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  /external/webrtc/test/testsupport/
fileutils.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
30 namespace webrtc { namespace
167 } // namespace webrtc
  /external/chromium_org/content/test/
webrtc_audio_device_test.h 20 #include "third_party/webrtc/common_types.h"
34 namespace webrtc { namespace
58 // Scoped class for WebRTC interfaces. Fetches the wrapped interface
59 // in the constructor via WebRTC's GetInterface mechanism and then releases
87 // This is useful for some WebRTC objects that have their own Create/Delete
197 // A very basic implementation of webrtc::Transport that acts as a transport
198 // but just forwards all calls to a local webrtc::VoENetwork implementation.
200 class WebRTCTransportImpl : public webrtc::Transport {
202 explicit WebRTCTransportImpl(webrtc::VoENetwork* network);
209 webrtc::VoENetwork* network_
    [all...]
  /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/
datachannel.cc 27 #include "talk/app/webrtc/datachannel.h"
31 #include "talk/app/webrtc/webrtcsession.h"
35 namespace webrtc { namespace
421 } // namespace webrtc
localvideosource.cc 28 #include "talk/app/webrtc/localvideosource.h"
32 #include "talk/app/webrtc/mediaconstraintsinterface.h"
36 using webrtc::MediaConstraintsInterface;
37 using webrtc::MediaSourceInterface;
39 namespace webrtc { namespace
58 } // namespace webrtc
325 namespace webrtc { namespace
330 const webrtc::MediaConstraintsInterface* constraints) {
355 const webrtc::MediaConstraintsInterface* constraints) {
442 } // namespace webrtc
    [all...]
mediastreamhandler.cc 28 #include "talk/app/webrtc/mediastreamhandler.h"
30 #include "talk/app/webrtc/localaudiosource.h"
31 #include "talk/app/webrtc/localvideosource.h"
32 #include "talk/app/webrtc/videosourceinterface.h"
34 namespace webrtc { namespace
441 } // namespace webrtc
mediastreamhandler_unittest.cc 28 #include "talk/app/webrtc/mediastreamhandler.h"
32 #include "talk/app/webrtc/audiotrack.h"
33 #include "talk/app/webrtc/localvideosource.h"
34 #include "talk/app/webrtc/mediastream.h"
35 #include "talk/app/webrtc/streamcollection.h"
36 #include "talk/app/webrtc/videotrack.h"
51 namespace webrtc { namespace
297 } // namespace webrtc
peerconnectionfactory.cc 28 #include "talk/app/webrtc/peerconnectionfactory.h"
30 #include "talk/app/webrtc/audiotrack.h"
31 #include "talk/app/webrtc/localaudiosource.h"
32 #include "talk/app/webrtc/localvideosource.h"
33 #include "talk/app/webrtc/mediastreamproxy.h"
34 #include "talk/app/webrtc/mediastreamtrackproxy.h"
35 #include "talk/app/webrtc/peerconnection.h"
36 #include "talk/app/webrtc/peerconnectionproxy.h"
37 #include "talk/app/webrtc/portallocatorfactory.h"
38 #include "talk/app/webrtc/videosourceproxy.h
118 namespace webrtc { namespace
    [all...]
webrtcsessiondescriptionfactory.cc 28 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
30 #include "talk/app/webrtc/jsep.h"
31 #include "talk/app/webrtc/jsepsessiondescription.h"
32 #include "talk/app/webrtc/mediaconstraintsinterface.h"
33 #include "talk/app/webrtc/mediastreamsignaling.h"
34 #include "talk/app/webrtc/webrtcsession.h"
36 namespace webrtc { namespace
45 static const char kWebRTCIdentityName[] = "WebRTC";
78 webrtc::CreateSessionDescriptionObserver* observer)
82 talk_base::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer
    [all...]
peerconnectioninterface.h 29 // http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
74 #include "talk/app/webrtc/datachannelinterface.h"
75 #include "talk/app/webrtc/dtmfsenderinterface.h"
76 #include "talk/app/webrtc/jsep.h"
77 #include "talk/app/webrtc/mediastreaminterface.h"
78 #include "talk/app/webrtc/statstypes.h"
91 namespace webrtc { namespace
122 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions
    [all...]
