Add logging of delay-based bandwidth estimate.

webrtc/logging/rtc_event_log/rtc_event_log_parser.h

 /*
  *  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.
  */
 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
@@ skipping 53 matching lines @@
   size_t GetNumberOfEvents() const;
 
   // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
   int64_t GetTimestamp(size_t index) const;
 
   // Reads the event type of the rtclog::Event at |index|.
   EventType GetEventType(size_t index) const;
 
   // Reads the header, direction, media type, header length and packet length
   // from the RTP event at |index|, and stores the values in the corresponding
-  // output parameters. The output parameters can be set to nullptr if those
-  // values aren't needed.
+  // output parameters. Each output parameter can be set to nullptr if that
+  // value isn't needed.
   // NB: The header must have space for at least IP_PACKET_SIZE bytes.
   void GetRtpHeader(size_t index,
                     PacketDirection* incoming,
                     MediaType* media_type,
                     uint8_t* header,
                     size_t* header_length,
                     size_t* total_length) const;
 
   // Reads packet, direction, media type and packet length from the RTCP event
   // at |index|, and stores the values in the corresponding output parameters.
-  // The output parameters can be set to nullptr if those values aren't needed.
+  // Each output parameter can be set to nullptr if that value isn't needed.
   // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
   void GetRtcpPacket(size_t index,
                      PacketDirection* incoming,
                      MediaType* media_type,
                      uint8_t* packet,
                      size_t* length) const;
 
   // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
   // Only the fields that are stored in the protobuf will be written.
   void GetVideoReceiveConfig(size_t index,
@@ skipping 11 matching lines @@
   // Reads a config event to a (non-NULL) AudioSendStream::Config struct.
   // Only the fields that are stored in the protobuf will be written.
   void GetAudioSendConfig(size_t index, AudioSendStream::Config* config) const;
 
   // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
   // in the output parameter ssrc. The output parameter can be set to nullptr
   // and in that case the function only asserts that the event is well formed.
   void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
 
   // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
-  // expected packets from the BWE event at |index| and stores the values in
-  // the corresponding output parameters. The output parameters can be set to
-  // nullptr if those values aren't needed.
-  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+  // expected packets from the loss based BWE event at |index| and stores the
+  // values in
+  // the corresponding output parameters. Each output parameter can be set to
+  // nullptr if that
+  // value isn't needed.
   void GetBwePacketLossEvent(size_t index,
                              int32_t* bitrate,
                              uint8_t* fraction_loss,
                              int32_t* total_packets) const;
 
+  // Reads bitrate and detector_state from the delay based BWE event at |index|
+  // and stores the values in the corresponding output parameters. Each output
+  // parameter can be set to nullptr if that
+  // value isn't needed.
+  void GetBwePacketDelayEvent(size_t index,
+                              int32_t* bitrate,
+                              BandwidthUsage* detector_state) const;
+
   // Reads a audio network adaptation event to a (non-NULL)
   // AudioNetworkAdaptor::EncoderRuntimeConfig struct. Only the fields that are
   // stored in the protobuf will be written.
   void GetAudioNetworkAdaptation(
       size_t index,
       AudioNetworkAdaptor::EncoderRuntimeConfig* config) const;
 
  private:
   std::vector<rtclog::Event> events_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_