Audio笔记之AudioPlayer流程

//下面是一个典型的播放序列:  
MediaPlayer player=new MediaPlayer()  
player->setDataSource(url,header);  
player->prepare();  
player->start();  
...  


在player阶段开始创建AudioPlayer对象,同时传入AudioSink作为输出,实际上是AudioOutput对象
并启动AudioOutput对象

status_t AwesomePlayer::play_l() {
    modifyFlags(SEEK_PREVIEW, CLEAR);

    if (mFlags & PLAYING) {
        return OK;
    }

    if (!(mFlags & PREPARED)) {
        status_t err = prepare_l();

        if (err != OK) {
            return err;
        }
    }

    modifyFlags(PLAYING, SET);
    modifyFlags(FIRST_FRAME, SET);

    if (mDecryptHandle != NULL) {
        int64_t position;
        getPosition(&position);
        mDrmManagerClient->setPlaybackStatus(mDecryptHandle,
                Playback::START, position / 1000);
    }

    if (mAudioSource != NULL) {
        if (mAudioPlayer == NULL) {
            if (mAudioSink != NULL) {
                bool allowDeepBuffering;
                int64_t cachedDurationUs;
                bool eos;
                if (mVideoSource == NULL
                        && (mDurationUs > AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US ||
                        (getCachedDuration_l(&cachedDurationUs, &eos) &&
                        cachedDurationUs > AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US))) {
                    allowDeepBuffering = true;
                } else {
                    allowDeepBuffering = false;
                }
                //创建AudioPlayer对象,将mAudioSink作为输出
                mAudioPlayer = new AudioPlayer(mAudioSink, allowDeepBuffering, this);
                mAudioPlayer->setSource(mAudioSource);

                mTimeSource = mAudioPlayer;

                // If there was a seek request before we ever started,
                // honor the request now.
                // Make sure to do this before starting the audio player
                // to avoid a race condition.
                seekAudioIfNecessary_l();
            }
        }

        CHECK(!(mFlags & AUDIO_RUNNING));

        if (mVideoSource == NULL) {
            // We don't want to post an error notification at this point,
            // the error returned from MediaPlayer::start() will suffice.
            // 开始启动AudioPlayer
            status_t err = startAudioPlayer_l(
                    false /* sendErrorNotification */);

            if (err != OK) {
                delete mAudioPlayer;
                mAudioPlayer = NULL;

                modifyFlags((PLAYING | FIRST_FRAME), CLEAR);

                if (mDecryptHandle != NULL) {
                    mDrmManagerClient->setPlaybackStatus(
                            mDecryptHandle, Playback::STOP, 0);
                }

                return err;
            }
        }
    }

    if (mTimeSource == NULL && mAudioPlayer == NULL) {
        mTimeSource = &mSystemTimeSource;
    }

    if (mVideoSource != NULL) {
        // Kick off video playback
        postVideoEvent_l();

        if (mAudioSource != NULL && mVideoSource != NULL) {
            postVideoLagEvent_l();
        }
    }

    if (mFlags & AT_EOS) {
        // Legacy behaviour, if a stream finishes playing and then
        // is started again, we play from the start...
        seekTo_l(0);
    }

    uint32_t params = IMediaPlayerService::kBatteryDataCodecStarted
        | IMediaPlayerService::kBatteryDataTrackDecoder;
    if ((mAudioSource != NULL) && (mAudioSource != mAudioTrack)) {
        params |= IMediaPlayerService::kBatteryDataTrackAudio;
    }
    if (mVideoSource != NULL) {
        params |= IMediaPlayerService::kBatteryDataTrackVideo;
    }
    addBatteryData(params);

    return OK;
}

在setDataSource阶段调用时创建的AudioOutput对象,并赋值给AwesomePlayer对象的
mAudioSink变量和MediaPlayer对象的mAudioOutput变量

sp<MediaPlayerBase> MediaPlayerService::Client::setDataSource_pre(
        player_type playerType)
{
    ALOGV("player type = %d", playerType);

