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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 RelEventStream { |
| 10 repeated RelEvent stream = 1; |
| 11 } |
| 12 |
| 13 |
| 14 message RelEvent { |
| 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 RTCP_EVENT = 2; |
| 25 DEBUG_EVENT = 3; |
| 26 RECEIVER_CONFIG_EVENT = 4; |
| 27 SENDER_CONFIG_EVENT = 5; |
| 28 AUDIO_RECEIVER_CONFIG_EVENT = 6; |
| 29 AUDIO_SENDER_CONFIG_EVENT = 7; |
| 30 } |
| 31 |
| 32 // required - Indicates the type of this event |
| 33 optional EventType type = 2; |
| 34 |
| 35 // optional - but required if type == RTP_EVENT |
| 36 optional RelRtpPacket rtp_packet = 3; |
| 37 |
| 38 // optional - but required if type == RTCP_EVENT |
| 39 optional RelRtcpPacket rtcp_packet = 4; |
| 40 |
| 41 // optional - but required if type == DEBUG_EVENT |
| 42 optional RelDebugEvent debug_event = 5; |
| 43 |
| 44 // optional - but required if type == RECEIVER_CONFIG_EVENT |
| 45 optional RelVideoReceiveConfig receiver_config = 6; |
| 46 |
| 47 // optional - but required if type == SENDER_CONFIG_EVENT |
| 48 optional RelVideoSendConfig sender_config = 7; |
| 49 |
| 50 // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT |
| 51 optional RelAudioReceiveConfig audio_receiver_config = 8; |
| 52 |
| 53 // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT |
| 54 optional RelAudioSendConfig audio_sender_config = 9; |
| 55 } |
| 56 |
| 57 |
| 58 message RelRtpPacket { |
| 59 // Indicates if the packet is incoming or outgoing with respect to the user |
| 60 // that is logging the data. |
| 61 enum Direction { |
| 62 UNKNOWN_DIRECTION = 0; |
| 63 OUTGOING = 1; |
| 64 INCOMING = 2; |
| 65 } |
| 66 enum PayloadType { |
| 67 UNKNOWN_TYPE = 0; |
| 68 AUDIO = 1; |
| 69 VIDEO = 2; |
| 70 RTX = 3; // TODO(terelius): Where does this get set? |
| 71 } |
| 72 |
| 73 // required |
| 74 optional Direction direction = 1; |
| 75 |
| 76 // required |
| 77 optional PayloadType type = 2; |
| 78 |
| 79 // required - The size of the packet including both payload and header. |
| 80 optional uint32 packet_length = 3; |
| 81 |
| 82 // required - The RTP header only. |
| 83 optional bytes header = 4; |
| 84 |
| 85 // Logging payloads for user data requires privacy review. Don't uncomment. |
| 86 // optional bytes payload = 5; |
| 87 } |
| 88 |
| 89 |
| 90 message RelRtcpPacket { |
| 91 // Indicates if the packet is incoming or outgoing with respect to the user |
| 92 // that is logging the data. |
| 93 enum Direction { |
| 94 UNKNOWN_DIRECTION = 0; |
| 95 OUTGOING = 1; |
| 96 INCOMING = 2; |
| 97 } |
| 98 enum PayloadType { |
| 99 UNKNOWN_TYPE = 0; |
| 100 AUDIO = 1; |
| 101 VIDEO = 2; |
| 102 } |
| 103 |
| 104 // required |
| 105 optional Direction direction = 1; |
| 106 |
| 107 // required |
| 108 optional PayloadType type = 2; |
| 109 |
| 110 // required - The whole packet including both payload and header. |
| 111 optional bytes data = 3; |
| 112 } |
| 113 |
| 114 |
| 115 message RelDebugEvent { |
| 116 // Indicates the type of the debug event. |
| 117 // LOG_START and LOG_END indicate the start and end of the log respectively. |
| 118 // AUDIO_PLAYOUT indicates a call to the PlayoutData10Ms() function in ACM. |
| 119 enum EventType { |
| 120 UNKNOWN_EVENT = 0; |
| 121 LOG_START = 1; |
| 122 LOG_END = 2; |
| 123 AUDIO_PLAYOUT = 3; |
| 124 } |
| 125 |
| 126 // required |
| 127 optional EventType type = 1; |
| 128 |
| 129 // An optional message that can be used to store additional information about |
| 130 // the debug event. |
| 131 optional string message = 2; |
| 132 } |
| 133 |
| 134 |
| 135 // TODO(terelius): Video and audio streams could in principle share SSRC, |
| 136 // so identifying a stream based only on SSRC might not work. |
