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    26 avril 2011, par

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  • Compute PTS and DTS correctly to sync audio and video ffmpeg C++

    14 août 2015, par Kaidul Islam

    I am trying to mux H264 encoded data and G711 PCM data into mov multimedia container. I am creating AVPacket from encoded data and initially the PTS and DTS value of video/audio frames is equivalent to AV_NOPTS_VALUE. So I calculated the DTS using current time information. My code -

    bool AudioVideoRecorder::WriteVideo(const unsigned char *pData, size_t iDataSize, bool const bIFrame) {
    
    .....................................
    
    .....................................
    
    .....................................
    
    AVPacket pkt = {0};
    
    av_init_packet(&pkt);
    
    int64_t dts = av_gettime();
    
    dts = av_rescale_q(dts, (AVRational){1, 1000000}, m_pVideoStream->time_base);
    
    int duration = 90000 / VIDEO_FRAME_RATE;
    
    if(m_prevVideoDts > 0LL) {
    
        duration = dts - m_prevVideoDts;
    
    }
    
    m_prevVideoDts = dts;
    

    
    pkt.pts = AV_NOPTS_VALUE;
    
    pkt.dts = m_currVideoDts;
    
    m_currVideoDts += duration;
    
    pkt.duration = duration;
    
    if(bIFrame) {
    
        pkt.flags |= AV_PKT_FLAG_KEY;
    
    }
    
    pkt.stream_index = m_pVideoStream->index;
    
    pkt.data = (uint8_t*) pData;
    
    pkt.size = iDataSize;
    

    
    int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
    

    
    if(ret < 0) {
    
        LogErr("Writing video frame failed.");
    
        return false;
    
    }
    

    
    Log("Writing video frame done.");
    

    
    av_free_packet(&pkt);
    
    return true;
    
}
    

    
bool AudioVideoRecorder::WriteAudio(const unsigned char *pEncodedData, size_t iDataSize) {
    
    .................................
    
    .................................
    
    .................................
    
    AVPacket pkt = {0};
    
    av_init_packet(&pkt);
    

    
    int64_t dts = av_gettime();
    
    dts = av_rescale_q(dts, (AVRational){1, 1000000}, (AVRational){1, 90000});
    
    int duration = AUDIO_STREAM_DURATION; // 20
    
    if(m_prevAudioDts > 0LL) {
    
        duration = dts - m_prevAudioDts;
    
    }
    
    m_prevAudioDts = dts;
    
    pkt.pts = AV_NOPTS_VALUE;
    
    pkt.dts = m_currAudioDts;
    
    m_currAudioDts += duration;
    
    pkt.duration = duration;
    

    
    pkt.stream_index = m_pAudioStream->index;
    
    pkt.flags |= AV_PKT_FLAG_KEY;
    
    pkt.data = (uint8_t*) pEncodedData;
    
    pkt.size = iDataSize;
    

    
    int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
    
    if(ret < 0) {
    
        LogErr("Writing audio frame failed: %d", ret);
    
        return false;
    
    }
    

    
    Log("Writing audio frame done.");
    

    
    av_free_packet(&pkt);
    
    return true;
    
}

    And I added stream like this -

    AVStream* AudioVideoRecorder::AddMediaStream(enum AVCodecID codecID) {
    
    ................................
    
    .................................   
    
    pStream = avformat_new_stream(m_pFormatCtx, codec);
    
    if (!pStream) {
    
        LogErr("Could not allocate stream.");
    
        return NULL;
    
    }
    
    pStream->id = m_pFormatCtx->nb_streams - 1;
    
    pCodecCtx = pStream->codec;
    
    pCodecCtx->codec_id = codecID;
    

    
    switch(codec->type) {
    
    case AVMEDIA_TYPE_VIDEO:
    
        pCodecCtx->bit_rate = VIDEO_BIT_RATE;
    
        pCodecCtx->width = PICTURE_WIDTH;
    
        pCodecCtx->height = PICTURE_HEIGHT;
    
        pStream->time_base = (AVRational){1, 90000};
    
        pStream->avg_frame_rate = (AVRational){90000, 1};
    
        pStream->r_frame_rate = (AVRational){90000, 1}; // though the frame rate is variable and around 15 fps
    
        pCodecCtx->pix_fmt = STREAM_PIX_FMT;
    
        m_pVideoStream = pStream;
    
        break;
    

    
    case AVMEDIA_TYPE_AUDIO:
    
        pCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
    
        pCodecCtx->bit_rate = AUDIO_BIT_RATE;
    
        pCodecCtx->sample_rate = AUDIO_SAMPLE_RATE;
    
        pCodecCtx->channels = 1;
    
        m_pAudioStream = pStream;
    
        break;
    

    
    default:
    
        break;
    
