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; |