/external/webrtc/webrtc/tools/frame_analyzer/ |
frame_analyzer.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 18 #include "webrtc/tools/frame_analyzer/video_quality_analysis.h" 19 #include "webrtc/tools/simple_command_line_parser.h" 60 webrtc::test::CommandLineParser parser; 89 webrtc::test::ResultsContainer results; 91 webrtc::test::RunAnalysis(parser.GetFlag("reference_file").c_str(), 97 webrtc::test::PrintAnalysisResults(label, &results); 98 webrtc::test::PrintMaxRepeatedAndSkippedFrames(label,
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/external/webrtc/talk/app/webrtc/objc/ |
RTCDataChannel.mm | 34 #include "talk/app/webrtc/datachannelinterface.h" 36 namespace webrtc { 70 // See https://code.google.com/p/webrtc/issues/detail?id=4773 for details. 86 webrtc::DataChannelInit _dataChannelInit; 141 - (const webrtc::DataChannelInit*)dataChannelInit { 148 rtc::scoped_ptr<webrtc::DataBuffer> _dataBuffer; 156 _dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary)); 174 - (instancetype)initWithDataBuffer:(const webrtc::DataBuffer&)buffer { 176 _dataBuffer.reset(new webrtc::DataBuffer(buffer)); 181 - (const webrtc::DataBuffer*)dataBuffer [all...] |
RTCMediaStreamTrack.mm | 35 namespace webrtc { 51 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _mediaTrack; 52 rtc::scoped_ptr<webrtc::RTCMediaStreamTrackObserver> _observer; 103 (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>) 113 _observer.reset(new webrtc::RTCMediaStreamTrackObserver(self)); 123 - (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)mediaTrack {
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RTCVideoRendererAdapter.mm | 35 namespace webrtc { 63 rtc::scoped_ptr<webrtc::RTCVideoRendererNativeAdapter> _adapter; 70 _adapter.reset(new webrtc::RTCVideoRendererNativeAdapter(self)); 75 - (webrtc::VideoRendererInterface*)nativeVideoRenderer {
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/external/webrtc/talk/app/webrtc/ |
sctputils_unittest.cc | 28 #include "talk/app/webrtc/sctputils.h" 29 #include "webrtc/base/bytebuffer.h" 30 #include "webrtc/base/gunit.h" 36 const webrtc::DataChannelInit& config) { 83 webrtc::DataChannelInit config; 88 ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet)); 93 webrtc::DataChannelInit output_config; 94 ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage( 105 webrtc::DataChannelInit config; 112 ASSERT_TRUE(webrtc::WriteDataChannelOpenMessage(label, config, &packet)) [all...] |
/external/webrtc/webrtc/call/ |
rtc_event_log.h | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 16 #include "webrtc/base/platform_file.h" 17 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/video_receive_stream.h" 19 #include "webrtc/video_send_stream.h" 21 namespace webrtc { namespace 56 // Logs configuration information for webrtc::VideoReceiveStream 58 const webrtc::VideoReceiveStream::Config& config) = 0; 60 // Logs configuration information for webrtc::VideoSendStream 62 const webrtc::VideoSendStream::Config& config) = 0 [all...] |
call_unittest.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 15 #include "webrtc/audio_state.h" 16 #include "webrtc/call.h" 17 #include "webrtc/test/mock_voice_engine.h" 23 webrtc::AudioState::Config audio_state_config; 25 webrtc::Call::Config config; 26 config.audio_state = webrtc::AudioState::Create(audio_state_config); 27 call_.reset(webrtc::Call::Create(config)); 30 webrtc::Call* operator->() { return call_.get(); } 33 testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_ 38 namespace webrtc { namespace [all...] |
rtc_event_log2rtp_dump.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 16 #include "webrtc/base/checks.h" 17 #include "webrtc/base/scoped_ptr.h" 18 #include "webrtc/call/rtc_event_log.h" 19 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 20 #include "webrtc/test/rtp_file_writer.h" 24 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" 26 #include "webrtc/call/rtc_event_log.pb.h" 97 webrtc::rtclog::EventStream event_stream [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
rtpcat.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 13 #include "webrtc/base/checks.h" 14 #include "webrtc/base/scoped_ptr.h" 15 #include "webrtc/test/rtp_file_reader.h" 16 #include "webrtc/test/rtp_file_writer.h" 19 using webrtc::test::RtpFileReader; 20 using webrtc::test::RtpFileWriter; 40 webrtc::test::RtpPacket packet;
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/external/webrtc/webrtc/modules/video_coding/codecs/test/ |
packet_manipulator.h | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 16 #include "webrtc/modules/video_coding/include/video_codec_interface.h" 17 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 18 #include "webrtc/test/testsupport/packet_reader.h" 20 namespace webrtc { namespace 86 virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0; 95 int ManipulatePackets(webrtc::EncodedImage* encoded_image) override; 113 } // namespace webrtc
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/external/webrtc/talk/media/webrtc/ |
webrtcvideoframe_unittest.cc | 31 #include "talk/media/webrtc/webrtcvideoframe.h" 32 #include "webrtc/test/fake_texture_frame.h" 59 webrtc::VideoRotation frame_rotation, 92 EXPECT_EQ(webrtc::kVideoRotation_0, frame.GetRotation()); 97 if (apply_rotation && (frame_rotation == webrtc::kVideoRotation_90 98 || frame_rotation == webrtc::kVideoRotation_270)) { 187 // Re-evaluate once WebRTC switches to libyuv 273 TestInit(640, 360, webrtc::kVideoRotation_0, true); 277 TestInit(601, 480, webrtc::kVideoRotation_0, true); 281 TestInit(360, 765, webrtc::kVideoRotation_0, true) [all...] |
simulcast.h | 33 #include "webrtc/base/basictypes.h" 34 #include "webrtc/config.h" 59 int GetTotalMaxBitrateBps(const std::vector<webrtc::VideoStream>& streams); 65 std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
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/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