    }
    

    
    /* Some formats want stream headers to be separate. */
    
    if (m_pOutputFmt->flags & AVFMT_GLOBALHEADER)
    
        m_pFormatCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
    

    
    return pStream;
    
}

    There are several problems with this calculation :

    1. The video is laggy and lags behind than audio increasingly with time.

    2. Suppose, an audio frame is received (WriteAudio(..)) little lately like 3 seconds, then the late frame should be started playing with 3 second delay, but it’s not. The delayed frame is played consecutively with previous frame.

    3. Sometimes I recorded for 40 seconds but the file duration is much like 2 minutes, but audio/video is played only few moments like 40 seconds and rest of the file contains nothing and seekbar jumps at en immediately after 40 seconds (tested in VLC).

    EDIT :

    According to Ronald S. Bultje’s suggestion, what I’ve understand :

    m_pAudioStream->time_base = (AVRational){1, 9000}; // actually no need to set as 9000 is already default value for audio as you said
    
m_pVideoStream->time_base = (AVRational){1, 9000};

    should be set as now both audio and video streams are now in same time base units.

    And for video :

    ...................
    
...................
    

    
int64_t dts = av_gettime(); // get current time in microseconds
    
dts *= 9000; 
    
dts /= 1000000; // 1 second = 10^6 microseconds
    
pkt.pts = AV_NOPTS_VALUE; // is it okay?
    
pkt.dts = dts;
    
// and no need to set pkt.duration, right?

    And for audio : (exactly same as video, right ?)

    ...................
    
...................
    

    
int64_t dts = av_gettime(); // get current time in microseconds
    
dts *= 9000; 
    
dts /= 1000000; // 1 second = 10^6 microseconds
    
pkt.pts = AV_NOPTS_VALUE; // is it okay?
    
pkt.dts = dts;
    
// and no need to set pkt.duration, right?

    And I think they are now like sharing same currDts, right ? Please correct me if I am wrong anywhere or missing anything.

    Also, if I want to use video stream time base as (AVRational){1, frameRate} and audio stream time base as (AVRational){1, sampleRate}, how the correct code should look like ?

    EDIT 2.0 :

    m_pAudioStream->time_base = (AVRational){1, VIDEO_FRAME_RATE};
    
m_pVideoStream->time_base = (AVRational){1, VIDEO_FRAME_RATE};

    And

    bool AudioVideoRecorder::WriteAudio(const unsigned char *pEncodedData, size_t iDataSize) {
    
    ...........................
    
    ......................
    
    AVPacket pkt = {0};
    
    av_init_packet(&pkt);
    

    
    int64_t dts = av_gettime() / 1000; // convert into millisecond
    
    dts = dts * VIDEO_FRAME_RATE;
    
    if(m_dtsOffset < 0) {
    
        m_dtsOffset = dts;
    
    }
    

    
    pkt.pts = AV_NOPTS_VALUE;
    
    pkt.dts = (dts - m_dtsOffset);
    

    
    pkt.stream_index = m_pAudioStream->index;
    
    pkt.flags |= AV_PKT_FLAG_KEY;
    
    pkt.data = (uint8_t*) pEncodedData;
    
    pkt.size = iDataSize;
    

    
    int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
    
    if(ret < 0) {
    
        LogErr("Writing audio frame failed: %d", ret);
    
        return false;
    
    }
    

    
    Log("Writing audio frame done.");
    

    
    av_free_packet(&pkt);
    
    return true;
    
}
    

    
bool AudioVideoRecorder::WriteVideo(const unsigned char *pData, size_t iDataSize, bool const bIFrame) {
    
    ........................................
    
