[Android N]MediaRecorder系列之StagefrightRecorder录制TS流flow(三)

CameraSource::start

先看CameraSource的start函数:

status_t CameraSource::start(MetaData *meta) {    ALOGV("start");    CHECK(!mStarted);    if (mInitCheck != OK) {        ALOGE("CameraSource is not initialized yet");        return mInitCheck;    }    char value[PROPERTY_VALUE_MAX];    if (property_get("media.stagefright.record-stats", value, NULL)        && (!strcmp(value, "1") || !strcasecmp(value, "true"))) {        mCollectStats = true;    }    mStartTimeUs = 0;    mNumInputBuffers = 0;    mEncoderFormat = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;    mEncoderDataSpace = HAL_DATASPACE_V0_BT709;    if (meta) {        //这里这个startTimeUs很重要，这里记录了开始录制的时间点，后面打时间戳的时候会用到        int64_t startTimeUs;        if (meta->findInt64(kKeyTime, &startTimeUs)) {            mStartTimeUs = startTimeUs;        }        int32_t nBuffers;        if (meta->findInt32(kKeyNumBuffers, &nBuffers)) {            CHECK_GT(nBuffers, 0);            mNumInputBuffers = nBuffers;        }        // apply encoder color format if specified        if (meta->findInt32(kKeyPixelFormat, &mEncoderFormat)) {            ALOGI("Using encoder format: %#x", mEncoderFormat);        }        if (meta->findInt32(kKeyColorSpace, &mEncoderDataSpace)) {            ALOGI("Using encoder data space: %#x", mEncoderDataSpace);        }    }    status_t err;     //这个才是真正的start的东西。    if ((err = startCameraRecording()) == OK) {        mStarted = true;    }    return err;}

status_t CameraSource::startCameraRecording() {    ALOGV("startCameraRecording");    // Reset the identity to the current thread because media server owns the    // camera and recording is started by the applications. The applications    // will connect to the camera in ICameraRecordingProxy::startRecording.    int64_t token = IPCThreadState::self()->clearCallingIdentity();    status_t err;    if (mVideoBufferMode == hardware::ICamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE) {        // Initialize buffer queue.        err = initBufferQueue(mVideoSize.width, mVideoSize.height, mEncoderFormat,                (android_dataspace_t)mEncoderDataSpace,                mNumInputBuffers > 0 ? mNumInputBuffers : 1);        if (err != OK) {            ALOGE("%s: Failed to initialize buffer queue: %s (err=%d)", __FUNCTION__,                    strerror(-err), err);            return err;        }    } else {        if (mNumInputBuffers > 0) {            err = mCamera->sendCommand(                CAMERA_CMD_SET_VIDEO_BUFFER_COUNT, mNumInputBuffers, 0);            // This could happen for CameraHAL1 clients; thus the failure is            // not a fatal error            if (err != OK) {                ALOGW("Failed to set video buffer count to %d due to %d",                    mNumInputBuffers, err);            }        }        //发送CMD去设置camer的参数        err = mCamera->sendCommand(            CAMERA_CMD_SET_VIDEO_FORMAT, mEncoderFormat, mEncoderDataSpace);        // This could happen for CameraHAL1 clients; thus the failure is        // not a fatal error        if (err != OK) {            ALOGW("Failed to set video encoder format/dataspace to %d, %d due to %d",                    mEncoderFormat, mEncoderDataSpace, err);        }        // Create memory heap to store buffers as VideoNativeMetadata.        createVideoBufferMemoryHeap(sizeof(VideoNativeHandleMetadata), kDefaultVideoBufferCount);    }    err = OK;    if (mCameraFlags & FLAGS_HOT_CAMERA) {        mCamera->unlock();        mCamera.clear();        //让camera开始recorder        if ((err = mCameraRecordingProxy->startRecording(                new ProxyListener(this))) != OK) {            ALOGE("Failed to start recording, received error: %s (%d)",                    strerror(-err), err);        }    } else {        mCamera->setListener(new CameraSourceListener(this));        mCamera->startRecording();        if (!mCamera->recordingEnabled()) {            err = -EINVAL;            ALOGE("Failed to start recording");        }    }    IPCThreadState::self()->restoreCallingIdentity(token);    return err;}

再往下就是cameraHAL的部分了，camera的整个框架和结构比较复杂，这里就先直接先跳过，我们就理解成，这个start函数会直接call到cameraHAL，开始进行录制操作，然后我们就等camera喷数据回来。

