Add logging of delay-based bandwidth estimate.

webrtc/logging/rtc_event_log/rtc_event_log.proto

 syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.rtclog;
 
 enum MediaType {
   ANY = 0;
   AUDIO = 1;
   VIDEO = 2;
   DATA = 3;
 }
@@ skipping 37 matching lines @@
 
   // optional - but required if type == RTCP_EVENT
   optional RtcpPacket rtcp_packet = 4;
 
   // optional - but required if type == AUDIO_PLAYOUT_EVENT
   optional AudioPlayoutEvent audio_playout_event = 5;
 
   // optional - but required if type == BWE_PACKET_LOSS_EVENT
   optional BwePacketLossEvent bwe_packet_loss_event = 6;
 
+  // optional - but required if type == BWE_PACKET_DELAY_EVENT
+  optional BwePacketDelayEvent bwe_packet_delay_event = 7;
+
   // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
   optional VideoReceiveConfig video_receiver_config = 8;
 
   // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
   optional VideoSendConfig video_sender_config = 9;
 
   // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
   optional AudioReceiveConfig audio_receiver_config = 10;
 
   // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
@@ skipping 42 matching lines @@
   // required - Fraction of lost packets since last receiver report
   // computed as floor( 256 * (#lost_packets / #total_packets) ).
   // The possible values range from 0 to 255.
   optional uint32 fraction_loss = 2;
 
   // TODO(terelius): Is this really needed? Remove or make optional?
   // required - Total number of packets that the BWE update is based on.
   optional int32 total_packets = 3;
 }
 
+message BwePacketDelayEvent {
+  enum DetectorState {
+    BWE_NORMAL = 0;
+    BWE_UNDERUSING = 1;
+    BWE_OVERUSING = 2;
+  }
+
+  // required - Bandwidth estimate (in bps) after the update.
+  optional int32 bitrate = 1;
+
+  // required - The state of the overuse detector.
+  optional DetectorState detector_state = 2;
+}
+
 // TODO(terelius): Video and audio streams could in principle share SSRC,
 // so identifying a stream based only on SSRC might not work.
 // It might be better to use a combination of SSRC and media type
 // or SSRC and port number, but for now we will rely on SSRC only.
 message VideoReceiveConfig {
   // required - Synchronization source (stream identifier) to be received.
   optional uint32 remote_ssrc = 1;
   // required - Sender SSRC used for sending RTCP (such as receiver reports).
   optional uint32 local_ssrc = 2;
 
@@ skipping 114 matching lines @@
   optional float uplink_packet_loss_fraction = 3;
 
   // Whether forward error correction (FEC) is turned on or off.
   optional bool enable_fec = 4;
 
   // Whether discontinuous transmission (DTX) is turned on or off.
   optional bool enable_dtx = 5;
 
   // Number of audio channels that each encoded packet consists of.
   optional uint32 num_channels = 6;
-}
+}