Android系统音频模块-Native层初始化工作

在前面一篇的文章中，我们知道了音频模块Java层所做的一些事情，总的来说还是比较简单的，下面我们继续学习和探索Native层中系统做了什么工作，首先先简单介绍下，Native层采用了C/S的架构方式，AudioTrack是属于Client端的，AudioFlinger和AudioPolicyService是属于服务端的，AudioFlinger是唯一可以调用HAL层接口的地方，而AudioPolicyService主要是对一些音频策略的选择和一些逻辑调用的中转，下面先来看看Client端AudioTrack都做了什么。

首先需要注意的是，我们本篇文章只描述Native层代码的初始化工作，因为整个Native层的代码和逻辑是整个音频模块的核心，想要在一篇文章中讲述完整那是不可能的，其中的很多地方的知识点都可以单独拿出来写一篇文章。

1. AudioTrack的初始化工作

在Java层JNI方法中，我们得知不仅通过AudioTrack的无参构造方法生成AudioTrack，还调用了它的set()方法，下面是一些重要的代码片段，它所在的路径是：/framework/av/media/libmedia .

//AudioTrack的无参构造方法
AudioTrack::AudioTrack()
    : mStatus(NO_INIT),
      mIsTimed(false),
      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
      mPreviousSchedulingGroup(SP_DEFAULT),
      mPausedPosition(0),
      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
    mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
    mAttributes.usage = AUDIO_USAGE_UNKNOWN;
    mAttributes.flags = 0x0;
    strcpy(mAttributes.tags, "");
}

status_t AudioTrack::set(
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t frameCount,
        audio_output_flags_t flags,
        callback_t cbf,
        void* user,
        uint32_t notificationFrames,
        const sp<IMemory>& sharedBuffer,
        bool threadCanCallJava,
        int sessionId,
        transfer_type transferType,
        const audio_offload_info_t *offloadInfo,
        int uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        bool doNotReconnect)
{


   //校验数据...(省略)

   //对参数的处理和初始化工作...(省略)

    // validate parameters
    if (!audio_is_valid_format(format)) {
        ALOGE("Invalid format %#x", format);
        return BAD_VALUE;
    }
    mFormat = format;

    if (!audio_is_output_channel(channelMask)) {
        ALOGE("Invalid channel mask %#x", channelMask);
        return BAD_VALUE;
    }

    //确定声道数
    mChannelMask = channelMask;
    uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
    mChannelCount = channelCount;

     //根据flags 确定framesize 大小
    if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
        if (audio_is_linear_pcm(format)) {
            mFrameSize = channelCount * audio_bytes_per_sample(format);
        } else {
            mFrameSize = sizeof(uint8_t);
        }
    } else {
        ALOG_ASSERT(audio_is_linear_pcm(format));
        mFrameSize = channelCount * audio_bytes_per_sample(format);
        // createTrack will return an error if PCM format is not supported by server,
        // so no need to check for specific PCM formats here
    }

    // sampling rate must be specified for direct outputs
    if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
        return BAD_VALUE;
    }
    mSampleRate = sampleRate;
    mOriginalSampleRate = sampleRate;
    mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;

    mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
    mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
    mSendLevel = 0.0f;
    // mFrameCount is initialized in createTrack_l
    mReqFrameCount = frameCount;
    mNotificationFramesReq = notificationFrames;
    mNotificationFramesAct = 0;
    // ...
    mAuxEffectId = 0;
    mFlags = flags;
    mCbf = cbf;

    //1. 如果Java传入的音频状态回调对象不为空，则开启线程处理回调
    if (cbf != NULL) {
        mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
        // thread begins in paused state, and will not reference us until start()
    }

    //2. 创建 sp<IAudioTrack> 对象，主要和服务端进行通信
    status_t status = createTrack_l();


    //... 参数的初始化，赋值工作

    return NO_ERROR;
}

这个方法实在太长了，有些不重要的我删除了，还对一些语句做了备注，最为主要的我用序列标出来了，下面就详细学习下这两步到底做了什么。

1.1. Native层状态回调

在我们AudioTrack调用set()方法中，创建了一条名叫AudioTrackThread线程，我们来看下它的数据结构：

    /* a small internal class to handle the callback */
    class AudioTrackThread : public Thread
    {
    public:
        AudioTrackThread(AudioTrack& 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 AudioTrack;
        virtual bool        threadLoop();
        AudioTrack&         mReceiver;
        virtual ~AudioTrackThread();
        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.
    };