    // create the right type of player
    sp<MediaPlayerBase> p = createPlayer(playerType);
    if (p == NULL) {
        return p;
    }

    if (!p->hardwareOutput()) {
        mAudioOutput = new AudioOutput(mAudioSessionId);
        static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
    }

    return p;
}
在player阶段构造完AudioPlayer之后,启动AudioPlayer对象
status_t AwesomePlayer::startAudioPlayer_l(bool sendErrorNotification) {
    CHECK(!(mFlags & AUDIO_RUNNING));

    if (mAudioSource == NULL || mAudioPlayer == NULL) {
        return OK;
    }

    if (!(mFlags & AUDIOPLAYER_STARTED)) {
        // 
        bool wasSeeking = mAudioPlayer->isSeeking();

        // We've already started the MediaSource in order to enable
        // the prefetcher to read its data.
        status_t err = mAudioPlayer->start(
                true /* sourceAlreadyStarted */);

        if (err != OK) {
            if (sendErrorNotification) {
                notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
            }

            return err;
        }

        modifyFlags(AUDIOPLAYER_STARTED, SET);

        if (wasSeeking) {
            CHECK(!mAudioPlayer->isSeeking());

            // We will have finished the seek while starting the audio player.
            postAudioSeekComplete();
        }
    } else {
        mAudioPlayer->resume();
    }

    modifyFlags(AUDIO_RUNNING, SET);

    mWatchForAudioEOS = true;

    return OK;
}

status_t AudioPlayer::start(bool sourceAlreadyStarted) {
    CHECK(!mStarted);
    CHECK(mSource != NULL);

    status_t err;
    if (!sourceAlreadyStarted) {
        err = mSource->start();

        if (err != OK) {
            return err;
        }
    }

    // We allow an optional INFO_FORMAT_CHANGED at the very beginning
    // of playback, if there is one, getFormat below will retrieve the
    // updated format, if there isn't, we'll stash away the valid buffer
    // of data to be used on the first audio callback.

    CHECK(mFirstBuffer == NULL);

    MediaSource::ReadOptions options;
    if (mSeeking) {
        options.setSeekTo(mSeekTimeUs);
        mSeeking = false;
    }

    mFirstBufferResult = mSource->read(&mFirstBuffer, &options);
    if (mFirstBufferResult == INFO_FORMAT_CHANGED) {
        ALOGV("INFO_FORMAT_CHANGED!!!");

        CHECK(mFirstBuffer == NULL);
        mFirstBufferResult = OK;
        mIsFirstBuffer = false;
    } else {
        mIsFirstBuffer = true;
    }

    sp<MetaData> format = mSource->getFormat();
    const char *mime;
    bool success = format->findCString(kKeyMIMEType, &mime);
    CHECK(success);
    CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW));

    success = format->findInt32(kKeySampleRate, &mSampleRate);
    CHECK(success);

    int32_t numChannels, channelMask;
    success = format->findInt32(kKeyChannelCount, &numChannels);
    CHECK(success);

    if(!format->findInt32(kKeyChannelMask, &channelMask)) {
        // log only when there's a risk of ambiguity of channel mask selection
        ALOGI_IF(numChannels > 2,
                "source format didn't specify channel mask, using (%d) channel order", numChannels);
        channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
    }

    if (mAudioSink.get() != NULL) {
        //启动AudioOutput对象,并注册回调函数,当AudioOutput输出完缓冲区里
        //的数据就会通过回调函数通知AudioPlayer填充数据
        status_t err = mAudioSink->open(
                mSampleRate, numChannels, channelMask, AUDIO_FORMAT_PCM_16_BIT,
                DEFAULT_AUDIOSINK_BUFFERCOUNT,
                &AudioPlayer::AudioSinkCallback,
                this,
                (mAllowDeepBuffering ?
                            AUDIO_OUTPUT_FLAG_DEEP_BUFFER :
                            AUDIO_OUTPUT_FLAG_NONE));
        if (err != OK) {
            if (mFirstBuffer != NULL) {
                mFirstBuffer->release();
                mFirstBuffer = NULL;
            }

