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I have set up an iOS broadcast extension and audio data is coming in through processSampleBuffer as a CMSampleBuffer.

I am sending this data through a Websocket connection to a Pion WebRTC sink, configured with MimeType: webrtc.MimeTypeOpus.

From the looks of it, the audio generated (RPSampleBufferType.audioApp) has 2 channels at a 44100 samplerate.

How do I convert the audio data to be suitable for playing through a WebRTC Opus audio stream?

I had some luck converting it to a PCMA audio stream by using a AudioConverterNew and

AudioStreamBasicDescription(
  mSampleRate: 8000.00,
  mFormatID: kAudioFormatLinearPCM,
  mFormatFlags: 0,
  mBytesPerPacket: 1,
  mFramesPerPacket: 1,
  mBytesPerFrame: 1,
  mChannelsPerFrame: 1,
  mBitsPerChannel: 8,
  mReserved: 0
)

but the audio mostly sounds garbled.

Jochen
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0 Answers0