我们可以看到它继承于Native层的Thread，其实在调用AudioTrackThread->run()的时候，最终会调用到它的threadLooper()，我们可以看下Thread.h的实现类Threads.cpp，代码位于：

./system/core/libutils/Threads.cpp

status_t Thread::run(const char* name, int32_t priority, size_t stack)
{
    Mutex::Autolock _l(mLock);

    //...
    bool res;
    //mCanCallJava表示是否需要attach虚拟机，这里是false.
    if (mCanCallJava) {
        res = createThreadEtc(_threadLoop,
                this, name, priority, stack, &mThread);
    } else {
        res = androidCreateRawThreadEtc(_threadLoop,
                this, name, priority, stack, &mThread);
    }
    //...
}

int androidCreateRawThreadEtc(android_thread_func_t entryFunction,
                               void *userData,
                               const char* threadName __android_unused,
                               int32_t threadPriority,
                               size_t threadStackSize,
                               android_thread_id_t *threadId)
{
    pthread_attr_t attr;
    pthread_attr_init(&attr);
    pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);

    //...

    errno = 0;
    pthread_t thread;
    //创建 unix thread. 可以看出 entryFunction就是上面方法传下的 _threadLoop
    int result = pthread_create(&thread, &attr,
                    (android_pthread_entry)entryFunction, userData);
    pthread_attr_destroy(&attr);
    if (result != 0) {
        ALOGE("androidCreateRawThreadEtc failed (entry=%p, res=%d, errno=%d)\n"
             "(android threadPriority=%d)",
            entryFunction, result, errno, threadPriority);
        return 0;
    }

    // Note that *threadID is directly available to the parent only, as it is
    // assigned after the child starts.  Use memory barrier / lock if the child
    // or other threads also need access.
    if (threadId != NULL) {
        *threadId = (android_thread_id_t)thread; // XXX: this is not portable
    }
    return 1;
}

int Thread::_threadLoop(void* user)
{
    Thread* const self = static_cast<Thread*>(user);

    //...

    bool first = true;

    //可以从下面代码看出，Thread线程会一直调用 threadloop 方法，当然这个循环也会阻塞，用于
    //和其他线程之间的协调工作.
    do {
        bool result;
        if (first) {
            first = false;
            self->mStatus = self->readyToRun();
            result = (self->mStatus == NO_ERROR);

            if (result && !self->exitPending()) {
                // Binder threads (and maybe others) rely on threadLoop
                // running at least once after a successful ::readyToRun()
                // (unless, of course, the thread has already been asked to exit
                // at that point).
                // This is because threads are essentially used like this:
                //   (new ThreadSubclass())->run();
                // The caller therefore does not retain a strong reference to
                // the thread and the thread would simply disappear after the
                // successful ::readyToRun() call instead of entering the
                // threadLoop at least once.
                result = self->threadLoop();
            }
        } else {
            result = self->threadLoop();
        }

        // establish a scope for mLock
        {
        Mutex::Autolock _l(self->mLock);
        if (result == false || self->mExitPending) {
            self->mExitPending = true;
            self->mRunning = false;
            // clear thread ID so that requestExitAndWait() does not exit if
            // called by a new thread using the same thread ID as this one.
            self->mThread = thread_id_t(-1);
            // note that interested observers blocked in requestExitAndWait are
            // awoken by broadcast, but blocked on mLock until break exits scope
            self->mThreadExitedCondition.broadcast();
            break;
        }
        }

        // Release our strong reference, to let a chance to the thread
        // to die a peaceful death.
        strong.clear();
        // And immediately, re-acquire a strong reference for the next loop
        strong = weak.promote();
    } while(strong != 0);

    return 0;
}

从上面一大片代码的分析，得出最后我们只要去分析 threadloop() 方法的实现即可，那么继续分析回调AudioTrack.cpp中对AudioTrackThread的threadloop()的实现：

bool AudioTrack::AudioTrackThread::threadLoop()
{
    {
        AutoMutex _l(mMyLock);
        if (mPaused) {
            mMyCond.wait(mMyLock);
            // caller will check for exitPending()
            return true;
        }
        if (mIgnoreNextPausedInt) {
            mIgnoreNextPausedInt = false;
            mPausedInt = false;
        }
        if (mPausedInt) {
            if (mPausedNs > 0) {
                (void) mMyCond.waitRelative(mMyLock, mPausedNs);
            } else {
                mMyCond.wait(mMyLock);
            }
            mPausedInt = false;
            return true;
        }
    }
    if (exitPending()) {
        return false;
    }
    nsecs_t ns = mReceiver.processAudioBuffer();
    switch (ns) {
    case 0:
        return true;
    case NS_INACTIVE:
        pauseInternal();
        return true;
    case NS_NEVER:
        return false;
    case NS_WHENEVER:
        // Event driven: call wake() when callback notifications conditions change.
        ns = INT64_MAX;
        // fall through
    default:
        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
        pauseInternal(ns);
        return true;
    }
}

