diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 5aa7228947..4b5bd3a292 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" @@ -176,12 +177,11 @@ class Call : public webrtc::Call, // 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_; - 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_; + RateCounter received_audio_bytes_per_second_counter_; + RateCounter received_video_bytes_per_second_counter_; + RateCounter received_rtcp_bytes_per_second_counter_; // TODO(holmer): Remove this lock once BitrateController no longer calls // OnNetworkChanged from multiple threads. @@ -239,12 +239,11 @@ Call::Call(const Call::Config& config) 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, true), + received_audio_bytes_per_second_counter_(clock_, nullptr, true), + received_video_bytes_per_second_counter_(clock_, nullptr, true), + received_rtcp_bytes_per_second_counter_(clock_, nullptr, true), estimated_send_bitrate_sum_kbits_(0), pacer_bitrate_sum_kbits_(0), min_allocated_send_bitrate_bps_(0), @@ -341,30 +340,31 @@ void Call::UpdateSendHistograms() { } void Call::UpdateReceiveHistograms() { - if (first_rtp_packet_received_ms_ == -1) - return; - int64_t elapsed_sec = - (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000; - if (elapsed_sec < metrics::kMinRunTimeInSeconds) - return; - 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) { + const int kMinRequiredPeriodicSamples = 5; + AggregatedStats video_bytes_per_sec = + received_video_bytes_per_second_counter_.GetStats(); + if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps", - video_bitrate_kbps); + video_bytes_per_sec.average * 8 / 1000); } - if (audio_bitrate_kbps > 0) { + AggregatedStats audio_bytes_per_sec = + received_audio_bytes_per_second_counter_.GetStats(); + if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps", - audio_bitrate_kbps); + audio_bytes_per_sec.average * 8 / 1000); } - if (rtcp_bitrate_bps > 0) { + AggregatedStats rtcp_bytes_per_sec = + received_rtcp_bytes_per_second_counter_.GetStats(); + if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps", - rtcp_bitrate_bps); + rtcp_bytes_per_sec.average * 8); + } + 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); } - RTC_LOGGED_HISTOGRAM_COUNTS_100000( - "WebRTC.Call.BitrateReceivedInKbps", - audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000); } PacketReceiver* Call::Receiver() { @@ -843,7 +843,11 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, // 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; + if (received_bytes_per_second_counter_.HasSample()) { + // First RTP packet has been received. + received_bytes_per_second_counter_.Add(static_cast(length)); + received_rtcp_bytes_per_second_counter_.Add(static_cast(length)); + } bool rtcp_delivered = false; if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { ReadLockScoped read_lock(*receive_crit_); @@ -889,16 +893,13 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, 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_ = last_rtp_packet_received_ms_; - uint32_t ssrc = ByteReader::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(static_cast(length)); + received_audio_bytes_per_second_counter_.Add(static_cast(length)); auto status = it->second->DeliverRtp(packet, length, packet_time) ? DELIVERY_OK : DELIVERY_PACKET_ERROR; @@ -910,7 +911,8 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, 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(static_cast(length)); + received_video_bytes_per_second_counter_.Add(static_cast(length)); auto status = it->second->DeliverRtp(packet, length, packet_time) ? DELIVERY_OK : DELIVERY_PACKET_ERROR; diff --git a/webrtc/video/stats_counter.cc b/webrtc/video/stats_counter.cc index 42d23adcfc..5bcef991d2 100644 --- a/webrtc/video/stats_counter.cc +++ b/webrtc/video/stats_counter.cc @@ -74,6 +74,10 @@ AggregatedStats StatsCounter::GetStats() { return aggregated_counter_->ComputeStats(); } +bool StatsCounter::HasSample() const { + return last_process_time_ms_ != -1; +} + bool StatsCounter::TimeToProcess() { int64_t now = clock_->TimeInMilliseconds(); if (last_process_time_ms_ == -1) diff --git a/webrtc/video/stats_counter.h b/webrtc/video/stats_counter.h index c272b6278d..ba1bd457eb 100644 --- a/webrtc/video/stats_counter.h +++ b/webrtc/video/stats_counter.h @@ -82,6 +82,9 @@ class StatsCounter { AggregatedStats GetStats(); + // Checks if a sample has been added (i.e. Add or Set called). + bool HasSample() const; + protected: StatsCounter(Clock* clock, bool include_empty_intervals, diff --git a/webrtc/video/stats_counter_unittest.cc b/webrtc/video/stats_counter_unittest.cc index 5dd8b72cd6..d054eaa4b2 100644 --- a/webrtc/video/stats_counter_unittest.cc +++ b/webrtc/video/stats_counter_unittest.cc @@ -72,6 +72,13 @@ TEST_F(StatsCounterTest, TestRegisterObserver) { EXPECT_EQ(1, observer->num_calls_); } +TEST_F(StatsCounterTest, HasSample) { + AvgCounter counter(&clock_, nullptr); + EXPECT_FALSE(counter.HasSample()); + counter.Add(1); + EXPECT_TRUE(counter.HasSample()); +} + TEST_F(StatsCounterTest, VerifyProcessInterval) { StatsCounterObserverImpl* observer = new StatsCounterObserverImpl(); AvgCounter counter(&clock_, observer);