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/external/webrtc/
DWATCHLISTS18 # NOTE: if you like this you might like webrtc-reviews@webrtc.org!
19 'filepath': '^webrtc/.*',
22 # webrtc/build/ and non-recursive contents of ./ and webrtc/
23 'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*',
26 'filepath': '^webrtc/[^/]*\.h$|'\
27 'webrtc/voice_engine/include/.*',
36 'filepath': 'webrtc/audio/.*',
39 'filepath': 'webrtc/call/.*',
42 'filepath': 'webrtc/video/.*',
45 'filepath': 'webrtc/voice_engine/.*',
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/external/webrtc/talk/
Dlibjingle.gyp53 'app/webrtc/java/jni/classreferenceholder.cc',
54 'app/webrtc/java/jni/classreferenceholder.h',
55 'app/webrtc/java/jni/jni_helpers.cc',
56 'app/webrtc/java/jni/jni_helpers.h',
57 'app/webrtc/java/jni/native_handle_impl.cc',
58 'app/webrtc/java/jni/native_handle_impl.h',
59 'app/webrtc/java/jni/peerconnection_jni.cc',
81 'app/webrtc/androidvideocapturer.cc',
82 'app/webrtc/androidvideocapturer.h',
83 'app/webrtc/java/jni/androidmediacodeccommon.h',
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Dlibjingle_tests.gyp63 'media/webrtc/fakewebrtccall.cc',
64 'media/webrtc/fakewebrtccall.h',
65 'media/webrtc/fakewebrtccommon.h',
66 'media/webrtc/fakewebrtcdeviceinfo.h',
67 'media/webrtc/fakewebrtcvcmfactory.h',
68 'media/webrtc/fakewebrtcvideocapturemodule.h',
69 'media/webrtc/fakewebrtcvideoengine.h',
70 'media/webrtc/fakewebrtcvoiceengine.h',
98 'media/webrtc/simulcast_unittest.cc',
99 'media/webrtc/webrtcmediaengine_unittest.cc',
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/external/webrtc/webrtc/voice_engine/test/auto_test/standard/
Daudio_processing_test.cc15 class RxCallback : public webrtc::VoERxVadCallback {
35 void TryEnablingAgcWithMode(webrtc::AgcModes agc_mode_to_set) { in TryEnablingAgcWithMode()
39 webrtc::AgcModes agc_mode = webrtc::kAgcDefault; in TryEnablingAgcWithMode()
46 void TryEnablingRxAgcWithMode(webrtc::AgcModes agc_mode_to_set) { in TryEnablingRxAgcWithMode()
50 webrtc::AgcModes agc_mode = webrtc::kAgcDefault; in TryEnablingRxAgcWithMode()
59 void TryEnablingEcWithMode(webrtc::EcModes ec_mode_to_set, in TryEnablingEcWithMode()
60 webrtc::EcModes expected_mode) { in TryEnablingEcWithMode()
64 webrtc::EcModes ec_mode = webrtc::kEcDefault; in TryEnablingEcWithMode()
72 void TryEnablingAecmWithMode(webrtc::AecmModes aecm_mode_to_set, in TryEnablingAecmWithMode()
77 webrtc::AecmModes aecm_mode = webrtc::kAecmEarpiece; in TryEnablingAecmWithMode()
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/external/webrtc/talk/media/webrtc/
Dfakewebrtccall.h50 class FakeAudioSendStream final : public webrtc::AudioSendStream {
58 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
60 const webrtc::AudioSendStream::Config& GetConfig() const;
61 void SetStats(const webrtc::AudioSendStream::Stats& stats);
68 void SignalNetworkState(webrtc::NetworkState state) override {} in SignalNetworkState()
76 webrtc::AudioSendStream::Stats GetStats() const override;
79 webrtc::AudioSendStream::Config config_;
80 webrtc::AudioSendStream::Stats stats_;
83 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
86 const webrtc::AudioReceiveStream::Config& config);
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Dfakewebrtccall.cc40 const webrtc::AudioSendStream::Config& config) : config_(config) { in FakeAudioSendStream()
44 const webrtc::AudioSendStream::Config&
50 const webrtc::AudioSendStream::Stats& stats) { in SetStats()
67 webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const { in GetStats()
72 const webrtc::AudioReceiveStream::Config& config) in FakeAudioReceiveStream()
77 const webrtc::AudioReceiveStream::Config&
83 const webrtc::AudioReceiveStream::Stats& stats) { in SetStats()
91 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const { in GetStats()
96 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) { in SetSink()
101 const webrtc::VideoSendStream::Config& config, in FakeVideoSendStream()
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Dwebrtcvoe.h49 explicit scoped_voe_engine(webrtc::VoiceEngine* e) : ptr(e) {} in scoped_voe_engine()
51 ~scoped_voe_engine() { if (ptr) VERIFY(webrtc::VoiceEngine::Delete(ptr)); } in ~scoped_voe_engine()
55 VERIFY(webrtc::VoiceEngine::Delete(ptr)); in reset()