其中很重要的一句mReceiver.processAudioBuffer()，其中的mReceiver就是AudioTrack本身，所以，go on …

nsecs_t AudioTrack::processAudioBuffer()
{
    //检测 mCblk 是否为空，mCblk 对应的是Java空间的回调方法
    LOG_ALWAYS_FATAL_IF(mCblk == NULL);

    mLock.lock();

    //...

    // Can only reference mCblk while locked
    int32_t flags = android_atomic_and(
        ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags);

    // Check for track invalidation
    if (flags & CBLK_INVALID) {
        // for offloaded tracks restoreTrack_l() will just update the sequence and clear
        // AudioSystem cache. We should not exit here but after calling the callback so
        // that the upper layers can recreate the track
        if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
            status_t status __unused = restoreTrack_l("processAudioBuffer");
            // FIXME unused status
            // after restoration, continue below to make sure that the loop and buffer events
            // are notified because they have been cleared from mCblk->mFlags above.
        }
    }

    //查看 音频服务端(消费者)是否是低负荷状态
    bool waitStreamEnd = mState == STATE_STOPPING;
    bool active = mState == STATE_ACTIVE;

    // Manage underrun callback, must be done under lock to avoid race with releaseBuffer()
    bool newUnderrun = false;
    if (flags & CBLK_UNDERRUN) {
#if 0
        // Currently in shared buffer mode, when the server reaches the end of buffer,
        // the track stays active in continuous underrun state.  It's up to the application
        // to pause or stop the track, or set the position to a new offset within buffer.
        // This was some experimental code to auto-pause on underrun.   Keeping it here
        // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content.
        if (mTransfer == TRANSFER_SHARED) {
            mState = STATE_PAUSED;
            active = false;
        }
#endif
        if (!mInUnderrun) {
            mInUnderrun = true;
            newUnderrun = true;
        }
    }

    //获取服务端的写指针
    size_t position = updateAndGetPosition_l();

    //检测是否达到警戒标志位
    bool markerReached = false;
    size_t markerPosition = mMarkerPosition;
    // FIXME fails for wraparound, need 64 bits
    if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) {
        mMarkerReached = markerReached = true;
    }

    // Determine number of new position callback(s) that will be needed, while locked
    size_t newPosCount = 0;
    size_t newPosition = mNewPosition;
    size_t updatePeriod = mUpdatePeriod;
    // FIXME fails for wraparound, need 64 bits
    if (updatePeriod > 0 && position >= newPosition) {
        newPosCount = ((position - newPosition) / updatePeriod) + 1;
        mNewPosition += updatePeriod * newPosCount;
    }

    // Cache other fields that will be needed soon
    uint32_t sampleRate = mSampleRate;
    float speed = mPlaybackRate.mSpeed;
    const uint32_t notificationFrames = mNotificationFramesAct;
    if (mRefreshRemaining) {
        mRefreshRemaining = false;
        mRemainingFrames = notificationFrames;
        mRetryOnPartialBuffer = false;
    }
    size_t misalignment = mProxy->getMisalignment();
    uint32_t sequence = mSequence;
    sp<AudioTrackClientProxy> proxy = mProxy;

    // Determine the number of new loop callback(s) that will be needed, while locked.
    int loopCountNotifications = 0;
    uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END

    //获取循环次数
    if (mLoopCount > 0) {
        int loopCount;
        size_t bufferPosition;
        mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
        loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
        loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
        mLoopCountNotified = loopCount; // discard any excess notifications
    } else if (mLoopCount < 0) {
        // FIXME: We're not accurate with notification count and position with infinite looping
        // since loopCount from server side will always return -1 (we could decrement it).
        size_t bufferPosition = mStaticProxy->getBufferPosition();
        loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
        loopPeriod = mLoopEnd - bufferPosition;
    } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
        size_t bufferPosition = mStaticProxy->getBufferPosition();
        loopPeriod = mFrameCount - bufferPosition;
    }

    mLock.unlock();

