Revert "Only include payload in bytes sent/received."
This reverts commit 74a1b4b1321b426392d4c32e4a02361226ad5358. Reason for revert: requested by chromium Original change's description: > Only include payload in bytes sent/received. > > According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and > https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent > statistic should not include headers or padding. > > Similarly, according to > https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes > received are calculated the same way as bytes sent (eg. not including > padding or headers). > > This change stops adding padding and headers to these statistics. > > Bug: webrtc:8516,webrtc:10525 > Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28647} TBR=steveanton@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,mellem@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8516, webrtc:10525 Change-Id: Ibd31a8264c19f0c6f57d8deb3974593d198046ab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147400 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28701}
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@ -767,7 +767,11 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
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if (statistician) {
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StreamDataCounters data_counters;
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statistician->GetReceiveStreamDataCounters(&data_counters);
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stats.bytesReceived = data_counters.transmitted.payload_bytes;
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// TODO(http://crbug.com/webrtc/10525): Bytes received should only include
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// payload bytes, not header and padding bytes.
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stats.bytesReceived = data_counters.transmitted.payload_bytes +
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data_counters.transmitted.header_bytes +
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data_counters.transmitted.padding_bytes;
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stats.packetsReceived = data_counters.transmitted.packets;
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stats.last_packet_received_timestamp_ms =
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data_counters.last_packet_received_timestamp_ms;
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@ -1078,8 +1078,13 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
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StreamDataCounters rtp_stats;
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StreamDataCounters rtx_stats;
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_rtpRtcpModule->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
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// TODO(https://crbug.com/webrtc/10525): Bytes sent should only include
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// payload bytes, not header and padding bytes.
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stats.bytesSent =
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rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
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rtp_stats.transmitted.payload_bytes +
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rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
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rtx_stats.transmitted.payload_bytes +
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rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
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// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
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// separate outbound-rtp stream objects.
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stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
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@ -2362,7 +2362,11 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
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it != stats.substreams.end(); ++it) {
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// TODO(pbos): Wire up additional stats, such as padding bytes.
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webrtc::VideoSendStream::StreamStats stream_stats = it->second;
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info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes;
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// TODO(http://crbug.com/webrtc/10525): Bytes sent should only include
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// payload bytes, not header and padding bytes.
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info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
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stream_stats.rtp_stats.transmitted.header_bytes +
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stream_stats.rtp_stats.transmitted.padding_bytes;
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info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
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info.total_packet_send_delay_ms += stream_stats.total_packet_send_delay_ms;
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// TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up
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@ -2779,7 +2783,9 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
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if (stats.current_payload_type != -1) {
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info.codec_payload_type = stats.current_payload_type;
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}
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info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes;
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info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
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stats.rtp_stats.transmitted.header_bytes +
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stats.rtp_stats.transmitted.padding_bytes;
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info.packets_rcvd = stats.rtp_stats.transmitted.packets;
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info.packets_lost = stats.rtcp_stats.packets_lost;
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@ -87,8 +87,6 @@ static const uint32_t kFlexfecSsrc = 5;
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static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE;
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static const uint32_t kDefaultRecvSsrc = 0;
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constexpr uint32_t kRtpHeaderSize = 12;
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static const char kUnsupportedExtensionName[] =
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"urn:ietf:params:rtp-hdrext:unsupported";
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@ -1605,8 +1603,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
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ASSERT_EQ(1U, info.senders.size());
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// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
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// For webrtc, bytes_sent does not include the RTP header length.
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EXPECT_EQ(info.senders[0].bytes_sent,
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NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
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EXPECT_GT(info.senders[0].bytes_sent, 0);
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EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
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EXPECT_EQ(0.0, info.senders[0].fraction_lost);
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ASSERT_TRUE(info.senders[0].codec_payload_type);
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@ -1629,8 +1626,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
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EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
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ASSERT_TRUE(info.receivers[0].codec_payload_type);
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EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
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EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
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info.receivers[0].bytes_rcvd);
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EXPECT_EQ(NumRtpBytes(), info.receivers[0].bytes_rcvd);
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EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
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EXPECT_EQ(0, info.receivers[0].packets_lost);
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// TODO(asapersson): Not set for webrtc. Handle missing stats.
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@ -1681,8 +1677,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
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ASSERT_EQ(1U, info.senders.size());
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// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
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// For webrtc, bytes_sent does not include the RTP header length.
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EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
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GetSenderStats(0).bytes_sent, kTimeout);
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EXPECT_GT(GetSenderStats(0).bytes_sent, 0);
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EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
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EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
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EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
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@ -1691,8 +1686,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
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for (size_t i = 0; i < info.receivers.size(); ++i) {
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EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
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EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
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EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
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GetReceiverStats(i).bytes_rcvd, kTimeout);
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EXPECT_EQ_WAIT(NumRtpBytes(), GetReceiverStats(i).bytes_rcvd, kTimeout);
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EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
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EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
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EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
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@ -5176,7 +5170,9 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
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cricket::VideoMediaInfo info;
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ASSERT_TRUE(channel_->GetStats(&info));
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EXPECT_EQ(stats.rtp_stats.transmitted.payload_bytes,
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EXPECT_EQ(stats.rtp_stats.transmitted.payload_bytes +
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stats.rtp_stats.transmitted.header_bytes +
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stats.rtp_stats.transmitted.padding_bytes,
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rtc::checked_cast<size_t>(info.receivers[0].bytes_rcvd));
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EXPECT_EQ(stats.rtp_stats.transmitted.packets,
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rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));
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