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 comment, rebase 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 96d060eb3013c943c0f40fb152a0ac41ac80fb0a..2a5df87cc4ac484fa36e9cd58577a56175dc7b19 100644

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

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

@@ -72,7 +72,6 @@ class VideoSendStream : public webrtc::VideoSendStream,

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

RtpStateMap GetRtpStates() const;

- int64_t GetRtt() const;

int GetPaddingNeededBps() const;

private:

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