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Searched
refs:webrtc
(Results
76 - 100
of
2050
) sorted by null
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/external/webrtc/webrtc/test/
run_loop.cc
2
* Copyright (c) 2013 The
WebRTC
project authors. All Rights Reserved.
10
#include "
webrtc
/test/run_loop.h"
14
namespace
webrtc
{
namespace
22
} // namespace
webrtc
/external/webrtc/webrtc/modules/video_coding/codecs/test/
videoprocessor.h
2
* Copyright (c) 2012 The
WebRTC
project authors. All Rights Reserved.
16
#include "
webrtc
/base/checks.h"
17
#include "
webrtc
/common_video/libyuv/include/scaler.h"
18
#include "
webrtc
/common_video/libyuv/include/webrtc_libyuv.h"
19
#include "
webrtc
/modules/video_coding/include/video_codec_interface.h"
20
#include "
webrtc
/modules/video_coding/codecs/test/packet_manipulator.h"
21
#include "
webrtc
/modules/video_coding/codecs/test/stats.h"
22
#include "
webrtc
/system_wrappers/include/tick_util.h"
23
#include "
webrtc
/test/testsupport/frame_reader.h"
24
#include "
webrtc
/test/testsupport/frame_writer.h
27
namespace
webrtc
{
namespace
[
all
...]
/external/webrtc/webrtc/modules/audio_coding/test/
Tester.cc
2
* Copyright (c) 2012 The
WebRTC
project authors. All Rights Reserved.
16
#include "
webrtc
/modules/audio_coding/include/audio_coding_module.h"
17
#include "
webrtc
/modules/audio_coding/test/APITest.h"
18
#include "
webrtc
/modules/audio_coding/test/EncodeDecodeTest.h"
19
#include "
webrtc
/modules/audio_coding/test/iSACTest.h"
20
#include "
webrtc
/modules/audio_coding/test/opus_test.h"
21
#include "
webrtc
/modules/audio_coding/test/PacketLossTest.h"
22
#include "
webrtc
/modules/audio_coding/test/TestAllCodecs.h"
23
#include "
webrtc
/modules/audio_coding/test/TestRedFec.h"
24
#include "
webrtc
/modules/audio_coding/test/TestStereo.h
[
all
...]
/external/webrtc/talk/app/webrtc/objc/
RTCDataChannel+Internal.h
30
#include "talk/app/
webrtc
/datachannelinterface.h"
31
#include "
webrtc
/base/scoped_ref_ptr.h"
35
@property(nonatomic, readonly) const
webrtc
::DataBuffer* dataBuffer;
37
- (instancetype)initWithDataBuffer:(const
webrtc
::DataBuffer&)buffer;
43
@property(nonatomic, readonly) const
webrtc
::DataChannelInit* dataChannelInit;
50
rtc::scoped_refptr<
webrtc
::DataChannelInterface> dataChannel;
53
(rtc::scoped_refptr<
webrtc
::DataChannelInterface>)dataChannel;
RTCAVFoundationVideoSource.mm
40
rtc::scoped_ptr<
webrtc
::AVFoundationVideoCapturer> capturer;
41
capturer.reset(new
webrtc
::AVFoundationVideoCapturer());
42
rtc::scoped_refptr<
webrtc
::VideoSourceInterface> source =
60
- (
webrtc
::AVFoundationVideoCapturer*)capturer {
64
webrtc
::AVFoundationVideoCapturer* foundationCapturer =
65
static_cast<
webrtc
::AVFoundationVideoCapturer*>(capturer);
RTCAudioTrack+Internal.h
30
#include "talk/app/
webrtc
/mediastreaminterface.h"
35
rtc::scoped_refptr<
webrtc
::AudioTrackInterface> audioTrack;
RTCAudioTrack.mm
41
- (rtc::scoped_refptr<
webrtc
::AudioTrackInterface>)audioTrack {
42
return static_cast<
webrtc
::AudioTrackInterface*>(self.mediaTrack.get());
RTCICEServer+Internal.h
30
#include "talk/app/
webrtc
/peerconnectioninterface.h"
36
readonly)
webrtc
::PeerConnectionInterface::IceServer iceServer;
RTCMediaConstraints+Internal.h
32
#include "talk/app/
