/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
H A D | peerconnectiontestwrapper.h | 31 #include "talk/app/webrtc/peerconnectioninterface.h" 32 #include "talk/app/webrtc/test/fakeaudiocapturemodule.h" 33 #include "talk/app/webrtc/test/fakeconstraints.h" 34 #include "talk/app/webrtc/test/fakevideotrackrenderer.h" 35 #include "webrtc/base/sigslot.h" 36 #include "webrtc/base/thread.h" 38 namespace webrtc { namespace 43 : public webrtc::PeerConnectionObserver, 44 public webrtc::CreateSessionDescriptionObserver, 53 bool CreatePc(const webrtc [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | Tester.cc | 16 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" 17 #include "webrtc/modules/audio_coding/main/test/APITest.h" 18 #include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h" 19 #include "webrtc/modules/audio_coding/main/test/iSACTest.h" 20 #include "webrtc/modules/audio_coding/main/test/opus_test.h" 21 #include "webrtc/modules/audio_coding/main/test/PacketLossTest.h" 22 #include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h" 23 #include "webrtc/modules/audio_coding/main/test/TestRedFec.h" 24 #include "webrtc/modules/audio_coding/main/test/TestStereo.h" 25 #include "webrtc/module [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
H A D | before_initialization_fixture.cc | 11 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h" 13 #include "webrtc/system_wrappers/interface/sleep.h" 16 : voice_engine_(webrtc::VoiceEngine::Create()) { 19 voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_); 20 voe_codec_ = webrtc::VoECodec::GetInterface(voice_engine_); 21 voe_volume_control_ = webrtc::VoEVolumeControl::GetInterface(voice_engine_); 22 voe_dtmf_ = webrtc::VoEDtmf::GetInterface(voice_engine_); 23 voe_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(voice_engine_); 24 voe_apm_ = webrtc::VoEAudioProcessing::GetInterface(voice_engine_); 25 voe_network_ = webrtc [all...] |
H A D | after_initialization_fixture.cc | 11 #include "webrtc/modules/audio_processing/include/audio_processing.h" 12 #include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h" 14 class TestErrorObserver : public webrtc::VoiceEngineObserver { 26 webrtc::Config config; 27 config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(false)); 28 webrtc::AudioProcessing* audioproc = webrtc::AudioProcessing::Create(config);
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | video_rtp_play_mt.cc | 13 #include "webrtc/modules/video_coding/main/test/receiver_tests.h" 14 #include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h" 15 #include "webrtc/system_wrappers/interface/event_wrapper.h" 16 #include "webrtc/system_wrappers/interface/thread_wrapper.h" 17 #include "webrtc/system_wrappers/interface/trace.h" 18 #include "webrtc/test/testsupport/fileutils.h" 20 using webrtc::rtpplayer::RtpPlayerInterface; 21 using webrtc::rtpplayer::VcmPayloadSinkFactory; 26 const webrtc::VCMVideoProtection kConfigProtectionMethod = 27 webrtc [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
H A D | audio_processing_test.cc | 11 #include "webrtc/test/testsupport/fileutils.h" 12 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" 13 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h" 15 class RxCallback : public webrtc::VoERxVadCallback { 35 void TryEnablingAgcWithMode(webrtc::AgcModes agc_mode_to_set) { 39 webrtc::AgcModes agc_mode = webrtc::kAgcDefault; 46 void TryEnablingRxAgcWithMode(webrtc::AgcModes agc_mode_to_set) { 50 webrtc::AgcModes agc_mode = webrtc [all...] |
/external/chromium_org/content/renderer/media/ |
H A D | mock_peer_connection_impl.h | 15 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" 22 class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { 27 virtual rtc::scoped_refptr<webrtc::StreamCollectionInterface> 29 virtual rtc::scoped_refptr<webrtc::StreamCollectionInterface> 32 webrtc::MediaStreamInterface* local_stream, 33 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; 35 webrtc::MediaStreamInterface* local_stream) OVERRIDE; 36 virtual rtc::scoped_refptr<webrtc::DtmfSenderInterface> 37 CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE; 38 virtual rtc::scoped_refptr<webrtc [all...] |