59 webrtc::VoiceEngine* get() const { return ptr; } in get()
61 webrtc::VoiceEngine* ptr;
92 : engine_(webrtc::VoiceEngine::Create()), processing_(engine_), in VoEWrapper()
97 VoEWrapper(webrtc::VoEAudioProcessing* processing, in VoEWrapper()
98 webrtc::VoEBase* base, in VoEWrapper()
99 webrtc::VoECodec* codec, in VoEWrapper()
100 webrtc::VoEHardware* hw, in VoEWrapper()
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Dfakewebrtcvoiceengine.h58 class FakeAudioProcessing : public webrtc::AudioProcessing {
67 webrtc::AudioProcessing::ChannelLayout input_layout,
68 webrtc::AudioProcessing::ChannelLayout output_layout,
69 webrtc::AudioProcessing::ChannelLayout reverse_layout));
71 const webrtc::ProcessingConfig& processing_config));
73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
85 WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
90 webrtc::AudioProcessing::ChannelLayout input_layout,
92 webrtc::AudioProcessing::ChannelLayout output_layout,
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Dfakewebrtcvideoengine.h61 class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
67 virtual int32_t InitDecode(const webrtc::VideoCodec*, int32_t) { in InitDecode()
71 virtual int32_t Decode(const webrtc::EncodedImage&, in Decode()
73 const webrtc::RTPFragmentationHeader*, in Decode()
74 const webrtc::CodecSpecificInfo*, in Decode()
81 webrtc::DecodedImageCallback*) { in RegisterDecodeCompleteCallback()
104 virtual webrtc::VideoDecoder* CreateVideoDecoder( in CreateVideoDecoder()
105 webrtc::VideoCodecType type) { in CreateVideoDecoder()
115 virtual void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) { in DestroyVideoDecoder()
122 void AddSupportedVideoCodecType(webrtc::VideoCodecType type) { in AddSupportedVideoCodecType()
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/external/webrtc/talk/app/webrtc/objc/
DRTCEnumConverter.mm30 #include "talk/app/webrtc/peerconnectioninterface.h"
35 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
37 case webrtc::PeerConnectionInterface::kIceConnectionNew:
39 case webrtc::PeerConnectionInterface::kIceConnectionChecking:
41 case webrtc::PeerConnectionInterface::kIceConnectionConnected:
43 case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
45 case webrtc::PeerConnectionInterface::kIceConnectionFailed:
47 case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
49 case webrtc::PeerConnectionInterface::kIceConnectionClosed:
51 case webrtc::PeerConnectionInterface::kIceConnectionMax:
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DRTCEnumConverter.h39 (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
42 (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
45 (webrtc::PeerConnectionInterface::SignalingState)nativeState;
47 + (webrtc::PeerConnectionInterface::StatsOutputLevel)
51 (webrtc::MediaSourceInterface::SourceState)nativeState;
53 + (webrtc::MediaStreamTrackInterface::TrackState)convertTrackStateToNative:
57 (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
60 (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnum;
62 + (webrtc::PeerConnectionInterface::IceTransportsType)nativeEnumForIceTransportsType:
66 (webrtc::PeerConnectionInterface::BundlePolicy)nativeEnum;
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/external/webrtc/tools/valgrind-webrtc/drmemory/
Dsuppressions.txt6 name=https://code.google.com/p/webrtc/issues/detail?id=2323 (1)
10 *!webrtc::`anonymous namespace'::WindowCapturerWin::Capture
15 name=<https://code.google.com/p/webrtc/issues/detail?id=2323 (2)>
17 *!webrtc::Desktop::GetThreadDesktop
18 *!webrtc::ScopedThreadDesktop::ScopedThreadDesktop
19 *!webrtc::ScreenCapturerWinGdi::ScreenCapturerWinGdi
20 *!webrtc::ScreenCapturer::Create
21 *!webrtc::ScreenCapturerTest::SetUp
25 name=<https://code.google.com/p/webrtc/issues/detail?id=2323 (3)>
27 *!webrtc::Desktop::GetThreadDesktop
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/external/webrtc/webrtc/modules/video_capture/mac/
Dvideo_capture_mac.mm18 #include "webrtc/modules/video_capture/device_info_impl.h"
19 #include "webrtc/modules/video_capture/video_capture_config.h"
20 #include "webrtc/modules/video_capture/video_capture_impl.h"
21 #include "webrtc/system_wrappers/include/ref_count.h"
22 #include "webrtc/system_wrappers/include/trace.h"
