背景需求
做语音识别和答录功能的app大都需要将通话录音上下行VOICE_DOWNLINK/VOICE_UPLINK单独分离实时转义识别。
两个解决方向:
1、AudioRecord 方法的立体声录制PCM数据混合音源了，二进制文件无法区分哪些数据是左声道，哪些是右声道。
通过修改AudioRecord立体声PCM录制方案，每8位一组数据，左声道8位在前，右声道8位在后。
2、使用两个实例同时运行MediaRecorder的VOICE_UPLINK，VOICE_DOWNLINK录制上行或下行数据，且数据不重用。
基于这两个方向我们都需要去看 Audio HAL 代码。
最终结论
一开始从方向二出发，由于安卓本身的录音机制同一时间只允许一个应用录音，首先需要破解这个机制，
Android Audio - 支持多应用同时录音_Android8.1修改方法 参考这个修改后，确实可以同时 start 两个 MediaRecorder。但最终问题是不管如何设置 AudioSource
VOICE_DOWNLINK 和 VOICE_UPLINK，保存录音文件内容都是一样的，依旧是上下行混在一起。最终发现底层 buffer 是同一个，所以
保存的数据一样，这块尝试改了好几天，没啥进展，具体分析过程看下文。
再尝试方向一，通过 AudioRecord 获取原始 pcm 数据按 8 位分离，最终通过这个搞定了！
分析过程
主要的代码目录位于
vendor/mediatek/proprietary/hardware/audio/common/V3/
framework 相关的分析这里就不写了，根据这篇查看
这里从 adev_open_input_stream() 接入
vendor\mediatek\proprietary\hardware\audio\common\aud_drv\audio_hw_hal.cpp
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags /*__unused*/,
const char *address __unused,
audio_source_t source /*__unused*/) {
#ifdef AUDIO_HAL_PROFILE_ENTRY_FUNCTION
AudioAutoTimeProfile _p(__func__);
#endif
struct legacy_audio_device *ladev = to_ladev(dev);
status_t status = (status_t) handle;;
struct legacy_stream_in *in;
int ret;
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyInputSource), source);
in = (struct legacy_stream_in *)calloc(1, sizeof(*in));
if (!in) {
return -ENOMEM;
}
//这里加了log打印，发现走的 openInputStreamWithFlags()
#ifdef UPLINK_LOW_LATENCY
ALOGD("-%s() %d", __FUNCTION__, __LINE__);
in->legacy_in = ladev->hwif->openInputStreamWithFlags(devices, (int *) &config->format,
&config->channel_mask, &config->sample_rate,
&status, (audio_in_acoustics_t)0, flags);
#else
ALOGD("-%s() %d", __FUNCTION__, __LINE__);
in->legacy_in = ladev->hwif->openInputStream(devices, (int *) &config->format,
&config->channel_mask, &config->sample_rate,
&status, (audio_in_acoustics_t)0);
#endif
if (!in->legacy_in) {
ret = status;
goto err_open;
}
in->legacy_in->setParameters(param.toString());
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
.....
进入 AudioALSAHardware::openInputStreamWithFlags
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAHardware.cpp
AudioMTKStreamInInterface *AudioALSAHardware::openInputStreamWithFlags(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status,
audio_in_acoustics_t acoustics,
audio_input_flags_t flags) {
ALOGD("-%s() %d", __FUNCTION__, __LINE__);
return mStreamManager->openInputStream(devices, format, channels, sampleRate, status, acoustics, flags);
}
继续跟进 AudioALSAStreamManager::openInputStream
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAStreamManager.cpp
AudioMTKStreamInInterface *AudioALSAStreamManager::openInputStream(
uint32_t devices,
int *format,
uint32_t *channels,
uint32_t *sampleRate,
status_t *status,
audio_in_acoustics_t acoustics,
uint32_t input_flag) {
.....
// create stream in
AudioALSAStreamIn *pAudioALSAStreamIn = new AudioALSAStreamIn();
audio_devices_t input_device = static_cast<audio_devices_t>(devices);
bool sharedDevice = (input_device & ~AUDIO_DEVICE_BIT_IN) & (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC | AUDIO_DEVICE_IN_WIRED_HEADSET);
.....
pAudioALSAStreamIn->set(devices, format, channels, sampleRate, status, acoustics, input_flag);
ALOGD("-%s(), in = %p, status = 0x%x, mStreamInVector.size() = %zu",
__FUNCTION__, pAudioALSAStreamIn, *status, mStreamInVector.size());
return pAudioALSAStreamIn;
AudioALSAStreamIn 非常重要，最后干活的将是这个对象,这里将 pAudioALSAStreamIn 返回到 audio_hw_hal.cpp 中
顺序调用 in->legacy_in->setParameters(param.toString())，其实调用 AudioALSAStreamIn->setParameters()
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAStreamIn.cpp
status_t AudioALSAStreamIn::setParameters(const String8 &keyValuePairs) {
ALOGD("+%s(): %s", __FUNCTION__, keyValuePairs.string());
AudioParameter param = AudioParameter(keyValuePairs);
/// keys
const String8 keyInputSource = String8(AudioParameter::keyInputSource);
const String8 keyRouting = String8(AudioParameter::keyRouting);
/// parse key value pairs
status_t status = NO_ERROR;
int value = 0;
int oldCount;
String8 value_str;
/// intput source
if (param.getInt(keyInputSource, value) == NO_ERROR) {
param.remove(keyInputSource);
oldCount = android_atomic_inc(&mLockCount);
// TODO(Harvey): input source
AL_AUTOLOCK(mLock);
ALOGV("%s() InputSource = %d", __FUNCTION__, value);
mStreamAttributeTarget.input_source = static_cast<audio_source_t>(value);
......
关键的地方在 mStreamAttributeTarget.input_source 这个值即上层 setAudioSource 传递对应 audio_source_t 值
system\media\audio\include\system\audio-base.h
typedef enum {
AUDIO_SOURCE_DEFAULT = 0,
AUDIO_SOURCE_MIC = 1,
AUDIO_SOURCE_VOICE_UPLINK = 2,
AUDIO_SOURCE_VOICE_DOWNLINK = 3,
AUDIO_SOURCE_VOICE_CALL = 4,
AUDIO_SOURCE_CAMCORDER = 5,
AUDIO_SOURCE_VOICE_RECOGNITION = 6,
AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
AUDIO_SOURCE_REMOTE_SUBMIX = 8,
AUDIO_SOURCE_UNPROCESSED = 9,
AUDIO_SOURCE_CNT = 10,
AUDIO_SOURCE_MAX = 9, // (CNT - 1)
AUDIO_SOURCE_FM_TUNER = 1998,
AUDIO_SOURCE_HOTWORD = 1999,
} audio_source_t;
一系列参数设置完成后，将开始 read 数据。回到 audio_hw_hal.cpp 中
static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
size_t bytes) {
#ifdef AUDIO_HAL_PROFILE_ENTRY_FUNCTION
AudioAutoTimeProfile _p(__func__, AUDIO_HAL_FUNCTION_READ_NS);
#endif
struct legacy_stream_in *in =
reinterpret_cast<struct legacy_stream_in *>(stream);
return in->legacy_in->read(buffer, bytes);
}
最终还是 AudioALSAStreamIn 干活，ssize_t AudioALSAStreamIn::read(void *buffer, ssize_t bytes)
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAStreamIn.cpp
ssize_t AudioALSAStreamIn::read(void *buffer, ssize_t bytes) {
ALOGV("%s(), bytes= %zu", __FUNCTION__, bytes);
if (tempDebugflag) {
ALOGD("%s()+, bytes= %zu", __FUNCTION__, bytes);
}
ssize_t ret_size = bytes;
/* fast record is RT thread and keep streamin lock.
so other thread can't get streamin lock. if necessary, read will active release CPU. */
int tryCount = 10;
while ((mLockCount != 0 || mSuspendLockCount != 0) && tryCount--) {
ALOGV("%s, free CPU, mLockCount = %d, mSuspendLockCount = %d, tryCount %d", __FUNCTION__, mLockCount, mSuspendLockCount, tryCount);
usleep(300);
if (tryCount == 0) {
ALOGD("%s, free CPU, mLockCount = %d, mSuspendLockCount = %d, tryCount %d", __FUNCTION__, mLockCount, mSuspendLockCount, tryCount);
}
}
....
