Fixig issues in bwe plot dynamics.

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 884 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_WITH_SSRC(1, "VideoTotBitrate[kbps]", now_ms,
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
                                     ActualSendBitrateKbit(), packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate[Kbps]", now_ms,
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
                                     FecOverheadRate() / 1000, packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate[Kbps]", now_ms,
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
                                     NackOverheadRate() / 1000, packet->Ssrc());
   } else {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate[kbps]", now_ms,
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
                                     ActualSendBitrateKbit(), packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate[Kbps]", now_ms,
+    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
                                     NackOverheadRate() / 1000, packet->Ssrc());
   }
 
   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();
@@ skipping 838 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = timestamp_offset_;
 
   return state;
 }
 
 }  // namespace webrtc