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• Slow audio-video sync drift when merging wav and mp4 with ffmpeg

  27 juin 2018, par charlie80

  I have an mp4 file with only a single video stream (no audio) and a wav audio file that I would like to add to the video using ffmpeg. The audio and the video have been recorded simultaneously during a conference, the former from a mixer output on a PC and the latter from a digital videocamera.

  I am using this ffmpeg command :

  ffmpeg -i incontro3.mp4 -itsoffset 18.39 -i audio_mix.wav -c:v copy -c:a aac final-video.mp4

  where I’m using the -itsoffset 18.39 option since I know that 18.39s is the video-audio delay.

  The problem I’m experiencing is that in the output file, while the audio is perfectly in sync with the video at the beginning, it slowly drifts out of sync during the movie.

  The output if ffprobe on the video file is :

  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'incontro3.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.25.100
  
    Duration: 00:47:22.56, start: 0.000000, bitrate: 888 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 886 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
  
      Metadata:
  
        handler_name    : VideoHandler

  and the ffprobe output for the audio file is :

  Input #0, wav, from 'audio_mix.wav':
  
    Metadata:
  
      track           : 5
  
      encoder         : Lavf57.25.100
  
    Duration: 00:46:32.20, bitrate: 1411 kb/s
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

  I’m using the latest ffmpeg Zeranoe windows build git-9591ca7 (2016-05-25).

  Thanks in anticipation for any help/ideas !

  ---

  UPDATE 1 : It looks like the problem is upstream the video-audio merging, and could be in the concatenation and conversion of the MTS files generated by the video camera into the mp4 video. I will follow up as I make any progress in understanding...

  ---

  UPDATE 2 : The problem is not in the initial merging of the MTS files generated by the camera. Or, at least, it occurs identically if I merge them with cat or with ffmpeg -f concat

  ---

  UPDATE 3 : Following @Mulvya’s suggestion, I observed that the drift rate is constant (at least as far as I can tell judging by eye). I also tried to superimpose the A/V tracks with another software, and the drift is exactly the same, thereby ruling out ffmpeg as culprit. My (bad) feeling is that the issue could be related to the internal clocks of the digital video camera and the laptop used for audio recording running at slightly different rates (see here the report of an identical issue I just found).

• 24kHz audio file problem : unsupported bitrate 64000

  28 novembre 2018, par R. Vait

  I use alexa audio tags a lot. I know that now audio tags support 24kHz audio files so tried converting my audio files from 16kHz. I used the provided command in the docs to do so :

  ffmpeg -i  -ac 2 -codec:a libmp3lame -b:a 48k -ar 24000 

  But when I try to play this file, I get an invalid response error, saying : Error: The audio is of an unsupported bitrate 64000. By looking into file details I clearly see, that bitrate is 48kbps and sample rate is 24kHz. I don’t see any value where it would say 64 or anything close to it.

  If I encode my file back to 16kHz it plays fine again.

  It seems that there is a problem with this command, because if I encode my files using audacity, they work with 24kHz. I still would prefer to use ffmpeg, because I need to encode a lot of files.

  I am asking, not about file format, format is correct. I need files in 24kHz sample rate and that is what causes issues. I saw another question about similar problem and others having the discussion there about sample rates, but no one was able to encode file to be 24kHz using ffmpeg.

  Did anyone had any luck on encoding files to 24kHz using ffmpeg ?

• ffmpeg avcodec_open2() return error -22 only in ubuntu

  26 octobre 2020, par User800222

  AS title,

  


  I'm working on a project on Mac with ffmpeg. It runs on my Mac to encode frames to vp9 & vp8. But when I run it on my docker image in a ubuntu, the function avcodec_open2(), returns -22 error. (only for vp9&vp8, h264 works).

  


  I wonder if the error comes from the setting parameters part ? But it's still weird to have it working on mac but not in ubuntu.

  


  Does anyone has a similar experience ?

  


  bool open_video(AVFormatContext * oc,AVCodec * codec,OutputStream * ost,AVDictionary * opt_arg, bool reduceQuality) {
    int ret;
    AVCodecContext * c = ost->st->codec;
    AVDictionary * opt = NULL;
    av_dict_copy(&opt,opt_arg,0);

    if(codec264){
      // http://arstechnica.com/civis/viewtopic.php?f=6&t=1239375
      av_dict_set(&opt,"refs",    "3",          0);
      av_dict_set(&opt,"vprofile","main",       0);  // 2 versions to support differents libav/ffmpeg
      av_dict_set(&opt,"profile", "main",       0);
      
      //av_dict_set(&opt,"preset","superfast",0);
      //av_dict_set(&opt,"preset","ultrafast",0);
      av_dict_set(&opt,"preset","superfast",0);
      //av_dict_set(&opt,"preset","slow",0);
      
      //https://mattgadient.com/2013/06/12/a-best-settings-guide-for-handbrake-0-9-9/
      //av_dict_set(&opt, "tune", "zerolatency", 0); //bluerry but very small size
      //av_dict_set(&opt, "tune", "fastdecode", 0); // hd but large size
      // 23 300k, 29 150k, 35 75k
      if(reduceQuality)
        av_dict_set(&opt, "crf", "35", 0); //35 +-6 decrease/increase bitrate half/twice
      else
        av_dict_set(&opt, "crf", "29", 0); //29 +-6 decrease/increase bitrate half/twice
    
    }
    else if (codecVP9)
    { 
        // Setting explanation:
        // http://wiki.webmproject.org/ffmpeg/vp9-encoding-guide
        // https://developers.google.com/media/vp9/live-encoding/
        // https://developers.google.com/media/vp9/settings/vod/
        // https://developers.google.com/media/vp9/bitrate-modes/

        if(reduceQuality) //crf is the quality value for VP9 (0-63), lower the better quality youll get 
          av_dict_set(&opt, "crf"    ,      "40",         0); 
        else
          av_dict_set(&opt, "crf"    ,      "45",         0);
        av_dict_set(&opt, "speed"  ,        "8" ,         0);
        av_dict_set(&opt, "quality",        "realtime",   0); //realtime is recommended
        //av_dict_set(&opt, "threads",        "4" ,         0);
        //av_dict_set(&opt, "tile-columns",   "2",          0);
        //av_dict_set(&opt, "frame-parallel", "1",          0);
        //av_dict_set(&opt, "row-mt",         "1",          0);
        av_dict_set(&opt, "b:v",            "1",         0);
        av_dict_set(&opt, "g",              "400",        0); //key frame interval (big difference in vid size)

        //av_dict_set(&opt, "maxrate",        "3k",         0);
        //av_dict_set(&opt, "minrate",        "1k",         0);
        //av_dict_set(&opt, "hwaccel",        "vaapi",      0);
        //av_dict_set(&opt, "c:v",            "libvpx-vp9", 0);
        
    }
    else if (codecVP8)
    {
        // https://trac.ffmpeg.org/wiki/Encode/VP8 
        av_dict_set(&opt, "passes"    ,        "1" ,         0);   
        av_dict_set(&opt, "cpu-used"  ,        "15" ,         0); 
        av_dict_set(&opt, "qmax"      ,        "63",         0);   
        av_dict_set(&opt, "rt"        ,          "",         0); // realtime setting
        //av_dict_set(&opt, "crf"       ,        "10",         0);   

        if(reduceQuality)
          av_dict_set(&opt, "qmin"     ,        "40",        0); // Lower quality, smaller sizes
        else
          av_dict_set(&opt, "qmin"     ,        "35",        0); // Higher quality, larger sizes
    }
    else 
    {
        if(reduceQuality)
            av_dict_set(&opt, "x265-params", "crf=29", 0);
        else
            av_dict_set(&opt, "x265-params", "crf=23", 0);
        
        av_dict_set(&opt, "preset", "ultrafast", 0);
        av_dict_set(&opt, "tune", "zerolatency", 0);
//        av_dict_set(&opt, "profile", "main", 0);
    }
      
    // Open the codec
    ret = avcodec_open2(c,codec,&opt);
    av_dict_free(&opt);
    if (ret < 0) {
        cout << "Could not open video codec. (error code:" << ret << ")\n";
        return false;
    } // if


  


  Edit : Does anyone know if the ffmpeg-linux using different parameters' name for setting from other platform's (like Mac) ?

  


  Eidt2 : It seems like even if I don't set any options parameters, it's still giving me -22 error. Ex : ret = avcodec_open2(c, codec, NULL)