webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Classes | Namespaces | Functions
end_to_end_tests.cc File Reference
#include <algorithm>
#include <list>
#include <map>
#include <memory>
#include <sstream>
#include <string>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/file.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/random.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/call/call.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakevideorenderer.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/null_transport.h"
#include "webrtc/test/rtcp_packet_parser.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/transport_adapter.h"
#include "webrtc/video_encoder.h"

Classes

class  webrtc::EndToEndTest
 
class  webrtc::EndToEndTest::UnusedTransport
 
class  webrtc::EndToEndTest::RequiredTransport
 
class  webrtc::CodecObserver
 
class  webrtc::FlexfecRenderObserver
 
class  webrtc::MultiStreamTest
 
struct  webrtc::MultiStreamTest::CodecSettings
 
class  webrtc::TransportFeedbackTester
 
class  webrtc::BandwidthStatsTest
 
class  webrtc::RtcpXrObserver
 
class  webrtc::EndToEndLogTest
 

Namespaces

 webrtc
 

Functions

 webrtc::INSTANTIATE_TEST_CASE_P (TestWithNewVideoJitterBuffer, EndToEndTest, ::testing::Values(new_jb_enabled, new_jb_disabled))
 
 webrtc::TEST_P (EndToEndTest, ReceiverCanBeStartedTwice)
 
 webrtc::TEST_P (EndToEndTest, ReceiverCanBeStoppedTwice)
 
 webrtc::TEST_P (EndToEndTest, ReceiverCanBeStoppedAndRestarted)
 
 webrtc::TEST_P (EndToEndTest, RendersSingleDelayedFrame)
 
 webrtc::TEST_P (EndToEndTest, TransmitsFirstFrame)
 
 webrtc::TEST_P (EndToEndTest, SendsAndReceivesVP8)
 
 webrtc::TEST_P (EndToEndTest, SendsAndReceivesVP8Rotation90)
 
 webrtc::TEST_P (EndToEndTest, SendsAndReceivesVP9)
 
 webrtc::TEST_P (EndToEndTest, SendsAndReceivesVP9VideoRotation90)
 
 webrtc::TEST_P (EndToEndTest, ReceiverUsesLocalSsrc)
 
 webrtc::TEST_P (EndToEndTest, ReceivesAndRetransmitsNack)
 
 webrtc::TEST_P (EndToEndTest, ReceivesNackAndRetransmitsAudio)
 
 webrtc::TEST_P (EndToEndTest, DISABLED_ReceivesUlpfec)
 
 webrtc::TEST_P (EndToEndTest, RecoversWithFlexfec)
 
 webrtc::TEST_P (EndToEndTest, RecoversWithFlexfecAndNack)
 
 webrtc::TEST_P (EndToEndTest, RecoversWithFlexfecAndSendsCorrespondingRtcp)
 
 webrtc::TEST_P (EndToEndTest, ReceivedUlpfecPacketsNotNacked)
 
 webrtc::TEST_P (EndToEndTest, DecodesRetransmittedFrame)
 
 webrtc::TEST_P (EndToEndTest, DecodesRetransmittedFrameOverRtx)
 
 webrtc::TEST_P (EndToEndTest, DecodesRetransmittedFrameByRed)
 
 webrtc::TEST_P (EndToEndTest, DecodesRetransmittedFrameByRedOverRtx)
 
 webrtc::TEST_P (EndToEndTest, ReceivesPliAndRecoversWithNack)
 
 webrtc::TEST_P (EndToEndTest, ReceivesPliAndRecoversWithoutNack)
 
 webrtc::TEST_P (EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode)
 
 webrtc::TEST_P (EndToEndTest, UsesRtcpCompoundMode)
 
 webrtc::TEST_P (EndToEndTest, UsesRtcpReducedSizeMode)
 
 webrtc::TEST_P (EndToEndTest, SendsAndReceivesMultipleStreams)
 