    .................................
    
    AVPacket pkt = {0};
    
    av_init_packet(&pkt);
    
    int64_t dts = av_gettime() / 1000;
    
    dts = dts * VIDEO_FRAME_RATE;
    
    if(m_dtsOffset < 0) {
    
        m_dtsOffset = dts;
    
    }
    
    pkt.pts = AV_NOPTS_VALUE;
    
    pkt.dts = (dts - m_dtsOffset);
    

    
    if(bIFrame) {
    
        pkt.flags |= AV_PKT_FLAG_KEY;
    
    }
    
    pkt.stream_index = m_pVideoStream->index;
    
    pkt.data = (uint8_t*) pData;
    
    pkt.size = iDataSize;
    

    
    int ret = av_interleaved_write_frame(m_pFormatCtx, &pkt);
    

    
    if(ret < 0) {
    
        LogErr("Writing video frame failed.");
    
        return false;
    
    }
    

    
    Log("Writing video frame done.");
    

    
    av_free_packet(&pkt);
    
    return true;
    
}

    Is the last change okay ? The video and audio seems synced. Only problem is - the audio is played without the delay regardless the packet arrived in delay.
    Like -

    packet arrival : 1 2 3 4... (then next frame arrived after 3 sec) .. 5

    audio played : 1 2 3 4 (no delay) 5

    EDIT 3.0 :

    zeroed audio sample data :

    AVFrame* pSilentData;
    
pSilentData = av_frame_alloc();
    
memset(&pSilentData->data[0], 0, iDataSize);
    

    
pkt.data = (uint8_t*) pSilentData;
    
pkt.size = iDataSize;
    

    
av_freep(&pSilentData->data[0]);
    
av_frame_free(&pSilentData);

    Is this okay ? But after writing this into file container, there are dot dot noise during playing the media. Whats the problem ?

    EDIT 4.0 :

    Well, For µ-Law audio the zero value is represented as 0xff. So -

    memset(&pSilentData->data[0], 0xff, iDataSize);

    solve my problem.

  • ffmpeg ffserver - create a mosaic from two 720p webcam feeds

    28 juillet 2015, par der_felix

    for a project i would like to take the video feeds (NO audio) of two logitech c920 webcams, put them side-by-side and stream them.
    the c920 is able to compress the video feed with h264 itself(if enabled) and delivers 1080p with upto 30fps.
    the stream is then loaded in an android app by a ffmpeg library and rendered to the screen.

    what i already know :
    i know that i can take multiple streams or input files and create a mosaic stream via the filter_complex module.
    http and h264 seem to be good for streaming, but other configurations are also welcom if they are faster/better.

    the question :
    but how can i start the cameras with v4l2, set the camera resoltution and camera internal encoding and use these streams to create the mosaic ?
    the mosaic should be unscaled (=2560x720px).

    and i very often get the error code 256 but didnt find a solution what it means.

    the system : laptop with usb3, ubuntu 15.04x64 ffmpeg 2.7.1 and ffserver 2.5.7

    thanks for your help

    ffserver config :

    HTTPPort 8080                
    
HTTPBindAddress 0.0.0.0      
    
MaxHTTPConnections 2000   
    
MaxClients 1000         
    
MaxBandwidth 50000
    
CustomLog -      
    
#NoDaemon       
    

    
<feed>         
    

    
File /tmp/feed1.ffm
    
Launch ffmpeg -f v4l2 - input_format h264 -i /dev/video0 -i /dev/video1 -size 1280x720 -r 30 -filter_complex "nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280"  -c:v libx264 -f mpegts
    

