webrtc_m130/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;
    LOG_END = 2;
    RTP_EVENT = 3;
    RTCP_EVENT = 4;
    AUDIO_PLAYOUT_EVENT = 5;
    LOSS_BASED_BWE_UPDATE = 6;
    DELAY_BASED_BWE_UPDATE = 7;
    VIDEO_RECEIVER_CONFIG_EVENT = 8;
    VIDEO_SENDER_CONFIG_EVENT = 9;
    AUDIO_RECEIVER_CONFIG_EVENT = 10;
    AUDIO_SENDER_CONFIG_EVENT = 11;
    AUDIO_NETWORK_ADAPTATION_EVENT = 16;
    BWE_PROBE_CLUSTER_CREATED_EVENT = 17;
    BWE_PROBE_RESULT_EVENT = 18;
    ALR_STATE_EVENT = 19;
  }

  // required - Indicates the type of this event
  optional EventType type = 2;

  oneof subtype {
    // required if type == RTP_EVENT
    RtpPacket rtp_packet = 3;

    // required if type == RTCP_EVENT
    RtcpPacket rtcp_packet = 4;

    // required if type == AUDIO_PLAYOUT_EVENT
    AudioPlayoutEvent audio_playout_event = 5;

    // required if type == LOSS_BASED_BWE_UPDATE
    LossBasedBweUpdate loss_based_bwe_update = 6;

    // required if type == DELAY_BASED_BWE_UPDATE
    DelayBasedBweUpdate delay_based_bwe_update = 7;

    // required if type == VIDEO_RECEIVER_CONFIG_EVENT
    VideoReceiveConfig video_receiver_config = 8;

    // required if type == VIDEO_SENDER_CONFIG_EVENT
    VideoSendConfig video_sender_config = 9;

    // required if type == AUDIO_RECEIVER_CONFIG_EVENT
    AudioReceiveConfig audio_receiver_config = 10;

    // required if type == AUDIO_SENDER_CONFIG_EVENT
    AudioSendConfig audio_sender_config = 11;

    // required if type == AUDIO_NETWORK_ADAPTATION_EVENT
    AudioNetworkAdaptation audio_network_adaptation = 16;

    // required if type == BWE_PROBE_CLUSTER_CREATED_EVENT
    BweProbeCluster probe_cluster = 17;

    // required if type == BWE_PROBE_RESULT_EVENT
    BweProbeResult probe_result = 18;

    // required if type == ALR_STATE_EVENT
    AlrState alr_state = 19;
  }
}

message RtpPacket {
  // required - True if the packet is incoming w.r.t. the user logging the data
  optional bool incoming = 1;

  optional MediaType type = 2 [deprecated = true];

  // 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;

  // optional - The probe cluster id.
  optional uint32 probe_cluster_id = 5;

  // 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;

  optional MediaType type = 2 [deprecated = true];

  // required - The whole packet including both payload and header.
  optional bytes packet_data = 3;
}

message AudioPlayoutEvent {
  // TODO(ivoc): Rename, we currently use the "remote" ssrc, i.e. identifying
  // the receive stream, while local_ssrc identifies the send stream, if any.
  // required - The SSRC of the audio stream associated with the playout event.
  optional uint32 local_ssrc = 2;
}

message LossBasedBweUpdate {
  // required - Bandwidth estimate (in bps) after the update.
  optional int32 bitrate_bps = 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;
}

message DelayBasedBweUpdate {
  enum DetectorState {
    BWE_NORMAL = 0;
    BWE_UNDERUSING = 1;
    BWE_OVERUSING = 2;
  }

  // required - Bandwidth estimate (in bps) after the update.
  optional int32 bitrate_bps = 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;

  // 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;
}

message AudioNetworkAdaptation {
  // Bit rate that the audio encoder is operating at.
  optional int32 bitrate_bps = 1;

  // Frame length that each encoded audio packet consists of.
  optional int32 frame_length_ms = 2;

  // Packet loss fraction that the encoder's forward error correction (FEC) is
  // optimized for.
  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;
}

message BweProbeCluster {
  // required - The id of this probe cluster.
  optional uint32 id = 1;

  // required - The bitrate in bps that this probe cluster is meant to probe.
  optional uint64 bitrate_bps = 2;

  // required - The minimum number of packets used to probe the given bitrate.
  optional uint32 min_packets = 3;

  // required - The minimum number of bytes used to probe the given bitrate.
  optional uint32 min_bytes = 4;
}

message BweProbeResult {
  // required - The id of this probe cluster.
  optional uint32 id = 1;

  enum ResultType {
    SUCCESS = 0;
    INVALID_SEND_RECEIVE_INTERVAL = 1;
    INVALID_SEND_RECEIVE_RATIO = 2;
    TIMEOUT = 3;
  }

  // required - The result of this probing attempt.
  optional ResultType result = 2;

  // optional - but required if result == SUCCESS. The resulting bitrate in bps.
  optional uint64 bitrate_bps = 3;
}

message AlrState {
  // required - If we are in ALR or not.
  optional bool in_alr = 1;
}