Video/Screenshare loopback tool.

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 42 matching lines @@
     rtc::Optional<uint64_t> bitrate_bps;
     rtc::Optional<ProbeFailureReason> failure_reason;
   };
 
   struct BweDelayBasedUpdate {
     uint64_t timestamp;
     int32_t bitrate_bps;
     BandwidthUsage detector_state;
   };
 
+  struct BweAckedBitrateEvent {
+    uint64_t timestamp;
+    int32_t bitrate_bps;
+  };
+
+  struct AlrStateEvent {
+    uint64_t timestamp;
+    bool in_alr;
+    uint32_t usage_bps;
+  };
+
+  struct PacketQueueTime {
+    uint64_t timestamp;
+    uint32_t ssrc;
+    int64_t queue_time_ms;
+  };
+
   enum EventType {
     UNKNOWN_EVENT = 0,
     LOG_START = 1,
     LOG_END = 2,
     RTP_EVENT = 3,
     RTCP_EVENT = 4,
     AUDIO_PLAYOUT_EVENT = 5,
     LOSS_BASED_BWE_UPDATE = 6,
     DELAY_BASED_BWE_UPDATE = 7,
     VIDEO_RECEIVER_CONFIG_EVENT = 8,
     VIDEO_SENDER_CONFIG_EVENT = 9,
     AUDIO_RECEIVER_CONFIG_EVENT = 10,
     AUDIO_SENDER_CONFIG_EVENT = 11,
     AUDIO_NETWORK_ADAPTATION_EVENT = 16,
     BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
-    BWE_PROBE_RESULT_EVENT = 18
+    BWE_PROBE_RESULT_EVENT = 18,
+    BWE_ACKED_BITRATE_EVENT = 19,
+    ALR_STATE_EVENT = 20,
+    PACKET_QUEUE_TIME = 21
   };
 
   enum class MediaType { ANY, AUDIO, VIDEO, DATA };
 
   // Reads an RtcEventLog file and returns true if parsing was successful.
   bool ParseFile(const std::string& file_name);
 
   // Reads an RtcEventLog from a string and returns true if successful.
   bool ParseString(const std::string& s);
 
@@ skipping 78 matching lines @@
   // stored in the protobuf will be written.
   void GetAudioNetworkAdaptation(size_t index,
                                  AudioEncoderRuntimeConfig* config) const;
 
   BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
 
   BweProbeResultEvent GetBweProbeResult(size_t index) const;
 
   MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
 
+  BweAckedBitrateEvent GetAckedBitrate(size_t index) const;
+
+  AlrStateEvent GetAlrState(size_t index) const;
+
+  PacketQueueTime GetQueueTime(size_t index) const;
+
  private:
   rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
   std::vector<rtclog::StreamConfig> GetVideoSendConfig(
       const rtclog::Event& event) const;
   rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
   rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
 
   std::vector<rtclog::Event> events_;
 
   struct Stream {
@@ skipping 16 matching lines @@
 
   // To find configured extensions map for given stream, what are needed to
   // parse a header.
   typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
   std::map<StreamId, webrtc::RtpHeaderExtensionMap*> rtp_extensions_maps_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_