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}

4 files changed, 26 insertions, 15 deletions

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 20aa217144..f248c99c6d 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -767,7 +767,11 @@ CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
   if (statistician) {
     StreamDataCounters data_counters;
     statistician->GetReceiveStreamDataCounters(&data_counters);
-    stats.bytesReceived = data_counters.transmitted.payload_bytes;
+    // TODO(http://crbug.com/webrtc/10525): Bytes received should only include
+    // payload bytes, not header and padding bytes.
+    stats.bytesReceived = data_counters.transmitted.payload_bytes +
+                          data_counters.transmitted.header_bytes +
+                          data_counters.transmitted.padding_bytes;
     stats.packetsReceived = data_counters.transmitted.packets;
     stats.last_packet_received_timestamp_ms =
         data_counters.last_packet_received_timestamp_ms;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index f00e0dcd8d..8ce33a46c1 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -1078,8 +1078,13 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
   _rtpRtcpModule->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  // TODO(https://crbug.com/webrtc/10525): Bytes sent should only include
+  // payload bytes, not header and padding bytes.
   stats.bytesSent =
-      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+      rtp_stats.transmitted.payload_bytes +
+      rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
+      rtx_stats.transmitted.payload_bytes +
+      rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
   // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
   // separate outbound-rtp stream objects.
   stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 7fa02487f5..9658ade62d 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -2362,7 +2362,11 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
        it != stats.substreams.end(); ++it) {
     // TODO(pbos): Wire up additional stats, such as padding bytes.
     webrtc::VideoSendStream::StreamStats stream_stats = it->second;
-    info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes;
+    // TODO(http://crbug.com/webrtc/10525): Bytes sent should only include
+    // payload bytes, not header and padding bytes.
+    info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
+                       stream_stats.rtp_stats.transmitted.header_bytes +
+                       stream_stats.rtp_stats.transmitted.padding_bytes;
     info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
     info.total_packet_send_delay_ms += stream_stats.total_packet_send_delay_ms;
     // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up
@@ -2779,7 +2783,9 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
   if (stats.current_payload_type != -1) {
     info.codec_payload_type = stats.current_payload_type;
   }
-  info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes;
+  info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
+                    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.packets_lost;
 
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 68c25be414..4874cf6200 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -87,8 +87,6 @@ static const uint32_t kFlexfecSsrc = 5;
 static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE;
 static const uint32_t kDefaultRecvSsrc = 0;
 
-constexpr uint32_t kRtpHeaderSize = 12;
-
 static const char kUnsupportedExtensionName[] =
     "urn:ietf:params:rtp-hdrext:unsupported";
 
@@ -1605,8 +1603,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
   ASSERT_EQ(1U, info.senders.size());
   // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
   // For webrtc, bytes_sent does not include the RTP header length.
-  EXPECT_EQ(info.senders[0].bytes_sent,
-            NumRtpBytes() - kRtpHeaderSize * NumRtpPackets());
+  EXPECT_GT(info.senders[0].bytes_sent, 0);
   EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
   EXPECT_EQ(0.0, info.senders[0].fraction_lost);
   ASSERT_TRUE(info.senders[0].codec_payload_type);
@@ -1629,8 +1626,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
   EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
   ASSERT_TRUE(info.receivers[0].codec_payload_type);
   EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
-  EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
-            info.receivers[0].bytes_rcvd);
+  EXPECT_EQ(NumRtpBytes(), info.receivers[0].bytes_rcvd);
   EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
   EXPECT_EQ(0, info.receivers[0].packets_lost);
   // TODO(asapersson): Not set for webrtc. Handle missing stats.
@@ -1681,8 +1677,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
   ASSERT_EQ(1U, info.senders.size());
   // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
   // For webrtc, bytes_sent does not include the RTP header length.
-  EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
-                 GetSenderStats(0).bytes_sent, kTimeout);
+  EXPECT_GT(GetSenderStats(0).bytes_sent, 0);
   EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
   EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
   EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
@@ -1691,8 +1686,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
   for (size_t i = 0; i < info.receivers.size(); ++i) {
     EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
     EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
-    EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
-                   GetReceiverStats(i).bytes_rcvd, kTimeout);
+    EXPECT_EQ_WAIT(NumRtpBytes(), GetReceiverStats(i).bytes_rcvd, kTimeout);
     EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
     EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
     EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
@@ -5176,7 +5170,9 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
 
   cricket::VideoMediaInfo info;
   ASSERT_TRUE(channel_->GetStats(&info));
-  EXPECT_EQ(stats.rtp_stats.transmitted.payload_bytes,
+  EXPECT_EQ(stats.rtp_stats.transmitted.payload_bytes +
+                stats.rtp_stats.transmitted.header_bytes +
+                stats.rtp_stats.transmitted.padding_bytes,
             rtc::checked_cast<size_t>(info.receivers[0].bytes_rcvd));
   EXPECT_EQ(stats.rtp_stats.transmitted.packets,
             rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));