// THIS FILE IS EXPERIMENTAL. BREAKING CHANGES MAY BE MADE AT ANY TIME
// WITHOUT PRIOR WARNING. THIS FILE SHOULD NOT BE USED IN PRODUCTION CODE.

syntax = "proto2";
option optimize_for = LITE_RUNTIME;
package webrtc.rtclog2;

// At the top level, a WebRTC event log is just an EventStream object. Note that
// concatenating multiple EventStreams in the same file is equivalent to a
// single EventStream object containing the same events. Hence, it is not
// necessary to wait for the entire log to be complete before beginning to
// write it to a file.
message EventStream {
  // Deprecated - Maintained for compatibility with the old event log.
  // TODO(terelius): Maybe we can remove this and instead check the stream for
  // presence of a version field. That requires a custom protobuf parser, but we
  // have that already anyway.
  repeated Event stream = 1 [deprecated = true];
  // required - The version number must be 2 for this version of the event log.
  optional uint32 version = 2;
  repeated IncomingRtpPackets incoming_rtp_packets = 3;
  repeated OutgoingRtpPackets outgoing_rtp_packets = 4;
  repeated IncomingRtcpPackets incoming_rtcp_packets = 5;
  repeated OutgoingRtcpPackets outgoing_rtcp_packets = 6;
  repeated AudioPlayoutEvents audio_playout_events = 7;
  // The field tags 8-15 are reserved for the most common events
  repeated BeginLogEvent begin_log_events = 16;
  repeated EndLogEvent end_log_events = 17;
  repeated LossBasedBweUpdates loss_based_bwe_updates = 18;
  repeated DelayBasedBweUpdates delay_based_bwe_updates = 19;
  repeated AudioNetworkAdaptations audio_network_adaptations = 20;
  repeated BweProbeCluster probe_clusters = 21;
  repeated BweProbeResultSuccess probe_success = 22;
  repeated BweProbeResultFailure probe_failure = 23;

  repeated AudioRecvStreamConfig audio_recv_stream_configs = 101;
  repeated AudioSendStreamConfig audio_send_stream_configs = 102;
  repeated VideoRecvStreamConfig video_recv_stream_configs = 103;
  repeated VideoSendStreamConfig video_send_stream_configs = 104;
}

// DEPRECATED.
message Event {
  // TODO(terelius): Do we want to preserve the old Event definition here?
}

message IncomingRtpPackets {
  optional int64 timestamp_ms = 1;

  // RTP marker bit, used to label boundaries within e.g. video frames.
  optional bool marker = 2;

  // RTP payload type.
  optional uint32 payload_type = 3;

  // RTP sequence number.
  optional uint32 sequence_number = 4;

  // RTP monotonic clock timestamp (not actual time).
  optional fixed32 rtp_timestamp = 5;

  // Synchronization source of this packet's RTP stream.
  optional fixed32 ssrc = 6;

  // TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.

  // required - The size of the packet including both payload and header.
  optional uint32 packet_size = 8;

  // Optional header extensions.
  optional int32 transmission_time_offset = 9;
  optional uint32 absolute_send_time = 10;
  optional uint32 transport_sequence_number = 11;
  optional uint32 audio_level = 12;
  // TODO(terelius): Add header extensions like video rotation, playout delay?

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes marker_deltas = 102;
  optional bytes payload_type_deltas = 103;
  optional bytes sequence_number_deltas = 104;
  optional bytes rtp_timestamp_deltas = 105;
  optional bytes ssrc_deltas = 106;
  optional bytes packet_size_deltas = 107;
  optional bytes transmission_time_offset_deltas = 108;
  optional bytes absolute_send_time_deltas = 109;
  optional bytes transport_sequence_number_deltas = 110;
  optional bytes audio_level_deltas = 111;
}

message OutgoingRtpPackets {
  optional int64 timestamp_ms = 1;

  // RTP marker bit, used to label boundaries within e.g. video frames.
  optional bool marker = 2;

  optional uint32 payload_type = 3;

  // RTP sequence number.
  optional uint32 sequence_number = 4;

  // RTP monotonic clock timestamp (not actual time).
  optional fixed32 rtp_timestamp = 5;

  // Synchronization source of this packet's RTP stream.
  optional fixed32 ssrc = 6;

  // TODO(terelius/dinor): Add CSRCs. Field number 7 reserved for this purpose.

