Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(512)

Side by Side Diff: webrtc/video/rtc_event_log.proto

Issue 1227923005: Split webrtc/video into webrtc/{audio,call,video}. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/video/rtc_event_log.cc ('k') | webrtc/video/rtc_event_log2rtp_dump.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
(Empty)
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 // required if type == AUDIO_PLAYOUT
111 optional uint32 local_ssrc = 2;
112 }
113
114
115 // TODO(terelius): Video and audio streams could in principle share SSRC,
116 // so identifying a stream based only on SSRC might not work.
117 // It might be better to use a combination of SSRC and media type
118 // or SSRC and port number, but for now we will rely on SSRC only.
119 message VideoReceiveConfig {
120 // required - Synchronization source (stream identifier) to be received.
121 optional uint32 remote_ssrc = 1;
122 // required - Sender SSRC used for sending RTCP (such as receiver reports).
123 optional uint32 local_ssrc = 2;
124
125 // Compound mode is described by RFC 4585 and reduced-size
126 // RTCP mode is described by RFC 5506.
127 enum RtcpMode {
128 RTCP_COMPOUND = 1;
129 RTCP_REDUCEDSIZE = 2;
130 }
131 // required - RTCP mode to use.
132 optional RtcpMode rtcp_mode = 3;
133
134 // required - Extended RTCP settings.
135 optional bool receiver_reference_time_report = 4;
136
137 // required - Receiver estimated maximum bandwidth.
138 optional bool remb = 5;
139
140 // Map from video RTP payload type -> RTX config.
141 repeated RtxMap rtx_map = 6;
142
143 // RTP header extensions used for the received stream.
144 repeated RtpHeaderExtension header_extensions = 7;
145
146 // List of decoders associated with the stream.
147 repeated DecoderConfig decoders = 8;
148 }
149
150
151 // Maps decoder names to payload types.
152 message DecoderConfig {
153 // required
154 optional string name = 1;
155
156 // required
157 optional sint32 payload_type = 2;
158 }
159
160
161 // Maps RTP header extension names to numerical IDs.
162 message RtpHeaderExtension {
163 // required
164 optional string name = 1;
165
166 // required
167 optional sint32 id = 2;
168 }
169
170
171 // RTX settings for incoming video payloads that may be received.
172 // RTX is disabled if there's no config present.
173 message RtxConfig {
174 // required - SSRC to use for the RTX stream.
175 optional uint32 rtx_ssrc = 1;
176
177 // required - Payload type to use for the RTX stream.
178 optional sint32 rtx_payload_type = 2;
179 }
180
181
182 message RtxMap {
183 // required
184 optional sint32 payload_type = 1;
185
186 // required
187 optional RtxConfig config = 2;
188 }
189
190
191 message VideoSendConfig {
192 // Synchronization source (stream identifier) for outgoing stream.
193 // One stream can have several ssrcs for e.g. simulcast.
194 // At least one ssrc is required.
195 repeated uint32 ssrcs = 1;
196
197 // RTP header extensions used for the outgoing stream.
198 repeated RtpHeaderExtension header_extensions = 2;
199
200 // List of SSRCs for retransmitted packets.
201 repeated uint32 rtx_ssrcs = 3;
202
203 // required if rtx_ssrcs is used - Payload type for retransmitted packets.
204 optional sint32 rtx_payload_type = 4;
205
206 // required - Canonical end-point identifier.
207 optional string c_name = 5;
208
209 // required - Encoder associated with the stream.
210 optional EncoderConfig encoder = 6;
211 }
212
213
214 // Maps encoder names to payload types.
215 message EncoderConfig {
216 // required
217 optional string name = 1;
218
219 // required
220 optional sint32 payload_type = 2;
221 }
222
223
224 message AudioReceiveConfig {
225 // TODO(terelius): Add audio-receive config.
226 }
227
228
229 message AudioSendConfig {
230 // TODO(terelius): Add audio-receive config.
231 }
OLDNEW
« no previous file with comments | « webrtc/video/rtc_event_log.cc ('k') | webrtc/video/rtc_event_log2rtp_dump.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698