    // get anchor time to account for callbacks.
    const nsecs_t timeBeforeCallbacks = systemTime();

    /**
     * 这里检测 音频流是否播放完，播放状态回调Java层
     */
    if (waitStreamEnd) {
        // FIXME:  Instead of blocking in proxy->waitStreamEndDone(), Callback thread
        // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function
        // (and make sure we don't callback for more data while we're stopping).
        // This helps with position, marker notifications, and track invalidation.
        struct timespec timeout;
        timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC;
        timeout.tv_nsec = 0;

        status_t status = proxy->waitStreamEndDone(&timeout);
        switch (status) {
        case NO_ERROR:
        case DEAD_OBJECT:
        case TIMED_OUT:
            if (status != DEAD_OBJECT) {
                //回调Java层 EVENT_STREAM_END 状态
                mCbf(EVENT_STREAM_END, mUserData, NULL);
            }
            {
                AutoMutex lock(mLock);
                // The previously assigned value of waitStreamEnd is no longer valid,
                // since the mutex has been unlocked and either the callback handler
                // or another thread could have re-started the AudioTrack during that time.
                waitStreamEnd = mState == STATE_STOPPING;
                if (waitStreamEnd) {
                    mState = STATE_STOPPED;
                    mReleased = 0;
                }
            }
            if (waitStreamEnd && status != DEAD_OBJECT) {
               return NS_INACTIVE;
            }
            break;
        }
        return 0;
    }

   //回调之前判断的状态...

    if (newUnderrun) {
        mCbf(EVENT_UNDERRUN, mUserData, NULL);
    }

    //循环播放回调
    while (loopCountNotifications > 0) {
        mCbf(EVENT_LOOP_END, mUserData, NULL);
        --loopCountNotifications;
    }
    if (flags & CBLK_BUFFER_END) {
        mCbf(EVENT_BUFFER_END, mUserData, NULL);
    }
    if (markerReached) {
        mCbf(EVENT_MARKER, mUserData, &markerPosition);
    }

    //播放多少音频帧回调一次
    while (newPosCount > 0) {
        size_t temp = newPosition;
        mCbf(EVENT_NEW_POS, mUserData, &temp);
        newPosition += updatePeriod;
        newPosCount--;
    }

    if (mObservedSequence != sequence) {
        mObservedSequence = sequence;
        mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
        // for offloaded tracks, just wait for the upper layers to recreate the track
        if (isOffloadedOrDirect()) {
            return NS_INACTIVE;
        }
    }

    // if inactive, then don't run me again until re-started
    if (!active) {
        return NS_INACTIVE;
    }

    //...

    // EVENT_MORE_DATA callback handling.
    // Timing for linear pcm audio data formats can be derived directly from the
    // buffer fill level.
    // Timing for compressed data is not directly available from the buffer fill level,
    // rather indirectly from waiting for blocking mode callbacks or waiting for obtain()
    // to return a certain fill level.

    struct timespec timeout;
    const struct timespec *requested = &ClientProxy::kForever;
    if (ns != NS_WHENEVER) {
        timeout.tv_sec = ns / 1000000000LL;
        timeout.tv_nsec = ns % 1000000000LL;
        ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000);
        requested = &timeout;
    }

    //如果还有音频流没有写完
    while (mRemainingFrames > 0) {

        Buffer audioBuffer;
        audioBuffer.frameCount = mRemainingFrames;
        size_t nonContig;
        //向共享内存块申请 一块可用的 buffer.
        status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
        LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
                "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
        requested = &ClientProxy::kNonBlocking;
        size_t avail = audioBuffer.frameCount + nonContig;
        ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
                mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
        if (err != NO_ERROR) {
            if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
                    (isOffloaded() && (err == DEAD_OBJECT))) {
                // FIXME bug 25195759
                return 1000000;
            }
            ALOGE("Error %d obtaining an audio buffer, giving up.", err);
            return NS_NEVER;
        }

        if (mRetryOnPartialBuffer && audio_is_linear_pcm(mFormat)) {
            mRetryOnPartialBuffer = false;
            if (avail < mRemainingFrames) {
                if (ns > 0) { // account for obtain time
                    const nsecs_t timeNow = systemTime();
                    ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks));
                }
                nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed);