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1 syntax = "proto2"; | 1 syntax = "proto2"; |
2 option optimize_for = LITE_RUNTIME; | 2 option optimize_for = LITE_RUNTIME; |
3 package webrtc.rtclog; | 3 package webrtc.rtclog; |
4 | 4 |
5 enum MediaType { | 5 enum MediaType { |
6 ANY = 0; | 6 ANY = 0; |
7 AUDIO = 1; | 7 AUDIO = 1; |
8 VIDEO = 2; | 8 VIDEO = 2; |
9 DATA = 3; | 9 DATA = 3; |
10 } | 10 } |
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30 LOG_END = 2; | 30 LOG_END = 2; |
31 RTP_EVENT = 3; | 31 RTP_EVENT = 3; |
32 RTCP_EVENT = 4; | 32 RTCP_EVENT = 4; |
33 AUDIO_PLAYOUT_EVENT = 5; | 33 AUDIO_PLAYOUT_EVENT = 5; |
34 BWE_PACKET_LOSS_EVENT = 6; | 34 BWE_PACKET_LOSS_EVENT = 6; |
35 BWE_PACKET_DELAY_EVENT = 7; | 35 BWE_PACKET_DELAY_EVENT = 7; |
36 VIDEO_RECEIVER_CONFIG_EVENT = 8; | 36 VIDEO_RECEIVER_CONFIG_EVENT = 8; |
37 VIDEO_SENDER_CONFIG_EVENT = 9; | 37 VIDEO_SENDER_CONFIG_EVENT = 9; |
38 AUDIO_RECEIVER_CONFIG_EVENT = 10; | 38 AUDIO_RECEIVER_CONFIG_EVENT = 10; |
39 AUDIO_SENDER_CONFIG_EVENT = 11; | 39 AUDIO_SENDER_CONFIG_EVENT = 11; |
40 AUDIO_NETWORK_ADAPTATION_EVENT = 16; | |
41 } | 40 } |
42 | 41 |
43 // required - Indicates the type of this event | 42 // required - Indicates the type of this event |
44 optional EventType type = 2; | 43 optional EventType type = 2; |
45 | 44 |
46 // optional - but required if type == RTP_EVENT | 45 // optional - but required if type == RTP_EVENT |
47 optional RtpPacket rtp_packet = 3; | 46 optional RtpPacket rtp_packet = 3; |
48 | 47 |
49 // optional - but required if type == RTCP_EVENT | 48 // optional - but required if type == RTCP_EVENT |
50 optional RtcpPacket rtcp_packet = 4; | 49 optional RtcpPacket rtcp_packet = 4; |
51 | 50 |
52 // optional - but required if type == AUDIO_PLAYOUT_EVENT | 51 // optional - but required if type == AUDIO_PLAYOUT_EVENT |
53 optional AudioPlayoutEvent audio_playout_event = 5; | 52 optional AudioPlayoutEvent audio_playout_event = 5; |
54 | 53 |
55 // optional - but required if type == BWE_PACKET_LOSS_EVENT | 54 // optional - but required if type == BWE_PACKET_LOSS_EVENT |
56 optional BwePacketLossEvent bwe_packet_loss_event = 6; | 55 optional BwePacketLossEvent bwe_packet_loss_event = 6; |
57 | 56 |
58 // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT | 57 // optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT |
59 optional VideoReceiveConfig video_receiver_config = 8; | 58 optional VideoReceiveConfig video_receiver_config = 8; |
60 | 59 |
61 // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT | 60 // optional - but required if type == VIDEO_SENDER_CONFIG_EVENT |
62 optional VideoSendConfig video_sender_config = 9; | 61 optional VideoSendConfig video_sender_config = 9; |
63 | 62 |
64 // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT | 63 // optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT |
65 optional AudioReceiveConfig audio_receiver_config = 10; | 64 optional AudioReceiveConfig audio_receiver_config = 10; |
66 | 65 |
67 // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT | 66 // optional - but required if type == AUDIO_SENDER_CONFIG_EVENT |
68 optional AudioSendConfig audio_sender_config = 11; | 67 optional AudioSendConfig audio_sender_config = 11; |
69 | |
70 // optional - but required if type == AUDIO_NETWORK_ADAPTATION_EVENT | |
71 optional AudioNetworkAdaptation audio_network_adaptation = 16; | |
72 } | 68 } |
73 | 69 |
74 message RtpPacket { | 70 message RtpPacket { |
75 // required - True if the packet is incoming w.r.t. the user logging the data | 71 // required - True if the packet is incoming w.r.t. the user logging the data |
76 optional bool incoming = 1; | 72 optional bool incoming = 1; |
77 | 73 |
78 // required | 74 // required |
79 optional MediaType type = 2; | 75 optional MediaType type = 2; |
80 | 76 |
81 // required - The size of the packet including both payload and header. | 77 // required - The size of the packet including both payload and header. |
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224 repeated RtpHeaderExtension header_extensions = 3; | 220 repeated RtpHeaderExtension header_extensions = 3; |
225 } | 221 } |
226 | 222 |
227 message AudioSendConfig { | 223 message AudioSendConfig { |
228 // required - Synchronization source (stream identifier) for outgoing stream. | 224 // required - Synchronization source (stream identifier) for outgoing stream. |
229 optional uint32 ssrc = 1; | 225 optional uint32 ssrc = 1; |
230 | 226 |
231 // RTP header extensions used for the outgoing audio stream. | 227 // RTP header extensions used for the outgoing audio stream. |
232 repeated RtpHeaderExtension header_extensions = 2; | 228 repeated RtpHeaderExtension header_extensions = 2; |
233 } | 229 } |
234 | |
235 message AudioNetworkAdaptation { | |
236 // Bit rate that the audio encoder is operating at. | |
237 optional int32 bitrate_bps = 1; | |
238 | |
239 // Frame length that each encoded audio packet consists of. | |
240 optional int32 frame_length_ms = 2; | |
241 | |
242 // Packet loss fraction that the encoder's forward error correction (FEC) is | |
243 // optimized for. | |
244 optional float uplink_packet_loss_fraction = 3; | |
245 | |
246 // Whether forward error correction (FEC) is turned on or off. | |
247 optional bool enable_fec = 4; | |
248 | |
249 // Whether discontinuous transmission (DTX) is turned on or off. | |
250 optional bool enable_dtx = 5; | |
251 | |
252 // Number of audio channels that each encoded packet consists of. | |
253 optional uint32 num_channels = 6; | |
254 } | |
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