| 137 // It might be better to use a combination of SSRC and media type |
| 138 // or SSRC and port number, but for now we will rely on SSRC only. |
| 139 message RelVideoReceiveConfig { |
| 140 // required - Synchronization source (stream identifier) to be received. |
| 141 optional uint32 remote_ssrc = 1; |
| 142 // required - Sender SSRC used for sending RTCP (such as receiver reports). |
| 143 optional uint32 local_ssrc = 2; |
| 144 |
| 145 // Compound mode is described by RFC 4585 and reduced-size |
| 146 // RTCP mode is described by RFC 5506. |
| 147 enum RtcpMode { |
| 148 RTCP_COMPOUND = 1; |
| 149 RTCP_REDUCEDSIZE = 2; |
| 150 } |
| 151 // required - RTCP mode to use. |
| 152 optional RtcpMode rtcp_mode = 3; |
| 153 |
| 154 // required - Extended RTCP settings. |
| 155 optional bool receiver_reference_time_report = 4; |
| 156 |
| 157 // required - Receiver estimated maximum bandwidth. |
| 158 optional bool remb = 5; |
| 159 |
| 160 // Map from video RTP payload type -> RTX config. |
| 161 repeated RtxMap rtx_map = 6; |
| 162 |
| 163 // RTP header extensions used for the received stream. |
| 164 repeated RtpHeaderExtension header_extensions = 7; |
| 165 |
| 166 // List of decoders associated with the stream. |
| 167 repeated DecoderConfig decoders = 8; |
| 168 } |
| 169 |
| 170 |
| 171 // Maps decoder names to payload types. |
| 172 message DecoderConfig { |
| 173 // required |
| 174 optional string name = 1; |
| 175 |
| 176 // required |
| 177 optional sint32 payload_type = 2; |
| 178 } |
| 179 |
| 180 |
| 181 // Maps RTP header extension names to numerical IDs. |
| 182 message RtpHeaderExtension { |
| 183 // required |
| 184 optional string name = 1; |
| 185 |
| 186 // required |
| 187 optional sint32 id = 2; |
| 188 } |
| 189 |
| 190 |
| 191 // RTX settings for incoming video payloads that may be received. |
| 192 // RTX is disabled if there's no config present. |
| 193 message RtxConfig { |
| 194 // required - SSRC to use for the RTX stream. |
| 195 optional uint32 rtx_ssrc = 1; |
| 196 |
| 197 // required - Payload type to use for the RTX stream. |
| 198 optional sint32 rtx_payload_type = 2; |
| 199 } |
| 200 |
| 201 |
| 202 message RtxMap { |
| 203 // required |
| 204 optional sint32 payload_type = 1; |
| 205 |
| 206 // required |
| 207 optional RtxConfig config = 2; |
| 208 } |
| 209 |
| 210 |
| 211 message RelVideoSendConfig { |
| 212 // Synchronization source (stream identifier) for outgoing stream. |
| 213 // One stream can have several ssrcs for e.g. simulcast. |
| 214 // At least one ssrc is required. |
| 215 repeated uint32 ssrcs = 1; |
| 216 |
| 217 // RTP header extensions used for the outgoing stream. |
| 218 repeated RtpHeaderExtension header_extensions = 2; |
| 219 |
| 220 // List of SSRCs for retransmitted packets. |
| 221 repeated uint32 rtx_ssrcs = 3; |
| 222 |
| 223 // required if rtx_ssrcs is used - Payload type for retransmitted packets. |
| 224 optional sint32 rtx_payload_type = 4; |
| 225 |
| 226 // required - Canonical end-point identifier. |
| 227 optional string c_name = 5; |
| 228 |
| 229 // required - Encoder associated with the stream. |
| 230 optional EncoderConfig encoder = 6; |
| 231 } |
| 232 |
| 233 |
| 234 // Maps encoder names to payload types. |
| 235 message EncoderConfig { |
| 236 // required |
| 237 optional string name = 1; |
| 238 |
| 239 // required |
| 240 optional sint32 payload_type = 2; |
| 241 } |
| 242 |
| 243 |
| 244 message RelAudioReceiveConfig { |
| 245 // TODO(terelius): Figure out what the requirements are from the audio team. |
| 246 } |
| 247 |
| 248 |
| 249 message RelAudioSendConfig { |
| 250 // TODO(terelius): Figure out what the requirements are from the audio team. |
| 251 } |
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