| Index: webrtc/audio/audio_receive_stream.h
|
| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
|
| index adac88387613a43a7f6220567a75c8e881fe313c..20ed4613a7ede16d1ad3aa8897cb957c8c282c34 100644
|
| --- a/webrtc/audio/audio_receive_stream.h
|
| +++ b/webrtc/audio/audio_receive_stream.h
|
| @@ -12,6 +12,7 @@
|
| #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
|
|
|
| #include <memory>
|
| +#include <vector>
|
|
|
| #include "webrtc/api/audio/audio_mixer.h"
|
| #include "webrtc/audio/audio_state.h"
|
| @@ -49,6 +50,7 @@
|
| int GetOutputLevel() const override;
|
| void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
|
| void SetGain(float gain) override;
|
| + std::vector<webrtc::RtpSource> GetSources() const override;
|
|
|
| // TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
|
| void OnRtpPacket(const RtpPacketReceived& packet);
|
|
|