| /external/webrtc/src/system_wrappers/source/ |
| critical_section_win.h | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
18 namespace webrtc { namespace
34 } // namespace webrtc
|
| data_log_helpers_unittest.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
16 using ::webrtc::DataLog;
20 webrtc::ValueContainer<int> v1(c);
22 webrtc::ValueContainer<int> v2(c);
35 webrtc::ValueContainer<double> v1(c);
37 webrtc::ValueContainer<double> v2(c);
51 webrtc::MultiValueContainer<int> m1(a, 3);
52 webrtc::MultiValueContainer<int> m2(b, 3);
53 webrtc::MultiValueContainer<int> m3(a, 3);
|
| data_log_c.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
22 return webrtc::DataLog::CreateLog();
26 return webrtc::DataLog::ReturnLog();
32 std::string combined = webrtc::DataLog::Combine(table_name, table_id);
41 return webrtc::DataLog::AddTable(table_name);
48 return webrtc::DataLog::AddColumn(table_name, column_name,
56 return webrtc::DataLog::InsertCell(table_name, column_name, value);
64 return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
71 return webrtc::DataLog::InsertCell(table_name, column_name, value);
79 return webrtc::DataLog::InsertCell(table_name, column_name, values, length)
[all...] |
| cpu_wrapper_unittest.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
20 using webrtc::CpuInfo;
21 using webrtc::CpuWrapper;
22 using webrtc::EventWrapper;
23 using webrtc::scoped_ptr;
24 using webrtc::Trace;
27 // http://code.google.com/p/webrtc/issues/detail?id=290
35 std::string trace_file = webrtc::test::OutputPath() +
38 Trace::SetLevelFilter(webrtc::kTraceAll);
|
| /external/webrtc/test/ |
| run_all_unittests.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
14 webrtc::test::TestSuite test_suite(argc, argv);
|
| test_suite.h | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
22 namespace webrtc { namespace
40 } // namespace webrtc
|
| /external/chromium_org/remoting/codec/ |
| video_decoder_verbatim.cc | 16 void VideoDecoderVerbatim::Initialize(const webrtc::DesktopSize& screen_size) {
30 webrtc::DesktopRegion region;
36 webrtc::DesktopRect rect =
37 webrtc::DesktopRect::MakeXYWH(proto_rect.x(),
43 if (!DoesRectContain(webrtc::DesktopRect::MakeSize(screen_size_), rect)) {
72 void VideoDecoderVerbatim::Invalidate(const webrtc::DesktopSize& view_size,
73 const webrtc::DesktopRegion& region) {
77 void VideoDecoderVerbatim::RenderFrame(const webrtc::DesktopSize& view_size,
78 const webrtc::DesktopRect& clip_area,
81 webrtc::DesktopRegion* output_region)
[all...] |
| video_encoder_verbatim.h | 10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
23 const webrtc::DesktopFrame& frame) OVERRIDE;
31 webrtc::DesktopSize screen_size_;
|
| /external/chromium_org/third_party/libjingle/source/talk/examples/peerconnection/client/ |
| conductor.h | 39 #include "talk/app/webrtc/mediastreaminterface.h"
40 #include "talk/app/webrtc/peerconnectioninterface.h"
43 namespace webrtc { namespace
45 } // namespace webrtc
52 : public webrtc::PeerConnectionObserver,
53 public webrtc::CreateSessionDescriptionObserver,
85 webrtc::PeerConnectionObserver::StateType state_changed) {}
86 virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
87 virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream);
90 virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate)
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| fakewebrtcvideocapturemodule.h | 32 #include "talk/media/webrtc/fakewebrtcdeviceinfo.h"
33 #include "talk/media/webrtc/webrtcvideocapturer.h"
37 // Fake class for mocking out webrtc::VideoCaptureModule.
