Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estima…

webrtc/call/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 16 matching lines @@
   // The different types of events that can occur, the UNKNOWN_EVENT entry
   // is added in case future EventTypes are added, in that case old code will
   // receive the new events as UNKNOWN_EVENT.
   enum EventType {
     UNKNOWN_EVENT = 0;
     LOG_START = 1;
     LOG_END = 2;
     RTP_EVENT = 3;
     RTCP_EVENT = 4;
     AUDIO_PLAYOUT_EVENT = 5;
-    VIDEO_RECEIVER_CONFIG_EVENT = 6;
-    VIDEO_SENDER_CONFIG_EVENT = 7;
-    AUDIO_RECEIVER_CONFIG_EVENT = 8;
-    AUDIO_SENDER_CONFIG_EVENT = 9;
+    BWE_PACKET_LOSS_EVENT = 6;
+    BWE_PACKET_DELAY_EVENT = 7;
+    VIDEO_RECEIVER_CONFIG_EVENT = 8;
+    VIDEO_SENDER_CONFIG_EVENT = 9;
+    AUDIO_RECEIVER_CONFIG_EVENT = 10;
+    AUDIO_SENDER_CONFIG_EVENT = 11;
   }
 
   // required - Indicates the type of this event
   optional EventType type = 2;
 
   // optional - but required if type == RTP_EVENT
   optional RtpPacket rtp_packet = 3;
 
   // 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 == VIDEO_RECEIVER_CONFIG_EVENT
-  optional VideoReceiveConfig video_receiver_config = 6;
+  optional VideoReceiveConfig video_receiver_config = 8;
 
   // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
-  optional VideoSendConfig video_sender_config = 7;
+  optional VideoSendConfig video_sender_config = 9;
 
   // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
-  optional AudioReceiveConfig audio_receiver_config = 8;
+  optional AudioReceiveConfig audio_receiver_config = 10;
 
   // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
-  optional AudioSendConfig audio_sender_config = 9;
+  optional AudioSendConfig audio_sender_config = 11;
 }
 
 
 message RtpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
 
   // required
   optional MediaType type = 2;
 
@@ skipping 16 matching lines @@
 
   // required - The whole packet including both payload and header.
   optional bytes packet_data = 3;
 }
 
 message AudioPlayoutEvent {
   // required - The SSRC of the audio stream associated with the playout event.
   optional uint32 local_ssrc = 2;
 }
 
+message BwePacketLossEvent {
+  // required - Bandwidth estimate (in bps) after the update.
+  optional int32 bitrate = 1;
+
+  // 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;
+}
 
 // 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 23 matching lines @@
   repeated DecoderConfig decoders = 8;
 }
 
 
 // Maps decoder names to payload types.
 message DecoderConfig {
   // required
   optional string name = 1;
 
   // required
-  optional sint32 payload_type = 2;
+  optional int32 payload_type = 2;
 }
 
 
 // Maps RTP header extension names to numerical IDs.
 message RtpHeaderExtension {
   // required
   optional string name = 1;
 
   // required
-  optional sint32 id = 2;
+  optional int32 id = 2;
 }
 
 
 // RTX settings for incoming video payloads that may be received.
 // RTX is disabled if there's no config present.
 message RtxConfig {
   // required - SSRC to use for the RTX stream.
   optional uint32 rtx_ssrc = 1;
 
   // required - Payload type to use for the RTX stream.
-  optional sint32 rtx_payload_type = 2;
+  optional int32 rtx_payload_type = 2;
 }
 
 
 message RtxMap {
   // required
-  optional sint32 payload_type = 1;
+  optional int32 payload_type = 1;
 
   // required
   optional RtxConfig config = 2;
 }
 
 
 message VideoSendConfig {
   // Synchronization source (stream identifier) for outgoing stream.
   // One stream can have several ssrcs for e.g. simulcast.
   // At least one ssrc is required.
   repeated uint32 ssrcs = 1;
 
   // RTP header extensions used for the outgoing stream.
   repeated RtpHeaderExtension header_extensions = 2;
 
   // List of SSRCs for retransmitted packets.
   repeated uint32 rtx_ssrcs = 3;
 
   // required if rtx_ssrcs is used - Payload type for retransmitted packets.
-  optional sint32 rtx_payload_type = 4;
+  optional int32 rtx_payload_type = 4;
 
   // required - Canonical end-point identifier.
   optional string c_name = 5;
 
   // required - Encoder associated with the stream.
   optional EncoderConfig encoder = 6;
 }
 
 
 // Maps encoder names to payload types.
 message EncoderConfig {
   // required
   optional string name = 1;
 
   // required
-  optional sint32 payload_type = 2;
+  optional int32 payload_type = 2;
 }
 
 
 message AudioReceiveConfig {
   // 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;
 
   // RTP header extensions used for the received audio stream.
   repeated RtpHeaderExtension header_extensions = 3;
 }
 
 
 message AudioSendConfig {
   // required - Synchronization source (stream identifier) for outgoing stream.
   optional uint32 ssrc = 1;
 
   // RTP header extensions used for the outgoing audio stream.
   repeated RtpHeaderExtension header_extensions = 2;
 }