webrtc::RtpPacketToSend Member List

This is the complete list of members for webrtc::RtpPacketToSend, including all inherited members.

AllocatePayload(size_t size_bytes)	webrtc::rtp::Packet
AllocatePayload(size_t size_bytes)	webrtc::rtp::Packet
Buffer() const	webrtc::rtp::Packet
Buffer() const	webrtc::rtp::Packet
capacity() const	webrtc::rtp::Packet
capacity() const	webrtc::rtp::Packet
capture_time_ms() const	webrtc::RtpPacketToSend	inline
capture_time_ms() const	webrtc::RtpPacketToSend	inline
Clear()	webrtc::rtp::Packet
Clear()	webrtc::rtp::Packet
CopyHeaderFrom(const Packet &packet)	webrtc::rtp::Packet
CopyHeaderFrom(const Packet &packet)	webrtc::rtp::Packet
Csrcs() const	webrtc::rtp::Packet
Csrcs() const	webrtc::rtp::Packet
data() const	webrtc::rtp::Packet
data() const	webrtc::rtp::Packet
ExtensionManager typedef	webrtc::rtp::Packet
ExtensionManager typedef	webrtc::rtp::Packet
ExtensionType typedef	webrtc::rtp::Packet
ExtensionType typedef	webrtc::rtp::Packet
FreeCapacity() const	webrtc::rtp::Packet
FreeCapacity() const	webrtc::rtp::Packet
GetExtension(Values...) const	webrtc::rtp::Packet
GetExtension(Values...) const	webrtc::rtp::Packet
GetHeader(RTPHeader *header) const	webrtc::rtp::Packet
GetHeader(RTPHeader *header) const	webrtc::rtp::Packet
HasExtension() const	webrtc::rtp::Packet
HasExtension() const	webrtc::rtp::Packet
headers_size() const	webrtc::rtp::Packet
headers_size() const	webrtc::rtp::Packet
IdentifyExtensions(const ExtensionManager &extensions)	webrtc::rtp::Packet
IdentifyExtensions(const ExtensionManager &extensions)	webrtc::rtp::Packet
kMaxExtensionHeaders	webrtc::rtp::Packet	static
Marker() const	webrtc::rtp::Packet
Marker() const	webrtc::rtp::Packet
MaxPayloadSize() const	webrtc::rtp::Packet
MaxPayloadSize() const	webrtc::rtp::Packet
operator=(const RtpPacketToSend &packet)=default	webrtc::RtpPacketToSend
operator=(const RtpPacketToSend &packet)=default	webrtc::RtpPacketToSend
webrtc::rtp::Packet::operator=(const Packet &)=default	webrtc::rtp::Packet	protected
Packet()	webrtc::rtp::Packet	protected
Packet(const ExtensionManager *extensions)	webrtc::rtp::Packet	explicitprotected
Packet(const Packet &)=default	webrtc::rtp::Packet	protected
Packet(const ExtensionManager *extensions, size_t capacity)	webrtc::rtp::Packet	protected
Packet()	webrtc::rtp::Packet	protected
Packet(const ExtensionManager *extensions)	webrtc::rtp::Packet	explicitprotected
Packet(const Packet &)=default	webrtc::rtp::Packet	protected
Packet(const ExtensionManager *extensions, size_t capacity)	webrtc::rtp::Packet	protected
padding_size() const	webrtc::rtp::Packet
padding_size() const	webrtc::rtp::Packet
Parse(const uint8_t *buffer, size_t size)	webrtc::rtp::Packet
Parse(rtc::ArrayView< const uint8_t > packet)	webrtc::rtp::Packet
Parse(rtc::CopyOnWriteBuffer packet)	webrtc::rtp::Packet
Parse(const uint8_t *buffer, size_t size)	webrtc::rtp::Packet
Parse(rtc::ArrayView< const uint8_t > packet)	webrtc::rtp::Packet
Parse(rtc::CopyOnWriteBuffer packet)	webrtc::rtp::Packet
payload() const	webrtc::rtp::Packet
payload() const	webrtc::rtp::Packet
payload_size() const	webrtc::rtp::Packet
payload_size() const	webrtc::rtp::Packet
PayloadType() const	webrtc::rtp::Packet
PayloadType() const	webrtc::rtp::Packet
ReserveExtension()	webrtc::rtp::Packet
ReserveExtension()	webrtc::rtp::Packet
RtpPacketToSend(const ExtensionManager *extensions)	webrtc::RtpPacketToSend	inlineexplicit
RtpPacketToSend(const RtpPacketToSend &packet)=default	webrtc::RtpPacketToSend
RtpPacketToSend(const ExtensionManager *extensions, size_t capacity)	webrtc::RtpPacketToSend	inline
RtpPacketToSend(const ExtensionManager *extensions)	webrtc::RtpPacketToSend	inlineexplicit
RtpPacketToSend(const RtpPacketToSend &packet)=default	webrtc::RtpPacketToSend
RtpPacketToSend(const ExtensionManager *extensions, size_t capacity)	webrtc::RtpPacketToSend	inline
SequenceNumber() const	webrtc::rtp::Packet
SequenceNumber() const	webrtc::rtp::Packet
set_capture_time_ms(int64_t time)	webrtc::RtpPacketToSend	inline
set_capture_time_ms(int64_t time)	webrtc::RtpPacketToSend	inline
SetCsrcs(const std::vector< uint32_t > &csrcs)	webrtc::rtp::Packet
SetCsrcs(const std::vector< uint32_t > &csrcs)	webrtc::rtp::Packet
SetExtension(Values...)	webrtc::rtp::Packet
SetExtension(Values...)	webrtc::rtp::Packet
SetMarker(bool marker_bit)	webrtc::rtp::Packet
SetMarker(bool marker_bit)	webrtc::rtp::Packet
SetPadding(uint8_t size_bytes, Random *random)	webrtc::rtp::Packet
SetPadding(uint8_t size_bytes, Random *random)	webrtc::rtp::Packet
SetPayloadSize(size_t size_bytes)	webrtc::rtp::Packet
SetPayloadSize(size_t size_bytes)	webrtc::rtp::Packet
SetPayloadType(uint8_t payload_type)	webrtc::rtp::Packet
SetPayloadType(uint8_t payload_type)	webrtc::rtp::Packet
SetSequenceNumber(uint16_t seq_no)	webrtc::rtp::Packet
SetSequenceNumber(uint16_t seq_no)	webrtc::rtp::Packet
SetSsrc(uint32_t ssrc)	webrtc::rtp::Packet
SetSsrc(uint32_t ssrc)	webrtc::rtp::Packet
SetTimestamp(uint32_t timestamp)	webrtc::rtp::Packet
SetTimestamp(uint32_t timestamp)	webrtc::rtp::Packet
size() const	webrtc::rtp::Packet
size() const	webrtc::rtp::Packet
Ssrc() const	webrtc::rtp::Packet
Ssrc() const	webrtc::rtp::Packet
Timestamp() const	webrtc::rtp::Packet
Timestamp() const	webrtc::rtp::Packet
~Packet()	webrtc::rtp::Packet	protectedvirtual
~Packet()	webrtc::rtp::Packet	protectedvirtual