38 class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
63 webrtc::VideoCaptureDataCallback& callback) {
72 webrtc::VideoCaptureFeedBack& callback) {
79 const webrtc::VideoCaptureCapability& cap) {
96 webrtc::VideoCaptureCapability& settings) {
109 webrtc::VideoCaptureRotation rotation) {
113 const webrtc::VideoCodec& codec)
[all...] |
| fakewebrtcvoiceengine.h | 41 #include "talk/media/webrtc/fakewebrtccommon.h"
42 #include "talk/media/webrtc/webrtcvoe.h"
43 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
44 #include "webrtc/common.h"
64 : public webrtc::VoEAudioProcessing,
65 public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
66 public webrtc::VoEFile, public webrtc::VoEHardware
[all...] |
| /external/chromium_org/remoting/host/ |
| screen_resolution.cc | 15 : dimensions_(webrtc::DesktopSize(0, 0)),
16 dpi_(webrtc::DesktopVector(0, 0)) {
19 ScreenResolution::ScreenResolution(const webrtc::DesktopSize& dimensions,
20 const webrtc::DesktopVector& dpi)
24 DCHECK(!dimensions.is_empty() || dimensions.equals(webrtc::DesktopSize()));
29 webrtc::DesktopSize ScreenResolution::ScaleDimensionsToDpi(
30 const webrtc::DesktopVector& new_dpi) const {
39 return webrtc::DesktopSize(width, height);
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
| RTCMediaSource+Internal.h | 30 #include "talk/app/webrtc/mediastreaminterface.h"
35 talk_base::scoped_refptr<webrtc::MediaSourceInterface> mediaSource;
38 (talk_base::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource;
|
| RTCMediaStream+Internal.h | 30 #include "talk/app/webrtc/mediastreamtrack.h"
35 talk_base::scoped_refptr<webrtc::MediaStreamInterface> mediaStream;
38 (talk_base::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
|
| RTCMediaStreamTrack+Internal.h | 30 #include "talk/app/webrtc/mediastreaminterface.h"
35 talk_base::scoped_refptr<webrtc::MediaStreamTrackInterface> mediaTrack;
38 (talk_base::scoped_refptr<webrtc::MediaStreamTrackInterface>)mediaTrack;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| mockpeerconnectionobservers.h | 35 #include "talk/app/webrtc/datachannelinterface.h"
37 namespace webrtc { namespace
40 : public webrtc::CreateSessionDescriptionObserver {
68 : public webrtc::SetSessionDescriptionObserver {
90 class MockDataChannelObserver : public webrtc::DataChannelObserver {
92 explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel)
110 talk_base::scoped_refptr<webrtc::DataChannelInterface> channel_;
115 class MockStatsObserver : public webrtc::StatsObserver {
120 virtual void OnComplete(const std::vector<webrtc::StatsReport>& reports) {
130 webrtc::StatsReport::kStatsValueNameAudioOutputLevel)
[all...] |
| peerconnectiontestwrapper.cc | 28 #include "talk/app/webrtc/fakeportallocatorfactory.h"
29 #include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
30 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
31 #include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
32 #include "talk/app/webrtc/videosourceinterface.h"
42 using webrtc::FakeConstraints;
43 using webrtc::FakeVideoTrackRenderer;
44 using webrtc::IceCandidateInterface;
45 using webrtc::MediaConstraintsInterface;
46 using webrtc::MediaStreamInterface
[all...] |
| /external/chromium_org/content/renderer/media/webrtc/ |
| webrtc_video_sink_adapter.h | 11 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
20 // webrtc:::VideoTrackInterface and a content::MediaStreamVideoSink.
24 : NON_EXPORTED_BASE(public webrtc::VideoRendererInterface),
25 NON_EXPORTED_BASE(public webrtc::ObserverInterface),
28 WebRtcVideoSinkAdapter(webrtc::VideoTrackInterface* video_track,
35 // webrtc::VideoRendererInterface implementation. May be called on
40 // webrtc::ObserverInterface implementation.
42 // for both WebRtc Audio and Video tracks.
51 scoped_refptr<webrtc::VideoTrackInterface> video_track_;
52 webrtc::MediaStreamTrackInterface::TrackState state_
[all...] |
| /external/chromium_org/third_party/libjingle/overrides/ |
| init_webrtc.h | 10 #include "third_party/webrtc/system_wrappers/interface/event_tracer.h"
20 namespace webrtc { namespace
22 } // namespace webrtc
25 webrtc::AudioDeviceModule* adm,
26 webrtc::AudioDeviceModule* adm_sc,
38 // as well as provide pointers back to a couple webrtc factory functions.