  // required - The size of the packet including both payload and header.
  optional uint32 packet_size = 8;

  // Optional header extensions.
  optional int32 transmission_time_offset = 9;
  optional uint32 absolute_send_time = 10;
  optional uint32 transport_sequence_number = 11;
  optional uint32 audio_level = 12;
  // TODO(terelius): Add header extensions like video rotation, playout delay?

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes marker_deltas = 102;
  optional bytes payload_type_deltas = 103;
  optional bytes sequence_number_deltas = 104;
  optional bytes rtp_timestamp_deltas = 105;
  optional bytes ssrc_deltas = 106;
  optional bytes packet_size_deltas = 107;
  optional bytes probe_cluster_id_deltas = 108;
  optional bytes transmission_time_offset_deltas = 109;
  optional bytes absolute_send_time_deltas = 110;
  optional bytes transport_sequence_number_deltas = 111;
}

message IncomingRtcpPackets {
  optional int64 timestamp_ms = 1;

  // required - The whole packet including both payload and header.
  optional bytes raw_packet = 2;
  // TODO(terelius): Feasible to log parsed RTCP instead?

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes raw_packet_deltas = 102;
}

message OutgoingRtcpPackets {
  optional int64 timestamp_ms = 1;

  // required - The whole packet including both payload and header.
  optional bytes raw_packet = 2;
  // TODO(terelius): Feasible to log parsed RTCP instead?

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes raw_packet_deltas = 102;
}

message AudioPlayoutEvents {
  optional int64 timestamp_ms = 1;

  // required - The SSRC of the audio stream associated with the playout event.
  optional uint32 local_ssrc = 2;

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes local_ssrc_deltas = 102;
}

message BeginLogEvent {
  optional int64 timestamp_ms = 1;
}

message EndLogEvent {
  optional int64 timestamp_ms = 1;
}

message LossBasedBweUpdates {
  optional int64 timestamp_ms = 1;

  // TODO(terelius): Update log interface to unsigned.
  // required - Bandwidth estimate (in bps) after the update.
  optional uint32 bitrate_bps = 2;

  // 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 = 3;

  // TODO(terelius): Is this really needed? Remove or make optional?
  // TODO(terelius): Update log interface to unsigned.
  // required - Total number of packets that the BWE update is based on.
  optional uint32 total_packets = 4;

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes bitrate_deltas_bps = 102;
  optional bytes fraction_loss_deltas = 103;
  optional bytes total_packets_deltas = 104;
}

message DelayBasedBweUpdates {
  optional int64 timestamp_ms = 1;

  // required - Bandwidth estimate (in bps) after the update.
  optional uint32 bitrate_bps = 2;

  enum DetectorState {
    BWE_NORMAL = 0;
    BWE_UNDERUSING = 1;
    BWE_OVERUSING = 2;
  }
  optional DetectorState detector_state = 3;

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes bitrate_deltas_bps = 102;
  optional bytes detector_state_deltas = 103;
}

// Maps RTP header extension names to numerical IDs.
message RtpHeaderExtensionConfig {
  // Optional IDs for the header extensions. Each ID is a 4-bit number that is
  // only set if that extension is configured.
  // TODO(terelius): Can we skip transmission_time_offset? When is it used?
  optional int32 transmission_time_offset_id = 1;
  optional int32 absolute_send_time_id = 2;
  optional int32 transport_sequence_number_id = 3;
  optional int32 audio_level_id = 4;
  // TODO(terelius): Add video_rotation and playout delay?
}

message VideoRecvStreamConfig {
  optional int64 timestamp_ms = 1;

  // required - Synchronization source (stream identifier) to be received.
  optional uint32 remote_ssrc = 2;

  // required - Sender SSRC used for sending RTCP (such as receiver reports).
  optional uint32 local_ssrc = 3;

  // required if RTX is configured
  optional uint32 rtx_ssrc = 4;

  // optional - RTP source stream ID
  optional bytes rsid = 5;

  // IDs for the header extension we care about. Only required if there are
  // header extensions configured.
  optional RtpHeaderExtensionConfig header_extensions = 6;

  // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
  // used, we also need a map between RTP payload type and RTX payload type.
}

message VideoSendStreamConfig {
  optional int64 timestamp_ms = 1;

