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• 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;

• Streaming webm with ffmpeg/ffserver

  20 octobre 2014, par Mediocre Gopher

  I’m attempting to cast my desktop screen to an ffserver and stream it as a webm. I’m using the following ffserver configuration :

  <feed>               # This is the input feed where FFmpeg will send
  
     File ./feed1.ffm            # video stream.
  
     FileMaxSize 1G              # Maximum file size for buffering video
  
     ACL allow 127.0.0.1
  
     ACL allow localhost
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed feed1.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 720x576           # Video resolution
  
     VideoFrameRate 25           # Video FPS
  
  
  
     AVOptionVideo cpu-used 10
  
     AVOptionVideo qmin 10
  
     AVOptionVideo qmax 42
  
     AVOptionVideo quality good
  
     AVOptionAudio flags +global_header
  
     PreRoll 15
  
     StartSendOnKey
  
     VideoBitRate 400            # Video bitrate
  
  </stream>

  And the following command on my desktop :

  ffmpeg -f x11grab -r 25 -s 1280x800 -i :0.0 -f alsa -i pulse http://127.0.0.1:8090/feed1.ffm

  With ffmpeg being version 2.4.2 and with libvpx enabled (latest on Arch). I get the error :

  [libvpx @ 0x20a21a0] CQ level 0 must be between minimum and maximum quantizer value (10-42)

  On the client side. As far as I can tell from calling ffmpeg -h full there’s no way of setting the cq-level, and setting qmin to 0 doesn’t work (it ends up as 3 for some reason, I guess ffmpeg enforces a minimum).

  This configuration seems to have worked for others on the internet, but I can’t see how if cq-level defaults 0. If anyone has any ideas I’d really appreciate it.

• avfilter/blend_modes : Always preserve constness

  7 septembre 2023, par Andreas Rheinhardt

  avfilter/blend_modes : Always preserve constness
  These casts cast const away temporarily ; they are safe, because
  
  the pointers that are initialized point to const data. But this
  
  is nevertheless not nice and leads to warnings when using
  
   Wcast-qual. blend_modes.c generates 546 (2*39*7) such warnings
  
  which is the majority of such warnings for FFmpeg as a whole.
  
  vf_blend.c and vf_blend_init.h also use this pattern ;
  
  they have also been changed.
  Reviewed-by : Paul B Mahol <onemda@gmail.com>
  
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
  

  • [DH] libavfilter/blend_modes.c
  • [DH] libavfilter/vf_blend.c
  • [DH] libavfilter/vf_blend_init.h