|
webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
|
This is the complete list of members for webrtc::MockRtpRtcp, including all inherited members.
| AddMixedCNAME(uint32_t ssrc, const char *cname)=0 | webrtc::RtpRtcp | pure virtual |
| AddMixedCNAME(uint32_t ssrc, const char *cname)=0 | webrtc::RtpRtcp | pure virtual |
| BitrateSent(uint32_t *total_rate, uint32_t *video_rate, uint32_t *fec_rate, uint32_t *nack_rate) const =0 | webrtc::RtpRtcp | pure virtual |
| BitrateSent(uint32_t *total_rate, uint32_t *video_rate, uint32_t *fec_rate, uint32_t *nack_rate) const =0 | webrtc::RtpRtcp | pure virtual |
| CreateRtpRtcp(const RtpRtcp::Configuration &configuration) | webrtc::RtpRtcp | static |
| CreateRtpRtcp(const RtpRtcp::Configuration &configuration) | webrtc::RtpRtcp | static |
| DataCountersRTP(size_t *bytes_sent, uint32_t *packets_sent) const =0 | webrtc::RtpRtcp | pure virtual |
| DataCountersRTP(size_t *bytes_sent, uint32_t *packets_sent) const =0 | webrtc::RtpRtcp | pure virtual |
| DeRegisterSendPayload(int8_t payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| DeRegisterSendPayload(int8_t payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| DeregisterSendRtpHeaderExtension(RTPExtensionType type)=0 | webrtc::RtpRtcp | pure virtual |
| DeregisterSendRtpHeaderExtension(RTPExtensionType type)=0 | webrtc::RtpRtcp | pure virtual |
| FlexfecSsrc() const =0 | webrtc::RtpRtcp | pure virtual |
| FlexfecSsrc() const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtcpStatisticsCallback()=0 | webrtc::RtpRtcp | pure virtual |
| GetRtcpStatisticsCallback()=0 | webrtc::RtpRtcp | pure virtual |
| GetRtpPacketLossStats(bool outgoing, uint32_t ssrc, struct RtpPacketLossStats *loss_stats) const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtpPacketLossStats(bool outgoing, uint32_t ssrc, struct RtpPacketLossStats *loss_stats) const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtpState() const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtpState() const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtxState() const =0 | webrtc::RtpRtcp | pure virtual |
| GetRtxState() const =0 | webrtc::RtpRtcp | pure virtual |
| GetSendChannelRtpStatisticsCallback() const =0 | webrtc::RtpRtcp | pure virtual |
| GetSendChannelRtpStatisticsCallback() const =0 | webrtc::RtpRtcp | pure virtual |
| GetSendStreamDataCounters(StreamDataCounters *rtp_counters, StreamDataCounters *rtx_counters) const =0 | webrtc::RtpRtcp | pure virtual |
| GetSendStreamDataCounters(StreamDataCounters *rtp_counters, StreamDataCounters *rtx_counters) const =0 | webrtc::RtpRtcp | pure virtual |
| HasBweExtensions() const =0 | webrtc::RtpRtcp | pure virtual |
| HasBweExtensions() const =0 | webrtc::RtpRtcp | pure virtual |
| IncomingRtcpPacket(const uint8_t *incoming_packet, size_t incoming_packet_length)=0 | webrtc::RtpRtcp | pure virtual |
| IncomingRtcpPacket(const uint8_t *incoming_packet, size_t incoming_packet_length)=0 | webrtc::RtpRtcp | pure virtual |
| MaxPayloadSize() const =0 | webrtc::RtpRtcp | pure virtual |
| MaxPayloadSize() const =0 | webrtc::RtpRtcp | pure virtual |
| MaxRtpPacketSize() const =0 | webrtc::RtpRtcp | pure virtual |
| MaxRtpPacketSize() const =0 | webrtc::RtpRtcp | pure virtual |
| MOCK_CONST_METHOD0(MaxPayloadSize, size_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(HasBweExtensions, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(StartTimestamp, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SequenceNumber, uint16_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetRtpState, RtpState()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetRtxState, RtpState()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SSRC, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtxSendStatus, int()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(FlexfecSsrc, rtc::Optional< uint32_t >()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair< int, int >()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(Sending, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SendingMedia, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RTCP, RtcpMode()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtcpXrRrtrStatus, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(REMB, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(TMMBR, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SelectiveRetransmissions, int()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(StorePackets, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetSendFrameCountObserver, FrameCountObserver *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, StreamDataCountersCallback *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(MaxPayloadSize, size_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(HasBweExtensions, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(StartTimestamp, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SequenceNumber, uint16_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetRtpState, RtpState()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetRtxState, RtpState()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SSRC, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtxSendStatus, int()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(FlexfecSsrc, rtc::Optional< uint32_t >()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair< int, int >()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(Sending, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SendingMedia, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RTCP, RtcpMode()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(RtcpXrRrtrStatus, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(REMB, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(TMMBR, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(SelectiveRetransmissions, int()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(StorePackets, bool()) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetSendFrameCountObserver, FrameCountObserver *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, StreamDataCountersCallback *(void)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t csrcs[kRtpCsrcSize])) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(EstimatedReceiveBandwidth, int(uint32_t *available_bandwidth)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(RemoteRTCPStat, int32_t(std::vector< RTCPReportBlock > *receive_blocks)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t *payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t csrcs[kRtpCsrcSize])) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(EstimatedReceiveBandwidth, int(uint32_t *available_bandwidth)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(RemoteRTCPStat, int32_t(std::vector< RTCPReportBlock > *receive_blocks)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t *payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(RemoteCNAME, int32_t(uint32_t remote_ssrc, char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(DataCountersRTP, int32_t(size_t *bytes_sent, uint32_t *packets_sent)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(GetSendStreamDataCounters, void(StreamDataCounters *, StreamDataCounters *)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(RemoteCNAME, int32_t(uint32_t remote_ssrc, char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(DataCountersRTP, int32_t(size_t *bytes_sent, uint32_t *packets_sent)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD2(GetSendStreamDataCounters, void(StreamDataCounters *, StreamDataCounters *)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD3(GetRtpPacketLossStats, void(bool, uint32_t, struct RtpPacketLossStats *)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD3(GetRtpPacketLossStats, void(bool, uint32_t, struct RtpPacketLossStats *)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD4(BitrateSent, void(uint32_t *total_rate, uint32_t *video_rate, uint32_t *fec_rate, uint32_t *nack_rate)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD4(BitrateSent, void(uint32_t *total_rate, uint32_t *video_rate, uint32_t *fec_rate, uint32_t *nack_rate)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD5(RemoteNTP, int32_t(uint32_t *received_ntp_secs, uint32_t *received_ntp_frac, uint32_t *rtcp_arrival_time_secs, uint32_t *rtcp_arrival_time_frac, uint32_t *rtcp_timestamp)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD5(RTT, int32_t(uint32_t remote_ssrc, int64_t *rtt, int64_t *avg_rtt, int64_t *min_rtt, int64_t *max_rtt)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD5(RemoteNTP, int32_t(uint32_t *received_ntp_secs, uint32_t *received_ntp_frac, uint32_t *rtcp_arrival_time_secs, uint32_t *rtcp_arrival_time_frac, uint32_t *rtcp_timestamp)) | webrtc::MockRtpRtcp | |
| MOCK_CONST_METHOD5(RTT, int32_t(uint32_t remote_ssrc, int64_t *rtt, int64_t *avg_rtt, int64_t *min_rtt, int64_t *max_rtt)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DeRegisterDefaultModule, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DefaultModuleRegistered, bool()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(NumberChildModules, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DeRegisterSyncModule, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(InitSender, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback *()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(RequestKeyFrame, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(TimeUntilNextProcess, int64_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(Process, void()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DeRegisterDefaultModule, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DefaultModuleRegistered, bool()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(NumberChildModules, uint32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(DeRegisterSyncModule, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(InitSender, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback *()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(RequestKeyFrame, int32_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(TimeUntilNextProcess, int64_t()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD0(Process, void()) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterDefaultModule, int32_t(RtpRtcp *module)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSyncModule, int32_t(RtpRtcp *module)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendTransport, int32_t(Transport *outgoing_transport)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst &voice_codec)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec &video_codec)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(DeregisterSendRtpHeaderExtension, int32_t(RTPExtensionType type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetStartTimestamp, void(uint32_t timestamp)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSequenceNumber, void(uint16_t seq)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtpState, void(const RtpState &rtp_state)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxState, void(const RtpState &rtp_state)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSSRC, void(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCsrcs, void(const std::vector< uint32_t > &csrcs)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCSRCStatus, int32_t(bool include)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxSendStatus, void(int modes)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxSsrc, void(uint32_t)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendingStatus, int32_t(bool sending)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendingMediaStatus, void(bool sending)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RemoveMixedCNAME, int32_t(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packet_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendCompoundRTCP, int32_t(const std::set< RTCPPacketType > &packet_types)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCPReferencePictureSelection, int32_t(uint64_t picture_id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCPSliceLossIndication, int32_t(uint8_t picture_id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RemoteRTCPStat, int32_t(RTCPSenderInfo *sender_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRTCPVoIPMetrics, int32_t(const RTCPVoIPMetric *voip_metric)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtcpXrRrtrStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetREMBStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTMMBRStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(OnBandwidthEstimateUpdate, void(uint16_t bandwidth_kbit)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSelectiveRetransmissions, int(uint8_t settings)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendNack, void(const std::vector< uint16_t > &sequence_numbers)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback &packet)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetAudioPacketSize, int32_t(uint16_t packet_size_samples)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendREDPayloadType, int32_t(int8_t payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendFrameCountObserver, void(FrameCountObserver *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendChannelRtpStatisticsCallback, void(StreamDataCountersCallback *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation &)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterDefaultModule, int32_t(RtpRtcp *module)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSyncModule, int32_t(RtpRtcp *module)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendTransport, int32_t(Transport *outgoing_transport)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst &voice_codec)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec &video_codec)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(DeregisterSendRtpHeaderExtension, int32_t(RTPExtensionType type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetStartTimestamp, void(uint32_t timestamp)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSequenceNumber, void(uint16_t seq)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtpState, void(const RtpState &rtp_state)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxState, void(const RtpState &rtp_state)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSSRC, void(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCsrcs, void(const