            if (!sourceAlreadyStarted) {
                mSource->stop();
            }

            return err;
        }

        mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
        mFrameSize = mAudioSink->frameSize();

        mAudioSink->start();
    } else {
        // playing to an AudioTrack, set up mask if necessary
        audio_channel_mask_t audioMask = channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER ?
                audio_channel_out_mask_from_count(numChannels) : channelMask;
        if (0 == audioMask) {
            return BAD_VALUE;
        }

        mAudioTrack = new AudioTrack(
                AUDIO_STREAM_MUSIC, mSampleRate, AUDIO_FORMAT_PCM_16_BIT, audioMask,
                0, AUDIO_OUTPUT_FLAG_NONE, &AudioCallback, this, 0);

        if ((err = mAudioTrack->initCheck()) != OK) {
            delete mAudioTrack;
            mAudioTrack = NULL;

            if (mFirstBuffer != NULL) {
                mFirstBuffer->release();
                mFirstBuffer = NULL;
            }

            if (!sourceAlreadyStarted) {
                mSource->stop();
            }

            return err;
        }

        mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
        mFrameSize = mAudioTrack->frameSize();
        // 实际调用AudioSink的实现类AudioOutput::start()
        // AudioOutput::start()又调用AudioTrack::start()开始输出数据
        mAudioTrack->start();
    }
    mStarted = true;
    mPinnedTimeUs = -1ll;

    return OK;
}


size_t AudioPlayer::AudioSinkCallback(
        MediaPlayerBase::AudioSink *audioSink,
        void *buffer, size_t size, void *cookie) {
    AudioPlayer *me = (AudioPlayer *)cookie;

    return me->fillBuffer(buffer, size);
}

对于fillBuffer函数,需要关注数据的输入和数据的输出,通过分析可以发现数据的输出就是AudioOutput对象
而数据的输入则是mSource对象,但是需要继续分析该对象的由来。
在有数据到来时,循环调用此callback函数调用fillBuffer函数进行填充数据
此函数的返回值size_done,代表已经处理的数据总大小,与传递进来的数据size(第二个参数)不一定相同

size_t AudioPlayer::fillBuffer(void *data, size_t size) {
    if (mNumFramesPlayed == 0) {
        ALOGV("AudioCallback");
    }

    if (mReachedEOS) {
        return 0;
    }

    bool postSeekComplete = false;
    bool postEOS = false;
    int64_t postEOSDelayUs = 0;

    size_t size_done = 0;
    size_t size_remaining = size;
    while (size_remaining > 0) {
        MediaSource::ReadOptions options;

        {
            Mutex::Autolock autoLock(mLock);

            if (mSeeking) {
                if (mIsFirstBuffer) {
                    if (mFirstBuffer != NULL) {
                        mFirstBuffer->release();
                        mFirstBuffer = NULL;
                    }
                    mIsFirstBuffer = false;
                }

                options.setSeekTo(mSeekTimeUs);

                if (mInputBuffer != NULL) {
                    mInputBuffer->release();
                    mInputBuffer = NULL;
                }

                mSeeking = false;
                if (mObserver) {
                    postSeekComplete = true;
                }
            }
        }

        if (mInputBuffer == NULL) {
            status_t err;

            if (mIsFirstBuffer) {
                mInputBuffer = mFirstBuffer;
                mFirstBuffer = NULL;
                err = mFirstBufferResult;

                mIsFirstBuffer = false;
            } else {
                //读取数据源AudioDecoder的数据
                err = mSource->read(&mInputBuffer, &options);
            }

            CHECK((err == OK && mInputBuffer != NULL)
                   || (err != OK && mInputBuffer == NULL));

            Mutex::Autolock autoLock(mLock);

            if (err != OK) {
                if (mObserver && !mReachedEOS) {
                    // We don't want to post EOS right away but only
                    // after all frames have actually been played out.

