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Unified Diff: webrtc/audio_receive_stream.h

Issue 1535963002: Wire-up BWE feedback for audio receive streams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix audio receive stream test expectation. Created 5 years ago
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« webrtc/audio/audio_send_stream_unittest.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/call/call.cc » ('j') | webrtc/call/mock/mock_congestion_controller.h » ('J')
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Index: webrtc/audio_receive_stream.h
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index daf45985d33c2523f85ee26891f4134643b66243..dc48bf6fac66bfa6bd17f8087090b6823514d237 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -73,6 +73,9 @@ class AudioReceiveStream : public ReceiveStream {
// Sender SSRC used for sending RTCP (such as receiver reports).
uint32_t local_ssrc = 0;
+ // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
the sun 2015/12/20 23:16:15 // Enable send side bandwidth estimation. // See .
stefan-webrtc 2015/12/21 08:01:29 Done.
+ bool transport_cc = false;
+
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
« webrtc/audio/audio_send_stream_unittest.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | webrtc/call/call.cc » ('j') | webrtc/call/mock/mock_congestion_controller.h » ('J')

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