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• Decoding and playing audio with ffmpeg and XAudio2 - frequency ratio wrong

  9 mars, par Brent de Carteret

  I'm using ffmpeg to decode audio and output it using the XAudio2 API, it works and plays synced with the video output using the pts. But it's high pitched (i.e. sounds like chipmunks).

  


  Setting breakpoints I can see it has set the correct sample rate from the audio codec in CreateSourceVoice. I'm stumped.

  


  Any help would be much appreciated.

  


  CDVDAUDIO.cpp

  


  #include "DVDAudioDevice.h"
   
HANDLE m_hBufferEndEvent;

CDVDAudio::CDVDAudio()
{
    m_pXAudio2 = NULL;
    m_pMasteringVoice = NULL;
    m_pSourceVoice = NULL;
    m_pWfx  = NULL;
    m_VoiceCallback = NULL;    
    m_hBufferEndEvent = CreateEvent(NULL, false, false, "Buffer end event");
}
    
CDVDAudio::~CDVDAudio()
{
    m_pXAudio2 = NULL;
    m_pMasteringVoice = NULL;
    m_pSourceVoice = NULL;
    m_pWfx  = NULL;
    m_VoiceCallback = NULL;
    CloseHandle(m_hBufferEndEvent);
    m_hBufferEndEvent = NULL;
}
    
bool CDVDAudio::Create(int iChannels, int iBitrate, int iBitsPerSample, bool bPasstrough)
{
    CoInitializeEx(NULL, COINIT_MULTITHREADED);
    HRESULT hr = XAudio2Create( &m_pXAudio2, 0, XAUDIO2_DEFAULT_PROCESSOR);
    
    if (SUCCEEDED(hr))
    {
        m_pXAudio2->CreateMasteringVoice( &m_pMasteringVoice );
    }
    
    // Create source voice
    WAVEFORMATEXTENSIBLE wfx;
    memset(&wfx, 0, sizeof(WAVEFORMATEXTENSIBLE));
    
    wfx.Format.wFormatTag           = WAVE_FORMAT_PCM;
    wfx.Format.nSamplesPerSec       = iBitrate;//pFFMpegData->pAudioCodecCtx->sample_rate;//48000 by default
    wfx.Format.nChannels            = iChannels;//pFFMpegData->pAudioCodecCtx->channels;
    wfx.Format.wBitsPerSample       = 16;
    wfx.Format.nBlockAlign          = wfx.Format.nChannels*16/8;
    wfx.Format.nAvgBytesPerSec      = wfx.Format.nSamplesPerSec * wfx.Format.nBlockAlign;
    wfx.Format.cbSize               = sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX);
    wfx.Samples.wValidBitsPerSample = wfx.Format.wBitsPerSample;
    
    if(wfx.Format.nChannels == 1)
    {
        wfx.dwChannelMask = SPEAKER_MONO;
    }
    else if(wfx.Format.nChannels == 2)
    {
        wfx.dwChannelMask = SPEAKER_STEREO;
    }
    else if(wfx.Format.nChannels == 5)
    {
        wfx.dwChannelMask = SPEAKER_5POINT1;
    }
    
    wfx.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
    
    unsigned int flags = 0;//XAUDIO2_VOICE_NOSRC;// | XAUDIO2_VOICE_NOPITCH;
        
    //Source voice
    m_VoiceCallback = new StreamingVoiceCallback(this);
    hr = m_pXAudio2->CreateSourceVoice(&m_pSourceVoice,(WAVEFORMATEX*)&wfx, 0 , 1.0f, m_VoiceCallback);
        
    if (!SUCCEEDED(hr))
        return false;
    
