| 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
|
| deleted file mode 100644
|
| index 232faec42871cfc20ef55ffc6f6e6b48c338eec0..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_coding/main/acm2/dump.proto
|
| +++ /dev/null
|
| @@ -1,169 +0,0 @@
|
| -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 ACMDumpEventStream {
|
| - repeated ACMDumpEvent stream = 1;
|
| -}
|
| -
|
| -
|
| -message ACMDumpEvent {
|
| - // 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;
|
| - DEBUG_EVENT = 2;
|
| - CONFIG_EVENT = 3;
|
| - }
|
| -
|
| - // required - Indicates the type of this event
|
| - optional EventType type = 2;
|
| -
|
| - // optional - but required if type == RTP_EVENT
|
| - optional ACMDumpRTPPacket packet = 3;
|
| -
|
| - // 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.
|
| - enum Direction {
|
| - UNKNOWN_DIRECTION = 0;
|
| - OUTGOING = 1;
|
| - INCOMING = 2;
|
| - }
|
| - enum PayloadType {
|
| - UNKNOWN_TYPE = 0;
|
| - AUDIO = 1;
|
| - VIDEO = 2;
|
| - RTX = 3;
|
| - }
|
| -
|
| - // required
|
| - optional Direction direction = 1;
|
| -
|
| - // required
|
| - optional PayloadType type = 2;
|
| -
|
| - // required - Contains the whole RTP packet (header+payload).
|
| - 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.
|
| - // 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.
|
| -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;
|
| -}
|
|
|