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1 syntax = "proto2"; | |
2 option optimize_for = LITE_RUNTIME; | |
3 package webrtc.rtclog; | |
4 | |
5 | |
6 enum MediaType { | |
7 ANY = 0; | |
8 AUDIO = 1; | |
9 VIDEO = 2; | |
10 DATA = 3; | |
11 } | |
12 | |
13 | |
14 // This is the main message to dump to a file, it can contain multiple event | |
15 // messages, but it is possible to append multiple EventStreams (each with a | |
16 // single event) to a file. | |
17 // This has the benefit that there's no need to keep all data in memory. | |
18 message EventStream { | |
19 repeated Event stream = 1; | |
20 } | |
21 | |
22 | |
23 message Event { | |
24 // required - Elapsed wallclock time in us since the start of the log. | |
25 optional int64 timestamp_us = 1; | |
26 | |
27 // The different types of events that can occur, the UNKNOWN_EVENT entry | |
28 // is added in case future EventTypes are added, in that case old code will | |
29 // receive the new events as UNKNOWN_EVENT. | |
30 enum EventType { | |
31 UNKNOWN_EVENT = 0; | |
32 RTP_EVENT = 1; | |
33 RTCP_EVENT = 2; | |
34 DEBUG_EVENT = 3; | |
35 VIDEO_RECEIVER_CONFIG_EVENT = 4; | |
36 VIDEO_SENDER_CONFIG_EVENT = 5; | |
37 AUDIO_RECEIVER_CONFIG_EVENT = 6; | |
38 AUDIO_SENDER_CONFIG_EVENT = 7; | |
39 } | |
40 | |
41 // required - Indicates the type of this event | |
42 optional EventType type = 2; | |
43 | |
44 // optional - but required if type == RTP_EVENT | |
45 optional RtpPacket rtp_packet = 3; | |
46 | |
47 // optional - but required if type == RTCP_EVENT | |
48 optional RtcpPacket rtcp_packet = 4; | |
49 | |
50 // optional - but required if type == DEBUG_EVENT | |
51 optional DebugEvent debug_event = 5; | |
52 | |
53 // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT | |
54 optional VideoReceiveConfig video_receiver_config = 6; | |
55 | |
56 // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT | |
57 optional VideoSendConfig video_sender_config = 7; | |
58 | |
59 // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT | |
60 optional AudioReceiveConfig audio_receiver_config = 8; | |
61 | |
62 // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT | |
63 optional AudioSendConfig audio_sender_config = 9; | |
64 } | |
65 | |
66 | |
67 message RtpPacket { | |
68 // required - True if the packet is incoming w.r.t. the user logging the data | |
69 optional bool incoming = 1; | |
70 | |
71 // required | |
72 optional MediaType type = 2; | |
73 | |
74 // required - The size of the packet including both payload and header. | |
75 optional uint32 packet_length = 3; | |
76 | |
77 // required - The RTP header only. | |
78 optional bytes header = 4; | |
79 | |
80 // Do not add code to log user payload data without a privacy review! | |
81 } | |
82 | |
83 | |
84 message RtcpPacket { | |
85 // required - True if the packet is incoming w.r.t. the user logging the data | |
86 optional bool incoming = 1; | |
87 | |
88 // required | |
89 optional MediaType type = 2; | |
90 | |
91 // required - The whole packet including both payload and header. | |
92 optional bytes packet_data = 3; | |
93 } | |
94 | |
95 | |
96 message DebugEvent { | |
97 // Indicates the type of the debug event. | |
98 // LOG_START and LOG_END indicate the start and end of the log respectively. | |
99 // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM. | |
100 enum EventType { | |
101 UNKNOWN_EVENT = 0; | |
102 LOG_START = 1; | |
103 LOG_END = 2; | |
104 AUDIO_PLAYOUT = 3; | |
105 } | |
106 | |
107 // required | |
108 optional EventType type = 1; | |
109 } | |
110 | |
111 | |
112 // TODO(terelius): Video and audio streams could in principle share SSRC, | |
113 // so identifying a stream based only on SSRC might not work. | |
