Adds logging of configuration information for VideoReceiveStream

webrtc/video/rtc_event_log.proto

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..671196f28bdf819126148f57f14025b6efee4f9e
--- /dev/null
+++ b/webrtc/video/rtc_event_log.proto
@@ -0,0 +1,251 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc;
+
+// 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 RelEventStream {

[stefan-webrtc, 2015/07/21 12:41:13]

Is there a need to have Rel infront of all names here? I don't know protobufs
well enough, so I have to ask. It can't be avoided by namespaces?

[terelius, 2015/07/23 15:54:01]

Yes, it is possible to use namespaces. I wanted to avoid it because the names
are already very long and I prefer not to have line breaks in simple lines such
as 
SomeType a = b;

Is webrtc::rtclog an acceptable namespace?

+  repeated RelEvent stream = 1;
+}
+
+
+message RelEvent {
+  // 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;
+    RECEIVER_CONFIG_EVENT = 4;

[pbos-webrtc, 2015/07/22 11:05:27]

VIDEO_

[terelius, 2015/07/23 15:54:01]

Done.

+    SENDER_CONFIG_EVENT = 5;

[pbos-webrtc, 2015/07/22 11:05:27]

VIDEO_

[terelius, 2015/07/23 15:54:01]

Done.

+    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 RelRtpPacket rtp_packet = 3;
+
+  // optional - but required if type == RTCP_EVENT
+  optional RelRtcpPacket rtcp_packet = 4;
+
+  // optional - but required if type == DEBUG_EVENT
+  optional RelDebugEvent debug_event = 5;
+
+  // optional - but required if type == RECEIVER_CONFIG_EVENT

[pbos-webrtc, 2015/07/22 11:05:27]

VIDEO_

[terelius, 2015/07/23 15:54:01]

Done.

+  optional RelVideoReceiveConfig receiver_config = 6;
+
+  // optional - but required if type == SENDER_CONFIG_EVENT

[pbos-webrtc, 2015/07/22 11:05:27]

VIDEO_

[terelius, 2015/07/23 15:54:01]

Done.

+  optional RelVideoSendConfig sender_config = 7;
+
+  // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
+  optional RelAudioReceiveConfig audio_receiver_config = 8;
+
+  // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
+  optional RelAudioSendConfig audio_sender_config = 9;
+}
+
+
+message RelRtpPacket {
+  // Indicates if the packet is incoming or outgoing with respect to the user
+  // that is logging the data.
+  enum Direction {
+    UNKNOWN_DIRECTION = 0;
+    OUTGOING = 1;
+    INCOMING = 2;
+  }
+  enum PayloadType {

[pbos-webrtc, 2015/07/22 11:05:27]

These aren't payload types.

[terelius, 2015/07/23 15:54:01]

No they are a legacy from an earlier version. We have discussed renaming them,
but I guess I can do it now.

+    UNKNOWN_TYPE = 0;
+    AUDIO = 1;
+    VIDEO = 2;
+    RTX = 3;  // TODO(terelius): Where does this get set?
+  }
+
+  // required
+  optional Direction direction = 1;
+
+  // required
+  optional PayloadType 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;
+
+  // Logging payloads for user data requires privacy review. Don't uncomment.
+  // optional bytes payload = 5;

[pbos-webrtc, 2015/07/22 11:05:27]

How would this be turned on/off? If this is premature, remove it completely.

[terelius, 2015/07/23 15:54:01]

I'll remove the // optional bytes payload = 5; part, but I will not remove the
warning against logging payload data without privacy review.

+}
+
+
+message RelRtcpPacket {
+  // Indicates if the packet is incoming or outgoing with respect to the user
+  // that is logging the data.
+  enum Direction {

[pbos-webrtc, 2015/07/22 11:05:27]

Can this be shared?

[terelius, 2015/07/23 15:54:01]

I'll change it to a bool.

+    UNKNOWN_DIRECTION = 0;
+    OUTGOING = 1;
+    INCOMING = 2;
+  }
+  enum PayloadType {
+    UNKNOWN_TYPE = 0;
+    AUDIO = 1;
+    VIDEO = 2;
+  }
+
+  // required
+  optional Direction direction = 1;
+
+  // required
+  optional PayloadType type = 2;
+
+  // required - The whole packet including both payload and header.
+  optional bytes data = 3;
+}
+
+
+message RelDebugEvent {
+  // 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;
+
+  // An optional message that can be used to store additional information about
+  // the debug event.
+  optional string message = 2;
+}
+
+
+// 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.

[pbos-webrtc, 2015/07/22 11:05:27]

We'll probably move away from media type as well, since there can be >1
transport per media type (just not in our implementation), meaning that video +
ssrc isn't even enough to identify the stream.

[terelius, 2015/07/23 15:54:01]

Ok, but there is nothing I can do about that unless we already have some better
way to identify streams.

+message RelVideoReceiveConfig {
+  // 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 RelVideoSendConfig {
+  // 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 RelAudioReceiveConfig {
+  // TODO(terelius): Figure out what the requirements are from the audio team.

[pbos-webrtc, 2015/07/22 11:05:27]

TODO(terelius): Add audio-receive config.

[terelius, 2015/07/23 15:54:01]

Changed comment.

+}
+
+
+message RelAudioSendConfig {
+  // TODO(terelius): Figure out what the requirements are from the audio team.

[pbos-webrtc, 2015/07/22 11:05:27]

TODO(terelius): Add audio-send config.

[terelius, 2015/07/23 15:54:01]

Changed comment.

+}