Add histogram stats for send delay for a sent video stream.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 550 matching lines @@
     for (auto& kv : video_receive_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
   }
 }
 
 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.send_time_ms?

[åsapersson, 2016/04/06 14:52:37]

Done.

+      return;
+  }
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
                             int64_t rtt_ms) {
   uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
       target_bitrate_bps, fraction_loss, rtt_ms);
 
   int pad_up_to_bitrate_bps = 0;
   {
     ReadLockScoped read_lock(*send_crit_);
@@ skipping 156 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc