渲染模块的主要功能如下。

1. 将音视频原始数据缓存到队列。
2. 音频数据消耗播放。
3. 视频数据消耗显示。
4. 音视频同步。
5. 播放器控制。

下面将音视频原始数据 缓存到队列。在\frameworks\av\media\libmediaplayerservice\

nuplayer'NuPlayerRenderer.cpp中，存在一个QueueEntry结构体和两个队列，代码如下：
  struct QueueEntry {
        sp<ABuffer> mBuffer;
        sp<AMessage> mNotifyConsumed;
        size_t mOffset;
        status_t mFinalResult;
        int32_t mBufferOrdinal;
    };
List<QueueEntry> mAudioQueue; //音频缓存队列
List<QueueEntry> mVideoQueue; //视频缓存队列
数据是在onQueueBuffer函数中进行添加：
void NuPlayer::Renderer::onQueueBuffer(const sp<AMessage> &msg) {
    int32_t audio;
    CHECK(msg->findInt32("audio", &audio));

    if (dropBufferIfStale(audio, msg)) {
        return;
    }

    if (audio) {
        mHasAudio = true;//音频
    } else {
        mHasVideo = true;//视频
    }

    if (mHasVideo) {
        if (mVideoScheduler == NULL) {
            mVideoScheduler = new VideoFrameScheduler();
            mVideoScheduler->init();
        }
    }

    sp<ABuffer> buffer;
    CHECK(msg->findBuffer("buffer", &buffer));

    sp<AMessage> notifyConsumed;
    CHECK(msg->findMessage("notifyConsumed", &notifyConsumed));

    QueueEntry entry;
    entry.mBuffer = buffer;
    entry.mNotifyConsumed = notifyConsumed;
    entry.mOffset = 0;
    entry.mFinalResult = OK;
    entry.mBufferOrdinal = ++mTotalBuffersQueued;

    if (audio) {
       Mutex::Autolock autoLock(mLock);
        mAudioQueue.push_back(entry);
        postDrainAudioQueue_l();
    } else {
        mVideoQueue.push_back(entry);
       postDrainVideoQueue();
    }

    Mutex::Autolock autoLock(mLock);
    if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
        return;
    }

    sp<ABuffer> firstAudioBuffer = (*mAudioQueue.begin()).mBuffer;
    sp<ABuffer> firstVideoBuffer = (*mVideoQueue.begin()).mBuffer;

    if (firstAudioBuffer == NULL || firstVideoBuffer == NULL) {
        // EOS signalled on either queue.
        syncQueuesDone_l();
        return;
    }

    int64_t firstAudioTimeUs;
    int64_t firstVideoTimeUs;
    CHECK(firstAudioBuffer->meta()
           ->findInt64("timeUs", &firstAudioTimeUs));
    CHECK(firstVideoBuffer->meta()
            ->findInt64("timeUs", &firstVideoTimeUs));

   int64_t diff = firstVideoTimeUs - firstAudioTimeUs;

    ALOGV("queueDiff = %.2f secs", diff / 1E6);

    if (diff > 100000ll) {
        // Audio data starts More than 0.1 secs before video.
       // Drop some audio.

        (*mAudioQueue.begin()).mNotifyConsumed->post();
        mAudioQueue.erase(mAudioQueue.begin());
        return;
    }

    syncQueuesDone_l();
}

那么音频是怎么播放的呢？在NuPlayerRenderer中，会先调用openAudioSink函数
void NuPlayer::Renderer::onAudioTearDown(AudioTearDownReason reason) {
    if (mAudioTornDown) {
        return;
    }
    mAudioTornDown = true;

    int64_t currentPositionUs;
    sp<AMessage> notify = mNotify->dup();
    if (getCurrentPosition(&currentPositionUs) == OK) {
        notify->setInt64("positionUs", currentPositionUs);
    }

    mAudioSink->stop();
    mAudioSink->flush();

    notify->setInt32("what", kWhatAudioTearDown);
    notify->setInt32("reason", reason);
    notify->post();
}

