Add logging of delay-based bandwidth estimate.

webrtc/logging/rtc_event_log/rtc_event_log.proto

Index: webrtc/logging/rtc_event_log/rtc_event_log.proto
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto
index e80772297997ce2b088404d8ff83395b71fab4f8..0da910a29f51ab62e1e84440a520260bdc2f1f2f 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.proto
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto
@@ -55,6 +55,9 @@ message Event {
   // 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;
 
@@ -117,6 +120,20 @@ message BwePacketLossEvent {
   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
@@ -251,4 +268,4 @@ message AudioNetworkAdaptation {
 
   // Number of audio channels that each encoded packet consists of.
   optional uint32 num_channels = 6;
-}
+}