    // Start sound
    hr = m_pSourceVoice->Start(0);
    
    if(!SUCCEEDED(hr))
        return false;
    
    return true;
}
    
DWORD CDVDAudio::AddPackets(unsigned char* data, DWORD len)
{  
    memset(&m_SoundBuffer,0,sizeof(XAUDIO2_BUFFER));
    m_SoundBuffer.AudioBytes = len;
    m_SoundBuffer.pAudioData = data;
    m_SoundBuffer.pContext = NULL;//(VOID*)data;
    XAUDIO2_VOICE_STATE state;
    
    while (m_pSourceVoice->GetState( &state ), state.BuffersQueued > 60)
    {
        WaitForSingleObject( m_hBufferEndEvent, INFINITE );
    }
    
    m_pSourceVoice->SubmitSourceBuffer( &m_SoundBuffer );
    return 0;
}
    
void CDVDAudio::Destroy()
{
    m_pMasteringVoice->DestroyVoice();
    m_pXAudio2->Release();
    m_pSourceVoice->DestroyVoice();
    delete m_VoiceCallback;
    m_VoiceCallback = NULL;
}


  


  CDVDAUdioCodecFFmpeg.cpp

  


  #include "DVDAudioCodecFFmpeg.h"
#include "Log.h"
    
CDVDAudioCodecFFmpeg::CDVDAudioCodecFFmpeg() : CDVDAudioCodec()
{
    m_iBufferSize = 0;
    m_pCodecContext = NULL;
    m_bOpenedCodec = false;
}
    
CDVDAudioCodecFFmpeg::~CDVDAudioCodecFFmpeg()
{
    Dispose();
}
    
bool CDVDAudioCodecFFmpeg::Open(AVCodecID codecID, int iChannels, int iSampleRate)
{
    AVCodec* pCodec;
    m_bOpenedCodec = false;
    av_register_all();
    pCodec = avcodec_find_decoder(codecID);
    m_pCodecContext = avcodec_alloc_context3(pCodec);//avcodec_alloc_context();
    avcodec_get_context_defaults3(m_pCodecContext, pCodec);
    
    if (!pCodec)
    {
        CLog::Log(LOGERROR, "CDVDAudioCodecFFmpeg::Open() Unable to find codec");
        return false;
    }
    
    m_pCodecContext->debug_mv = 0;
    m_pCodecContext->debug = 0;
    m_pCodecContext->workaround_bugs = 1;
    
    if (pCodec->capabilities & CODEC_CAP_TRUNCATED)
        m_pCodecContext->flags |= CODEC_FLAG_TRUNCATED;
    
    m_pCodecContext->channels = iChannels;
    m_pCodecContext->sample_rate = iSampleRate;
    //m_pCodecContext->bits_per_sample = 24;
     
    /* //FIXME BRENT
        if( ExtraData && ExtraSize > 0 )
        {
            m_pCodecContext->extradata_size = ExtraSize;
            m_pCodecContext->extradata = m_dllAvCodec.av_mallocz(ExtraSize + FF_INPUT_BUFFER_PADDING_SIZE);
            memcpy(m_pCodecContext->extradata, ExtraData, ExtraSize);
        }
    */
    
    // set acceleration
    //m_pCodecContext->dsp_mask = FF_MM_FORCE | FF_MM_MMX | FF_MM_MMXEXT | FF_MM_SSE; //BRENT
    
    if (avcodec_open2(m_pCodecContext, pCodec, NULL) < 0)
    {
        CLog::Log(LOGERROR, "CDVDAudioCodecFFmpeg::Open() Unable to open codec");
        Dispose();
        return false;
    }
    
    m_bOpenedCodec = true;
    return true;
}
    
void CDVDAudioCodecFFmpeg::Dispose()
{
    if (m_pCodecContext)
    {
        if (m_bOpenedCodec)
            avcodec_close(m_pCodecContext);
        m_bOpenedCodec = false;
        av_free(m_pCodecContext);
        m_pCodecContext = NULL;
    }
    m_iBufferSize = 0;
}

int CDVDAudioCodecFFmpeg::Decode(BYTE* pData, int iSize)
{
    int iBytesUsed;
    if (!m_pCodecContext) return -1;
    