114 // It might be better to use a combination of SSRC and media type | |
115 // or SSRC and port number, but for now we will rely on SSRC only. | |
116 message VideoReceiveConfig { | |
117 // required - Synchronization source (stream identifier) to be received. | |
118 optional uint32 remote_ssrc = 1; | |
119 // required - Sender SSRC used for sending RTCP (such as receiver reports). | |
120 optional uint32 local_ssrc = 2; | |
121 | |
122 // Compound mode is described by RFC 4585 and reduced-size | |
123 // RTCP mode is described by RFC 5506. | |
124 enum RtcpMode { | |
125 RTCP_COMPOUND = 1; | |
126 RTCP_REDUCEDSIZE = 2; | |
127 } | |
128 // required - RTCP mode to use. | |
129 optional RtcpMode rtcp_mode = 3; | |
130 | |
131 // required - Extended RTCP settings. | |
132 optional bool receiver_reference_time_report = 4; | |
133 | |
134 // required - Receiver estimated maximum bandwidth. | |
135 optional bool remb = 5; | |
136 | |
137 // Map from video RTP payload type -> RTX config. | |
138 repeated RtxMap rtx_map = 6; | |
139 | |
140 // RTP header extensions used for the received stream. | |
141 repeated RtpHeaderExtension header_extensions = 7; | |
142 | |
143 // List of decoders associated with the stream. | |
144 repeated DecoderConfig decoders = 8; | |
145 } | |
146 | |
147 | |
148 // Maps decoder names to payload types. | |
149 message DecoderConfig { | |
150 // required | |
151 optional string name = 1; | |
152 | |
153 // required | |
154 optional sint32 payload_type = 2; | |
155 } | |
156 | |
157 | |
158 // Maps RTP header extension names to numerical IDs. | |
159 message RtpHeaderExtension { | |
160 // required | |
161 optional string name = 1; | |
162 | |
163 // required | |
164 optional sint32 id = 2; | |
165 } | |
166 | |
167 | |
168 // RTX settings for incoming video payloads that may be received. | |
169 // RTX is disabled if there's no config present. | |
170 message RtxConfig { | |
171 // required - SSRC to use for the RTX stream. | |
172 optional uint32 rtx_ssrc = 1; | |
173 | |
174 // required - Payload type to use for the RTX stream. | |
175 optional sint32 rtx_payload_type = 2; | |
176 } | |
177 | |
178 | |
179 message RtxMap { | |
180 // required | |
181 optional sint32 payload_type = 1; | |
182 | |
183 // required | |
184 optional RtxConfig config = 2; | |
185 } | |
186 | |
187 | |
188 message VideoSendConfig { | |
189 // Synchronization source (stream identifier) for outgoing stream. | |
190 // One stream can have several ssrcs for e.g. simulcast. | |
191 // At least one ssrc is required. | |
192 repeated uint32 ssrcs = 1; | |
193 | |
194 // RTP header extensions used for the outgoing stream. | |
195 repeated RtpHeaderExtension header_extensions = 2; | |
196 | |
197 // List of SSRCs for retransmitted packets. | |
198 repeated uint32 rtx_ssrcs = 3; | |
199 | |
200 // required if rtx_ssrcs is used - Payload type for retransmitted packets. | |
201 optional sint32 rtx_payload_type = 4; | |
202 | |
203 // required - Canonical end-point identifier. | |
204 optional string c_name = 5; | |
205 | |
206 // required - Encoder associated with the stream. | |
207 optional EncoderConfig encoder = 6; | |
208 } | |
209 | |
210 | |
211 // Maps encoder names to payload types. | |
212 message EncoderConfig { | |
213 // required | |
214 optional string name = 1; | |
215 | |
216 // required | |
217 optional sint32 payload_type = 2; | |
218 } | |
219 | |
220 | |
221 message AudioReceiveConfig { | |
222 // TODO(terelius): Add audio-receive config. | |
223 } | |
224 | |
225 | |
226 message AudioSendConfig { | |
227 // TODO(terelius): Add audio-receive config. | |
228 } | |
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