/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/ |
H A D | moz.build | 10 "/media/webrtc/trunk/webrtc/api/array_view_gn", 31 "/media/webrtc/trunk/webrtc/api/call_api_gn", 32 "/media/webrtc/trunk/webrtc/api/optional_gn", 33 "/media/webrtc/trunk/webrtc/api/refcountedbase_gn", 34 "/media/webrtc/trunk/webrtc/api/transport_api_gn", 38 "/media/webrtc/trunk/webrtc/audio/audio_gn", 41 "/media/webrtc/trunk/webrtc/call/call_gn", 45 "/media/webrtc/trunk/webrtc/call/rtp_sender_gn", 60 "/media/webrtc/trunk/webrtc/media/rtc_media_gn", 136 "/media/webrtc/trunk/webrtc/video/video_gn", [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/ |
H A D | moz.build | 10 "/media/webrtc/trunk/webrtc/api/audio_mixer_api_gn", 11 "/media/webrtc/trunk/webrtc/api/call_api_gn", 12 "/media/webrtc/trunk/webrtc/api/transport_api_gn", 13 "/media/webrtc/trunk/webrtc/api/video_frame_api_gn", 14 "/media/webrtc/trunk/webrtc/audio/audio_gn", 16 "/media/webrtc/trunk/webrtc/base/gtest_prod_gn", 18 "/media/webrtc/trunk/webrtc/base/rtc_numerics_gn", 19 "/media/webrtc/trunk/webrtc/base/rtc_task_queue_gn", 20 "/media/webrtc/trunk/webrtc/call/call_gn", 77 "/media/webrtc/trunk/webrtc/video/video_gn", [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capture_generic_gn/ |
H A D | moz.build | 20 FINAL_LIBRARY = "webrtc" 28 "/media/webrtc/trunk/webrtc/" 34 "/media/webrtc/trunk/webrtc/modules/desktop_capture/cropped_desktop_frame.cc", 38 "/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_capturer.cc", 41 "/media/webrtc/trunk/webrtc/modules/desktop_capture/desktop_device_info.cc", 44 "/media/webrtc/trunk/webrtc/modules/desktop_capture/differ_block.cc", 46 "/media/webrtc/trunk/webrtc/modules/desktop_capture/mouse_cursor.cc", 47 "/media/webrtc/trunk/webrtc/modules/desktop_capture/resolution_tracker.cc", 48 "/media/webrtc/trunk/webrtc/modules/desktop_capture/rgba_color.cc", 50 "/media/webrtc/trunk/webrtc/modules/desktop_capture/window_finder.cc" [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/gtest/ |
H A D | moz.build | 10 include('/media/webrtc/webrtc.mozbuild') 38 'webrtc', 143 # '../webrtc/call/call_unittest.cc', 420 '../webrtc/test/fake_decoder.cc', 421 '../webrtc/test/fake_encoder.cc', 423 '../webrtc/test/field_trial.cc', 424 '../webrtc/test/frame_generator.cc', 427 '../webrtc/test/frame_utils.cc', 429 '../webrtc/test/rtp_file_reader.cc', 433 '../webrtc/test/test_main.cc', [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/ |
H A D | moz.build | 10 "/third_party/libwebrtc/webrtc/api/array_view_gn", 31 "/third_party/libwebrtc/webrtc/api/call_api_gn", 32 "/third_party/libwebrtc/webrtc/api/optional_gn", 34 "/third_party/libwebrtc/webrtc/api/transport_api_gn", 38 "/third_party/libwebrtc/webrtc/audio/audio_gn", 41 "/third_party/libwebrtc/webrtc/call/call_gn", 45 "/third_party/libwebrtc/webrtc/call/rtp_sender_gn", 60 "/third_party/libwebrtc/webrtc/media/rtc_media_gn", 135 "/third_party/libwebrtc/webrtc/video/video_gn", 138 "/third_party/libwebrtc/webrtc/webrtc_common_gn", [all …]
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/ |
H A D | moz.build | 10 "/third_party/libwebrtc/webrtc/api/array_view_gn", 31 "/third_party/libwebrtc/webrtc/api/call_api_gn", 32 "/third_party/libwebrtc/webrtc/api/optional_gn", 34 "/third_party/libwebrtc/webrtc/api/transport_api_gn", 38 "/third_party/libwebrtc/webrtc/audio/audio_gn", 41 "/third_party/libwebrtc/webrtc/call/call_gn", 45 "/third_party/libwebrtc/webrtc/call/rtp_sender_gn", 60 "/third_party/libwebrtc/webrtc/media/rtc_media_gn", 135 "/third_party/libwebrtc/webrtc/video/video_gn", 138 "/third_party/libwebrtc/webrtc/webrtc_common_gn", [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/common_audio/common_audio_c_gn/ |
