Create Audio Sending Stream

user.cpp

webrtc::AudioSendStream* createAudioSendStream(
  uint32_t ssrc,
  uint8_t payloadType,
  webrtc::Transport* transport,
  rtc::scoped_refptr<webrtc::AudioEncoderFactory> audioEncoderFactory,
  webrtc::Call* call)
{
    webrtc::AudioSendStream::Config config{transport};
    config.rtp.ssrc = ssrc;
    config.rtp.extensions = {{"urn:ietf:params:rtp-hdrext:ssrc-audio-level", 1}};
    config.encoder_factory = audioEncoderFactory;
    const webrtc::SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
    config.send_codec_spec =
      webrtc::AudioSendStream::Config::SendCodecSpec(payloadType, kOpusFormat);
    webrtc::AudioSendStream* audioStream = call->CreateAudioSendStream(config);
    audioStream->Start();
    return audioStream;
}