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Unified Diff: webrtc/video/rtp_stream_receiver.h

Issue 2181383002: Add NACK rate throttling for audio channels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed race in test Created 4 years, 5 months ago
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« webrtc/video/end_to_end_tests.cc ('K') | « webrtc/video/end_to_end_tests.cc ('k') | webrtc/video/rtp_stream_receiver.cc » ('j') | no next file with comments »
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Index: webrtc/video/rtp_stream_receiver.h
diff --git a/webrtc/video/rtp_stream_receiver.h b/webrtc/video/rtp_stream_receiver.h
index d83d1bcd366c31f5284ad30ab76b222e74a69204..a73a4ca83fb9bd5be644e5cdcdc775271bc9ecd4 100644
--- a/webrtc/video/rtp_stream_receiver.h
+++ b/webrtc/video/rtp_stream_receiver.h
@@ -62,7 +62,8 @@ class RtpStreamReceiver : public RtpData, public RtpFeedback,
VieRemb* remb,
const VideoReceiveStream::Config* config,
ReceiveStatisticsProxy* receive_stats_proxy,
- ProcessThread* process_thread);
+ ProcessThread* process_thread,
+ RateLimiter* retranmission_rate_limiter);
stefan-webrtc 2016/07/28 15:06:15 retransmission
sprang_webrtc 2016/07/28 15:17:18 Done.
~RtpStreamReceiver();
bool SetReceiveCodec(const VideoCodec& video_codec);
« webrtc/video/end_to_end_tests.cc ('K') | « webrtc/video/end_to_end_tests.cc ('k') | webrtc/video/rtp_stream_receiver.cc » ('j') | no next file with comments »

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