/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_neteq_unittest.cc | 13 namespace webrtc { namespace
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/external/chromium_org/third_party/webrtc/common_audio/ |
H A D | wav_header_unittest.cc | 14 #include "webrtc/common_audio/wav_header.h" 15 #include "webrtc/system_wrappers/interface/compile_assert.h" 21 EXPECT_TRUE(webrtc::CheckWavParameters(1, 8000, webrtc::kWavFormatPcm, 1, 0)); 23 webrtc::CheckWavParameters(0, 8000, webrtc::kWavFormatPcm, 1, 0)); 25 webrtc::CheckWavParameters(-1, 8000, webrtc::kWavFormatPcm, 1, 0)); 26 EXPECT_FALSE(webrtc::CheckWavParameters(1, 0, webrtc [all...] |
/external/chromium_org/third_party/webrtc/system_wrappers/source/ |
H A D | logcat_trace_context.cc | 11 #include "webrtc/system_wrappers/interface/logcat_trace_context.h" 16 #include "webrtc/system_wrappers/interface/logging.h" 18 namespace webrtc { namespace 23 // to DEBUG because they are highly verbose in webrtc code (which is 26 case webrtc::kTraceStateInfo: return ANDROID_LOG_DEBUG; 27 case webrtc::kTraceWarning: return ANDROID_LOG_WARN; 28 case webrtc::kTraceError: return ANDROID_LOG_ERROR; 29 case webrtc::kTraceCritical: return ANDROID_LOG_FATAL; 30 case webrtc::kTraceApiCall: return ANDROID_LOG_VERBOSE; 31 case webrtc [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_device/dummy/ |
H A D | audio_device_utility_dummy.cc | 10 #include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h" 12 namespace webrtc { namespace 14 } // namespace webrtc
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/external/chromium_org/third_party/webrtc/modules/video_capture/ |
H A D | ensure_initialized.h | 11 namespace webrtc { namespace 14 // Ensure any necessary initialization of webrtc::videocapturemodule has 19 } // namespace webrtc.
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/external/chromium_org/third_party/webrtc/test/testsupport/ |
H A D | always_passing_unittest.cc | 13 namespace webrtc { namespace 19 } // namespace webrtc
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/external/chromium_org/remoting/host/ |
H A D | screen_resolution.h | 10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 18 ScreenResolution(const webrtc::DesktopSize& dimensions, 19 const webrtc::DesktopVector& dpi); 22 webrtc::DesktopSize ScaleDimensionsToDpi( 23 const webrtc::DesktopVector& new_dpi) const; 26 const webrtc::DesktopSize& dimensions() const { return dimensions_; } 29 const webrtc::DesktopVector& dpi() const { return dpi_; } 39 webrtc::DesktopSize dimensions_; 40 webrtc::DesktopVector dpi_;
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H A D | screen_resolution_unittest.cc | 16 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10)); 20 webrtc::DesktopSize(), webrtc::DesktopVector(10, 10)); 24 webrtc::DesktopSize(1, 1), webrtc::DesktopVector(0, 0)); 30 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10)); 32 EXPECT_TRUE(webrtc::DesktopSize(50, 50).equals( 33 resolution.ScaleDimensionsToDpi(webrtc [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/ |
H A D | tb_interfaces.h | 16 #include "webrtc/base/constructormagic.h" 17 #include "webrtc/common_types.h" 18 #include "webrtc/video_engine/include/vie_base.h" 19 #include "webrtc/video_engine/include/vie_capture.h" 20 #include "webrtc/video_engine/include/vie_codec.h" 21 #include "webrtc/video_engine/include/vie_image_process.h" 22 #include "webrtc/video_engine/include/vie_network.h" 23 #include "webrtc/video_engine/include/vie_render.h" 24 #include "webrtc/video_engine/include/vie_rtp_rtcp.h" 25 #include "webrtc/video_engin [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/primitives/ |
H A D | base_primitives.h | 14 namespace webrtc { namespace 26 void TestI420CallSetup(webrtc::ViECodec* codec_interface, 27 webrtc::VideoEngine* video_engine, 28 webrtc::ViEBase* base_interface, 29 webrtc::ViENetwork* network_interface, 30 webrtc::ViERTP_RTCP* rtp_rtcp_interface,
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
H A D | before_initialization_fixture.h | 16 #include "webrtc/common.h" 17 #include "webrtc/common_types.h" 18 #include "webrtc/engine_configurations.h" 19 #include "webrtc/test/testsupport/gtest_disable.h" 20 #include "webrtc/voice_engine/include/voe_audio_processing.h" 21 #include "webrtc/voice_engine/include/voe_base.h" 22 #include "webrtc/voice_engine/include/voe_codec.h" 23 #include "webrtc/voice_engine/include/voe_dtmf.h" 24 #include "webrtc/voice_engine/include/voe_errors.h" 25 #include "webrtc/voice_engin [all...] |
/external/chromium_org/third_party/webrtc/modules/video_capture/mac/ |
H A D | video_capture_mac.mm | 18 #include "webrtc/modules/video_capture/device_info_impl.h" 19 #include "webrtc/modules/video_capture/video_capture_config.h" 20 #include "webrtc/modules/video_capture/video_capture_impl.h" 21 #include "webrtc/system_wrappers/interface/ref_count.h" 22 #include "webrtc/system_wrappers/interface/trace.h" 30 #include "webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit.h" 31 #include "webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.h" 34 namespace webrtc 50 WEBRTC_TRACE(webrtc::kTraceError, webrtc [all...] |
/external/chromium_org/remoting/client/plugin/ |
H A D | pepper_view.h | 19 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 20 #include "third_party/webrtc/modules/desktop_capture/desktop_region.h" 26 namespace webrtc { namespace 28 } // namespace webrtc 48 virtual void ApplyBuffer(const webrtc::DesktopSize& view_size, 49 const webrtc::DesktopRect& clip_area, 50 webrtc::DesktopFrame* buffer, 51 const webrtc::DesktopRegion& region, 52 const webrtc::DesktopRegion& shape) OVERRIDE; 53 virtual void ReturnBuffer(webrtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