void NuPlayer::Renderer::startAudioOffloadPauseTimeout() {
    if (offloadingAudio()) {
        mWakeLock->acquire();
        sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, this);
        msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
        msg->post(kOffloadPauseMaxUs);
    }
}

void NuPlayer::Renderer::cancelAudioOffloadPauseTimeout() {
    if (offloadingAudio()) {
        mWakeLock->release(true);
        ++mAudioOffloadPauseTimeoutGeneration;
    }
}

status_t NuPlayer::Renderer::onOpenAudioSink(
        const sp<AMessage> &format,
       bool offloadOnly,
        bool hasVideo,
       uint32_t flags) {
    ALOGV("openAudioSink: offloadOnly(%d) offloadingAudio(%d)",
            offloadOnly, offloadingAudio());
    bool audioSinkChanged = false;

    int32_t numChannels;
    CHECK(format->findInt32("channel-count", &numChannels));

    int32_t channelMask;
    if (!format->findInt32("channel-mask", &channelMask)) {
        // signal to the AudioSink to derive the mask from count.
        channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
    }

    int32_t sampleRate;
    CHECK(format->findInt32("sample-rate", &sampleRate));

    if (offloadingAudio()) {
        audio_format_t audioFormat = AUDIO_FORMAT_PCM_16_BIT;
       AString mime;
        CHECK(format->findString("mime", &mime));
        status_t err = mapMimeToAudioFormat(audioFormat, mime.c_str());

        if (err != OK) {
            ALOGE("Couldn't map mime \"%s\" to a valid "
                    "audio_format", mime.c_str());
            onDisableOffloadAudio();
        } else {
            ALOGV("Mime \"%s\" mapped to audio_format 0x%x",
                   mime.c_str(), audioFormat);

            int avgBitRate = -1;
            format->findInt32("bit-rate", &avgBitRate);

            int32_t aacProfile = -1;
           if (audioFormat == AUDIO_FORMAT_AAC
                    && format->findInt32("aac-profile", &aacProfile)) {
                // Redefine AAC format as per aac profile
               mapAACProfileToAudioFormat(
                        audioFormat,
                        aacProfile);
            }

            audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
            offloadInfo.duration_us = -1;
           format->findInt64(
                    "durationUs", &offloadInfo.duration_us);
            offloadInfo.sample_rate = sampleRate;
           offloadInfo.channel_mask = channelMask;
            offloadInfo.format = audioFormat;
            offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
            offloadInfo.bit_rate = avgBitRate;
           offloadInfo.has_video = hasVideo;
            offloadInfo.is_streaming = true;

           if (memcmp(&mCurrentOffloadInfo, &offloadInfo, sizeof(offloadInfo)) == 0) {
                ALOGV("openAudioSink: no change in offload mode");
                // no change from previous configuration, everything ok.
                return OK;
            }
            mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;

            ALOGV("openAudioSink: try to open AudioSink in offload mode");
            uint32_t offloadFlags = flags;
            offloadFlags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
            offloadFlags &= ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
            audioSinkChanged = true;
            mAudioSink->close();

            err = mAudioSink->open(
                    sampleRate,
                    numChannels,
                    (audio_channel_mask_t)channelMask,
                    audioFormat,
                    0 /* bufferCount - unused */,
                   &NuPlayer::Renderer::AudioSinkCallback,
                    this,
                    (audio_output_flags_t)offloadFlags,
                    &offloadInfo);

            if (err == OK) {
               err = mAudioSink->setPlaybackRate(mPlaybackSettings);
            }

           if (err == OK) {
                // If the playback is offloaded to h/w, we pass
                // the HAL some metadata information.
                // We don't want to do this for PCM because it
               // will be going through the AudioFlinger mixer
                // before reaching the hardware.
                // TODO
                mCurrentOffloadInfo = offloadInfo;
                if (!mPaused) { // for preview mode, don't start if paused
                   err = mAudioSink->start();
               }
                ALOGV_IF(err == OK, "openAudioSink: offload succeeded");
            }
            if (err != OK) {
                // Clean up, fall back to non offload mode.
                mAudioSink->close();
                onDisableOffloadAudio();
                mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
                ALOGV("openAudioSink: offload failed");
            } else {
               mUseAudioCallback = true;  // offload mode transfers data through callback
                ++mAudioDrainGeneration;  // discard pending kWhatDrainAudioQueue message.
            }
       }
    }
    if (!offloadOnly && !offloadingAudio()) {
        ALOGV("openAudioSink: open AudioSink in NON-offload mode");
        uint32_t pcmFlags = flags;
        pcmFlags &= ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;