/external/chromium_org/remoting/client/ |
H A D | frame_consumer.h | 10 namespace webrtc { namespace 16 } // namespace webrtc 37 virtual void ApplyBuffer(const webrtc::DesktopSize& view_size, 38 const webrtc::DesktopRect& clip_area, 39 webrtc::DesktopFrame* buffer, 40 const webrtc::DesktopRegion& region, 41 const webrtc::DesktopRegion& shape) = 0; 46 virtual void ReturnBuffer(webrtc::DesktopFrame* buffer) = 0; 49 virtual void SetSourceSize(const webrtc::DesktopSize& source_size, 50 const webrtc [all...] |
/external/chromium_org/remoting/codec/ |
H A D | video_decoder_verbatim.h | 11 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 12 #include "third_party/webrtc/modules/desktop_capture/desktop_region.h" 26 virtual void Initialize(const webrtc::DesktopSize& screen_size) OVERRIDE; 28 virtual void Invalidate(const webrtc::DesktopSize& view_size, 29 const webrtc::DesktopRegion& region) OVERRIDE; 30 virtual void RenderFrame(const webrtc::DesktopSize& view_size, 31 const webrtc::DesktopRect& clip_area, 34 webrtc::DesktopRegion* output_region) OVERRIDE; 35 virtual const webrtc::DesktopRegion* GetImageShape() OVERRIDE; 39 webrtc [all...] |
/external/chromium_org/remoting/host/ |
H A D | fake_mouse_cursor_monitor.cc | 9 #include "third_party/webrtc/modules/desktop_capture/desktop_frame.h" 10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 11 #include "third_party/webrtc/modules/desktop_capture/mouse_cursor.h" 20 webrtc::MouseCursorMonitor::Callback* callback, 21 webrtc::MouseCursorMonitor::Mode mode) { 37 scoped_ptr<webrtc::DesktopFrame> desktop_frame( 38 new webrtc::BasicDesktopFrame(webrtc::DesktopSize(kWidth, kHeight))); 40 webrtc::DesktopFrame::kBytesPerPixel * kWidth * kHeight); 42 scoped_ptr<webrtc [all...] |
H A D | shaped_desktop_capturer.h | 9 #include "third_party/webrtc/modules/desktop_capture/desktop_capturer.h" 16 class ShapedDesktopCapturer : public webrtc::DesktopCapturer, 17 public webrtc::DesktopCapturer::Callback { 19 ShapedDesktopCapturer(scoped_ptr<webrtc::DesktopCapturer> screen_capturer, 23 // webrtc::DesktopCapturer interface. 24 virtual void Start(webrtc::DesktopCapturer::Callback* callback) OVERRIDE; 25 virtual void Capture(const webrtc::DesktopRegion& region) OVERRIDE; 28 // webrtc::DesktopCapturer::Callback interface. 29 virtual webrtc::SharedMemory* CreateSharedMemory(size_t size) OVERRIDE; 30 virtual void OnCaptureCompleted(webrtc [all...] |
H A D | desktop_shape_tracker_mac.cc | 8 #include "third_party/webrtc/modules/desktop_capture/desktop_capture_options.h" 13 webrtc::DesktopCaptureOptions options) {
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H A D | desktop_shape_tracker_x11.cc | 8 #include "third_party/webrtc/modules/desktop_capture/desktop_capture_options.h" 13 webrtc::DesktopCaptureOptions options) {
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
H A D | webrtcvideoengine2_unittest.h | 34 #include "webrtc/call.h" 35 #include "webrtc/video_receive_stream.h" 36 #include "webrtc/video_send_stream.h" 39 class FakeVideoSendStream : public webrtc::VideoSendStream { 41 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 42 const webrtc::VideoEncoderConfig& encoder_config); 43 webrtc::VideoSendStream::Config GetConfig(); 44 std::vector<webrtc::VideoStream> GetVideoStreams(); 47 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const; 50 virtual webrtc [all...] |