std::vector< uint32_t > &csrcs)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCSRCStatus, int32_t(bool include)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxSendStatus, void(int modes)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtxSsrc, void(uint32_t)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendingStatus, int32_t(bool sending)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendingMediaStatus, void(bool sending)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RemoveMixedCNAME, int32_t(uint32_t ssrc)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCP, int32_t(RTCPPacketType packet_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendCompoundRTCP, int32_t(const std::set< RTCPPacketType > &packet_types)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCPReferencePictureSelection, int32_t(uint64_t picture_id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendRTCPSliceLossIndication, int32_t(uint8_t picture_id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RemoteRTCPStat, int32_t(RTCPSenderInfo *sender_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRTCPVoIPMetrics, int32_t(const RTCPVoIPMetric *voip_metric)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetRtcpXrRrtrStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetREMBStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTMMBRStatus, void(bool enable)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(OnBandwidthEstimateUpdate, void(uint16_t bandwidth_kbit)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSelectiveRetransmissions, int(uint8_t settings)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendNack, void(const std::vector< uint16_t > &sequence_numbers)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback &packet)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetAudioPacketSize, int32_t(uint16_t packet_size_samples)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetSendREDPayloadType, int32_t(int8_t payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendFrameCountObserver, void(FrameCountObserver *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(RegisterSendChannelRtpStatisticsCallback, void(StreamDataCountersCallback *)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation &)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(IncomingRtcpPacket, int32_t(const uint8_t *incoming_packet, size_t packet_length)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterVideoSendPayload, void(int payload_type, const char *payload_name)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterSendRtpHeaderExtension, int32_t(RTPExtensionType type, uint8_t id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetRtxSendPayloadType, void(int, int)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(TimeToSendPadding, size_t(size_t bytes, const PacedPacketInfo &pacing_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterRtcpObservers, void(RtcpIntraFrameObserver *intra_frame_callback, RtcpBandwidthObserver *bandwidth_callback)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(AddMixedCNAME, int32_t(uint32_t ssrc, const char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetREMBData, void(uint32_t bitrate, const std::vector< uint32_t > &ssrcs)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SendNACK, int32_t(const uint16_t *nack_list, uint16_t size)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetStorePacketsStatus, void(bool enable, uint16_t number_to_store)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetRTPAudioLevelIndicationStatus, int32_t(bool enable, uint8_t id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetUlpfecConfig, void(int red_payload_type, int fec_payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetFecParameters, bool(const FecProtectionParams &delta_params, const FecProtectionParams &key_params)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(IncomingRtcpPacket, int32_t(const uint8_t *incoming_packet, size_t packet_length)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterVideoSendPayload, void(int payload_type, const char *payload_name)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterSendRtpHeaderExtension, int32_t(RTPExtensionType type, uint8_t id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetRtxSendPayloadType, void(int, int)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(TimeToSendPadding, size_t(size_t bytes, const PacedPacketInfo &pacing_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(RegisterRtcpObservers, void(RtcpIntraFrameObserver *intra_frame_callback, RtcpBandwidthObserver *bandwidth_callback)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(AddMixedCNAME, int32_t(uint32_t ssrc, const char cname[RTCP_CNAME_SIZE])) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetREMBData, void(uint32_t