        const PcmInfo info = {
                (audio_channel_mask_t)channelMask,
                (audio_output_flags_t)pcmFlags,
                AUDIO_FORMAT_PCM_16_BIT, // TODO: change to audioFormat
                numChannels,
                sampleRate
       };
        if (memcmp(&mCurrentPcmInfo, &info, sizeof(info)) == 0) {
            ALOGV("openAudioSink: no change in pcm mode");
            // no change from previous configuration, everything ok.
           return OK;
        }

        audioSinkChanged = true;
        mAudioSink->close();
        mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
        // Note: It is possible to set up the callback, but not use it to send audio data.
        // This requires a fix in AudioSink to explicitly specify the transfer mode.
        mUseAudioCallback = getUseAudioCallbackSetting();
        if (mUseAudioCallback) {
           ++mAudioDrainGeneration;  // discard pending kWhatDrainAudioQueue message.
        }

        // Compute the desired buffer size.
        // For callback mode, the amount of time before wakeup is about half the buffer size.
       const uint32_t frameCount =
                (unsigned long long)sampleRate * getAudioSinkPcmMsSetting() / 1000;

        // The doNotReconnect means AudioSink will signal back and let NuPlayer to re-construct
        // AudioSink. We don't want this when there's video because it will cause a video seek to
        // the previous I frame. But we do want this when there's only audio because it will give
        // NuPlayer a chance to switch from non-offload mode to offload mode.
        // So we only set doNotReconnect when there's no video.
       const bool doNotReconnect = !hasVideo;
       status_t err = mAudioSink->open(
                    sampleRate,
                    numChannels,
                    (audio_channel_mask_t)channelMask,
                    AUDIO_FORMAT_PCM_16_BIT,
                    0 /* bufferCount - unused */,
                    mUseAudioCallback ? &NuPlayer::Renderer::AudioSinkCallback : NULL,
                    mUseAudioCallback ? this : NULL,
                    (audio_output_flags_t)pcmFlags,
                    NULL,
                    doNotReconnect,
                    frameCount);
       if (err == OK) {
            err = mAudioSink->setPlaybackRate(mPlaybackSettings);
        }
        if (err != OK) {
            ALOGW("openAudioSink: non offloaded open failed status: %d", err);
            mAudioSink->close();
           mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
            return err;
       }
        mCurrentPcmInfo = info;
        if (!mPaused) { // for preview mode, don't start if paused
           mAudioSink->start();
        }
    }
    if (audioSinkChanged) {
        onAudioSinkChanged();
    }
    mAudioTornDown = false;
    return OK;
}



void NuPlayer::Renderer::postDrainAudioQueue_l(int64_t delayUs) {
    if (mDrainAudioQueuePending || mSyncQueues || mUseAudioCallback) {
        return;
   }

   if (mAudioQueue.empty()) {
        return;
   }

    // FIXME: if paused, wait until AudioTrack stop() is complete before delivering data.
    if (mPaused) {
        const int64_t diffUs = mPauseDrainAudioAllowedUs - ALooper::GetNowUs();
       if (diffUs > delayUs) {
            delayUs = diffUs;
        }
    }

    mDrainAudioQueuePending = true;
    sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, this);
    msg->setInt32("drainGeneration", mAudioDrainGeneration);
    msg->post(delayUs);
}

上面主要是发送了一个消息kWhatDrainAudioQueue,找到对应接收消息的地方，代码如 下：

   case kWhatDrainAudioQueue:
        {
            mDrainAudioQueuePending = false;

            int32_t generation;
            CHECK(msg->findInt32("drainGeneration", &generation));
            if (generation != getDrainGeneration(true /* audio */)) {
                break;
            }

            if (onDrainAudioQueue()) {
                uint32_t numFramesPlayed;
                CHECK_EQ(mAudioSink->getPosition(&numFramesPlayed),
                         (status_t)OK);

                uint32_t numFramesPendingPlayout =
                    mNumFramesWritten - numFramesPlayed;

