| Index: webrtc/audio_receive_stream.h
|
| diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
|
| index daf45985d33c2523f85ee26891f4134643b66243..8cab094f4bed211946f878cbdbda25708aa75886 100644
|
| --- a/webrtc/audio_receive_stream.h
|
| +++ b/webrtc/audio_receive_stream.h
|
| @@ -73,6 +73,12 @@ class AudioReceiveStream : public ReceiveStream {
|
| // Sender SSRC used for sending RTCP (such as receiver reports).
|
| uint32_t local_ssrc = 0;
|
|
|
| + // Enable feedback for send side bandwidth estimation.
|
| + // See
|
| + // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
|
| + // for details.
|
| + bool transport_cc = false;
|
| +
|
| // RTP header extensions used for the received stream.
|
| std::vector<RtpExtension> extensions;
|
| } rtp;
|
|
|