| /external/webrtc/test/fuzzers/ |
| D | BUILD.gn | 38 "../../modules/rtp_rtcp:rtp_rtcp_format",
77 deps = [ "../../modules/rtp_rtcp" ]
84 "../../modules/rtp_rtcp",
85 "../../modules/rtp_rtcp:rtp_video_header",
93 "../../modules/rtp_rtcp",
94 "../../modules/rtp_rtcp:rtp_video_header",
126 "../../modules/rtp_rtcp",
127 "../../modules/rtp_rtcp:rtp_rtcp_format",
136 "../../modules/rtp_rtcp",
137 "../../modules/rtp_rtcp:rtp_rtcp_format",
[all …]
|
| /external/webrtc/call/ |
| D | rtp_video_sender.cc | 40 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp, in RtpStreamSender() argument
43 : rtp_rtcp(std::move(rtp_rtcp)), in RtpStreamSender()
259 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp( in CreateRtpStreamSenders() local
261 rtp_rtcp->SetSendingStatus(false); in CreateRtpStreamSenders()
262 rtp_rtcp->SetSendingMediaStatus(false); in CreateRtpStreamSenders()
263 rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); in CreateRtpStreamSenders()
265 rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize); in CreateRtpStreamSenders()
268 video_config.rtp_sender = rtp_rtcp->RtpSender(); in CreateRtpStreamSenders()
291 rtp_streams.emplace_back(std::move(rtp_rtcp), std::move(sender_video), in CreateRtpStreamSenders()
400 transport->packet_router()->AddSendRtpModule(stream.rtp_rtcp.get(), in RtpVideoSender()
[all …]
|
| D | BUILD.gn | 58 "../modules/rtp_rtcp:rtp_rtcp_format",
100 "../modules/rtp_rtcp:rtp_rtcp_format",
124 "../modules/rtp_rtcp",
125 "../modules/rtp_rtcp:rtp_rtcp_format",
167 "../modules/rtp_rtcp",
168 "../modules/rtp_rtcp:rtp_rtcp_format",
169 "../modules/rtp_rtcp:rtp_video_header",
272 "../modules/rtp_rtcp",
273 "../modules/rtp_rtcp:rtp_rtcp_format",
319 "../modules/rtp_rtcp:rtp_rtcp_format",
[all …]
|
| /external/webrtc/audio/voip/test/ |
| D | BUILD.gn | 39 "../../../modules/rtp_rtcp:rtp_rtcp",
40 "../../../modules/rtp_rtcp:rtp_rtcp_format",
60 "../../../modules/rtp_rtcp:rtp_rtcp",
77 "../../../modules/rtp_rtcp:rtp_rtcp",
78 "../../../modules/rtp_rtcp:rtp_rtcp_format",
|
| D | audio_egress_unittest.cc | 40 auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(rtp_config); in CreateRtpStack() local
41 rtp_rtcp->SetSendingMediaStatus(false); in CreateRtpStack()
42 rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); in CreateRtpStack()
43 return rtp_rtcp; in CreateRtpStack()
|
| /external/webrtc/audio/voip/ |
| D | BUILD.gn | 47 "../../modules/rtp_rtcp",
48 "../../modules/rtp_rtcp:rtp_rtcp_format",
71 "../../modules/rtp_rtcp",
72 "../../modules/rtp_rtcp:rtp_rtcp_format",
94 "../../modules/rtp_rtcp",
95 "../../modules/rtp_rtcp:rtp_rtcp_format",
|
| D | audio_egress.cc | 20 AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp, in AudioEgress() argument
23 : rtp_rtcp_(rtp_rtcp), in AudioEgress()
|
| D | audio_ingress.cc | 38 RtpRtcpInterface* rtp_rtcp, in AudioIngress() argument
46 rtp_rtcp_(rtp_rtcp), in AudioIngress()
|
| /external/webrtc/modules/pacing/ |
| D | BUILD.gn | 58 "../rtp_rtcp",
59 "../rtp_rtcp:rtp_rtcp_format",
109 "../rtp_rtcp",
110 "../rtp_rtcp:mock_rtp_rtcp",
111 "../rtp_rtcp:rtp_rtcp_format",
|
| /external/webrtc/ |
| D | Android.bp | 2303 "modules/rtp_rtcp/source/rtp_video_header.cc",
3087 "modules/rtp_rtcp/include/report_block_data.cc",
3088 "modules/rtp_rtcp/include/rtp_rtcp_defines.cc",
3089 "modules/rtp_rtcp/source/rtcp_packet.cc",
3090 "modules/rtp_rtcp/source/rtcp_packet/app.cc",