                    // These are the number of frames submitted to the
                    // AudioTrack that you haven't heard yet.
                    uint32_t numFramesPendingPlayout =
                        getNumFramesPendingPlayout();

                    // These are the number of frames we're going to
                    // submit to the AudioTrack by returning from this
                    // callback.
                    uint32_t numAdditionalFrames = size_done / mFrameSize;

                    numFramesPendingPlayout += numAdditionalFrames;
                    
                    int64_t timeToCompletionUs =
                        (1000000ll * numFramesPendingPlayout) / mSampleRate;

                    ALOGV("total number of frames played: %lld (%lld us)",
                            (mNumFramesPlayed + numAdditionalFrames),
                            1000000ll * (mNumFramesPlayed + numAdditionalFrames)
                                / mSampleRate);

                    ALOGV("%d frames left to play, %lld us (%.2f secs)",
                         numFramesPendingPlayout,
                         timeToCompletionUs, timeToCompletionUs / 1E6);

                    postEOS = true;
                    if (mAudioSink->needsTrailingPadding()) {
                        postEOSDelayUs = timeToCompletionUs + mLatencyUs;
                    } else {
                        postEOSDelayUs = 0;
                    }
                }

                mReachedEOS = true;
                mFinalStatus = err;
                break;
            }

            if (mAudioSink != NULL) {
                mLatencyUs = (int64_t)mAudioSink->latency() * 1000;
            } else {
                mLatencyUs = (int64_t)mAudioTrack->latency() * 1000;
            }
            <pre name="code" class="cpp">            //取得一帧数据在媒体文件中存储的时间戳mPositionTimeMediaUs
            CHECK(mInputBuffer->meta_data()->findInt64(
                        kKeyTime, &mPositionTimeMediaUs));
            //计算一帧数据实际播放位置的时间戳
            mPositionTimeRealUs =
                ((mNumFramesPlayed + size_done / mFrameSize) * 1000000)
                    / mSampleRate;
            //这两个时间戳,在AwesomePlayer::onVideoEvent()中,用于计算音视频同步的依据
            ALOGV("buffer->size() = %d, "
                 "mPositionTimeMediaUs=%.2f mPositionTimeRealUs=%.2f",
                 mInputBuffer->range_length(),
                 mPositionTimeMediaUs / 1E6, mPositionTimeRealUs / 1E6);
        }

        if (mInputBuffer->range_length() == 0) {
            mInputBuffer->release();
            mInputBuffer = NULL;

            continue;
        }

        size_t copy = size_remaining;
        if (copy > mInputBuffer->range_length()) {
            copy = mInputBuffer->range_length();
        }

        memcpy((char *)data + size_done,
               (const char *)mInputBuffer->data() + mInputBuffer->range_offset(),
               copy);

        mInputBuffer->set_range(mInputBuffer->range_offset() + copy,
                                mInputBuffer->range_length() - copy);

        size_done += copy;
        size_remaining -= copy;
    }

    {
        Mutex::Autolock autoLock(mLock);
        mNumFramesPlayed += size_done / mFrameSize;
        mNumFramesPlayedSysTimeUs = ALooper::GetNowUs();

        if (mReachedEOS) {
            mPinnedTimeUs = mNumFramesPlayedSysTimeUs;
        } else {
            mPinnedTimeUs = -1ll;
        }
    }

    if (postEOS) {
        mObserver->postAudioEOS(postEOSDelayUs);
    }

    if (postSeekComplete) {
        mObserver->postAudioSeekComplete();
    }

    return size_done;
}

在prepare阶段,通过分析发现数据的输入mSource是Decoder之后的数据,该数据是经过OMXCodec构造的Decoder
实现的,具体Decoder需要根据Audio的具体格式来生成,如果是RAW数据,则直接输出。
status_t AwesomePlayer::prepareAsync_l() {

    modifyFlags(PREPARING, SET);
    mAsyncPrepareEvent = new AwesomeEvent(
            this, &AwesomePlayer::onPrepareAsyncEvent);

    mQueue.postEvent(mAsyncPrepareEvent);

