| Index: webrtc/video/rtc_event_log.proto
|
| diff --git a/webrtc/video/rtc_event_log.proto b/webrtc/video/rtc_event_log.proto
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..7e4e699e338c1bcbaf44e397942d600d2dbd11a4
|
| --- /dev/null
|
| +++ b/webrtc/video/rtc_event_log.proto
|
| @@ -0,0 +1,228 @@
|
| +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;
|
| + RTP_EVENT = 1;
|
| + RTCP_EVENT = 2;
|
| + DEBUG_EVENT = 3;
|
| + VIDEO_RECEIVER_CONFIG_EVENT = 4;
|
| + VIDEO_SENDER_CONFIG_EVENT = 5;
|
| + AUDIO_RECEIVER_CONFIG_EVENT = 6;
|
| + AUDIO_SENDER_CONFIG_EVENT = 7;
|
| + }
|
| +
|
| + // 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 == DEBUG_EVENT
|
| + optional DebugEvent debug_event = 5;
|
| +
|
| + // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
|
| + optional VideoReceiveConfig video_receiver_config = 6;
|
| +
|
| + // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
|
| + optional VideoSendConfig video_sender_config = 7;
|
| +
|
| + // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
|
| + optional AudioReceiveConfig audio_receiver_config = 8;
|
| +
|
| + // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
|
| + optional AudioSendConfig audio_sender_config = 9;
|
| +}
|
| +
|
| +
|
| +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 DebugEvent {
|
| + // 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;
|
| +}
|
| +
|
| +
|
| +// 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 - Extended RTCP settings.
|
| + optional bool receiver_reference_time_report = 4;
|
| +
|
| + // required - Receiver estimated maximum bandwidth.
|
| + optional bool remb = 5;
|
| +
|
| + // Map from video RTP payload type -> RTX config.
|
| + repeated RtxMap rtx_map = 6;
|
| +
|
| + // RTP header extensions used for the received stream.
|
| + repeated RtpHeaderExtension header_extensions = 7;
|
| +
|
| + // List of decoders associated with the stream.
|
| + repeated DecoderConfig decoders = 8;
|
| +}
|
| +
|
| +
|
| +// 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 - SSRC to use for the RTX stream.
|
| + optional uint32 rtx_ssrc = 1;
|
| +
|
| + // required - Payload type to use for the RTX stream.
|
| + optional sint32 rtx_payload_type = 2;
|
| +}
|
| +
|
| +
|
| +message RtxMap {
|
| + // required
|
| + optional sint32 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 sint32 rtx_payload_type = 4;
|
| +
|
| + // required - Canonical end-point identifier.
|
| + optional string c_name = 5;
|
| +
|
| + // required - Encoder associated with the stream.
|
| + optional EncoderConfig encoder = 6;
|
| +}
|
| +
|
| +
|
| +// Maps encoder names to payload types.
|
| +message EncoderConfig {
|
| + // required
|
| + optional string name = 1;
|
| +
|
| + // required
|
| + optional sint32 payload_type = 2;
|
| +}
|
| +
|
| +
|
| +message AudioReceiveConfig {
|
| + // TODO(terelius): Add audio-receive config.
|
| +}
|
| +
|
| +
|
| +message AudioSendConfig {
|
| + // TODO(terelius): Add audio-receive config.
|
| +}
|
|
|