Rewrote pacer and bandwidth UMA stats.

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 96 matching lines @@
 
   VoiceEngine* voice_engine() {
     internal::AudioState* audio_state =
         static_cast<internal::AudioState*>(config_.audio_state.get());
     if (audio_state)
       return audio_state->voice_engine();
     else
       return nullptr;
   }
 
-  void UpdateSendHistograms();
+  void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
 
   const Clock* const clock_;
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<CallStats> call_stats_;
   const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
@@ skipping 17 matching lines @@
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
   RtcEventLog* event_log_ = nullptr;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
-  rtc::RateTracker received_video_bytes_per_sec_;
-  rtc::RateTracker received_audio_bytes_per_sec_;
-  rtc::RateTracker received_rtcp_bytes_per_sec_;
+  int64_t received_video_bytes_;
+  int64_t received_audio_bytes_;
+  int64_t received_rtcp_bytes_;

[åsapersson, 2015/11/26 09:14:32]

size_t?

[stefan-webrtc, 2015/11/26 10:07:56]

Done.

+  int64_t first_rtp_packet_received_ms_;
+  int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
-  int64_t first_rtp_packet_received_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
-  rtc::RateTracker estimated_send_bitrate_kbps_ GUARDED_BY(&bitrate_crit_);
-  rtc::RateTracker pacer_bitrate_kbps_ GUARDED_BY(&bitrate_crit_);
-  uint32_t target_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
-  uint32_t pacer_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
-  int64_t last_bitrate_update_ms_ GUARDED_BY(&bitrate_crit_);
+  int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+  int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);

[åsapersson, 2015/11/26 09:14:32]

size_t

[stefan-webrtc, 2015/11/26 10:07:56]

Done.

+  int num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);

[åsapersson, 2015/11/26 09:14:32]

size_t?

[stefan-webrtc, 2015/11/26 10:07:56]

Done.

 
   const rtc::scoped_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
 }  // namespace internal
 
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       call_stats_(new CallStats()),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
-      received_video_bytes_per_sec_(1000, 1),
-      received_audio_bytes_per_sec_(1000, 1),
-      received_rtcp_bytes_per_sec_(1000, 1),
+      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),
-      first_rtp_packet_received_ms_(-1),
-      estimated_send_bitrate_kbps_(1000, 1),
-      pacer_bitrate_kbps_(1000, 1),
-      target_bitrate_bps_(0),
-      pacer_bitrate_bps_(0),
-      last_bitrate_update_ms_(-1),
+      estimated_send_bitrate_sum_kbits_(0),
+      pacer_bitrate_sum_kbits_(0),
+      num_bitrate_updates_(0),
       congestion_controller_(
           new CongestionController(module_process_thread_.get(),
                                    call_stats_.get(),
                                    this)) {
   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,
@@ skipping 26 matching lines @@
   RTC_CHECK(audio_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
   module_process_thread_->DeRegisterModule(call_stats_.get());
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
 void Call::UpdateSendHistograms() {
-  if (first_packet_sent_ms_ == -1)
+  if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
-  rtc::CritScope lock(&bitrate_crit_);
-  int send_bitrate_kbps = estimated_send_bitrate_kbps_.ComputeTotalRate();
-  int pacer_bitrate_kbps = pacer_bitrate_kbps_.ComputeTotalRate();
+  int send_bitrate_kbps =
+      estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
+  int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
   if (send_bitrate_kbps > 0) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
                                 send_bitrate_kbps);
   }
   if (pacer_bitrate_kbps > 0) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
                                 pacer_bitrate_kbps);
   }
 }
 
 void Call::UpdateReceiveHistograms() {
   if (first_rtp_packet_received_ms_ == -1)
     return;
   int64_t elapsed_sec =
-      (clock_->TimeInMilliseconds() - first_rtp_packet_received_ms_) / 1000;
+      (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
-  int audio_bitrate_kbps =
-      received_audio_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
-  int video_bitrate_kbps =
-      received_video_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
-  int rtcp_bitrate_bps = received_rtcp_bytes_per_sec_.ComputeTotalRate() * 8;
+  int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
+  int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
+  int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
   if (video_bitrate_kbps > 0) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
                                 video_bitrate_kbps);
   }
   if (audio_bitrate_kbps > 0) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
                                 audio_bitrate_kbps);
   }
   if (rtcp_bitrate_bps > 0) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
@@ skipping 280 matching lines @@
 }
 