30 #include "webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit.h"
31 #include "webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_info.h"
34 namespace webrtc
50 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
57 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
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/external/webrtc/webrtc/modules/video_capture/windows/
Dvideo_capture_ds.cc21 namespace webrtc namespace
80 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
91 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
100 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
107 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
119 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
128 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
143 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
148 WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, _id, in Init()
170 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in StartCapture()
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Ddevice_info_ds.cc22 namespace webrtc namespace
92 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id, in DeviceInfoDS()
117 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in Init()
166 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, in GetDeviceInfo()
210 WEBRTC_TRACE(webrtc::kTraceError, in GetDeviceInfo()
211 webrtc::kTraceVideoCapture, _id, in GetDeviceInfo()
225 WEBRTC_TRACE(webrtc::kTraceError, in GetDeviceInfo()
226 webrtc::kTraceVideoCapture, _id, in GetDeviceInfo()
241 WEBRTC_TRACE(webrtc::kTraceError, in GetDeviceInfo()
242 webrtc::kTraceVideoCapture, _id, in GetDeviceInfo()
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Dneteq_rtpplay.cc43 using webrtc::NetEq;
44 using webrtc::WebRtcRTPHeader;
148 std::string CodecName(webrtc::NetEqDecoder codec) { in CodecName()
150 case webrtc::NetEqDecoder::kDecoderPCMu: in CodecName()
152 case webrtc::NetEqDecoder::kDecoderPCMa: in CodecName()
154 case webrtc::NetEqDecoder::kDecoderILBC: in CodecName()
156 case webrtc::NetEqDecoder::kDecoderISAC: in CodecName()
158 case webrtc::NetEqDecoder::kDecoderISACswb: in CodecName()
160 case webrtc::NetEqDecoder::kDecoderOpus: in CodecName()
162 case webrtc::NetEqDecoder::kDecoderPCM16B: in CodecName()
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/external/webrtc/webrtc/modules/audio_coding/test/
DTester.cc30 using webrtc::Trace;
38 Trace::SetTraceFile((webrtc::test::OutputPath() + in TEST()
40 webrtc::TestAllCodecs(ACM_TEST_MODE).Perform(); in TEST()
50 Trace::SetTraceFile((webrtc::test::OutputPath() +
52 webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
63 Trace::SetTraceFile((webrtc::test::OutputPath() +
65 webrtc::TestRedFec().Perform();
77 Trace::SetTraceFile((webrtc::test::OutputPath() +
79 webrtc::ISACTest(ACM_TEST_MODE).Perform();
92 Trace::SetTraceFile((webrtc::test::OutputPath() +
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/external/webrtc/talk/app/webrtc/test/
Dpeerconnectiontestwrapper.h38 : 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);
56 webrtc::PeerConnectionInterface::SignalingState new_state) {} in OnSignalingChange()
58 webrtc::PeerConnectionObserver::StateType state_changed) {} in OnStateChange()
59 virtual void OnAddStream(webrtc::MediaStreamInterface* stream);
60 virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) {} in OnRemoveStream()
61 virtual void OnDataChannel(webrtc::DataChannelInterface* data_channel);
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/external/webrtc/webrtc/system_wrappers/source/
Dlogcat_trace_context.cc18 namespace webrtc { namespace
26 case webrtc::kTraceStateInfo: return ANDROID_LOG_DEBUG; in AndroidLogPriorityFromWebRtcLogLevel()
27 case webrtc::kTraceWarning: return ANDROID_LOG_WARN; in AndroidLogPriorityFromWebRtcLogLevel()
28 case webrtc::kTraceError: return ANDROID_LOG_ERROR; in AndroidLogPriorityFromWebRtcLogLevel()
29 case webrtc::kTraceCritical: return ANDROID_LOG_FATAL; in AndroidLogPriorityFromWebRtcLogLevel()
30 case webrtc::kTraceApiCall: return ANDROID_LOG_VERBOSE; in AndroidLogPriorityFromWebRtcLogLevel()
31 case webrtc::kTraceModuleCall: return ANDROID_LOG_VERBOSE; in AndroidLogPriorityFromWebRtcLogLevel()
32 case webrtc::kTraceMemory: return ANDROID_LOG_VERBOSE; in AndroidLogPriorityFromWebRtcLogLevel()