    //Copy into a FFMpeg AVPAcket again
    AVPacket packet;
    av_init_packet(&packet);
    
    packet.data=pData;
    packet.size=iSize;
    
    int iOutputSize = AVCODEC_MAX_AUDIO_FRAME_SIZE; //BRENT
    
    iBytesUsed = avcodec_decode_audio3(m_pCodecContext, (int16_t *)m_buffer, &iOutputSize/*m_iBufferSize*/, &packet);

    m_iBufferSize = iOutputSize;//BRENT

    return iBytesUsed;
}

int CDVDAudioCodecFFmpeg::GetData(BYTE** dst)
{
    *dst = m_buffer;
    return m_iBufferSize;
}

void CDVDAudioCodecFFmpeg::Reset()
{
    if (m_pCodecContext)
        avcodec_flush_buffers(m_pCodecContext);
}

int CDVDAudioCodecFFmpeg::GetChannels()
{
    if (m_pCodecContext)
        return m_pCodecContext->channels;
    return 0;
}

int CDVDAudioCodecFFmpeg::GetSampleRate()
{
    if (m_pCodecContext)
        return m_pCodecContext->sample_rate;
    return 0;
}
    
int CDVDAudioCodecFFmpeg::GetBitsPerSample()
{
    if (m_pCodecContext)
        return 16;
    return 0;
}


  


  CDVDPlayerAudio.cpp

  


  #include "DVDPlayerAudio.h"
#include "DVDDemuxUtils.h"
#include "Log.h"
    
#include 
#include "DVDAudioCodecFFmpeg.h" //FIXME Move to a codec factory!!
    
CDVDPlayerAudio::CDVDPlayerAudio(CDVDClock* pClock) : CThread()
{
    m_pClock = pClock;
    m_pAudioCodec = NULL;
    m_bInitializedOutputDevice = false;
    m_iSourceChannels = 0;
    m_audioClock = 0;
    
    //  m_currentPTSItem.pts = DVD_NOPTS_VALUE;
    //  m_currentPTSItem.timestamp = 0;
    
    SetSpeed(DVD_PLAYSPEED_NORMAL);
      
    InitializeCriticalSection(&m_critCodecSection);
    m_messageQueue.SetMaxDataSize(10 * 16 * 1024);
    //  g_dvdPerformanceCounter.EnableAudioQueue(&m_packetQueue);
}

CDVDPlayerAudio::~CDVDPlayerAudio()
{
    //  g_dvdPerformanceCounter.DisableAudioQueue();

    // close the stream, and don't wait for the audio to be finished
    CloseStream(true);
    DeleteCriticalSection(&m_critCodecSection);
}

bool CDVDPlayerAudio::OpenStream( CDemuxStreamAudio *pDemuxStream )
{
    // should always be NULL!!!!, it will probably crash anyway when deleting m_pAudioCodec here.
    if (m_pAudioCodec)
    {
        CLog::Log(LOGFATAL, "CDVDPlayerAudio::OpenStream() m_pAudioCodec != NULL");
        return false;
    }
    
    AVCodecID codecID = pDemuxStream->codec;
    
    CLog::Log(LOGNOTICE, "Finding audio codec for: %i", codecID);
    //m_pAudioCodec = CDVDFactoryCodec::CreateAudioCodec( pDemuxStream ); 
    m_pAudioCodec = new CDVDAudioCodecFFmpeg; //FIXME BRENT Codec Factory needed!
    
    if (!m_pAudioCodec->Open(pDemuxStream->codec, pDemuxStream->iChannels, pDemuxStream->iSampleRate))
    {
        m_pAudioCodec->Dispose();
        delete m_pAudioCodec;
        m_pAudioCodec = NULL;
        return false;
    }
    
    if ( !m_pAudioCodec )
    {
        CLog::Log(LOGERROR, "Unsupported audio codec");
        return false;
    }
    
    m_codec = pDemuxStream->codec;
    m_iSourceChannels = pDemuxStream->iChannels;
    m_messageQueue.Init();
    
    CLog::Log(LOGNOTICE, "Creating audio thread");
    Create();
    
    return true;
}

void CDVDPlayerAudio::CloseStream(bool bWaitForBuffers)
{
    // wait until buffers are empty
    if (bWaitForBuffers)
        m_messageQueue.WaitUntilEmpty();
    
    // send abort message to the audio queue
    m_messageQueue.Abort();
    
    CLog::Log(LOGNOTICE, "waiting for audio thread to exit");
    
    // shut down the audio_decode thread and wait for it
    StopThread(); // will set this->m_bStop to true
    this->WaitForThreadExit(INFINITE);
    
    // uninit queue
    m_messageQueue.End();
    