bitrate, const std::vector< uint32_t > &ssrcs)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SendNACK, int32_t(const uint16_t *nack_list, uint16_t size)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetStorePacketsStatus, void(bool enable, uint16_t number_to_store)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetRTPAudioLevelIndicationStatus, int32_t(bool enable, uint8_t id)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetUlpfecConfig, void(int red_payload_type, int fec_payload_type)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD2(SetFecParameters, bool(const FecProtectionParams &delta_params, const FecProtectionParams &key_params)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD3(SendTelephoneEventOutband, int32_t(uint8_t key, uint16_t time_ms, uint8_t level)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD3(SendTelephoneEventOutband, int32_t(uint8_t key, uint16_t time_ms, uint8_t level)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD4(IncomingAudioNTP, int32_t(uint32_t audio_received_ntp_secs, uint32_t audio_received_ntp_frac, uint32_t audio_rtcp_arrival_time_secs, uint32_t audio_rtcp_arrival_time_frac)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD4(SetRTCPApplicationSpecificData, int32_t(uint8_t sub_type, uint32_t name, const uint8_t *data, uint16_t length)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD4(IncomingAudioNTP, int32_t(uint32_t audio_received_ntp_secs, uint32_t audio_received_ntp_frac, uint32_t audio_rtcp_arrival_time_secs, uint32_t audio_rtcp_arrival_time_frac)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD4(SetRTCPApplicationSpecificData, int32_t(uint8_t sub_type, uint32_t name, const uint8_t *data, uint16_t length)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD5(TimeToSendPacket, bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD5(TimeToSendPacket, bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD9(SendOutgoingData, bool(FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_video_header, uint32_t *frame_id_out)) | webrtc::MockRtpRtcp | |
| MOCK_METHOD9(SendOutgoingData, bool(FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_video_header, uint32_t *frame_id_out)) | webrtc::MockRtpRtcp | |
| Process()=0 | webrtc::Module | pure virtual |
| Process()=0 | webrtc::Module | pure virtual |
| ProcessThreadAttached(ProcessThread *) | webrtc::Module | inlinevirtual |
| ProcessThreadAttached(ProcessThread *) | webrtc::Module | inlinevirtual |
| RegisterRtcpStatisticsCallback(RtcpStatisticsCallback *callback)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterRtcpStatisticsCallback(RtcpStatisticsCallback *callback)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendChannelRtpStatisticsCallback(StreamDataCountersCallback *callback)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendChannelRtpStatisticsCallback(StreamDataCountersCallback *callback)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendPayload(const CodecInst &voice_codec)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendPayload(const VideoCodec &video_codec)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendPayload(const CodecInst &voice_codec)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendPayload(const VideoCodec &video_codec)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendRtpHeaderExtension(RTPExtensionType type, uint8_t id)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterSendRtpHeaderExtension(RTPExtensionType type, uint8_t id)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterVideoSendPayload(int payload_type, const char *payload_name)=0 | webrtc::RtpRtcp | pure virtual |
| RegisterVideoSendPayload(int payload_type, const char *payload_name)=0 | webrtc::RtpRtcp | pure virtual |
| REMB() const =0 | webrtc::RtpRtcp | pure virtual |
| REMB() const =0 | webrtc::RtpRtcp | pure virtual |
| remote_ssrc_ | webrtc::MockRtpRtcp | |
| RemoteCNAME(uint32_t remote_ssrc, char cname[RTCP_CNAME_SIZE]) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoteCNAME(uint32_t remote_ssrc, char cname[RTCP_CNAME_SIZE]) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoteNTP(uint32_t *received_ntp_secs, uint32_t *received_ntp_frac, uint32_t *rtcp_arrival_time_secs, uint32_t *rtcp_arrival_time_frac, uint32_t *rtcp_timestamp) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoteNTP(uint32_t *received_ntp_secs, uint32_t *received_ntp_frac, uint32_t *rtcp_arrival_time_secs, uint32_t *rtcp_arrival_time_frac, uint32_t *rtcp_timestamp) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoteRTCPStat(RTCPSenderInfo *sender_info)=0 | webrtc::RtpRtcp | pure virtual |
| RemoteRTCPStat(std::vector< RTCPReportBlock > *receive_blocks) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoteRTCPStat(RTCPSenderInfo *sender_info)=0 | webrtc::RtpRtcp | pure virtual |
