Made AudioReceiveStream a mixer participant.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 11a1c5815ecdd7da7f7264f985559d856c8203a5..d05934297cc0c943a779c9be6c1bae0c0014af71 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -272,6 +272,15 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
+AudioMixer::Source::AudioFrameWithInfo
+AudioReceiveStream::GetAudioFrameWithInfo(int sample_rate_hz) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz);
+}
+
+int AudioReceiveStream::Ssrc() {
+  return config_.rtp.local_ssrc;
+}
+
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());