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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::RTPReceiverStrategy, including all inherited members.
| CheckPayloadChanged(int8_t payload_type, PayloadUnion *specific_payload, bool *should_discard_changes) | webrtc::RTPReceiverStrategy | virtual |
| CheckPayloadChanged(int8_t payload_type, PayloadUnion *specific_payload, bool *should_discard_changes) | webrtc::RTPReceiverStrategy | virtual |
| CreateAudioStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | static |
| CreateAudioStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | static |
| CreateVideoStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | static |
| CreateVideoStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | static |
| crit_sect_ | webrtc::RTPReceiverStrategy | protected |
| data_callback_ | webrtc::RTPReceiverStrategy | protected |
| Energy(uint8_t array_of_energy[kRtpCsrcSize]) const | webrtc::RTPReceiverStrategy | virtual |
| Energy(uint8_t array_of_energy[kRtpCsrcSize]) const | webrtc::RTPReceiverStrategy | virtual |
| GetLastMediaSpecificPayload(PayloadUnion *payload) const | webrtc::RTPReceiverStrategy | |
| GetLastMediaSpecificPayload(PayloadUnion *payload) const | webrtc::RTPReceiverStrategy | |
| GetTelephoneEventHandler()=0 | webrtc::RTPReceiverStrategy | pure virtual |
| GetTelephoneEventHandler()=0 | webrtc::RTPReceiverStrategy | pure virtual |
| InvokeOnInitializeDecoder(RtpFeedback *callback, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion &specific_payload) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| InvokeOnInitializeDecoder(RtpFeedback *callback, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion &specific_payload) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| last_payload_ | webrtc::RTPReceiverStrategy | protected |
| OnNewPayloadTypeCreated(const CodecInst &audio_codec)=0 | webrtc::RTPReceiverStrategy | pure virtual |
| OnNewPayloadTypeCreated(const CodecInst &audio_codec)=0 | webrtc::RTPReceiverStrategy | pure virtual |
| ParseRtpPacket(WebRtcRTPHeader *rtp_header, const PayloadUnion &specific_payload, bool is_red, const uint8_t *payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet)=0 | webrtc::RTPReceiverStrategy | pure virtual |
| ParseRtpPacket(WebRtcRTPHeader *rtp_header, const PayloadUnion &specific_payload, bool is_red, const uint8_t *payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet)=0 | webrtc::RTPReceiverStrategy | pure virtual |
| ProcessDeadOrAlive(uint16_t last_payload_length) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| ProcessDeadOrAlive(uint16_t last_payload_length) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| RTPReceiverStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | explicitprotected |
| RTPReceiverStrategy(RtpData *data_callback) | webrtc::RTPReceiverStrategy | explicitprotected |
| SetLastMediaSpecificPayload(const PayloadUnion &payload) | webrtc::RTPReceiverStrategy | |
| SetLastMediaSpecificPayload(const PayloadUnion &payload) | webrtc::RTPReceiverStrategy | |
| ShouldReportCsrcChanges(uint8_t payload_type) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| ShouldReportCsrcChanges(uint8_t payload_type) const =0 | webrtc::RTPReceiverStrategy | pure virtual |
| ~RTPReceiverStrategy() | webrtc::RTPReceiverStrategy | inlinevirtual |
| ~RTPReceiverStrategy() | webrtc::RTPReceiverStrategy | inlinevirtual |
1.8.13