CameraSourceListener的postDataTimestamp会收到数据抛回来的广播，数据就存在dataPtr

void CameraSourceListener::postDataTimestamp(        nsecs_t timestamp, int32_t msgType, const sp<IMemory>& dataPtr) {    sp<CameraSource> source = mSource.promote();    if (source.get() != NULL) {        source->dataCallbackTimestamp(timestamp/1000, msgType, dataPtr);    }}

source->dataCallbackTimestamp(timestamp/1000, msgType, dataPtr);就会跑入:

void CameraSource::dataCallbackTimestamp(int64_t timestampUs,        int32_t msgType __unused, const sp<IMemory> &data) {    ALOGV("dataCallbackTimestamp: timestamp %lld us", (long long)timestampUs);    Mutex::Autolock autoLock(mLock);    if (shouldSkipFrameLocked(timestampUs)) {        releaseOneRecordingFrame(data);        return;    }    //这个flag表示有多少frame已经收到了    ++mNumFramesReceived;    CHECK(data != NULL && data->size() > 0);    //mFramesReceived这个里面存着camera抓到的一帧，帧数据    mFramesReceived.push_back(data);    int64_t timeUs = mStartTimeUs + (timestampUs - mFirstFrameTimeUs);    //mFrameTimes,打上时间戳    mFrameTimes.push_back(timeUs);    ALOGV("initial delay: %" PRId64 ", current time stamp: %" PRId64,        mStartTimeUs, timeUs);    mFrameAvailableCondition.signal();}

这个mFramesReceived先记住一下，等下后面read的时候就会用到了。

AudioSource::start

status_t AudioSource::start(MetaData *params) {    Mutex::Autolock autoLock(mLock);    if (mStarted) {        return UNKNOWN_ERROR;    }    if (mInitCheck != OK) {        return NO_INIT;    }    mTrackMaxAmplitude = false;    mMaxAmplitude = 0;    mInitialReadTimeUs = 0;    mStartTimeUs = 0;    int64_t startTimeUs;    //mStartTimeUs 记下开始的时间，为后面打算时间戳做准备    if (params && params->findInt64(kKeyTime, &startTimeUs)) {        mStartTimeUs = startTimeUs;    }    //会跑到HAL的start，不跟了    status_t err = mRecord->start();    if (err == OK) {        mStarted = true;    } else {        mRecord.clear();    }    return err;}

然后就等数据报回来：

status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {    int64_t timeUs = systemTime() / 1000ll;    // Estimate the real sampling time of the 1st sample in this buffer    // from AudioRecord's latency. (Apply this adjustment first so that    // the start time logic is not affected.)    timeUs -= mRecord->latency() * 1000LL;    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);    Mutex::Autolock autoLock(mLock);    if (!mStarted) {        ALOGW("Spurious callback from AudioRecord. Drop the audio data.");        return OK;    }    // Drop retrieved and previously lost audio data.    if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {        (void) mRecord->getInputFramesLost();        ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs);        return OK;    }    if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {        mInitialReadTimeUs = timeUs;        // Initial delay        if (mStartTimeUs > 0) {            mStartTimeUs = timeUs - mStartTimeUs;        } else {            // Assume latency is constant.            mStartTimeUs += mRecord->latency() * 1000;        }        mPrevSampleTimeUs = mStartTimeUs;    }    size_t numLostBytes = 0;    if (mNumFramesReceived > 0) {  // Ignore earlier frame lost        // getInputFramesLost() returns the number of lost frames.        // Convert number of frames lost to number of bytes lost.        numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();    }    CHECK_EQ(numLostBytes & 1, 0u);    CHECK_EQ(audioBuffer.size & 1, 0u);    if (numLostBytes > 0) {        // Loss of audio frames should happen rarely; thus the LOGW should        // not cause a logging spam        ALOGW("Lost audio record data: %zu bytes", numLostBytes);    }    while (numLostBytes > 0) {        size_t bufferSize = numLostBytes;        if (numLostBytes > kMaxBufferSize) {            numLostBytes -= kMaxBufferSize;            bufferSize = kMaxBufferSize;        } else {            numLostBytes = 0;        }        MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);        memset(lostAudioBuffer->data(), 0, bufferSize);        lostAudioBuffer->set_range(0, bufferSize);        queueInputBuffer_l(lostAudioBuffer, timeUs);    }    if (audioBuffer.size == 0) {        ALOGW("Nothing is available from AudioRecord callback buffer");        return OK;    }    const size_t bufferSize = audioBuffer.size;    MediaBuffer *buffer = new MediaBuffer(bufferSize);    memcpy((uint8_t *) buffer->data(),            audioBuffer.i16, audioBuffer.size);    buffer->set_range(0, bufferSize);    //在这个函数中，把buffer装到mBuffersReceived    queueInputBuffer_l(buffer, timeUs);    return OK;}

mBuffersReceived也记住一个，后面read的时候，就会用到了。

mFramesReceived和mBuffersReceived都是raw data，没有经过encode的数据，encode在后面的read中会再处理。