blob: b14306e3628d3982079fcfb906af93a5d34e7507 [file] [log] [blame]
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;
}