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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;

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