背景需求
做語音識別和答錄功能的app大都需要將通話錄音上下行VOICE_DOWNLINK/VOICE_UPLINK單獨分離實時轉義識別，
兩個解決方向:
1、AudioRecord 方法的立體聲錄制PCM資料混合音源了，二進制檔案無法區分哪些資料是左聲道，哪些是右聲道，
通過修改AudioRecord立體聲PCM錄制方案，每8位一組資料，左聲道8位在前，右聲道8位在后，
2、使用兩個實體同時運行MediaRecorder的VOICE_UPLINK，VOICE_DOWNLINK錄制上行或下行資料，且資料不重用，
基于這兩個方向我們都需要去看 Audio HAL 代碼，
最終結論
一開始從方向二出發，由于安卓本身的錄音機制同一時間只允許一個應用錄音，首先需要破解這個機制，
Android Audio - 支持多應用同時錄音_Android8.1修改方法 參考這個修改后，確實可以同時 start 兩個 MediaRecorder，但最終問題是不管如何設定 AudioSource
VOICE_DOWNLINK 和 VOICE_UPLINK，保存錄音檔案內容都是一樣的，依舊是上下行混在一起，最終發現底層 buffer 是同一個，所以
保存的資料一樣，這塊嘗試改了好幾天，沒啥進展，具體分析程序看下文，
再嘗試方向一，通過 AudioRecord 獲取原始 pcm 資料按 8 位分離，最終通過這個搞定了！
分析程序
主要的代碼目錄位于
vendor/mediatek/proprietary/hardware/audio/common/V3/
framework 相關的分析這里就不寫了，根據這篇查看
Android 5.1 Audio HAL分析
這里從 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;
}
//這里加了log列印，發現走的 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;
.....
進入 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);
}
繼續跟進 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，最后干活的將是這個物件,這里將 pAudioALSAStreamIn 回傳到 audio_hw_hal.cpp 中
順序呼叫 in->legacy_in->setParameters(param.toString())，其實呼叫 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);
......
關鍵的地方在 mStreamAttributeTarget.input_source 這個值即上層 setAudioSource 傳遞對應 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;
一系列引數設定完成后，將開始 read 資料，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);
}
最侄訓是 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();
}
看到這 open() 方法，這個方法有點東西
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 將剛剛的引數一并傳遞
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);
當前正在通話中，且音頻源為 VOICE_CALL、VOICE_UPLINK、VOICE_DOWNLINK，初始化 AudioALSACaptureHandlerVoice 并 return
回到剛剛 AudioALSAStreamIn::open 中，往下順序呼叫 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;
}
來到這里終于有點意思了，看到根據不同的音頻源，實體化對應的 AudioALSACaptureDataProvider
這里加了log列印，發現同時開啟兩個錄音執行緒，音頻源分別設定 VOICE_UPLINK VOICE_DOWNLINK，但列印的
都是最后一個，也就是前面的被覆寫了，繼續往下將 provider 傳參到 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--
}
一頓各種初始化完成后，朋友們我們終于要開始錄音了， mBliSrc->open();
回到剛剛對應的 Provider，這里以 AudioALSACaptureDataProviderVoiceUL 為例
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);
}
這里 SpeechDataProcessingHandler 很關鍵，從字面理解為處理語音資料 handler
來看下 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__);
}
搞了一個執行緒一直在讀取錄音資料 threadLoop() 這個方法尤其重要，加log看具體的執行程序
收到資料后在 processSpeechPacket() 中處理
剛剛的 AudioALSACaptureDataProviderVoiceUL::open() 中最后一句呼叫 recordOn(RECORD_TYPE_UL)
RECORD_TYPE_UL 型別定義在 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;
}
可以看到此處并沒有使用剛剛傳遞的 RECORD_TYPE_UL，而是默認混合 RECORD_TYPE_MIX，是我太天真了，以為將此處修改為
type 就能正常錄取 UL 和 DL，繼續進入 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));
}
這樣底層開啟錄音后，剛剛的 threadLoop(void *arg) 中將回圈收取資料，然后根據 rawPcmDir 是 RECORD_TYPE_UL
RECORD_TYPE_DL RECORD_TYPE_MIX 判斷對應 Provider 是否存在，存在呼叫 provideModemRecordDataToProvider(ringBuf)
將資料分發出去，
當呼叫 mMediaRecorderd.stop(); mMediaRecorder.reset(); mMediaRecorder.release(); 后，底層將結束錄音，
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 釋放指標 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() 呼叫到 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，最終通過修改 audio hal 層 + AudioRecord 錄取 VOICE_CALL 通道方案完美解決，
其它
過濾關鍵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
Android 5.1 Audio HAL分析
Android 錄音實作（AudioRecord）
音頻framework與中間層分析
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標籤：其他
上一篇：2021-08-13