Audio-AudioRecord create(一)

AudioRecord create相关代码流程、重点关键接口、参数解析以及相关log share分析比对。

1.frameworks/base/media/java/android/media/AudioRecord.java

参数解析：

audioSource: MediaRecorder.AudioSource.MIC，详细定义需check MediaRecorder.AudioSource;

sampleRateInHz：默认采样率，单位Hz，44100Hz是唯一能保证在所有设备上工作的采样率；

channelConfig：描述音频通道设置，AudioFormat.CHANNEL_CONFIGURATION_MONO保证能在所有设备上工作；

audioFormat：音频数据保证支持此格式，这里设置为AudioFormat.ENCODING_16BIT；

bufferSizeInBytes：在录制过程中,音频数据写入缓冲区的字节数,getMinVufferSize()获取到的值。

主要的解析过程在native_setup函数中处理，需要注意的是区别于不同audio调用的session Id.

@SystemApipublic AudioRecord(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,int sessionId) throws IllegalArgumentException {mRecordingState = RECORDSTATE_STOPPED;...///*add by shengjie for parse AudioRecorder//1.REMOTE_SUBMIX这个音源是咋么处理的，没有跟，之后有项目需要的话可以跟一下//*/// is this AudioRecord using REMOTE_SUBMIX at full volume?if (attributes.getCapturePreset() == MediaRecorder.AudioSource.REMOTE_SUBMIX) {final AudioAttributes.Builder filteredAttr = new AudioAttributes.Builder();final Iterator<String> tagsIter = attributes.getTags().iterator();while (tagsIter.hasNext()) {final String tag = tagsIter.next();if (tag.equalsIgnoreCase(SUBMIX_FIXED_VOLUME)) {mIsSubmixFullVolume = true;Log.v(TAG, "Will record from REMOTE_SUBMIX at full fixed volume");} else { // SUBMIX_FIXED_VOLUME: is not to be propagated to the native layersfilteredAttr.addTag(tag);}}filteredAttr.setInternalCapturePreset(attributes.getCapturePreset());mAudioAttributes = filteredAttr.build();} else {mAudioAttributes = attributes;}...///*add by shengjie for parse AudioRecord//2.audioParamCheck对参数进行合法性检查，赋值mRecordSource、mSampleRate、mAudioFormat//*/audioParamCheck(attributes.getCapturePreset(), rate, encoding);///*add by shengjie for parse AudioRecord//3.获取声道数以及声道掩码，单声道掩码为0x10，双声道掩码为0x0c；//*/if ((format.getPropertySetMask()& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {mChannelIndexMask = format.getChannelIndexMask();mChannelCount = format.getChannelCount();}if ((format.getPropertySetMask()& AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {mChannelMask = getChannelMaskFromLegacyConfig(format.getChannelMask(), false);mChannelCount = format.getChannelCount();} else if (mChannelIndexMask == 0) {mChannelMask = getChannelMaskFromLegacyConfig(AudioFormat.CHANNEL_IN_DEFAULT, false);mChannelCount =  AudioFormat.channelCountFromInChannelMask(mChannelMask);}audioBuffSizeCheck(bufferSizeInBytes);int[] sampleRate = new int[] {mSampleRate};///*add by shengjie for parse AudioRecord//一个Session就是一个会话，每个会话都有一个独一无二的Id来标识。该Id的最终管理在AudioFlinger中。//一个会话可以被多个AudioTrack对象和MediaPlayer共用。//共用一个Session的AudioTrack和MediaPlayer共享相同的AudioEffect（音效）。//*/int[] session = new int[1];session[0] = sessionId;//TODO: update native initialization when information about hardware init failure//      due to capture device already open is available.int initResult = native_setup( new WeakReference<AudioRecord>(this),mAudioAttributes, sampleRate, mChannelMask, mChannelIndexMask,mAudioFormat, mNativeBufferSizeInBytes,session, getCurrentOpPackageName(), 0 /*nativeRecordInJavaObj*/);if (initResult != SUCCESS) {loge("Error code "+initResult+" when initializing native AudioRecord object.");return; // with mState == STATE_UNINITIALIZED}mSampleRate = sampleRate[0];mSessionId = session[0];mState = STATE_INITIALIZED;}