/// check open
if (mStandby == true) {
status = open();
}
看到这 open() 方法，这个方法有点东西
status_t AudioALSAStreamIn::open() {
// call open() only when mLock is locked.
ASSERT(AL_TRYLOCK(mLock) != 0);
ALOGD("%s()", __FUNCTION__);
status_t status = NO_ERROR;
if (mStandby == true) {
// create capture handler
ASSERT(mCaptureHandler == NULL);
mCaptureHandler = mStreamManager->createCaptureHandler(&mStreamAttributeTarget);
if (mCaptureHandler == NULL) {
status = BAD_VALUE;
return status;
}
mStandby = false;
// open audio hardware
status = mCaptureHandler->open();
ASSERT(status == NO_ERROR);
OpenPCMDump();
}
return status;
}
mStreamManager->createCaptureHandler 将刚刚的参数一并传递
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAStreamManager.cpp
AudioALSACaptureHandlerBase *AudioALSAStreamManager::createCaptureHandler(
stream_attribute_t *stream_attribute_target) {
ALOGD("+%s(), mAudioMode = %d, input_source = %d, input_device = 0x%x, mBypassDualMICProcessUL=%d, sample_rate=%d",
__FUNCTION__, mAudioMode, stream_attribute_target->input_source, stream_attribute_target->input_device, mBypassDualMICProcessUL, stream_attribute_target->sample_rate);
//AL_AUTOLOCK(mLock);
.....
else if ((stream_attribute_target->input_source == AUDIO_SOURCE_VOICE_UPLINK) || (stream_attribute_target->input_source == AUDIO_SOURCE_VOICE_DOWNLINK) ||
(stream_attribute_target->input_source == AUDIO_SOURCE_VOICE_CALL) ||
((isModeInPhoneCall() == true) && (WCNChipController::GetInstance()->IsBTMergeInterfaceSupported() == false) && (stream_attribute_target->input_device == AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET))) {
if ((isModeInPhoneCall() == false) ||
(SpeechDriverFactory::GetInstance()->GetSpeechDriver()->GetApSideModemStatus(SPEECH_STATUS_MASK) == false)) {
ALOGD("Can not open PhoneCall Record now !! Because it is not PhoneCallMode or speech driver is not ready, return NULL");
AL_UNLOCK(mLock);
return NULL;
}
pCaptureHandler = new AudioALSACaptureHandlerVoice(stream_attribute_target);
当前正在通话中，且音频源为 VOICE_CALL、VOICE_UPLINK、VOICE_DOWNLINK，初始化 AudioALSACaptureHandlerVoice 并 return
回到刚刚 AudioALSAStreamIn::open 中，往下顺序调用 mCaptureHandler->open();
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureHandlerVoice.cpp
status_t AudioALSACaptureHandlerVoice::open() {
ALOGD("+%s(), input_device = 0x%x, input_source = 0x%x",
__FUNCTION__, mStreamAttributeTarget->input_device, mStreamAttributeTarget->input_source);
//+open(), input_device = 0x80000004, input_source = 0x3
//+open(), input_device = 0x80000080, input_source = 0x3
ASSERT(mCaptureDataClient == NULL);
AudioALSACaptureDataProviderBase *provider = NULL;//
