Index: webrtc/api/rtpsender.cc |
diff --git a/webrtc/api/rtpsender.cc b/webrtc/api/rtpsender.cc |
index c5db92946d200ab0d643107cfa871fe4e6c519a6..822b7f457e47e96506f13303a002b614320c6624 100644 |
--- a/webrtc/api/rtpsender.cc |
+++ b/webrtc/api/rtpsender.cc |
@@ -36,7 +36,7 @@ void LocalAudioSinkAdapter::OnData(const void* audio_data, |
} |
} |
-void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) { |
+void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) { |
rtc::CritScope lock(&lock_); |
ASSERT(!sink || !sink_); |
sink_ = sink; |
@@ -194,9 +194,9 @@ void AudioRtpSender::SetAudioSend() { |
} |
#endif |
- cricket::AudioRenderer* renderer = sink_adapter_.get(); |
- ASSERT(renderer != nullptr); |
- provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer); |
+ cricket::AudioSource* source = sink_adapter_.get(); |
+ ASSERT(source != nullptr); |
+ provider_->SetAudioSend(ssrc_, track_->enabled(), options, source); |
} |
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track, |