2.frameworks/base/core/jni/android_media_AudioRecord.cpp

主要工作：调用AudioRecord.cpp的set函数接口，并设置
audiorecord_callback_cookie *lpCallbackData回调用于传输音频数据buffer。

static jint
android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,jobject jaa, jintArray jSampleRate, jint channelMask, jint channelIndexMask,jint audioFormat, jint buffSizeInBytes, jintArray jSession, jstring opPackageName,jlong nativeRecordInJavaObj)
{//ALOGV(">> Entering android_media_AudioRecord_setup");//ALOGV("sampleRate=%d, audioFormat=%d, channel mask=%x, buffSizeInBytes=%d "//     "nativeRecordInJavaObj=0x%llX",//     sampleRateInHertz, audioFormat, channelMask, buffSizeInBytes, nativeRecordInJavaObj);...sp<AudioRecord> lpRecorder = 0;audiorecord_callback_cookie *lpCallbackData = NULL;// if we pass in an existing *Native* AudioRecord, we don't need to create/initialize one.if (nativeRecordInJavaObj == 0) {if (jaa == 0) {ALOGE("Error creating AudioRecord: invalid audio attributes");return (jint) AUDIO_JAVA_ERROR;}if (jSampleRate == 0) {ALOGE("Error creating AudioRecord: invalid sample rates");return (jint) AUDIO_JAVA_ERROR;}///*add by shengjie for parse AudioRecord setup//1.got sample Rate from new AlsaRecord(44100hz/16khz)jint elements[1];env->GetIntArrayRegion(jSampleRate, 0, 1, elements);int sampleRateInHertz = elements[0];//*/// channel index mask takes priority over channel position masks.if (channelIndexMask) {// Java channel index masks need the representation bits set.localChanMask = audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_INDEX,channelIndexMask);}// Java channel position masks map directly to the native definitionif (!audio_is_input_channel(localChanMask)) {ALOGE("Error creating AudioRecord: channel mask %#x is not valid.", localChanMask);return (jint) AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK;}///*add by shengjie for parse AudioRecord setup//2.got channel count from new AlsaRecord(1/2/4 ch)uint32_t channelCount = audio_channel_count_from_in_mask(localChanMask);//*/// compare the format against the Java constants///*add by shengjie for parse AudioRecord setup//3.got audio format from new AlsaRecord(16bit/32bit)audio_format_t format = audioFormatToNative(audioFormat);if (format == AUDIO_FORMAT_INVALID) {ALOGE("Error creating AudioRecord: unsupported audio format %d.", audioFormat);return (jint) AUDIORECORD_ERROR_SETUP_INVALIDFORMAT;}//*////*add by shengjie for parse AudioRecord setupsize_t bytesPerSample = audio_bytes_per_sample(format);if (buffSizeInBytes == 0) {ALOGE("Error creating AudioRecord: frameCount is 0.");return (jint) AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT;}//4.1 每个采样帧大小:所有通道数 * 每一帧所占字节数size_t frameSize = channelCount * bytesPerSample;//4.2 采样帧数量：最小缓冲区大小 / 每个采样帧大小size_t frameCount = buffSizeInBytes / frameSize;//*/ScopedUtfChars opPackageNameStr(env, opPackageName);// create an uninitialized AudioRecord objectlpRecorder = new AudioRecord(String16(opPackageNameStr.c_str()));// read the AudioAttributes valuesauto paa = JNIAudioAttributeHelper::makeUnique();jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {return jStatus;}ALOGV("AudioRecord_setup for source=%d tags=%s flags=%08x", paa->source, paa->tags, paa->flags);audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE;if (paa->flags & AUDIO_FLAG_HW_HOTWORD) {flags = AUDIO_INPUT_FLAG_HW_HOTWORD;}// create the callback information:// this data will be passed with every AudioRecord callbacklpCallbackData = new audiorecord_callback_cookie;lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);// we use a weak reference so the AudioRecord object can be garbage collected.///*add by shengjie for parse AudioRecord setup//5.AudioRecord.java的指针绑定到lpCallbackData回调数据中，这样就能把数据通过回调的方式通知到上层lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);lpCallbackData->busy = false;//*////*add by shengjie for AudioRecord setup//6.调用AudioRecord.cpp的set函数，flags，他的类型为audio_input_flags_t，定义在audio-base.h中，作为音频输入的标志，此处为AUDIO_INPUT_FLAG_NONE，没有做特殊设定const status_t status = lpRecorder->set(paa->source,sampleRateInHertz,format,        // word length, PCMlocalChanMask,frameCount,recorderCallback,// callback_tlpCallbackData,// void* user0,             // notificationFrames,true,          // threadCanCallJavasessionId,AudioRecord::TRANSFER_DEFAULT,flags,-1, -1,        // default uid, pidpaa.get());//*/...// Set caller name so it can be logged in destructor.// MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_JAVAlpRecorder->setCallerName("java");} else { // end if nativeRecordInJavaObj == 0)lpRecorder = (AudioRecord*)nativeRecordInJavaObj;// TODO: We need to find out which members of the Java AudioRecord might need to be...// create the callback information:// this data will be passed with every AudioRecord callbacklpCallbackData = new audiorecord_callback_cookie;lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);// we use a weak reference so the AudioRecord object can be garbage collected.lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);lpCallbackData->busy = false;}nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);if (nSession == NULL) {ALOGE("Error creating AudioRecord: Error retrieving session id pointer");goto native_init_failure;}// read the audio session ID back from AudioRecord in case a new session was created during set()nSession[0] = lpRecorder->getSessionId();env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);nSession = NULL;...///*add by shengjie for parse AudioRecord setup//7.把lpRecorder对象以及lpCallbackData回调保存到javaAudioRecordFields的相应字段中// save our newly created C++ AudioRecord in the "nativeRecorderInJavaObj" field// of the Java objectsetAudioRecord(env, thiz, lpRecorder);// save our newly created callback information in the "nativeCallbackCookie" field// of the Java object (in mNativeCallbackCookie) so we can free the memory in finalize()env->SetLongField(thiz, javaAudioRecordFields.nativeCallbackCookie, (jlong)lpCallbackData);//*/return (jint) AUDIO_JAVA_SUCCESS;...///*Error Handler Process*///...}

