| /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,
|
| /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 {
|
| RTCVideoRendererAdapter.mm | 35 namespace webrtc {
63 rtc::scoped_ptr<webrtc::RTCVideoRendererNativeAdapter> _adapter;
70 _adapter.reset(new webrtc::RTCVideoRendererNativeAdapter(self));
75 - (webrtc::VideoRendererInterface*)nativeVideoRenderer {
|
| /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;
|
| /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
|
| /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,
|
| /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);
|
| /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
|
| 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_;
|
| /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 {
|
| /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
|
| /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
|
| /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
|
| /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
|
| /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
|
| /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,
|
