webrtc/video/video_send_stream.h - Issue 1669623004: Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

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Unified Diff: webrtc/video/video_send_stream.h

Issue 1669623004: Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Addressed comments Created 4 years, 10 months ago

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Index: webrtc/video/video_send_stream.h

diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h

index 4a58e6f9ad4f71b05a6e6330f655792bfde1ba67..61d0b4653db004d5ecdfa3d4ace977309921b156 100644

--- a/webrtc/video/video_send_stream.h

+++ b/webrtc/video/video_send_stream.h

@@ -69,7 +69,6 @@ class VideoSendStream : public webrtc::VideoSendStream,

typedef std::map<uint32_t, RtpState> RtpStateMap;

RtpStateMap GetRtpStates() const;

- int64_t GetRtt() const;

int GetPaddingNeededBps() const;

private:

« webrtc/video/end_to_end_tests.cc ('K') | « webrtc/video/send_statistics_proxy.cc ('k') | webrtc/video/video_send_stream.cc » ('j') | no next file with comments »