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  /external/webrtc/webrtc/modules/video_coding/test/
video_rtp_play.cc 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/modules/video_coding/test/receiver_tests.h"
12 #include "webrtc/modules/video_coding/test/vcm_payload_sink_factory.h"
13 #include "webrtc/system_wrappers/include/trace.h"
14 #include "webrtc/test/testsupport/fileutils.h"
19 const webrtc::VCMVideoProtection kConfigProtectionMethod =
20 webrtc::kProtectionNack;
33 std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt";
34 webrtc::Trace::CreateTrace();
35 webrtc::Trace::SetTraceFile(trace_file.c_str())
    [all...]
test_util.h 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
20 #include "webrtc/base/constructormagic.h"
21 #include "webrtc/modules/include/module_common_types.h"
22 #include "webrtc/modules/video_coding/include/video_coding.h"
23 #include "webrtc/system_wrappers/include/event_wrapper.h"
28 class NullEvent : public webrtc::EventWrapper {
36 virtual webrtc::EventTypeWrapper Wait(unsigned long max_time) { // NOLINT
37 return webrtc::kEventTimeout;
47 class NullEventFactory : public webrtc::EventFactory {
51 virtual webrtc::EventWrapper* CreateEvent() { return new NullEvent;
    [all...]
receiver_tests.h 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
17 #include "webrtc/common_types.h"
18 #include "webrtc/modules/include/module_common_types.h"
19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
20 #include "webrtc/modules/video_coding/include/video_coding.h"
21 #include "webrtc/modules/video_coding/test/test_util.h"
22 #include "webrtc/modules/video_coding/test/video_source.h"
23 #include "webrtc/typedefs.h"
25 class RtpDataCallback : public webrtc::NullRtpData {
27 explicit RtpDataCallback(webrtc::VideoCodingModule* vcm) : vcm_(vcm) {
    [all...]
  /external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/
before_initialization_fixture.cc 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
13 #include "webrtc/system_wrappers/include/sleep.h"
16 : voice_engine_(webrtc::VoiceEngine::Create()) {
19 voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_);
20 voe_codec_ = webrtc::VoECodec::GetInterface(voice_engine_);
21 voe_volume_control_ = webrtc::VoEVolumeControl::GetInterface(voice_engine_);
22 voe_dtmf_ = webrtc::VoEDtmf::GetInterface(voice_engine_);
23 voe_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(voice_engine_);
24 voe_apm_ = webrtc::VoEAudioProcessing::GetInterface(voice_engine_)
    [all...]
after_initialization_fixture.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/modules/audio_processing/include/audio_processing.h"
12 #include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
14 class TestErrorObserver : public webrtc::VoiceEngineObserver {
26 webrtc::Config config;
27 config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(false));
28 webrtc::AudioProcessing* audioproc = webrtc::AudioProcessing::Create(config);
  /external/webrtc/webrtc/api/objc/
RTCMediaSource.mm 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
13 #import "webrtc/api/objc/RTCMediaSource+Private.h"
16 rtc::scoped_refptr<webrtc::MediaSourceInterface> _nativeMediaSource;
30 - (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
35 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
43 + (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
47 return webrtc::MediaSourceInterface::kInitializing;
49 return webrtc::MediaSourceInterface::kLive;
51 return webrtc::MediaSourceInterface::kEnded;
53 return webrtc::MediaSourceInterface::kMuted
    [all...]
RTCIceCandidate+Private.h 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
13 #include "talk/app/webrtc/jsep.h"
14 #include "webrtc/base/scoped_ptr.h"
25 rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate;
32 (webrtc::IceCandidateInterface *)candidate;
RTCMediaSource+Private.h 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
13 #include "talk/app/webrtc/mediastreaminterface.h"
24 rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
28 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
31 + (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
35 (webrtc::MediaSourceInterface::SourceState)nativeState;
RTCMediaStreamTrack+Private.h 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved.
13 #include "talk/app/webrtc/mediastreaminterface.h"
14 #include "webrtc/base/scoped_ptr.h"
26 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
32 (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
35 + (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
39 (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
  /external/webrtc/webrtc/voice_engine/test/auto_test/standard/
audio_processing_test.cc 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/test/testsupport/fileutils.h"
12 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
13 #include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
15 class RxCallback : public webrtc::VoERxVadCallback {
35 void TryEnablingAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
39 webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
46 void TryEnablingRxAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
50 webrtc::AgcModes agc_mode = webrtc::kAgcDefault
    [all...]
