Android音频子系统（十） |
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Android framework中的代码每个平台基本都是大同小异，只有Hal上代码才是厂商特制，每个平台都不相同，这里以MTK平台为例，记录下MTK平台Hal audio音频录音代码流程。 调用大致流程如下 AudioALSAHardwa: openInputStream() AudioALSAStreamManager: openInputStream() AudioALSAStreamIn: AudioALSAStreamIn //new AudioALSAStreamIn AudioALSAStreamIn: set() //done, devices: 0x80000004, flags: 0, acoustics: 0x0, format: 0x1, sampleRate: 48000/48000, num_channels: 12/2, buffer_size: 3840, tempDebugflag: 0 AudioALSAStreamIn: checkOpenStreamFormat AudioALSAStreamIn: checkOpenStreamChannels AudioALSAStreamIn: checkOpenStreamSampleRate AudioALSAHardware: createAudioPatch() AudioALSAStreamManager: setParameters() //IOport = 70, keyValuePairs = input_source=1;routing=-2147483644 AudioALSAStreamIn: setParameters() AudioALSAStreamManager: routingInputDevice()// input_device: 0x80000004 => 0x80000004 AudioALSAStreamIn: read() AudioALSAStreamIn: open() AudioALSAStreamManager: createCaptureHandler() AudioALSAStreamManager: ulStreamAttributeTargetCustomization AudioALSACaptureDataProviderNormal: AudioALSACaptureDataProviderNormal() AudioALSACaptureDataProviderBase: AudioALSACaptureDataProviderBase() AudioALSACaptureHandlerNormal: AudioALSACaptureHandlerNormal() AudioALSACaptureHandlerNormal: init() AudioALSACaptureHandlerNormal: open()//input_device = 0x80000004, input_source = 0x1, sample_rate=48000, num_channels=2 //里面配置period_count channels... AudioALSACaptureDataProviderBase: AudioALSACaptureDataProviderBase() AudioALSACaptureDataProviderDspRaw: AudioALSACaptureDataProviderDspRaw() AudioALSACaptureDataClientAurisysNormal: AudioALSACaptureDataClientAurisysNormal(+) //mCaptureDataClient = new AudioALSACaptureDataClientAurisysNormal AudioALSACaptureDataProviderBase: configStreamAttribute() //audio_mode: 0 => 0, input_device: 0x0 => 0x80000004, flag: 0x0 => 0x0, input_source: 0->1, output_device: 0x0 => 0x2, sample_rate: 0 => 48000, period_us: 0 => 0, DSP out sample_rate: 0 => 48000 AudioALSACaptureDataProviderBase: attach AudioALSACaptureDataProviderDspRaw: open(+) //里面配置format、channels AudioALSADeviceConfigManager: ApplyDeviceTurnonSequenceByName() DeviceName = ADDA_TO_CAPTURE1 descriptor->DeviceStatusCounte = 0 AudioALSACaptureDataProviderBase: enablePmicInputDevice AudioALSAHardwareResourceManager: +startInputDevice() AudioALSADeviceConfigManager: ApplyDeviceTurnonSequenceByName() DeviceName = builtin_Mic_DualMic descriptor->DeviceStatusCounte = 0 AudioALSACaptureDataProviderBase: getInputSampleRate() AudioMTKGainController: +SetCaptureGain() //mode=0, source=1, input device=0x80000004, output device=0x2 AudioMTKGainController: ApplyMicGain() AudioALSACaptureDataProviderDspRaw: openApHwPcm(), mPcm = 0xf2b55260 AudioDspStreamManager: addCaptureDataProvider() //添加到mCaptureDataProviderVector AudioDspStreamManager: openCaptureDspHwPcm(), mDspHwPcm = 0xf2b55340 pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderDspRaw::readThread, (void *)this); //创建线程 AudioALSACaptureDataProviderDspRaw: +readThread() AudioALSACaptureDataProviderBase: waitPcmStart AudioALSACaptureDataProviderBase: pcm_start AudioALSACaptureDataProviderBase: pcmRead AudioALSACaptureDataProviderBase: provideCaptureDataToAllClients AudioALSACaptureDataClientAurisysNormal: copyCaptureDataToClient() //提供数据给client，然后继续Read，如此循环 pthread_create(&hProcessThread, NULL, AudioALSACaptureDataClientAurisysNormal::processThread, (void *)this); //创建线程 AudioALSACaptureDataClientAurisysNormal: processThread AudioALSACaptureHandlerNormal: read() bytes = mCaptureDataClient->read(buffer, bytes); 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) { //检查参数配置 if (format == NULL || channels == NULL || sampleRate == NULL || status == NULL) { ALOGE("%s(), NULL pointer!! format = %p, channels = %p, sampleRate = %p, status = %p", __FUNCTION__, format, channels, sampleRate, status); if (status != NULL) { *status = INVALID_OPERATION; } return NULL; } ALOGD("%s(), devices = 0x%x, format = 0x%x, channels = 0x%x, sampleRate = %d, status = %d, acoustics = 0x%x, input_flag 0x%x", __FUNCTION__, devices, *format, *channels, *sampleRate, *status, acoustics, input_flag); // create stream in AudioALSAStreamIn *pAudioALSAStreamIn = new AudioALSAStreamIn(); //主要set一些参数 pAudioALSAStreamIn->set(devices, format, channels, sampleRate, status, acoustics, input_flag); pAudioALSAStreamIn->setIdentity(mStreamInIndex); mStreamInVector.add(mStreamInIndex, pAudioALSAStreamIn);//添加到mStreamInVector return pAudioALSAStreamIn; } status_t AudioALSAStreamIn::set( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, audio_in_acoustics_t acoustics, uint32_t flags) { // check format if (checkOpenStreamFormat(static_cast(devices), format) == false) { *status = BAD_VALUE; } // check channel mask if (checkOpenStreamChannels(static_cast(devices), channels) == false) { *status = BAD_VALUE; } // check sample rate if (checkOpenStreamSampleRate(static_cast(devices), sampleRate) == false) { *status = BAD_VALUE; } // config stream attribute，主要配置参数 if (*status == NO_ERROR) { // format mStreamAttributeTarget.audio_format = static_cast(*format); // channel mStreamAttributeTarget.audio_channel_mask = static_cast(*channels); mStreamAttributeTarget.num_channels = popcount(*channels); // sample rate mStreamAttributeTarget.sample_rate = *sampleRate; // devices mStreamAttributeTarget.input_device = static_cast(devices); // acoustics flags mStreamAttributeTarget.acoustics_mask = static_cast(acoustics); } }上层服务会通过setParameters下发参数给HAL，这里下发的参数是：IOport = 70, keyValuePairs = input_source=1;routing=-2147483644 status_t AudioALSAStreamIn::setParameters(const String8 &keyValuePairs) { /// routing if (param.getInt(keyRouting, value) == NO_ERROR) { status = mStreamManager->routingInputDevice(this, mStreamAttributeTarget.input_device, inputdevice); } } //setParameters里面解析参数，本次参数主要是要route操作 status_t AudioALSAStreamManager::routingInputDevice(AudioALSAStreamIn *pAudioALSAStreamIn, const audio_devices_t current_input_device, audio_devices_t input_device) { //如果要route的device和上次的一样，则直接返回 if (input_device == AUDIO_DEVICE_NONE || input_device == current_input_device) { ALOGW("-%s(), input_device(0x%x) is AUDIO_DEVICE_NONE(0x%x) or current_input_device(0x%x), return", __FUNCTION__, input_device, AUDIO_DEVICE_NONE, current_input_device); return NO_ERROR; } //如果要切换设备，先suspend挂起 setAllInputStreamsSuspend(true, false); standbyAllInputStreams(); if (mStreamInVector.size() > 0) { for (size_t i = 0; i if (pAudioALSAStreamIn == mStreamInVector[i]) { status = mStreamInVector[i]->routing(input_device); ASSERT(status == NO_ERROR); } } else { //这里就是实际route操作 status = mStreamInVector[i]->routing(input_device); ASSERT(status == NO_ERROR); } } } //suspend恢复 setAllInputStreamsSuspend(false, false); }接下来就是最重要的read调用了，上层服务会调用read函数来读取录音数据： ssize_t AudioALSAStreamIn::read(void *buffer, ssize_t bytes) { /// check open if (mStandby == true) { status = open(); } }open是放在read里面来做的： status_t AudioALSAStreamIn::open() { if (mStandby == true) { // create capture handler ASSERT(mCaptureHandler == NULL); mCaptureHandler = mStreamManager->createCaptureHandler(&mStreamAttributeTarget); if (mCaptureHandler == NULL) { status = BAD_VALUE; return status; } // open audio hardware status = mCaptureHandler->open(); mStandby = false; } return status; }先创建CaptureHandler，不同的设备和源会创建不同的CaptureHandler执行类，之后open对应的CaptureHandler。 可以留一下createCaptureHandler的逻辑，主要还是对设备做区分： AudioALSACaptureHandlerBase *AudioALSAStreamManager::createCaptureHandler( // Init input stream attribute here配置stream_attribute_target stream_attribute_target->audio_mode = mAudioMode; // set mode to stream attribute for mic gain setting stream_attribute_target->output_devices = current_output_devices; // set output devices to stream attribute for mic gain setting and BesRecord parameter stream_attribute_target->micmute = mMicMute; //客制化通路，设置APP2场景可以使得normal 录音也走voip，这样就可以跑aec算法消除回声 /* StreamAttribute customization for scene */ ulStreamAttributeTargetCustomization(stream_attribute_target); //语音唤醒 if (stream_attribute_target->input_source == AUDIO_SOURCE_HOTWORD) { if (mAudioALSAVoiceWakeUpController->getVoiceWakeUpEnable() == false) { mAudioALSAVoiceWakeUpController->setVoiceWakeUpEnable(true); } if (mVoiceWakeUpNeedOn == true) { mAudioALSAVoiceWakeUpController->SeamlessRecordEnable(); } pCaptureHandler = new AudioALSACaptureHandlerVOW(stream_attribute_target); } else if (stream_attribute_target->input_source == AUDIO_SOURCE_VOICE_UNLOCK || stream_attribute_target->input_source == AUDIO_SOURCE_ECHO_REFERENCE) { pCaptureHandler = new AudioALSACaptureHandlerSyncIO(stream_attribute_target); //实网通话 } else if (isPhoneCallOpen() == true) { pCaptureHandler = new AudioALSACaptureHandlerVoice(stream_attribute_target); //客制化AEC场景 } else if ((stream_attribute_target->NativePreprocess_Info.PreProcessEffect_AECOn == true) || (stream_attribute_target->input_source == AUDIO_SOURCE_VOICE_COMMUNICATION) || (stream_attribute_target->input_source == AUDIO_SOURCE_CUSTOMIZATION1) || //MagiASR enable AEC (stream_attribute_target->input_source == AUDIO_SOURCE_CUSTOMIZATION2)) { //Normal REC with AEC AudioALSAHardwareResourceManager::getInstance()->setHDRRecord(false); // turn off HRD record for VoIP switch (stream_attribute_target->input_device) { case AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET: { if (stream_attribute_target->output_devices & AUDIO_DEVICE_OUT_ALL_SCO) { pCaptureHandler = new AudioALSACaptureHandlerAEC(stream_attribute_target); } else { pCaptureHandler = new AudioALSACaptureHandlerBT(stream_attribute_target); } break; } case AUDIO_DEVICE_IN_USB_DEVICE: case AUDIO_DEVICE_IN_USB_HEADSET: #if defined(MTK_AURISYS_FRAMEWORK_SUPPORT) pCaptureHandler = new AudioALSACaptureHandlerAEC(stream_attribute_target); #else pCaptureHandler = new AudioALSACaptureHandlerUsb(stream_attribute_target); #endif break; default: { if (isAdspOptionEnable() && ((isCaptureOffload(stream_attribute_target) && !isIEMsOn && !AudioALSACaptureDataProviderNormal::getInstance()->getNormalOn()) || isBleInputDevice(stream_attribute_target->input_device))) { pCaptureHandler = new AudioALSACaptureHandlerDsp(stream_attribute_target); } else { pCaptureHandler = new AudioALSACaptureHandlerAEC(stream_attribute_target); } break; } }//switch } else { switch (stream_attribute_target->input_device) { case AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET: { pCaptureHandler = new AudioALSACaptureHandlerBT(stream_attribute_target); break; } case AUDIO_DEVICE_IN_USB_DEVICE: case AUDIO_DEVICE_IN_USB_HEADSET: pCaptureHandler = new AudioALSACaptureHandlerUsb(stream_attribute_target); break; case AUDIO_DEVICE_IN_BUILTIN_MIC: case AUDIO_DEVICE_IN_BACK_MIC: case AUDIO_DEVICE_IN_WIRED_HEADSET: case AUDIO_DEVICE_IN_BLE_HEADSET: case AUDIO_DEVICE_IN_BUS: default: { if (AudioSmartPaController::getInstance()->isInCalibration()) { pCaptureHandler = new AudioALSACaptureHandlerNormal(stream_attribute_target); break; } if (isAdspOptionEnable() && !(AUDIO_INPUT_FLAG_MMAP_NOIRQ & stream_attribute_target->mAudioInputFlags) &&((isCaptureOffload(stream_attribute_target) && !isIEMsOn && !AudioALSACaptureDataProviderNormal::getInstance()->getNormalOn()) || isBleInputDevice(stream_attribute_target->input_device))) { if (isPhoneCallOpen() == true) { pCaptureHandler = new AudioALSACaptureHandlerVoice(stream_attribute_target); } else { pCaptureHandler = new AudioALSACaptureHandlerDsp(stream_attribute_target); } } else { pCaptureHandler = new AudioALSACaptureHandlerNormal(stream_attribute_target); } break; } }//switch } // save capture handler object in vector mCaptureHandlerVector.add(mCaptureHandlerIndex, pCaptureHandler); return pCaptureHandler; }createCaptureHandler里面逻辑还是比较多的，这里我们分析AUDIO_DEVICE_IN_BUILTIN_MIC，也就是普通MIC录音场景，也就是AudioALSACaptureHandlerNormal。 status_t AudioALSACaptureHandlerNormal::open() { if (!AudioSmartPaController::getInstance()->isInCalibration()) { if (isAdspOptionEnable() && (AudioDspStreamManager::getInstance()->getDspRawInHandlerEnable(mStreamAttributeTarget->mAudioInputFlags) > 0) && (AudioDspStreamManager::getInstance()->getDspInHandlerEnable(mStreamAttributeTarget->mAudioInputFlags) > 0) && !isIEMsOn && !AudioALSACaptureDataProviderNormal::getInstance()->getNormalOn()) { mCaptureDataClient = new AudioALSACaptureDataClientAurisysNormal(AudioALSACaptureDataProviderDspRaw::getInstance(), mStreamAttributeTarget, NULL); // NULL: w/o AEC } else { mCaptureDataClient = new AudioALSACaptureDataClientAurisysNormal(AudioALSACaptureDataProviderNormal::getInstance(), mStreamAttributeTarget, NULL); // NULL: w/o AEC } } else { mCaptureDataClient = new AudioALSACaptureDataClientAurisysNormal(AudioALSACaptureDataProviderEchoRefExt::getInstance(), mStreamAttributeTarget, NULL); // NULL: w/o AEC } }这里主要走的是AudioALSACaptureDataClientAurisysNormal，DataProvider是AudioALSACaptureDataProviderDspRaw（这个后面表述）。 继续看下AudioALSACaptureDataClientAurisysNormal AudioALSACaptureDataClientAurisysNormal::AudioALSACaptureDataClientAurisysNormal( AudioALSACaptureDataProviderBase *pCaptureDataProvider, stream_attribute_t *stream_attribute_target, AudioALSACaptureDataProviderBase *pCaptureDataProviderEchoRef) { // config attribute for input device mCaptureDataProvider->configStreamAttribute(mStreamAttributeTarget); // attach client to capture data provider (after data buf ready) mCaptureDataProvider->attach(this); // get latency (for library, but not data provider) mLatency = (IsLowLatencyCapture()) ? UPLINK_LOW_LATENCY_MS : UPLINK_NORMAL_LATENCY_MS; if (mAudioALSAVolumeController != NULL) { mAudioALSAVolumeController->SetCaptureGain(mStreamAttributeTarget->audio_mode, mStreamAttributeTarget->input_source, mStreamAttributeTarget->input_device, mStreamAttributeTarget->output_devices); } // create