Use RateCounter for received bitrate 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 25 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/clock.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/stats_counter.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 115 matching lines @@
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
   std::unique_ptr<webrtc::RtcEventLog> event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   int64_t received_video_bytes_;
   int64_t received_audio_bytes_;
   int64_t received_rtcp_bytes_;

[stefan-webrtc, 2016/09/06 11:55:58]

Should we do the same thing for these three? In that case maybe they can replace
the new RateCounter too?

[åsapersson, 2016/09/06 15:13:23]

Also updated video, audio and rctp stats to use RateCounter.

   int64_t first_rtp_packet_received_ms_;
   int64_t last_rtp_packet_received_ms_;
   int64_t first_packet_sent_ms_;
+  RateCounter received_bytes_per_second_counter_;
 
   // TODO(holmer): Remove this lock once BitrateController no longer calls
   // 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_);
   uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
   uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
 
@@ skipping 38 matching lines @@
       video_network_state_(kNetworkUp),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
       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),
+      received_bytes_per_second_counter_(clock_, nullptr, false),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       min_allocated_send_bitrate_bps_(0),
       num_bitrate_updates_(0),
       configured_max_padding_bitrate_bps_(0),
       remb_(clock_),
       congestion_controller_(
           new CongestionController(clock_, this, &remb_, event_log_.get())),
       video_send_delay_stats_(new SendDelayStats(clock_)),
       start_ms_(clock_->TimeInMilliseconds()) {
@@ skipping 92 matching lines @@
                                        video_bitrate_kbps);
   }
   if (audio_bitrate_kbps > 0) {
     RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
                                        audio_bitrate_kbps);
   }
   if (rtcp_bitrate_bps > 0) {
     RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
                                        rtcp_bitrate_bps);
   }
-  RTC_LOGGED_HISTOGRAM_COUNTS_100000(
-      "WebRTC.Call.BitrateReceivedInKbps",
-      audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
+  const int kMinRequiredPeriodicSamples = 5;
+  AggregatedStats recv_bytes_per_sec =
+      received_bytes_per_second_counter_.GetStats();
+  if (recv_bytes_per_sec.num_samples >= kMinRequiredPeriodicSamples) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
+                                       recv_bytes_per_sec.average * 8 / 1000);
+  }
 }
 
 PacketReceiver* Call::Receiver() {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
@@ skipping 444 matching lines @@
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
   // TODO(pbos): 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_ += length;
+  received_bytes_per_second_counter_.Add(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 (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     ReadLockScoped read_lock(*receive_crit_);
@@ skipping 35 matching lines @@
   last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
   if (first_rtp_packet_received_ms_ == -1)
     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_ += length;
+      received_bytes_per_second_counter_.Add(length);
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
       if (status == DELIVERY_OK)
         event_log_->LogRtpHeader(kIncomingPacket, 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_ += length;
+      received_bytes_per_second_counter_.Add(length);
       auto status = it->second->DeliverRtp(packet, length, packet_time)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
       if (status == DELIVERY_OK)
         event_log_->LogRtpHeader(kIncomingPacket, 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