Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
These are already existed in StreamDataCounters. This CL takes care of
the plumbing of these values to the standard stats collector.
TBR=solenberg@webrtc.org
Bug: webrtc:10447
Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27663}diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7804f55..283fd9a 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -389,7 +389,9 @@
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
stats.bytes_sent = call_stats.bytesSent;
+ stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
stats.packets_sent = call_stats.packetsSent;
+ stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
// RTT isn't known until a RTCP report is received. Until then, VoiceEngine
// returns 0 to indicate an error value.
if (call_stats.rttMs > 0) {