statstypes.h 40 namespace webrtc { namespace
158 } // namespace webrtc
mediastreamsignaling.cc 28 #include "talk/app/webrtc/mediastreamsignaling.h"
32 #include "talk/app/webrtc/audiotrack.h"
33 #include "talk/app/webrtc/mediastreamproxy.h"
34 #include "talk/app/webrtc/mediaconstraintsinterface.h"
35 #include "talk/app/webrtc/mediastreamtrackproxy.h"
36 #include "talk/app/webrtc/videotrack.h"
43 namespace webrtc { namespace
145 AudioTrackInterface* AddAudioTrack(webrtc::MediaStreamInterface* stream,
151 VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
162 track->set_state(webrtc::MediaStreamTrackInterface::kLive)
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peerconnection.cc 28 #include "talk/app/webrtc/peerconnection.h"
32 #include "talk/app/webrtc/dtmfsender.h"
33 #include "talk/app/webrtc/jsepicecandidate.h"
34 #include "talk/app/webrtc/jsepsessiondescription.h"
35 #include "talk/app/webrtc/mediastreamhandler.h"
36 #include "talk/app/webrtc/streamcollection.h"
43 using webrtc::PeerConnectionInterface;
82 explicit CandidateMsg(const webrtc::JsepIceCandidate* candidate)
85 talk_base::scoped_ptr<const webrtc::JsepIceCandidate> candidate;
90 webrtc::SetSessionDescriptionObserver* observer
250 namespace webrtc { namespace
    [all...]
statscollector.cc 28 #include "talk/app/webrtc/statscollector.h"
35 namespace webrtc { namespace
158 webrtc::MediaStreamTrackInterface* track = tracks[j];
571 } // namespace webrtc
webrtcsession.cc 28 #include "talk/app/webrtc/webrtcsession.h"
34 #include "talk/app/webrtc/jsepicecandidate.h"
35 #include "talk/app/webrtc/jsepsessiondescription.h"
36 #include "talk/app/webrtc/mediaconstraintsinterface.h"
37 #include "talk/app/webrtc/mediastreamsignaling.h"
38 #include "talk/app/webrtc/peerconnectioninterface.h"
39 #include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
55 namespace webrtc { namespace
766 return webrtc::GetTrackIdBySsrc(
773 return webrtc::GetTrackIdBySsrc
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  /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/
PeerConnection.java 29 package org.webrtc;
36 * http://www.webrtc.org/reference/native-apis, which in turn is inspired by the
37 * JS APIs: http://dev.w3.org/2011/webrtc/editor/webrtc.html and
  /external/webrtc/src/common_audio/resampler/include/
resampler.h 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
21 namespace webrtc namespace
114 } // namespace webrtc
  /external/webrtc/src/modules/audio_processing/
audio_processing_impl.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
32 #include "external/webrtc/src/modules/audio_processing/debug.pb.h"
34 #include "webrtc/audio_processing/debug.pb.h"
38 namespace webrtc { namespace
40 /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
41 webrtc::kTraceAudioProcessing,
557 /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
558 webrtc::kTraceAudioProcessing,
652 } // namespace webrtc
echo_cancellation_impl.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
22 namespace webrtc { namespace
383 } // namespace webrtc
gain_control_impl.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
21 namespace webrtc { namespace
391 } // namespace webrtc
  /external/webrtc/src/system_wrappers/source/
data_log.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
24 namespace webrtc { namespace
455 } // namespace webrtc
trace_impl.cc 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
32 namespace webrtc { namespace
827 } // namespace webrtc
  /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
webrtcvideoengine.h 37 #include "talk/media/webrtc/webrtccommon.h"
38 #include "talk/media/webrtc/webrtcexport.h"
39 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
41 #include "webrtc/video_engine/include/vie_base.h"
48 namespace webrtc { namespace
85 public webrtc::TraceCallback,
135 webrtc::VideoDecoder* CreateExternalDecoder(webrtc::VideoCodecType type);
137 void DestroyExternalDecoder(webrtc::VideoDecoder* decoder);
142 webrtc::VideoEncoder* CreateExternalEncoder(webrtc::VideoCodecType type)
    [all...]
fakewebrtcvideoengine.h 38 #include "talk/media/webrtc/fakewebrtccommon.h"
39 #include "talk/media/webrtc/webrtcvideodecoderfactory.h"
40 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
41 #include "talk/media/webrtc/webrtcvie.h"
43 namespace webrtc { namespace
45 bool operator==(const webrtc::VideoCodec& c1, const webrtc::VideoCodec& c2) {
63 // WebRtc channel id and capture id share the same number space.
71 // Fake class for mocking out webrtc::VideoDecoder
72 class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder
    [all...]
  /external/webrtc/src/modules/audio_processing/interface/
audio_processing.h 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
19 namespace webrtc { namespace
595 } // namespace webrtc
  /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/javatests/src/org/webrtc/
PeerConnectionTest.java 28 package org.webrtc;
33 import org.webrtc.PeerConnection.IceConnectionState;
34 import org.webrtc.PeerConnection.IceGatheringState;
35 import org.webrtc.PeerConnection.SignalingState;
478 // Uncomment to get ALL WebRTC tracing and SENSITIVE libjingle logging.
493 // supported (https://code.google.com/p/webrtc/issues/detail?id=1408).
524 // https://code.google.com/p/webrtc/issues/detail?id=1253 is fixed.
642 // supported (https://code.google.com/p/webrtc/issues/detail?id=1408).

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