H A D | webrtcvie.h | 32 #include "talk/media/webrtc/webrtccommon.h" 33 #include "webrtc/base/common.h" 34 #include "webrtc/common_types.h" 35 #include "webrtc/modules/interface/module_common_types.h" 36 #include "webrtc/modules/video_capture/include/video_capture.h" 37 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" 38 #include "webrtc/modules/video_render/include/video_render.h" 39 #include "webrtc/video_engine/include/vie_base.h" 40 #include "webrtc/video_engine/include/vie_capture.h" 41 #include "webrtc/video_engin [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
H A D | RTCEnumConverter.h | 32 #include "talk/app/webrtc/peerconnectioninterface.h" 37 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState; 40 (webrtc::PeerConnectionInterface::IceGatheringState)nativeState; 43 (webrtc::PeerConnectionInterface::SignalingState)nativeState; 45 + (webrtc::PeerConnectionInterface::StatsOutputLevel) 49 (webrtc::MediaSourceInterface::SourceState)nativeState; 51 + (webrtc::MediaStreamTrackInterface::TrackState)convertTrackStateToNative: 55 (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
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H A D | RTCICECandidate+Internal.h | 30 #include "talk/app/webrtc/peerconnectioninterface.h" 35 webrtc::IceCandidateInterface* candidate; 37 - (id)initWithCandidate:(const webrtc::IceCandidateInterface*)candidate;
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H A D | RTCStatsReport+Internal.h | 30 #include "talk/app/webrtc/statstypes.h" 34 - (instancetype)initWithStatsReport:(const webrtc::StatsReport&)statsReport;
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H A D | RTCMediaConstraints.mm | 36 #include "webrtc/base/scoped_ptr.h" 42 rtc::scoped_ptr<webrtc::RTCMediaConstraintsNative> _constraints; 43 webrtc::MediaConstraintsInterface::Constraints _mandatory; 44 webrtc::MediaConstraintsInterface::Constraints _optional; 53 new webrtc::RTCMediaConstraintsNative(_mandatory, _optional)); 58 + (webrtc::MediaConstraintsInterface::Constraints)constraintsFromArray: 60 webrtc::MediaConstraintsInterface::Constraints constraints; 62 constraints.push_back(webrtc::MediaConstraintsInterface::Constraint( 72 - (const webrtc::RTCMediaConstraintsNative*)constraints {
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/external/chromium_org/remoting/client/jni/ |
H A D | jni_frame_consumer.h | 15 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 21 namespace webrtc { namespace 23 } // namespace webrtc 43 virtual void ApplyBuffer(const webrtc::DesktopSize& view_size, 44 const webrtc::DesktopRect& clip_area, 45 webrtc::DesktopFrame* buffer, 46 const webrtc::DesktopRegion& region, 47 const webrtc::DesktopRegion& shape) OVERRIDE; 48 virtual void ReturnBuffer(webrtc::DesktopFrame* buffer) OVERRIDE; 49 virtual void SetSourceSize(const webrtc [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/mock/ |
H A D | mock_packet_manipulator.h | 14 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h" 19 #include "webrtc/typedefs.h" 20 #include "webrtc/video_frame.h" 22 namespace webrtc { namespace 27 MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image)); 31 } // namespace webrtc
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/external/chromium_org/third_party/webrtc/test/ |
H A D | mock_transport.h | 15 #include "webrtc/transport.h" 17 namespace webrtc { namespace 19 class MockTransport : public webrtc::Transport { 26 } // namespace webrtc
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/external/chromium_org/remoting/protocol/ |
H A D | mouse_input_filter.h | 10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h" 25 void set_input_size(const webrtc::DesktopSize& size); 28 void set_output_size(const webrtc::DesktopSize& size); 34 webrtc::DesktopSize input_max_; 35 webrtc::DesktopSize output_max_;
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/ |
H A D | CameraDesc.java | 11 package org.webrtc.webrtcdemo;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | background_noise_unittest.cc | 13 #include "webrtc/modules/audio_coding/neteq/background_noise.h" 17 namespace webrtc { namespace 26 } // namespace webrtc
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