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Searched
refs:webrtc
(Results
51 - 75
of
2050
) sorted by null
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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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