| RemoteRTCPStat(std::vector< RTCPReportBlock > *receive_blocks) const =0 | webrtc::RtpRtcp | pure virtual |
| RemoveMixedCNAME(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| RemoveMixedCNAME(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| RequestKeyFrame()=0 | webrtc::RtpRtcp | pure virtual |
| RequestKeyFrame()=0 | webrtc::RtpRtcp | pure virtual |
| RTCP() const =0 | webrtc::RtpRtcp | pure virtual |
| RTCP() const =0 | webrtc::RtpRtcp | pure virtual |
| RtcpXrRrtrStatus() const =0 | webrtc::RtpRtcp | pure virtual |
| RtcpXrRrtrStatus() const =0 | webrtc::RtpRtcp | pure virtual |
| RTT(uint32_t remote_ssrc, int64_t *rtt, int64_t *avg_rtt, int64_t *min_rtt, int64_t *max_rtt) const =0 | webrtc::RtpRtcp | pure virtual |
| RTT(uint32_t remote_ssrc, int64_t *rtt, int64_t *avg_rtt, int64_t *min_rtt, int64_t *max_rtt) const =0 | webrtc::RtpRtcp | pure virtual |
| RtxSendStatus() const =0 | webrtc::RtpRtcp | pure virtual |
| RtxSendStatus() const =0 | webrtc::RtpRtcp | pure virtual |
| SelectiveRetransmissions() const =0 | webrtc::RtpRtcp | pure virtual |
| SelectiveRetransmissions() const =0 | webrtc::RtpRtcp | pure virtual |
| SendCompoundRTCP(const std::set< RTCPPacketType > &rtcp_packet_types)=0 | webrtc::RtpRtcp | pure virtual |
| SendCompoundRTCP(const std::set< RTCPPacketType > &rtcp_packet_types)=0 | webrtc::RtpRtcp | pure virtual |
| SendFeedbackPacket(const rtcp::TransportFeedback &packet)=0 | webrtc::RtpRtcp | pure virtual |
| SendFeedbackPacket(const rtcp::TransportFeedback &packet)=0 | webrtc::RtpRtcp | pure virtual |
| Sending() const =0 | webrtc::RtpRtcp | pure virtual |
| Sending() const =0 | webrtc::RtpRtcp | pure virtual |
| SendingMedia() const =0 | webrtc::RtpRtcp | pure virtual |
| SendingMedia() const =0 | webrtc::RtpRtcp | pure virtual |
| SendNACK(const uint16_t *nack_list, uint16_t size)=0 | webrtc::RtpRtcp | pure virtual |
| SendNACK(const uint16_t *nack_list, uint16_t size)=0 | webrtc::RtpRtcp | pure virtual |
| SendNack(const std::vector< uint16_t > &sequence_numbers)=0 | webrtc::RtpRtcp | pure virtual |
| SendNack(const std::vector< uint16_t > &sequence_numbers)=0 | webrtc::RtpRtcp | pure virtual |
| SendOutgoingData(FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_video_header, uint32_t *transport_frame_id_out)=0 | webrtc::RtpRtcp | pure virtual |
| SendOutgoingData(FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation, const RTPVideoHeader *rtp_video_header, uint32_t *transport_frame_id_out)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCP(RTCPPacketType rtcp_packet_type)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCP(RTCPPacketType rtcp_packet_type)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCPReferencePictureSelection(uint64_t picture_id)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCPReferencePictureSelection(uint64_t picture_id)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCPSliceLossIndication(uint8_t picture_id)=0 | webrtc::RtpRtcp | pure virtual |
| SendRTCPSliceLossIndication(uint8_t picture_id)=0 | webrtc::RtpRtcp | pure virtual |
| SendTelephoneEventOutband(uint8_t key, uint16_t time_ms, uint8_t level)=0 | webrtc::RtpRtcp | pure virtual |
| SendTelephoneEventOutband(uint8_t key, uint16_t time_ms, uint8_t level)=0 | webrtc::RtpRtcp | pure virtual |
| SequenceNumber() const =0 | webrtc::RtpRtcp | pure virtual |
| SequenceNumber() const =0 | webrtc::RtpRtcp | pure virtual |
| SetAudioLevel(uint8_t level_dbov)=0 | webrtc::RtpRtcp | pure virtual |
| SetAudioLevel(uint8_t level_dbov)=0 | webrtc::RtpRtcp | pure virtual |
| SetAudioPacketSize(uint16_t packet_size_samples)=0 | webrtc::RtpRtcp | pure virtual |
| SetAudioPacketSize(uint16_t packet_size_samples)=0 | webrtc::RtpRtcp | pure virtual |
| SetCNAME(const char *cname)=0 | webrtc::RtpRtcp | pure virtual |
| SetCNAME(const char *cname)=0 | webrtc::RtpRtcp | pure virtual |
| SetCsrcs(const std::vector< uint32_t > &csrcs)=0 | webrtc::RtpRtcp | pure virtual |
| SetCsrcs(const std::vector< uint32_t > &csrcs)=0 | webrtc::RtpRtcp | pure virtual |
| SetFecParameters(const FecProtectionParams &delta_params, const FecProtectionParams &key_params)=0 | webrtc::RtpRtcp | pure virtual |
| SetFecParameters(const FecProtectionParams *delta_params, const FecProtectionParams *key_params) | webrtc::RtpRtcp | |
| SetFecParameters(const FecProtectionParams &delta_params, const FecProtectionParams &key_params)=0 | webrtc::RtpRtcp | pure virtual |
| SetFecParameters(const FecProtectionParams *delta_params, const FecProtectionParams *key_params) | webrtc::RtpRtcp | |
| SetKeyFrameRequestMethod(KeyFrameRequestMethod method)=0 | webrtc::RtpRtcp | pure virtual |
| SetKeyFrameRequestMethod(KeyFrameRequestMethod method)=0 | webrtc::RtpRtcp | pure virtual |
| SetMaxRtpPacketSize(size_t size)=0 | webrtc::RtpRtcp | pure virtual |
