/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
rtp_to_text.cc | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 14 #include "webrtc/base/format_macros.h" 15 #include "webrtc/base/scoped_ptr.h" 16 #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" 17 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 18 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" 19 #include "webrtc/test/rtp_file_reader.h" 22 webrtc::test::RtpFileReader* reader; 23 webrtc::RtpHeaderParser* parser; 29 rtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(reader) [all...] |
/external/webrtc/webrtc/modules/video_coding/ |
encoded_frame.h | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 16 #include "webrtc/common_types.h" 17 #include "webrtc/common_video/include/video_image.h" 18 #include "webrtc/modules/include/module_common_types.h" 19 #include "webrtc/modules/video_coding/include/video_codec_interface.h" 20 #include "webrtc/modules/video_coding/include/video_coding_defines.h" 22 namespace webrtc { namespace 27 explicit VCMEncodedFrame(const webrtc::EncodedImage& rhs); 52 const webrtc::EncodedImage& EncodedImage() const { 53 return static_cast<const webrtc::EncodedImage&>(*this) [all...] |
/external/webrtc/webrtc/modules/video_coding/test/ |
video_source.h | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 16 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" 17 #include "webrtc/typedefs.h" 58 webrtc::VideoType type = webrtc::kI420); 63 webrtc::VideoType type = webrtc::kI420); 68 webrtc::VideoType GetType() const { return _type; } 81 webrtc::VideoType _type;
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tester_main.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 15 #include "webrtc/modules/video_coding/include/video_coding.h" 16 #include "webrtc/modules/video_coding/test/receiver_tests.h" 17 #include "webrtc/test/testsupport/fileutils.h" 24 webrtc::test::ProjectRootPath() + "/resources/foreman_cif.yuv", 27 webrtc::test::OutputPath() + 31 namespace webrtc { namespace 63 } // namespace webrtc 66 // Initialize WebRTC fileutils.h so paths to resources can be resolved. 67 webrtc::test::SetExecutablePath(argv[0]) [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
codec_test.cc | 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 14 #include "webrtc/call/rtc_event_log.h" 15 #include "webrtc/test/test_suite.h" 16 #include "webrtc/test/testsupport/fileutils.h" 17 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" 18 #include "webrtc/voice_engine/voice_engine_defines.h" 32 webrtc::CodecInst codec_instance_; 35 static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) { 64 webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid [all...] |
rtp_rtcp_extensions.cc | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/include/module_common_types.h" 12 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 13 #include "webrtc/system_wrappers/include/atomic32.h" 14 #include "webrtc/system_wrappers/include/sleep.h" 15 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h" 22 class ExtensionVerifyTransport : public webrtc::Transport { 25 : parser_(webrtc::RtpHeaderParser::Create()), 33 const webrtc::PacketOptions& options) override { 34 webrtc::RTPHeader header [all...] |
/external/webrtc/talk/media/webrtc/ |
fakewebrtcvoiceengine.h | 37 #include "talk/media/webrtc/fakewebrtccommon.h" 38 #include "talk/media/webrtc/webrtcvoe.h" 39 #include "webrtc/base/basictypes.h" 40 #include "webrtc/base/checks.h" 41 #include "webrtc/base/gunit.h" 42 #include "webrtc/base/stringutils.h" 43 #include "webrtc/config.h" 44 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" 45 #include "webrtc/modules/audio_processing/include/audio_processing.h" 58 class FakeAudioProcessing : public webrtc::AudioProcessing [all...] |
webrtcvideoframe.h | 32 #include "webrtc/base/buffer.h" 33 #include "webrtc/base/refcount.h" 34 #include "webrtc/base/scoped_ref_ptr.h" 35 #include "webrtc/common_types.h" 36 #include "webrtc/common_video/include/video_frame_buffer.h" 45 WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer, 47 webrtc::VideoRotation rotation); 65 webrtc::VideoRotation rotation); 86 webrtc::VideoRotation rotation, 101 rtc::scoped_refptr<webrtc::VideoFrameBuffer> GetVideoFrameBuffer( [all...] |
