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• Transcode to opus by Fluent-ffmpeg or ffmpeg from command line

  14 juillet 2017, par shamaleyte

  My purpose is to transcode a webm file into opus file.
  It works just fine as the following ;

  ffmpeg -i input.webm -vn -c:a copy output.opus

  But the generated opus file always starts from 4rd or 5th seconds when I play it. It seems like that the first seconds are lost. Any idea why it happens ?

  >ffmpeg -i x.webm -vn -c:a copy x1.opus
  
  ffmpeg version N-86175-g64ea4d1 Copyright (c) 2000-2017 the FFmpeg 
  
  developers
  
  built with gcc 6.3.0 (GCC)
  
  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid -
  
  -enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-
  
  avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls 
  
  --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-
  
  libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-
  
  libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb -
  
  -enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --
  
  enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --
  
  enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab -
  
  -enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-
  
  libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-
  
  libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  
    libavutil      55. 63.100 / 55. 63.100
  
    libavcodec     57. 96.101 / 57. 96.101
  
    libavformat    57. 72.101 / 57. 72.101
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 90.100 /  6. 90.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, matroska,webm, from 'x.webm':
  
    Metadata:
  
    encoder         : libwebm-0.2.1.0
  
    creation_time   : 2017-06-19T20:50:21.722000Z
  
  Duration: 00:00:32.33, start: 0.000000, bitrate: 134 kb/s
  
  Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  Stream #0:1(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 
  
  4:3, 16.67 fps, 16.67 tbr, 1k tbn, 1k tbc (default)
  
  Output #0, opus, to 'x1.opus':
  
  Metadata:
  
  encoder         : Lavf57.72.101
  
  Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  Metadata:
  
    encoder         : Lavf57.72.101
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=     114kB time=00:00:32.33 bitrate=  28.8kbits/s speed=3.22e+003x
  
  video:0kB audio:111kB subtitle:0kB other streams:0kB global headers:0kB 
  
  muxing overhead: 2.152229%

  It is jumping from 0 to 4th second .
  Please take a look at this screencast.
  https://www.screenmailer.com/v/52IXnpAarHavwJE

  This is the sample video file that I tried to transcode : https://drive.google.com/open?id=0B2sa3oV_Y3X_ZmVWX3MzTlRPSmc

  So I guess the transcoding starts right at the point that the voice comes in, why is that ?

• How to convert a Stream on the fly with FFMpegCore ?

  18 octobre 2023, par Adrian

  For a school project, I need to stream videos that I get from torrents while they are downloading on the server.
When the video is a .mp4 file, there's no problem, but I must also be able to stream .mkv files, and for that I need to convert them into .mp4 before sending them to the client, and I can't find a way to convert my Stream that I get from MonoTorrents with FFMpegCore into a Stream that I can send to my client.

  


  Here is the code I wrote to simply download and stream my torrent :

  


  var cEngine = new ClientEngine();

var manager = await cEngine.AddStreamingAsync(GenerateMagnet(torrent), ) ?? throw new Exception("An error occurred while creating the torrent manager");

await manager.StartAsync();
await manager.WaitForMetadataAsync();

var videoFile = manager.Files.OrderByDescending(f => f.Length).FirstOrDefault();
if (videoFile == null)
    return Results.NotFound();

var stream = await manager.StreamProvider!.CreateStreamAsync(videoFile, true);
return Results.File(stream, contentType: "video/mp4", fileDownloadName: manager.Name, enableRangeProcessing: true);


  


  I saw that the most common way to convert videos is by using ffmpeg. .NET has a package called FFMpefCore that is a wrapper for ffmpeg.

  


  To my previous code, I would add right before the return :

  


  if (!videoFile.Path.EndsWith(".mp4"))
{
    var outputStream = new MemoryStream();
    FFMpegArguments
        .FromPipeInput(new StreamPipeSource(stream), options =>
        {
            options.ForceFormat("mp4");
        })
        .OutputToPipe(new StreamPipeSink(outputStream))
        .ProcessAsynchronously();
    return Results.File(outputStream, contentType: "video/mp4", fileDownloadName: manager.Name, enableRangeProcessing: true);
}


  


  I unfortunately can't get a "live" Stream to send to my client.

