Android 语音遥控器的整体分析-底层实现机制分析

前面我们知道Writer类中开了一个线程在进行实际的录音。

Writer.addSource()中，有这样一段代码：

    const char *kHeader = isWide ? "#!AMR-WB\n" : "#!AMR\n";
    ssize_t n = strlen(kHeader);
    if (write(mFd, kHeader, n) != n) {
        return ERROR_IO;
    }

其实就是在录音之前先将音频的一些参数信息写进去。

在StagefrightRecorder::createAudioSource()中，有一段代码：

  sp audioEncoder =
        OMXCodec::Create(client.interface(), encMeta,
                         true /* createEncoder */, audioSource);
    mAudioSourceNode = audioSource;

    return audioEncoder;

这里是创建一个编码器

在OMXCodec::Create()中是这样的，先调用findMatchingCodecs查找匹配的编码器，如果没有匹配的就返回，，如果需要添加编码器，那么就添加一个：

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

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

                return softwareCodec;
            }
        }

因此，添加音频编码器就在这里添加咯！！！

static sp InstantiateSoftwareEncoder(
        const char *name, const sp &source,
        const sp &meta) {
    struct FactoryInfo {
        const char *name;
        sp (*CreateFunc)(const sp &, const sp &);
    };

    static const FactoryInfo kFactoryInfo[] = {
        FACTORY_REF(AACEncoder)
    };
    for (size_t i = 0;
         i < sizeof(kFactoryInfo) / sizeof(kFactoryInfo[0]); ++i) {
        if (!strcmp(name, kFactoryInfo[i].name)) {
            return (*kFactoryInfo[i].CreateFunc)(source, meta);
        }
    }

    return NULL;
}

由这个create中的匹配过程，可以知道，编解码器存放的地方是：struct MediaCodecList中,以一个向量保存

\\frameworks\av\media\libstagefright\MediaCodecList.h

    Vector mCodecInfos;

ssize_t MediaCodecList::findCodecByType(
        const char *type, bool encoder, size_t startIndex) const {
    ssize_t typeIndex = mTypes.indexOfKey(type);

    if (typeIndex < 0) {
        return -ENOENT;
    }
    while (startIndex < mCodecInfos.size()) {
        const CodecInfo &info = mCodecInfos.itemAt(startIndex);

        if (info.mIsEncoder == encoder && (info.mTypes & typeMask)) {
            return startIndex;
        }

        ++startIndex;
    }

    return -ENOENT;
}

对编码器的使用，也就是线程里的 err = mSource->read(&buffer);  由read（）函数可以知道音频数据来源是 List mBuffersReceived这个Buffer：

status_t AudioSource::read(
        MediaBuffer **out, const ReadOptions *options) {
    Mutex::Autolock autoLock(mLock);
    *out = NULL;

    if (mInitCheck != OK) {
        return NO_INIT;
    }

    while (mStarted && mBuffersReceived.empty()) {
        mFrameAvailableCondition.wait(mLock);
    }
    if (!mStarted) {
        return OK;
    }
    MediaBuffer *buffer = *mBuffersReceived.begin();
    mBuffersReceived.erase(mBuffersReceived.begin());
    ++mNumClientOwnedBuffers;
    buffer->setObserver(this);
    buffer->add_ref();

    // Mute/suppress the recording sound
    int64_t timeUs;
    CHECK(buffer->meta_data()->findInt64(kKeyTime, &timeUs));
    int64_t elapsedTimeUs = timeUs - mStartTimeUs;
    if (elapsedTimeUs < kAutoRampStartUs) {
        memset((uint8_t *) buffer->data(), 0, buffer->range_length());
    } else if (elapsedTimeUs < kAutoRampStartUs + kAutoRampDurationUs) {
        int32_t autoRampDurationFrames =
                    (kAutoRampDurationUs * mSampleRate + 500000LL) / 1000000LL;

        int32_t autoRampStartFrames =
                    (kAutoRampStartUs * mSampleRate + 500000LL) / 1000000LL;

        int32_t nFrames = mNumFramesReceived - autoRampStartFrames;
        rampVolume(nFrames, autoRampDurationFrames,
                (uint8_t *) buffer->data(), buffer->range_length());
    }

    // Track the max recording signal amplitude.
    if (mTrackMaxAmplitude) {
        trackMaxAmplitude(
            (int16_t *) buffer->data(), buffer->range_length() >> 1);
    }

    *out = buffer;
    return OK;
}

 而跟踪这个mBuffersReceived，可以知道Android在构造AudioSource实例的时候会指定一个数据源inputSource和回调函数AudioRecordCallbackFunction，在AudioSource中会构造AudioRecord对象，实际录音过程中就是在AudioRecord中通过这个回调函数中将数据一帧一帧填充进去的。

