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webkit
2cdf99a9e3038c7e01b3c37e8ad903ecbe5eecf1
https://github.com/WebKit/webkit
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This is the complete list of members for webrtc::RtpPacketizer, including all inherited members.
| Create(RtpVideoCodecTypes type, size_t max_payload_len, const RTPVideoTypeHeader *rtp_type_header, FrameType frame_type) | webrtc::RtpPacketizer | static |
| Create(RtpVideoCodecTypes type, size_t max_payload_len, const RTPVideoTypeHeader *rtp_type_header, FrameType frame_type) | webrtc::RtpPacketizer | static |
| GetProtectionType()=0 | webrtc::RtpPacketizer | pure virtual |
| GetProtectionType()=0 | webrtc::RtpPacketizer | pure virtual |
| GetStorageType(uint32_t retransmission_settings)=0 | webrtc::RtpPacketizer | pure virtual |
| GetStorageType(uint32_t retransmission_settings)=0 | webrtc::RtpPacketizer | pure virtual |
| NextPacket(RtpPacketToSend *packet, bool *last_packet)=0 | webrtc::RtpPacketizer | pure virtual |
| NextPacket(RtpPacketToSend *packet, bool *last_packet)=0 | webrtc::RtpPacketizer | pure virtual |
| SetPayloadData(const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)=0 | webrtc::RtpPacketizer | pure virtual |
| SetPayloadData(const uint8_t *payload_data, size_t payload_size, const RTPFragmentationHeader *fragmentation)=0 | webrtc::RtpPacketizer | pure virtual |
| ToString()=0 | webrtc::RtpPacketizer | pure virtual |
| ToString()=0 | webrtc::RtpPacketizer | pure virtual |
| ~RtpPacketizer() | webrtc::RtpPacketizer | inlinevirtual |
| ~RtpPacketizer() | webrtc::RtpPacketizer | inlinevirtual |
1.8.13