Add logging of delay-based bandwidth estimate.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 70 matching lines @@
       return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
   }
   RTC_NOTREACHED();
   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
 }
 
+BandwidthUsage GetRuntimeDetectorState(
+    rtclog::BwePacketDelayEvent::DetectorState detector_state) {
+  switch (detector_state) {
+    case rtclog::BwePacketDelayEvent::BWE_NORMAL:
+      return kBwNormal;
+    case rtclog::BwePacketDelayEvent::BWE_UNDERUSING:
+      return kBwUnderusing;
+    case rtclog::BwePacketDelayEvent::BWE_OVERUSING:
+      return kBwOverusing;
+  }
+  RTC_NOTREACHED();
+  return kBwNormal;
+}
+
 std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
   uint64_t varint = 0;
   for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
     // The most significant bit of each byte is 0 if it is the last byte in
     // the varint and 1 otherwise. Thus, we take the 7 least significant bits
     // of each byte and shift them 7 bits for each byte read previously to get
     // the (unsigned) integer.
     int byte = stream.get();
     if (stream.eof()) {
       return std::make_pair(varint, false);
@@ skipping 363 matching lines @@
   RTC_CHECK(loss_event.has_fraction_loss());
   if (fraction_loss != nullptr) {
     *fraction_loss = loss_event.fraction_loss();
   }
   RTC_CHECK(loss_event.has_total_packets());
   if (total_packets != nullptr) {
     *total_packets = loss_event.total_packets();
   }
 }
 
+void ParsedRtcEventLog::GetBwePacketDelayEvent(
+    size_t index,
+    int32_t* bitrate,
+    BandwidthUsage* detector_state) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PACKET_DELAY_EVENT);
+  RTC_CHECK(event.has_bwe_packet_delay_event());
+  const rtclog::BwePacketDelayEvent& delay_event =
+      event.bwe_packet_delay_event();
+  RTC_CHECK(delay_event.has_bitrate());
+  if (bitrate != nullptr) {
+    *bitrate = delay_event.bitrate();
+  }
+  RTC_CHECK(delay_event.has_detector_state());
+  if (detector_state != nullptr) {
+    *detector_state = GetRuntimeDetectorState(delay_event.detector_state());
+  }
+}
+
 void ParsedRtcEventLog::GetAudioNetworkAdaptation(
     size_t index,
     AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
   RTC_CHECK(event.has_audio_network_adaptation());
   const rtclog::AudioNetworkAdaptation& ana_event =
       event.audio_network_adaptation();
   if (ana_event.has_bitrate_bps())
     config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
   if (ana_event.has_enable_fec())
     config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
   if (ana_event.has_enable_dtx())
     config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
   if (ana_event.has_frame_length_ms())
     config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
   if (ana_event.has_num_channels())
     config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
   if (ana_event.has_uplink_packet_loss_fraction())
     config->uplink_packet_loss_fraction =
         rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
 }
 
 }  // namespace webrtc