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_) {
+    stream->OnSentPacket(sent_packet.packet_id);

[stefan-webrtc, 2015/12/10 08:37:49]

Should this method return if the packet belonged to that send stream so that we
don't have to loop over all send streams? Or would it make sense to actually
move send delay tracking to Call instead? Maybe we want to track this per
stream.

[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,
                             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