Android audio 三 AudioRecord 分析下

frameworks/av/media/libmedia/AudioRecord.cpp

AudioRecord 是Android 拾音對象。

實例化 AudioRecord 對象後，啓動錄音的步驟：

sp< AudioRecord> audiorecord = new AudioRecord(......)
audiorecord->start();
audiorecord->read(......)
audiorecord->stop()

先來看看 AudioRecord 的構造函數和析構函數；

AudioRecord::AudioRecord

AudioRecord::AudioRecord(......) 調用了 status_t AudioRecord::set(......) 接下來分析 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.h
            mAudioRecordThread->requestExitAndWait();
            mAudioRecordThread.clear();
        }
        // No lock here: worst case we remove a NULL callback which will be a nop
        if(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 聲道採樣率音頻幀長度 讀取音頻設備的數據的方式

重點如下：

// frameworks/av/media/libmedia/AudioRecord.cpp
status_t AudioRecord::set(......)
{
    ......
    // 實例化 AudioRecordThread
    if(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
	// 獲取系統服務 AudioFlinger ，並調用 AudioFlinger->openRecord ，重點
    status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);
	......
}

AudioRecord::set 源碼如下：

// 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);

    // 選擇數據傳輸方式
    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 successfully
    if(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 attributes
        memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
        ALOGV("Building AudioRecord with attributes: source=%d flags=0x%x tags=[%s]",
              mAttributes.source, mAttributes.flags, mAttributes.tags);
    }
    // 採樣率
    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
	// 判斷 AudioFlinger 是否支持該格式
    if(!audio_is_valid_format(format) || !audio_is_linear_pcm(format))
    {
        ALOGE("Format %#x is not linear pcm", format);
        return BAD_VALUE;
    }
    mFormat = format;
	// 聲道檢查
    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;
    // 音頻幀長度
    if(audio_is_linear_pcm(format))
    {
        mFrameSize = channelCount * audio_bytes_per_sample(format);
    }
    else
    {
        mFrameSize = sizeof(uint8_t);
    }

    // mFrameCount is initialized in openRecord_l
    mReqFrameCount = frameCount;

    mNotificationFramesReq = notificationFrames;
    // mNotificationFramesAct is initialized in openRecord_l

    if(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;
    // 實例化 AudioRecordThread
    if(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
	// 獲取系統服務 AudioFlinger ，並調用 AudioFlinger->openRecord ，重點
    status_t status = openRecord_l(0 /*epoch*/, mOpPackageName);

    if(status != NO_ERROR)
    {
        if(mAudioRecordThread != 0)
        {
            mAudioRecordThread->requestExit();   // see comment in AudioRecord.h
            mAudioRecordThread->requestExitAndWait();
            mAudioRecordThread.clear();
        }
        return status;
    }

    mStatus = NO_ERROR;
    mUserData = user;
    // TODO: add audio hardware input latency here
    if(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 的聲明如下：

發現並沒有 run 成員函數，AudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);

我們知道 AudioRecordThread 繼承了一個線程類 Thread， Thread 類中聲明瞭成員 run

在線程函數 Thread::run 創建線程 Thread::_threadLoop(void* user)

_threadLoop 循環調用 self->threadLoop();
self->threadLoop() 調用 AudioRecord.processAudioBuffer();
AudioRecord.processAudioBuffer() 調用 AudioRecord::restoreRecord_l(const char *from)
AudioRecord::restoreRecord_l(const char *from) 調用 AudioFlinger->openRecord(......)
AudioFlinger 是承上啓下的作用，爲應用層提供訪問接口，並通過 HAL 來管理音頻設備
下一步是分析對象 AudioFlinger，後續有時間，我會寫博客分析的 AudioFlinger 。（這裏不做分析）

AudioRecord::openRecord_l() 也是調用 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 boundary
    void        resume();   // allow thread to execute, if not requested to exit
    void        wake();     // wake to handle changed notification conditions.

private:
    void        pauseInternal(nsecs_t ns = 0LL);
    // like pause(), but only used internally within thread

    friend class AudioRecord;
    virtual bool        threadLoop();
    AudioRecord&        mReceiver;
    virtual ~AudioRecordThread();
    Mutex               mMyLock;    // Thread::mLock is private
    Condition           mMyCond;    // Thread::mThreadExitedCondition is private
    bool                mPaused;    // whether thread is requested to pause at next loop entry
    bool                mPausedInt; // whether thread internally requests pause
    nsecs_t             mPausedNs;  // if mPausedInt then associated timeout, otherwise ignored
    bool                mIgnoreNextPausedInt;   // skip any internal pause and go immediately
    // to processAudioBuffer() as state may have changed
    // since pause time calculated.
};

AudioRecord::star

開始拾音

// 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 buffer
    const 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 0
    mProxy->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

AudioRecord::read 調用 AudioRecord::obtainBuffer(......) 獲取音頻設備的拾音數據
AudioRecord::obtainBuffer(......) 調用 AudioRecordClientProxy::obtainBuffer(......)

AudioRecord::stop

AudioRecord::stop 源碼如下：

IAudioRecord::stop() 通過 binder 停止 Audio Record 服務。

AudioRecordThread::pause(); 暫停線程

變量 mPaused 在函數開頭的地方判斷返回 true ，暫停線程。

// 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);
    }
}

Android audio 三 AudioRecord 分析下

AudioRecord::AudioRecord

AudioRecordThread 和 AudioRecord::openRecord_l()

AudioRecord::star

AudioRecord::read

AudioRecord::stop

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