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• avfilter/af_afir : adjust min partition size

  5 janvier 2019, par Paul B Mahol

  avfilter/af_afir : adjust min partition size
  Minimal value allowed by our FFT is 16 thus min partition size is 8.
  

  • [DH] doc/filters.texi
  • [DH] libavfilter/af_afir.c

• Rendering video by ffmpeg.wasm in browser occured an error

  15 septembre 2022, par James Bor

  When a local video renderer uses the ffmpeg.wasm library in the Chrome browser, very often an error with the SBOX_FATAL_MEMORY_EXCEEDED code occurs during the rendering process. The standard command set is used. The code below is half fake because it is very long, but describes an approximate action algorithm. Computer performance and RAM capacity do not affect the video, files used - minimal size. Has anyone experienced this and how can we solve it ?
Error screen

  


  const videoGenerate = async (project) => {
  const ffmpeg = createFFmpeg({
      corePath: 'ffmpeg/ffmpeg-core.js',
      workerPath: 'ffmpeg/ffmpeg-core.worker.js'
  });
  await loadFfmpeg(ffmpeg);
  project.projectName = "Default";
  project.fileType = "video/mp4";

  const resultVideo = {
    title: `${project.projectName}ConcatenatedVideo.mp4`,
  };
  // *For fetchFile method and ffmpeg.FS('writeFile', title, file);
  await uploadObjects(project.projectName, ffmpeg);
  // *
  const command = ['-i', project.video, resultVideo.title];
  await ffmpeg.run(...command);
  await ffmpeg.FS("unlink", resultVideo.title);
  resultVideo["blob"] = ffmpeg.FS('readFile', title);
  return resultVideo.blob;
};


  


  These dependencies are used : "@ffmpeg/core" : " 0.8.5", "@ffmpeg/ffmpeg" : " 0.9.7". Upgrading the library to the latest version does not work either.

  


• "Error : more samples than frame size" while encoding audio to opus codec using FFMPEG

  28 avril 2023, par lokit khemka

  I am converting audio from codec AAC to Opus using libavcodec library of FFMPEG. The input codec details are as follows : Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 6 channels, fltp, 391 kb/s (default)

  


  The codec options that I have used for the output encoding are as follows :

  


      int OUTPUT_CHANNELS = 2;
    int OUTPUT_BIT_RATE = 32000;
int sample_rate = 48000;
    encoder_sc->audio_avcc->channels = OUTPUT_CHANNELS;
    encoder_sc->audio_avcc->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
    encoder_sc->audio_avcc->sample_rate = sample_rate;
    encoder_sc->audio_avcc->sample_fmt = encoder_sc->audio_avc->sample_fmts[0];
    encoder_sc->audio_avcc->bit_rate = OUTPUT_BIT_RATE;
    encoder_sc->audio_avcc->time_base = (AVRational){1, sample_rate};


  


  I am using the code in the file as it is, with minimal changes : https://github.com/leandromoreira/ffmpeg-libav-tutorial/blob/master/3_transcoding.c for reference. Look for the function prepare_audio_encoder in the file.

  


  When the run the program, I keep getting the error : " more samples than frame size". I don't know much about Audio Processing, so I cannot debug this error. Any help is greatly appreciated.