Rename some variables and methods in RTC event log.

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 51 matching lines @@
     case rtclog::Event::LOG_START:
       return ParsedRtcEventLog::EventType::LOG_START;
     case rtclog::Event::LOG_END:
       return ParsedRtcEventLog::EventType::LOG_END;
     case rtclog::Event::RTP_EVENT:
       return ParsedRtcEventLog::EventType::RTP_EVENT;
     case rtclog::Event::RTCP_EVENT:
       return ParsedRtcEventLog::EventType::RTCP_EVENT;
     case rtclog::Event::AUDIO_PLAYOUT_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
-    case rtclog::Event::BWE_PACKET_LOSS_EVENT:
-      return ParsedRtcEventLog::EventType::BWE_PACKET_LOSS_EVENT;
-    case rtclog::Event::BWE_PACKET_DELAY_EVENT:
-      return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
+    case rtclog::Event::LOSS_BASED_BWE_UPDATE:
+      return ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE;
+    case rtclog::Event::DELAY_BASED_BWE_UPDATE:
+      return ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE;
     case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
       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) {
+    rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
   switch (detector_state) {
-    case rtclog::BwePacketDelayEvent::BWE_NORMAL:
+    case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
       return kBwNormal;
-    case rtclog::BwePacketDelayEvent::BWE_UNDERUSING:
+    case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
       return kBwUnderusing;
-    case rtclog::BwePacketDelayEvent::BWE_OVERUSING:
+    case rtclog::DelayBasedBweUpdate::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
@@ skipping 345 matching lines @@
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
   RTC_CHECK(event.has_audio_playout_event());
   const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
   RTC_CHECK(loss_event.has_local_ssrc());
   if (ssrc != nullptr) {
     *ssrc = loss_event.local_ssrc();
   }
 }
 
-void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
-                                              int32_t* bitrate,
+void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
+                                              int32_t* bitrate_bps,
                                               uint8_t* fraction_loss,
                                               int32_t* total_packets) 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_LOSS_EVENT);
-  RTC_CHECK(event.has_bwe_packet_loss_event());
-  const rtclog::BwePacketLossEvent& loss_event = event.bwe_packet_loss_event();
-  RTC_CHECK(loss_event.has_bitrate());
-  if (bitrate != nullptr) {
-    *bitrate = loss_event.bitrate();
+  RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
+  RTC_CHECK(event.has_loss_based_bwe_update());
+  const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
+  RTC_CHECK(loss_event.has_bitrate_bps());
+  if (bitrate_bps != nullptr) {
+    *bitrate_bps = loss_event.bitrate_bps();
   }
   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(
+void ParsedRtcEventLog::GetDelayBasedBweUpdate(
     size_t index,
-    int32_t* bitrate,
+    int32_t* bitrate_bps,
     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_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
+  RTC_CHECK(event.has_delay_based_bwe_update());
+  const rtclog::DelayBasedBweUpdate& delay_event =
+      event.delay_based_bwe_update();
+  RTC_CHECK(delay_event.has_bitrate_bps());
+  if (bitrate_bps != nullptr) {
+    *bitrate_bps = delay_event.bitrate_bps();
   }
   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 {
@@ skipping 13 matching lines @@
   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