从Android API AudioRecorder构造过程开始：
构造函数调用了native_setup native函数，在android_media_AudioRecorder.cpp jni文件中，基本的API接口都对应了native接口。
private native final int native_setup(Object audiorecord_this,
Object attributes,
int[] sampleRate, int channelMask, int channelIndexMask, int audioFormat,
int buffSizeInBytes, int[] sessionId, String opPackageName,
long nativeRecordInJavaObj);
private native final int native_start(int syncEvent, int sessionId);
private native final int native_read_in_byte_array(byte[] audioData,
int offsetInBytes, int sizeInBytes, boolean isBlocking);android_media_AudioRecord_setup
lpRecorder = new AudioRecord(String16(opPackageNameStr.c_str()));
lpRecorder->set，调用了openRecord_l，并且访问AF openRecord拿到IAudioRecord。
sp<IAudioRecord> AudioFlinger::openRecord
recordingAllowed(opPackageName, tid, clientUid) 根据包名等检查权限；
RecordThread *thread = checkRecordThread_l(input) 根据audio_io_handle从mRecordThreads中拿到recordthread
client = registerPid(pid) 创建了一个AudioFlinger::Client对象，clinet主要包含一块1024*1024的内存，将client加入mClients.add(pid, client);
AudioFlinger::RecordThread::createRecordTrack_l 使用recordthread创建track
recordHandle = new RecordHandle(recordTrack); 利用recordtrack生成recordHandle
return recordHandle ; RecordHandle继承自BnAudioRecord，返回类型为sp<IAudioRecord>的bindler对象，返回给AudioRecorder中使用
IAudioFlinger中是AudioFlinger的Bindler调用接口，client层也就是AudioRecorder.cpp运行的层，通过AudioSystem提供的接口拿到AF的bindler对象，实际过程是通过ServiceManager查找Bindler服务得到bindler对象，同时在AudioSystem保存了静态值。
sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger()
AF的bindler对象执行openRcorder的操作,是一个标准的Bindler调用，删除了很多代码仅保留Bindler调用，data和reply分别是调用的序列化和返回的序列化，执行remote()->transact进行Bindler调用，返回了IAudioRecord Bindler对象。
注意Bindler调用的code为OPEN_RECORD，可根据code在服务端区分调用；
virtual sp<IAudioRecord> openRecord( audio_io_handle_t input, uint32_t sampleRate, audio_format_t format,...) { Parcel data, reply; sp<IAudioRecord> record; data.writeInt32((int32_t) input); data.writeInt32(format); status_t lStatus = remote()->transact(OPEN_RECORD, data, &reply); if (lStatus != NO_ERROR) { ALOGE("openRecord error: %s", strerror(-lStatus)); } else { size_t lNotificationFrames = (size_t) reply.readInt64(); lStatus = reply.readInt32(); record = interface_cast<IAudioRecord>(reply.readStrongBinder()); cblk = interface_cast<IMemory>(reply.readStrongBinder()); buffers = interface_cast<IMemory>(reply.readStrongBinder()) }在Android系统中代码设计时，通常Bindler Server接收的实现和Client调用的实现在一个文件中，这里在IAudioFlinger中，BnAudioFlinger表示server端的接口，onTransact收到Bindler调用；
下面代码仅保留Bindler调用部分，这里因为BnAudioFlinger是AudioFlinger的父类，调用openRecord是AudioFlinger中成员，这样就一个调用执行到了音频服务AudioFlinger中。并且返回也序列化处理了，包含Bindler对象。
status_t BnAudioFlinger::onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) { case OPEN_RECORD: { CHECK_INTERFACE(IAudioFlinger, data, reply); audio_io_handle_t input = (audio_io_handle_t) data.readInt32(); sp<IAudioRecord> record = openRecord(input, sampleRate, format, channelMask, opPackageName, &frameCount, &flags, pid, tid, clientUid, &sessionId, ¬ificationFrames, cblk, buffers, &status); reply->writeInt32(status); reply->writeStrongBinder(IInterface::asBinder(record)); reply->writeStrongBinder(IInterface::asBinder(cblk)); reply->writeStrongBinder(IInterface::asBinder(buffers));
AudioFlinger::RecordThread::createRecordTrack_l 使用recordthread创建track
track = new RecordTrack(this, client, sampleRate,
format, channelMask, frameCount, NULL, sessionId, uid,
*flags, TrackBase::TYPE_DEFAULT); 里面包含了RecordBufferConverter AudioRecordServerProxy ResamplerBufferProvider
mTracks.add(track); mTracks是一个vector
return一个 sp<RecordTrack>对象
AudioRecorder在Client端拿到了可以访问AF的Bindler对象IAudioRecord，赋值给mAudioRecord，这里还涉及一个共享内存的使用，非常重要；
到此Recorder的初始化就完成了。
API
startRecording 对应的过程如下
native_start 调用native的start
android_media_AudioRecord_start jni函数start
sp<AudioRecord> lpRecorder = getAudioRecord(env, thiz); 拿到AudioRecorder对象，之前将这个对象创建好后保存在了java层；