AudioSource::AudioSource(
        audio_source_t inputSource, uint32_t sampleRate, uint32_t channelCount)
    : mRecord(NULL),
      mStarted(false),
      mSampleRate(sampleRate),
      mPrevSampleTimeUs(0),
      mNumFramesReceived(0),
      mNumClientOwnedBuffers(0) {
    ALOGV("sampleRate: %d, channelCount: %d", sampleRate, channelCount);
    CHECK(channelCount == 1 || channelCount == 2);

    size_t minFrameCount;
    status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
                                           sampleRate,
                                           AUDIO_FORMAT_PCM_16_BIT,
                                           audio_channel_in_mask_from_count(channelCount));
    if (status == OK) {
        // make sure that the AudioRecord callback never returns more than the maximum
        // buffer size
        int frameCount = kMaxBufferSize / sizeof(int16_t) / channelCount;

        // make sure that the AudioRecord total buffer size is large enough
        int bufCount = 2;
        while ((bufCount * frameCount) < minFrameCount) {
            bufCount++;
        }

        mRecord = new AudioRecord(
                    inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
                    audio_channel_in_mask_from_count(channelCount),
                    bufCount * frameCount,
                    AudioRecordCallbackFunction,
                    this,
                    frameCount);
        mInitCheck = mRecord->initCheck();
    } else {
        mInitCheck = status;
    }
}

如果想知道录音设备，比如MIC中的数据是怎么通过回调函数写到Buffer中，那就要继续跟踪传到AudioRecord中的回调函数AudioRecordCallbackFunction了。

分析AudioRecord可以知道，在创建AudioRecord的时候会创建一个线程

 if (cbf != NULL) {
        mAudioRecordThread = new AudioRecordThread(*this, threadCanCallJava);
        mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
    }

调用回调的这个过程就是在这个AudioRecordThread线程中操作的。

bool AudioRecord::AudioRecordThread::threadLoop()
{
    {
        AutoMutex _l(mMyLock);
        if (mPaused) {
            mMyCond.wait(mMyLock);
            // caller will check for exitPending()
            return true;
        }
    }
    if (!mReceiver.processAudioBuffer(this)) {
        pause();
    }
    return true;
}

一次次在processAudioBuffer(this)中调用回调函数（也就是AudioRecord.mCbf）。

 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);

前面都是初始化构建过程，现在来看下AudioRecord::start,我们知道其中会调用IAudioRecord类型对象mAudioRecord的start函数。

status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
    status_t ret = NO_ERROR;
    sp t = mAudioRecordThread;

    ALOGV("start, sync event %d trigger session %d", event, triggerSession);

    AutoMutex lock(mLock);
    // acquire a strong reference on the IAudioRecord and IMemory so that they cannot be destroyed
    // while we are accessing the cblk
    sp audioRecord = mAudioRecord;
    sp iMem = mCblkMemory;
    audio_track_cblk_t* cblk = mCblk;

    if (!mActive) {
        mActive = true;

        cblk->lock.lock();
        if (!(cblk->flags & CBLK_INVALID)) {
            cblk->lock.unlock();
            ALOGV("mAudioRecord->start()");
            ret = mAudioRecord->start(event, triggerSession);
            cblk->lock.lock();
            if (ret == DEAD_OBJECT) {
                android_atomic_or(CBLK_INVALID, &cblk->flags);
            }
        }
        if (cblk->flags & CBLK_INVALID) {
            audio_track_cblk_t* temp = cblk;
            ret = restoreRecord_l(temp);
            cblk = temp;
        }
        cblk->lock.unlock();
        if (ret == NO_ERROR) {
            mNewPosition = cblk->user + mUpdatePeriod;
            cblk->bufferTimeoutMs = (event == AudioSystem::SYNC_EVENT_NONE) ? MAX_RUN_TIMEOUT_MS :
                                            AudioSystem::kSyncRecordStartTimeOutMs;
            cblk->waitTimeMs = 0;
            if (t != 0) {
                t->resume();
            } else {
                mPreviousPriority = getpriority(PRIO_PROCESS, 0);
                get_sched_policy(0, &mPreviousSchedulingGroup);
                androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
            }
        } else {
            mActive = false;
        }
    }

    return ret;
}

分析围绕IAudioRecord接口的类图

可以知道，最终调用的是AudioFinger中RecordHandler这个类的start()函数。

status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
        int triggerSession) {
    ALOGV("RecordHandle::start()");
    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
}

后面再分析AudioFlinger怎么操作硬件。 

这篇博文提到了怎么在AudioFlinger中处理音频多路输出的情况：

http://blog.csdn.net/hgl868/article/details/6888502