Searched refs:webrtc (Results 1 - 25 of 2804) sorted by relevance

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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Dacm_neteq_unittest.cc13 namespace webrtc { namespace
/external/chromium_org/third_party/webrtc/common_audio/
H A Dwav_header_unittest.cc14 #include "webrtc/common_audio/wav_header.h"
15 #include "webrtc/system_wrappers/interface/compile_assert.h"
21 EXPECT_TRUE(webrtc::CheckWavParameters(1, 8000, webrtc::kWavFormatPcm, 1, 0));
23 webrtc::CheckWavParameters(0, 8000, webrtc::kWavFormatPcm, 1, 0));
25 webrtc::CheckWavParameters(-1, 8000, webrtc::kWavFormatPcm, 1, 0));
26 EXPECT_FALSE(webrtc::CheckWavParameters(1, 0, webrtc
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/external/chromium_org/third_party/webrtc/system_wrappers/source/
H A Dlogcat_trace_context.cc11 #include "webrtc/system_wrappers/interface/logcat_trace_context.h"
16 #include "webrtc/system_wrappers/interface/logging.h"
18 namespace webrtc { namespace
23 // to DEBUG because they are highly verbose in webrtc code (which is
26 case webrtc::kTraceStateInfo: return ANDROID_LOG_DEBUG;
27 case webrtc::kTraceWarning: return ANDROID_LOG_WARN;
28 case webrtc::kTraceError: return ANDROID_LOG_ERROR;
29 case webrtc::kTraceCritical: return ANDROID_LOG_FATAL;
30 case webrtc::kTraceApiCall: return ANDROID_LOG_VERBOSE;
31 case webrtc
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/external/chromium_org/third_party/webrtc/modules/audio_device/dummy/
H A Daudio_device_utility_dummy.cc10 #include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h"
12 namespace webrtc { namespace
14 } // namespace webrtc
/external/chromium_org/third_party/webrtc/modules/video_capture/
H A Densure_initialized.h11 namespace webrtc { namespace
14 // Ensure any necessary initialization of webrtc::videocapturemodule has
19 } // namespace webrtc.
/external/chromium_org/third_party/webrtc/test/testsupport/
H A Dalways_passing_unittest.cc13 namespace webrtc { namespace
19 } // namespace webrtc
/external/chromium_org/remoting/host/
H A Dscreen_resolution.h10 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
18 ScreenResolution(const webrtc::DesktopSize& dimensions,
19 const webrtc::DesktopVector& dpi);
22 webrtc::DesktopSize ScaleDimensionsToDpi(
23 const webrtc::DesktopVector& new_dpi) const;
26 const webrtc::DesktopSize& dimensions() const { return dimensions_; }
29 const webrtc::DesktopVector& dpi() const { return dpi_; }
39 webrtc::DesktopSize dimensions_;
40 webrtc::DesktopVector dpi_;
H A Dscreen_resolution_unittest.cc16 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10));
20 webrtc::DesktopSize(), webrtc::DesktopVector(10, 10));
24 webrtc::DesktopSize(1, 1), webrtc::DesktopVector(0, 0));
30 webrtc::DesktopSize(100, 100), webrtc::DesktopVector(10, 10));
32 EXPECT_TRUE(webrtc::DesktopSize(50, 50).equals(
33 resolution.ScaleDimensionsToDpi(webrtc
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/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/
H A Dtb_interfaces.h16 #include "webrtc/base/constructormagic.h"
17 #include "webrtc/common_types.h"
18 #include "webrtc/video_engine/include/vie_base.h"
19 #include "webrtc/video_engine/include/vie_capture.h"
20 #include "webrtc/video_engine/include/vie_codec.h"
21 #include "webrtc/video_engine/include/vie_image_process.h"
22 #include "webrtc/video_engine/include/vie_network.h"
23 #include "webrtc/video_engine/include/vie_render.h"
24 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
25 #include "webrtc/video_engin
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/primitives/
H A Dbase_primitives.h14 namespace webrtc { namespace
26 void TestI420CallSetup(webrtc::ViECodec* codec_interface,
27 webrtc::VideoEngine* video_engine,
28 webrtc::ViEBase* base_interface,
29 webrtc::ViENetwork* network_interface,
30 webrtc::ViERTP_RTCP* rtp_rtcp_interface,
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/
H A Dbefore_initialization_fixture.h16 #include "webrtc/common.h"
17 #include "webrtc/common_types.h"
18 #include "webrtc/engine_configurations.h"
19 #include "webrtc/test/testsupport/gtest_disable.h"
20 #include "webrtc/voice_engine/include/voe_audio_processing.h"
21 #include "webrtc/voice_engine/include/voe_base.h"
22 #include "webrtc/voice_engine/include/voe_codec.h"
23 #include "webrtc/voice_engine/include/voe_dtmf.h"
24 #include "webrtc/voice_engine/include/voe_errors.h"
25 #include "webrtc/voice_engin
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/external/chromium_org/third_party/webrtc/modules/video_capture/mac/
H A 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/interface/ref_count.h"
22 #include "webrtc/system_wrappers/interface/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
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/external/chromium_org/remoting/client/plugin/
H A Dpepper_view.h19 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
20 #include "third_party/webrtc/modules/desktop_capture/desktop_region.h"
26 namespace webrtc { namespace
28 } // namespace webrtc
