AudioTrack::createTrack()

status_t AudioTrack::createTrack(        int streamType,        uint32_t sampleRate,        int format,        int channelCount,        int frameCount,        uint32_t flags,        const sp& sharedBuffer,        audio_io_handle_t output,        bool enforceFrameCount){    status_t status;    const sp& audioFlinger = AudioSystem::get_audio_flinger();    if (audioFlinger == 0) {       LOGE("Could not get audioflinger");       return NO_INIT;    }    int afSampleRate;    if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {        return NO_INIT;    }    int afFrameCount;    if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {        return NO_INIT;    }    uint32_t afLatency;    if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {        return NO_INIT;    }    mNotificationFramesAct = mNotificationFramesReq;    if (!AudioSystem::isLinearPCM(format)) {        if (sharedBuffer != 0) {            frameCount = sharedBuffer->size();        }    } else {        // Ensure that buffer depth covers at least audio hardware latency        uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);        if (minBufCount < 2) minBufCount = 2;        int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;        if (sharedBuffer == 0) {            if (frameCount == 0) {                frameCount = minFrameCount;            }            if (mNotificationFramesAct == 0) {                mNotificationFramesAct = frameCount/2;            }            // Make sure that application is notified with sufficient margin            // before underrun            if (mNotificationFramesAct > (uint32_t)frameCount/2) {                mNotificationFramesAct = frameCount/2;            }            if (frameCount < minFrameCount) {                if (enforceFrameCount) {                    LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);                    return BAD_VALUE;                } else {                    frameCount = minFrameCount;                }            }        } else {            // Ensure that buffer alignment matches channelcount            if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {                LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);                return BAD_VALUE;            }            frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);        }    }    sp track = audioFlinger->createTrack(getpid(),                                                      streamType,                                                      sampleRate,                                                      format,                                                      channelCount,                                                      frameCount,                                                      ((uint16_t)flags) << 16,                                                      sharedBuffer,                                                      output,                                                      &mSessionId,                                                      &status);    if (track == 0) {        LOGE("AudioFlinger could not create track, status: %d", status);        return status;    }    sp cblk = track->getCblk();    if (cblk == 0) {        LOGE("Could not get control block");        return NO_INIT;    }    mAudioTrack.clear();    mAudioTrack = track;    mCblkMemory.clear();    mCblkMemory = cblk;    mCblk = static_cast(cblk->pointer());    mCblk->flags |= CBLK_DIRECTION_OUT;    if (sharedBuffer == 0) {        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);    } else {        mCblk->buffers = sharedBuffer->pointer();         // Force buffer full condition as data is already present in shared memory        mCblk->stepUser(mCblk->frameCount);    }    mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);    mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);    mAudioTrack->attachAuxEffect(mAuxEffectId);    mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;    mCblk->waitTimeMs = 0;    mRemainingFrames = mNotificationFramesAct;    mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;    return NO_ERROR;}//--> client buffer //audio flinger client move data from client buffer to hardware buffer//--> hardware buffer//hardware playback porcess extract data from hardware buffer//what is "frame"//-----------------------------------------------------------------------------------------------------//A frame is a set of samples, one per channel, at a particular instant in time. For stereophonic audio, //a frame consists of two samples. For Dolby 5.1 Surround Sound, a frame would consist of six samples //(left channel, center channel, right channel, rear right, rear left, and the low frequency channel). //For monophonic audio, a frame is equivalent to one sample.//Retrieved from "http://alsa.opensrc.org/Frame"//1个采样点只针对一个声道，而实际上可能会有一或多个声道.//由于不能用一个独立的单位来表示全部声道一次采样的数据量，也就引出了Frame的概念.//Frame的大小，就是一个采样点的字节数×声道数.//另外，在目前的声卡驱动程序中，其内部缓冲区也是采用Frame作为单位来分配和管理的.//what is "bufCount"//-------------------------------------------------------------------------------------------------------//bufCount 表示一个frame里面的字节数,用来描述一个时域的采样点所占的字节宽度,(注意:不等同于channel数.)//what is "frame count"//-------------------------------------------------------------------------------------------------------//frameFount 表示一个buffer所能够缓冲的frame的个数,也就是时域的点数 //所以://afFrameCount -- 硬件buffer的大小,以frame为单位.//afSampleRate -- 硬件采样率.//afFrameCount/afSampleRate -- 硬件buffer保存的数据能够播放的时间是多少,换算成ms就是//                             1000*(afFrameCount/afSampleRate)//硬件buffer的大小是既定的,如果硬件要求的延迟硬件buffer不能提供,则需要多开几个软件buffer,以缓冲更多的数据,提供更长的播放时间.//如果硬件要求的延迟为afLatency(ms),//那么硬件延迟要求对应的最小的软件buffer的数目为://minBufCount=afLatency/1000*(afFrameCount/afSampleRate)//但这里并未给出每个软件buffer的大小.//暂记每个软件buffer的大小(以frame为单位)为bufsize,采样率转换之前的采样率为sampleRate//由于每个软件buffer和硬件buffer中的数据所对应的播放时长是一样的.所以://bufsize/sampleRate=afFrameCount/afSampleRate;//bufsize=afFrameCount*sampleRate/afSampleRate;//则所有软件buffer加在一起大小为//minFrameCount=minBufCount*afFrameCount*sampleRate/afSampleRate;
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