system/media/audio/include/system/audio-base.h

typedef enum {AUDIO_INPUT_FLAG_NONE       = 0x0,AUDIO_INPUT_FLAG_FAST       = 0x1,AUDIO_INPUT_FLAG_HW_HOTWORD = 0x2,AUDIO_INPUT_FLAG_RAW        = 0x4,AUDIO_INPUT_FLAG_SYNC       = 0x8,AUDIO_INPUT_FLAG_MMAP_NOIRQ = 0x10,AUDIO_INPUT_FLAG_VOIP_TX    = 0x20,AUDIO_INPUT_FLAG_HW_AV_SYNC = 0x40,AUDIO_INPUT_FLAG_DIRECT     = 0x80,
} audio_input_flags_t;

3.frameworks/av/media/libaudioclient/AudioRecord.cpp

解析相关的音频参数，主要是创建AudioRecordThread和createRecord_l

status_t AudioRecord::set(audio_source_t inputSource,uint32_t sampleRate,audio_format_t format,audio_channel_mask_t channelMask,size_t frameCount,callback_t cbf,void* user,uint32_t notificationFrames,bool threadCanCallJava,audio_session_t sessionId,transfer_type transferType,audio_input_flags_t flags,uid_t uid,pid_t pid,const audio_attributes_t* pAttributes,audio_port_handle_t selectedDeviceId,audio_microphone_direction_t selectedMicDirection,float microphoneFieldDimension)
{status_t status = NO_ERROR;uint32_t channelCount;pid_t callingPid;pid_t myPid;// Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.ALOGV("%s(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, ""notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s ""uid %d, pid %d",__func__,inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);mTracker.reset(new RecordingActivityTracker());mSelectedDeviceId = selectedDeviceId;mSelectedMicDirection = selectedMicDirection;mSelectedMicFieldDimension = microphoneFieldDimension;...///*add by shengjie for parse AudioRecord set//1.get trabsferType TRANSFER_SYNC/TRANSFER_CALLBACKmTransfer = transferType;//*/...if (pAttributes == NULL) {mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;mAttributes.source = inputSource;if (inputSource == AUDIO_SOURCE_VOICE_COMMUNICATION|| inputSource == AUDIO_SOURCE_CAMCORDER) {mAttributes.flags |= AUDIO_FLAG_CAPTURE_PRIVATE;}} else {// stream type shouldn't be looked at, this track has audio attributesmemcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));ALOGV("%s(): Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",__func__, mAttributes.source, mAttributes.flags, mAttributes.tags);}mSampleRate = sampleRate;// these below should probably come from the audioFlinger too...if (format == AUDIO_FORMAT_DEFAULT) {format = AUDIO_FORMAT_PCM_16_BIT;}// validate parameters// AudioFlinger capture only supports linear PCMif (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {ALOGE("%s(): Format %#x is not linear pcm", __func__, format);status = BAD_VALUE;goto exit;}mFormat = format;if (!audio_is_input_channel(channelMask)) {ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);status = BAD_VALUE;goto exit;}mChannelMask = channelMask;channelCount = audio_channel_count_from_in_mask(channelMask);mChannelCount = channelCount;if (audio_is_linear_pcm(format)) {mFrameSize = channelCount * audio_bytes_per_sample(format);} else {mFrameSize = sizeof(uint8_t);}// mFrameCount is initialized in createRecord_lmReqFrameCount = frameCount;mNotificationFramesReq = notificationFrames;// mNotificationFramesAct is initialized in createRecord_lmSessionId = sessionId;ALOGV("%s(): mSessionId %d", __func__, mSessionId);callingPid = IPCThreadState::self()->getCallingPid();myPid = getpid();if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {mClientUid = IPCThreadState::self()->getCallingUid();} else {mClientUid = uid;}if (pid == -1 || (callingPid != myPid)) {mClientPid = callingPid;} else {mClientPid = pid;}mOrigFlags = mFlags = flags;mCbf = cbf;///*add by shengjie for parse AudioRecord set//2.new AudioRecordThread to handle releated processif (cbf != NULL) {mAudioRecordThread = new AudioRecordThread(*this);mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);// thread begins in paused state, and will not reference us until start()}//*/// create the IAudioRecord///*add by shengjie for parse AudioRecord set//3.use createRecord_l to invork audio related devices/input parameters{AutoMutex lock(mLock);status = createRecord_l(0 /*epoch*/, mOpPackageName);}//*/ALOGV("%s(%d): status %d", __func__, mPortId, status);if (status != NO_ERROR) {if (mAudioRecordThread != 0) {mAudioRecordThread->requestExit();   // see comment in AudioRecord.hmAudioRecordThread->requestExitAndWait();mAudioRecordThread.clear();}goto exit;}mUserData = user;// TODO: add audio hardware input latency heremLatency = (1000LL * mFrameCount) / mSampleRate;mMarkerPosition = 0;mMarkerReached = false;mNewPosition = 0;mUpdatePeriod = 0;AudioSystem::acquireAudioSessionId(mSessionId, mClientPid, mClientUid);mSequence = 1;mObservedSequence = mSequence;mInOverrun = false;mFramesRead = 0;mFramesReadServerOffset = 0;///*error handler process*///
}

createRecord_l方法具体实现如下：

重要接口:AudioFliger.cpp createRecord_l后续会进行一系列参数的解析与分发；

// must be called with mLock held
status_t AudioRecord::createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
{const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();IAudioFlinger::CreateRecordInput input;IAudioFlinger::CreateRecordOutput output;sp<media::IAudioRecord> record;...///*loding input related parameters*////*add by shengjie for parse AudioRecord createRecord_l//1.IAudioFliger.cpp通过Binder进程间通信调用AudioFliger.cpp的createRecord,承担包括音频input device构建getInputForAttr、createRecordTrack_l、getOrphanEffectChain_l等所有功能。ALOGI("wsj-audioFlinger->createRecord_l %s(): mSessionId : %d, input flags = %08x", __func__, mSessionId, mFlags);record = audioFlinger->createRecord(input,output,&status);//*/...ALOGI("wsj-audioFlinger->createRecord_l ater, output mSessionId : %d, output flags = %08x", mSessionId, mFlags);// Starting address of buffers in shared memory.// The buffers are either immediately after the control block,// or in a separate area at discretion of server.void *buffers;if (output.buffers == 0) {buffers = cblk + 1;} else {// TODO: Using unsecurePointer() has some associated security pitfalls//       (see declaration for details).//       Either document why it is safe in this case or address the//       issue (e.g. by copying).buffers = output.buffers->unsecurePointer();if (buffers == NULL) {ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);status = NO_INIT;goto exit;}}// invariant that mAudioRecord != 0 is true only after set() returns successfullyif (mAudioRecord != 0) {IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);mDeathNotifier.clear();}mAudioRecord = record;mCblkMemory = output.cblk;mBufferMemory = output.buffers;IPCThreadState::self()->flushCommands();mCblk = cblk;...//mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creationif (mDeviceCallback != 0) {if (mInput != AUDIO_IO_HANDLE_NONE) {AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);}ALOGI("wsj-audioFlinger->createRecord_l ater, oAudioSystem::addAudioDeviceCallback output.inputId: %d, output.portId: %d", output.inputId, output.portId);AudioSystem::addAudioDeviceCallback(this, output.inputId, output.portId);}...// update proxyALOGI("wsj-audioFlinger->createRecord_l after, new AudioRecordClientProxy, mFrameCount: %zu, mFrameSize: %zu", mFrameCount, mFrameSize);mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);mProxy->setEpoch(epoch);mProxy->setMinimum(mNotificationFramesAct);DeathNotifier = new DeathNotifier(this);IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this);...
}