H A D | moz.build | 26 "/media/webrtc/trunk/webrtc/", 27 "/media/webrtc/trunk/webrtc/common_audio/resampler/include/", 28 "/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/", 29 "/media/webrtc/trunk/webrtc/common_audio/vad/include/" 33 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c", 34 "/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c" 38 "/media/webrtc/trunk/webrtc/common_audio/fft4g.c", 39 "/media/webrtc/trunk/webrtc/common_audio/ring_buffer.c", 70 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c", 71 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_gmm.c", [all …]
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/ |
H A D | WATCHLISTS | 18 # NOTE: if you like this you might like webrtc-reviews@webrtc.org! 19 'filepath': 'webrtc/.*', 22 # webrtc/build/ and non-recursive contents of ./ and webrtc/ 23 'filepath': '^[^/]*$|webrtc/[^/]*$|webrtc/build/.*', 26 'filepath': 'webrtc/[^/]*\.h$|'\ 37 'webrtc/video/.*', 92 'root_files': ['andrew@webrtc.org', 96 'peah@webrtc.org'], 102 'peah@webrtc.org' 127 'minyue@webrtc.org'], [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/common_audio/common_audio_c_gn/ |
H A D | moz.build | 19 FINAL_LIBRARY = "webrtc" 27 "/media/webrtc/trunk/webrtc/common_audio/resampler/include/", 28 "/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/", 29 "/media/webrtc/trunk/webrtc/common_audio/vad/include/" 33 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_core.c", 34 "/media/webrtc/trunk/webrtc/common_audio/vad/webrtc_vad.c" 38 "/media/webrtc/trunk/webrtc/common_audio/fft4g.c", 39 "/media/webrtc/trunk/webrtc/common_audio/ring_buffer.c", 71 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_filterbank.c", 72 "/media/webrtc/trunk/webrtc/common_audio/vad/vad_gmm.c", [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_gn/ |
H A D | moz.build | 21 FINAL_LIBRARY = "webrtc" 28 "/media/webrtc/trunk/webrtc/", 29 "/media/webrtc/trunk/webrtc/common_audio/resampler/include/", 30 "/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/", 31 "/media/webrtc/trunk/webrtc/common_audio/vad/include/", 37 "/media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.cc", 47 "/media/webrtc/trunk/webrtc/modules/audio_processing/rms_level.cc" 51 "/media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core.cc", 55 "/media/webrtc/trunk/webrtc/modules/audio_processing/aec3/aec3_fft.cc", 86 "/media/webrtc/trunk/webrtc/modules/audio_processing/agc/agc.cc", [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc_overrides/ |
H A D | BUILD.gn | 14 "//third_party/webrtc/api:callfactory_api", 21 "//third_party/webrtc/api:rtc_error", 22 "//third_party/webrtc/api:rtc_stats_api", 25 "//third_party/webrtc/api:scoped_refptr", 53 "//third_party/webrtc/common_video", 74 "//third_party/webrtc/p2p:rtc_p2p", 77 "//third_party/webrtc/pc:rtc_pc", 79 "//third_party/webrtc/rtc_base", 91 "//third_party/webrtc/stats", 119 # in https://cs.chromium.org/chromium/src/third_party/webrtc/webrtc.gni [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/blink/renderer/modules/peerconnection/ |
H A D | mock_peer_connection_impl.h | 42 webrtc::RTCError SetParameters( 163 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 167 return webrtc::RTCErrorOr< in AddTransceiver() 170 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 174 return webrtc::RTCErrorOr< in AddTransceiver() 178 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 181 return webrtc::RTCErrorOr< in AddTransceiver() 184 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 188 return webrtc::RTCErrorOr< in AddTransceiver() 199 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack( [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/blink/renderer/modules/peerconnection/ |
