Is there any function for custom audio source in webRTC c++?

Question

I am using android app native code and I am using some audio file and audio processing. I need to send audio data (raw audio frame) to webRTC. But I am not able to find any API of webRTC to add custom audio source(not default audio source like mic).

I understand that I need to add AudioTrackInterface and for that I need to add AudioSourceInterface.

Answer 1

This method is actual for version 66 of WebRTC. It is not so simple and maybe not clear, but it really works. I try to explain main idea:

I inherit webrtc::AudioDeviceModule and override some methods for emulate 'Virtual audio device' for virtual playout and recording. On calls I just call standard AudioDeviceModule base methods with some modifications:

int16_t PlayoutDevices() => call base method, but return base + 1

int16_t RecordingDevices() => return base + 1

int32_t PlayoutDeviceName => return my virtual device name and GUID

int32_t RecordingDeviceName => return my virtual device name and GUID

void SendFrameP => return my virtual device audio data

void ReceiveFrameP => use received audio data by my virtual device

etc methods => just look at webrtc::AudioDeviceModule implementation.

Then you can use your own AudioDeviceModule as parameter to webrtc::CreatePeerConnectionFactory function and provide audio data as recording device and receive data as playout device.