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1 syntax = "proto2"; | |
2 option optimize_for = LITE_RUNTIME; | |
3 package webrtc; | |
4 | |
5 // This is the main message to dump to a file, it can contain multiple event | |
6 // messages, but it is possible to append multiple EventStreams (each with a | |
7 // single event) to a file. | |
8 // This has the benefit that there's no need to keep all data in memory. | |
9 message ACMDumpEventStream { | |
10 repeated ACMDumpEvent stream = 1; | |
11 } | |
12 | |
13 | |
14 message ACMDumpEvent { | |
15 // required - Elapsed wallclock time in us since the start of the log. | |
16 optional int64 timestamp_us = 1; | |
17 | |
18 // The different types of events that can occur, the UNKNOWN_EVENT entry | |
19 // is added in case future EventTypes are added, in that case old code will | |
20 // receive the new events as UNKNOWN_EVENT. | |
21 enum EventType { | |
22 UNKNOWN_EVENT = 0; | |
23 RTP_EVENT = 1; | |
24 DEBUG_EVENT = 2; | |
25 CONFIG_EVENT = 3; | |
26 } | |
27 | |
28 // required - Indicates the type of this event | |
29 optional EventType type = 2; | |
30 | |
31 // optional - but required if type == RTP_EVENT | |
32 optional ACMDumpRTPPacket packet = 3; | |
33 | |
34 // optional - but required if type == DEBUG_EVENT | |
35 optional ACMDumpDebugEvent debug_event = 4; | |
36 | |
37 // optional - but required if type == CONFIG_EVENT | |
38 optional ACMDumpConfigEvent config = 5; | |
39 } | |
40 | |
41 | |
42 message ACMDumpRTPPacket { | |
43 // Indicates if the packet is incoming or outgoing with respect to the user | |
44 // that is logging the data. | |
45 enum Direction { | |
46 UNKNOWN_DIRECTION = 0; | |
47 OUTGOING = 1; | |
48 INCOMING = 2; | |
49 } | |
50 enum PayloadType { | |
51 UNKNOWN_TYPE = 0; | |
52 AUDIO = 1; | |
53 VIDEO = 2; | |
54 RTX = 3; | |
55 } | |
56 | |
57 // required | |
58 optional Direction direction = 1; | |
59 | |
60 // required | |
61 optional PayloadType type = 2; | |
62 | |
63 // required - Contains the whole RTP packet (header+payload). | |
64 optional bytes RTP_data = 3; | |
65 } | |
66 | |
67 | |
68 message ACMDumpDebugEvent { | |
69 // Indicates the type of the debug event. | |
70 // LOG_START and LOG_END indicate the start and end of the log respectively. | |
71 // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM. | |
72 enum EventType { | |
73 UNKNOWN_EVENT = 0; | |
74 LOG_START = 1; | |
75 LOG_END = 2; | |
76 AUDIO_PLAYOUT = 3; | |
77 } | |
78 | |
79 // required | |
80 optional EventType type = 1; | |
81 | |
82 // An optional message that can be used to store additional information about | |
83 // the debug event. | |
84 optional string message = 2; | |
85 } | |
86 | |
87 | |
88 // TODO(terelius): Video and audio streams could in principle share SSRC, | |
89 // so identifying a stream based only on SSRC might not work. | |
90 // It might be better to use a combination of SSRC and media type | |
91 // or SSRC and port number, but for now we will rely on SSRC only. | |
92 message ACMDumpConfigEvent { | |
93 // Synchronization source (stream identifier) to be received. | |
94 optional uint32 remote_ssrc = 1; | |
95 | |
96 // RTX settings for incoming video payloads that may be received. RTX is | |
97 // disabled if there's no config present. | |
98 optional RtcpConfig rtcp_config = 3; | |
99 | |
100 // Map from video RTP payload type -> RTX config. | |
101 repeated RtxMap rtx_map = 4; | |
102 | |
103 // RTP header extensions used for the received stream. | |
104 repeated RtpHeaderExtension header_extensions = 5; | |
105 | |
106 // List of decoders associated with the stream. | |
107 repeated DecoderConfig decoders = 6; | |
108 } | |
109 | |
110 | |
111 // Maps decoder names to payload types. | |
112 message DecoderConfig { | |
113 // required | |
114 optional string name = 1; | |
115 | |
116 // required | |
117 optional sint32 payload_type = 2; | |
118 } | |
119 | |
120 | |
121 // Maps RTP header extension names to numerical ids. | |
122 message RtpHeaderExtension { | |
123 // required | |
124 optional string name = 1; | |
125 | |
126 // required | |
127 optional sint32 id = 2; | |
128 } | |
129 | |
130 | |
131 // RTX settings for incoming video payloads that may be received. | |
132 // RTX is disabled if there's no config present. | |
133 message RtxConfig { | |
134 // required - SSRCs to use for the RTX streams. | |
135 optional uint32 ssrc = 1; | |
136 | |
137 // required - Payload type to use for the RTX stream. | |
138 optional sint32 payload_type = 2; | |
139 } | |
140 | |
141 | |
142 message RtxMap { | |
143 // required | |
144 optional sint32 payload_type = 1; | |
145 | |
146 // required | |
147 optional RtxConfig config = 2; | |
148 } | |
149 | |
150 | |
151 // Configuration information for RTCP. | |
152 // For bandwidth estimation purposes it is more interesting to log the | |
153 // RTCP messages that the sender receives, but we will support logging | |
154 // at the receiver side too. | |
155 message RtcpConfig { | |
156 // Sender SSRC used for sending RTCP (such as receiver reports). | |
157 optional uint32 local_ssrc = 1; | |
158 | |
159 // RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size | |
160 // RTCP mode is described by RFC 5506. | |
161 enum RtcpMode {RTCP_COMPOUND = 1; RTCP_REDUCEDSIZE = 2;} | |
162 optional RtcpMode rtcp_mode = 2; | |
163 | |
164 // Extended RTCP settings. | |
165 optional bool receiver_reference_time_report = 3; | |
166 | |
167 // Receiver estimated maximum bandwidth. | |
168 optional bool remb = 4; | |
169 } | |
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