| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
| 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::PeerConnectionInterface::IceGatheringState)nativeState
[all...] |
| RTCSessionDescription+Internal.h | 30 #include "talk/app/webrtc/jsep.h"
31 #include "talk/app/webrtc/webrtcsession.h"
36 - (webrtc::SessionDescriptionInterface *)sessionDescription;
39 (const webrtc::SessionDescriptionInterface*)sessionDescription;
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideodecoderfactory.h | 32 #include "webrtc/common_types.h"
34 namespace webrtc { namespace
44 virtual webrtc::VideoDecoder* CreateVideoDecoder(
45 webrtc::VideoCodecType type) = 0;
48 virtual void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) = 0;
|
| webrtcvie.h | 33 #include "talk/media/webrtc/webrtccommon.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_engine/include/vie_codec.h"
42 #include "webrtc/video_engine/include/vie_errors.h
[all...] |
| webrtcvoe.h | 33 #include "talk/media/webrtc/webrtccommon.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_engine/include/voe_external_media.h"
43 #include "webrtc/voice_engine/include/voe_file.h
[all...] |
| /external/webrtc/src/system_wrappers/interface/ |
| sleep.h | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
15 namespace webrtc { namespace
22 } // namespace webrtc
|
| /external/chromium_org/remoting/codec/ |
| 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;
|
| 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::DesktopRegion updated_region_
[all...] |
| video_decoder_vpx.h | 12 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
13 #include "third_party/webrtc/modules/desktop_capture/desktop_region.h"
28 virtual void Initialize(const webrtc::DesktopSize& screen_size) OVERRIDE;
30 virtual void Invalidate(const webrtc::DesktopSize& view_size,
31 const webrtc::DesktopRegion& region) OVERRIDE;
32 virtual void RenderFrame(const webrtc::DesktopSize& view_size,
33 const webrtc::DesktopRect& clip_area,
36 webrtc::DesktopRegion* output_region) OVERRIDE;
37 virtual const webrtc::DesktopRegion* GetImageShape() OVERRIDE;
45 void UpdateImageShapeRegion(webrtc::DesktopRegion* new_desktop_shape)
[all...] |
| video_encoder.h | 10 namespace webrtc { namespace
12 } // namespace webrtc
25 virtual scoped_ptr<VideoPacket> Encode(const webrtc::DesktopFrame& frame) = 0;
|
| /external/chromium_org/remoting/host/ |
| 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_;
|
| 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::DesktopVector(5, 5))))
[all...] |
| /external/chromium_org/remoting/client/plugin/ |
| 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) OVERRIDE;
52 virtual void ReturnBuffer(webrtc::DesktopFrame* buffer) OVERRIDE;
53 virtual void SetSourceSize(const webrtc::DesktopSize& source_size
[all...] |
| /external/chromium_org/remoting/client/ |
| frame_producer.h | 10 namespace webrtc { namespace
15 } // namespace webrtc
29 virtual void DrawBuffer(webrtc::DesktopFrame* buffer) = 0;
34 virtual void InvalidateRegion(const webrtc::DesktopRegion& region) = 0;
42 virtual void SetOutputSizeAndClip(const webrtc::DesktopSize& view_size,
43 const webrtc::DesktopRect& clip_area) = 0;
46 virtual const webrtc::DesktopRegion* GetBufferShape() = 0;
|
| 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) = 0;
45 virtual void ReturnBuffer(webrtc::DesktopFrame* buffer) = 0;
48 virtual void SetSourceSize(const webrtc::DesktopSize& source_size,
49 const webrtc::DesktopVector& dpi) = 0;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| 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(talk_base::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::MediaStreamInterface* stream
[all...] |
| testsdpstrings.h | 33 namespace webrtc { namespace
81 "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"
145 } // namespace webrtc
|
| 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"
38 namespace webrtc { namespace
43 : public webrtc::PeerConnectionObserver,
44 public webrtc::CreateSessionDescriptionObserver,
53 bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
58 webrtc::PeerConnectionInterface::SignalingState new_state) {}
60 webrtc::PeerConnectionObserver::StateType state_changed) {
[all...] |
| /external/chromium_org/content/renderer/media/ |
| media_stream_track_extra_data.h | 14 namespace webrtc { namespace
16 } // namespace webrtc
23 MediaStreamTrackExtraData(webrtc::MediaStreamTrackInterface* track,
27 const scoped_refptr<webrtc::MediaStreamTrackInterface>& track() const {
33 scoped_refptr<webrtc::MediaStreamTrackInterface> track_;
|
| mock_peer_connection_impl.h | 14 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
21 class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface {
26 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
28 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
31 webrtc::MediaStreamInterface* local_stream,
32 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
34 webrtc::MediaStreamInterface* local_stream) OVERRIDE;
35 virtual talk_base::scoped_refptr<webrtc::DtmfSenderInterface>
36 CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE;
37 virtual talk_base::scoped_refptr<webrtc::DataChannelInterface
[all...] |
| /external/chromium_org/remoting/base/ |
| 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);
69 webrtc::DesktopRect AlignRect(const webrtc::DesktopRect& rect);
74 webrtc::DesktopRect ScaleRect(const webrtc::DesktopRect& rect
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/ |
| AudioSource.java | 28 package org.webrtc;
|
| AudioTrack.java | 28 package org.webrtc;
|
| StatsObserver.java | 28 package org.webrtc;
30 /** Interface for observing Stats reports (see webrtc::StatsObservers). */
|
| /external/webrtc/src/system_wrappers/source/ |
| cpu_no_op.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
15 namespace webrtc { namespace
22 } // namespace webrtc
|
