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; |
-} |