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 22 matching lines @@
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
+#include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 
 namespace webrtc {
 
 const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace internal {
@@ skipping 117 matching lines @@
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
   std::map<std::string, rtc::NetworkRoute> network_routes_;
 
   VieRemb remb_;
   const std::unique_ptr<CongestionController> congestion_controller_;
+  const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
@@ skipping 13 matching lines @@
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
       last_rtp_packet_received_ms_(-1),
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
-      congestion_controller_(new CongestionController(clock_, this, &remb_)) {
+      congestion_controller_(new CongestionController(clock_, this, &remb_)),
+      video_send_delay_stats_(new SendDelayStats(clock_)) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   if (config.audio_state.get()) {
     ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
@@ skipping 172 matching lines @@
   UpdateAggregateNetworkState();
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
+  video_send_delay_stats_->AddSsrcs(config);
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
-      congestion_controller_.get(), bitrate_allocator_.get(), &remb_, config,
-      encoder_config, suspended_video_send_ssrcs_);
+      congestion_controller_.get(), bitrate_allocator_.get(),
+      video_send_delay_stats_.get(), &remb_, config, encoder_config,
+      suspended_video_send_ssrcs_);
   {
     WriteLockScoped write_lock(*send_crit_);
     for (uint32_t ssrc : config.rtp.ssrcs) {
       RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
       video_send_ssrcs_[ssrc] = send_stream;
     }
     video_send_streams_.insert(send_stream);
   }
   send_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
@@ skipping 232 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();
+  video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
+                                        clock_->TimeInMilliseconds());
   congestion_controller_->OnSentPacket(sent_packet);
 }
 
 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;
   {
@@ skipping 168 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