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Unified Diff: webrtc/video/rtc_event_log.proto

Issue 1230973005: Adds logging of configuration information for VideoReceiveStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments from stefan. Created 5 years, 5 months ago
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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.
+}
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