#ifndef LEGACY_VOICE_RECORD
switch (mStreamAttributeTarget->input_source) {
#ifdef INCALL_DL_RECORD_DISABLED
default: {
provider = AudioALSACaptureDataProviderVoiceUL::getInstance();
break;
}
#else
case AUDIO_SOURCE_VOICE_DOWNLINK: {
provider = AudioALSACaptureDataProviderVoiceDL::getInstance();
break;
}
case AUDIO_SOURCE_VOICE_UPLINK: {
provider = AudioALSACaptureDataProviderVoiceUL::getInstance();
break;
}
case AUDIO_SOURCE_VOICE_CALL: {
provider = AudioALSACaptureDataProviderVoiceMix::getInstance();
break;
}
default: {
provider = AudioALSACaptureDataProviderVoiceUL::getInstance();
break;
}
#endif
}
#else
// legacy voice record
provider = AudioALSACaptureDataProviderVoice::getInstance();
#endif
#if defined(MTK_AURISYS_FRAMEWORK_SUPPORT)
mCaptureDataClient = new AudioALSACaptureDataClientAurisysNormal(provider, mStreamAttributeTarget, NULL);
#else
mCaptureDataClient = new AudioALSACaptureDataClient(provider, mStreamAttributeTarget);
#endif
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
来到这里终于有点意思了，看到根据不同的音频源，实例化对应的 AudioALSACaptureDataProvider
这里加了log打印，发现同时开启两个录音线程，音频源分别设置 VOICE_UPLINK VOICE_DOWNLINK，但打印的
都是最后一个，也就是前面的被覆盖了。继续往下将 provider 传参到 new AudioALSACaptureDataClient(provider, mStreamAttributeTarget);
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureDataClient.cpp
AudioALSACaptureDataClient::AudioALSACaptureDataClient(AudioALSACaptureDataProviderBase *pCaptureDataProvider, stream_attribute_t *stream_attribute_target) :
mCaptureDataProvider(pCaptureDataProvider),
mAudioSpeechEnhanceInfoInstance(AudioSpeechEnhanceInfo::getInstance()),
mStreamAttributeSource(mCaptureDataProvider->getStreamAttributeSource()),
mStreamAttributeTarget(stream_attribute_target),
mBliSrc(NULL),
mMicMute(false),
mMuteTransition(false),
mSPELayer(NULL),
mAudioALSAVolumeController(AudioVolumeFactory::CreateAudioVolumeController()),
mChannelRemixOp(CHANNEL_REMIX_NOP),
mBesRecTuningEnable(false),
//echoref+++
mCaptureDataProviderEchoRef(NULL),
mStreamAttributeSourceEchoRef(NULL),
mStreamAttributeTargetEchoRef(NULL),
mBliSrcEchoRef(NULL),
mBliSrcEchoRefBesRecord(NULL)
//echoref---
{
ALOGD("%s()", __FUNCTION__);
// init member struct
memset((void *)&mEchoRefRawDataBuf, 0, sizeof(mEchoRefRawDataBuf));
memset((void *)&mEchoRefSrcDataBuf, 0, sizeof(mEchoRefSrcDataBuf));
// raw data
memset((void *)&mRawDataBuf, 0, sizeof(mRawDataBuf));
mRawDataBuf.pBufBase = new char[kClientBufferSize];
mRawDataBuf.bufLen = kClientBufferSize;
mRawDataBuf.pRead = mRawDataBuf.pBufBase;
mRawDataBuf.pWrite = mRawDataBuf.pBufBase;
ASSERT(mRawDataBuf.pBufBase != NULL);
....