H A D | RTCEnumConverter.mm | 30 #include "talk/app/webrtc/peerconnectioninterface.h" 35 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState { 37 case webrtc::PeerConnectionInterface::kIceConnectionNew: 39 case webrtc::PeerConnectionInterface::kIceConnectionChecking: 41 case webrtc::PeerConnectionInterface::kIceConnectionConnected: 43 case webrtc::PeerConnectionInterface::kIceConnectionCompleted: 45 case webrtc::PeerConnectionInterface::kIceConnectionFailed: 47 case webrtc::PeerConnectionInterface::kIceConnectionDisconnected: 49 case webrtc::PeerConnectionInterface::kIceConnectionClosed: 55 (webrtc [all...] |
/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/tools/ |
H A D | bwe_rtp.h | 16 namespace webrtc { namespace 29 webrtc::Clock* clock, 30 webrtc::RemoteBitrateObserver* observer, 31 webrtc::test::RtpFileReader** rtp_reader, 32 webrtc::RtpHeaderParser** parser, 33 webrtc::RemoteBitrateEstimator** estimator,
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H A D | bwe_rtp.cc | 11 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" 16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" 19 #include "webrtc/test/rtp_file_reader.h" 25 webrtc::Clock* clock, 26 webrtc::RemoteBitrateObserver* observer, 27 webrtc::test::RtpFileReader** rtp_reader, 28 webrtc::RtpHeaderParser** parser, 29 webrtc [all...] |
/external/chromium_org/third_party/webrtc/tools/force_mic_volume_max/ |
H A D | force_mic_volume_max.cc | 15 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 16 #include "webrtc/test/channel_transport/include/channel_transport.h" 17 #include "webrtc/voice_engine/include/voe_audio_processing.h" 18 #include "webrtc/voice_engine/include/voe_base.h" 19 #include "webrtc/voice_engine/include/voe_volume_control.h" 22 webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create(); 28 webrtc::VoEBase* base = webrtc::VoEBase::GetInterface(voe); 29 webrtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvoe.h | 32 #include "talk/media/webrtc/webrtccommon.h" 33 #include "webrtc/base/common.h" 35 #include "webrtc/common_types.h" 36 #include "webrtc/modules/audio_device/include/audio_device.h" 37 #include "webrtc/voice_engine/include/voe_audio_processing.h" 38 #include "webrtc/voice_engine/include/voe_base.h" 39 #include "webrtc/voice_engine/include/voe_codec.h" 40 #include "webrtc/voice_engine/include/voe_dtmf.h" 41 #include "webrtc/voice_engine/include/voe_errors.h" 42 #include "webrtc/voice_engin [all...] |
/external/chromium_org/remoting/codec/ |
H A D | video_decoder.h | 11 namespace webrtc { namespace 15 } // namespace webrtc 30 virtual void Initialize(const webrtc::DesktopSize& screen_size) = 0; 39 virtual void Invalidate(const webrtc::DesktopSize& view_size, 40 const webrtc::DesktopRegion& region) = 0; 54 virtual void RenderFrame(const webrtc::DesktopSize& view_size, 55 const webrtc::DesktopRect& clip_area, 58 webrtc::DesktopRegion* output_region) = 0; 62 virtual const webrtc::DesktopRegion* GetImageShape() = 0;
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/external/chromium_org/third_party/webrtc/modules/audio_device/test/android/audio_device_android_test/src/org/webrtc/voiceengine/ |
H A D | AudioDeviceAndroid.java | 1 ../../../../../../../source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
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/external/chromium_org/tools/perf/benchmarks/ |
H A D | webrtc.py | 5 from measurements import webrtc namespace 12 test = webrtc.WebRTC
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | fakemediastreamsignaling.h | 32 #include "talk/app/webrtc/audiotrack.h" 33 #include "talk/app/webrtc/mediastreamsignaling.h" 34 #include "talk/app/webrtc/videotrack.h" 44 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling, 45 public webrtc::MediaStreamSignalingObserver { 48 webrtc::MediaStreamSignaling(rtc::Thread::Current(), this, 89 virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) { 91 virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) { 93 virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) { 95 virtual void OnAddLocalAudioTrack(webrtc [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | video_rtp_play.cc | 11 #include "webrtc/modules/video_coding/main/test/receiver_tests.h" 12 #include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h" 13 #include "webrtc/system_wrappers/interface/trace.h" 14 #include "webrtc/test/testsupport/fileutils.h" 19 const webrtc::VCMVideoProtection kConfigProtectionMethod = 20 webrtc::kProtectionNack; 31 std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt"; 32 webrtc::Trace::CreateTrace(); 33 webrtc::Trace::SetTraceFile(trace_file.c_str()); 34 webrtc [all...] |
/external/chromium_org/remoting/base/ |
H A D | util.h | 11 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 45 const webrtc::DesktopSize& source_size, 46 const webrtc::DesktopRect& source_buffer_rect, 49 const webrtc::DesktopSize& dest_size, 50 const webrtc::DesktopRect& dest_buffer_rect, 51 const webrtc::DesktopRect& dest_rect); 56 webrtc::DesktopRect AlignRect(const webrtc::DesktopRect& rect); 61 webrtc::DesktopRect ScaleRect(const webrtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/ |
H A D | AudioSource.java | 28 package org.webrtc;
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