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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::RtpRtcp, 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 |
| 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 |
| 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