// init SRC
if (mStreamAttributeSource->sample_rate != mStreamAttributeTarget->sample_rate) {
ALOGD("sample_rate: %d => %d, num_channels: %d => %d, audio_format: 0x%x => 0x%x",
mStreamAttributeSource->sample_rate, mStreamAttributeTarget->sample_rate,
mStreamAttributeSource->num_channels, mStreamAttributeSource->num_channels,
mStreamAttributeSource->audio_format, mStreamAttributeTarget->audio_format);
SRC_PCM_FORMAT SrcFormat = mStreamAttributeTarget->audio_format == AUDIO_FORMAT_PCM_16_BIT ? SRC_IN_Q1P15_OUT_Q1P15 : SRC_IN_Q1P31_OUT_Q1P31;
mBliSrc = newMtkAudioSrc(
mStreamAttributeSource->sample_rate, mStreamAttributeSource->num_channels,
mStreamAttributeTarget->sample_rate, mStreamAttributeSource->num_channels,
SrcFormat);
mBliSrc->open();
}
if (mStreamAttributeTarget->BesRecord_Info.besrecord_enable) {
//move CheckNeedBesRecordSRC to here for mStreamAttributeSource info
CheckNeedBesRecordSRC();
}
CheckChannelRemixOp();
//debug++
bTempDebug = mAudioSpeechEnhanceInfoInstance->GetDebugStatus();
//debug--
}
一顿各种初始化完成后，朋友们我们终于要开始录音了。 mBliSrc->open();
回到刚刚对应的 Provider，这里以 AudioALSACaptureDataProviderVoiceUL 为例
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureDataProviderVoiceUL.cpp
status_t AudioALSACaptureDataProviderVoiceUL::open() {
ALOGD("%s()", __FUNCTION__);
ASSERT(mEnable == false);
SpeechDataProcessingHandler::getInstance()->getStreamAttributeSource(SpeechDataProcessingHandler::VOICE_UL_CALLER, &mStreamAttributeSource);
mPeriodBufferSize = getPeriodBufSize(&mStreamAttributeSource, UPLINK_NORMAL_LATENCY_MS);
mRawDataRingBuf.bufLen = mPeriodBufferSize * 4;
/* Setup ringbuf */
mRawDataRingBuf.pBufBase = new char[mRawDataRingBuf.bufLen];
mRawDataRingBuf.pRead = mRawDataRingBuf.pBufBase;
mRawDataRingBuf.pWrite = mRawDataRingBuf.pBufBase;
mRawDataRingBuf.pBufEnd = mRawDataRingBuf.pBufBase + mRawDataRingBuf.bufLen;
ALOGD("%s(), mStreamAttributeSource: audio_format = %d, num_channels = %d, audio_channel_mask = %x, sample_rate = %d, periodBufferSize = %d\n",
__FUNCTION__, mStreamAttributeSource.audio_format, mStreamAttributeSource.num_channels, mStreamAttributeSource.audio_channel_mask, mStreamAttributeSource.sample_rate, mPeriodBufferSize);
mEnable = true;
OpenPCMDump(LOG_TAG);
return SpeechDataProcessingHandler::getInstance()->recordOn(RECORD_TYPE_UL);
}
这里 SpeechDataProcessingHandler 很关键，从字面理解为处理语音数据 handler
来看下 SpeechDataProcessingHandler::getInstance() 都干了啥
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\SpeechDataProcessingHandler.cpp
SpeechDataProcessingHandler::SpeechDataProcessingHandler() {
ALOGD("+%s()", __FUNCTION__);
// Init thread resources
mStopThreadFlag = false;
mBliSrcUL = NULL;
mBliSrcDL = NULL;
mSrcSampleRateUL = 0;
mSrcSampleRateDL = 0;
mSpeechRecordOn = false;
int ret;
ret = pthread_cond_init(&mSpeechDataNotifyEvent, NULL);
if (ret != 0) {