    return OK;
}
void AwesomePlayer::onPrepareAsyncEvent() {
    Mutex::Autolock autoLock(mLock);


    if (mVideoTrack != NULL && mVideoSource == NULL) {
        status_t err = initVideoDecoder();

        if (err != OK) {
            abortPrepare(err);
            return;
        }
    }

    if (mAudioTrack != NULL && mAudioSource == NULL) {
        status_t err = initAudioDecoder();

        if (err != OK) {
            abortPrepare(err);
            return;
        }
    }
}
status_t AwesomePlayer::initAudioDecoder() {
    ATRACE_CALL();

    sp<MetaData> meta = mAudioTrack->getFormat();

    const char *mime;
    CHECK(meta->findCString(kKeyMIMEType, &mime));

    if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_RAW)) {
        mAudioSource = mAudioTrack;
    } else {
                // 根据mAudioTrack提取文件的格式生成Decoder
                mAudioSource = OMXCodec::Create(
                mClient.interface(), mAudioTrack->getFormat(),
                false, // createEncoder
                mAudioTrack);
    }

    if (mAudioSource != NULL) {
        int64_t durationUs;
        if (mAudioTrack->getFormat()->findInt64(kKeyDuration, &durationUs)) {
            Mutex::Autolock autoLock(mMiscStateLock);
            if (mDurationUs < 0 || durationUs > mDurationUs) {
                mDurationUs = durationUs;
            }
        }

        status_t err = mAudioSource->start();

        if (err != OK) {
            mAudioSource.clear();
            return err;
        }
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_QCELP)) {
        // For legacy reasons we're simply going to ignore the absence
        // of an audio decoder for QCELP instead of aborting playback
        // altogether.
        return OK;
    }

    if (mAudioSource != NULL) {
        Mutex::Autolock autoLock(mStatsLock);
        TrackStat *stat = &mStats.mTracks.editItemAt(mStats.mAudioTrackIndex);
        const char *component;
        if (!mAudioSource->getFormat()
                ->findCString(kKeyDecoderComponent, &component)) {
            component = "none";
        }

        stat->mDecoderName = component;
    }

    return mAudioSource != NULL ? OK : UNKNOWN_ERROR;
}

void AwesomePlayer::setAudioSource(sp<MediaSource> source) {
    CHECK(source != NULL);

    mAudioTrack = source;
} 
分析mAudioTack的由来,是在AwesomePlayer设置数据源的时候通过extractor解析出来的
status_t AwesomePlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
    // Attempt to approximate overall stream bitrate by summing all
    // tracks' individual bitrates, if not all of them advertise bitrate,
    // we have to fail.

    int64_t totalBitRate = 0;

    mExtractor = extractor;
    for (size_t i = 0; i < extractor->countTracks(); ++i) {
        sp<MetaData> meta = extractor->getTrackMetaData(i);

        int32_t bitrate;
        if (!meta->findInt32(kKeyBitRate, &bitrate)) {
            const char *mime;
            CHECK(meta->findCString(kKeyMIMEType, &mime));
            ALOGV("track of type '%s' does not publish bitrate", mime);

            totalBitRate = -1;
            break;
        }

        totalBitRate += bitrate;
    }

    mBitrate = totalBitRate;

    ALOGV("mBitrate = %lld bits/sec", mBitrate);

    {
        Mutex::Autolock autoLock(mStatsLock);
        mStats.mBitrate = mBitrate;
        mStats.mTracks.clear();
        mStats.mAudioTrackIndex = -1;
        mStats.mVideoTrackIndex = -1;
    }

    bool haveAudio = false;
    bool haveVideo = false;
    for (size_t i = 0; i < extractor->countTracks(); ++i) {
        sp<MetaData> meta = extractor->getTrackMetaData(i);

        const char *_mime;
        CHECK(meta->findCString(kKeyMIMEType, &_mime));

        String8 mime = String8(_mime);

        if (!haveVideo && !strncasecmp(mime.string(), "video/", 6)) {
            setVideoSource(extractor->getTrack(i));
            haveVideo = true;

