1syntax = "proto2";
2option optimize_for = LITE_RUNTIME;
3package webrtc.rtclog;
4
5
6enum 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.
18message EventStream {
19  repeated Event stream = 1;
20}
21
22
23message 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    LOG_START = 1;
33    LOG_END = 2;
34    RTP_EVENT = 3;
35    RTCP_EVENT = 4;
36    AUDIO_PLAYOUT_EVENT = 5;
37    BWE_PACKET_LOSS_EVENT = 6;
38    BWE_PACKET_DELAY_EVENT = 7;
39    VIDEO_RECEIVER_CONFIG_EVENT = 8;
40    VIDEO_SENDER_CONFIG_EVENT = 9;
41    AUDIO_RECEIVER_CONFIG_EVENT = 10;
42    AUDIO_SENDER_CONFIG_EVENT = 11;
43  }
44
45  // required - Indicates the type of this event
46  optional EventType type = 2;
47
48  // optional - but required if type == RTP_EVENT
49  optional RtpPacket rtp_packet = 3;
50
51  // optional - but required if type == RTCP_EVENT
52  optional RtcpPacket rtcp_packet = 4;
53
54  // optional - but required if type == AUDIO_PLAYOUT_EVENT
55  optional AudioPlayoutEvent audio_playout_event = 5;
56
57  // optional - but required if type == BWE_PACKET_LOSS_EVENT
58  optional BwePacketLossEvent bwe_packet_loss_event = 6;
59
60  // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
61  optional VideoReceiveConfig video_receiver_config = 8;
62
63  // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
64  optional VideoSendConfig video_sender_config = 9;
65
66  // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
67  optional AudioReceiveConfig audio_receiver_config = 10;
68
69  // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
70  optional AudioSendConfig audio_sender_config = 11;
71}
72
73
74message RtpPacket {
75  // required - True if the packet is incoming w.r.t. the user logging the data
76  optional bool incoming = 1;
77
78  // required
79  optional MediaType type = 2;
80
81  // required - The size of the packet including both payload and header.
82  optional uint32 packet_length = 3;
83
84  // required - The RTP header only.
85  optional bytes header = 4;
86
87  // Do not add code to log user payload data without a privacy review!
88}
89
90
91message RtcpPacket {
92  // required - True if the packet is incoming w.r.t. the user logging the data
93  optional bool incoming = 1;
94
95  // required
96  optional MediaType type = 2;
97
98  // required - The whole packet including both payload and header.
99  optional bytes packet_data = 3;
100}
101
102message AudioPlayoutEvent {
103  // required - The SSRC of the audio stream associated with the playout event.
104  optional uint32 local_ssrc = 2;
105}
106
107message BwePacketLossEvent {
108  // required - Bandwidth estimate (in bps) after the update.
109  optional int32 bitrate = 1;
110
111  // required - Fraction of lost packets since last receiver report
112  // computed as floor( 256 * (#lost_packets / #total_packets) ).
113  // The possible values range from 0 to 255.
114  optional uint32 fraction_loss = 2;
115
116  // TODO(terelius): Is this really needed? Remove or make optional?
117  // required - Total number of packets that the BWE update is based on.
118  optional int32 total_packets = 3;
119}
120
121// TODO(terelius): Video and audio streams could in principle share SSRC,
122// so identifying a stream based only on SSRC might not work.
123// It might be better to use a combination of SSRC and media type
124// or SSRC and port number, but for now we will rely on SSRC only.
125message VideoReceiveConfig {
126  // required - Synchronization source (stream identifier) to be received.
127  optional uint32 remote_ssrc = 1;
128  // required - Sender SSRC used for sending RTCP (such as receiver reports).
129  optional uint32 local_ssrc = 2;
130
131  // Compound mode is described by RFC 4585 and reduced-size
132  // RTCP mode is described by RFC 5506.
133  enum RtcpMode {
134    RTCP_COMPOUND = 1;
135    RTCP_REDUCEDSIZE = 2;
136  }
137  // required - RTCP mode to use.
138  optional RtcpMode rtcp_mode = 3;
139
140  // required - Receiver estimated maximum bandwidth.
141  optional bool remb = 4;
142
143  // Map from video RTP payload type -> RTX config.
144  repeated RtxMap rtx_map = 5;
145
146  // RTP header extensions used for the received stream.
147  repeated RtpHeaderExtension header_extensions = 6;
148
149  // List of decoders associated with the stream.
150  repeated DecoderConfig decoders = 7;
151}
152
153
154// Maps decoder names to payload types.
155message DecoderConfig {
156  // required
157  optional string name = 1;
158
159  // required
160  optional int32 payload_type = 2;
161}
162
163
164// Maps RTP header extension names to numerical IDs.
165message RtpHeaderExtension {
166  // required
167  optional string name = 1;
168
169  // required
170  optional int32 id = 2;
171}
172
173
174// RTX settings for incoming video payloads that may be received.
175// RTX is disabled if there's no config present.
176message RtxConfig {
177  // required - SSRC to use for the RTX stream.
178  optional uint32 rtx_ssrc = 1;
179
180  // required - Payload type to use for the RTX stream.
181  optional int32 rtx_payload_type = 2;
182}
183
184
185message RtxMap {
186  // required
187  optional int32 payload_type = 1;
188
189  // required
190  optional RtxConfig config = 2;
191}
192
193
194message VideoSendConfig {
195  // Synchronization source (stream identifier) for outgoing stream.
196  // One stream can have several ssrcs for e.g. simulcast.
197  // At least one ssrc is required.
198  repeated uint32 ssrcs = 1;
199
200  // RTP header extensions used for the outgoing stream.
201  repeated RtpHeaderExtension header_extensions = 2;
202
203  // List of SSRCs for retransmitted packets.
204  repeated uint32 rtx_ssrcs = 3;
205
206  // required if rtx_ssrcs is used - Payload type for retransmitted packets.
207  optional int32 rtx_payload_type = 4;
208
209  // required - Encoder associated with the stream.
210  optional EncoderConfig encoder = 5;
211}
212
213
214// Maps encoder names to payload types.
215message EncoderConfig {
216  // required
217  optional string name = 1;
218
219  // required
220  optional int32 payload_type = 2;
221}
222
223
224message AudioReceiveConfig {
225  // required - Synchronization source (stream identifier) to be received.
226  optional uint32 remote_ssrc = 1;
227
228  // required - Sender SSRC used for sending RTCP (such as receiver reports).
229  optional uint32 local_ssrc = 2;
230
231  // RTP header extensions used for the received audio stream.
232  repeated RtpHeaderExtension header_extensions = 3;
233}
234
235
236message AudioSendConfig {
237  // required - Synchronization source (stream identifier) for outgoing stream.
238  optional uint32 ssrc = 1;
239
240  // RTP header extensions used for the outgoing audio stream.
241  repeated RtpHeaderExtension header_extensions = 2;
242}
243