rtpsenderreceiver_unittest.cc File Reference

#include <memory>
#include <string>
#include <utility>
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/engine/fakewebrtccall.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/fakemediacontroller.h"
#include "webrtc/pc/localaudiosource.h"
#include "webrtc/pc/mediastream.h"
#include "webrtc/pc/remoteaudiosource.h"
#include "webrtc/pc/rtpreceiver.h"
#include "webrtc/pc/rtpsender.h"
#include "webrtc/pc/streamcollection.h"
#include "webrtc/pc/test/fakevideotracksource.h"
#include "webrtc/pc/videotrack.h"
#include "webrtc/pc/videotracksource.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"

Classes
class	webrtc::RtpSenderReceiverTest

Namespaces
	webrtc

Functions
	webrtc::TEST_F (RtpSenderReceiverTest, AddAndDestroyAudioRtpSender)
	webrtc::TEST_F (RtpSenderReceiverTest, AddAndDestroyVideoRtpSender)
	webrtc::TEST_F (RtpSenderReceiverTest, AddAndDestroyAudioRtpReceiver)
	webrtc::TEST_F (RtpSenderReceiverTest, AddAndDestroyVideoRtpReceiver)
	webrtc::TEST_F (RtpSenderReceiverTest, LocalAudioSourceOptionsApplied)
	webrtc::TEST_F (RtpSenderReceiverTest, LocalAudioTrackDisable)
	webrtc::TEST_F (RtpSenderReceiverTest, RemoteAudioTrackDisable)
	webrtc::TEST_F (RtpSenderReceiverTest, LocalVideoTrackDisable)
	webrtc::TEST_F (RtpSenderReceiverTest, RemoteVideoTrackState)
	webrtc::TEST_F (RtpSenderReceiverTest, RemoteVideoTrackDisable)
	webrtc::TEST_F (RtpSenderReceiverTest, RemoteAudioTrackSetVolume)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderSsrcSetToZero)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderSsrcSetToZero)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderTrackSetToNull)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderTrackSetToNull)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderSsrcChanged)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderSsrcChanged)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderCanSetParameters)
	webrtc::TEST_F (RtpSenderReceiverTest, SetAudioMaxSendBitrate)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderCanSetParameters)
	webrtc::TEST_F (RtpSenderReceiverTest, SetVideoMaxSendBitrate)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioReceiverCanSetParameters)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoReceiverCanSetParameters)
	webrtc::TEST_F (RtpSenderReceiverTest, PropagatesVideoTrackContentHint)
	webrtc::TEST_F (RtpSenderReceiverTest, PropagatesVideoTrackContentHintForScreencastSource)
	webrtc::TEST_F (RtpSenderReceiverTest, PropagatesVideoTrackContentHintSetBeforeEnabling)
	webrtc::TEST_F (RtpSenderReceiverTest, AudioSenderHasDtmfSender)
	webrtc::TEST_F (RtpSenderReceiverTest, VideoSenderDoesNotHaveDtmfSender)
	webrtc::TEST_F (RtpSenderReceiverTest, CanInsertDtmf)
	webrtc::TEST_F (RtpSenderReceiverTest, CanNotInsertDtmf)
	webrtc::TEST_F (RtpSenderReceiverTest, InsertDtmf)
	webrtc::TEST_F (RtpSenderReceiverTest, TestOnDestroyedSignal)