3091 "modules/rtp_rtcp/source/rtcp_packet/bye.cc",
3092 "modules/rtp_rtcp/source/rtcp_packet/common_header.cc",
3093 "modules/rtp_rtcp/source/rtcp_packet/compound_packet.cc",
3094 "modules/rtp_rtcp/source/rtcp_packet/dlrr.cc",
3095 "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc",
[all …]
|
| D | WATCHLISTS | 91 'rtp_rtcp': {
92 'filepath': 'modules/rtp_rtcp/.*'
178 'rtp_rtcp': ['mflodman@webrtc.org',
|
| /external/webrtc/modules/remote_bitrate_estimator/ |
| D | BUILD.gn | 53 "../../modules/rtp_rtcp:rtp_rtcp_format",
81 "../rtp_rtcp",
94 "../../modules/rtp_rtcp",
131 "../rtp_rtcp:rtp_rtcp_format",
|
| /external/webrtc/modules/video_coding/ |
| D | BUILD.gn | 27 "../../modules/rtp_rtcp:rtp_video_header",
117 "../rtp_rtcp:rtp_video_header",
212 "../rtp_rtcp",
213 "../rtp_rtcp:rtp_rtcp_format",
278 "../../modules/rtp_rtcp:rtp_video_header",
286 "../rtp_rtcp:rtp_rtcp_format",
287 "../rtp_rtcp:rtp_video_header",
346 "../../modules/rtp_rtcp",
362 "../rtp_rtcp:rtp_rtcp_format",
443 "../rtp_rtcp:rtp_rtcp_format",
[all …]
|
| /external/webrtc/test/peer_scenario/tests/ |
| D | BUILD.gn | 22 "../../../modules/rtp_rtcp:rtp_rtcp",
|
| /external/webrtc/pc/scenario_tests/ |
| D | BUILD.gn | 17 "../../modules/rtp_rtcp:rtp_rtcp",
|
| /external/webrtc/logging/ |
| D | BUILD.gn | 147 "../modules/rtp_rtcp:rtp_rtcp_format",
217 "../modules/rtp_rtcp:rtp_rtcp_format",
319 "../modules/rtp_rtcp",
320 "../modules/rtp_rtcp:rtp_rtcp_format",
372 "../modules/rtp_rtcp:rtp_rtcp_format",
395 "../modules/rtp_rtcp",
396 "../modules/rtp_rtcp:rtp_rtcp_format",
|
| /external/webrtc/modules/rtp_rtcp/ |
| D | BUILD.gn | 130 rtc_library("rtp_rtcp") {
137 "include/rtp_rtcp.h", # deprecated
332 ":rtp_rtcp",
381 ":rtp_rtcp",
398 ":rtp_rtcp",
419 ":rtp_rtcp",
432 ":rtp_rtcp",
523 ":rtp_rtcp",
|
| /external/webrtc/video/ |
| D | BUILD.gn | 106 "../modules/rtp_rtcp",
107 "../modules/rtp_rtcp:rtp_rtcp_format",
108 "../modules/rtp_rtcp:rtp_video_header",
320 "../modules/rtp_rtcp",
321 "../modules/rtp_rtcp:rtp_rtcp_format",
625 "../modules/rtp_rtcp",
626 "../modules/rtp_rtcp:mock_rtp_rtcp",
627 "../modules/rtp_rtcp:rtp_rtcp_format",
|
| D | DEPS | 13 "+modules/rtp_rtcp",
|
| /external/webrtc/test/scenario/ |
| D | BUILD.gn | 119 "../../modules/rtp_rtcp",
120 "../../modules/rtp_rtcp:mock_rtp_rtcp",
121 "../../modules/rtp_rtcp:rtp_rtcp_format",
|
| /external/webrtc/modules/congestion_controller/rtp/ |
| D | BUILD.gn | 64 "../../rtp_rtcp:rtp_rtcp_format",
94 "../../rtp_rtcp:rtp_rtcp_format",
|
| /external/webrtc/audio/ |
| D | BUILD.gn | 82 "../modules/rtp_rtcp",
83 "../modules/rtp_rtcp:rtp_rtcp_format",
169 "../modules/rtp_rtcp:mock_rtp_rtcp",
170 "../modules/rtp_rtcp:rtp_rtcp_format",
|
| D | audio_send_stream_unittest.cc | 206 MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; } in rtp_rtcp() function
665 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
689 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
711 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
773 EXPECT_CALL(*helper.rtp_rtcp(), in TEST()
833 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
849 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
864 EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead) in TEST()
|
| /external/webrtc/logging/rtc_event_log/ |
| D | DEPS | 6 "+modules/rtp_rtcp",
|
| /external/webrtc/modules/audio_mixer/ |
| D | DEPS | 10 "+modules/rtp_rtcp",
|