            // Set the presentation/display size
            int32_t displayWidth, displayHeight;
            bool success = meta->findInt32(kKeyDisplayWidth, &displayWidth);
            if (success) {
                success = meta->findInt32(kKeyDisplayHeight, &displayHeight);
            }
            if (success) {
                mDisplayWidth = displayWidth;
                mDisplayHeight = displayHeight;
            }

            {
                Mutex::Autolock autoLock(mStatsLock);
                mStats.mVideoTrackIndex = mStats.mTracks.size();
                mStats.mTracks.push();
                TrackStat *stat =
                    &mStats.mTracks.editItemAt(mStats.mVideoTrackIndex);
                stat->mMIME = mime.string();
            }
        } else if (!haveAudio && !strncasecmp(mime.string(), "audio/", 6)) {
            //将extractor解析出来的AudioTrack对象赋值给mAudioTrack变量
            setAudioSource(extractor->getTrack(i));
            haveAudio = true;
            mActiveAudioTrackIndex = i;

            {
                Mutex::Autolock autoLock(mStatsLock);
                mStats.mAudioTrackIndex = mStats.mTracks.size();
                mStats.mTracks.push();
                TrackStat *stat =
                    &mStats.mTracks.editItemAt(mStats.mAudioTrackIndex);
                stat->mMIME = mime.string();
            }

            if (!strcasecmp(mime.string(), MEDIA_MIMETYPE_AUDIO_VORBIS)) {
                // Only do this for vorbis audio, none of the other audio
                // formats even support this ringtone specific hack and
                // retrieving the metadata on some extractors may turn out
                // to be very expensive.
                sp<MetaData> fileMeta = extractor->getMetaData();
                int32_t loop;
                if (fileMeta != NULL
                        && fileMeta->findInt32(kKeyAutoLoop, &loop) && loop != 0) {
                    modifyFlags(AUTO_LOOPING, SET);
                }
            }
        } else if (!strcasecmp(mime.string(), MEDIA_MIMETYPE_TEXT_3GPP)) {
            addTextSource_l(i, extractor->getTrack(i));
        }
    }

    if (!haveAudio && !haveVideo) {
        if (mWVMExtractor != NULL) {
            return mWVMExtractor->getError();
        } else {
            return UNKNOWN_ERROR;
        }
    }

    mExtractorFlags = extractor->flags();

    return OK;
}
//通过解析文件的格式,生成对应的extractor对象
status_t AwesomePlayer::setDataSource_l(
        const sp<DataSource> &dataSource) {
    sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);

    if (extractor == NULL) {
        return UNKNOWN_ERROR;
    }

    if (extractor->getDrmFlag()) {
        checkDrmStatus(dataSource);
    }

    return setDataSource_l(extractor);
}

sp<MediaExtractor> MediaExtractor::Create(
        const sp<DataSource> &source, const char *mime) {
    sp<AMessage> meta;

    String8 tmp;
    if (mime == NULL) {
        float confidence;
        if (!source->sniff(&tmp, &confidence, &meta)) {
            ALOGV("FAILED to autodetect media content.");

            return NULL;
        }

        mime = tmp.string();
        ALOGV("Autodetected media content as '%s' with confidence %.2f",
             mime, confidence);
    }

    bool isDrm = false;
    // DRM MIME type syntax is "drm+type+original" where
    // type is "es_based" or "container_based" and
    // original is the content's cleartext MIME type
    if (!strncmp(mime, "drm+", 4)) {
        const char *originalMime = strchr(mime+4, '+');
        if (originalMime == NULL) {
            // second + not found
            return NULL;
        }
        ++originalMime;
        if (!strncmp(mime, "drm+es_based+", 13)) {
            // DRMExtractor sets container metadata kKeyIsDRM to 1
            return new DRMExtractor(source, originalMime);
        } else if (!strncmp(mime, "drm+container_based+", 20)) {
            mime = originalMime;
            isDrm = true;
        } else {
            return NULL;
        }
    }

