Convert AAC to Opus audio format

[clone1018]

We presumably receive AAC in from our RTMP streaming clients, however our FTL & WebRTC stack require Opus. We need to convert the audio to Opus before sending it over the RTP stream.

Relevant lines:

rtmp-ingest/rtmp.go

Lines 170 to 221 in 98e8d35

	func (h *ConnHandler) OnAudio(timestamp uint32, payload io.Reader) error {
	return nil
	// Convert AAC to opus
	// https://github.com/notedit/rtc-rtmp - Has c bindings
	//var audio flvtag.AudioData
	//if err := flvtag.DecodeAudioData(payload, &audio); err != nil {
	// return err
	//}
	//
	//data := new(bytes.Buffer)
	//if _, err := io.Copy(data, audio.Data); err != nil {
	// return err
	//}
	//outBuf := data.Bytes()
	//
	//const bufferSize = 1000 // choose any buffer size you like. 1k is plenty.
	//
	//// Check the frame size. You don't need to do this if you trust your input.
	//frameSize := len(outBuf) // must be interleaved if stereo
	//frameSizeMs := float32(frameSize) / 2 * 1000 / float32(h.audioClockRate)
	//switch frameSizeMs {
	//case 2.5, 5, 10, 20, 40, 60:
	// // Good.
	//default:
	// h.log.Errorf("Illegal frame size: %d bytes (%f ms)", frameSize, frameSizeMs)
	// return fmt.Errorf("Illegal frame size: %d bytes (%f ms)", frameSize, frameSizeMs)
	//}
	//
	//opusData := make([]byte, bufferSize)
	//n, err := h.audioEncoder.Encode(outBuf, opusData)
	//if err != nil {
	// return err
	//}
	//opusOutput := opusData[:n]
	//
	//samples := uint32(len(opusData)) + h.audioClockRate
	//// Of course this doesn't fucking work, it's an h264 packetizer
	//packets := h.audioPacketizer.Packetize(opusOutput, samples)
	//
	//for _, p := range packets {
	// buf, err := p.Marshal()
	// if err != nil {
	// return err
	// }
	// if err := h.stream.WriteRTP(buf); err != nil {
	// return err
	// }
	//}
	//
	//return nil
	}

From my cursory research, it seemed like a native program would be require to accomplish this. There's no pure-Go solution I could find.

[clone1018]

This was completed in #8 and is working well.