Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 
 namespace webrtc {
@@ skipping 864 matching lines @@
 
   // |capture_time_ms| <= 0 is considered invalid.
   // TODO(holmer): This should be changed all over Video Engine so that negative
   // time is consider invalid, while 0 is considered a valid time.
   if (packet->capture_time_ms() > 0) {
     packet->SetExtension<TransmissionOffset>(
         kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
   }
   packet->SetExtension<AbsoluteSendTime>(now_ms);
 
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "SentBitrate[Kbps]", now_ms, \
+                                  ActualSendBitrateKbit(), packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "FecBitrate[Kbps]", now_ms,  \
+                                  FecOverheadRate()/1000, packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "NackBitrate[Kbps]", now_ms, \
+                                  NackOverheadRate()/1000, packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoBitrate[bps]", now_ms, \
+                                  VideoBitrateSent()/1000, packet->Ssrc());

[stefan-webrtc, 2016/09/02 11:13:04]

Same here, remove '\' and add spaces around '/'.

Btw, shouldn't we track the audio bitrate too? Maybe something like:

 BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "MediaBitrate[bps]", now_ms,
    video_ ? VideoBitrateSent() / 1000 : AudioBitrateSent() / 1000,   
    packet->Ssrc());

Or update VideoBitrateSent to also include audio packets and rename it
MediaBitrateSent. We never use RtpSender for audio and video at the same time
anyway.

+
   if (paced_sender_) {
     uint16_t seq_no = packet->SequenceNumber();
     uint32_t ssrc = packet->Ssrc();
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
     size_t payload_length = packet->payload_size();
     packet_history_.PutRtpPacket(std::move(packet), storage, false);
 
     paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms,
@@ skipping 804 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc