| Index: webrtc/call/audio_receive_stream.h
|
| diff --git a/webrtc/call/audio_receive_stream.h b/webrtc/call/audio_receive_stream.h
|
| index e3bdd452473666bc68bebd16116944b84176035f..3959da1369eeb1cd11dc9eecad6b536d93b789b8 100644
|
| --- a/webrtc/call/audio_receive_stream.h
|
| +++ b/webrtc/call/audio_receive_stream.h
|
| @@ -18,7 +18,6 @@
|
|
|
| #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
|
| #include "webrtc/api/call/transport.h"
|
| -#include "webrtc/api/rtpreceiverinterface.h"
|
| #include "webrtc/base/optional.h"
|
| #include "webrtc/base/scoped_ref_ptr.h"
|
| #include "webrtc/common_types.h"
|
| @@ -134,8 +133,6 @@
|
| // is potentially forwarded to any attached AudioSinkInterface implementation.
|
| virtual void SetGain(float gain) = 0;
|
|
|
| - virtual std::vector<RtpSource> GetSources() const = 0;
|
| -
|
| protected:
|
| virtual ~AudioReceiveStream() {}
|
| };
|
|
|