Use RateCounter for received bitrate stats.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index ceec963ee29a04b2f2f9f8f11dfa446d0861db86..bc336157478bf06794e3afdb5703a673af1c1d75 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -43,6 +43,7 @@
 #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"
@@ -182,6 +183,7 @@ class Call : public webrtc::Call,
   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.
@@ -240,6 +242,7 @@ Call::Call(const Call::Config& config)
       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),
@@ -352,9 +355,13 @@ void Call::UpdateReceiveHistograms() {
     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() {
@@ -819,6 +826,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
   //             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_);
@@ -874,6 +882,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     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;
@@ -886,6 +895,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     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;