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DuplicatingThread需要与蓝牙结合起来使用,它的存在与Audio硬件结构息息相关。读者可参考图7-12“智能手机硬件架构图”来理解。当一份数据同时需要发送给DSP和蓝牙A2DP设备时,DuplicatingThread就派上用场了。在分析DuplicatingThread前,还是应该了解一下它的来龙去脉。 1. DuplicatingThread的来历 DuplicatingThread和蓝牙的A2DP设备有关系。可先假设有一个蓝牙立体声耳机已经连接上了,接着从setDeviceConnectionState开始分析,代码如下所示: **AudioPolicyManagerBase.cpp** ~~~ status_t AudioPolicyManagerBase::setDeviceConnectionState( AudioSystem::audio_devicesdevice, AudioSystem::device_connection_state state, const char *device_address) { ...... switch (state) { case AudioSystem::DEVICE_STATE_AVAILABLE: mAvailableOutputDevices |= device; #ifdef WITH_A2DP if (AudioSystem::isA2dpDevice(device)) { //专门处理A2DP设备的连接 status_t status = handleA2dpConnection(device, device_address); } #endif ...... ~~~ 对于A2DP设备,有专门的函数handleA2dpConnection处理,代码如下所示: **AudioPolicyManagerBase.cpp** ~~~ status_tAudioPolicyManagerBase::handleA2dpConnection( AudioSystem::audio_devicesdevice, const char*device_address) { AudioOutputDescriptor *outputDesc = new AudioOutputDescriptor(); outputDesc->mDevice= device; //先为mA2dpOutput创建一个MixerThread,这个和mHardwareOutput一样 mA2dpOutput =mpClientInterface->openOutput(&outputDesc->mDevice, &outputDesc->mSamplingRate, &outputDesc->mFormat, &outputDesc->mChannels, &outputDesc->mLatency, outputDesc->mFlags); if (mA2dpOutput) { /* a2dpUsedForSonification永远返回true,表示属于SONIFCATION策略的音频流声音需要 同时从蓝牙和DSP中传出。属于SONIFCATION策略的音频流类型可查看前面关于getStrategy的 分析,来电铃声、短信通知等属于这一类 */ if(a2dpUsedForSonification()) { /* 创建一个DuplicateOutput,注意它的参数,第一个是蓝牙MixerThread 第二个是DSPMixerThread */ mDuplicatedOutput = mpClientInterface->openDuplicateOutput( mA2dpOutput, mHardwareOutput); } if(mDuplicatedOutput != 0 || !a2dpUsedForSonification()) { if (a2dpUsedForSonification()) { //创建一个AudioOutputDescriptor对象 AudioOutputDescriptor *dupOutputDesc = new AudioOutputDescriptor(); dupOutputDesc->mOutput1 = mOutputs.valueFor(mHardwareOutput); dupOutputDesc->mOutput2 = mOutputs.valueFor(mA2dpOutput); ...... //保存mDuplicatedOutput和dupOutputDesc键值对 addOutput(mDuplicatedOutput, dupOutputDesc); ...... } } } ...... ~~~ 这里,最重要的函数是openDuplicateOutput。它和openOutput一样,最终的处理都是在AF中。去那里看看,代码如下所示: **AudioFlinger.cpp** ~~~ int AudioFlinger::openDuplicateOutput(intoutput1, int output2) { Mutex::Autolock_l(mLock); //output1对应蓝牙的MixerThread MixerThread*thread1 = checkMixerThread_l(output1); //output2对应DSP的MixerThread MixerThread *thread2 = checkMixerThread_l(output2); //①创建DuplicatingThread,注意它第二个参数使用的,是代表蓝牙的MixerThread DuplicatingThread *thread = new DuplicatingThread(this, thread1,++mNextThreadId); //②加入代表DSP的MixerThread thread->addOutputTrack(thread2); mPlaybackThreads.add(mNextThreadId, thread); returnmNextThreadId;//返回DuplicatingThread的索引 } ~~~ 从现在起,MixerThread要简写为MT,而DuplicatingThread则简写为DT。 OK,这里面有两个重要的函数调用,一起来看。 2. DuplicatingThread和OutputTrack 先看DT的构造函数,代码如下所示: **AudioFlinger.cpp** ~~~ AudioFlinger::DuplicatingThread::DuplicatingThread(constsp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread*mainThread,int id) : MixerThread(audioFlinger,mainThread->getOutput(), id), mWaitTimeMs(UINT_MAX) { //DT是MT的派生类,所以先要完成基类的构造,还记得MT的构造吗?它会创建一个AudioMixer对象 mType =PlaybackThread::DUPLICATING; //把代表DSP的MT加入进来,咱们看看 addOutputTrack(mainThread); } ~~~ **AudioFlinger.cpp** ~~~ voidAudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) { intframeCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); //构造一个OutputTrack,它的第一个参数是MT OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, this, mSampleRate, mFormat, mChannelCount,frameCount); if(outputTrack->cblk() != NULL) { thread->setStreamVolume(AudioSystem::NUM_STREAM_TYPES, 1.0f); //把这个outputTrack加入到mOutputTracks数组保存 mOutputTracks.add(outputTrack); updateWaitTime(); } } ~~~ 此时,当下面两句代码执行完: ~~~ DuplicatingThread *thread = newDuplicatingThread(this, thread1,++mNextThreadId); thread->addOutputTrack(thread2); ~~~ DT分别构造了两个OutputTrack,一个对应蓝牙的MT,一个对应DSP的MT。现在来看OutputTrack为何方神圣,代码如下所示: **AudioFlinger.cpp** ~~~ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( const wp<ThreadBase>& thread, DuplicatingThread*sourceThread, uint32_t sampleRate, int format,int channelCount,int frameCount) :Track(thread,NULL, AudioSystem::NUM_STREAM_TYPES, sampleRate, format, channelCount, frameCount, NULL),//最后这个参数为NULL mActive(false),mSourceThread(sourceThread) { /* OutputTrack从Track派生,所以需要先调用基类的构造,还记得Track构造函数 中的事情吗?它会创建一块内存,至于是不是共享内存,由Track构造函数的最后一个参数决定。 如果该值为NULL,表示没有客户端参与,则会在本进程内创建一块内存,这块内存的结构如 图7-4所示,前边为CB对象,后边为数据缓冲 */ //下面的这个thread对象为MT PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); if(mCblk != NULL) { mCblk->out = 1;//表示DT将往MT中写数据 //和前面所分析的AT、AF中的处理何其相似! mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); mCblk->volume[0] = mCblk->volume[1] = 0x1000; mOutBuffer.frameCount = 0; //把这个Track加到MT的Track中 playbackThread->mTracks.add(this); } ~~~ 明白了吗?图7-16表示的是openDuplicateOutput的结果: :-: ![](http://img.blog.csdn.net/20150802161026784?watermark/2/text/aHR0cDovL2Jsb2cuY3Nkbi5uZXQv/font/5a6L5L2T/fontsize/400/fill/I0JBQkFCMA==/dissolve/70/gravity/Center) 图7-16 openDuplicateOutput的结果示意图 图7-16说明(以蓝牙MT为例): - 蓝牙MT的Track中有一个成员为OutputTrack0。 - DT的mOutputTracks也有一个成员指向OutputTrack0。这就好像DT是MT的客户端一样,它和前面分析的AT是AF的客户端类似。 - 红色部分代表数据传递用的缓冲。 3. DT的客户端AT DT是从MT中派生的,根据AP和AT的交互流程,当AT创建的流类型对应策略为SONIFACATION时,它会从AP中得到代表DT的线程索引号。由于DT没有重载createTrack_l,所以这个过程也会创建一个Track对象(和MT创建Track对象一样)。此时的结果,将导致图7-16变成图7-17。 :-: ![](http://img.blog.csdn.net/20150802160934318?watermark/2/text/aHR0cDovL2Jsb2cuY3Nkbi5uZXQv/font/5a6L5L2T/fontsize/400/fill/I0JBQkFCMA==/dissolve/70/gravity/Center) 图7-17 有AT的DT全景图 图7-17把DT的工作方式表达得非常清晰了。一个DT配合两个OutputTrack中的进程内缓冲,把来自AT的数据原封不动地发给蓝牙MT和DSP MT,这简直就是个数据中继器!。不过俗话说得好,道理虽简单,实现却复杂。来看DT是如何完成这一复杂而艰巨的任务的吧。 4. DT的线程函数 DT的线程函数代码如下所示: **AudioFlinger.cpp** ~~~ boolAudioFlinger::DuplicatingThread::threadLoop() { int16_t* curBuf = mMixBuffer; Vector< sp<Track> > tracksToRemove; uint32_t mixerStatus = MIXER_IDLE; nsecs_t standbyTime = systemTime(); size_tmixBufferSize = mFrameCount*mFrameSize; SortedVector< sp<OutputTrack> > outputTracks; while(!exitPending()) { processConfigEvents(); mixerStatus = MIXER_IDLE; { ...... //处理配置请求,和MT处理一样 const SortedVector< wp<Track> >& activeTracks =mActiveTracks; for (size_t i = 0; i < mOutputTracks.size(); i++) { outputTracks.add(mOutputTracks[i]); } //如果AT的Track停止了,则需要停止和MT共享的OutputTrack ifUNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || mSuspended) { if (!mStandby) { for (size_t i = 0; i <outputTracks.size(); i++) { outputTracks[i]->stop(); } mStandby = true; mBytesWritten = 0; } ...... //DT从MT派生,天然具有混音的功能,所以这部分功能和MT一致 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); } if(LIKELY(mixerStatus == MIXER_TRACKS_READY)) { //outputsReady将检查OutputTracks对应的MT状态 if (outputsReady(outputTracks)) { mAudioMixer->process(curBuf);//使用AudioMixer对象混音 } else { memset(curBuf, 0, mixBufferSize); } sleepTime = 0; writeFrames = mFrameCount; } ...... if (sleepTime == 0) { standbyTime = systemTime() +kStandbyTimeInNsecs; for (size_t i = 0; i < outputTracks.size(); i++) { //将混音后的数据写到outputTrack中 outputTracks[i]->write(curBuf, writeFrames); } mStandby = false; mBytesWritten += mixBufferSize; }else { usleep(sleepTime); } tracksToRemove.clear(); outputTracks.clear(); } returnfalse; } ~~~ 现在,来自远端进程AT的数据已得到了混音,这一份混音后的数据还将通过调用OutputTrack的write完成DT到其他两个MT的传输。注意,这里除了AT使用的Track外,还有DT和两个MT共享的OutputTrack。AT调用的start,将导致DT的Track加入到活跃数组中,但另外两个OutputTrack还没调用start。这些操作又是在哪里做的呢?来看write函数: **AudioFlinger.cpp** ~~~ boolAudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) { //注意,此处的OutputTrack是DT和MT共享的 Buffer *pInBuffer; BufferinBuffer; uint32_t channels = mCblk->channels; booloutputBufferFull = false; inBuffer.frameCount = frames; inBuffer.i16 = data; uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); if(!mActive && frames != 0) { //如果此Track没有活跃,则调用start激活 start(); ...... } /* 现在,AF中的数据传递有三个线程:一个DT,两个MT。MT作为DT的二级消费者, 可能由于某种原因来不及消费数据,所以DT中提供了一个缓冲队列mBufferQueue, 把MT来不及消费的数据保存在这个缓冲队列中。注意这个缓冲队列容纳的临时缓冲 个数是有限制的,其限制值由kMaxOverFlowBuffers控制,初始化为10个 */ while(waitTimeLeftMs) { //先消耗保存在缓冲队列的数据 if(mBufferQueue.size()) { pInBuffer = mBufferQueue.itemAt(0); }else { pInBuffer = &inBuffer; } ...... //获取可写缓冲,下面这句代码是否和AT中对应的代码很相似? if(obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS){ ...... break; } uint32_toutFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount: pInBuffer->frameCount; //将数据拷贝到DT和MT共享的那块缓冲中去 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t)); //更新写位置 mCblk->stepUser(outFrames); pInBuffer->frameCount-= outFrames; pInBuffer->i16 += outFrames * channels; mOutBuffer.frameCount -= outFrames; mOutBuffer.i16 += outFrames * channels; ...... }//while 结束 if(inBuffer.frameCount) { sp<ThreadBase> thread = mThread.promote(); if(thread != 0 && !thread->standby()) { if (mBufferQueue.size() < kMaxOverFlowBuffers) { pInBuffer = new Buffer; pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels]; pInBuffer->frameCount = inBuffer.frameCount; pInBuffer->i16 = pInBuffer->mBuffer; //拷贝旧数据到新的临时缓冲 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount *channels * sizeof(int16_t)); //保存这个临时缓冲 mBufferQueue.add(pInBuffer); } } } //如果数据全部写完 if(pInBuffer->frameCount == 0) { if (mBufferQueue.size()) { mBufferQueue.removeAt(0); delete [] pInBuffer->mBuffer; delete pInBuffer;//释放缓冲队列对应的数据缓冲 } else { break; } } } ...... return outputBufferFull; } ~~~ 数据就这样从AT通过DT的帮助,传输到蓝牙的MT和DSP的MT中了。这种方式继数据传输比直接使用MT传输要缓慢。 到这里,对DT的讲解就告一段落了。本人觉得,DT的实现是AF代码中最美妙的地方,多学习这些优秀代码,有助于提高学习者的水平。 >[info] **说明**:DT还有别的一些细节本书中没有涉及,读者可以结合自己的情况进行分析和理解。