Searched refs:webrtc (Results 51 - 75 of 2804) sorted by relevance

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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/
H A Dpeerconnectiontestwrapper.h31 #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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DTester.cc16 #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
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/
H A Dbefore_initialization_fixture.cc11 #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
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H A Dafter_initialization_fixture.cc11 #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);
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dvideo_rtp_play_mt.cc13 #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
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/
H A Daudio_processing_test.cc11 #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
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/external/chromium_org/content/renderer/media/
H A Dmock_peer_connection_impl.h15 #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
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/external/chromium_org/remoting/client/
H A Dframe_consumer.h10 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
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/external/chromium_org/remoting/codec/
H A Dvideo_decoder_verbatim.h11 #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
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/external/chromium_org/remoting/host/
H A Dfake_mouse_cursor_monitor.cc9 #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
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H A Dshaped_desktop_capturer.h9 #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
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H A Ddesktop_shape_tracker_mac.cc8 #include "third_party/webrtc/modules/desktop_capture/desktop_capture_options.h"
13 webrtc::DesktopCaptureOptions options) {
H A Ddesktop_shape_tracker_x11.cc8 #include "third_party/webrtc/modules/desktop_capture/desktop_capture_options.h"
13 webrtc::DesktopCaptureOptions options) {
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
H A Dwebrtcvideoengine2_unittest.h34 #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
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H A Dwebrtcvie.h32 #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
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/
H A DRTCEnumConverter.h32 #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;
H A DRTCICECandidate+Internal.h30 #include "talk/app/webrtc/peerconnectioninterface.h"
35 webrtc::IceCandidateInterface* candidate;
37 - (id)initWithCandidate:(const webrtc::IceCandidateInterface*)candidate;
H A DRTCStatsReport+Internal.h30 #include "talk/app/webrtc/statstypes.h"
34 - (instancetype)initWithStatsReport:(const webrtc::StatsReport&)statsReport;
H A DRTCMediaConstraints.mm36 #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 {
/external/chromium_org/remoting/client/jni/
H A Djni_frame_consumer.h15 #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
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/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test/mock/
H A Dmock_packet_manipulator.h14 #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
/external/chromium_org/third_party/webrtc/test/
H A Dmock_transport.h15 #include "webrtc/transport.h"
17 namespace webrtc { namespace
19 class MockTransport : public webrtc::Transport {
26 } // namespace webrtc
/external/chromium_org/remoting/protocol/
H A Dmouse_input_filter.h10 #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_;
/external/chromium_org/third_party/webrtc/examples/android/media_demo/src/org/webrtc/webrtcdemo/
H A DCameraDesc.java11 package org.webrtc.webrtcdemo;
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Dbackground_noise_unittest.cc13 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
17 namespace webrtc { namespace
26 } // namespace webrtc

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