lpRecorder->start((AudioSystem::sync_event_t)event, (audio_session_t) triggerSession) 调用了AudioRecorder的start， triggerSession在java API传过来，默认为0
status_t AudioRecord::start
status = mAudioRecord->start(event, triggerSession); 调用前面从AF拿到的IAudioRecorder Bindler对象mAudioRecord，这个里面只有两个接口可调用， transact的code代表Bindler调用的type；
enum { UNUSED_WAS_GET_CBLK = IBinder::FIRST_CALL_TRANSACTION, START, STOP };status_t BnAudioRecord::onTransact中执行server端的start，RecordHandle继承自BnAudioRecord，因此在BnAudioRecorder中调用start是 AudioFlinger::RecordHandle::start成员函数；
在audioflinger/Tracks.cpp中实现 AudioFlinger::RecordHandle
mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 实调用际执行track的方法；RecordHandle应该是对RecordTrack的Bindler封装；
status_t AudioFlinger::RecordThread::RecordTrack::start RecordTrack和RecordThread关联很密切；
track的定义在RecordTrack.h中，实现在Tracks.cpp, 都在audioflinger目录下；
recordThread->start(this, event, triggerSession); 对RecordThread调用start, 将当前track对象this传入了RecordThread, event使用SYNC_EVENT_NONE（java传入，通过AudioSystem映射）
AudioFlinger::RecordThread::start
mActiveTracks.add(recordTrack); 加入active队列
mActiveTracksGen++; active计数增加
recordTrack->mResamplerBufferProvider->reset(); 重置一些...
// clear any converter state as new data will be discontinuous
recordTrack->mRecordBufferConverter->reset();
recordTrack->mState = TrackBase::STARTING_2; starting状态
// signal thread to start
mWaitWorkCV.broadcast(); 通知线程start
下面应该看如何使用这个mActiveTracks和里面的track的；
AudioFlinger::RecordThread::threadLoop() AudioFlinger和AudioPolicyService启动阶段创建的Recorder线程执行体；
for (;;) { thread loop
size_t size = mActiveTracks.size();
mWaitWorkCV.wait(mLock); active的track为0时，利用CV wait，当recorder start时将创建的track加入队列，CV信号通知这里开始工作；
for (size_t i = 0; i < size; ) { 遍历每个recordtrack
activeTrack = mActiveTracks[i];
switch判断track状态， 在mActiveTracks中的也有可能处在未active状态： PAUSING STARTING_1 STARTING_2 ACTIVE IDLE
activeTracks.add(activeTrack); 将active状态的加入局部队列；
在record的threadloop中还有一个核心的操作，从hw中读取音频数据到缓存；
threadloop从hw中读取音频数据到缓存；mRsmpInBuffer是缓存buffer，mRsmpInRear 是buffer写数据偏移，
// Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. // Only the client(s) that are too slow will overrun. But if even the fastest client is too // slow, then this RecordThread will overrun by not calling HAL read often enough. // If destination is non-contiguous, first read past the nominal end of buffer, then // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); ssize_t framesRead; // If an NBAIO source is present, use it to read the normal capture's data if (mPipeSource != 0) { size_t framesToRead = mBufferSize / mFrameSize; framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesToRead); if (framesRead == 0) { // since pipe is non-blocking, simulate blocking input sleepUs = (framesToRead * 1000000LL) / mSampleRate; } // otherwise use the HAL / AudioStreamIn directly } else { ATRACE_BEGIN("read"); ssize_t bytesRead = mInput->stream->read(mInput->stream, (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); ATRACE_END(); if (bytesRead < 0) { framesRead = bytesRead; } else { framesRead = bytesRead / mFrameSize; } }
record threadloop中对真正active的track局部队列遍历
size = activeTracks.size();
// loop over each active track
for (size_t i = 0; i < size; i++) {
activeTrack = activeTracks[i];
针对每个track循环更新buffer
track更新buffer过程，
activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun);
mResamplerBufferProvider这个是track的一个成员对象，track创建时创建了它，并且将track的this引用传入，因此mResamplerBufferProvider拥有了track的引用，track拥有RecordThread的应用，在sync中能拿到recordThread的引用；
thread中保存有音频buffer的状态，将这些状态更新到mResamplerBufferProvider
const int32_t rear = recordThread->mRsmpInRear;
const int32_t front = mRsmpInFront;
const ssize_t filled = rear - front;
// process frames from the RecordThread buffer provider to the RecordTrack buffer
framesOut = activeTrack->mRecordBufferConverter->convert(activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
// update frame information and push timestamp out
activeTrack->updateTrackFrameInfo(activeTrack->mServerProxy->framesReleased(),
mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER],
mSampleRate, mTimestamp);
ResamplerBufferProvider和RecordBufferConverter的实现都在AudioFlinger.cpp中；这两类的对象都属于一个track，辅助track实现音频capture的过程；
RecorderBufferConverter在构建时，创建了一个mResampler = AudioResampler::create 用于重采样；
RecordBufferConverter::conver过程就是用mResampler重采样，并将输入拷贝到activeTrack->mSink.raw
API read对应的过程如下
native_read_in_byte_array jni实现
lpRecorder->read 调用AudioRecorder native read
ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking)
status_t err = obtainBuffer(&audioBuffer,
blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 循环调用获取音频数到audioBuffer
obtainBuffer执行过程用到了在openRecord阶段创建的AudioRecordClientProxy，
mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize);
buffers = bufferMem->pointer(); buffers共享内存中缓冲区的起始地址。
bufferMem是AF openRecord过程生成的共享内存；
cblk指向iMem，是缓冲区控制块；
sp<IMemory> iMem; // for cblk
sp<IMemory> bufferMem;
sp<IAudioRecord> record = audioFlinger->openRecord(input,
mSampleRate,
mFormat,
mChannelMask,
opPackageName,
&temp,
&flags,
mClientPid,
tid,
mClientUid,
&mSessionId,
¬ificationFrames,
iMem,
bufferMem,
&status);
class AudioRecordClientProxy : public ClientProxy 继承自ClientProxy, 在AudioTrackShared.h中定义，还有AudioTrack也有一样的实现；主要是为client和server之间跨进程内存共享数据传输；
obtainBuffer是父类ClientProxy的成员，obtainBuffer是计算共享内存读取的位置，将指针指向该点；
buffer->mFrameCount = part1;
buffer->mRaw = part1 > 0 ?
&((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL; mBuffers是创建Porxy时传入的共享内存对象；
obtainBuffer是一个阻塞式调用，通过for(;;)循环获取AF服务中产生的音频数据，在打断或拿到数据后返回；
通过上面提到的控制块中数据判断是否有可用数据，write to rear, read from front
int32_t front; int32_t rear; if (mIsOut) { // The barrier following the read of mFront is probably redundant. // We're about to perform a conditional branch based on 'filled', // which will force the processor to observe the read of mFront // prior to allowing data writes starting at mRaw. // However, the processor may support speculative execution, // and be unable to undo speculative writes into shared memory. // The barrier will prevent such speculative execution. front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); rear = cblk->u.mStreaming.mRear; } else { // On the other hand, this barrier is required. rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear); front = cblk->u.mStreaming.mFront; } // write to rear, read from front ssize_t filled = rear - front;
AudioRecord::read调用obtainBuffer获得共享内存的读取地址后，拷贝数据到用户buffer，完成一次数据read，obtainBuffer过程只获取一帧数据，read可循环的调用obtainBuffer获取多帧数据到用户buffer；
size_t bytesRead = audioBuffer.size;
memcpy(buffer, audioBuffer.i8, bytesRead);
AudioRecord在client端调用AudioFlinger server端时传入IMemory类型的两个共享内存对象；
iMem应用控制，bufferMem是共享内存buffer；
sp<IMemory> iMem; // for cblk sp<IMemory> bufferMem; sp<IAudioRecord> record = audioFlinger->openRecord(input, mSampleRate, mFormat, mChannelMask, opPackageName, &temp, &flags, mClientPid, tid, mClientUid, &mSessionId, ¬ificationFrames, iMem, bufferMem, &status);
sp<IAudioRecord> AudioFlinger::openRecord 共享内存实现；利用recorderTrack的方法获得；一个client调用一次openRecord对应一个track，因此每个client和server之间唯一一个共享内存用于音频传递；
cblk = recordTrack->getCblk();
buffers = recordTrack->getBuffers();
recordTrack包装成Bindler对象IAudioRecord返回给client，实际每个client调用server都是通过对应的IAudioRecord；
recordHandle = new RecordHandle(recordTrack);
return recordHandle; 返回值是sp<IAudioRecord>， RecordHandle继承自BnAudioRecord;
AudioFlinger openRecord过程创建了RecordTrack，这里涉及两个队列：
mTracks和mActiveTracks
mTracks：刚创建的加入这个队列；
mActiveTracks：start后的加入这个队列；
RecordTrack继承自TrackBase，并且引用一个AudioFlinger::Client对象，clinet主要包含一块1024*1024的内存；client是在RecordTrack构建之前创建，构建时传给RecordTrack，代表了一个client在server中的实例；
TrackBase的构建过程创建了共享内存；
mCblkMemory和mBufferMemory对应控制块和共享内存，在下面调用中返回：
cblk = recordTrack->getCblk();
buffers = recordTrack->getBuffers();
上面过程是读共享内存的过程，还有些共享内存的过程，在RecodThread的threadloop中；
status_t status = activeTrack->getNextBuffer(&activeTrack->mSink);
status_t status = mServerProxy->obtainBuffer(&buf) 实际执行ServerProxy的obtainBuffer， 和ClientProxy的原理一样；