    MediaExtractor *ret = NULL;
    if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG4)
            || !strcasecmp(mime, "audio/mp4")) {
        ret = new MPEG4Extractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_MPEG)) {
        ret = new MP3Extractor(source, meta);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_NB)
            || !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_WB)) {
        ret = new AMRExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_FLAC)) {
        ret = new FLACExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WAV)) {
        ret = new WAVExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_OGG)) {
        ret = new OggExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MATROSKA)) {
        ret = new MatroskaExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
        ret = new MPEG2TSExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WVM)) {
        // Return now.  WVExtractor should not have the DrmFlag set in the block below.
        return new WVMExtractor(source);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC_ADTS)) {
        ret = new AACExtractor(source, meta);
    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2PS)) {
        ret = new MPEG2PSExtractor(source);
    }

    if (ret != NULL) {
       if (isDrm) {
           ret->setDrmFlag(true);
       } else {
           ret->setDrmFlag(false);
       }
    }

    return ret;
}

//补充下OMXCodec生成具体Decoder的过程,调用成功后会返回一个OMXCodec对象
sp<MediaSource> OMXCodec::Create(
        const sp<IOMX> &omx,
        const sp<MetaData> &meta, bool createEncoder,
        const sp<MediaSource> &source,
        const char *matchComponentName,
        uint32_t flags,
        const sp<ANativeWindow> &nativeWindow) {
    int32_t requiresSecureBuffers;
    if (source->getFormat()->findInt32(
                kKeyRequiresSecureBuffers,
                &requiresSecureBuffers)
            && requiresSecureBuffers) {
        flags |= kIgnoreCodecSpecificData;
        flags |= kUseSecureInputBuffers;
    }

    const char *mime;
    bool success = meta->findCString(kKeyMIMEType, &mime);
    CHECK(success);

    Vector<CodecNameAndQuirks> matchingCodecs;
    findMatchingCodecs(
            mime, createEncoder, matchComponentName, flags, &matchingCodecs);

    if (matchingCodecs.isEmpty()) {
        ALOGV("No matching codecs! (mime: %s, createEncoder: %s, "
                "matchComponentName: %s, flags: 0x%x)",
                mime, createEncoder ? "true" : "false", matchComponentName, flags);
        return NULL;
    }

    sp<OMXCodecObserver> observer = new OMXCodecObserver;
    IOMX::node_id node = 0;

    for (size_t i = 0; i < matchingCodecs.size(); ++i) {
        const char *componentNameBase = matchingCodecs[i].mName.string();
        uint32_t quirks = matchingCodecs[i].mQuirks;
        const char *componentName = componentNameBase;

        AString tmp;
        if (flags & kUseSecureInputBuffers) {
            tmp = componentNameBase;
            tmp.append(".secure");

            componentName = tmp.c_str();
        }

        if (createEncoder) {
            sp<MediaSource> softwareCodec =
                InstantiateSoftwareEncoder(componentName, source, meta);

            if (softwareCodec != NULL) {
                ALOGV("Successfully allocated software codec '%s'", componentName);

                return softwareCodec;
            }
        }

        ALOGV("Attempting to allocate OMX node '%s'", componentName);

        if (!createEncoder
                && (quirks & kOutputBuffersAreUnreadable)
                && (flags & kClientNeedsFramebuffer)) {
            if (strncmp(componentName, "OMX.SEC.", 8)) {
                // For OMX.SEC.* decoders we can enable a special mode that
                // gives the client access to the framebuffer contents.