| SetMaxRtpPacketSize(size_t size)=0 | webrtc::RtpRtcp | pure virtual |
| SetMaxTransferUnit(uint16_t size) | webrtc::RtpRtcp | inlinevirtual |
| SetMaxTransferUnit(uint16_t size) | webrtc::RtpRtcp | inlinevirtual |
| SetREMBData(uint32_t bitrate, const std::vector< uint32_t > &ssrcs)=0 | webrtc::RtpRtcp | pure virtual |
| SetREMBData(uint32_t bitrate, const std::vector< uint32_t > &ssrcs)=0 | webrtc::RtpRtcp | pure virtual |
| SetREMBStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetREMBStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetRemoteSSRC(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetRemoteSSRC(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPApplicationSpecificData(uint8_t sub_type, uint32_t name, const uint8_t *data, uint16_t length)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPApplicationSpecificData(uint8_t sub_type, uint32_t name, const uint8_t *data, uint16_t length)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPStatus(RtcpMode method)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPStatus(RtcpMode method)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPVoIPMetrics(const RTCPVoIPMetric *VoIPMetric)=0 | webrtc::RtpRtcp | pure virtual |
| SetRTCPVoIPMetrics(const RTCPVoIPMetric *VoIPMetric)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtcpXrRrtrStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtcpXrRrtrStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtpState(const RtpState &rtp_state)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtpState(const RtpState &rtp_state)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSendPayloadType(int payload_type, int associated_payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSendPayloadType(int payload_type, int associated_payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSendStatus(int modes)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSendStatus(int modes)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSsrc(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxSsrc(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxState(const RtpState &rtp_state)=0 | webrtc::RtpRtcp | pure virtual |
| SetRtxState(const RtpState &rtp_state)=0 | webrtc::RtpRtcp | pure virtual |
| SetSelectiveRetransmissions(uint8_t settings)=0 | webrtc::RtpRtcp | pure virtual |
| SetSelectiveRetransmissions(uint8_t settings)=0 | webrtc::RtpRtcp | pure virtual |
| SetSendingMediaStatus(bool sending)=0 | webrtc::RtpRtcp | pure virtual |
| SetSendingMediaStatus(bool sending)=0 | webrtc::RtpRtcp | pure virtual |
| SetSendingStatus(bool sending)=0 | webrtc::RtpRtcp | pure virtual |
| SetSendingStatus(bool sending)=0 | webrtc::RtpRtcp | pure virtual |
| SetSequenceNumber(uint16_t seq)=0 | webrtc::RtpRtcp | pure virtual |
| SetSequenceNumber(uint16_t seq)=0 | webrtc::RtpRtcp | pure virtual |
| SetSSRC(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetSSRC(uint32_t ssrc)=0 | webrtc::RtpRtcp | pure virtual |
| SetStartTimestamp(uint32_t timestamp)=0 | webrtc::RtpRtcp | pure virtual |
| SetStartTimestamp(uint32_t timestamp)=0 | webrtc::RtpRtcp | pure virtual |
| SetStorePacketsStatus(bool enable, uint16_t numberToStore)=0 | webrtc::RtpRtcp | pure virtual |
| SetStorePacketsStatus(bool enable, uint16_t numberToStore)=0 | webrtc::RtpRtcp | pure virtual |
| SetTMMBRStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetTMMBRStatus(bool enable)=0 | webrtc::RtpRtcp | pure virtual |
| SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type)=0 | webrtc::RtpRtcp | pure virtual |
| SetVideoBitrateAllocation(const BitrateAllocation &bitrate)=0 | webrtc::RtpRtcp | pure virtual |
| SetVideoBitrateAllocation(const BitrateAllocation &bitrate)=0 | webrtc::RtpRtcp | pure virtual |
| SSRC() const =0 | webrtc::RtpRtcp | pure virtual |
| SSRC() const =0 | webrtc::RtpRtcp | pure virtual |
| StartTimestamp() const =0 | webrtc::RtpRtcp | pure virtual |
| StartTimestamp() const =0 | webrtc::RtpRtcp | pure virtual |
| StorePackets() const =0 | webrtc::RtpRtcp | pure virtual |
| StorePackets() const =0 | webrtc::RtpRtcp | pure virtual |
| TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)=0 | webrtc::RtpRtcp | pure virtual |
| TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo &pacing_info)=0 | webrtc::RtpRtcp | pure virtual |
| TimeToSendPadding(size_t bytes, const PacedPacketInfo &pacing_info)=0 | webrtc::RtpRtcp | pure virtual |
| TimeToSendPadding(size_t bytes, const PacedPacketInfo &pacing_info)=0 | webrtc::RtpRtcp | pure virtual |
| TimeUntilNextProcess()=0 | webrtc::Module | pure virtual |
| TimeUntilNextProcess()=0 | webrtc::Module | pure virtual |
| TMMBR() const =0 | webrtc::RtpRtcp | pure virtual |
| TMMBR() const =0 | webrtc::RtpRtcp | pure virtual |
| ~Module() | webrtc::Module | inlineprotectedvirtual |
| ~Module() | webrtc::Module | inlineprotectedvirtual |
1.8.13