    CLog::Log(LOGNOTICE, "Deleting audio codec");
    if (m_pAudioCodec)
    {
        m_pAudioCodec->Dispose();
        delete m_pAudioCodec;
        m_pAudioCodec = NULL;
    }
    
    // flush any remaining pts values
    //FlushPTSQueue(); //FIXME BRENT
}

void CDVDPlayerAudio::OnStartup()
{
    CThread::SetName("CDVDPlayerAudio");
    pAudioPacket = NULL;
    m_audioClock = 0;
    audio_pkt_data = NULL;
    audio_pkt_size = 0;
  
    //  g_dvdPerformanceCounter.EnableAudioDecodePerformance(ThreadHandle());
}

void CDVDPlayerAudio::Process()
{
    CLog::Log(LOGNOTICE, "running thread: CDVDPlayerAudio::Process()");

    int result;
    
    // silence data
    BYTE silence[1024];
    memset(silence, 0, 1024);
    
    DVDAudioFrame audioframe;
    
    __int64 iClockDiff=0;
    while (!m_bStop)
    {
        //Don't let anybody mess with our global variables
        EnterCriticalSection(&m_critCodecSection);
        result = DecodeFrame(audioframe, m_speed != DVD_PLAYSPEED_NORMAL); // blocks if no audio is available, but leaves critical section before doing so
        LeaveCriticalSection(&m_critCodecSection);
    
        if ( result & DECODE_FLAG_ERROR ) 
        {      
            CLog::Log(LOGERROR, "CDVDPlayerAudio::Process - Decode Error. Skipping audio frame");
            continue;
        }
    
        if ( result & DECODE_FLAG_ABORT )
        {
            CLog::Log(LOGDEBUG, "CDVDPlayerAudio::Process - Abort received, exiting thread");
            break;
        }
    
        if ( result & DECODE_FLAG_DROP ) //FIXME BRENT
        {
            /*  //frame should be dropped. Don't let audio move ahead of the current time thou
                //we need to be able to start playing at any time
                //when playing backwards, we try to keep as small buffers as possible
    
                // set the time at this delay
                AddPTSQueue(audioframe.pts, m_dvdAudio.GetDelay());
            */
            if (m_speed > 0)
            {
                __int64 timestamp = m_pClock->GetAbsoluteClock() + (audioframe.duration * DVD_PLAYSPEED_NORMAL) / m_speed;
                while ( !m_bStop && timestamp > m_pClock->GetAbsoluteClock() )
                    Sleep(1);
            }
            continue;
        }
    
        if ( audioframe.size > 0 ) 
        {
            // we have successfully decoded an audio frame, open up the audio device if not already done
            if (!m_bInitializedOutputDevice)
            {
                m_bInitializedOutputDevice = InitializeOutputDevice();
            }
    
            //Add any packets play
            m_dvdAudio.AddPackets(audioframe.data, audioframe.size);
    
            // store the delay for this pts value so we can calculate the current playing
            //AddPTSQueue(audioframe.pts, m_dvdAudio.GetDelay() - audioframe.duration);//BRENT
        }
    
        // if we where asked to resync on this packet, do so here
        if ( result & DECODE_FLAG_RESYNC )
        {
            CLog::Log(LOGDEBUG, "CDVDPlayerAudio::Process - Resync recieved.");
            //while (!m_bStop && (unsigned int)m_dvdAudio.GetDelay() > audioframe.duration ) Sleep(5); //BRENT
            m_pClock->Discontinuity(CLOCK_DISC_NORMAL, audioframe.pts);
        }
    
        #ifdef USEOLDSYNC
        //Clock should be calculated after packets have been added as m_audioClock points to the 
        //time after they have been played
    
        const __int64 iCurrDiff = (m_audioClock - m_dvdAudio.GetDelay()) - m_pClock->GetClock();
        const __int64 iAvDiff = (iClockDiff + iCurrDiff)/2;
    
