diff --git a/webrtc/common_types.h b/webrtc/common_types.h index 43ae642375..daaf1ff973 100644 --- a/webrtc/common_types.h +++ b/webrtc/common_types.h @@ -22,6 +22,7 @@ #include "webrtc/api/video/video_timing.h" #include "webrtc/rtc_base/array_view.h" #include "webrtc/rtc_base/checks.h" +#include "webrtc/rtc_base/deprecation.h" #include "webrtc/rtc_base/optional.h" #include "webrtc/typedefs.h" @@ -156,13 +157,19 @@ enum FrameType { struct RtcpStatistics { RtcpStatistics() : fraction_lost(0), - cumulative_lost(0), - extended_max_sequence_number(0), + packets_lost(0), + extended_highest_sequence_number(0), jitter(0) {} uint8_t fraction_lost; - uint32_t cumulative_lost; - uint32_t extended_max_sequence_number; + union { + uint32_t packets_lost; + RTC_DEPRECATED uint32_t cumulative_lost; + }; + union { + uint32_t extended_highest_sequence_number; + RTC_DEPRECATED uint32_t extended_max_sequence_number; + }; uint32_t jitter; }; diff --git a/webrtc/media/engine/webrtcvideoengine.cc b/webrtc/media/engine/webrtcvideoengine.cc index 2651a9a3bb..6813ada0ca 100644 --- a/webrtc/media/engine/webrtcvideoengine.cc +++ b/webrtc/media/engine/webrtcvideoengine.cc @@ -2037,7 +2037,7 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo( stream_stats.rtp_stats.transmitted.header_bytes + stream_stats.rtp_stats.transmitted.padding_bytes; info.packets_sent += stream_stats.rtp_stats.transmitted.packets; - info.packets_lost += stream_stats.rtcp_stats.cumulative_lost; + info.packets_lost += stream_stats.rtcp_stats.packets_lost; if (stream_stats.width > info.send_frame_width) info.send_frame_width = stream_stats.width; if (stream_stats.height > info.send_frame_height) @@ -2445,7 +2445,7 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo( stats.rtp_stats.transmitted.header_bytes + stats.rtp_stats.transmitted.padding_bytes; info.packets_rcvd = stats.rtp_stats.transmitted.packets; - info.packets_lost = stats.rtcp_stats.cumulative_lost; + info.packets_lost = stats.rtcp_stats.packets_lost; info.fraction_lost = static_cast(stats.rtcp_stats.fraction_lost) / (1 << 8); diff --git a/webrtc/media/engine/webrtcvideoengine_unittest.cc b/webrtc/media/engine/webrtcvideoengine_unittest.cc index d6a98de8d2..f31ac512b7 100644 --- a/webrtc/media/engine/webrtcvideoengine_unittest.cc +++ b/webrtc/media/engine/webrtcvideoengine_unittest.cc @@ -3709,7 +3709,7 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) { stats.rtp_stats.transmitted.header_bytes = 3; stats.rtp_stats.transmitted.padding_bytes = 4; stats.rtp_stats.transmitted.packets = 5; - stats.rtcp_stats.cumulative_lost = 6; + stats.rtcp_stats.packets_lost = 6; stats.rtcp_stats.fraction_lost = 7; stream->SetStats(stats); @@ -3721,7 +3721,7 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) { info.receivers[0].bytes_rcvd); EXPECT_EQ(stats.rtp_stats.transmitted.packets, info.receivers[0].packets_rcvd); - EXPECT_EQ(stats.rtcp_stats.cumulative_lost, info.receivers[0].packets_lost); + EXPECT_EQ(stats.rtcp_stats.packets_lost, info.receivers[0].packets_lost); EXPECT_EQ(static_cast(stats.rtcp_stats.fraction_lost) / (1 << 8), info.receivers[0].fraction_lost); } diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc index 66f653790d..31fb74ee82 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc @@ -92,9 +92,9 @@ void Convert(const webrtc::NetEqNetworkStatistics& stats_raw, void Convert(const webrtc::RtcpStatistics& stats_raw, webrtc::neteq_unittest::RtcpStatistics* stats) { stats->set_fraction_lost(stats_raw.fraction_lost); - stats->set_cumulative_lost(stats_raw.cumulative_lost); + stats->set_cumulative_lost(stats_raw.packets_lost); stats->set_extended_max_sequence_number( - stats_raw.extended_max_sequence_number); + stats_raw.extended_highest_sequence_number); stats->set_jitter(stats_raw.jitter); } diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc index 0263e763ef..3f8ef0e4e9 100644 --- a/webrtc/modules/audio_coding/neteq/rtcp.cc +++ b/webrtc/modules/audio_coding/neteq/rtcp.cc @@ -56,24 +56,24 @@ void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { void Rtcp::GetStatistics(bool no_reset, RtcpStatistics* stats) { // Extended highest sequence number received. - stats->extended_max_sequence_number = + stats->extended_highest_sequence_number = (static_cast(cycles_) << 16) + max_seq_no_; // Calculate expected number of packets and compare it with the number of // packets that were actually received. The cumulative number of lost packets // can be extracted. uint32_t expected_packets = - stats->extended_max_sequence_number - base_seq_no_ + 1; + stats->extended_highest_sequence_number - base_seq_no_ + 1; if (received_packets_ == 0) { // No packets received, assume none lost. - stats->cumulative_lost = 0; + stats->packets_lost = 0; } else if (expected_packets > received_packets_) { - stats->cumulative_lost = expected_packets - received_packets_; - if (stats->cumulative_lost > 0xFFFFFF) { - stats->cumulative_lost = 0xFFFFFF; + stats->packets_lost = expected_packets - received_packets_; + if (stats->packets_lost > 0xFFFFFF) { + stats->packets_lost = 0xFFFFFF; } } else { - stats->cumulative_lost = 0; + stats->packets_lost = 0; } // Fraction lost since last report. diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc index 9c3e342c8a..57f3f73411 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc @@ -139,8 +139,8 @@ RTCPReportBlock RembReceiver::BuildReportBlock( RtcpStatistics stats; RTC_DCHECK(statistician->GetStatistics(&stats, true)); report_block.fractionLost = stats.fraction_lost; - report_block.cumulativeLost = stats.cumulative_lost; - report_block.extendedHighSeqNum = stats.extended_max_sequence_number; + report_block.cumulativeLost = stats.packets_lost; + report_block.extendedHighSeqNum = stats.extended_highest_sequence_number; report_block.jitter = stats.jitter; return report_block; } diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc index 150c88b426..7ac86ee0f2 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -265,8 +265,8 @@ RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() { // We need a counter for cumulative loss too. // TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24. cumulative_loss_ += missing; - stats.cumulative_lost = cumulative_loss_; - stats.extended_max_sequence_number = + stats.packets_lost = cumulative_loss_; + stats.extended_highest_sequence_number = (received_seq_wraps_ << 16) + received_seq_max_; // Note: internal jitter value is in Q4 and needs to be scaled by 1/16. stats.jitter = jitter_q4_ >> 4; @@ -514,12 +514,12 @@ std::vector ReceiveStatistics::RtcpReportBlocks( rtcp::ReportBlock& block = result.back(); block.SetMediaSsrc(statistician.first); block.SetFractionLost(stats.fraction_lost); - if (!block.SetCumulativeLost(stats.cumulative_lost)) { + if (!block.SetCumulativeLost(stats.packets_lost)) { LOG(LS_WARNING) << "Cumulative lost is oversized."; result.pop_back(); continue; } - block.SetExtHighestSeqNum(stats.extended_max_sequence_number); + block.SetExtHighestSeqNum(stats.extended_highest_sequence_number); block.SetJitter(stats.jitter); } return result; diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index 97f6fa3bdc..4264ac4582 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -201,14 +201,14 @@ TEST_F(ReceiveStatisticsTest, RtcpCallbacks) { EXPECT_EQ(1u, callback.num_calls_); EXPECT_EQ(callback.ssrc_, kSsrc1); - EXPECT_EQ(statistics.cumulative_lost, callback.stats_.cumulative_lost); - EXPECT_EQ(statistics.extended_max_sequence_number, - callback.stats_.extended_max_sequence_number); + EXPECT_EQ(statistics.packets_lost, callback.stats_.packets_lost); + EXPECT_EQ(statistics.extended_highest_sequence_number, + callback.stats_.extended_highest_sequence_number); EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost); EXPECT_EQ(statistics.jitter, callback.stats_.jitter); EXPECT_EQ(51, statistics.fraction_lost); - EXPECT_EQ(1u, statistics.cumulative_lost); - EXPECT_EQ(5u, statistics.extended_max_sequence_number); + EXPECT_EQ(1u, statistics.packets_lost); + EXPECT_EQ(5u, statistics.extended_highest_sequence_number); EXPECT_EQ(4u, statistics.jitter); receive_statistics_->RegisterRtcpStatisticsCallback(NULL); diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc index 1f1d4f81fa..94783b721f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc @@ -992,8 +992,9 @@ void RTCPReceiver::TriggerCallbacksFromRtcpPacket( if (stats_callback_) { for (const auto& report_block : packet_information.report_blocks) { RtcpStatistics stats; - stats.cumulative_lost = report_block.cumulativeLost; - stats.extended_max_sequence_number = report_block.extendedHighSeqNum; + stats.packets_lost = report_block.cumulativeLost; + stats.extended_highest_sequence_number = + report_block.extendedHighSeqNum; stats.fraction_lost = report_block.fractionLost; stats.jitter = report_block.jitter; diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index de08958ffd..4813cc4bb3 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -1047,15 +1047,14 @@ TEST_F(RtcpReceiverTest, Callbacks) { rtcp::ReceiverReport rr1; rr1.SetSenderSsrc(kSenderSsrc); rr1.AddReportBlock(rb1); - EXPECT_CALL( - callback, - StatisticsUpdated( - AllOf(Field(&RtcpStatistics::fraction_lost, kFractionLoss), - Field(&RtcpStatistics::cumulative_lost, kCumulativeLoss), - Field(&RtcpStatistics::extended_max_sequence_number, - kSequenceNumber), - Field(&RtcpStatistics::jitter, kJitter)), - kReceiverMainSsrc)); + EXPECT_CALL(callback, + StatisticsUpdated( + AllOf(Field(&RtcpStatistics::fraction_lost, kFractionLoss), + Field(&RtcpStatistics::packets_lost, kCumulativeLoss), + Field(&RtcpStatistics::extended_highest_sequence_number, + kSequenceNumber), + Field(&RtcpStatistics::jitter, kJitter)), + kReceiverMainSsrc)); EXPECT_CALL(rtp_rtcp_impl_, OnReceivedRtcpReportBlocks(_)); EXPECT_CALL(bandwidth_observer_, OnReceivedRtcpReceiverReport(_, _, _)); InjectRtcpPacket(rr1); diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc index 106b513712..394abe8552 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -3177,8 +3177,8 @@ TEST_F(EndToEndTest, GetStats) { receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0; receive_stats_filled_["StatisticsUpdated"] |= - stats.rtcp_stats.cumulative_lost != 0 || - stats.rtcp_stats.extended_max_sequence_number != 0 || + stats.rtcp_stats.packets_lost != 0 || + stats.rtcp_stats.extended_highest_sequence_number != 0 || stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0; receive_stats_filled_["DataCountersUpdated"] |= @@ -3244,8 +3244,8 @@ TEST_F(EndToEndTest, GetStats) { const VideoSendStream::StreamStats& stream_stats = it->second; send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |= - stream_stats.rtcp_stats.cumulative_lost != 0 || - stream_stats.rtcp_stats.extended_max_sequence_number != 0 || + stream_stats.rtcp_stats.packets_lost != 0 || + stream_stats.rtcp_stats.extended_highest_sequence_number != 0 || stream_stats.rtcp_stats.fraction_lost != 0; send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |= diff --git a/webrtc/video/receive_statistics_proxy_unittest.cc b/webrtc/video/receive_statistics_proxy_unittest.cc index 822fc453f4..6112dd7fd4 100644 --- a/webrtc/video/receive_statistics_proxy_unittest.cc +++ b/webrtc/video/receive_statistics_proxy_unittest.cc @@ -245,15 +245,15 @@ TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsRtcpStats) { RtcpStatistics rtcp_stats; rtcp_stats.fraction_lost = kFracLost; - rtcp_stats.cumulative_lost = kCumLost; - rtcp_stats.extended_max_sequence_number = kExtSeqNum; + rtcp_stats.packets_lost = kCumLost; + rtcp_stats.extended_highest_sequence_number = kExtSeqNum; rtcp_stats.jitter = kJitter; statistics_proxy_->StatisticsUpdated(rtcp_stats, kRemoteSsrc); VideoReceiveStream::Stats stats = statistics_proxy_->GetStats(); EXPECT_EQ(kFracLost, stats.rtcp_stats.fraction_lost); - EXPECT_EQ(kCumLost, stats.rtcp_stats.cumulative_lost); - EXPECT_EQ(kExtSeqNum, stats.rtcp_stats.extended_max_sequence_number); + EXPECT_EQ(kCumLost, stats.rtcp_stats.packets_lost); + EXPECT_EQ(kExtSeqNum, stats.rtcp_stats.extended_highest_sequence_number); EXPECT_EQ(kJitter, stats.rtcp_stats.jitter); } @@ -359,14 +359,14 @@ TEST_F(ReceiveStatisticsProxyTest, PacketLossHistogramIsUpdated) { // One report block received. RtcpStatistics rtcp_stats1; - rtcp_stats1.cumulative_lost = kCumLost1; - rtcp_stats1.extended_max_sequence_number = kExtSeqNum1; + rtcp_stats1.packets_lost = kCumLost1; + rtcp_stats1.extended_highest_sequence_number = kExtSeqNum1; statistics_proxy_->StatisticsUpdated(rtcp_stats1, kRemoteSsrc); // Two report blocks received. RtcpStatistics rtcp_stats2; - rtcp_stats2.cumulative_lost = kCumLost2; - rtcp_stats2.extended_max_sequence_number = kExtSeqNum2; + rtcp_stats2.packets_lost = kCumLost2; + rtcp_stats2.extended_highest_sequence_number = kExtSeqNum2; statistics_proxy_->StatisticsUpdated(rtcp_stats2, kRemoteSsrc); // Two received report blocks but min run time has not passed. @@ -392,8 +392,8 @@ TEST_F(ReceiveStatisticsProxyTest, PacketLossHistogramIsUpdated) { TEST_F(ReceiveStatisticsProxyTest, PacketLossHistogramIsNotUpdatedIfLessThanTwoReportBlocksAreReceived) { RtcpStatistics rtcp_stats1; - rtcp_stats1.cumulative_lost = 1; - rtcp_stats1.extended_max_sequence_number = 10; + rtcp_stats1.packets_lost = 1; + rtcp_stats1.extended_highest_sequence_number = 10; // Min run time has passed but no received report block. fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000); diff --git a/webrtc/video/report_block_stats.cc b/webrtc/video/report_block_stats.cc index dee5662c3c..f36feb9e67 100644 --- a/webrtc/video/report_block_stats.cc +++ b/webrtc/video/report_block_stats.cc @@ -34,9 +34,9 @@ void ReportBlockStats::Store(const RtcpStatistics& rtcp_stats, uint32_t remote_ssrc, uint32_t source_ssrc) { RTCPReportBlock block; - block.cumulativeLost = rtcp_stats.cumulative_lost; + block.cumulativeLost = rtcp_stats.packets_lost; block.fractionLost = rtcp_stats.fraction_lost; - block.extendedHighSeqNum = rtcp_stats.extended_max_sequence_number; + block.extendedHighSeqNum = rtcp_stats.extended_highest_sequence_number; block.jitter = rtcp_stats.jitter; block.remoteSSRC = remote_ssrc; block.sourceSSRC = source_ssrc; diff --git a/webrtc/video/report_block_stats_unittest.cc b/webrtc/video/report_block_stats_unittest.cc index d9863a712f..d8bd7be49b 100644 --- a/webrtc/video/report_block_stats_unittest.cc +++ b/webrtc/video/report_block_stats_unittest.cc @@ -57,9 +57,9 @@ class ReportBlockStatsTest : public ::testing::Test { RtcpStatistics RtcpReportBlockToRtcpStatistics( const RTCPReportBlock& stats) { RtcpStatistics block; - block.cumulative_lost = stats.cumulativeLost; + block.packets_lost = stats.cumulativeLost; block.fraction_lost = stats.fractionLost; - block.extended_max_sequence_number = stats.extendedHighSeqNum; + block.extended_highest_sequence_number = stats.extendedHighSeqNum; block.jitter = stats.jitter; return block; } diff --git a/webrtc/video/send_statistics_proxy_unittest.cc b/webrtc/video/send_statistics_proxy_unittest.cc index 274aca8d52..de42fe8b2a 100644 --- a/webrtc/video/send_statistics_proxy_unittest.cc +++ b/webrtc/video/send_statistics_proxy_unittest.cc @@ -133,9 +133,9 @@ class SendStatisticsProxyTest : public ::testing::Test { EXPECT_EQ(a.rtp_stats.fec.packets, b.rtp_stats.fec.packets); EXPECT_EQ(a.rtcp_stats.fraction_lost, b.rtcp_stats.fraction_lost); - EXPECT_EQ(a.rtcp_stats.cumulative_lost, b.rtcp_stats.cumulative_lost); - EXPECT_EQ(a.rtcp_stats.extended_max_sequence_number, - b.rtcp_stats.extended_max_sequence_number); + EXPECT_EQ(a.rtcp_stats.packets_lost, b.rtcp_stats.packets_lost); + EXPECT_EQ(a.rtcp_stats.extended_highest_sequence_number, + b.rtcp_stats.extended_highest_sequence_number); EXPECT_EQ(a.rtcp_stats.jitter, b.rtcp_stats.jitter); } } @@ -157,8 +157,8 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) { // Add statistics with some arbitrary, but unique, numbers. uint32_t offset = ssrc * sizeof(RtcpStatistics); - ssrc_stats.rtcp_stats.cumulative_lost = offset; - ssrc_stats.rtcp_stats.extended_max_sequence_number = offset + 1; + ssrc_stats.rtcp_stats.packets_lost = offset; + ssrc_stats.rtcp_stats.extended_highest_sequence_number = offset + 1; ssrc_stats.rtcp_stats.fraction_lost = offset + 2; ssrc_stats.rtcp_stats.jitter = offset + 3; callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); @@ -168,8 +168,8 @@ TEST_F(SendStatisticsProxyTest, RtcpStatistics) { // Add statistics with some arbitrary, but unique, numbers. uint32_t offset = ssrc * sizeof(RtcpStatistics); - ssrc_stats.rtcp_stats.cumulative_lost = offset; - ssrc_stats.rtcp_stats.extended_max_sequence_number = offset + 1; + ssrc_stats.rtcp_stats.packets_lost = offset; + ssrc_stats.rtcp_stats.extended_highest_sequence_number = offset + 1; ssrc_stats.rtcp_stats.fraction_lost = offset + 2; ssrc_stats.rtcp_stats.jitter = offset + 3; callback->StatisticsUpdated(ssrc_stats.rtcp_stats, ssrc); diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc index 961108d9b9..e8a9da2245 100644 --- a/webrtc/video/video_receive_stream.cc +++ b/webrtc/video/video_receive_stream.cc @@ -128,8 +128,8 @@ std::string VideoReceiveStream::Stats::ToString(int64_t time_ms) const { ss << "min_playout_delay_ms: " << min_playout_delay_ms << ", "; ss << "discarded: " << discarded_packets << ", "; ss << "sync_offset_ms: " << sync_offset_ms << ", "; - ss << "cum_loss: " << rtcp_stats.cumulative_lost << ", "; - ss << "max_ext_seq: " << rtcp_stats.extended_max_sequence_number << ", "; + ss << "cum_loss: " << rtcp_stats.packets_lost << ", "; + ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", "; ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", "; ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", "; ss << "pli: " << rtcp_packet_type_counts.pli_packets; diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc index d55ad017ff..8d587f82b3 100644 --- a/webrtc/video/video_send_stream.cc +++ b/webrtc/video/video_send_stream.cc @@ -260,8 +260,8 @@ std::string VideoSendStream::StreamStats::ToString() const { ss << "retransmit_bps: " << retransmit_bitrate_bps << ", "; ss << "avg_delay_ms: " << avg_delay_ms << ", "; ss << "max_delay_ms: " << max_delay_ms << ", "; - ss << "cum_loss: " << rtcp_stats.cumulative_lost << ", "; - ss << "max_ext_seq: " << rtcp_stats.extended_max_sequence_number << ", "; + ss << "cum_loss: " << rtcp_stats.packets_lost << ", "; + ss << "max_ext_seq: " << rtcp_stats.extended_highest_sequence_number << ", "; ss << "nack: " << rtcp_packet_type_counts.nack_packets << ", "; ss << "fir: " << rtcp_packet_type_counts.fir_packets << ", "; ss << "pli: " << rtcp_packet_type_counts.pli_packets; diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 047f3e19dd..d630575cbc 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -2675,8 +2675,8 @@ int Channel::GetRTPStatistics(CallStatistics& stats) { } stats.fractionLost = statistics.fraction_lost; - stats.cumulativeLost = statistics.cumulative_lost; - stats.extendedMax = statistics.extended_max_sequence_number; + stats.cumulativeLost = statistics.packets_lost; + stats.extendedMax = statistics.extended_highest_sequence_number; stats.jitterSamples = statistics.jitter; // --- RTT