lib manager,aurisys用于dsp里面传递算法 CreateAurisysLibManager(); // depop，MTK在录音时会丢弃前面一段数据避免杂音 drop_ms = getDropMs(mStreamAttributeTarget); if (drop_ms) { if ((drop_ms % mLatency) != 0) { // drop data size need to align interrupt rate drop_ms = ((drop_ms / mLatency) + 1) * mLatency; // cell() } mDropPopSize = (audio_bytes_per_sample(mStreamAttributeTarget->audio_format) * mStreamAttributeTarget->num_channels * mStreamAttributeTarget->sample_rate * drop_ms) / 1000; } // processThread ret = pthread_create(&hProcessThread, NULL, AudioALSACaptureDataClientAurisysNormal::processThread, (void *)this); } 配置AttributeDataProvider attach，提供录音数据计算Latency ，录音间隔延时设置gain增益创建aurisys，算法会用到计算录音开头需要丢弃的数据大小mDropPopSize，避免录音开头有杂音创建线程去读取录音数据 void *AudioALSACaptureDataClientAurisysNormal::processThread(void *arg) { /* process thread created */ client->mProcessThreadLaunched = true; /* get buffer address */ raw_ul = &client->mRawDataBuf;//ringbuf地址 processed = &client->mProcessedDataBuf; while (client->mEnable == true) { data_count_raw_ul = audio_ringbuf_count(raw_ul); // data not reary, wait data if ((data_count_raw_ul mRawDataPeriodBufSize) || (client->IsAECEnable() == true && ((client->mIsEchoRefDataSync == false && client->isNeedSkipSyncEchoRef() == false) || data_count_raw_aec mEchoRefDataPeriodBufSize))) { wait_result = AL_WAIT_MS(client->mRawDataBufLock, MAX_PROCESS_DATA_WAIT_TIME_OUT_MS); } // copy data audio_pool_buf_copy_from_ringbuf(ul_in, raw_ul, client->mRawDataPeriodBufSize); aurisys_process_ul_only(manager, ul_in, ul_out, ul_aec); // depop if (client->mDropPopSize > 0) { ALOGV("data_count %u, mDropPopSize %u, %dL", data_count, client->mDropPopSize, __LINE__); if (data_count >= client->mDropPopSize) { audio_ringbuf_drop_data(&ul_out->ringbuf, client->mDropPopSize); data_count -= client->mDropPopSize; client->mDropPopSize = 0; } else { audio_ringbuf_drop_data(&ul_out->ringbuf, data_count); client->mDropPopSize -= data_count; data_count = 0; } } // copy to processed buf and signal read() audio_ringbuf_copy_from_linear(processed, effect_buf, data_count); } pthread_exit(NULL); return NULL; }线程里面会不停循环的从client->mRawDataBuf这个ringbuf的里面copy数据出来，这个数据也就是录音数据，同时会通过aurisys ul过一道算法处理，然后通过之前在AudioALSACaptureDataClientAurisysNormal算出的mDropPopSize，丢弃一段数据避免录音杂音。 那么问题来了，为什么ringbuf里面会有录音数据呢？是谁提供的呢？ 这就是之前说的：DataProvider是AudioALSACaptureDataProviderDspRaw 之前在AudioALSACaptureDataClientAurisysNormal::AudioALSACaptureDataClientAurisysNormal中有调用：mCaptureDataProvider->attach(this); mCaptureDataProvider也就是传入的AudioALSACaptureDataProviderDspRaw！ void AudioALSACaptureDataProviderBase::attach(IAudioALSACaptureDataClient *pCaptureDataClient) { mCaptureDataClientVector.add(pCaptureDataClient->getIdentity(), pCaptureDataClient); size = (uint32_t)mCaptureDataClientVector.size(); // open pcm interface when 1st attach if (size == 1) { mOpenIndex++; open(); } else { if (!hasLowLatencyCapture && pCaptureDataClient->IsLowLatencyCapture()) { // update HW interrupt rate by HW sample rate updateReadSize(getPeriodBufSize(pStreamAttr, UPLINK_NORMAL_LATENCY_MS) * lowLatencyMs / UPLINK_NORMAL_LATENCY_MS); if (mCaptureDataProviderType != CAPTURE_PROVIDER_DSP) { mHardwareResourceManager->setULInterruptRate(mStreamAttributeSource.sample_rate * lowLatencyMs / 1000); } else if (isAdspOptionEnable()) { AudioDspStreamManager::getInstance()->UpdateCaptureDspLatency(); } } enablePmicInputDevice(true); } }第一次录音时，size必定等于1，所以会进入open函数： status_t AudioALSACaptureDataProviderDspRaw::open() { unsigned int feature_id = CAPTURE_RAW_FEATURE_ID; //发送给DSP需要录音原数据 mAudioMessengerIPI->registerAdspFeature(feature_id); if (AudioALSAHardwareResourceManager::getInstance()->getNumPhoneMicSupport() > 2 && mStreamAttributeSource.input_device != AUDIO_DEVICE_IN_WIRED_HEADSET) { mApTurnOnSequence = AUDIO_CTL_ADDA_TO_CAPTURE1_4CH; } else { mApTurnOnSequence = AUDIO_CTL_ADDA_TO_CAPTURE1; } //打开对应通路控件 AudioALSADeviceConfigManager::getInstance()->ApplyDeviceTurnonSequenceByName(mApTurnOnSequence); //配置mStreamAttributeSource参数 /* Reset frames readed counter */ mStreamAttributeSource.Time_Info.total_frames_readed = 0; mStreamAttributeSource.sample_rate = getInputSampleRate(mStreamAttributeSource.input_device, mStreamAttributeSource.output_devices); mStreamAttributeSource.audio_format = AUDIO_FORMAT_PCM_8_24_BIT; if (mStreamAttributeSource.input_device == AUDIO_DEVICE_IN_WIRED_HEADSET || mStreamAttributeSource.input_source == AUDIO_SOURCE_UNPROCESSED){ mStreamAttributeSource.num_channels = 1; } else { mStreamAttributeSource.num_channels = 2; } mStreamAttributeSource.latency = mlatency; //配置mConfig setApHwPcm(); //DMA传输完成之后的回调函数：processDmaMsgWrapper mAudioMessengerIPI->registerDmaCbk( TASK_SCENE_CAPTURE_RAW, 0x2000, 0xF000, processDmaMsgWrapper, this); mAudioALSAVolumeController->SetCaptureGain(mStreamAttributeSource.audio_mode, mStreamAttributeSource.input_source, mStreamAttributeSource.input_device, mStreamAttributeSource.output_devices); openApHwPcm();// pcmOpen AudioDspStreamManager::getInstance()->addCaptureDataProvider(this);//pcm_prepare pcm_start int ret = pthread_create(&hReadThread, NULL, AudioALSACaptureDataProviderDspRaw::readThread, (void *)this); }open里面，主要就是操作到底层里面pcm_open和pcm_start，开始录音，然后就是最重要的readThread线程了！ void *AudioALSACaptureDataProviderDspRaw::readThread(void *arg) { pDataProvider->waitPcmStart(); // read raw data from alsa driver char linear_buffer[kReadBufferSizeNormal]; while (pDataProvider->mEnable == true) { //从底层读取录音数据，放到linear_buffer ret = pDataProvider->pcmRead(pDataProvider->mPcm, linear_buffer, kReadBufferSize); // use ringbuf format to save buffer info pDataProvider->mPcmReadBuf.pBufBase = linear_buffer; pDataProvider->mPcmReadBuf.bufLen = Read_Size + 1; // +1: avoid pRead == pWrite pDataProvider->mPcmReadBuf.pRead = linear_buffer; pDataProvider->mPcmReadBuf.pWrite = linear_buffer + Read_Size; pDataProvider->provideCaptureDataToAllClients(open_index); } pthread_exit(NULL); return NULL; }重点来了，这里就是在线程中不断的从alsa底层中读取到录音数据到linear_buffer中，然后放到pDataProvider->mPcmReadBuf.pBufBase中，提供给所有clien，也就是我们AudioALSACaptureDataClientAurisysNormal。 void AudioALSACaptureDataProviderBase::provideCaptureDataToAllClients(const uint32_t open_index) { for (size_t i = 0; i pcm_read_buf_wrap.base = pcm_read_buf.pBufBase; pcm_read_buf_wrap.read = pcm_read_buf.pRead; pcm_read_buf_wrap.write = pcm_read_buf.pWrite; pcm_read_buf_wrap.size = pcm_read_buf.bufLen; audio_ringbuf_copy_from_ringbuf_all(&mRawDataBuf, &pcm_read_buf_wrap); }最后就是将数据填装到mRawDataBuf了，这样就可以在AudioALSACaptureDataClientAurisysNormal::processThread中拿到录音数据了。 总的来说，录音流程还是比较简单的，核心就是AudioALSACaptureDataProviderDspRaw::readThread负责生产数据，AudioALSACaptureDataClientAurisysNormal::processThread负责消耗数据，一个生产者-消费者模型。 |
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