webrtc
/mediastreaminterface.h"
38
webrtc
::RTCMediaConstraintsNative *constraints;
RTCPeerConnectionFactory+Internal.h
30
#include "talk/app/
webrtc
/peerconnectionfactory.h"
31
#include "
webrtc
/base/scoped_ptr.h"
36
webrtc
::PeerConnectionFactoryInterface> nativeFactory;
RTCPeerConnectionInterface+Internal.h
28
#import "talk/app/
webrtc
/objc/public/RTCPeerConnectionInterface.h"
30
#include "talk/app/
webrtc
/peerconnectioninterface.h"
35
webrtc
::PeerConnectionInterface::RTCConfiguration nativeConfiguration;
RTCVideoSource+Internal.h
30
#include "talk/app/
webrtc
/videosourceinterface.h"
35
rtc::scoped_refptr<
webrtc
::VideoSourceInterface>videoSource;
/external/webrtc/talk/media/webrtc/
fakewebrtccall.cc
28
#include "talk/media/
webrtc
/fakewebrtccall.h"
34
#include "
webrtc
/base/checks.h"
35
#include "
webrtc
/base/gunit.h"
36
#include "
webrtc
/audio/audio_sink.h"
40
const
webrtc
::AudioSendStream::Config& config) : config_(config) {
44
const
webrtc
::AudioSendStream::Config&
50
const
webrtc
::AudioSendStream::Stats& stats) {
67
webrtc
::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
72
const
webrtc
::AudioReceiveStream::Config& config)
77
const
webrtc
::AudioReceiveStream::Config
[
all
...]
fakewebrtcvideoengine.h
36
#include "talk/media/
webrtc
/fakewebrtccommon.h"
37
#include "talk/media/
webrtc
/webrtcvideodecoderfactory.h"
38
#include "talk/media/
webrtc
/webrtcvideoencoderfactory.h"
39
#include "
webrtc
/base/basictypes.h"
40
#include "
webrtc
/base/criticalsection.h"
41
#include "
webrtc
/base/gunit.h"
42
#include "
webrtc
/base/stringutils.h"
43
#include "
webrtc
/base/thread_annotations.h"
44
#include "
webrtc
/modules/video_coding/include/video_error_codes.h"
45
#include "
webrtc
/video_decoder.h
[
all
...]
webrtcmediaengine.h
35
#include "
webrtc
/config.h"
37
namespace
webrtc
{
namespace
50
webrtc
::AudioDeviceModule* adm,
59
// Convert cricket::RtpHeaderExtension:s to
webrtc
::RtpExtension:s, discarding
63
std::vector<
webrtc
::RtpExtension> FilterRtpExtensions(
/external/webrtc/webrtc/api/objc/
RTCMediaStreamTrack.mm
2
* Copyright 2015 The
WebRTC
project authors. All Rights Reserved.
13
#import "
webrtc
/api/objc/RTCMediaStreamTrack+Private.h"
14
#import "
webrtc
/base/objc/NSString+StdString.h"
17
rtc::scoped_refptr<
webrtc
::MediaStreamTrackInterface> _nativeTrack;
51
- (rtc::scoped_refptr<
webrtc
::MediaStreamTrackInterface>)nativeTrack {
56
(rtc::scoped_refptr<
webrtc
::MediaStreamTrackInterface>)nativeTrack {
64
+ (
webrtc
::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
68
return
webrtc
::MediaStreamTrackInterface::kInitializing;
70
return
webrtc
::MediaStreamTrackInterface::kLive;
72
return
webrtc
::MediaStreamTrackInterface::kEnded
[
all
...]
RTCMediaConstraints+Private.h
2
* Copyright 2015 The
WebRTC
project authors. All Rights Reserved.
13
#include "talk/app/
webrtc
/mediaconstraintsinterface.h"
14
#include "
webrtc
/base/scoped_ptr.h"
16
namespace
webrtc
{
namespace
33
} // namespace
webrtc
44
- (rtc::scoped_ptr<
webrtc
::MediaConstraints>)nativeConstraints;
47
+ (
webrtc
::MediaConstraintsInterface::Constraints)
/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/
conference_transport.h
2
* Copyright (c) 2015 The
WebRTC
project authors. All Rights Reserved.