        //Check for discontinuity in the stream, use a moving average to
        //eliminate highfreq fluctuations of large packet sizes
        if ( ABS(iAvDiff) > 5000 ) // sync clock if average diff is bigger than 5 msec 
        {
            //Wait until only the new audio frame which triggered the discontinuity is left
            //then set disc state
            while (!m_bStop && (unsigned int)m_dvdAudio.GetBytesInBuffer() > audioframe.size )
                Sleep(5);
    
            m_pClock->Discontinuity(CLOCK_DISC_NORMAL, m_audioClock - m_dvdAudio.GetDelay());
            CLog::("CDVDPlayer:: Detected Audio Discontinuity, syncing clock. diff was: %I64d, %I64d, av: %I64d", iClockDiff, iCurrDiff, iAvDiff);
            iClockDiff = 0;
        }
        else
        {
            //Do gradual adjustments (not working yet)
            //m_pClock->AdjustSpeedToMatch(iClock + iAvDiff);
            iClockDiff = iCurrDiff;
        }
        #endif
    }
}

void CDVDPlayerAudio::OnExit()
{
    //g_dvdPerformanceCounter.DisableAudioDecodePerformance();
  
    // destroy audio device
    CLog::Log(LOGNOTICE, "Closing audio device");
    m_dvdAudio.Destroy();
    m_bInitializedOutputDevice = false;

    CLog::Log(LOGNOTICE, "thread end: CDVDPlayerAudio::OnExit()");
}

// decode one audio frame and returns its uncompressed size
int CDVDPlayerAudio::DecodeFrame(DVDAudioFrame &audioframe, bool bDropPacket)
{
    CDVDDemux::DemuxPacket* pPacket = pAudioPacket;
    int n=48000*2*16/8, len;
    
    //Store amount left at this point, and what last pts was
    unsigned __int64 first_pkt_pts = 0;
    int first_pkt_size = 0; 
    int first_pkt_used = 0;
    int result = 0;
    
    // make sure the sent frame is clean
    memset(&audioframe, 0, sizeof(DVDAudioFrame));
    
    if (pPacket)
    {
        first_pkt_pts = pPacket->pts;
        first_pkt_size = pPacket->iSize;
        first_pkt_used = first_pkt_size - audio_pkt_size;
    }
     
    for (;;)
    {
        /* NOTE: the audio packet can contain several frames */
        while (audio_pkt_size > 0)
        {
            len = m_pAudioCodec->Decode(audio_pkt_data, audio_pkt_size);
            if (len < 0)
            {
                /* if error, we skip the frame */
                audio_pkt_size=0;
                m_pAudioCodec->Reset();
                break;
            }
    
            // fix for fucked up decoders //FIXME BRENT
            if( len > audio_pkt_size )
            {        
                CLog::Log(LOGERROR, "CDVDPlayerAudio:DecodeFrame - Codec tried to consume more data than available. Potential memory corruption");        
                audio_pkt_size=0;
                m_pAudioCodec->Reset();
                assert(0);
            }
    
            // get decoded data and the size of it
            audioframe.size = m_pAudioCodec->GetData(&audioframe.data);
            audio_pkt_data += len;
            audio_pkt_size -= len;
    
            if (audioframe.size <= 0)
                continue;
    
            audioframe.pts = m_audioClock;
    
            // compute duration.
            n = m_pAudioCodec->GetChannels() * m_pAudioCodec->GetBitsPerSample() / 8 * m_pAudioCodec->GetSampleRate();
            if (n > 0)
            {
                // safety check, if channels == 0, n will result in 0, and that will result in a nice divide exception
                audioframe.duration = (unsigned int)(((__int64)audioframe.size * DVD_TIME_BASE) / n);
    
                // increase audioclock to after the packet
                m_audioClock += audioframe.duration;
            }
    
            //If we are asked to drop this packet, return a size of zero. then it won't be played
            //we currently still decode the audio.. this is needed since we still need to know it's 
            //duration to make sure clock is updated correctly.
            if ( bDropPacket )
            {
                result |= DECODE_FLAG_DROP;
            }
            return result;
        }
    
        // free the current packet
        if (pPacket)
        {
            CDVDDemuxUtils::FreeDemuxPacket(pPacket); //BRENT FIXME
            pPacket = NULL;
            pAudioPacket = NULL;
        }
    
        if (m_messageQueue.RecievedAbortRequest())
            return DECODE_FLAG_ABORT;
    
