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Unified Diff: webrtc/call/call.cc

Issue 1478253002: Add histogram stats for send delay for a sent video stream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 4682e926bd719bc4a01fed9129e90ad1a5114dd7..b7980382cfa16664685c469e5e18f4838c696e0e 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -568,6 +568,11 @@ void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (first_packet_sent_ms_ == -1)
first_packet_sent_ms_ = clock_->TimeInMilliseconds();
congestion_controller_->OnSentPacket(sent_packet);
+
+ ReadLockScoped read_lock(*send_crit_);
+ for (VideoSendStream* stream : video_send_streams_) {
+ stream->OnSentPacket(sent_packet.packet_id);
stefan-webrtc 2015/12/10 08:37:49 Should this method return if the packet belonged t
åsapersson 2015/12/15 14:28:26 Done. See other comment about per stream.
+ }
}
void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
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