ALOGE("mSpeechDataNotifyEvent create fail!!!");
}
ret = pthread_mutex_init(&mSpeechDataNotifyMutex, NULL);
if (ret != 0) {
ALOGE("nSpeechDataNotifyMutex create fail!!!");
}
ret = pthread_create(&mSpeechDataProcessingThread, NULL, SpeechDataProcessingHandler::threadLoop, (void *)this);
if (ret != 0) {
ALOGE("mSpeechDataProcessingThread create fail!!!");
} else {
ALOGD("mSpeechDataProcessingThread = %lu created", mSpeechDataProcessingThread);
}
ALOGD("-%s()", __FUNCTION__);
}
搞了一个线程一直在读取录音数据 threadLoop() 这个方法尤其重要，加log看具体的执行过程
收到数据后在 processSpeechPacket() 中处理
刚刚的 AudioALSACaptureDataProviderVoiceUL::open() 中最后一句调用 recordOn(RECORD_TYPE_UL)
RECORD_TYPE_UL 类型定义在 SpeechType.h 中
vendor\mediatek\proprietary\hardware\audio\common\include\SpeechType.h
enum record_type_t {
RECORD_TYPE_UL = 0,
RECORD_TYPE_DL = 1,
RECORD_TYPE_MIX = 2
};
static int mUserCounter = 0;
status_t SpeechDataProcessingHandler::recordOn(record_type_t type __unused) {
ALOGD("+%s(),type_record = %d\n", __FUNCTION__, type);
AL_AUTOLOCK(speechDataProcessingHandlerLock);
mUserCounter++;
if (mUserCounter == 1) {
SpeechDriverFactory::GetInstance()->GetSpeechDriver()->RecordOn(RECORD_TYPE_MIX);
//SpeechDriverFactory::GetInstance()->GetSpeechDriver()->RecordOn(type);
ALOGD("%s(), First user, record on.\n", __FUNCTION__);
} else {
ALOGD("%s(), Record already on. user = %d\n", __FUNCTION__, mUserCounter);
}
ALOGD("-%s()\n", __FUNCTION__);
return NO_ERROR;
}
可以看到此处并没有使用刚刚传递的 RECORD_TYPE_UL，而是默认混合 RECORD_TYPE_MIX。是我太天真了，以为将此处修改为
type 就能正常录取 UL 和 DL。继续进入 SpeechDriver()->RecordOn(RECORD_TYPE_MIX)
vendor\mediatek\proprietary\hardware\audio\common\V3\speech_driver\SpeechDriverLAD.cpp
/*==============================================================================
* Recording Control
*============================================================================*/
status_t SpeechDriverLAD::RecordOn(record_type_t type_record) {
ALOGD("%s(), sample_rate = %d, channel = %d, type_record = %d, MSG_A2M_RECORD_RAW_PCM_ON", __FUNCTION__, mRecordSampleRateType, mRecordChannelType, type_record);
uint16_t param_16bit;
SetApSideModemStatus(RAW_RECORD_STATUS_MASK);
mRecordType = type_record;
pCCCI->SetPcmRecordType(type_record);
param_16bit = mRecordSampleRateType | (mRecordChannelType << 4);
return pCCCI->SendMessageInQueue(pCCCI->InitCcciMailbox(MSG_A2M_RECORD_RAW_PCM_ON, param_16bit, 0));
}
这样底层开启录音后，刚刚的 threadLoop(void *arg) 中将循环收取数据，然后根据 rawPcmDir 是 RECORD_TYPE_UL
RECORD_TYPE_DL RECORD_TYPE_MIX 判断对应 Provider 是否存在，存在调用 provideModemRecordDataToProvider(ringBuf)
将数据分发出去。
当调用 mMediaRecorderd.stop(); mMediaRecorder.reset(); mMediaRecorder.release(); 后，底层将结束录音。
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSAStreamIn.cpp
回到 AudioALSAStreamIn::close() 中
status_t AudioALSAStreamIn::close() {
// call close() only when mLock is locked.
