rtp_sender_unittest.cc File Reference

#include <memory>
#include <vector>
#include "webrtc/base/buffer.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_transport.h"
#include "webrtc/typedefs.h"

Classes
class	webrtc::MockRtpPacketSender
class	webrtc::MockTransportSequenceNumberAllocator
class	webrtc::MockSendPacketObserver
class	webrtc::MockTransportFeedbackObserver
class	webrtc::RtpSenderTest
class	webrtc::RtpSenderTestWithoutPacer
class	webrtc::RtpSenderVideoTest
class	webrtc::RtpSenderAudioTest

Namespaces
	webrtc

Functions
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AllocatePacketSetCsrc)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AllocatePacketReserveExtensions)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AssignSequenceNumberAdvanceSequenceNumber)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AssignSequenceNumberFailsOnNotSending)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, NoAllocationIfNotRegistered)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, OnSendPacketUpdated)
	webrtc::TEST_F (RtpSenderTest, SendsPacketsWithTransportSequenceNumber)
	webrtc::TEST_F (RtpSenderTest, TrafficSmoothingWithExtensions)
	webrtc::TEST_F (RtpSenderTest, TrafficSmoothingRetransmits)
	webrtc::TEST_F (RtpSenderTest, SendPadding)
	webrtc::TEST_F (RtpSenderTest, OnSendPacketUpdated)
	webrtc::TEST_F (RtpSenderTest, OnSendPacketNotUpdatedForRetransmits)
	webrtc::TEST_F (RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator)
	webrtc::TEST_F (RtpSenderTest, SendRedundantPayloads)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, SendGenericVideo)
	webrtc::TEST_F (RtpSenderTest, SendFlexfecPackets)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, SendFlexfecPackets)
	webrtc::TEST_F (RtpSenderTest, FecOverheadRate)
	webrtc::TEST_F (RtpSenderTest, FrameCountCallbacks)
	webrtc::TEST_F (RtpSenderTest, BitrateCallbacks)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, StreamDataCountersCallbacks)
	webrtc::TEST_F (RtpSenderAudioTest, SendAudio)
	webrtc::TEST_F (RtpSenderAudioTest, SendAudioWithAudioLevelExtension)
	webrtc::TEST_F (RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, BytesReportedCorrectly)
	webrtc::TEST_F (RtpSenderTestWithoutPacer, RespectsNackBitrateLimit)
	webrtc::TEST_F (RtpSenderVideoTest, KeyFrameHasCVO)
	webrtc::TEST_F (RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged)
	webrtc::TEST_F (RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero)
	webrtc::TEST_F (RtpSenderVideoTest, SendVideoWithCameraAndFlipCVO)
	webrtc::TEST_F (RtpSenderTest, OnOverheadChanged)
	webrtc::TEST_F (RtpSenderTest, DoesNotUpdateOverheadOnEqualSize)
	webrtc::TEST_F (RtpSenderTest, AddOverheadToTransportFeedbackObserver)
	webrtc::TEST_F (RtpSenderTest, SendAudioPadding)

Variable Documentation

last_packet_id_

int last_packet_id_

sent_packets_

std::vector<RtpPacketReceived> sent_packets_

total_bytes_sent_

size_t total_bytes_sent_