  /external/webrtc/talk/app/webrtc/test/
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"
35 #include "webrtc/base/sigslot.h"
38 : public webrtc::PeerConnectionObserver,
39 public webrtc::CreateSessionDescriptionObserver,
48 bool CreatePc(const webrtc::MediaConstraintsInterface* constraints);
50 rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
52 const webrtc::DataChannelInit& init)
    [all...]
  /external/webrtc/talk/app/webrtc/objc/
RTCICECandidate+Internal.h 30 #include "talk/app/webrtc/peerconnectioninterface.h"
35 webrtc::IceCandidateInterface* candidate;
37 - (id)initWithCandidate:(const webrtc::IceCandidateInterface*)candidate;
RTCStatsReport+Internal.h 30 #include "talk/app/webrtc/statstypes.h"
34 - (instancetype)initWithStatsReport:(const webrtc::StatsReport&)statsReport;
RTCAVFoundationVideoSource+Internal.h 34 @property(nonatomic, readonly) webrtc::AVFoundationVideoCapturer* capturer;
RTCMediaConstraints.mm 36 #include "webrtc/base/scoped_ptr.h"
42 rtc::scoped_ptr<webrtc::RTCMediaConstraintsNative> _constraints;
43 webrtc::MediaConstraintsInterface::Constraints _mandatory;
44 webrtc::MediaConstraintsInterface::Constraints _optional;
53 new webrtc::RTCMediaConstraintsNative(_mandatory, _optional));
58 + (webrtc::MediaConstraintsInterface::Constraints)constraintsFromArray:
60 webrtc::MediaConstraintsInterface::Constraints constraints;
62 constraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
72 - (const webrtc::RTCMediaConstraintsNative*)constraints {
  /external/webrtc/webrtc/audio/
audio_send_stream.h 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
14 #include "webrtc/audio_send_stream.h"
15 #include "webrtc/audio_state.h"
16 #include "webrtc/base/thread_checker.h"
17 #include "webrtc/base/scoped_ptr.h"
19 namespace webrtc { namespace
28 class AudioSendStream final : public webrtc::AudioSendStream {
30 AudioSendStream(const webrtc::AudioSendStream::Config& config,
31 const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
35 // webrtc::SendStream implementation
    [all...]
  /external/webrtc/webrtc/modules/audio_coding/codecs/isac/
locked_bandwidth_info.cc 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
13 namespace webrtc { namespace
22 } // namespace webrtc
  /external/webrtc/webrtc/modules/audio_coding/neteq/
background_noise_unittest.cc 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
13 #include "webrtc/modules/audio_coding/neteq/background_noise.h"
17 namespace webrtc { namespace
26 } // namespace webrtc
  /external/webrtc/webrtc/modules/audio_coding/neteq/tools/
neteq_performance_test.h 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
14 #include "webrtc/typedefs.h"
16 namespace webrtc { namespace
30 } // namespace webrtc
  /external/webrtc/webrtc/modules/audio_device/android/
audio_common.h 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
14 namespace webrtc { namespace
30 } // namespace webrtc
  /external/webrtc/webrtc/modules/desktop_capture/
cropped_desktop_frame.h 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
14 #include "webrtc/modules/desktop_capture/desktop_frame.h"
16 namespace webrtc { namespace
22 } // namespace webrtc
  /external/webrtc/webrtc/modules/video_coding/utility/
qp_parser.h 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
14 #include "webrtc/modules/video_coding/encoded_frame.h"
16 namespace webrtc { namespace
28 } // namespace webrtc
  /external/webrtc/webrtc/modules/video_render/
video_render_internal.h 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
17 namespace webrtc { namespace
19 // In order to be able to use the internal webrtc video render
23 } // namespace webrtc
  /external/webrtc/webrtc/system_wrappers/source/
cpu_info.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
11 #include "webrtc/system_wrappers/include/cpu_info.h"
13 #include "webrtc/base/systeminfo.h"
15 namespace webrtc { namespace
21 } // namespace webrtc
critical_section.cc 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
13 #include "webrtc/system_wrappers/source/critical_section_win.h"
15 #include "webrtc/system_wrappers/source/critical_section_posix.h"
18 namespace webrtc { namespace
28 } // namespace webrtc

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