| 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;
|
| +}
|
|
|