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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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10
webrtc/call/call.cc
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@ -511,15 +511,7 @@ Call::Stats Call::GetStats() const {
stats.send_bandwidth_bps = send_bandwidth;
stats.recv_bandwidth_bps = recv_bandwidth;
stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs();
{
ReadLockScoped read_lock(*send_crit_);
// TODO(solenberg): Add audio send streams.
for (const auto& kv : video_send_ssrcs_) {
int rtt_ms = kv.second->GetRtt();
if (rtt_ms > 0)
stats.rtt_ms = rtt_ms;
}
}
stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
return stats;
}