                ALOGW("Component '%s' does not give the client access to "
                     "the framebuffer contents. Skipping.",
                     componentName);

                continue;
            }
        }

        status_t err = omx->allocateNode(componentName, observer, &node);
        if (err == OK) {
            ALOGV("Successfully allocated OMX node '%s'", componentName);

            sp<OMXCodec> codec = new OMXCodec(
                    omx, node, quirks, flags,
                    createEncoder, mime, componentName,
                    source, nativeWindow);

            observer->setCodec(codec);

            err = codec->configureCodec(meta);

            if (err == OK) {
                if (!strcmp("OMX.Nvidia.mpeg2v.decode", componentName)) {
                    codec->mFlags |= kOnlySubmitOneInputBufferAtOneTime;
                }

                return codec;
            }

            ALOGV("Failed to configure codec '%s'", componentName);
        }
    }

    return NULL;
}
//负责获取解码后的数据,供AudioPlayer使用
status_t OMXCodec::read(
        MediaBuffer **buffer, const ReadOptions *options) {
    status_t err = OK;
    *buffer = NULL;

    Mutex::Autolock autoLock(mLock);

    if (mState != EXECUTING && mState != RECONFIGURING) {
        return UNKNOWN_ERROR;
    }

    bool seeking = false;
    int64_t seekTimeUs;
    ReadOptions::SeekMode seekMode;
    if (options && options->getSeekTo(&seekTimeUs, &seekMode)) {
        seeking = true;
    }

    if (mInitialBufferSubmit) {
        mInitialBufferSubmit = false;

        if (seeking) {
            CHECK(seekTimeUs >= 0);
            mSeekTimeUs = seekTimeUs;
            mSeekMode = seekMode;

            // There's no reason to trigger the code below, there's
            // nothing to flush yet.
            seeking = false;
            mPaused = false;
        }

        drainInputBuffers();

        if (mState == EXECUTING) {
            // Otherwise mState == RECONFIGURING and this code will trigger
            // after the output port is reenabled.
            fillOutputBuffers();
        }
    }

    if (seeking) {
        while (mState == RECONFIGURING) {
            if ((err = waitForBufferFilled_l()) != OK) {
                return err;
            }
        }

        if (mState != EXECUTING) {
            return UNKNOWN_ERROR;
        }

        CODEC_LOGV("seeking to %lld us (%.2f secs)", seekTimeUs, seekTimeUs / 1E6);

        mSignalledEOS = false;

        CHECK(seekTimeUs >= 0);
        mSeekTimeUs = seekTimeUs;
        mSeekMode = seekMode;

        mFilledBuffers.clear();

        CHECK_EQ((int)mState, (int)EXECUTING);

        bool emulateInputFlushCompletion = !flushPortAsync(kPortIndexInput);
        bool emulateOutputFlushCompletion = !flushPortAsync(kPortIndexOutput);

        if (emulateInputFlushCompletion) {
            onCmdComplete(OMX_CommandFlush, kPortIndexInput);
        }

        if (emulateOutputFlushCompletion) {
            onCmdComplete(OMX_CommandFlush, kPortIndexOutput);
        }

        while (mSeekTimeUs >= 0) {
            if ((err = waitForBufferFilled_l()) != OK) {
                return err;
            }
        }
    }

    while (mState != ERROR && !mNoMoreOutputData && mFilledBuffers.empty()) {
        if ((err = waitForBufferFilled_l()) != OK) {
            return err;
        }
    }

    if (mState == ERROR) {
        return UNKNOWN_ERROR;
    }

    if (mFilledBuffers.empty()) {
        return mSignalledEOS ? mFinalStatus : ERROR_END_OF_STREAM;
    }

    if (mOutputPortSettingsHaveChanged) {
        mOutputPortSettingsHaveChanged = false;

        return INFO_FORMAT_CHANGED;
    }

    size_t index = *mFilledBuffers.begin();
    mFilledBuffers.erase(mFilledBuffers.begin());

    BufferInfo *info = &mPortBuffers[kPortIndexOutput].editItemAt(index);
    CHECK_EQ((int)info->mStatus, (int)OWNED_BY_US);
    info->mStatus = OWNED_BY_CLIENT;

    info->mMediaBuffer->add_ref();
    if (mSkipCutBuffer != NULL) {
        mSkipCutBuffer->submit(info->mMediaBuffer);
    }
    *buffer = info->mMediaBuffer;

    return OK;
}











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