48 webrtc::GetCategoryEnabledPtr trace_get_category_enabled,
49 webrtc::AddTraceEventPtr trace_add_trace_event,
55 // Load and initialize the shared WebRTC module (libpeerconnection).
56 // Call this explicitly to load and initialize the WebRTC module (e.g. befor
[all...] |
| /external/chromium_org/content/browser/renderer_host/media/ |
| desktop_capture_device.cc | 18 #include "third_party/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h"
19 #include "third_party/webrtc/modules/desktop_capture/desktop_capture_options.h"
20 #include "third_party/webrtc/modules/desktop_capture/desktop_capturer.h"
21 #include "third_party/webrtc/modules/desktop_capture/desktop_frame.h"
22 #include "third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor.h"
23 #include "third_party/webrtc/modules/desktop_capture/screen_capturer.h"
24 #include "third_party/webrtc/modules/desktop_capture/window_capturer.h"
36 webrtc::DesktopRect ComputeLetterboxRect(
37 const webrtc::DesktopSize& max_size,
38 const webrtc::DesktopSize& source_size)
[all...] |
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/ |
| rtp_parser.cc | 32 if (rtp_header->webrtc.header.payloadType == parser_config_.payload_type &&
33 rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
60 rtp_header->webrtc.header.markerBit = marker;
61 rtp_header->webrtc.header.payloadType = payload_type;
62 rtp_header->webrtc.header.sequenceNumber = sequence_number;
63 rtp_header->webrtc.header.timestamp = rtp_timestamp;
64 rtp_header->webrtc.header.ssrc = ssrc;
65 rtp_header->webrtc.header.numCSRCs = cc;
68 rtp_header->webrtc.type.Audio.numEnergy = rtp_header->webrtc.header.numCSRCs
[all...] |
| /external/chromium_org/content/renderer/media/ |
| media_stream_audio_processor.h | 14 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
15 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
16 #include "third_party/webrtc/modules/interface/module_common_types.h"
24 namespace webrtc { namespace
30 // This class owns an object of webrtc::AudioProcessing which contains signal
37 const webrtc::MediaConstraintsInterface* constraints);
44 // Push the render audio to webrtc::AudioProcessing for analysis. This is
63 // webrtc::AudioProcessing to help processing the data.
84 // Helper to initialize the WebRtc AudioProcessing.
86 const webrtc::MediaConstraintsInterface* constraints)
[all...] |
| peer_connection_handler_base.cc | 29 const webrtc::MediaConstraintsInterface* constraints) {
30 webrtc::MediaStreamInterface* native_stream =
39 webrtc::MediaStreamInterface* native_stream =
|
| peer_connection_identity_service.h | 12 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
17 // This class is associated with a peer connection and handles WebRTC DTLS
20 : public webrtc::DTLSIdentityServiceInterface {
26 // webrtc::DTLSIdentityServiceInterface implementation.
30 webrtc::DTLSIdentityRequestObserver* observer) OVERRIDE;
41 talk_base::scoped_refptr<webrtc::DTLSIdentityRequestObserver>
|
| /external/chromium_org/remoting/client/ |
| frame_consumer_proxy.cc | 10 #include "third_party/webrtc/modules/desktop_capture/desktop_frame.h"
11 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
12 #include "third_party/webrtc/modules/desktop_capture/desktop_region.h"
24 void FrameConsumerProxy::ApplyBuffer(const webrtc::DesktopSize& view_size,
25 const webrtc::DesktopRect& clip_area,
26 webrtc::DesktopFrame* buffer,
27 const webrtc::DesktopRegion& region) {
39 void FrameConsumerProxy::ReturnBuffer(webrtc::DesktopFrame* buffer) {
51 const webrtc::DesktopSize& source_size,
52 const webrtc::DesktopVector& source_dpi)
[all...] |
