Android audio 三 AudioRecord 分析下

Android audio 一 源码路径

Android audio 二 AudioRecord 分析上

Android audio 三 AudioRecord 分析下

Android audio 四 AudioTrack 分析上

Android audio 五 AudioTrack 分析下

Android audio 六 AudioRecord AudiTrack 拾音放音例子

frameworks/av/media/libmedia/AudioRecord.cpp

AudioRecord 是Android 拾音对象。

实例化 AudioRecord 对象后，启动录音的步骤：

1.  sp< AudioRecord> audiorecord = new AudioRecord(......)
2.  audiorecord->start();
3.  audiorecord->read(......)
4.  audiorecord->stop()

先来看看 AudioRecord 的构造函数和析构函数；

AudioRecord::AudioRecord

AudioRecord::AudioRecord(......)  调用了 status_t AudioRecord::set(......)  接下来分析 set(......) ；

// frameworks/av/media/libmedia/AudioRecord.cpp
AudioRecord::AudioRecord(const String16 &opPackageName): mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName), mSessionId(AUDIO_SESSION_ALLOCATE),mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
}AudioRecord::AudioRecord(audio_source_t inputSource,uint32_t sampleRate,audio_format_t format,audio_channel_mask_t channelMask,const String16& opPackageName,size_t frameCount,callback_t cbf,void* user,uint32_t notificationFrames,audio_session_t sessionId,transfer_type transferType,audio_input_flags_t flags,int uid,pid_t pid,const audio_attributes_t* pAttributes): mActive(false),mStatus(NO_INIT),mOpPackageName(opPackageName),mSessionId(AUDIO_SESSION_ALLOCATE),mPreviousPriority(ANDROID_PRIORITY_NORMAL),mPreviousSchedulingGroup(SP_DEFAULT),mProxy(NULL),mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{mStatus = set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,uid, pid, pAttributes);
}AudioRecord::~AudioRecord()
{if(mStatus == NO_ERROR){// Make sure that callback function exits in the case where// it is looping on buffer empty condition in obtainBuffer().// Otherwise the callback thread will never exit.stop();if(mAudioRecordThread != 0){mProxy->interrupt();mAudioRecordThread->requestExit();  // see comment in AudioRecord.hmAudioRecordThread->requestExitAndWait();mAudioRecordThread.clear();}// No lock here: worst case we remove a NULL callback which will be a nopif(mDeviceCallback != 0 && mInput != AUDIO_IO_HANDLE_NONE){AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mInput);}IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);mAudioRecord.clear();mCblkMemory.clear();mBufferMemory.clear();IPCThreadState::self()->flushCommands();ALOGV("~AudioRecord, releasing session id %d",mSessionId);AudioSystem::releaseAudioSessionId(mSessionId, -1 /*pid*/);}
}

status_t AudioRecord::set(......) 

配置 AudioRecord 的 声道 采样率 音频帧长度  读取音频设备的数据的方式

重点如下：

// frameworks/av/media/libmedia/AudioRecord.cpp
status_t AudioRecord::set(......)
{......// 实例化 AudioRecordThreadif(cbf != NULL){mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);// thread begins in paused state, and will not reference us until start()}// create the IAudioRecord// 获取系统服务 AudioFlinger ，并调用 AudioFlinger->openRecord ，重点status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);......
}

 

