Android audio 五 AudioTrack 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

该小节简要分析 AudioTrack.cpp 

文件：

frameworks/av/media/libmedia/AudioTrack.cpp

 

AudioTrack native 方法实现的播放音频的逻辑和 java 层类似，详情参考上小节 AudioTrack 播放音频的步骤。

 

先看下 AudioTrack 本地方法的构造和析构函数 ：

在构造函数中调用 set(......)函数完成 AudioTrack 的初始化。

AudioTrack::AudioTrack()
    : mStatus(NO_INIT),
      mState(STATE_STOPPED),
      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, "");
}

AudioTrack::AudioTrack(
        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,
        int32_t notificationFrames,
        audio_session_t sessionId,
        transfer_type transferType,
        const audio_offload_info_t *offloadInfo,
        int uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        bool doNotReconnect,
        float maxRequiredSpeed)
    : mStatus(NO_INIT),
      mState(STATE_STOPPED),
      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
      mPreviousSchedulingGroup(SP_DEFAULT),
      mPausedPosition(0),
      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
    mStatus = set(streamType, sampleRate, format, channelMask,
            frameCount, flags, cbf, user, notificationFrames,
            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
            offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
}

AudioTrack::AudioTrack(
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        const sp<IMemory>& sharedBuffer,
        audio_output_flags_t flags,
        callback_t cbf,
        void* user,
        int32_t notificationFrames,
        audio_session_t sessionId,
        transfer_type transferType,
        const audio_offload_info_t *offloadInfo,
        int uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        bool doNotReconnect,
        float maxRequiredSpeed)
    : mStatus(NO_INIT),
      mState(STATE_STOPPED),
      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
      mPreviousSchedulingGroup(SP_DEFAULT),
      mPausedPosition(0),
      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE)
{
    mStatus = set(streamType, sampleRate, format, channelMask,
            0 /*frameCount*/, flags, cbf, user, notificationFrames,
            sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
            uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
}

AudioTrack::~AudioTrack()
{
    if (mStatus == NO_ERROR) {
        // Make sure that callback function exits in the case where
        // it is looping on buffer full condition in obtainBuffer().
        // Otherwise the callback thread will never exit.
        stop();
        if (mAudioTrackThread != 0) {
            mProxy->interrupt();
            mAudioTrackThread->requestExit();   // see comment in AudioTrack.h
            mAudioTrackThread->requestExitAndWait();
            mAudioTrackThread.clear();
        }
        // No lock here: worst case we remove a NULL callback which will be a nop
        if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) {
            AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput);
        }
        IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this);
        mAudioTrack.clear();
        mCblkMemory.clear();
        mSharedBuffer.clear();
        IPCThreadState::self()->flushCommands();
        ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d",
                mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid);
        AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
    }
}

 

AudioTrack::set

AudioTrack::set的代码很长，这里简要叙述下，不做贴出源码。

1. 根据 transferType 选择传输模式 TRANSFER_SHARED/TRANSFER_SYNC/TRANSFER_CALLBACK 
   默认使用 AudioTrack::TRANSFER_SYNC ；
2. 检查输出音频的格式和音频的通道数是否合法；
3. 创建并运行放音线程 AudioTrackThread ；
4. 调用 AudioTrack::createTrack_l()；
   1. 检查传输的音频参数和 AudioSystem 的参数是否匹配；
   2. 调用 audioFlinger->createTrack(......);   Android 的音频离不开 AudioFlinger ，
      AudioFlinger 是承上启下的作用，承上(为上层提供访问接口)，启下(通过HAL来管理音频设备)。是整个音频系统的核心与难点。
5. 使用 binder 创建一个链接 Service/Client ： 

       mDeathNotifier = new DeathNotifier(this);
       IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);

 

AudioTrack::write（播放音频）

write(......) 调用 AudioTrack::obtainBuffer(......) 发送数据

AudioTrack::obtainBuffer(......) 中使用 AudioTrackClientProxy::obtainBuffer(......) 发送音频数据。

virtual status_t AudioTrackClientProxy::obtainBuffer(Buffer* buffer, bool ackFlush = false);

这里用到了 virtual 虚函数，放音服务的成员实现 obtainBuffer(......) 的业务逻辑，即播放音频。

ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
{
    if (mTransfer != TRANSFER_SYNC) {
        return INVALID_OPERATION;
    }

    if (isDirect()) {
        AutoMutex lock(mLock);
        int32_t flags = android_atomic_and(
                            ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
                            &mCblk->mFlags);
        if (flags & CBLK_INVALID) {
            return DEAD_OBJECT;
        }
    }

    if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
        // Sanity-check: user is most-likely passing an error code, and it would
        // make the return value ambiguous (actualSize vs error).
        ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);
        return BAD_VALUE;
    }

    size_t written = 0;
    Buffer audioBuffer;

    while (userSize >= mFrameSize) {
        audioBuffer.frameCount = userSize / mFrameSize;
        // 发送数据
        status_t err = obtainBuffer(&audioBuffer,
                blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking);
        if (err < 0) {
            if (written > 0) {
                break;
            }
            return ssize_t(err);
        }

        size_t toWrite = audioBuffer.size;
        memcpy(audioBuffer.i8, buffer, toWrite);
        buffer = ((const char *) buffer) + toWrite;
        userSize -= toWrite;
        written += toWrite;

        releaseBuffer(&audioBuffer);
    }

    if (written > 0) {
        mFramesWritten += written / mFrameSize;
    }
    return written;
}