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Mot : - Tags -/wav

Autres articles (69)

  • Websites made ​​with MediaSPIP

    2 mai 2011, par

    This page lists some websites based on MediaSPIP.

  • Creating farms of unique websites

    13 avril 2011, par

    MediaSPIP platforms can be installed as a farm, with a single "core" hosted on a dedicated server and used by multiple websites.
    This allows (among other things) : implementation costs to be shared between several different projects / individuals rapid deployment of multiple unique sites creation of groups of like-minded sites, making it possible to browse media in a more controlled and selective environment than the major "open" (...)

  • Other interesting software

    13 avril 2011, par

    We don’t claim to be the only ones doing what we do ... and especially not to assert claims to be the best either ... What we do, we just try to do it well and getting better ...
    The following list represents softwares that tend to be more or less as MediaSPIP or that MediaSPIP tries more or less to do the same, whatever ...
    We don’t know them, we didn’t try them, but you can take a peek.
    Videopress
    Website : http://videopress.com/
    License : GNU/GPL v2
    Source code : (...)

Sur d’autres sites (7167)

  • 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.

    


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