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Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()

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Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
This commit is contained in:
sprang
2016-02-19 09:03:26 -08:00
committed by Commit bot
parent 45c44f0b94
commit e2d83d6560
12 changed files with 92 additions and 141 deletions
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15
webrtc/video/video_send_stream.cc
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@ -571,21 +571,6 @@ void VideoSendStream::SignalNetworkState(NetworkState state) {
vie_channel_.SetRTCPMode(RtcpMode::kOff);
}
int64_t VideoSendStream::GetRtt() const {
webrtc::RtcpStatistics rtcp_stats;
uint16_t frac_lost;
uint32_t cumulative_lost;
uint32_t extended_max_sequence_number;
uint32_t jitter;
int64_t rtt_ms;
if (vie_channel_.GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
&extended_max_sequence_number,
&jitter, &rtt_ms) == 0) {
return rtt_ms;
}
return -1;
}
int VideoSendStream::GetPaddingNeededBps() const {
return vie_encoder_.GetPaddingNeededBps();
}