AudioRecord::set 源码如下：

// frameworks/av/media/libmedia/AudioRecord.cpp
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,int uid,pid_t pid,const audio_attributes_t* pAttributes)
{ALOGV("set(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, ""notificationFrames %u, sessionId %d, transferType %d, flags %#x, opPackageName %s ""uid %d, pid %d",inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,sessionId, transferType, flags, String8(mOpPackageName).string(), uid, pid);// 选择数据传输方式switch(transferType){case TRANSFER_DEFAULT:if(cbf == NULL || threadCanCallJava){transferType = TRANSFER_SYNC;}else{transferType = TRANSFER_CALLBACK;}break;case TRANSFER_CALLBACK:if(cbf == NULL){ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL");return BAD_VALUE;}break;case TRANSFER_OBTAIN:case TRANSFER_SYNC:break;default:ALOGE("Invalid transfer type %d", transferType);return BAD_VALUE;}mTransfer = transferType;// invariant that mAudioRecord != 0 is true only after set() returns successfullyif(mAudioRecord != 0){ALOGE("Track already in use");return INVALID_OPERATION;}if(pAttributes == NULL){memset(&mAttributes, 0, sizeof(audio_attributes_t));mAttributes.source = inputSource;}else{// stream type shouldn't be looked at, this track has audio attributesmemcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",mAttributes.source, mAttributes.flags, mAttributes.tags);}// 采样率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 PCM// 判断 AudioFlinger 是否支持该格式if(!audio_is_valid_format(format) || !audio_is_linear_pcm(format)){ALOGE("Format %#x is not linear pcm", format);return BAD_VALUE;}mFormat = format;// 声道检查if(!audio_is_input_channel(channelMask)){ALOGE("Invalid channel mask %#x", channelMask);return BAD_VALUE;}mChannelMask = channelMask;uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);mChannelCount = channelCount;// 音频帧长度if(audio_is_linear_pcm(format)){mFrameSize = channelCount * audio_bytes_per_sample(format);}else{mFrameSize = sizeof(uint8_t);}// mFrameCount is initialized in openRecord_lmReqFrameCount = frameCount;mNotificationFramesReq = notificationFrames;// mNotificationFramesAct is initialized in openRecord_lif(sessionId == AUDIO_SESSION_ALLOCATE){mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);}else{mSessionId = sessionId;}ALOGV("set(): mSessionId %d", mSessionId);int callingpid = IPCThreadState::self()->getCallingPid();int mypid = getpid();if(uid == -1 || (callingpid != mypid)){mClientUid = IPCThreadState::self()->getCallingUid();}else{mClientUid = uid;}if(pid == -1 || (callingpid != mypid)){mClientPid = callingpid;}else{mClientPid = pid;}mOrigFlags = mFlags = flags;mCbf = cbf;// 实例化 AudioRecordThreadif(cbf != NULL){mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);// thread begins in paused state, and will not reference us until start()}// create the IAudioRecord// 获取系统服务 AudioFlinger ，并调用 AudioFlinger->openRecord ，重点status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);if(status != NO_ERROR){if(mAudioRecordThread != 0){mAudioRecordThread->requestExit();   // see comment in AudioRecord.hmAudioRecordThread->requestExitAndWait();mAudioRecordThread.clear();}return status;}mStatus = NO_ERROR;mUserData = user;// TODO: add audio hardware input latency hereif(mTransfer == TRANSFER_CALLBACK ||mTransfer == TRANSFER_SYNC){mLatency = (1000 * mNotificationFramesAct) / sampleRate;}else{mLatency = (1000 * mFrameCount) / sampleRate;}mMarkerPosition = 0;mMarkerReached = false;mNewPosition = 0;mUpdatePeriod = 0;AudioSystem::acquireAudioSessionId(mSessionId, -1);mSequence = 1;mObservedSequence = mSequence;mInOverrun = false;mFramesRead = 0;mFramesReadServerOffset = 0;return NO_ERROR;
}

 

AudioRecordThread 和 AudioRecord::openRecord_l() 

 AudioRcorcdThread 的声明如下： 

发现并没有 run 成员函数，AudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);

我们知道 AudioRecordThread 继承了一个线程类 Thread， Thread 类中声明了成员 run 

在线程函数 Thread::run 创建线程 Thread::_threadLoop(void* user)

1.  _threadLoop 循环调用 self->threadLoop();
2. self->threadLoop() 调用 AudioRecord.processAudioBuffer();  
3. AudioRecord.processAudioBuffer() 调用 AudioRecord::restoreRecord_l(const char *from)
4. AudioRecord::restoreRecord_l(const char *from) 调用 AudioFlinger->openRecord(......)
5.  AudioFlinger 是承上启下的作用，为应用层提供访问接口，并通过 HAL 来管理音频设备
6. 下一步是分析对象 AudioFlinger，后续有时间，我会写博客分析的 AudioFlinger 。（这里不做分析）  

AudioRecord::openRecord_l() 也是调用 AudioFlinger->openRecord(......)

