BWE Simulation Framework: Standard plot logging

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc

 /*
  *  Copyright (c) 2013 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 250 matching lines @@
   ss << name << "_" << flow_id;
   name_ = ss.str();
 }
 
 RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
                                      const FlowIds& flow_ids,
                                      const char* name)
     : PacketProcessor(listener, flow_ids, kRegular),
       packets_per_second_stats_(),
       kbps_stats_(),
-      name_(),
+      name_(std::string(name) + "_A"),  // All flow ids.

[stefan-webrtc, 2015/07/24 09:30:42]

I don't fully understand this. What if a RateCounterFilter is created only to
cover a subset of flows?

[magalhaesc, 2015/07/24 11:46:40]

Right, I didn't consider that.
Replaced back with all flow ids.

       start_plotting_time_ms_(0) {
-  std::stringstream ss;
-  ss << name << "_";
-  for (int flow_id : flow_ids) {
-    ss << flow_id << ",";
-  }
-  name_ = ss.str();
 }
 
 RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
                                      const FlowIds& flow_ids,
                                      const char* name,
                                      int64_t start_plotting_time_ms)
     : RateCounterFilter(listener, flow_ids, name) {
   start_plotting_time_ms_ = start_plotting_time_ms;
 }
 
@@ skipping 11 matching lines @@
 Stats<double> RateCounterFilter::GetBitrateStats() const {
   return kbps_stats_;
 }
 
 void RateCounterFilter::Plot(int64_t timestamp_ms) {
   uint32_t plot_kbps = 0;
   if (timestamp_ms >= start_plotting_time_ms_) {
     plot_kbps = rate_counter_.bits_per_second() / 1000.0;
   }
   BWE_TEST_LOGGING_CONTEXT(name_.c_str());
-  BWE_TEST_LOGGING_PLOT(0, "Throughput_#1", timestamp_ms, plot_kbps);
+  BWE_TEST_LOGGING_PLOT(0, "Throughput_kbps#2", timestamp_ms, plot_kbps);

[stefan-webrtc, 2015/07/24 09:30:42]

Isn't #1 the bitrate axis and #2 the delay axis?

[magalhaesc, 2015/07/24 13:51:40]

Yes.
I was using d for delay, will go back to the standard:
-- 1 if the alignment is with the left axis.
-- 2 if the alignment is with the right axis.

   RTC_UNUSED(plot_kbps);
 }
 
 void RateCounterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   assert(in_out);
   for (const Packet* packet : *in_out) {
     rate_counter_.UpdateRates(packet->send_time_us(),
                               static_cast<int>(packet->payload_size()));
   }
   packets_per_second_stats_.Push(rate_counter_.packets_per_second());
@@ skipping 510 matching lines @@
 uint32_t PeriodicKeyFrameSource::NextPacketSize(uint32_t frame_size,
                                                 uint32_t remaining_payload) {
   uint32_t fragments =
       (frame_size + (kMaxPayloadSizeBytes - 1)) / kMaxPayloadSizeBytes;
   uint32_t avg_size = (frame_size + fragments - 1) / fragments;
   return std::min(avg_size, remaining_payload);
 }
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc