| Index: webrtc/video/rtc_event_log.proto
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| diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/video/rtc_event_log.proto
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..671196f28bdf819126148f57f14025b6efee4f9e
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| --- /dev/null
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| +++ b/webrtc/video/rtc_event_log.proto
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| @@ -0,0 +1,251 @@
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| +syntax = "proto2";
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| +option optimize_for = LITE_RUNTIME;
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| +package webrtc;
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| +
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| +// This is the main message to dump to a file, it can contain multiple event
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| +// messages, but it is possible to append multiple EventStreams (each with a
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| +// single event) to a file.
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| +// This has the benefit that there's no need to keep all data in memory.
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| +message RelEventStream {
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| +  repeated RelEvent stream = 1;
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| +}
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| +
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| +
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| +message RelEvent {
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| +  // required - Elapsed wallclock time in us since the start of the log.
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| +  optional int64 timestamp_us = 1;
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| +
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| +  // The different types of events that can occur, the UNKNOWN_EVENT entry
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| +  // is added in case future EventTypes are added, in that case old code will
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| +  // receive the new events as UNKNOWN_EVENT.
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| +  enum EventType {
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| +    UNKNOWN_EVENT = 0;
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| +    RTP_EVENT = 1;
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| +    RTCP_EVENT = 2;
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| +    DEBUG_EVENT = 3;
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| +    RECEIVER_CONFIG_EVENT = 4;
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| +    SENDER_CONFIG_EVENT = 5;
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| +    AUDIO_RECEIVER_CONFIG_EVENT = 6;
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| +    AUDIO_SENDER_CONFIG_EVENT = 7;
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| +  }
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| +
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| +  // required - Indicates the type of this event
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| +  optional EventType type = 2;
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| +
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| +  // optional - but required if type == RTP_EVENT
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| +  optional RelRtpPacket rtp_packet = 3;
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| +
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| +  // optional - but required if type == RTCP_EVENT
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| +  optional RelRtcpPacket rtcp_packet = 4;
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| +
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| +  // optional - but required if type == DEBUG_EVENT
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| +  optional RelDebugEvent debug_event = 5;
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| +
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| +  // optional - but required if type == RECEIVER_CONFIG_EVENT
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| +  optional RelVideoReceiveConfig receiver_config = 6;
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| +
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| +  // optional - but required if type == SENDER_CONFIG_EVENT
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| +  optional RelVideoSendConfig sender_config = 7;
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| +
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| +  // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
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| +  optional RelAudioReceiveConfig audio_receiver_config = 8;
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| +
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| +  // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
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| +  optional RelAudioSendConfig audio_sender_config = 9;
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| +}
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| +
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| +
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| +message RelRtpPacket {
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| +  // Indicates if the packet is incoming or outgoing with respect to the user
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| +  // that is logging the data.
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| +  enum Direction {
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| +    UNKNOWN_DIRECTION = 0;
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| +    OUTGOING = 1;
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| +    INCOMING = 2;
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| +  }
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| +  enum PayloadType {
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| +    UNKNOWN_TYPE = 0;
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| +    AUDIO = 1;
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| +    VIDEO = 2;
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| +    RTX = 3;  // TODO(terelius): Where does this get set?
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| +  }
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| +
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| +  // required
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| +  optional Direction direction = 1;
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| +
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| +  // required
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| +  optional PayloadType type = 2;
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| +
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| +  // required - The size of the packet including both payload and header.
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| +  optional uint32 packet_length = 3;
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| +
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| +  // required - The RTP header only.
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| +  optional bytes header = 4;
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| +
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| +  // Logging payloads for user data requires privacy review. Don't uncomment.
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| +  // optional bytes payload = 5;
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| +}
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| +
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| +
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| +message RelRtcpPacket {
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| +  // Indicates if the packet is incoming or outgoing with respect to the user
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| +  // that is logging the data.
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| +  enum Direction {
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| +    UNKNOWN_DIRECTION = 0;
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| +    OUTGOING = 1;
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| +    INCOMING = 2;
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| +  }
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| +  enum PayloadType {
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| +    UNKNOWN_TYPE = 0;
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| +    AUDIO = 1;
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| +    VIDEO = 2;
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| +  }
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| +
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| +  // required
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| +  optional Direction direction = 1;
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| +
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| +  // required
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| +  optional PayloadType type = 2;
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| +
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| +  // required - The whole packet including both payload and header.
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| +  optional bytes data = 3;
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| +}
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| +
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| +
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| +message RelDebugEvent {
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| +  // Indicates the type of the debug event.
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| +  // LOG_START and LOG_END indicate the start and end of the log respectively.
