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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::MockRtcEventLog, including all inherited members.
| Create() | webrtc::RtcEventLog | static |
| Create(const Clock *clock) | webrtc::RtcEventLog | inlinestatic |
| Create() | webrtc::RtcEventLog | static |
| Create(const Clock *clock) | webrtc::RtcEventLog | inlinestatic |
| CreateNull() | webrtc::RtcEventLog | static |
| CreateNull() | webrtc::RtcEventLog | static |
| LogAudioNetworkAdaptation(const AudioNetworkAdaptor::EncoderRuntimeConfig &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioNetworkAdaptation(const AudioNetworkAdaptor::EncoderRuntimeConfig &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioPlayout(uint32_t ssrc)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioPlayout(uint32_t ssrc)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioReceiveStreamConfig(const webrtc::AudioReceiveStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioReceiveStreamConfig(const webrtc::AudioReceiveStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioSendStreamConfig(const webrtc::AudioSendStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogAudioSendStreamConfig(const webrtc::AudioSendStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogDelayBasedBweUpdate(int32_t bitrate_bps, BandwidthUsage detector_state)=0 | webrtc::RtcEventLog | pure virtual |
| LogDelayBasedBweUpdate(int32_t bitrate_bps, BandwidthUsage detector_state)=0 | webrtc::RtcEventLog | pure virtual |
| LogLossBasedBweUpdate(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets)=0 | webrtc::RtcEventLog | pure virtual |
| LogLossBasedBweUpdate(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeClusterCreated(int id, int bitrate_bps, int min_probes, int min_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeClusterCreated(int id, int bitrate_bps, int min_probes, int min_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeResultFailure(int id, ProbeFailureReason failure_reason)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeResultFailure(int id, ProbeFailureReason failure_reason)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeResultSuccess(int id, int bitrate_bps)=0 | webrtc::RtcEventLog | pure virtual |
| LogProbeResultSuccess(int id, int bitrate_bps)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtcpPacket(PacketDirection direction, MediaType media_type, const uint8_t *packet, size_t length)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtcpPacket(PacketDirection direction, MediaType media_type, const uint8_t *packet, size_t length)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtpHeader(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtpHeader(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length, int probe_cluster_id)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtpHeader(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length)=0 | webrtc::RtcEventLog | pure virtual |
| LogRtpHeader(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length, int probe_cluster_id)=0 | webrtc::RtcEventLog | pure virtual |
| LogVideoReceiveStreamConfig(const webrtc::VideoReceiveStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogVideoReceiveStreamConfig(const webrtc::VideoReceiveStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogVideoSendStreamConfig(const webrtc::VideoSendStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| LogVideoSendStreamConfig(const webrtc::VideoSendStream::Config &config)=0 | webrtc::RtcEventLog | pure virtual |
| MOCK_METHOD0(StopLogging, void()) | webrtc::MockRtcEventLog | |
| MOCK_METHOD0(StopLogging, void()) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogVideoReceiveStreamConfig, void(const webrtc::VideoReceiveStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogVideoSendStreamConfig, void(const webrtc::VideoSendStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioReceiveStreamConfig, void(const webrtc::AudioReceiveStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioSendStreamConfig, void(const webrtc::AudioSendStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioNetworkAdaptation, void(const AudioNetworkAdaptor::EncoderRuntimeConfig &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogVideoReceiveStreamConfig, void(const webrtc::VideoReceiveStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogVideoSendStreamConfig, void(const webrtc::VideoSendStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioReceiveStreamConfig, void(const webrtc::AudioReceiveStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioSendStreamConfig, void(const webrtc::AudioSendStream::Config &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD1(LogAudioNetworkAdaptation, void(const AudioNetworkAdaptor::EncoderRuntimeConfig &config)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(StartLogging, bool(const std::string &file_name, int64_t max_size_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(StartLogging, bool(rtc::PlatformFile log_file, int64_t max_size_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogDelayBasedBweUpdate, void(int32_t bitrate_bps, BandwidthUsage detector_state)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogProbeResultFailure, void(int id, ProbeFailureReason failure_reason)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(StartLogging, bool(const std::string &file_name, int64_t max_size_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(StartLogging, bool(rtc::PlatformFile log_file, int64_t max_size_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogDelayBasedBweUpdate, void(int32_t bitrate_bps, BandwidthUsage detector_state)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogProbeResultSuccess, void(int id, int bitrate_bps)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD2(LogProbeResultFailure, void(int id, ProbeFailureReason failure_reason)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD3(LogLossBasedBweUpdate, void(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD3(LogLossBasedBweUpdate, void(int32_t bitrate_bps, uint8_t fraction_loss, int32_t total_packets)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogRtpHeader, void(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogRtcpPacket, void(PacketDirection direction, MediaType media_type, const uint8_t *packet, size_t length)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogProbeClusterCreated, void(int id, int bitrate_bps, int min_probes, int min_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogRtpHeader, void(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogRtcpPacket, void(PacketDirection direction, MediaType media_type, const uint8_t *packet, size_t length)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD4(LogProbeClusterCreated, void(int id, int bitrate_bps, int min_probes, int min_bytes)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD5(LogRtpHeader, void(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length, int probe_cluster_id)) | webrtc::MockRtcEventLog | |
| MOCK_METHOD5(LogRtpHeader, void(PacketDirection direction, MediaType media_type, const uint8_t *header, size_t packet_length, int probe_cluster_id)) | webrtc::MockRtcEventLog | |
| ParseRtcEventLog(const std::string &file_name, rtclog::EventStream *result) | webrtc::RtcEventLog | static |
| ParseRtcEventLog(const std::string &file_name, rtclog::EventStream *result) | webrtc::RtcEventLog | static |
| StartLogging(const std::string &file_name, int64_t max_size_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| StartLogging(rtc::PlatformFile platform_file, int64_t max_size_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| StartLogging(const std::string &file_name) | webrtc::RtcEventLog | inline |
| StartLogging(rtc::PlatformFile platform_file) | webrtc::RtcEventLog | inline |
| StartLogging(const std::string &file_name, int64_t max_size_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| StartLogging(rtc::PlatformFile platform_file, int64_t max_size_bytes)=0 | webrtc::RtcEventLog | pure virtual |
| StartLogging(const std::string &file_name) | webrtc::RtcEventLog | inline |
| StartLogging(rtc::PlatformFile platform_file) | webrtc::RtcEventLog | inline |
| StopLogging()=0 | webrtc::RtcEventLog | pure virtual |
| StopLogging()=0 | webrtc::RtcEventLog | pure virtual |
| ~RtcEventLog() | webrtc::RtcEventLog | inlinevirtual |
| ~RtcEventLog() | webrtc::RtcEventLog | inlinevirtual |
1.8.13