Chromium Code Reviews| Index: webrtc/call/audio_receive_stream.h |
| diff --git a/webrtc/call/audio_receive_stream.h b/webrtc/call/audio_receive_stream.h |
| index 3959da1369eeb1cd11dc9eecad6b536d93b789b8..fca84918910f381424a284f80092db02f2d906d3 100644 |
| --- a/webrtc/call/audio_receive_stream.h |
| +++ b/webrtc/call/audio_receive_stream.h |
| @@ -18,6 +18,7 @@ |
| #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" |
| @@ -133,6 +134,8 @@ class AudioReceiveStream { |
| // is potentially forwarded to any attached AudioSinkInterface implementation. |
| virtual void SetGain(float gain) = 0; |
| + virtual const std::vector<RtpSource> GetSources() = 0; |
|
hbos
2017/04/05 11:15:41
Make the return value non-const since it's returne
Zhi Huang
2017/04/06 03:09:49
Done.
|
| + |
| protected: |
| virtual ~AudioReceiveStream() {} |
| }; |