Moved RtcEventLog files from call/ to logging/ (new top level dir)

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;
 }
 
-
 // This is the main message to dump to a file, it can contain multiple event
 // messages, but it is possible to append multiple EventStreams (each with a
 // single event) to a file.
 // This has the benefit that there's no need to keep all data in memory.
 message EventStream {
   repeated Event stream = 1;
 }
 
-
 message Event {
   // required - Elapsed wallclock time in us since the start of the log.
   optional int64 timestamp_us = 1;
 
   // 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;
@@ skipping 30 matching lines @@
   // 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
   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;
 
   // required - The size of the packet including both payload and header.
   optional uint32 packet_length = 3;
 
   // required - The RTP header only.
   optional bytes header = 4;
 
   // Do not add code to log user payload data without a privacy review!
 }
 
-
 message RtcpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
 
   // required
   optional MediaType type = 2;
 
   // required - The whole packet including both payload and header.
   optional bytes packet_data = 3;
 }
@@ skipping 42 matching lines @@
   // Map from video RTP payload type -> RTX config.
   repeated RtxMap rtx_map = 5;
 
   // RTP header extensions used for the received stream.
   repeated RtpHeaderExtension header_extensions = 6;
 
   // List of decoders associated with the stream.
   repeated DecoderConfig decoders = 7;
 }
 
-
 // Maps decoder names to payload types.
 message DecoderConfig {
   // required
   optional string name = 1;
 
   // required
   optional int32 payload_type = 2;
 }
 
-
 // Maps RTP header extension names to numerical IDs.
 message RtpHeaderExtension {
   // required
   optional string name = 1;
 
   // required
   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 int32 rtx_payload_type = 2;
 }
 
-
 message RtxMap {
   // required
   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 int32 rtx_payload_type = 4;
 
   // required - Encoder associated with the stream.
   optional EncoderConfig encoder = 5;
 }
 
-
 // Maps encoder names to payload types.
 message EncoderConfig {
   // required
   optional string name = 1;
 
   // required
   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;
 }