  // Synchronization source (stream identifier) for outgoing stream.
  // One stream can have several ssrcs for e.g. simulcast.
  optional uint32 ssrc = 2;

  // SSRC for the RTX stream
  optional uint32 rtx_ssrc = 3;

  // RTP source stream ID
  optional bytes rsid = 4;

  // IDs for the header extension we care about. Only required if there are
  // header extensions configured.
  optional RtpHeaderExtensionConfig header_extensions = 5;

  // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
  // used, we also need a map between RTP payload type and RTX payload type.
}

message AudioRecvStreamConfig {
  optional int64 timestamp_ms = 1;

  // required - Synchronization source (stream identifier) to be received.
  optional uint32 remote_ssrc = 2;

  // required - Sender SSRC used for sending RTCP (such as receiver reports).
  optional uint32 local_ssrc = 3;

  // Field number 4 reserved for RTX SSRC.

  // optional - RTP source stream ID
  optional bytes rsid = 5;

  // IDs for the header extension we care about. Only required if there are
  // header extensions configured.
  optional RtpHeaderExtensionConfig header_extensions = 6;

  // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
  // used, we also need a map between RTP payload type and RTX payload type.
}

message AudioSendStreamConfig {
  optional int64 timestamp_ms = 1;

  // Synchronization source (stream identifier) for outgoing stream.
  // One stream can have several ssrcs for e.g. simulcast.
  optional uint32 ssrc = 2;

  // Field number 3 reserved for RTX SSRC

  // RTP source stream ID
  optional bytes rsid = 4;

  // IDs for the header extension we care about. Only required if there are
  // header extensions configured.
  optional RtpHeaderExtensionConfig header_extensions = 5;

  // TODO(terelius): Do we need codec-payload mapping? If so and rtx_ssrc is
  // used, we also need a map between RTP payload type and RTX payload type.
}

message AudioNetworkAdaptations {
  optional int64 timestamp_ms = 1;

  // Bit rate that the audio encoder is operating at.
  // TODO(terelius): Signed vs unsigned?
  optional int32 bitrate_bps = 2;

  // Frame length that each encoded audio packet consists of.
  // TODO(terelius): Signed vs unsigned?
  optional int32 frame_length_ms = 3;

  // Packet loss fraction that the encoder's forward error correction (FEC) is
  // optimized for.
  optional float uplink_packet_loss_fraction = 4;

  // Whether forward error correction (FEC) is turned on or off.
  optional bool enable_fec = 5;

  // Whether discontinuous transmission (DTX) is turned on or off.
  optional bool enable_dtx = 6;

  // Number of audio channels that each encoded packet consists of.
  optional uint32 num_channels = 7;

  // Delta encodings
  optional bytes timestamp_deltas_ms = 101;
  optional bytes bitrate_deltas_bps = 102;
  optional bytes frame_length_deltas_ms = 103;
  optional bytes uplink_packet_loss_fraction_deltas = 104;
  optional bytes enable_fec_deltas = 105;
  optional bytes enable_dtx_deltas = 106;
  optional bytes num_channels_deltas = 107;
}

message BweProbeCluster {
  optional int64 timestamp_ms = 1;

  // required - The id of this probe cluster.
  optional uint32 id = 2;

  // required - The bitrate in bps that this probe cluster is meant to probe.
  optional uint32 bitrate_bps = 3;

  // required - The minimum number of packets used to probe the given bitrate.
  optional uint32 min_packets = 4;

  // required - The minimum number of bytes used to probe the given bitrate.
  optional uint32 min_bytes = 5;
}

message BweProbeResultSuccess {
  optional int64 timestamp_ms = 1;

  // required - The id of this probe cluster.
  optional uint32 id = 2;

  // required - The resulting bitrate in bps.
  optional uint32 bitrate_bps = 3;
}

message BweProbeResultFailure {
  optional int64 timestamp_ms = 1;

  // required - The id of this probe cluster.
  optional uint32 id = 2;

  enum FailureReason {
    UNKNOWN = 0;
    INVALID_SEND_RECEIVE_INTERVAL = 1;
    INVALID_SEND_RECEIVE_RATIO = 2;
    TIMEOUT = 3;
  }

  // required
  optional FailureReason failure = 3;
}