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

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Unified Diff: webrtc/video/vie_channel.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: Cleanup Created 4 years, 11 months ago

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Index: webrtc/video/vie_channel.h

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

index 9f5e1ff90de397f6579b300282af48f9e2a2017f..8010385725ed8506a41a76cd6b3076f8c3d1573d 100644

--- a/webrtc/video/vie_channel.h

+++ b/webrtc/video/vie_channel.h

@@ -434,8 +434,6 @@ class ViEChannel : public VCMFrameTypeCallback,

int max_nack_reordering_threshold_;

I420FrameCallback* pre_render_callback_ GUARDED_BY(crit_);

- const rtc::scoped_ptr<ReportBlockStats> report_block_stats_sender_;

int64_t time_of_first_rtt_ms_ GUARDED_BY(crit_);

int64_t rtt_sum_ms_ GUARDED_BY(crit_);

int64_t last_rtt_ms_ GUARDED_BY(crit_);

« webrtc/video/send_statistics_proxy.cc ('K') | « webrtc/video/video_send_stream.cc ('k') | webrtc/video/vie_channel.cc » ('j') | webrtc/video/vie_channel.cc » ('J')