| /external/webrtc/webrtc/video/ |
| D | vie_channel.cc | 153 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) in Init()
154 rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); in Init()
186 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { in ~ViEChannel()
187 module_process_thread_->DeRegisterModule(rtp_rtcp); in ~ViEChannel()
188 delete rtp_rtcp; in ~ViEChannel()
382 for (RtpRtcp* rtp_rtcp : deregistered_modules) { in SetSendCodec()
383 rtp_rtcp->SetSendingStatus(false); in SetSendCodec()
384 rtp_rtcp->SetSendingMediaStatus(false); in SetSendCodec()
388 for (RtpRtcp* rtp_rtcp : registered_modules) { in SetSendCodec()
389 rtp_rtcp->DeRegisterSendPayload(video_codec.plType); in SetSendCodec()
[all …]
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| D | vie_remb.cc | 39 void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) { in AddReceiveChannel() argument
40 assert(rtp_rtcp); in AddReceiveChannel()
43 if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) != in AddReceiveChannel()
49 receive_modules_.push_back(rtp_rtcp); in AddReceiveChannel()
52 void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) { in RemoveReceiveChannel() argument
53 assert(rtp_rtcp); in RemoveReceiveChannel()
58 if ((*it) == rtp_rtcp) { in RemoveReceiveChannel()
65 void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) { in AddRembSender() argument
66 assert(rtp_rtcp); in AddRembSender()
71 if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) != in AddRembSender()
[all …]
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| D | vie_remb.h | 35 void AddReceiveChannel(RtpRtcp* rtp_rtcp);
38 void RemoveReceiveChannel(RtpRtcp* rtp_rtcp);
41 void AddRembSender(RtpRtcp* rtp_rtcp);
44 void RemoveRembSender(RtpRtcp* rtp_rtcp);
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| D | vie_sync_module.cc | 25 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { in UpdateMeasurements() argument
34 if (0 != rtp_rtcp.RemoteNTP(&ntp_secs, in UpdateMeasurements()
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| D | BUILD.gn | 66 "../modules/rtp_rtcp",
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| D | vie_receiver.cc | 418 for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_) in InsertRTCPPacket()
419 rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); in InsertRTCPPacket()
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| D | webrtc_video.gypi | 16 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
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| D | video_receive_stream.cc | 202 vie_channel_->rtp_rtcp()); in VideoReceiveStream()
297 vie_channel_->rtp_rtcp()); in ~VideoReceiveStream()
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| /external/webrtc/webrtc/modules/ |
| D | modules.gyp | 21 'rtp_rtcp/rtp_rtcp.gypi',
35 'rtp_rtcp/test/testFec/test_fec.gypi',
65 'rtp_rtcp',
92 'rtp_rtcp/test/testFec/test_fec.cc',
136 'rtp_rtcp',
294 'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
295 'rtp_rtcp/source/byte_io_unittest.cc',
296 'rtp_rtcp/source/fec_receiver_unittest.cc',
297 'rtp_rtcp/source/fec_test_helper.cc',
298 'rtp_rtcp/source/fec_test_helper.h',
[all …]
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator.gypi | 114 'rtp_rtcp',
125 '../rtp_rtcp/rtp_rtcp.gypi',
132 'rtp_rtcp',
147 '../rtp_rtcp/rtp_rtcp.gypi',
154 'rtp_rtcp',
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| /external/webrtc/webrtc/test/fuzzers/ |
| D | BUILD.gn | 29 "../../modules/rtp_rtcp",
38 "../../modules/rtp_rtcp",
47 "../../modules/rtp_rtcp",
65 "../../modules/rtp_rtcp/",
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| /external/webrtc/webrtc/voice_engine/test/android/android_test/jni/ |
| D | android_test.cc | 85 if (!veData1.rtp_rtcp) \
128 VoERTP_RTCP* rtp_rtcp; member
1192 veData.rtp_rtcp = VoERTP_RTCP::GetInterface(veData.ve); in GetSubApis()
1193 if (!veData.rtp_rtcp) in GetSubApis()
1316 if (veData.rtp_rtcp) in ReleaseSubApis()
1318 if (0 != veData.rtp_rtcp->Release()) in ReleaseSubApis()
1326 veData.rtp_rtcp = NULL; in ReleaseSubApis()
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| /external/webrtc/ |
| D | WATCHLISTS | 86 'rtp_rtcp': {
87 'filepath': 'webrtc/modules/rtp_rtcp/.*'
177 'rtp_rtcp': ['mflodman@webrtc.org',
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| /external/webrtc/webrtc/voice_engine/test/cmd_test/ |
| D | voe_cmd_test.cc | 61 VoERTP_RTCP* rtp_rtcp = NULL; variable
135 rtp_rtcp = VoERTP_RTCP::GetInterface(m_voe); in main()
196 if (rtp_rtcp) in main()
197 rtp_rtcp->Release(); in main()
469 res = rtp_rtcp->SetREDStatus(chan, true, cinst.pltype); in RunTest()
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| /external/webrtc/webrtc/call/ |
| D | webrtc_call.gypi | 12 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
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| D | BUILD.gn | 32 "../modules/rtp_rtcp",
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| /external/webrtc/webrtc/modules/pacing/ |
| D | pacing.gypi | 17 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
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| D | BUILD.gn | 31 "../rtp_rtcp",
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| /external/webrtc/webrtc/modules/rtp_rtcp/test/testFec/ |
| D | test_fec.gypi | 16 'rtp_rtcp',
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| /external/webrtc/webrtc/modules/video_coding/ |
| D | video_coding_test.gypi | 14 'rtp_rtcp',
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| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api_rtcp.cc | 55 explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} in TestRtpFeedback() argument
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| /external/webrtc/webrtc/modules/rtp_rtcp/ |
| D | rtp_rtcp.gypi | 12 'target_name': 'rtp_rtcp',
26 'include/rtp_rtcp.h',
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| D | BUILD.gn | 11 source_set("rtp_rtcp") {
19 "include/rtp_rtcp.h",
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | nack_rtx_unittest.cc | 56 explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} in TestRtpFeedback() argument
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| /external/webrtc/webrtc/ |
| D | webrtc_tests.gypi | 187 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
240 'modules/modules.gyp:rtp_rtcp',
|