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Unified Diff: webrtc/modules/audio_coding/main/acm2/dump.proto

Issue 1202833003: Added fields for configuration information to the protobuf format (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Minor fix of whitespace Created 5 years, 6 months ago
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Index: webrtc/modules/audio_coding/main/acm2/dump.proto
diff --git a/webrtc/modules/audio_coding/main/acm2/dump.proto b/webrtc/modules/audio_coding/main/acm2/dump.proto
index 416bb7a61b6e08beaae960023dac1260216aa0bf..232faec42871cfc20ef55ffc6f6e6b48c338eec0 100644
--- a/webrtc/modules/audio_coding/main/acm2/dump.proto
+++ b/webrtc/modules/audio_coding/main/acm2/dump.proto
@@ -10,6 +10,7 @@ message ACMDumpEventStream {
repeated ACMDumpEvent stream = 1;
}
+
message ACMDumpEvent {
// required - Elapsed wallclock time in us since the start of the log.
optional int64 timestamp_us = 1;
@@ -21,6 +22,7 @@ message ACMDumpEvent {
UNKNOWN_EVENT = 0;
RTP_EVENT = 1;
DEBUG_EVENT = 2;
+ CONFIG_EVENT = 3;
}
// required - Indicates the type of this event
@@ -31,8 +33,12 @@ message ACMDumpEvent {
// optional - but required if type == DEBUG_EVENT
optional ACMDumpDebugEvent debug_event = 4;
+
+ // optional - but required if type == CONFIG_EVENT
+ optional ACMDumpConfigEvent config = 5;
}
+
message ACMDumpRTPPacket {
// Indicates if the packet is incoming or outgoing with respect to the user
// that is logging the data.
@@ -58,6 +64,7 @@ message ACMDumpRTPPacket {
optional bytes RTP_data = 3;
}
+
message ACMDumpDebugEvent {
// Indicates the type of the debug event.
// LOG_START and LOG_END indicate the start and end of the log respectively.
@@ -75,4 +82,88 @@ message ACMDumpDebugEvent {
// 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.
+message ACMDumpConfigEvent {
+ // Synchronization source (stream identifier) to be received.
+ optional uint32 remote_ssrc = 1;
+
+ // RTX settings for incoming video payloads that may be received. RTX is
+ // disabled if there's no config present.
+ optional RtcpConfig rtcp_config = 3;
+
+ // Map from video RTP payload type -> RTX config.
+ repeated RtxMap rtx_map = 4;
+
+ // RTP header extensions used for the received stream.
+ repeated RtpHeaderExtension header_extensions = 5;
+
+ // List of decoders associated with the stream.
+ repeated DecoderConfig decoders = 6;
+}
+
+
+// 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 - SSRCs to use for the RTX streams.
+ optional uint32 ssrc = 1;
+
+ // required - Payload type to use for the RTX stream.
+ optional sint32 payload_type = 2;
+}
+
+
+message RtxMap {
+ // required
+ optional sint32 payload_type = 1;
+
+ // required
+ optional RtxConfig config = 2;
+}
+
+
+// Configuration information for RTCP.
+// For bandwidth estimation purposes it is more interesting to log the
+// RTCP messages that the sender receives, but we will support logging
+// at the receiver side too.
+message RtcpConfig {
+ // Sender SSRC used for sending RTCP (such as receiver reports).
+ optional uint32 local_ssrc = 1;
+
+ // RTCP mode to use. 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;}
+ optional RtcpMode rtcp_mode = 2;
+
+ // Extended RTCP settings.
+ optional bool receiver_reference_time_report = 3;
+
+ // Receiver estimated maximum bandwidth.
+ optional bool remb = 4;
+}
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