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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: 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..7edd3c5394262da55d6287bdc21c9f742db98036 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -568,6 +568,12 @@ 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_) {
+ if (stream->OnSentPacket(sent_packet.packet_id))
stefan-webrtc 2016/01/18 19:48:25 Should there be a TODO here to use the sent_packet
åsapersson 2016/04/06 14:52:37 Done.
+ return;
+ }
}
void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
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