ASSERT(AL_TRYLOCK(mLock) != 0);
ALOGD("%s()", __FUNCTION__);
status_t status = NO_ERROR;
if (mStandby == false) {
mStandby = true;
ASSERT(mCaptureHandler != NULL);
// close audio hardware
status = mCaptureHandler->close();
if (status != NO_ERROR) {
ALOGE("%s(), close() fail!!", __FUNCTION__);
}
ClosePCMDump();
// destroy playback handler
mStreamManager->destroyCaptureHandler(mCaptureHandler);
mCaptureHandler = NULL;
}
ASSERT(mCaptureHandler == NULL);
return status;
}
AudioALSACaptureHandlerVoice::close()
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureHandlerVoice.cpp
delete 释放指针 mCaptureDataClient
status_t AudioALSACaptureHandlerVoice::close() {
ALOGD("+%s()", __FUNCTION__);
ASSERT(mCaptureDataClient != NULL);
delete mCaptureDataClient;
ALOGD("-%s()", __FUNCTION__);
return NO_ERROR;
}
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureDataClient.cpp
AudioALSACaptureDataClient::~AudioALSACaptureDataClient() {
ALOGD("%s()", __FUNCTION__);
.....
if (mBliSrc != NULL) {
mBliSrc->close();
deleteMtkAudioSrc(mBliSrc);
mBliSrc = NULL;
}
//TODO: Sam, add here for temp
//BesRecord+++
StopBesRecord();
....
mBliSrc->close() 调用到 AudioALSACaptureDataProviderVoiceUL::close()
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\AudioALSACaptureDataProviderVoiceUL.cpp
status_t AudioALSACaptureDataProviderVoiceUL::close() {
ALOGD("%s()", __FUNCTION__);
mEnable = false;
delete[] mRawDataRingBuf.pBufBase;
memset(&mRawDataRingBuf, 0, sizeof(mRawDataRingBuf));
ClosePCMDump();
return SpeechDataProcessingHandler::getInstance()->recordOff(RECORD_TYPE_UL);
}
vendor\mediatek\proprietary\hardware\audio\common\V3\aud_drv\SpeechDataProcessingHandler.cpp
status_t SpeechDataProcessingHandler::recordOff(record_type_t type __unused) {
ALOGD("+%s()\n", __FUNCTION__);
AL_AUTOLOCK(speechDataProcessingHandlerLock);
mUserCounter--;
if (mUserCounter == 0) {
SpeechDriverFactory::GetInstance()->GetSpeechDriver()->RecordOff(RECORD_TYPE_MIX);
//SpeechDriverFactory::GetInstance()->GetSpeechDriver()->RecordOff(type);
ALOGD("%s(), No user, record off!\n", __FUNCTION__);
} else {
ALOGD("%s(), Record is still using. user = %d\n", __FUNCTION__, mUserCounter);
}
ALOGD("-%s()\n", __FUNCTION__);
#endif
return NO_ERROR;
}
vendor\mediatek\proprietary\hardware\audio\common\V3\speech_driver\SpeechDriverLAD.cpp
status_t SpeechDriverLAD::RecordOn(record_type_t type_record) {
ALOGD("%s(), sample_rate = %d, channel = %d, type_record = %d, MSG_A2M_RECORD_RAW_PCM_ON", __FUNCTION__, mRecordSampleRateType, mRecordChannelType, type_record);
uint16_t param_16bit;
SetApSideModemStatus(RAW_RECORD_STATUS_MASK);
mRecordType = type_record;
pCCCI->SetPcmRecordType(type_record);
param_16bit = mRecordSampleRateType | (mRecordChannelType << 4);
return pCCCI->SendMessageInQueue(pCCCI->InitCcciMailbox(MSG_A2M_RECORD_RAW_PCM_ON, param_16bit, 0));
}
到此流程基本就分析完了。最终通过修改 audio hal 层 + AudioRecord 录取 VOICE_CALL 通道方案完美解决。
其它
过滤关键tag值
AudioALSAStream AudioALSAStreamIn AudioALSAHardware AudioALSACaptureDataProvider
2019-01-01 10:09:57.325 280-954/? D/AudioALSAHardware: +setParameters(): A2dpSuspended=true