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| +  // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM.
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| +  enum EventType {
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| +    UNKNOWN_EVENT = 0;
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| +    LOG_START = 1;
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| +    LOG_END = 2;
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| +    AUDIO_PLAYOUT = 3;
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| +  }
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| +
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| +  // required
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| +  optional EventType type = 1;
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| +
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| +  // An optional message that can be used to store additional information about
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| +  // the debug event.
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| +  optional string message = 2;
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| +}
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| +
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| +
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| +// TODO(terelius): Video and audio streams could in principle share SSRC,
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| +// so identifying a stream based only on SSRC might not work.
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| +// It might be better to use a combination of SSRC and media type
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| +// or SSRC and port number, but for now we will rely on SSRC only.
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| +message RelVideoReceiveConfig {
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| +  // required - Synchronization source (stream identifier) to be received.
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| +  optional uint32 remote_ssrc = 1;
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| +  // required - Sender SSRC used for sending RTCP (such as receiver reports).
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| +  optional uint32 local_ssrc = 2;
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| +
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| +  // Compound mode is described by RFC 4585 and reduced-size
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| +  // RTCP mode is described by RFC 5506.
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| +  enum RtcpMode {
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| +    RTCP_COMPOUND = 1;
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| +    RTCP_REDUCEDSIZE = 2;
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| +  }
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| +  // required - RTCP mode to use.
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| +  optional RtcpMode rtcp_mode = 3;
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| +
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| +  // required - Extended RTCP settings.
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| +  optional bool receiver_reference_time_report = 4;
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| +
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| +  // required - Receiver estimated maximum bandwidth.
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| +  optional bool remb = 5;
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| +
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| +  // Map from video RTP payload type -> RTX config.
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| +  repeated RtxMap rtx_map = 6;
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| +
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| +  // RTP header extensions used for the received stream.
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| +  repeated RtpHeaderExtension header_extensions = 7;
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| +
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| +  // List of decoders associated with the stream.
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| +  repeated DecoderConfig decoders = 8;
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| +}
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| +
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| +
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| +// Maps decoder names to payload types.
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| +message DecoderConfig {
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| +  // required
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| +  optional string name = 1;
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| +
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| +  // required
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| +  optional sint32 payload_type = 2;
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| +}
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| +
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| +
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| +// Maps RTP header extension names to numerical IDs.
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| +message RtpHeaderExtension {
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| +  // required
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| +  optional string name = 1;
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| +
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| +  // required
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| +  optional sint32 id = 2;
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| +}
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| +
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| +
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| +// RTX settings for incoming video payloads that may be received.
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| +// RTX is disabled if there's no config present.
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| +message RtxConfig {
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| +  // required - SSRC to use for the RTX stream.
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| +  optional uint32 rtx_ssrc = 1;
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| +
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| +  // required - Payload type to use for the RTX stream.
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| +  optional sint32 rtx_payload_type = 2;
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| +}
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| +
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| +
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| +message RtxMap {
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| +  // required
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| +  optional sint32 payload_type = 1;
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| +
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| +  // required
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| +  optional RtxConfig config = 2;
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| +}
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| +
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| +
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| +message RelVideoSendConfig {
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| +  // Synchronization source (stream identifier) for outgoing stream.
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| +  // One stream can have several ssrcs for e.g. simulcast.
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| +  // At least one ssrc is required.
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| +  repeated uint32 ssrcs = 1;
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| +
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| +  // RTP header extensions used for the outgoing stream.
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| +  repeated RtpHeaderExtension header_extensions = 2;
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| +
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| +  // List of SSRCs for retransmitted packets.
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| +  repeated uint32 rtx_ssrcs = 3;
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| +
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| +  // required if rtx_ssrcs is used - Payload type for retransmitted packets.
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| +  optional sint32 rtx_payload_type = 4;
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| +
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| +  // required - Canonical end-point identifier.
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| +  optional string c_name = 5;
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| +
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| +  // required - Encoder associated with the stream.
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| +  optional EncoderConfig encoder = 6;
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| +}
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| +
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| +
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| +// Maps encoder names to payload types.
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| +message EncoderConfig {
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| +  // required
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| +  optional string name = 1;
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| +
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| +  // required
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| +  optional sint32 payload_type = 2;
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| +}
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| +
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| +
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| +message RelAudioReceiveConfig {
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| +  // TODO(terelius): Figure out what the requirements are from the audio team.
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| +}
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| +
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| +
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| +message RelAudioSendConfig {
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| +  // TODO(terelius): Figure out what the requirements are from the audio team.
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| +}
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| 
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