 webrtc::TEST_P (EndToEndTest, AssignsTransportSequenceNumbers)
 
 webrtc::TEST_P (EndToEndTest, VideoReceivesTransportFeedback)
 
 webrtc::TEST_P (EndToEndTest, VideoTransportFeedbackNotConfigured)
 
 webrtc::TEST_P (EndToEndTest, AudioReceivesTransportFeedback)
 
 webrtc::TEST_P (EndToEndTest, AudioTransportFeedbackNotConfigured)
 
 webrtc::TEST_P (EndToEndTest, AudioVideoReceivesTransportFeedback)
 
 webrtc::TEST_P (EndToEndTest, ObserversEncodedFrames)
 
 webrtc::TEST_P (EndToEndTest, ReceiveStreamSendsRemb)
 
 webrtc::TEST_P (EndToEndTest, VerifySendSideBweStats)
 
 webrtc::TEST_P (EndToEndTest, VerifyRecvSideBweStats)
 
 webrtc::TEST_P (EndToEndTest, RembWithSendSideBwe)
 
 webrtc::TEST_P (EndToEndTest, VerifyNackStats)
 
 webrtc::TEST_P (EndToEndTest, VerifyHistogramStatsWithRtx)
 
 webrtc::TEST_P (EndToEndTest, VerifyHistogramStatsWithRed)
 
 webrtc::TEST_P (EndToEndTest, VerifyHistogramStatsWithScreenshare)
 
 webrtc::TEST_P (EndToEndTest, ReportsSetEncoderRates)
 
 webrtc::TEST_P (EndToEndTest, GetStats)
 
 webrtc::TEST_P (EndToEndTest, TestExtendedReportsWithRrtrWithoutTargetBitrate)
 
 webrtc::TEST_P (EndToEndTest, TestExtendedReportsWithoutRrtrWithoutTargetBitrate)
 
 webrtc::TEST_P (EndToEndTest, TestExtendedReportsWithRrtrWithTargetBitrate)
 
 webrtc::TEST_P (EndToEndTest, TestExtendedReportsWithoutRrtrWithTargetBitrate)
 
 webrtc::TEST_P (EndToEndTest, TestReceivedRtpPacketStats)
 
 webrtc::TEST_P (EndToEndTest, SendsSetSsrc)
 
 webrtc::TEST_P (EndToEndTest, SendsSetSimulcastSsrcs)
 
 webrtc::TEST_P (EndToEndTest, CanSwitchToUseAllSsrcs)
 
 webrtc::TEST_P (EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs)
 
 webrtc::TEST_P (EndToEndTest, RestartingSendStreamPreservesRtpState)
 
 webrtc::TEST_P (EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx)
 
 webrtc::TEST_P (EndToEndTest, RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced)
 
 webrtc::TEST_P (EndToEndTest, RespectsNetworkState)
 
 webrtc::TEST_P (EndToEndTest, CallReportsRttForSender)
 
 webrtc::TEST_P (EndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown)
 
 webrtc::TEST_P (EndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown)
 
 webrtc::TEST_P (EndToEndTest, NewVideoReceiveStreamsRespectVideoNetworkDown)
 
 webrtc::TEST_P (EndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown)
 
void webrtc::VerifyEmptyNackConfig (const NackConfig &config)
 
void webrtc::VerifyEmptyUlpfecConfig (const UlpfecConfig &config)
 
void webrtc::VerifyEmptyFlexfecConfig (const VideoSendStream::Config::Rtp::Flexfec &config)
 
 webrtc::TEST_P (EndToEndTest, VerifyDefaultSendConfigParameters)
 
 webrtc::TEST_P (EndToEndTest, VerifyDefaultVideoReceiveConfigParameters)
 
 webrtc::TEST_P (EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters)
 
 webrtc::TEST_P (EndToEndTest, TransportSeqNumOnAudioAndVideo)
 
 webrtc::TEST_P (EndToEndLogTest, LogsEncodedFramesWhenRequested)