33 case webrtc::kTraceTimer: return ANDROID_LOG_VERBOSE; in AndroidLogPriorityFromWebRtcLogLevel()
34 case webrtc::kTraceStream: return ANDROID_LOG_VERBOSE; in AndroidLogPriorityFromWebRtcLogLevel()
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/external/webrtc/webrtc/modules/video_coding/test/
Dvideo_rtp_play.cc19 const webrtc::VCMVideoProtection kConfigProtectionMethod =
20 webrtc::kProtectionNack;
33 std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt"; in RtpPlay()
34 webrtc::Trace::CreateTrace(); in RtpPlay()
35 webrtc::Trace::SetTraceFile(trace_file.c_str()); in RtpPlay()
36 webrtc::Trace::set_level_filter(webrtc::kTraceAll); in RtpPlay()
38 webrtc::rtpplayer::PayloadTypes payload_types; in RtpPlay()
39 payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple( in RtpPlay()
40 kDefaultUlpFecPayloadType, "ULPFEC", webrtc::kVideoCodecULPFEC)); in RtpPlay()
41 payload_types.push_back(webrtc::rtpplayer::PayloadCodecTuple( in RtpPlay()
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/external/webrtc/webrtc/modules/video_coding/codecs/vp8/
Dvp8_sequence_coder.cc23 class Vp8SequenceCoderEncodeCallback : public webrtc::EncodedImageCallback {
28 int Encoded(const webrtc::EncodedImage& encoded_image,
29 const webrtc::CodecSpecificInfo* codecSpecificInfo,
30 const webrtc::RTPFragmentationHeader*);
32 webrtc::EncodedImage encoded_image() { return encoded_image_; } in encoded_image()
36 webrtc::EncodedImage encoded_image_;
46 const webrtc::EncodedImage& encoded_image, in Encoded()
47 const webrtc::CodecSpecificInfo* codecSpecificInfo, in Encoded()
48 const webrtc::RTPFragmentationHeader* fragmentation) { in Encoded()
68 class Vp8SequenceCoderDecodeCallback : public webrtc::DecodedImageCallback {
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/external/webrtc/webrtc/modules/video_capture/test/
Dvideo_capture_unittest.cc29 using webrtc::CriticalSectionWrapper;
30 using webrtc::CriticalSectionScoped;
31 using webrtc::SleepMs;
32 using webrtc::TickTime;
33 using webrtc::VideoCaptureAlarm;
34 using webrtc::VideoCaptureCapability;
35 using webrtc::VideoCaptureDataCallback;
36 using webrtc::VideoCaptureFactory;
37 using webrtc::VideoCaptureFeedBack;
38 using webrtc::VideoCaptureModule;
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/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/
Dbefore_initialization_fixture.h57 webrtc::VoiceEngine* voice_engine_;
58 webrtc::VoEBase* voe_base_;
59 webrtc::VoECodec* voe_codec_;
60 webrtc::VoEVolumeControl* voe_volume_control_;
61 webrtc::VoEDtmf* voe_dtmf_;
62 webrtc::VoERTP_RTCP* voe_rtp_rtcp_;
63 webrtc::VoEAudioProcessing* voe_apm_;
64 webrtc::VoENetwork* voe_network_;
65 webrtc::VoEFile* voe_file_;
66 webrtc::VoEVideoSync* voe_vsync_;
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/external/webrtc/webrtc/modules/video_coding/codecs/test/
Dvideoprocessor.h27 namespace webrtc {
103 webrtc::VideoCodec* codec_settings;
110 const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
162 VideoProcessorImpl(webrtc::VideoEncoder* encoder,
163 webrtc::VideoDecoder* decoder,
175 void FrameEncoded(const webrtc::EncodedImage& encodedImage);
177 void FrameDecoded(const webrtc::VideoFrame& image);
180 int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
181 const webrtc::TickTime& stop);
193 webrtc::VideoEncoder* encoder_;
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/
Dconference_transport.cc23 static const webrtc::CodecInst kCodecInst =
40 : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), in ConferenceTransport()
41 stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), in ConferenceTransport()
42 packet_event_(webrtc::EventWrapper::Create()), in ConferenceTransport()
46 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { in ConferenceTransport()
48 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, in ConferenceTransport()
51 local_voe_ = webrtc::VoiceEngine::Create(); in ConferenceTransport()
52 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); in ConferenceTransport()
53 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); in ConferenceTransport()
54 local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_); in ConferenceTransport()
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