        // read next packet and return -1 on error
        LeaveCriticalSection(&m_critCodecSection); //Leave here as this might stall a while
    
        CDVDMsg* pMsg;
        MsgQueueReturnCode ret = m_messageQueue.Get(&pMsg, INFINITE);
        EnterCriticalSection(&m_critCodecSection);
            
        if (MSGQ_IS_ERROR(ret) || ret == MSGQ_ABORT)
            return DECODE_FLAG_ABORT;
    
        if (pMsg->IsType(CDVDMsg::DEMUXER_PACKET))
        {
            CDVDMsgDemuxerPacket* pMsgDemuxerPacket = (CDVDMsgDemuxerPacket*)pMsg;
            pPacket = pMsgDemuxerPacket->GetPacket();
            pMsgDemuxerPacket->m_pPacket = NULL; // XXX, test
            pAudioPacket = pPacket;
            audio_pkt_data = pPacket->pData;
            audio_pkt_size = pPacket->iSize;
        }
        else
        {
            // other data is not used here, free if
            // msg itself will still be available
            pMsg->Release();
        }
 
        // if update the audio clock with the pts
        if (pMsg->IsType(CDVDMsg::DEMUXER_PACKET) || pMsg->IsType(CDVDMsg::GENERAL_RESYNC))
        {
            if (pMsg->IsType(CDVDMsg::GENERAL_RESYNC))
            { 
                //player asked us to sync on this package
                CDVDMsgGeneralResync* pMsgGeneralResync = (CDVDMsgGeneralResync*)pMsg;
                result |= DECODE_FLAG_RESYNC;
                m_audioClock = pMsgGeneralResync->GetPts();
            }
            else if (pPacket->pts != DVD_NOPTS_VALUE) // CDVDMsg::DEMUXER_PACKET, pPacket is already set above
            {
                if (first_pkt_size == 0) 
                { 
                    //first package
                    m_audioClock = pPacket->pts;        
                }
                else if (first_pkt_pts > pPacket->pts)
                { 
                    //okey first packet in this continous stream, make sure we use the time here        
                    m_audioClock = pPacket->pts;        
                }
                else if ((unsigned __int64)m_audioClock < pPacket->pts || (unsigned __int64)m_audioClock > pPacket->pts)
                {
                    //crap, moved outsided correct pts
                    //Use pts from current packet, untill we find a better value for it.
                    //Should be ok after a couple of frames, as soon as it starts clean on a packet
                    m_audioClock = pPacket->pts;
                }
                else if (first_pkt_size == first_pkt_used)
                {
                    //Nice starting up freshly on the start of a packet, use pts from it
                    m_audioClock = pPacket->pts;
                }
            }
        }
        pMsg->Release();
    }
}

void CDVDPlayerAudio::SetSpeed(int speed)
{ 
    m_speed = speed;
  
    //if (m_speed == DVD_PLAYSPEED_PAUSE) m_dvdAudio.Pause(); //BRENT FIXME
    //else m_dvdAudio.Resume();
}
    
bool CDVDPlayerAudio::InitializeOutputDevice()
{
    int iChannels = m_pAudioCodec->GetChannels();
    int iSampleRate = m_pAudioCodec->GetSampleRate();
    int iBitsPerSample = m_pAudioCodec->GetBitsPerSample();
    //bool bPasstrough = m_pAudioCodec->NeedPasstrough(); //BRENT
    
    if (iChannels == 0 || iSampleRate == 0 || iBitsPerSample == 0)
    {
        CLog::Log(LOGERROR, "Unable to create audio device, (iChannels == 0 || iSampleRate == 0 || iBitsPerSample == 0)");
        return false;
    }
    
    CLog::Log(LOGNOTICE, "Creating audio device with codec id: %i, channels: %i, sample rate: %i", m_codec, iChannels, iSampleRate);
    if (m_dvdAudio.Create(iChannels, iSampleRate, iBitsPerSample, /*bPasstrough*/0)) // always 16 bit with ffmpeg ? //BRENT Passthrough needed?
    {
        return true;
    }
    
    CLog::Log(LOGERROR, "Failed Creating audio device with codec id: %i, channels: %i, sample rate: %i", m_codec, iChannels, iSampleRate);
    return false;
}


  


• Low latency video player on android

  20 mai 2021, par Louis Blenner

  I'd like to be able to stream the video from my webcam to an Android app with a latency below 500ms, on my local network.