    
</feed>
    

    
<stream>
    

    
Feed feed1.ffm
    
Format mpegts      
    
VideoBitRate 1024   
    
#VideoBufferSize 1024
    
VideoFrameRate 30      
    
#VideoSize hd720      
    
VideoSize 2560x720
    
#VideoIntraOnly         
    
#VideoGopSize 12      
    
VideoCodec libx264      
    
NoAudio            
    
VideoQMin 3         
    
VideoQMax 31
    
NoDefaults
    

    
</stream>
    

    
<stream>        
    
   Format status
    
   #Only allow local people to get the status
    
   ACL allow localhost
    
   ACL allow 192.168.0.0 192.168.255.255
    
</stream>

    output :

    ubuntu@ubuntu:~$ ffserver
    
ffserver version 2.5.7-0ubuntu0.15.04.1 Copyright (c) 2000-2015 the FFmpeg developers
    
  built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
    
  configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
    
  libavutil      54. 15.100 / 54. 15.100
    
  libavcodec     56. 13.100 / 56. 13.100
    
  libavformat    56. 15.102 / 56. 15.102
    
  libavdevice    56.  3.100 / 56.  3.100
    
  libavfilter     5.  2.103 /  5.  2.103
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  1.100 /  1.  1.100
    
  libpostproc    53.  3.100 / 53.  3.100
    
Tue Jul 28 10:13:44 2015 FFserver started.
    
Tue Jul 28 10:13:44 2015 Launch command line: ffmpeg -f v4l2 - input_format h264 -i /dev/video0 -i /dev/video1 -size 1280x720 -r 30 -filter_complex nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280 -c:v libx264 -f mpegts http://127.0.0.1:8080/feed1.ffm
    
feed1.ffm: Pid 17388 exited with status 256 after 0 seconds

    Hey guys !

    Here is our plan b for the mosaic stream.

    Alternative config :

    HTTPPort 8080         
    
HTTPBindAddress 0.0.0.0   
    
MaxHTTPConnections 2000   
    
MaxClients 1000         
    
MaxBandwidth 50000      
    
CustomLog -         
    

    
<feed>
    
File /tmp/feedlinks.ffm
    
Launch ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video0 -video_size 1280x720 -r 30
    
</feed>
    

    
<feed>
    
File /tmp/feedrechts.ffm
    
Launch ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video1 -video_size 1280x720 -r 30 
    
</feed>
    

    
<stream>
    
Feed feedlinks.ffm
    
Format mpegts
    
VideoBitRate 512
    
VideoFrameRate 30
    
VideoSize hd720
    
VideoCodec libx264
    
NoAudio
    
VideoQMin 3
    
VideoQMax 31
    
</stream>
    

    
<stream>
    
Feed feedrechts.ffm
    
Format mpegts
    
VideoBitRate 512
    
VideoFrameRate 30
    
VideoSize hd720
    
VideoCodec libx264
    
NoAudio
    
VideoQMin 3
    
VideoQMax 31
    
</stream>
    

    
<feed>
    
File /tmp/feedmosaic.ffm
    
Launch ffmpeg -i http://localhost:8080/testlinks.mpg -i http://localhost:8080/testrechts.mpg -filter_complex "nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280" -c:v libx264 -preset ultrafast -f mpegts
    
</feed>
    

    
<stream>
    
Feed feedmosaic.ffm
    
Format mpegts         # Format of the stream
    
VideoFrameRate 30      # Number of frames per second
    
VideoSize 2560x720
    
VideoCodec libx264      # Choose your codecs.
    
NoAudio            # Suppress audio
    
VideoQMin 3         # Videoquality ranges from 1 - 31 (worst to best)
    
VideoQMax 31
    
NoDefaults
    
</stream>
    

    
<stream>           # Server status URL
    
   Format status
    
   # Only allow local people to get the status
    
   ACL allow localhost
    
   ACL allow 192.168.0.0 192.168.255.255
    
   ACL allow 192.168.178.0 192.168.255.255
    
</stream>

    And this is the new output :

    ubuntu@ubuntu:~$ ffserver
    
ffserver version 2.5.7-0ubuntu0.15.04.1 Copyright (c) 2000-2015 the FFmpeg developers
    
  built with gcc 4.9.2 (Ubuntu 4.9.2-10ubuntu13)
    
  configuration: --prefix=/usr --extra-version=0ubuntu0.15.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --shlibdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --enable-shared --disable-stripping --enable-avresample --enable-avisynth --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libshine --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libwavpack --enable-libwebp --enable-libxvid --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzvbi --enable-libzmq --enable-frei0r --enable-libvpx --enable-libx264 --enable-libsoxr --enable-gnutls --enable-openal --enable-libopencv --enable-librtmp --enable-libx265
    
