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. |
+} |