RTPencode.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 28 #include "webrtc/base/checks.h" 29 #include "webrtc/typedefs.h" 32 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" 33 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 85 webrtc::NetEqDecoder* codec, 91 int NetEQTest_init_coders(webrtc::NetEqDecoder coder, 97 void defineCodecs(webrtc::NetEqDecoder* usedCodec, int* noOfCodecs); 98 int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels); 99 size_t NetEQTest_encode(webrtc::NetEqDecoder coder [all...] |
/external/webrtc/talk/app/webrtc/java/jni/ |
native_handle_impl.h | 34 #include "webrtc/common_video/include/video_frame_buffer.h" 35 #include "webrtc/common_video/rotation.h" 49 class AndroidTextureBuffer : public webrtc::NativeHandleBuffer { 62 webrtc::VideoRotation rotation);
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/external/webrtc/talk/app/webrtc/test/ |
androidtestinitializer.cc | 28 #include "talk/app/webrtc/test/androidtestinitializer.h" 39 #include "talk/app/webrtc/java/jni/classreferenceholder.h" 40 #include "talk/app/webrtc/java/jni/jni_helpers.h" 41 #include "webrtc/base/checks.h" 42 #include "webrtc/base/ssladapter.h" 43 #include "webrtc/voice_engine/include/voe_base.h" 45 namespace webrtc { namespace 65 webrtc::VoiceEngine::SetAndroidObjects(jvm, context); 74 } // namespace webrtc
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fakedatachannelprovider.h | 31 #include "talk/app/webrtc/datachannel.h" 33 class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { 60 bool ConnectDataChannel(webrtc::DataChannel* data_channel) override { 70 void DisconnectDataChannel(webrtc::DataChannel* data_channel) override { 101 for (webrtc::DataChannel *ch : std::set<webrtc::DataChannel*>( 122 std::set<webrtc::DataChannel*>::iterator it; 139 bool IsConnected(webrtc::DataChannel* data_channel) const { 157 std::set<webrtc::DataChannel*> connected_channels_;
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/external/webrtc/webrtc/api/objctests/ |
RTCMediaConstraintsTest.mm | 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 13 #include "webrtc/base/gunit.h" 15 #import "webrtc/api/objc/RTCMediaConstraints.h" 16 #import "webrtc/api/objc/RTCMediaConstraints+Private.h" 17 #import "webrtc/base/objc/NSString+StdString.h" 32 rtc::scoped_ptr<webrtc::MediaConstraints> nativeConstraints = 35 webrtc::MediaConstraintsInterface::Constraints nativeMandatory = 39 webrtc::MediaConstraintsInterface::Constraints nativeOptional = 46 (webrtc::MediaConstraintsInterface::Constraints)nativeConstraints {
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/external/webrtc/webrtc/ |
config.cc | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 10 #include "webrtc/config.h" 15 namespace webrtc { namespace 34 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"; 42 return name == webrtc::RtpExtension::kAbsSendTime || 43 name == webrtc::RtpExtension::kAudioLevel || 44 name == webrtc::RtpExtension::kTransportSequenceNumber; 48 return name == webrtc::RtpExtension::kTOffset || 49 name == webrtc::RtpExtension::kAbsSendTime || 50 name == webrtc::RtpExtension::kVideoRotation | [all...] |
/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
overuse_detector.h | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 15 #include "webrtc/base/constructormagic.h" 16 #include "webrtc/modules/include/module_common_types.h" 17 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" 18 #include "webrtc/typedefs.h" 20 namespace webrtc { namespace 54 webrtc::OverUseDetectorOptions options_; 64 } // namespace webrtc
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/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
pli.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h" 13 #include "webrtc/base/checks.h" 14 #include "webrtc/base/logging.h" 16 using webrtc::RTCPUtility::RtcpCommonHeader; 18 namespace webrtc { namespace 70 } // namespace webrtc
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/external/webrtc/webrtc/modules/video_coding/codecs/h264/ |
h264_objc.mm | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 12 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h" 18 namespace webrtc { 33 } // namespace webrtc
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/external/webrtc/webrtc/test/channel_transport/ |
udp_socket_manager_wrapper.h | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 14 #include "webrtc/system_wrappers/include/static_instance.h" 15 #include "webrtc/typedefs.h" 17 namespace webrtc { namespace 58 friend UdpSocketManager* webrtc::GetStaticInstance<UdpSocketManager>( 68 } // namespace webrtc
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/external/webrtc/webrtc/test/linux/ |
glx_renderer.h | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 17 #include "webrtc/test/gl/gl_renderer.h" 18 #include "webrtc/typedefs.h" 20 namespace webrtc { namespace 29 void RenderFrame(const webrtc::VideoFrame& frame, int delta) override; 46 } // webrtc
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/external/webrtc/webrtc/test/ |
test_suite.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/test/test_suite.h" 16 #include "webrtc/base/logging.h" 17 #include "webrtc/test/testsupport/fileutils.h" 18 #include "webrtc/test/testsupport/trace_to_stderr.h" 19 #include "webrtc/test/field_trial.h" 25 "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/" 26 " will assign the group Enable to field trial WebRTC-FooFeature."); 28 namespace webrtc { namespace 39 webrtc::test::InitFieldTrialsFromString(FLAGS_force_fieldtrials) [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
fake_media_process.h | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 15 class FakeMediaProcess : public webrtc::VoEMediaProcess { 19 const webrtc::ProcessingTypes type,
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