H A D | mock_peer_connection_impl.h | 44 webrtc::RTCError SetParameters( 165 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 169 return webrtc::RTCErrorOr< in AddTransceiver() 172 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 176 return webrtc::RTCErrorOr< in AddTransceiver() 180 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 183 return webrtc::RTCErrorOr< in AddTransceiver() 186 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> 190 return webrtc::RTCErrorOr< in AddTransceiver() 201 webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack( [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc_overrides/ |
H A D | BUILD.gn | 14 "//third_party/webrtc/api:callfactory_api", 20 "//third_party/webrtc/api:rtc_error", 21 "//third_party/webrtc/api:rtc_stats_api", 23 "//third_party/webrtc/api:scoped_refptr", 48 "//third_party/webrtc/common_video", 68 "//third_party/webrtc/p2p:rtc_p2p", 71 "//third_party/webrtc/pc:rtc_pc", 73 "//third_party/webrtc/rtc_base", 85 "//third_party/webrtc/stats", 108 # in https://cs.chromium.org/chromium/src/third_party/webrtc/webrtc.gni [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/ilbc_c_gn/ |
H A D | moz.build | 19 FINAL_LIBRARY = "webrtc" 26 "/media/webrtc/trunk/webrtc/", 27 "/media/webrtc/trunk/webrtc/common_audio/resampler/include/", 28 "/media/webrtc/trunk/webrtc/common_audio/signal_processing/include/", 29 "/media/webrtc/trunk/webrtc/common_audio/vad/include/" 33 "/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c", 36 "/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c", 48 "/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/decode.c", 51 "/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c", 52 "/media/webrtc/trunk/webrtc/modules/audio_coding/codecs/ilbc/encode.c", [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/media/engine/ |
H A D | fake_webrtc_call.h | 71 webrtc::AudioSendStream::Stats GetStats( 76 webrtc::AudioSendStream::Config config_; 77 webrtc::AudioSendStream::Stats stats_; 133 : public webrtc::VideoSendStream, 190 webrtc::VideoCodecVP8 vp8; 191 webrtc::VideoCodecVP9 vp9; 192 webrtc::VideoCodecH264 h264; 199 webrtc::VideoSendStream::Stats stats_; 283 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 311 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/media/engine/ |
H A D | fake_webrtc_call.h | 72 webrtc::AudioSendStream::Stats GetStats( 77 webrtc::AudioSendStream::Config config_; 78 webrtc::AudioSendStream::Stats stats_; 135 : public webrtc::VideoSendStream, 196 webrtc::VideoCodecVP8 vp8; 197 webrtc::VideoCodecVP9 vp9; 198 webrtc::VideoCodecH264 h264; 205 webrtc::VideoSendStream::Stats stats_; 289 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 317 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; [all …]
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/dports/net-im/tg_owt/tg_owt-d578c76/src/media/engine/ |
H A D | fake_webrtc_call.h | 73 webrtc::AudioSendStream::Stats GetStats( 78 webrtc::AudioSendStream::Config config_; 79 webrtc::AudioSendStream::Stats stats_; 137 : public webrtc::VideoSendStream, 198 webrtc::VideoCodecVP8 vp8; 199 webrtc::VideoCodecVP9 vp9; 200 webrtc::VideoCodecH264 h264; 207 webrtc::VideoSendStream::Stats stats_; 282 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 314 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; [all …]
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/media/engine/ |
H A D | fake_webrtc_call.h | 71 webrtc::AudioSendStream::Stats GetStats( 76 webrtc::AudioSendStream::Config config_; 77 webrtc::AudioSendStream::Stats stats_; 134 : public webrtc::VideoSendStream, 195 webrtc::VideoCodecVP8 vp8; 196 webrtc::VideoCodecVP9 vp9; 197 webrtc::VideoCodecH264 h264; 204 webrtc::VideoSendStream::Stats stats_; 288 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 316 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/media/engine/ |
