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• FFMPEG C++ API audio/video sync, video is longer [closed]

  11 mai 2023, par Gábor Gombor

  I create a video from frames in C++ with a given FPS and supply it to FFMPEG API. I record from an Unreal engine viewport and feed FFMPEG with the images. In this interval I have also an audio track in FLAC which I want sync with the video. When the music ends, I close the video and merge them, but the final video has sync problems, the video is a little bit longer than the audio, so I will have an increasing delay. For example I record 0:55 secs, the audio is ok=same length, but the video from frames will be 0:56 secs.

  


  I think the following code is problematic :

  


  bool MyVideoExporter::writeFrame(OutputStream* oStream, AVFrame* frame, int& framePTS)
{
    auto* packet = oStream->pkt->pkt;
    auto* codecContext = oStream->enc->codecContext;

    frame->pts = framePTS;
    frame->pkt_dts = frame->pts;

    auto ret = avcodec_send_frame(codecContext, frame);
    if (ret >= 0) {
        auto retVal = 0;
        while (retVal >= 0) {
            retVal = avcodec_receive_packet(codecContext, packet);
            if (retVal == AVERROR(EAGAIN) || retVal == AVERROR_EOF) break;
            else if (retVal < 0) {
                return false;
            }

            // rescale to audio, usually 1/44100
            av_packet_rescale_ts(packet, m_audiotimestamp, oStream->st->time_base);
            // rescale to FPS, usually 1/30 or 1/60
            av_packet_rescale_ts(packet, codecContext->time_base, oStream->st->time_base);

            packet->stream_index = oStream->st->index;

            retVal = av_interleaved_write_frame(m_avFormatContext.avFormatContext, packet);
            if (retVal < 0) {
                return false;
            }

            framePTS++;
        }

        return retVal == AVERROR_EOF;
    }

    return false;
}



  


  Any idea what is wrong ?

  


  I tried change the order of av_packet_rescale_ts lines or move the frame increase code to another places, but getting far worse results.

  


• FFMPEG How to insert short video and audio in longer video in specific time, size and position

  24 novembre 2020, par Milos Dimitrijevic

  FFMPEG How to insert short video (4sec) in longer video in specific time (from 15th to 19th second) and specific size. Long video is 1920x1080px and inserted video should be 1720x820px, positioned in the middle ?

  


  file info..
Duration : 00:00:22.07, start : 0.000000, bitrate : 3982 kb/s
Stream #0:0(und) : Video : h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 3900 kb/s, 29 fps, 29 tbr, 14848 tbn, 58 tbc (default)
Metadata :
handler_name : VideoHandler
Stream #0:1(und) : Audio : aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
Metadata :
handler_name : SoundHandler
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'short.mp4' :
Metadata :
major_brand : mp42
minor_version : 19529854
compatible_brands : mp42isom
creation_time : 2020-10-25T11:01:10.000000Z
Duration : 00:00:03.02, start : 0.000000, bitrate : 14405 kb/s
Stream #1:0(eng) : Audio : aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 319 kb/s (default)
Metadata :
creation_time : 2020-10-25T11:01:10.000000Z
handler_name : Sound Media Handler
Stream #1:1(eng) : Video : h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 14078 kb/s, 59.94 fps, 59.94 tbr, 60k tbn, 119.88 tbc (default)
Metadata :
creation_time : 2020-10-25T11:01:10.000000Z
handler_name : Video Media Handler
encoder : AVC Coding

  


• ffmpeg removing silence makes mp3 longer ?

  13 août 2017, par pocketg99

  I’ve been using the following command to attempt to remove silent segments from an mp3 file

  ffmpeg -i "podcasts/audio1.mp3" -af silenceremove=1:0:-50dB "/tmp/pod-sil.mp3"

  For some reason the resulting mp3 is twice as log as the input mp3. It is not half as fast. There does not appear to be any duplicated audio. There is some silence, but not an hour’s worth. For a given portion of the input file, you can find the same thing in the output file by going to twice the timestamp of the input file.

  The files are long so I have not yet listened to them all the way through. I really have no idea where the extra length is coming from, the files seem normal.

  Here is the full output from ffmpeg

  ffmpeg version 2.8.11-0ubuntu0.16.04.1 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --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-librtmp --enable-libschroedinger --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  [mp3 @ 0x21880e0] Skipping 0 bytes of junk at 0. 
  
  [mp3 @ 0x21880e0] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from 'podcasts/audio1.mp3':
  
    Duration: 01:00:00.20, start: 0.000000, bitrate: 320 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 320 kb/s
  
  File '/tmp/pod-sil.mp3' already exists. Overwrite ? [y/N] y
  
  Output #0, mp3, to '/tmp/pod-sil.mp3':
  
    Metadata:
  
      TSSE            : Lavf56.40.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp
  
      Metadata:
  
        encoder         : Lavc56.60.100 libmp3lame
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [libmp3lame @ 0x21999e0] Trying to remove 1152 samples, but the queue is empty
  
  size=   56253kB time=01:00:00.16 bitrate= 128.0kbits/s    
  
  video:0kB audio:56253kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000439%