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• 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.

  


• Does the Remote I/O audio unit set the number of channels in the buffer ?

  25 novembre 2013, par awfulcode

  I'm using kxmovie (it's a ffmpeg-based video player) as a base for an app and I'm trying to figure out how the RemoteI/O unit works on iOS when the only thing connected to a device is headphones and we're playing a track with more than 2 channels (say a surround 6 track channel). It seems like it is going with the output channel setting and the buffer only has 2 channels. Is this because of Core Audio's pull structure ? And if so, what's happening to the other channels in the track ? Are they being downmixed or simply ignored ?

  The code for the render callback connected to the remoteio unit is here :

  - (BOOL) renderFrames: (UInt32) numFrames
  
                 ioData: (AudioBufferList *) ioData
  
  {
  
      NSLog(@"Number of channels in buffer: %lu",ioData->mNumberBuffers);
  
  
  
      for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
          memset(ioData->mBuffers[iBuffer].mData, 0, ioData->mBuffers[iBuffer].mDataByteSize);
  
      }
  
  
  
  
  
      if (_playing && _outputBlock ) {
  
  
  
          // Collect data to render from the callbacks
  
          _outputBlock(_outData, numFrames, _numOutputChannels);
  
  
  
          // Put the rendered data into the output buffer
  
          if (_numBytesPerSample == 4) // then we've already got floats
  
          {
  
              float zero = 0.0;
  
  
  
              for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
  
  
                  int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
  
  
  
                  for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
  
                      vDSP_vsadd(_outData+iChannel, _numOutputChannels, &zero, (float *)ioData->mBuffers[iBuffer].mData, thisNumChannels, numFrames);
  
                  }
  
              }
  
          }
  
          else if (_numBytesPerSample == 2) // then we need to convert SInt16 -> Float (and also scale)
  
          {
  
              float scale = (float)INT16_MAX;
  
              vDSP_vsmul(_outData, 1, &scale, _outData, 1, numFrames*_numOutputChannels);
  
  
  
              for (int iBuffer=0; iBuffer < ioData->mNumberBuffers; ++iBuffer) {
  
  
  
                  int thisNumChannels = ioData->mBuffers[iBuffer].mNumberChannels;
  
  
  
                  for (int iChannel = 0; iChannel < thisNumChannels; ++iChannel) {
  
                      vDSP_vfix16(_outData+iChannel, _numOutputChannels, (SInt16 *)ioData->mBuffers[iBuffer].mData+iChannel, thisNumChannels, numFrames);
  
                  }
  
              }
  
  
  
          }        
  
      }
  
  
  
      return noErr;
  
  }

  Thanks !

  edit : Here's the code for the ASBD (_ouputFormat). It's getting its values straight from the remoteio. You can also check the whole method file here.

  if (checkError(AudioUnitGetProperty(_audioUnit,
  
                                      kAudioUnitProperty_StreamFormat,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &_outputFormat,
  
                                      &size),
  
                 "Couldn't get the hardware output stream format"))
  
      return NO;
  
  
  
  
  
  _outputFormat.mSampleRate = _samplingRate;
  
  if (checkError(AudioUnitSetProperty(_audioUnit,
  
                                      kAudioUnitProperty_StreamFormat,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &_outputFormat,
  
                                      size),
  
                 "Couldn't set the hardware output stream format")) {
  
  
  
      // just warning
  
  }
  
  
  
  _numBytesPerSample = _outputFormat.mBitsPerChannel / 8;
  
  _numOutputChannels = _outputFormat.mChannelsPerFrame;
  
  
  
  NSLog(@"Current output bytes per sample: %ld", _numBytesPerSample);
  
  NSLog(@"Current output num channels: %ld", _numOutputChannels);
  
  
  
  // Slap a render callback on the unit
  
  AURenderCallbackStruct callbackStruct;
  
  callbackStruct.inputProc = renderCallback;
  
  callbackStruct.inputProcRefCon = (__bridge void *)(self);
  
  
  
  if (checkError(AudioUnitSetProperty(_audioUnit,
  
                                      kAudioUnitProperty_SetRenderCallback,
  
                                      kAudioUnitScope_Input,
  
                                      0,
  
                                      &callbackStruct,
  
                                      sizeof(callbackStruct)),
  
                 "Couldn't set the render callback on the audio unit"))
  
      return NO;

• avformat/matroskadec : Check min_luminance more thoroughly

  16 février 2021, par Andreas Rheinhardt

  avformat/matroskadec : Check min_luminance more thoroughly
  In the absence of an explicitly coded minimal luminance, the current
  
  code inferred it to be -1, an invalid value. Yet it did not check the
  
  value lateron at all, so that if a valid maximum luminance is
  
  encountered, but no minimal luminance, an invalid minimal luminance of
  
   1 is exported. If an minimal luminance element with a negative value is
  
  present, it is exported, too. This can be simply fixed by adding a check
  
  for the value of the element.
  Yet given that a minimal luminance of zero Cd/m² is legal and can be
  
  coded with a length of zero, we must not use a fake default value to
  
  find out whether the element is present or not. Therefore this patch
  
  uses an explicit counter for it.
  While just at it, also check for max_luminance > min_luminance.
  Reviewed-by : Ridley Combs <rcombs@rcombs.me>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavformat/matroskadec.c