  libavutil      54. 15.100 / 54. 15.100
    
  libavcodec     56. 13.100 / 56. 13.100
    
  libavformat    56. 15.102 / 56. 15.102
    
  libavdevice    56.  3.100 / 56.  3.100
    
  libavfilter     5.  2.103 /  5.  2.103
    
  libavresample   2.  1.  0 /  2.  1.  0
    
  libswscale      3.  1.101 /  3.  1.101
    
  libswresample   1.  1.100 /  1.  1.100
    
  libpostproc    53.  3.100 / 53.  3.100
    
/etc/ffserver.conf:44: Setting default value for video bit rate tolerance = 128000. Use NoDefaults to disable it.
    
/etc/ffserver.conf:44: Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
    
/etc/ffserver.conf:44: Setting default value for video max rate = 1024000. Use NoDefaults to disable it.
    
/etc/ffserver.conf:44: Setting default value for video buffer size = 1024000. Use NoDefaults to disable it.
    
/etc/ffserver.conf:61: Setting default value for video bit rate tolerance = 128000. Use NoDefaults to disable it.
    
/etc/ffserver.conf:61: Setting default value for video rate control equation = tex^qComp. Use NoDefaults to disable it.
    
/etc/ffserver.conf:61: Setting default value for video max rate = 1024000. Use NoDefaults to disable it.
    
/etc/ffserver.conf:61: Setting default value for video buffer size = 1024000. Use NoDefaults to disable it.
    
Tue Jul 28 11:13:01 2015 Codec bitrates do not match for stream 0
    
Tue Jul 28 11:13:01 2015 FFserver started.
    
Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video0 -video_size 1280x720 -r 30 http://127.0.0.1:8080/feedlinks.ffm
    
Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -f v4l2 -input_format h264 -vcodec h264 -i /dev/video1 -video_size 1280x720 -r 30 http://127.0.0.1:8080/feedrechts.ffm
    
Tue Jul 28 11:13:01 2015 Launch command line: ffmpeg -i http://localhost:8080/testlinks.mpg -i http://localhost:8080/testrechts.mpg -filter_complex nullsrc=size=2560x720 [base]; [0:v] setpts=PTS-STARTPTS [left]; [1:v] setpts=PTS-STARTPTS [right]; [base][left] overlay=shortest=1 [tmp1]; [tmp1][right] overlay=shortest=1:x=1280 -c:v libx264 -preset ultrafast -f mpegts http://127.0.0.1:8080/feedmosaic.ffm
    
Tue Jul 28 11:13:02 2015 127.0.0.1 - - [GET] "/feedlinks.ffm HTTP/1.1" 200 4175
    
Tue Jul 28 11:13:02 2015 127.0.0.1 - - [GET] "/feedrechts.ffm HTTP/1.1" 200 4175
    
Tue Jul 28 11:13:18 2015 127.0.0.1 - - [POST] "/feedmosaic.ffm HTTP/1.1" 200 4096
    
Tue Jul 28 11:13:18 2015 127.0.0.1 - - [GET] "/testlinks.mpg HTTP/1.1" 200 2130291
    
Tue Jul 28 11:13:18 2015 127.0.0.1 - - [GET] "/testrechts.mpg HTTP/1.1" 200 1244999
    
feedmosaic.ffm: Pid 18775 exited with status 256 after 17 seconds

    Thanks for your help !

  • Ffmpeg : how to keep orientation when trimming video file ?

    7 mars 2013, par Alex

    I have a video file which I capture from my Android program and save as an mp4 video.

    In this my Android program I use

    setOrientationHint(90)

    call to indicate to a videoplayer that my camera has been rotated 90 degrees.

    I'm not really sure what setOrientationHint(90) does but with it I can see the file properly oriented when it plays in the video player. If not, then a video player orients my file incorrectly.

    Now I trim this file using FFMPEG command (here in.mp4, out.mp4, 1000 and 2000 are just for example)

    ffmpeg -i in.mp4 -ss 1000 -t 2000 -vcodec copy -acodec

    However, the resulting file is again wrongly oriented in the player.

    I wonder what should I do to keep the orientation hint in the trimmed video file ?

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