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.
  */
 
@@ skipping 11 matching lines @@
 #include "webrtc/call/rtc_event_log.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"
 
+#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+
 namespace webrtc {
 
 namespace {
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
 constexpr size_t kMaxPaddingLength = 224;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr size_t kRtpHeaderLength = 12;
 constexpr uint16_t kMaxInitRtpSeqNumber = 32767;  // 2^15 -1.
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr int kBitrateStatisticsWindowMs = 1000;
@@ skipping 855 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);
 
+  if (video_) {
+    BWE_TEST_LOGGING_PLOT(1, "SentBitrate[Kbps]", \
+      now_ms, ActualSendBitrateKbit());
+    BWE_TEST_LOGGING_PLOT(1, "FecBitrate[Kbps]", \
+      now_ms, FecOverheadRate()/1000);
+    BWE_TEST_LOGGING_PLOT(1, "NackBitrate[Kbps]", \
+      now_ms, NackOverheadRate()/1000);
+    BWE_TEST_LOGGING_PLOT(1, "VideoBitrate[bps]", \
+      now_ms, VideoBitrateSent()/1000);
+  }
+
+  if (!video_) {
+    BWE_TEST_LOGGING_PLOT(1, "AudioSentBitrate[Kbps]", \
+      now_ms, ActualSendBitrateKbit());
+  }
+

[stefan-webrtc, 2016/09/01 14:07:36]

Note that these are measured per ssrc, so I think you have to add ssrc to the
logs. Or if you are only interested in per transport you could choose to do
something different. Maybe we're not even interested in knowing if it's audio,
video or retransmits etc?

[Gaetano Carlucci, 2016/09/01 16:06:26]

yes, adding ssrc should be enough 

   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