rtc_event_log_unittest.cc File Reference

#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/base/buffer.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/random.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.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_to_send.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"

Classes
class	webrtc::ConfigReadWriteTest
class	webrtc::AudioReceiveConfigReadWriteTest
class	webrtc::AudioSendConfigReadWriteTest
class	webrtc::VideoReceiveConfigReadWriteTest
class	webrtc::VideoSendConfigReadWriteTest
class	webrtc::AudioNetworkAdaptationReadWriteTest

Namespaces
	webrtc

Functions
RtpPacketToSend	webrtc::GenerateRtpPacket (const RtpHeaderExtensionMap *extensions, uint32_t csrcs_count, size_t packet_size, Random *prng)
rtc::Buffer	webrtc::GenerateRtcpPacket (Random *prng)
void	webrtc::GenerateVideoReceiveConfig (uint32_t extensions_bitvector, VideoReceiveStream::Config *config, Random *prng)
void	webrtc::GenerateVideoSendConfig (uint32_t extensions_bitvector, VideoSendStream::Config *config, Random *prng)
void	webrtc::GenerateAudioReceiveConfig (uint32_t extensions_bitvector, AudioReceiveStream::Config *config, Random *prng)
void	webrtc::GenerateAudioSendConfig (uint32_t extensions_bitvector, AudioSendStream::Config *config, Random *prng)
void	webrtc::GenerateAudioNetworkAdaptation (uint32_t extensions_bitvector, AudioNetworkAdaptor::EncoderRuntimeConfig *config, Random *prng)
void	webrtc::LogSessionAndReadBack (size_t rtp_count, size_t rtcp_count, size_t playout_count, size_t bwe_loss_count, uint32_t extensions_bitvector, uint32_t csrcs_count, unsigned int random_seed)
	webrtc::TEST (RtcEventLogTest, LogSessionAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogEventAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogLossBasedBweUpdateAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogDelayBasedBweUpdateAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogProbeClusterCreatedAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogProbeResultSuccessAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogProbeResultFailureAndReadBack)
	webrtc::TEST (RtcEventLogTest, LogAudioReceiveConfig)
	webrtc::TEST (RtcEventLogTest, LogAudioSendConfig)
	webrtc::TEST (RtcEventLogTest, LogVideoReceiveConfig)
	webrtc::TEST (RtcEventLogTest, LogVideoSendConfig)
	webrtc::TEST (RtcEventLogTest, LogAudioNetworkAdaptation)