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

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;
-}
-
-
-// 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;
-    LOG_END = 2;
-    RTP_EVENT = 3;
-    RTCP_EVENT = 4;
-    AUDIO_PLAYOUT_EVENT = 5;
-    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 = 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
-  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;
-}
-
-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;
-
-  // Compound mode is described by RFC 4585 and reduced-size
-  // RTCP mode is described by RFC 5506.
-  enum RtcpMode {
-    RTCP_COMPOUND = 1;
-    RTCP_REDUCEDSIZE = 2;
-  }
-  // required - RTCP mode to use.
-  optional RtcpMode rtcp_mode = 3;
-
-  // required - Receiver estimated maximum bandwidth.
-  optional bool remb = 4;
-
-  // 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;
-}