19
#include "
webrtc
/base/basictypes.h"
20
#include "
webrtc
/base/platform_thread.h"
21
#include "
webrtc
/base/scoped_ptr.h"
22
#include "
webrtc
/common_types.h"
23
#include "
webrtc
/modules/rtp_rtcp/include/rtp_header_parser.h"
24
#include "
webrtc
/system_wrappers/include/critical_section_wrapper.h"
25
#include "
webrtc
/system_wrappers/include/event_wrapper.h"
26
#include "
webrtc
/voice_engine/include/voe_base.h"
27
#include "
webrtc
/voice_engine/include/voe_codec.h
[
all
...]
/frameworks/av/media/libeffects/preprocessing/
Android.mk
14
external/
webrtc
\
15
external/
webrtc
/
webrtc
/modules/include \
16
external/
webrtc
/
webrtc
/modules/audio_processing/include \
/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/
bwe_rtp.cc
2
* Copyright (c) 2014 The
WebRTC
project authors. All Rights Reserved.
11
#include "
webrtc
/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
20
#include "
webrtc
/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
21
#include "
webrtc
/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
22
#include "
webrtc
/modules/rtp_rtcp/include/rtp_header_parser.h"
23
#include "
webrtc
/modules/rtp_rtcp/include/rtp_payload_registry.h"
24
#include "
webrtc
/test/rtp_file_reader.h"
71
webrtc
::Clock* clock,
72
webrtc
::RemoteBitrateObserver* observer,
73
webrtc
::test::RtpFileReader** rtp_reader
[
all
...]
/external/webrtc/talk/app/webrtc/java/jni/
androidmediaencoder_jni.h
34
#include "talk/app/
webrtc
/java/jni/jni_helpers.h"
35
#include "talk/media/
webrtc
/webrtcvideoencoderfactory.h"
49
webrtc
::VideoEncoder* CreateVideoEncoder(
webrtc
::VideoCodecType type)
52
void DestroyVideoEncoder(
webrtc
::VideoEncoder* encoder) override;
/external/webrtc/webrtc/examples/peerconnection/client/
conductor.h
2
* Copyright 2012 The
WebRTC
Project Authors. All rights reserved.
20
#include "talk/app/
webrtc
/mediastreaminterface.h"
21
#include "talk/app/
webrtc
/peerconnectioninterface.h"
22
#include "
webrtc
/examples/peerconnection/client/main_wnd.h"
23
#include "
webrtc
/examples/peerconnection/client/peer_connection_client.h"
24
#include "
webrtc
/base/scoped_ptr.h"
26
namespace
webrtc
{
namespace
28
} // namespace
webrtc
35
: public
webrtc
::PeerConnectionObserver,
36
public
webrtc
::CreateSessionDescriptionObserver
[
all
...]
/external/webrtc/webrtc/modules/video_capture/ios/
rtc_video_capture_ios_objc.h
2
* Copyright (c) 2013 The
WebRTC
project authors. All Rights Reserved.
17
#include "
webrtc
/modules/video_capture/ios/video_capture_ios.h"
26
@property
webrtc
::VideoRotation frameRotation;
31
- (id)initWithOwner:(
webrtc
::videocapturemodule::VideoCaptureIos*)owner
35
(const
webrtc
::VideoCaptureCapability&)capability;
/external/webrtc/talk/app/webrtc/
fakemediacontroller.h
31
#include "talk/app/
webrtc
/mediacontroller.h"
32
#include "
webrtc
/base/checks.h"
36
class FakeMediaController : public
webrtc
::MediaControllerInterface {
39
webrtc
::Call* call)
45
webrtc
::Call* call_w() override { return call_; }
52
webrtc
::Call* call_;
/external/webrtc/talk/app/webrtc/java/src/org/webrtc/
SdpObserver.java
28
package org.
webrtc
;
Completed in 846 milliseconds
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