| OLD | NEW |
| (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 } | |
| OLD | NEW |