webrtc/modules/audio_coding/main/acm2/dump.proto

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

// 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 ACMDumpEventStream {
  repeated ACMDumpEvent stream = 1;
}


message ACMDumpEvent {
  // 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;
    RTP_EVENT = 1;
    DEBUG_EVENT = 2;
    CONFIG_EVENT = 3;
  }

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

  // optional - but required if type == RTP_EVENT
  optional ACMDumpRTPPacket packet = 3;

  // optional - but required if type == DEBUG_EVENT
  optional ACMDumpDebugEvent debug_event = 4;

  // optional - but required if type == CONFIG_EVENT
  optional ACMDumpConfigEvent config = 5;
}


message ACMDumpRTPPacket {
  // Indicates if the packet is incoming or outgoing with respect to the user
  // that is logging the data.
  enum Direction {
    UNKNOWN_DIRECTION = 0;
    OUTGOING = 1;
    INCOMING = 2;
  }
  enum PayloadType {
    UNKNOWN_TYPE = 0;
    AUDIO = 1;
    VIDEO = 2;
    RTX = 3;
  }

  // required
  optional Direction direction = 1;

  // required
  optional PayloadType type = 2;

  // required - Contains the whole RTP packet (header+payload).
  optional bytes RTP_data = 3;
}


message ACMDumpDebugEvent {
  // Indicates the type of the debug event.
  // LOG_START and LOG_END indicate the start and end of the log respectively.
  // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM.
  enum EventType {
    UNKNOWN_EVENT = 0;
    LOG_START = 1;
    LOG_END = 2;
    AUDIO_PLAYOUT = 3;
  }

  // required
  optional EventType type = 1;

  // An optional message that can be used to store additional information about
  // the debug event.
  optional string message = 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 ACMDumpConfigEvent {
  // Synchronization source (stream identifier) to be received.
  optional uint32 remote_ssrc = 1;

  // RTX settings for incoming video payloads that may be received. RTX is
  // disabled if there's no config present.
  optional RtcpConfig rtcp_config = 3;

  // Map from video RTP payload type -> RTX config.
  repeated RtxMap rtx_map = 4;

  // RTP header extensions used for the received stream.
  repeated RtpHeaderExtension header_extensions = 5;

  // List of decoders associated with the stream.
  repeated DecoderConfig decoders = 6;
}


// Maps decoder names to payload types.
message DecoderConfig {
  // required
  optional string name = 1;

  // required
  optional sint32 payload_type = 2;
}


// Maps RTP header extension names to numerical ids.
message RtpHeaderExtension {
  // required
  optional string name = 1;

  // required
  optional sint32 id = 2;
}


// RTX settings for incoming video payloads that may be received.
// RTX is disabled if there's no config present.
message RtxConfig {
  // required - SSRCs to use for the RTX streams.
  optional uint32 ssrc = 1;

  // required - Payload type to use for the RTX stream.
  optional sint32 payload_type = 2;
}


message RtxMap {
  // required
  optional sint32 payload_type = 1;

  // required
  optional RtxConfig config = 2;
}


// Configuration information for RTCP.
// For bandwidth estimation purposes it is more interesting to log the
// RTCP messages that the sender receives, but we will support logging
// at the receiver side too.
message RtcpConfig {
  // Sender SSRC used for sending RTCP (such as receiver reports).
  optional uint32 local_ssrc = 1;

  // RTCP mode to use. 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;}
  optional RtcpMode rtcp_mode = 2;

  // Extended RTCP settings.
  optional bool receiver_reference_time_report = 3;

  // Receiver estimated maximum bandwidth.
  optional bool remb = 4;
}