48 virtual void ApplyBuffer(const webrtc::DesktopSize& view_size,
49 const webrtc::DesktopRect& clip_area,
50 webrtc::DesktopFrame* buffer,
51 const webrtc::DesktopRegion& region,
52 const webrtc::DesktopRegion& shape) OVERRIDE;
53 virtual void ReturnBuffer(webrtc
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/
H A 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:
55 (webrtc
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/external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/tools/
H A Dbwe_rtp.h16 namespace webrtc { namespace
29 webrtc::Clock* clock,
30 webrtc::RemoteBitrateObserver* observer,
31 webrtc::test::RtpFileReader** rtp_reader,
32 webrtc::RtpHeaderParser** parser,
33 webrtc::RemoteBitrateEstimator** estimator,
H A Dbwe_rtp.cc11 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
16 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
17 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
19 #include "webrtc/test/rtp_file_reader.h"
25 webrtc::Clock* clock,
26 webrtc::RemoteBitrateObserver* observer,
27 webrtc::test::RtpFileReader** rtp_reader,
28 webrtc::RtpHeaderParser** parser,
29 webrtc
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/external/chromium_org/third_party/webrtc/tools/force_mic_volume_max/
H A Dforce_mic_volume_max.cc15 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
16 #include "webrtc/test/channel_transport/include/channel_transport.h"
17 #include "webrtc/voice_engine/include/voe_audio_processing.h"
18 #include "webrtc/voice_engine/include/voe_base.h"
19 #include "webrtc/voice_engine/include/voe_volume_control.h"
22 webrtc::VoiceEngine* voe = webrtc::VoiceEngine::Create();
28 webrtc::VoEBase* base = webrtc::VoEBase::GetInterface(voe);
29 webrtc
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
H A Dwebrtcvoe.h32 #include "talk/media/webrtc/webrtccommon.h"
33 #include "webrtc/base/common.h"
35 #include "webrtc/common_types.h"
36 #include "webrtc/modules/audio_device/include/audio_device.h"
37 #include "webrtc/voice_engine/include/voe_audio_processing.h"
38 #include "webrtc/voice_engine/include/voe_base.h"
39 #include "webrtc/voice_engine/include/voe_codec.h"
40 #include "webrtc/voice_engine/include/voe_dtmf.h"
41 #include "webrtc/voice_engine/include/voe_errors.h"
42 #include "webrtc/voice_engin
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/external/chromium_org/remoting/codec/
H A Dvideo_decoder.h11 namespace webrtc { namespace
15 } // namespace webrtc
30 virtual void Initialize(const webrtc::DesktopSize& screen_size) = 0;
39 virtual void Invalidate(const webrtc::DesktopSize& view_size,
40 const webrtc::DesktopRegion& region) = 0;
54 virtual void RenderFrame(const webrtc::DesktopSize& view_size,
55 const webrtc::DesktopRect& clip_area,
58 webrtc::DesktopRegion* output_region) = 0;
62 virtual const webrtc::DesktopRegion* GetImageShape() = 0;
/external/chromium_org/third_party/webrtc/modules/audio_device/test/android/audio_device_android_test/src/org/webrtc/voiceengine/
H A DAudioDeviceAndroid.java1 ../../../../../../../source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
/external/chromium_org/tools/perf/benchmarks/
H A Dwebrtc.py5 from measurements import webrtc namespace
12 test = webrtc.WebRTC
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/
H A Dfakemediastreamsignaling.h32 #include "talk/app/webrtc/audiotrack.h"
33 #include "talk/app/webrtc/mediastreamsignaling.h"
34 #include "talk/app/webrtc/videotrack.h"
44 class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
45 public webrtc::MediaStreamSignalingObserver {
48 webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
89 virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {
91 virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {
93 virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {
95 virtual void OnAddLocalAudioTrack(webrtc
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dvideo_rtp_play.cc11 #include "webrtc/modules/video_coding/main/test/receiver_tests.h"
12 #include "webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.h"
13 #include "webrtc/system_wrappers/interface/trace.h"
14 #include "webrtc/test/testsupport/fileutils.h"
19 const webrtc::VCMVideoProtection kConfigProtectionMethod =
20 webrtc::kProtectionNack;
31 std::string trace_file = webrtc::test::OutputPath() + "receiverTestTrace.txt";
32 webrtc::Trace::CreateTrace();
33 webrtc::Trace::SetTraceFile(trace_file.c_str());
34 webrtc
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/external/chromium_org/remoting/base/
H A Dutil.h11 #include "third_party/webrtc/modules/desktop_capture/desktop_geometry.h"
45 const webrtc::DesktopSize& source_size,
46 const webrtc::DesktopRect& source_buffer_rect,
49 const webrtc::DesktopSize& dest_size,
50 const webrtc::DesktopRect& dest_buffer_rect,
51 const webrtc::DesktopRect& dest_rect);
56 webrtc::DesktopRect AlignRect(const webrtc::DesktopRect& rect);
61 webrtc::DesktopRect ScaleRect(const webrtc
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/
H A DAudioSource.java28 package org.webrtc;

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