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defs:webrtc
(Results
226 - 250
of
253
) sorted by null
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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
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// in the constructor via
WebRTC
's GetInterface mechanism and then releases
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// This is useful for some
WebRTC
objects that have their own Create/Delete
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// A very basic implementation of
webrtc
::Transport that acts as a transport
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// but just forwards all calls to a local
webrtc
::VoENetwork implementation.
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class WebRTCTransportImpl : public
webrtc
::Transport {
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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"
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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
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namespace
webrtc
{
namespace
[
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...]
webrtcsessiondescriptionfactory.cc
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#include "talk/app/
webrtc
/webrtcsessiondescriptionfactory.h"
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#include "talk/app/
webrtc
/jsep.h"
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#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
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// http://dev.w3.org/2011/
webrtc
/editor/
webrtc
.html#peer-to-peer-connections.
74
#include "talk/app/
webrtc
/datachannelinterface.h"
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#include "talk/app/
webrtc
/dtmfsenderinterface.h"
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#include "talk/app/
webrtc
/jsep.h"
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#include "talk/app/
webrtc
/mediastreaminterface.h"
78
#include "talk/app/
webrtc
/statstypes.h"
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namespace
webrtc
{
namespace
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// See http://dev.w3.org/2011/
webrtc
/editor/
webrtc
.html#state-definitions
[
all
...]
statstypes.h
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namespace
webrtc
{
namespace
158
} // namespace
webrtc
mediastreamsignaling.cc
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#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"
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#include "talk/app/
webrtc
/videotrack.h"
43
namespace
webrtc
{
namespace
145
AudioTrackInterface* AddAudioTrack(
webrtc
::MediaStreamInterface* stream,
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VideoTrackInterface* AddVideoTrack(
webrtc
::MediaStreamInterface* stream,
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track->set_state(
webrtc
::MediaStreamTrackInterface::kLive)
[
all
...]
peerconnection.cc
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#include "talk/app/
webrtc
/peerconnection.h"
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#include "talk/app/
webrtc
/dtmfsender.h"
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#include "talk/app/
webrtc
/jsepicecandidate.h"
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#include "talk/app/
webrtc
/jsepsessiondescription.h"
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#include "talk/app/
webrtc
/mediastreamhandler.h"
36
#include "talk/app/
webrtc
/streamcollection.h"
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using
webrtc
::PeerConnectionInterface;
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explicit CandidateMsg(const
webrtc
::JsepIceCandidate* candidate)
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talk_base::scoped_ptr<const
webrtc
::JsepIceCandidate> candidate;
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webrtc
::SetSessionDescriptionObserver* observer
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namespace
webrtc
{
namespace
[
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...]
statscollector.cc
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#include "talk/app/
webrtc
/statscollector.h"
35
namespace
webrtc
{
namespace
158
webrtc
::MediaStreamTrackInterface* track = tracks[j];
571
} // namespace
webrtc
webrtcsession.cc
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#include "talk/app/
webrtc
/webrtcsession.h"
34
#include "talk/app/
webrtc
/jsepicecandidate.h"
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#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(
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return
webrtc
::GetTrackIdBySsrc
[
all
...]
/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
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* 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,
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/*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
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* 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"
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#include "talk/media/
webrtc
/webrtcexport.h"
39
#include "talk/media/
webrtc
/webrtcvideoencoderfactory.h"
41
#include "
webrtc
/video_engine/include/vie_base.h"
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namespace
webrtc
{
namespace
85
public
webrtc
::TraceCallback,
135
webrtc
::VideoDecoder* CreateExternalDecoder(
webrtc
::VideoCodecType type);
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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.
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// Fake class for mocking out
webrtc
::VideoDecoder
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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).
Completed in 1495 milliseconds
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