H A D | fakewebrtccall.h | 68 webrtc::AudioSendStream::Stats GetStats( 73 webrtc::AudioSendStream::Config config_; 74 webrtc::AudioSendStream::Stats stats_; 120 : public webrtc::VideoSendStream, 177 webrtc::VideoCodecVP8 vp8; 178 webrtc::VideoCodecVP9 vp9; 185 webrtc::VideoSendStream::Stats stats_; 234 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 258 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; 312 webrtc::Call::Config config_; [all …]
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/media/engine/ |
H A D | fakewebrtccall.h | 68 webrtc::AudioSendStream::Stats GetStats( 73 webrtc::AudioSendStream::Config config_; 74 webrtc::AudioSendStream::Stats stats_; 120 : public webrtc::VideoSendStream, 177 webrtc::VideoCodecVP8 vp8; 178 webrtc::VideoCodecVP9 vp9; 185 webrtc::VideoSendStream::Stats stats_; 234 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 258 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; 312 webrtc::Call::Config config_; [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/media/engine/ |
H A D | fakewebrtccall.h | 68 webrtc::AudioSendStream::Stats GetStats( 73 webrtc::AudioSendStream::Config config_; 74 webrtc::AudioSendStream::Stats stats_; 120 : public webrtc::VideoSendStream, 177 webrtc::VideoCodecVP8 vp8; 178 webrtc::VideoCodecVP9 vp9; 185 webrtc::VideoSendStream::Stats stats_; 234 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 258 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; 312 webrtc::Call::Config config_; [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/media/engine/ |
H A D | fakewebrtccall.h | 66 webrtc::AudioSendStream::Config config_; 67 webrtc::AudioSendStream::Stats stats_; 107 : public webrtc::VideoSendStream, 163 webrtc::VideoCodecVP8 vp8; 164 webrtc::VideoCodecVP9 vp9; 170 webrtc::VideoSendStream::Stats stats_; 219 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { 224 webrtc::Call::Config GetConfig() const; 243 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; 292 webrtc::Call::Config config_; [all …]
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/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/voice_engine/test/auto_test/standard/ |
H A D | audio_processing_test.cc | 39 webrtc::AgcModes agc_mode = webrtc::kAgcDefault; in TryEnablingAgcWithMode() 50 webrtc::AgcModes agc_mode = webrtc::kAgcDefault; in TryEnablingRxAgcWithMode() 64 webrtc::EcModes ec_mode = webrtc::kEcDefault; in TryEnablingEcWithMode() 90 webrtc::NsModes ns_mode = webrtc::kNsDefault; in TryEnablingNsWithMode() 102 webrtc::NsModes ns_mode = webrtc::kNsDefault; in TryEnablingRxNsWithMode() 160 webrtc::EcModes ec_mode = webrtc::kEcDefault; in TEST_F() 181 webrtc::EcModes ec_mode = webrtc::kEcDefault; in TEST_F() 195 webrtc::EcModes ec_mode = webrtc::kEcDefault; in TEST_F() 272 webrtc::NsModes ns_mode = webrtc::kNsDefault; in TEST_F() 299 webrtc::NsModes ns_mode = webrtc::kNsDefault; in TEST_F() [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/rtc_base/rtc_base_approved_generic_gn/ |
H A D | moz.build | 26 "/media/webrtc/trunk/webrtc/" 32 "/media/webrtc/trunk/webrtc/rtc_base/base64.cc", 33 "/media/webrtc/trunk/webrtc/rtc_base/bitbuffer.cc", 37 "/media/webrtc/trunk/webrtc/rtc_base/checks.cc", 40 "/media/webrtc/trunk/webrtc/rtc_base/event.cc", 42 "/media/webrtc/trunk/webrtc/rtc_base/file.cc", 43 "/media/webrtc/trunk/webrtc/rtc_base/flags.cc", 45 "/media/webrtc/trunk/webrtc/rtc_base/location.cc", 46 "/media/webrtc/trunk/webrtc/rtc_base/logging.cc", 57 "/media/webrtc/trunk/webrtc/rtc_base/random.cc", [all …]
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