                // This is how long the audio sink will have data to
                // play back.
                int64_t delayUs =
                   mAudioSink->msecsPerFrame()
                        * numFramesPendingPlayout * 1000ll;
               if (mPlaybackRate > 1.0f) {
                    delayUs /= mPlaybackRate;
                }

               // Let's give it more data after about half that time
                // has elapsed.
                Mutex::Autolock autoLock(mLock);
                postDrainAudioQueue_l(delayUs / 2);
            }
          break;
       }

主要是有一个进行判断的onDrainAudioQueue函数，判断是否需要重新向AudioSink写入 数据，代码如下：

bool NuPlayer::Renderer::onDrainAudioQueue() {
    // do not drain audio during teardown as queued buffers may be invalid.
    if (mAudioTornDown) {
        return false;
    }
    // TODO: This call to getPosition checks if AudioTrack has been created
    // in AudioSink before draining audio. If AudioTrack doesn't exist, then
    // CHECKs on getPosition will fail.
    // We still need to figure out why AudioTrack is not created when
    // this function is called. One possible reason could be leftover
    // audio. Another possible place is to check whether decoder
    // has received INFO_FORMAT_CHANGED as the first buffer since
    // AudioSink is opened there, and possible interactions with flush
    // immediately after start. Investigate error message
    // "vorbis_dsp_synthesis returned -135", along with RTSP.
    uint32_t numFramesPlayed;
    if (mAudioSink->getPosition(&numFramesPlayed) != OK) {
        // When getPosition fails, renderer will not reschedule the draining
        // unless new samples are queued.
       // If we have pending EOS (or "eos" marker for discontinuities), we need
        // to post these now as NuPlayerDecoder might be waiting for it.
        drainAudioQueueUntilLastEOS();

        ALOGW("onDrainAudioQueue(): audio sink is not ready");
        return false;
    }

#if 0
    ssize_t numFramesAvailableToWrite =
        mAudioSink->frameCount() - (mNumFramesWritten - numFramesPlayed);

    if (numFramesAvailableToWrite == mAudioSink->frameCount()) {
        ALOGI("audio sink underrun");
    } else {
        ALOGV("audio queue has %d frames left to play",
             mAudioSink->frameCount() - numFramesAvailableToWrite);
    }
#endif

    uint32_t prevFramesWritten = mNumFramesWritten;
    while (!mAudioQueue.empty()) {
        QueueEntry *entry = &*mAudioQueue.begin();

        mLastAudioBufferDrained = entry->mBufferOrdinal;

        if (entry->mBuffer == NULL) {
            // EOS
            int64_t postEOSDelayUs = 0;
            if (mAudioSink->needsTrailingPadding()) {
                postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs());
           }
            notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs);

            mAudioQueue.erase(mAudioQueue.begin());
            entry = NULL;
           if (mAudioSink->needsTrailingPadding()) {
               // If we're not in gapless playback (i.e. through setNextPlayer), we
                // need to stop the track here, because that will play out the last
              // little bit at the end of the file. Otherwise short files won't play.
                mAudioSink->stop();
                mNumFramesWritten = 0;
            }
            return false;
        }

        // ignore 0-sized buffer which could be EOS marker with no data
       if (entry->mOffset == 0 && entry->mBuffer->size() > 0) {
           int64_t mediaTimeUs;
            CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
            ALOGV("onDrainAudioQueue: rendering audio at media time %.2f secs",
                  mediaTimeUs / 1E6);
            onNewAudioMediaTime(mediaTimeUs);
       }

        size_t copy = entry->mBuffer->size() - entry->mOffset;

        ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset,
                                            copy, false /* blocking */);
        if (written < 0) {
            // An error in AudioSink write. Perhaps the AudioSink was not properly opened.
            if (written == WOULD_BLOCK) {
                ALOGV("AudioSink write would block when writing %zu bytes", copy);
            } else {
                ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy);
                // This can only happen when AudioSink was opened with doNotReconnect flag set to
                // true, in which case the NuPlayer will handle the reconnect.
                notifyAudioTearDown();
            }
            break;
        }

        entry->mOffset += written;
        if (entry->mOffset == entry->mBuffer->size()) {
            entry->mNotifyConsumed->post();
            mAudioQueue.erase(mAudioQueue.begin());

           entry = NULL;
      }

       size_t copiedFrames = written / mAudioSink->frameSize();
        mNumFramesWritten += copiedFrames;

       {
            Mutex::Autolock autoLock(mLock);
           int64_t maxTimeMedia;
            maxTimeMedia =
               mAnchorTimeMediaUs +
                        (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
                               * 1000LL * mAudioSink->msecsPerFrame());
           mMediaClock->updateMaxTimeMedia(maxTimeMedia);

            notifyIfMediaRenderingStarted_l();
        }

        if (written != (ssize_t)copy) {
            // A short count was received from AudioSink::write()
            //
            // AudioSink write is called in non-blocking mode.
            // It may return with a short count when:
            //
           // 1) Size to be copied is not a multiple of the frame size. We consider this fatal.
            // 2) The data to be copied exceeds the available buffer in AudioSink.
           // 3) An error occurs and data has been partially copied to the buffer in AudioSink.
            // 4) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded.

            // (Case 1)
          // Must be a multiple of the frame size.  If it is not a multiple of a frame size, it
           // needs to fail, as we should not carry over fractional frames between calls.
            CHECK_EQ(copy % mAudioSink->frameSize(), 0);

            // (Case 2, 3, 4)
            // Return early to the caller.
            // Beware of calling immediately again as this may busy-loop if you are not careful.
            ALOGV("AudioSink write short frame count %zd < %zu", written, copy);
           break;
       }
    }

    // calculate whether we need to reschedule another write.
    bool reschedule = !mAudioQueue.empty()
            && (!mPaused
                || prevFramesWritten != mNumFramesWritten); // permit pause to fill buffers
    //ALOGD("reschedule:%d  empty:%d  mPaused:%d  prevFramesWritten:%u  mNumFramesWritten:%u",
    //        reschedule, mAudioQueue.empty(), mPaused, prevFramesWritten, mNumFramesWritten);
    return reschedule;
}

到这里，已经很清楚了，音频播放流程如下：先打开音频后端，然后当向音频队列中发送 数据时，音频队列同时向音频后端写入数据，以供播放音频。
那么视频显示呢？同样是在视频原始数据进入视频队列后，开始执行postDrain VideoQueue 函数：

void NuPlayer::Renderer::postDrainVideoQueue() {
   if (mDrainVideoQueuePending
            || getSyncQueues()
           || (mPaused && mVideoSampleReceived)) {
        return;
    }

    if (mVideoQueue.empty()) {
        return;
    }

    QueueEntry &entry = *mVideoQueue.begin();

    sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, this);
    msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));

    if (entry.mBuffer == NULL) {
        // EOS doesn't carry a timestamp.
        msg->post();
        mDrainVideoQueuePending = true;
        return;
    }

    int64_t delayUs;
    int64_t nowUs = ALooper::GetNowUs();
    int64_t realTimeUs;
    if (mFlags & FLAG_REAL_TIME) {
        int64_t mediaTimeUs;
        CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
        realTimeUs = mediaTimeUs;
    } else {
       int64_t mediaTimeUs;
        CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));

        {
            Mutex::Autolock autoLock(mLock);
           if (mAnchorTimeMediaUs < 0) {
                mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
                mAnchorTimeMediaUs = mediaTimeUs;
                realTimeUs = nowUs;
            } else {
                realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
           }
        }
        if (!mHasAudio) {
            // smooth out videos >= 10fps
            mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
        }