// frameworks/av/media/libmedia/AudioRecord.cpp/* a small internal class to handle the callback */
class AudioRecordThread : public Thread
{
public:AudioRecordThread(AudioRecord& receiver, bool bCanCallJava = false);// Do not call Thread::requestExitAndWait() without first calling requestExit().// Thread::requestExitAndWait() is not virtual, and the implementation doesn't do enough.virtual void        requestExit();void        pause();    // suspend thread from execution at next loop boundaryvoid        resume();   // allow thread to execute, if not requested to exitvoid        wake();     // wake to handle changed notification conditions.private:void        pauseInternal(nsecs_t ns = 0LL);// like pause(), but only used internally within threadfriend class AudioRecord;virtual bool        threadLoop();AudioRecord&        mReceiver;virtual ~AudioRecordThread();Mutex               mMyLock;    // Thread::mLock is privateCondition           mMyCond;    // Thread::mThreadExitedCondition is privatebool                mPaused;    // whether thread is requested to pause at next loop entrybool                mPausedInt; // whether thread internally requests pausensecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignoredbool                mIgnoreNextPausedInt;   // skip any internal pause and go immediately// to processAudioBuffer() as state may have changed// since pause time calculated.
};

 

AudioRecord::star

开始拾音

// frameworks/av/media/libmedia/AudioRecord.cpp
status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
{ALOGV("start, sync event %d trigger session %d", event, triggerSession);SEEMPLOG_RECORD(71,"");AutoMutex lock(mLock);if(mActive){return NO_ERROR;}// discard data in bufferconst uint32_t framesFlushed = mProxy->flush();mFramesReadServerOffset -= mFramesRead + framesFlushed;mFramesRead = 0;mProxy->clearTimestamp();  // timestamp is invalid until next server push// reset current position as seen by client to 0mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());// force refresh of remaining frames by processAudioBuffer() as last// read before stop could be partial.mRefreshRemaining = true;mNewPosition = mProxy->getPosition() + mUpdatePeriod;int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);// we reactivate markers (mMarkerPosition != 0) as the position is reset to 0.// This is legacy behavior.  This is not done in stop() to avoid a race condition// where the last marker event is issued twice.mMarkerReached = false;mActive = true;status_t status = NO_ERROR;if(!(flags & CBLK_INVALID)){status = mAudioRecord->start(event, triggerSession);if(status == DEAD_OBJECT){flags |= CBLK_INVALID;}}if(flags & CBLK_INVALID){status = restoreRecord_l("start");}if(status != NO_ERROR){mActive = false;ALOGE("start() status %d", status);}else{sp<AudioRecordThread> t = mAudioRecordThread;if(t != 0){t->resume();}else{mPreviousPriority = getpriority(PRIO_PROCESS, 0);get_sched_policy(0, &mPreviousSchedulingGroup);androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);}}return status;
}

 

AudioRecord::read

1. AudioRecord::read 调用 AudioRecord::obtainBuffer(......) 获取音频设备的拾音数据
2. AudioRecord::obtainBuffer(......)  调用 AudioRecordClientProxy::obtainBuffer(......)

 

AudioRecord::stop

AudioRecord::stop 源码如下：

IAudioRecord::stop()  通过 binder 停止 Audio Record 服务。

  AudioRecordThread::pause();  暂停线程

变量 mPaused  在函数开头的地方判断返回 true ，暂停线程。

// frameworks/av/media/libmedia/AudioRecord.cpp
void AudioRecord::stop()
{AutoMutex lock(mLock);if (!mActive) {return;}mActive = false;mProxy->interrupt();mAudioRecord->stop();// Note: legacy handling - stop does not clear record marker and// periodic update position; we update those on start().sp<AudioRecordThread> t = mAudioRecordThread;if (t != 0) {t->pause();} else {setpriority(PRIO_PROCESS, 0, mPreviousPriority);set_sched_policy(0, mPreviousSchedulingGroup);}
}