  


  To capture and send the video over the network, I use ffmpeg.

  


  ffmpeg -f v4l2 -i /dev/video0 -preset ultrafast -tune zerolatency -vcodec libx264 -an -vf format=yuv420p -f mpegts  udp://192.168.1.155:5000


  


  This command takes the webcam as an input, convert it and send it to a device using the mpegts protocol.
  
This is not a requirement, if another technique could work, I could change the way I send the video.

  


  I am able to read the video on another PC from the local network with a latency below 500 ms, using commands like

  


  gst-launch-1.0 -v udpsrc port=5000 ! video/mpegts ! tsdemux ! h264parse ! avdec_h264 ! fpsdisplaysink sync=false


  


  or

  


  mpv udp://0.0.0.0:5000 --no-cache --untimed --no-demuxer-thread --video-sync=audio --vd-lavc-threads=1 


  


  So it is possible to have this range of latency.
  
I'd like to have the same thing on Android.

  


  Here are my tries to do that.

  


  Exoplayer

  


  After looking at the different players available on Android studio, it seems like Exoplayer is the go-to choice.
  
I tried different options indicated in the live-streaming documentation, but I always end up with a stream taking seconds to start and with a latency of seconds.
  
I tried to add a Button to seek to the default position of the windows, but it results in a loading of several seconds.

  


  DefaultExtractorsFactory extractorsFactory =
                new DefaultExtractorsFactory()
                        .setTsExtractorFlags(DefaultTsPayloadReaderFactory.FLAG_IGNORE_AAC_STREAM);

        player = new SimpleExoPlayer.Builder(this)
                .setMediaSourceFactory(
                        new DefaultMediaSourceFactory(this, extractorsFactory))
                .setLoadControl(new DefaultLoadControl.Builder()
                        .setBufferDurationsMs(DefaultLoadControl.DEFAULT_MIN_BUFFER_MS, DefaultLoadControl.DEFAULT_MAX_BUFFER_MS, 200, 200)
                        .build())
                .build();
        MyPlayerView playerView = findViewById(R.id.player_view);
        // Bind the player to the view.
        playerView.setPlayer(player);
        // Build the media item.
        MediaItem mediaItem = new MediaItem.Builder()
                .setUri(Uri.parse("udp://0.0.0.0:5000"))
                .setLiveMaxOffsetMs(500)
                .setLiveTargetOffsetMs(0)
                .setLiveMinOffsetMs(0)
                .build();
        // Set the media item to be played.
        player.setMediaItem(mediaItem);
        // Prepare the player.
        player.setPlayWhenReady(true);
        player.prepare();
        //player.seekToDefaultPosition();


  


  This issue is about the same issue and the conclusion was that Exoplayer was not fit for this use case.

  


  


  I'll be honest, ultra low-latency like this isn't ExoPlayer's main use-case

  


  


  Vlc

  


  Another try was to use the Vlc library.
  
But I was unable to have the same low latency stream as with the two previous players with Vlc.
  
I tried changing the preferences of Vlc to stream as fast as possible as described here

  


  Input/Codecs -> x264 preset: ultrafast - zerolatency
Input/Codecs -> Access Module: UDP input
Input/Codecs -> Clock Jitter: 500
Audio: disable audio


  


  I also tried reducing the different buffers.
  
However, I still have a latency of more than 1 seconds with that.

  


  Gstreamer

  


  Another try was to create a react-native project to use the different players available here.
  
One player that seemed promising was react-native-gstreamer because it uses gstreamer which is able to stream with low latency (gst-launch command).
  
But the library is now outdated.

  


  Question

  


  There were other tries, but none were successful.
  
Is there a problem with one of my approaches ?
  
And if not, Is there a player on Android (that I missed) which is able to achieve low latency stream like gstream or mpv on linux ?

  


• MP4Box / FFMPEG concat loses audio after first clip

  17 novembre 2017, par user1615343

  So I am certainly no expert when it comes to either of these tools, but I have a web-based project that’s executing commands on an Amazon Linux server to concatenate two video files that are uploaded.