 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);
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
                             int64_t rtt_ms) {
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  int64_t time_since_last_update_ms = 0;
-  {
-    rtc::CritScope lock(&bitrate_crit_);
-    if (last_bitrate_update_ms_ >= 0)
-      time_since_last_update_ms = now_ms - last_bitrate_update_ms_;
-    estimated_send_bitrate_kbps_.AddSamples(
-        time_since_last_update_ms * (target_bitrate_bps_ / 1000) / 1000);
-    pacer_bitrate_kbps_.AddSamples(time_since_last_update_ms *
-                                   (pacer_bitrate_bps_ / 1000) / 1000);
-    target_bitrate_bps_ = target_bitrate_bps;
-    last_bitrate_update_ms_ = now_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_);
     // No need to update as long as we're not sending.
     if (video_send_streams_.empty())
       return;
 
     for (VideoSendStream* stream : video_send_streams_)
       pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
   }
   // Allocated bitrate might be higher than bitrate estimate if enforcing min
   // bitrate, or lower if estimate is higher than the sum of max bitrates, so
   // set the pacer bitrate to the maximum of the two.
   uint32_t pacer_bitrate_bps =
       std::max(target_bitrate_bps, allocated_bitrate_bps);
   {
     rtc::CritScope lock(&bitrate_crit_);
-    pacer_bitrate_bps_ = pacer_bitrate_bps;
+    // We only update these stats if we have send streams, and assume that
+    // OnNetworkChanged is called roughly with a fixed frequency.
+    estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
+    pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
+    ++num_bitrate_updates_;
   }
   congestion_controller_->UpdatePacerBitrate(
       target_bitrate_bps / 1000,
       PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
       pad_up_to_bitrate_bps / 1000);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (voice_engine() == nullptr || sync_group.empty())
@@ skipping 42 matching lines @@
   }
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   // TODO(pbos): Figure out what channel needs it actually.
   //             Do NOT broadcast! Also make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
-  received_rtcp_bytes_per_sec_.AddSamples(length);
+  received_rtcp_bytes_ += length;
   bool rtcp_delivered = false;
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*receive_crit_);
     for (VideoReceiveStream* stream : video_receive_streams_) {
       if (stream->DeliverRtcp(packet, length)) {
         rtcp_delivered = true;
         if (event_log_)
           event_log_->LogRtcpPacket(true, media_type, packet, length);
       }
     }
@@ skipping 12 matching lines @@
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   // Minimum RTP header size.
   if (length < 12)
     return DELIVERY_PACKET_ERROR;
 
+  last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
   if (first_rtp_packet_received_ms_ == -1)
-    first_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
+    first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
 
   uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
   ReadLockScoped read_lock(*receive_crit_);
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     auto it = audio_receive_ssrcs_.find(ssrc);
     if (it != audio_receive_ssrcs_.end()) {
-      received_audio_bytes_per_sec_.AddSamples(length);
+      received_audio_bytes_ += length;
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
       if (status == DELIVERY_OK && event_log_)
         event_log_->LogRtpHeader(true, media_type, packet, length);
       return status;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     auto it = video_receive_ssrcs_.find(ssrc);
     if (it != video_receive_ssrcs_.end()) {
-      received_video_bytes_per_sec_.AddSamples(length);
+      received_video_bytes_ += length;
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
       if (status == DELIVERY_OK && event_log_)
         event_log_->LogRtpHeader(true, media_type, packet, length);
       return status;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
     MediaType media_type,
     const uint8_t* packet,
     size_t length,
     const PacketTime& packet_time) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // 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