webkit  2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
Classes | Namespaces | Functions
call_perf_tests.cc File Reference
#include <algorithm>
#include <limits>
#include <memory>
#include <string>
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call/call.h"
#include "webrtc/config.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/drifting_clock.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/gtest.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/voice_engine/include/voe_base.h"

Classes

class  webrtc::CallPerfTest
 
class  webrtc::VideoRtcpAndSyncObserver
 

Namespaces

 webrtc
 

Functions

 webrtc::TEST_F (CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift)
 
 webrtc::TEST_F (CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift)
 
 webrtc::TEST_F (CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift)
 
 webrtc::TEST_F (CallPerfTest, CaptureNtpTimeWithNetworkDelay)
 
 webrtc::TEST_F (CallPerfTest, CaptureNtpTimeWithNetworkJitter)
 
 webrtc::TEST_F (CallPerfTest, ReceivesCpuOveruseAndUnderuse)
 
 webrtc::TEST_F (CallPerfTest, PadsToMinTransmitBitrate)
 
 webrtc::TEST_F (CallPerfTest, NoPadWithoutMinTransmitBitrate)
 
 webrtc::TEST_F (CallPerfTest, KeepsHighBitrateWhenReconfiguringSender)