| 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,
|
|
|