| /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;
|
| 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
|
| 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
|