  Both files are converted to mp4s first using FFMPEG, and those play perfectly in a browser after conversion :

  ffmpeg -i file1.mpg -c:v libx264 -crf 22 -c:a aac -strict -2 -movflags faststart file2.mp4

  Then, I attempt to combine these two resulting mp4s into a single mp4. I tried using FFMPEG to do this but to no avail. Switching to try MP4Box got me much closer : the videos are concatenated together, but the audio stops playing at the end of the first clip, and the second clip is silent.

  MP4Box -force-cat -keepsys -add file.mp4 -cat file2.mp4 out.mp4

  I’ve tried varying versions of the above command with no better results. Any input is greatly appreciated.

  EDIT : info on .mp4 files using

  ffmpeg -i file1.mp4 -i file2.mp4

  ffmpeg -i 1510189259715DogRunsintoGlassDoor_315a03a8e20acfc.mp4 -i
  1510189273549NewhouseMoonMoonneverseenstairsbeforefunnydog_285a03a8e6aab25.mp4

  ffmpeg version N-61041-g52a2138 Copyright (c) 2000-2014 the FFmpeg
  developers

  built on Mar 2 2014 05:45:04 with gcc 4.6 (Debian 4.6.3-1)

  configuration : —prefix=/root/ffmpeg-static/64bit
  —extra-cflags=’-I/root/ffmpeg-static/64bit/include -static’ —extra-ldflags=’-L/root/ffmpeg-static/64bit/lib -static’ —extra-libs=’-lxml2 -lexpat -lfreetype’ —enable-static —disable-shared —disable-ffserver —disable-doc —enable-bzlib —enable-zlib —enable-postproc —enable-runtime-cpudetect —enable-libx264 —enable-gpl —enable-libtheora —enable-libvorbis —enable-libmp3lame —enable-gray —enable-libass —enable-libfreetype —enable-libopenjpeg —enable-libspeex —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-version3 —enable-libvpx

  libavutil 52. 66.100 / 52. 66.100

  libavcodec 55. 52.102 / 55. 52.102

  libavformat 55. 33.100 / 55. 33.100

  libavdevice 55. 10.100 / 55. 10.100

  libavfilter 4. 2.100 / 4. 2.100

  libswscale 2. 5.101 / 2. 5.101

  libswresample 0. 18.100 / 0. 18.100

  libpostproc 52. 3.100 / 52. 3.100

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
  ’1510189259715DogRunsintoGlassDoor_315a03a8e20acfc.mp4’ :

  Metadata :

  major_brand : isom

  minor_version : 512

  compatible_brands : isomiso2avc1mp41

  encoder : Lavf55.33.100

  Duration : 00:00:04.92, start : 0.023220, bitrate : 634 kb/s

  Stream #0:0(und) : Video : h264 (High) (avc1 / 0x31637661), yuv420p,
  360x360 [SAR 1:1 DAR 1:1], 501 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
  (default)

  Metadata :

  handler_name : VideoHandler

  Stream #0:1(und) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, mono,
  fltp, 132 kb/s (default)

  Metadata :

  handler_name : SoundHandler

  Input #1, mov,mp4,m4a,3gp,3g2,mj2, from
  ’1510189273549NewhouseMoonMoonneverseenstairsbeforefunnydog_285a03a8e6aab25.mp4’ :

  Metadata :

  major_brand : isom

  minor_version : 512

  compatible_brands : isomiso2avc1mp41

  encoder : Lavf55.33.100

  Duration : 00:00:18.79, start : 0.023220, bitrate : 455 kb/s

  Stream #1:0(und) : Video : h264 (High) (avc1 / 0x31637661), yuv420p,
  362x360 [SAR 1:1 DAR 181:180], 320 kb/s, 29.94 fps, 29.94 tbr, 11976
  tbn, 59.88 tbc (default)

  Metadata :

  handler_name : VideoHandler

  Stream #1:1(eng) : Audio : aac (mp4a / 0x6134706D), 44100 Hz, stereo,
  fltp, 129 kb/s (default)

  Metadata :

  handler_name : SoundHandler

  At least one output file must be specified