Decoding Audio Track

[edwardsPaul421]

Summary

Hello,
I'm trying to write the audio I receive to PulseAudio Source-Output using Pulseaudio Simple API . Therefore, I need to decode my track from OPUS to PCM format.
I'm using hraban opus lib to decode the Payload of the track.
I get the decoded payload in PCM format in a int16[]. But, in order to write it to my pa source I need to write []byte.
I'm doing the conversion like that : binary.Write(stream, binary.LittleEndian, pcm).
However, when I play the audio I have a bit of "metalic voice" and cuts.
The problem is when writing the track using oggwriter, It works fine.

[ashellunts]

Hi @edwardsPaul421 , if you can provide minimal reproducer, I can try to help you.

[Sean-Der]

Hey @edwardsPaul421

Anything me or @ashellunts can do? Happy to leave this open for a couple more days :)

This could be a great repo for you to share in https://github.com/pion/awesome-pion could help out others.

thanks

[edwardsPaul421]

Hey @edwardsPaul421

Anything me or @ashellunts can do? Happy to leave this open for a couple more days :)

This could be a great repo for you to share in https://github.com/pion/awesome-pion could help out others.

thanks

Hi @Sean-Der @ashellunts sorry for the late reply.
I'm trying to decode RTP Packet's Payload (Which in format of OPUS) to PCM (Raw).
I'm using hraban opus lib to decode the Payload of the track, as mentioned above.
I'm also using mediadevices/opus lib to create codec like this:

	opusParams, err := opus.NewParams()
	if err != nil {
		panic(err)
	}
	codecSelector := mediadevices.NewCodecSelector(
		mediadevices.WithAudioEncoders(&opusParams),
	)

	mediaEngine := webrtc.MediaEngine{}
	codecSelector.Populate(&mediaEngine)
	api := webrtc.NewAPI(webrtc.WithMediaEngine(&mediaEngine))
	peerConnection, err := api.NewPeerConnection(config)
	if err != nil {
		panic(err)
	}

The problem is that both libs are defining the following C function:

int bridge_encoder_set_bitrate(OpusEncoder *e, opus_int32 bitrate)
{
	return opus_encoder_ctl(e, OPUS_SET_BITRATE(bitrate));
}

Therefore, I get a double definition error.

I've tried to avoid using mediadevices to create the Opus decode and to use the following code I saw here in one of the examples:

        m := &webrtc.MediaEngine{}
	if err := m.RegisterCodec(webrtc.RTPCodecParameters{
		RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus, ClockRate: 48000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: nil},
		PayloadType:        111,
	}, webrtc.RTPCodecTypeAudio); err != nil {
		panic(err)
	}

But the program just dont work after I send the sdp.

[brotherdang]

@edwardsPaul421 Have you solve the problem ?? I have the same problem.

[Sean-Der]

Sorry this didn't addressed sooner

LiveKit's SIP implementation does an Encode + Decode of Opus to connect with telephony endpoints. See media.go for an example of how to do this!