- params32.codec.options.vorbis.max_bit_rate;
- params.codec.options.vorbis.min_bit_rate =
- params32.codec.options.vorbis.min_bit_rate;
- params.codec.options.vorbis.quality =
- params32.codec.options.vorbis.quality;
- break;
- case SND_AUDIOCODEC_REAL:
- params.codec.options.real.num_regions =
- params32.codec.options.real.num_regions;
- params.codec.options.real.quant_bits =
- params32.codec.options.real.quant_bits;
- params.codec.options.real.start_region =
- params32.codec.options.real.start_region;
- break;
- case SND_AUDIOCODEC_FLAC:
- params.codec.options.flac.gain =
- params32.codec.options.flac.gain;
- params.codec.options.flac.num =
- params32.codec.options.flac.num;
- break;
- case SND_AUDIOCODEC_DTS:
- case SND_AUDIOCODEC_DTS_PASS_THROUGH:
- case SND_AUDIOCODEC_DTS_LBR:
- case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
- case SND_AUDIOCODEC_DTS_TRANSCODE_LOOPBACK:
- break;
- case SND_AUDIOCODEC_AC3:
- case SND_AUDIOCODEC_EAC3:
- params.codec.options.ddp.params_length =
- params32.codec.options.ddp.params_length;
- memcpy(params.codec.options.ddp.params_value,
- params32.codec.options.ddp.params_value,
- sizeof(params32.codec.options.ddp.params_value));
- memcpy(params.codec.options.ddp.params_id,
- params32.codec.options.ddp.params_id,
- sizeof(params32.codec.options.ddp.params_id));
- break;
- default:
- params.codec.options.generic.bw =
- params32.codec.options.generic.bw;
- break;
- }
- if (!err)
- err = msm_compr_ioctl_shared(substream, cmd, &params);
- break;
- }
- default:
- err = msm_compr_ioctl_shared(substream, cmd, arg);
- }
-bail_out:
- return err;
-
-}
-#endif
-static int msm_compr_ioctl(struct snd_pcm_substream *substream,
- unsigned int cmd, void *arg)
-{
- int err = 0;
- if (!substream) {
- pr_err("%s: Invalid params\n", __func__);
- return -EINVAL;
- }
- pr_debug("%s called with cmd = %d\n", __func__, cmd);
- switch (cmd) {
- case SNDRV_COMPRESS_TSTAMP: {
- struct snd_compr_tstamp tstamp;
- if (!arg) {
- pr_err("%s: Invalid params Tstamp\n", __func__);
- return -EINVAL;
- }
- err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
- if (err)
- pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
- __func__, err);
- if (!err && copy_to_user(arg, &tstamp, sizeof(tstamp))) {
- pr_err("%s: copytouser failed COMPRESS_TSTAMP\n",
- __func__);
- err = -EFAULT;
- }
- break;
- }
- case SNDRV_COMPRESS_GET_CAPS: {
- struct snd_compr_caps cap;
- if (!arg) {
- pr_err("%s: Invalid params getcaps\n", __func__);
- return -EINVAL;
- }
- pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
- err = msm_compr_ioctl_shared(substream, cmd, &cap);
- if (err)
- pr_err("%s: GET_CAPS failed rc %d\n",
- __func__, err);
- if (!err && copy_to_user(arg, &cap, sizeof(cap))) {
- pr_err("%s: copytouser failed GET_CAPS\n",
- __func__);
- err = -EFAULT;
- }
- break;
- }
- case SNDRV_COMPRESS_SET_PARAMS: {
- struct snd_compr_params params;
- if (!arg) {
- pr_err("%s: Invalid params setparam\n", __func__);
- return -EINVAL;
- }
- if (copy_from_user(&params, arg,
- sizeof(struct snd_compr_params))) {
- pr_err("%s: SET_PARAMS\n", __func__);
- return -EFAULT;
- }
- err = msm_compr_ioctl_shared(substream, cmd, &params);
- if (err)
- pr_err("%s: SET_PARAMS failed rc %d\n",
- __func__, err);
- break;
- }
- default:
- err = msm_compr_ioctl_shared(substream, cmd, arg);
- }
- return err;
-}
-
-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 = prtd->session_id;
- 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,
-#ifdef CONFIG_COMPAT
- .compat_ioctl = msm_compr_compat_ioctl,
-#endif
-};
-
-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 int msm_compr_probe(struct platform_device *pdev)
-{
-
- 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 = 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");
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
deleted file mode 100644
index d6e3ec6..0000000
--- a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2012, 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.
- */
-
-#ifndef _MSM_COMPR_H
-#define _MSM_COMPR_H
-#include <sound/apr_audio-v2.h>
-#include <sound/q6asm-v2.h>
-#include <sound/compress_params.h>
-#include <sound/compress_offload.h>
-#include <sound/compress_driver.h>
-
-#include "msm-pcm-q6-v2.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*/
--
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