2019-01-01 10:09:57.325 280-954/? W/AudioALSAHardware: setParameters(), still have param.size() = 1, remain param = "A2dpSuspended=true"
2019-01-01 10:09:57.325 280-954/? D/AudioALSAHardware: -setParameters(): A2dpSuspended=true
2019-01-01 10:09:57.953 280-954/? D/AudioALSAHardware: +routing createAudioPatch Mixer->1
2019-01-01 10:09:57.955 280-954/? D/AudioALSAHardwareResourceManager: +startInputDevice(), new_device = 0x80000004, mInputDevice = 0x0, mStartInputDeviceCount = 0, mMicInverse = 0, mNumPhoneMicSupport = 2
2019-01-01 10:09:57.957 280-954/? D/AudioALSAHardwareResourceManager: +startOutputDevice(), new_devices = 0x1, mOutputDevices = 0x0, mStartOutputDevicesCount = 0 SampleRate = 16000
2019-01-01 10:09:58.042 280-954/? D/AudioALSAHardware: +routing createAudioPatch 80000004->Mixer Src 3
2019-01-01 10:09:58.768 280-954/? D/AudioALSAHardware: +routing createAudioPatch Mixer->2
2019-01-01 10:09:58.790 280-954/? D/AudioALSAHardwareResourceManager: +stopOutputDevice(), mOutputDevices = 0x1, mStartOutputDevicesCount = 1
2019-01-01 10:09:58.791 280-954/? D/AudioALSAHardwareResourceManager: +stopInputDevice(), mInputDevice = 0x80000004, stop_device = 0x80000004, mStartInputDeviceCount = 1, mMicInverse = 0, mNumPhoneMicSupport = 2
2019-01-01 10:09:58.794 280-954/? D/AudioALSAHardwareResourceManager: +startInputDevice(), new_device = 0x80000080, mInputDevice = 0x0, mStartInputDeviceCount = 0, mMicInverse = 0, mNumPhoneMicSupport = 2
2019-01-01 10:09:58.796 280-954/? D/AudioALSAHardwareResourceManager: +startOutputDevice(), new_devices = 0x2, mOutputDevices = 0x0, mStartOutputDevicesCount = 0 SampleRate = 16000
2019-01-01 10:09:58.991 280-954/? D/AudioALSAHardware: setVoiceVolume(), volume = 0.258523, mUseTuningVolume = 1
2019-01-01 10:09:59.000 280-954/? D/AudioALSAHardware: +routing createAudioPatch Mixer->2
2019-01-01 10:09:59.027 280-954/? D/AudioALSAHardware: +routing createAudioPatch 80000080->Mixer Src 3
2019-01-01 10:09:59.448 280-954/? D/AudioALSAHardware: +routing createAudioPatch Mixer->1
2019-01-01 10:09:59.470 280-954/? D/AudioALSAHardwareResourceManager: +stopOutputDevice(), mOutputDevices = 0x2, mStartOutputDevicesCount = 1
2019-01-01 10:09:59.479 280-954/? D/AudioALSAHardwareResourceManager: +stopInputDevice(), mInputDevice = 0x80000080, stop_device = 0x80000080, mStartInputDeviceCount = 1, mMicInverse = 0, mNumPhoneMicSupport = 2
2019-01-01 10:09:59.482 280-954/? D/AudioALSAHardwareResourceManager: +startInputDevice(), new_device = 0x80000004, mInputDevice = 0x0, mStartInputDeviceCount = 0, mMicInverse = 0, mNumPhoneMicSupport = 2
2019-01-01 10:09:59.484 280-954/? D/AudioALSAHardwareResourceManager: +startOutputDevice(), new_devices = 0x1, mOutputDevices = 0x0, mStartOutputDevicesCount = 0 SampleRate = 16000
2019-01-01 10:09:59.607 280-954/? D/AudioALSAHardware: setVoiceVolume(), volume = 0.230409, mUseTuningVolume = 1
2019-01-01 10:09:59.610 280-954/? D/AudioALSAHardware: +routing createAudioPatch Mixer->1
2019-01-01 10:09:59.627 280-954/? D/AudioALSAHardware: +routing createAudioPatch 80000004->Mixer Src 3
2019-01-01 10:10:07.965 280-412/? D/AudioALSAHardware: +setParameters(): A2dpSuspended=false