  


• AVAssetWriter creating mp4 with no sound in last 50msec

  12 août 2015, par Joseph K

  I’m working on a project involving live streaming from the iPhone’s camera.

  To minimize loss during AVAssetWriter finishWriting, I use an array of 2 asset writers and swap them whenever I need to create an mp4 fragment out of the recorded buffers.

  Code responsible for capturing Audio & Video sample buffers

  func captureOutput(captureOutput: AVCaptureOutput!, didOutputSampleBuffer sampleBuffer: CMSampleBuffer!, fromConnection connection: AVCaptureConnection!) {
  
  
  
      if CMSampleBufferDataIsReady(sampleBuffer) <= 0 {
  
              println("Skipped sample because it was not ready")
  
              return
  
          }
  
  
  
      if captureOutput == audioOutput  {
  
  
  
          if audioWriterBuffers[0].readyForMoreMediaData() {
  
              if !writers[0].appendAudio(sampleBuffer) { println("Failed to append: \(recordingPackages[0].name). Error: \(recordingPackages[0].outputWriter.error.localizedDescription)") }
  
              else {
  
                  writtenAudioFrames++
  
  
  
                  if writtenAudioFrames == framesPerFragment {
  
                      writeFragment()
  
                  }
  
              }
  
          }
  
          else {
  
              println("Skipped audio sample; it is not ready.")
  
          }
  
      }
  
  
  
      else if captureOutput == videoOutput {
  
          //Video sample buffer
  
          if videoWriterBuffers[0].readyForMoreMediaData() {
  
              //Call startSessionAtSourceTime if needed
  
              //Append sample buffer with a source time
  
          }
  
      }
  
  }

  Code responsible for the writing and swapping

  func writeFragment() {
  
      writtenAudioFrames = 0
  
  
  
      swap(&writers[0], &writers[1])
  
      if !writers[0].startWriting() {println( "Failed to start OTHER writer writing") }
  
      else { startTime  = CFAbsoluteTimeGetCurrent() }
  
  
  
      audioWriterBuffers[0].markAsFinished()
  
      videoWriterBuffers[0].markAsFinished()
  
  
  
      writers[1].outputWriter.finishWritingWithCompletionHandler { () -> Void in
  
          println("Finish Package record Writing, now Resetting")
  
          //
  
          // Handle written MP4 fragment code 
  
          //
  
  
  
          //Reset Writer
  
          //Basically reallocate it as a new AVAssetWriter with a given URL and MPEG4 file Type and add inputs to it
  
          self.resetWriter()
  
      }

  The issue at hand

  The written MP4 fragments are being sent over to a local sandbox server to be analyzed.

  When MP4 fragments are stitched together using FFMpeg, there is a noticeable glitch in sound due to the fact that there is not audio at the last 50msec of every fragment.

  My audio AVAssetWriterInput’s settings are the following :

  static let audioSettings: [NSObject : AnyObject]! =
  
  [
  
      AVFormatIDKey : NSNumber(integer: kAudioFormatMPEG4AAC),
  
      AVNumberOfChannelsKey : NSNumber(integer: 1),
  
      AVSampleRateKey : NSNumber(int: 44100),
  
      AVEncoderBitRateKey : NSNumber(int: 64000),
  
  ]

  As such, I encode 44 audio sample buffers every second. They are all being successfully appended.

  Further resources

  Here’s a waveform display of the audio stream after concatenating the mp4 fragments

  

   !! Note that my fragments are about 2secs in length.
   !! Note that I’m focusing on audio since video frames are extremely smooth when jumping from one fragment to another.

  Any idea as to what is causing this ? I can provide further code or info if needed.