webrtcvideoframefactory_unittest.cc | 31 #include "talk/media/webrtc/webrtcvideoframe.h" 32 #include "talk/media/webrtc/webrtcvideoframefactory.h" 39 void InitFrame(webrtc::VideoRotation frame_rotation) { 61 webrtc::VideoRotation src_rotation, 70 EXPECT_EQ(dest_frame->GetRotation(), webrtc::kVideoRotation_0); 71 if (src_rotation == webrtc::kVideoRotation_90 || 72 src_rotation == webrtc::kVideoRotation_270) { 85 InitFrame(webrtc::kVideoRotation_270); 93 VerifyFrame(frame.get(), webrtc::kVideoRotation_270, new_width, new_height, 98 VerifyFrame(frame.get(), webrtc::kVideoRotation_270, new_width / 2 [all...] |
webrtcvoiceengine.h | 36 #include "talk/media/webrtc/webrtccommon.h" 37 #include "talk/media/webrtc/webrtcvoe.h" 39 #include "webrtc/audio_state.h" 40 #include "webrtc/base/buffer.h" 41 #include "webrtc/base/scoped_ptr.h" 42 #include "webrtc/base/stream.h" 43 #include "webrtc/base/thread_checker.h" 44 #include "webrtc/call.h" 45 #include "webrtc/common.h" 46 #include "webrtc/config.h [all...] |
/external/webrtc/talk/app/webrtc/java/jni/ |
native_handle_impl.cc | 28 #include "talk/app/webrtc/java/jni/native_handle_impl.h" 30 #include "talk/app/webrtc/java/jni/jni_helpers.h" 31 #include "webrtc/base/bind.h" 32 #include "webrtc/base/checks.h" 33 #include "webrtc/base/keep_ref_until_done.h" 34 #include "webrtc/base/scoped_ptr.h" 35 #include "webrtc/base/scoped_ref_ptr.h" 36 #include "webrtc/base/logging.h" 38 using webrtc::NativeHandleBuffer; 42 void RotateMatrix(float a[16], webrtc::VideoRotation rotation) [all...] |
androidvideocapturer_jni.h | 34 #include "talk/app/webrtc/androidvideocapturer.h" 35 #include "talk/app/webrtc/java/jni/jni_helpers.h" 36 #include "webrtc/base/asyncinvoker.h" 37 #include "webrtc/base/criticalsection.h" 38 #include "webrtc/base/thread_checker.h" 39 #include "webrtc/common_video/include/i420_buffer_pool.h" 49 class AndroidVideoCapturerJni : public webrtc::AndroidVideoCapturerDelegate { 58 webrtc::AndroidVideoCapturer* capturer) override; 88 void (webrtc::AndroidVideoCapturer::*method)(Args...), 96 webrtc::I420BufferPool buffer_pool_ [all...] |
/external/webrtc/webrtc/modules/video_coding/codecs/tools/ |
video_quality_measurement.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 23 #include "webrtc/base/format_macros.h" 24 #include "webrtc/common_types.h" 25 #include "webrtc/modules/video_coding/codecs/test/packet_manipulator.h" 26 #include "webrtc/modules/video_coding/codecs/test/stats.h" 27 #include "webrtc/modules/video_coding/codecs/test/videoprocessor.h" 28 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h" 29 #include "webrtc/modules/video_coding/include/video_coding.h" 30 #include "webrtc/system_wrappers/include/trace.h" 31 #include "webrtc/test/testsupport/frame_reader.h [all...] |
/external/webrtc/webrtc/modules/video_capture/linux/ |
device_info_linux.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/video_capture/linux/device_info_linux.h" 23 #include "webrtc/system_wrappers/include/ref_count.h" 24 #include "webrtc/system_wrappers/include/trace.h" 27 namespace webrtc namespace 60 WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__); 89 WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__); 119 WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id [all...] |
/external/webrtc/webrtc/api/objctests/ |
RTCSessionDescriptionTest.mm | 2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 13 #include "webrtc/base/gunit.h" 15 #import "webrtc/api/objc/RTCSessionDescription.h" 16 #import "webrtc/api/objc/RTCSessionDescription+Private.h" 17 #import "webrtc/base/objc/NSString+StdString.h" 35 webrtc::SessionDescriptionInterface *nativeDescription = 47 webrtc::SessionDescriptionInterface *nativeDescription; 49 nativeDescription = webrtc::CreateSessionDescription( 50 webrtc::SessionDescriptionInterface::kAnswer, 57 EXPECT_EQ(webrtc::SessionDescriptionInterface::kAnswer [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_performance_test.cc | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h" 13 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" 14 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" 15 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" 16 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" 17 #include "webrtc/system_wrappers/include/clock.h" 18 #include "webrtc/test/testsupport/fileutils.h" 19 #include "webrtc/typedefs.h" 21 using webrtc::NetEq 26 namespace webrtc { namespace [all...] |
/external/webrtc/webrtc/modules/video_capture/windows/ |
device_info_ds.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/video_capture/windows/device_info_ds.h" 13 #include "webrtc/modules/video_capture/video_capture_config.h" 14 #include "webrtc/modules/video_capture/video_capture_delay.h" 15 #include "webrtc/modules/video_capture/windows/help_functions_ds.h" 16 #include "webrtc/system_wrappers/include/ref_count.h" 17 #include "webrtc/system_wrappers/include/trace.h" 22 namespace webrtc namespace 92 WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id [all...] |
/external/webrtc/talk/app/webrtc/objc/ |
RTCPeerConnection.mm | 48 #include "talk/app/webrtc/jsep.h" 53 namespace webrtc { 144 rtc::scoped_ptr<webrtc::RTCPeerConnectionObserver> _observer; 145 rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection; 149 rtc::scoped_ptr<const webrtc::IceCandidateInterface> iceCandidate( 166 rtc::scoped_refptr<webrtc::DataChannelInterface> dataChannel = 174 rtc::scoped_refptr<webrtc::RTCCreateSessionDescriptionObserver> 176 webrtc::RTCCreateSessionDescriptionObserver>(delegate, self)); 182 rtc::scoped_refptr<webrtc::RTCCreateSessionDescriptionObserver> 184 webrtc::RTCCreateSessionDescriptionObserver>(delegate, self)) [all...] |
/external/webrtc/webrtc/common_video/ |
i420_buffer_pool.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/common_video/include/i420_buffer_pool.h" 13 #include "webrtc/base/checks.h" 18 class PooledI420Buffer : public webrtc::VideoFrameBuffer { 21 const rtc::scoped_refptr<webrtc::I420Buffer>& buffer) 29 const uint8_t* data(webrtc::PlaneType type) const override { 32 uint8_t* MutableData(webrtc::PlaneType type) override { 38 int stride(webrtc::PlaneType type) const override { 49 rtc::scoped_refptr<webrtc::I420Buffer> buffer_; 54 namespace webrtc { namespace [all...] |
/external/webrtc/talk/app/webrtc/ |
datachannel_unittest.cc | 28 #include "talk/app/webrtc/datachannel.h" 29 #include "talk/app/webrtc/sctputils.h" 30 #include "talk/app/webrtc/test/fakedatachannelprovider.h" 31 #include "webrtc/base/gunit.h" 33 using webrtc::DataChannel; 34 using webrtc::SctpSidAllocator; 36 class FakeDataChannelObserver : public webrtc::DataChannelObserver { 51 void OnMessage(const webrtc::DataBuffer& buffer) { 103 webrtc::InternalDataChannelInit init_; 137 EXPECT_EQ(webrtc::DataChannelInterface::kConnecting [all...] |
localaudiosource_unittest.cc | 28 #include "talk/app/webrtc/localaudiosource.h" 33 #include "talk/app/webrtc/test/fakeconstraints.h" 36 #include "webrtc/base/gunit.h" 38 using webrtc::LocalAudioSource; 39 using webrtc::MediaConstraintsInterface; 40 using webrtc::MediaSourceInterface; 41 using webrtc::PeerConnectionFactoryInterface; 44 webrtc::FakeConstraints constraints; 72 webrtc::FakeConstraints constraints; 80 webrtc::FakeConstraints constraints [all...] |
/external/webrtc/webrtc/tools/psnr_ssim_analyzer/ |
psnr_ssim_analyzer.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 19 #include "webrtc/tools/frame_analyzer/video_quality_analysis.h" 20 #include "webrtc/tools/simple_command_line_parser.h" 34 int size = webrtc::test::GetI420FrameSize(width, height); 43 read_result &= (y4m_mode) ? webrtc::test::ExtractFrameFromY4mFile( 45 webrtc::test::ExtractFrameFromYuvFile(reference_file_name, width, 47 read_result &= webrtc::test::ExtractFrameFromYuvFile(test_file_name, width, 54 double result_psnr = webrtc::test::CalculateMetrics( 55 webrtc::test::kPSNR, ref_frame, test_frame, width, height); 56 double result_ssim = webrtc::test::CalculateMetrics [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/extended/ |
agc_config_test.cc | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" 22 webrtc::AgcConfig default_agc_config_; 26 webrtc::AgcConfig agc_config; 37 webrtc::AgcConfig agc_config = default_agc_config_; 51 webrtc::AgcConfig agc_config; 56 webrtc::AgcConfig actual_config; 69 webrtc::AgcConfig agc_config; 80 webrtc::AgcConfig agc_config = default_agc_config_; 94 webrtc::AgcConfig agc_config [all...] |
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
app_unittest.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" 18 using webrtc::rtcp::App; 19 using webrtc::rtcp::RawPacket; 20 using webrtc::RTCPUtility::RtcpCommonHeader; 21 using webrtc::RTCPUtility::RtcpParseCommonHeader; 23 namespace webrtc { namespace 81 } // namespace webrtc
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pli_unittest.cc | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h" 14 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" 16 using webrtc::rtcp::Pli; 17 using webrtc::rtcp::RawPacket; 18 using webrtc::RTCPUtility::RtcpCommonHeader; 19 using webrtc::RTCPUtility::RtcpParseCommonHeader; 21 namespace webrtc { namespace 66 } // namespace webrtc
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