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 609 matching lines @@
   LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
                << (aggregate_state == kNetworkUp ? "up" : "down");
 
   congestion_controller_->SignalNetworkState(aggregate_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_) {
+    // TODO(asapersson): Use sent_packet.send_time_ms.
+    if (stream->OnSentPacket(sent_packet.packet_id))
+      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 158 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