Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(14)

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

Issue 2702203002: Reland of Delete class SSRCDatabase, and its global ssrc registry. (Closed)
Patch Set: Move SetRtcpReceiverSsrcs call to ModuleRtpRtcpImpl::SetSendingStatus. Update tests to set SSRC ear… Created 3 years, 10 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 | « no previous file | webrtc/modules/rtp_rtcp/BUILD.gn » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 }
(...skipping 84 matching lines...) Expand 10 before | Expand all | Expand 10 after
95 optional bool incoming = 1; 95 optional bool incoming = 1;
96 96
97 // required 97 // required
98 optional MediaType type = 2; 98 optional MediaType type = 2;
99 99
100 // required - The whole packet including both payload and header. 100 // required - The whole packet including both payload and header.
101 optional bytes packet_data = 3; 101 optional bytes packet_data = 3;
102 } 102 }
103 103
104 message AudioPlayoutEvent { 104 message AudioPlayoutEvent {
105 // TODO(ivoc): Rename, we currently use the "remote" ssrc, i.e. identifying
106 // the receive stream, while local_ssrc identifies the send stream, if any.
105 // required - The SSRC of the audio stream associated with the playout event. 107 // required - The SSRC of the audio stream associated with the playout event.
106 optional uint32 local_ssrc = 2; 108 optional uint32 local_ssrc = 2;
107 } 109 }
108 110
109 message BwePacketLossEvent { 111 message BwePacketLossEvent {
110 // required - Bandwidth estimate (in bps) after the update. 112 // required - Bandwidth estimate (in bps) after the update.
111 optional int32 bitrate = 1; 113 optional int32 bitrate = 1;
112 114
113 // required - Fraction of lost packets since last receiver report 115 // required - Fraction of lost packets since last receiver report
114 // computed as floor( 256 * (#lost_packets / #total_packets) ). 116 // computed as floor( 256 * (#lost_packets / #total_packets) ).
(...skipping 147 matching lines...) Expand 10 before | Expand all | Expand 10 after
262 264
263 // Whether forward error correction (FEC) is turned on or off. 265 // Whether forward error correction (FEC) is turned on or off.
264 optional bool enable_fec = 4; 266 optional bool enable_fec = 4;
265 267
266 // Whether discontinuous transmission (DTX) is turned on or off. 268 // Whether discontinuous transmission (DTX) is turned on or off.
267 optional bool enable_dtx = 5; 269 optional bool enable_dtx = 5;
268 270
269 // Number of audio channels that each encoded packet consists of. 271 // Number of audio channels that each encoded packet consists of.
270 optional uint32 num_channels = 6; 272 optional uint32 num_channels = 6;
271 } 273 }
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/BUILD.gn » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698