4.frameworks/av/media/libaudioclient/IAudioFlinger.cpp

virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input,CreateRecordOutput& output,status_t *status){Parcel data, reply;sp<media::IAudioRecord> record;data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());if (status == nullptr) {return record;}ALOGI("wsj-IaudioFlinger->createRecord input.writeToParcel(&data)");input.writeToParcel(&data);///*add by shengjie for IAudioFlinger createRecord//通过Binder进程间通信拿到share bufferstatus_t lStatus = remote()->transact(CREATE_RECORD, data, &reply);if (lStatus != NO_ERROR) {ALOGE("createRecord transaction error %d", lStatus);*status = DEAD_OBJECT;return record;}//*/*status = reply.readInt32();if (*status != NO_ERROR) {ALOGE("createRecord returned error %d", *status);return record;}record = interface_cast<media::IAudioRecord>(reply.readStrongBinder());if (record == 0) {ALOGE("createRecord returned a NULL IAudioRecord with status OK");*status = DEAD_OBJECT;return record;}ALOGI("wsj-IaudioFlinger->createRecord output.readFromParcel(&reply)");output.readFromParcel(&reply);return record;}

5.frameworks/av/services/audioflinger/AudioFlinger.cpp

getInputForAttr:设置audio input相关信息对应的audio device

checkRecordThread_l:创建recordThread去做recording相关的配置

getOrphanEffectChain_l:下发对应的不同增益

sp<media::IAudioRecord> AudioFlinger::createRecord(const CreateRecordInput& input,CreateRecordOutput& output,status_t *status)
{sp<RecordThread::RecordTrack> recordTrack;sp<RecordHandle> recordHandle;sp<Client> client;status_t lStatus;audio_session_t sessionId = input.sessionId;audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;...///*add by shengjie for parse AudioFlinger createRecord//1.parse input related info and check all audio devices infoALOGI("wsj-audioFlinger->createRecord getInputForAttr, sessionId: %d, clientPid: %d, clientUid: %d, input.opPackageName: %s, output.flags: %d", sessionId, clientPid, clientUid, String8(input.opPackageName).string(), output.flags);lStatus = AudioSystem::getInputForAttr(&input.attr, &output.inputId,input.riid,sessionId,// FIXME compare to AudioTrackclientPid,clientUid,input.opPackageName,&input.config,output.flags, &output.selectedDeviceId, &portId);//*/{Mutex::Autolock _l(mLock);RecordThread *thread = checkRecordThread_l(output.inputId);if (thread == NULL) {ALOGE("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);lStatus = BAD_VALUE;goto Exit;}...///*add by shengjie for AudioFlinger createRecord//2.create Record Tract to check recording audio buffer and dataALOGI("wsj-createRecord() AudioFlinger::createRecord lSessionId: %d input %d", sessionId, output.inputId);recordTrack = thread->createRecordTrack_l(client, input.attr, &output.sampleRate,input.config.format, input.config.channel_mask,&output.frameCount, sessionId,&output.notificationFrameCount,callingPid, clientUid, &output.flags,input.clientInfo.clientTid,&lStatus, portId,input.opPackageName);//*/...// Check if one effect chain was awaiting for an AudioRecord to be created on this// session and move it to this thread.///*add by shengjie for parse AudioFlinger createRecord//3.enable audio input channer related effectALOGI("wsj-createRecord() AudioFlinger::createRecord getOrphanEffectChain_l");sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);//*/if (chain != 0) {Mutex::Autolock _l(thread->mLock);thread->addEffectChain_l(chain);}break;}// End of retry loop.// The lack of indentation is deliberate, to reduce code churn and ease merges.}output.cblk = recordTrack->getCblk();output.buffers = recordTrack->getBuffers();output.portId = portId;// return handle to clientrecordHandle = new RecordHandle(recordTrack);...*status = lStatus;return recordHandle;
}

6.相关流程Log解析

总结：

下面的文章会对AudioFlinger createRecord里面涉及的getInputForAttr、checkRecordThread_l、getOrphanEffectChain_l接口进行更详细的解析。

对流程的追踪，目的是能在这套流程下能随便插入一些项目中关于音频麦克阵列、数据处理的一些特殊配置及要求，能在整体宏观的session、client管控下另外走一些特殊的通道满足项目需求。