        // Heuristics to handle situation when media time changed without a
        // discontinuity. If we have not drained an audio buffer that was
        // received after this buffer, repost in 10 msec. Otherwise repost
        // in 500 msec.
        delayUs = realTimeUs - nowUs;
        if (delayUs > 500000) {
            int64_t postDelayUs = 500000;
           if (mHasAudio && (mLastAudioBufferDrained - entry.mBufferOrdinal) <= 0) {
                postDelayUs = 10000;
            }
            msg->setWhat(kWhatPostDrainVideoQueue);
            msg->post(postDelayUs);
           mVideoScheduler->restart();
            ALOGI("possible video time jump of %dms, retrying in %dms",
                    (int)(delayUs / 1000), (int)(postDelayUs / 1000));
            mDrainVideoQueuePending = true;
            return;
        }
    }

    realTimeUs = mVideoScheduler->schedule(realTimeUs * 1000) / 1000;
    int64_t twoVsyncsUs = 2 * (mVideoScheduler->getVsyncPeriod() / 1000);

    delayUs = realTimeUs - nowUs;

    ALOGW_IF(delayUs > 500000, "unusually high delayUs: %" PRId64, delayUs);
    // post 2 display refreshes before rendering is due
    msg->post(delayUs > twoVsyncsUs ? delayUs - twoVsyncsUs : 0);

    mDrainVideoQueuePending = true;
}

这里先发送一个kWhatDrainVideoQueue的消息，然后进行音视频同步。发送kWhatDrain- VideoQueue消息后，会触发调用onDrainVideoQueue函数：

void NuPlayer::Renderer::onDrainVideoQueue() {
    if (mVideoQueue.empty()) {
        return;
    }

    QueueEntry *entry = &*mVideoQueue.begin();

    if (entry->mBuffer == NULL) {
        // EOS

       notifyEOS(false /* audio */, entry->mFinalResult);

        mVideoQueue.erase(mVideoQueue.begin());
        entry = NULL;

        setVideoLateByUs(0);
        return;
    }

    int64_t nowUs = -1;
    int64_t realTimeUs;
    if (mFlags & FLAG_REAL_TIME) {
        CHECK(entry->mBuffer->meta()->findInt64("timeUs", &realTimeUs));
    } else {
        int64_t mediaTimeUs;
        CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));

        nowUs = ALooper::GetNowUs();
        realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
    }

    bool tooLate = false;

    if (!mPaused) {
       if (nowUs == -1) {
            nowUs = ALooper::GetNowUs();
       }
        setVideoLateByUs(nowUs - realTimeUs);
        tooLate = (mVideoLateByUs > 40000);

        if (tooLate) {
            ALOGV("video late by %lld us (%.2f secs)",
                 (long long)mVideoLateByUs, mVideoLateByUs / 1E6);
        } else {
            int64_t mediaUs = 0;
            mMediaClock->getMediaTime(realTimeUs, &mediaUs);
            ALOGV("rendering video at media time %.2f secs",
                    (mFlags & FLAG_REAL_TIME ? realTimeUs :
                    mediaUs) / 1E6);
        }
    } else {
        setVideoLateByUs(0);
        if (!mVideoSampleReceived && !mHasAudio) {
            // This will ensure that the first frame after a flush won't be used as anchor
           // when renderer is in paused state, because resume can happen any time after seek.
            Mutex::Autolock autoLock(mLock);
            clearAnchorTime_l();
        }
    }

   entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000ll);
    entry->mNotifyConsumed->setInt32("render", !tooLate);
   entry->mNotifyConsumed->post();
    mVideoQueue.erase(mVideoQueue.begin());
    entry = NULL;

    mVideoSampleReceived = true;

    if (!mPaused) {
        if (!mVideoRenderingStarted) {
            mVideoRenderingStarted = true;
            notifyVideoRenderingStart();
        }
        Mutex::Autolock autoLock(mLock);
        notifyIfMediaRenderingStarted_l();//向上通知开始渲染视频
    }
}

NuPlayer::Renderer使用的是以视频为基准的同步机制，解码后的音频数据时间戳如果大 于视频数据时间戳，直接丢弃音频包，然后直接渲染视频。同步机制主要位于视频缓冲区处理 部分的onDrainVideoQueue和音频缓冲区处理部分的onDrainVideoQueue中。音视频的渲染都 釆用类似定时器的机制，只不过视频显示需要依赖于实际解码器，音频播放需要依赖于 AudioSink 的接口。