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()); |