Add a histogram metric tracking for how long audio RTP packets are sent

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.
  */
 
 #include <string.h>
 #include <algorithm>
 #include <map>
 #include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/audio/time_interval.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream_impl.h"
 #include "webrtc/call/rtp_stream_receiver_controller.h"
 #include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ skipping 289 matching lines @@
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   RateCounter received_bytes_per_second_counter_;
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
   rtc::Optional<int64_t> first_received_rtp_audio_ms_;
   rtc::Optional<int64_t> last_received_rtp_audio_ms_;
   rtc::Optional<int64_t> first_received_rtp_video_ms_;
   rtc::Optional<int64_t> last_received_rtp_video_ms_;
+  TimeInterval sent_rtp_audio_timer_ms_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // OnNetworkChanged from multiple threads.
   rtc::CriticalSection bitrate_crit_;
   uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
   AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
 
   std::map<std::string, rtc::NetworkRoute> network_routes_;
@@ skipping 161 matching lines @@
 
 void Call::UpdateHistograms() {
   RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.LifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 }
 
 void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) {
   if (first_sent_packet_ms == -1)
     return;
+  if (!sent_rtp_audio_timer_ms_.Empty()) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds",
+        sent_rtp_audio_timer_ms_.Length() / 1000);
+  }
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats send_bitrate_stats =
       estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
   if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
                                 send_bitrate_stats.average);
@@ skipping 118 matching lines @@
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (AudioReceiveStream* stream : audio_receive_streams_) {
       if (stream->config().rtp.local_ssrc == ssrc) {
         stream->AssociateSendStream(nullptr);
       }
     }
   }
   UpdateAggregateNetworkState();
+  sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_RUN_ON(&configuration_thread_checker_);
   event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       &audio_receiver_controller_, transport_send_->packet_router(), config,
@@ skipping 754 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc