C types for gstrtp1 library
Different types of feedback messages.
Different types of SDES content.
Different RTCP packet types.
Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs according to the IANA registry.
Additional RTP buffer flags. These flags can potentially be used on any buffers carrying RTP packets.
Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp). They can conflict with other extended buffer flags.
Additional mapping flags for [gstrtp.rtpbuffer.RTPBuffer.map].
Direction to which to apply the RTP Header Extension
Flags that apply to a RTP Audio/Video header extension.
Standard predefined fixed payload types.
The official list is at:
http://www.iana.org/assignments/rtp-parametersAudio: reserved: 19 unassigned: 20-23,
Video: unassigned: 24, 27, 29, 30, 35-71, 77-95 Reserved for RTCP conflict avoidance: 72-76
The transfer profile to use.
The GstRTPCBuffer helper functions makes it easy to parse and create regular #GstBuffer objects that contain compound RTCP packets. These buffers are typically of 'application/x-rtcp' #GstCaps.
An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can retrieve with [gstrtp.rtcpbuffer.RTCPBuffer.getFirstPacket]. #GstRTCPPacket acts as a pointer into the RTCP buffer; you can move to the next packet with [gstrtp.rtcppacket.RTCPPacket.moveToNext].
Data structure that points to a packet at @offset in @buffer. The size of the structure is made public to allow stack allocations.
GstRTCPBuffer * rtcppointer to RTCP bufferuint offsetoffset of packet in buffer datagboolean paddingubyte countGstRTCPType typeushort lengthuint itemOffsetuint itemCountuint entryOffsetProvides a base class for audio RTP payloaders for frame or sample based audio codecs (constant bitrate)
This class derives from GstRTPBasePayload. It can be used for payloading audio codecs. It will only work with constant bitrate codecs. It supports both frame based and sample based codecs. It takes care of packing up the audio data into RTP packets and filling up the headers accordingly. The payloading is done based on the maximum MTU (mtu) and the maximum time per packet (max-ptime). The general idea is to divide large data buffers into smaller RTP packets. The RTP packet size is the minimum of either the MTU, max-ptime (if set) or available data. The RTP packet size is always larger or equal to min-ptime (if set). If min-ptime is not set, any residual data is sent in a last RTP packet. In the case of frame based codecs, the resulting RTP packets always contain full frames.
To use this base class, your child element needs to call either [gstrtp.rtpbase_audio_payload.RTPBaseAudioPayload.setFrameBased] or [gstrtp.rtpbase_audio_payload.RTPBaseAudioPayload.setSampleBased]. This is usually done in the element's _init() function. Then, the child element must call either [gstrtp.rtpbase_audio_payload.RTPBaseAudioPayload.setFrameOptions], [gstrtp.rtpbase_audio_payload.RTPBaseAudioPayload.setSampleOptions] or gst_rtp_base_audio_payload_set_samplebits_options. Since GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element must set any variables or call/override any functions required by that base class. The child element does not need to override any other functions specific to GstRTPBaseAudioPayload.
GstRTPBasePayload payloadGstClockTime baseTsint frameSizeint frameDurationint sampleSizevoid *[4] GstReservedBase class for audio RTP payloader.
Provides a base class for RTP depayloaders
In order to handle RTP header extensions correctly if the depayloader aggregates multiple RTP packet payloads into one output buffer this class provides the function [gstrtp.rtpbase_depayload.RTPBaseDepayload.setAggregateHdrextEnabled]. If the aggregation is enabled the virtual functions @GstRTPBaseDepayload.process or @GstRTPBaseDepayload.process_rtp_packet must tell the base class what happens to the current RTP packet. By default the base class assumes that the packet payload is used with the next output buffer.
If the RTP packet will not be used with an output buffer [gstrtp.rtpbase_depayload.RTPBaseDepayload.dropped] must be called. A typical situation would be if we are waiting for a keyframe.
If the RTP packet will be used but not with the current output buffer but with the next one [gstrtp.rtpbase_depayload.RTPBaseDepayload.delayed] must be called. This may happen if the current RTP packet signals the start of a new output buffer and the currently processed output buffer will be pushed first. The undelay happens implicitly once the current buffer has been pushed or [gstrtp.rtpbase_depayload.RTPBaseDepayload.flush] has been called.
If [gstrtp.rtpbase_depayload.RTPBaseDepayload.flush] is called all RTP packets that have not been dropped since the last output buffer are dropped, e.g. if an output buffer is discarded due to malformed data. This may or may not include the current RTP packet depending on the 2nd parameter @keep_current.
Be aware that in case [gstrtp.rtpbase_depayload.RTPBaseDepayload.pushList] is used each buffer will see the same list of RTP header extensions.
GstElement parentGstPad * sinkpadGstPad * srcpaduint clockRateGstSegment segmentgboolean needNewsegmentGstRTPBaseDepayloadPrivate * privvoid *[4] GstReservedBase class for RTP depayloaders.
GstElementClass parentClassthe parent classgboolean function(GstRTPBaseDepayload * filter, GstCaps * caps) setCapsconfigure the depayloaderGstBuffer * function(GstRTPBaseDepayload * base, GstBuffer * in_) processprocess incoming rtp packets. Subclass must implement either this method or @processrtppacket to process incoming rtp packets. If the child returns a buffer without a valid timestamp, the timestamp...gboolean function(GstRTPBaseDepayload * filter, GstEvent * event) packetLostsignal the depayloader about packet lossgboolean function(GstRTPBaseDepayload * filter, GstEvent * event) handleEventcustom event handlingGstBuffer * function(GstRTPBaseDepayload * base, GstRTPBuffer * rtpBuffer) processRtpPacketSame as the process virtual function, but slightly more efficient, since it is passed the rtp buffer structure that has already been mapped (with GSTMAPREAD) by the base class and thus does not hav...void *[3] GstReservedProvides a base class for RTP payloaders
GstElement elementGstPad * sinkpadGstPad * srcpaduint tsBaseushort seqnumBasechar * mediachar * encodingNamegboolean dynamicuint clockRateint tsOffsetuint timestampshort seqnumOffsetushort seqnumlong maxPtimeuint ptuint ssrcuint currentSsrcuint mtuGstSegment segmentulong minPtimeulong ptimeulong ptimeMultipleGstRTPBasePayloadPrivate * privvoid *[4] GstReservedBase class for audio RTP payloader.
GstElementClass parentClassthe parent classGstCaps * function(GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter) getCapsget desired capsgboolean function(GstRTPBasePayload * payload, GstCaps * caps) setCapsconfigure the payloaderGstFlowReturn function(GstRTPBasePayload * payload, GstBuffer * buffer) handleBufferprocess datagboolean function(GstRTPBasePayload * payload, GstEvent * event) sinkEventcustom event handling on the sinkpadgboolean function(GstRTPBasePayload * payload, GstEvent * event) srcEventcustom event handling on the srcpadgboolean function(GstRTPBasePayload * payload, GstPad * pad, GstQuery * query) querycustom query handlingvoid *[4] GstReservedThe GstRTPBuffer helper functions makes it easy to parse and create regular #GstBuffer objects that contain RTP payloads. These buffers are typically of 'application/x-rtp' #GstCaps.
GstBuffer * bufferpointer to RTP bufferuint stateinternal statevoid *[4] dataarray of datasize_t[4] sizearray of sizeGstMapInfo[4] maparray of #GstMapInfoInstance struct for a RTP Audio/Video header extension.
Base class for RTP Header extensions.
GstElementClass parentClassthe parent classGstRTPHeaderExtensionFlags function(GstRTPHeaderExtension * ext) getSupportedFlagsretrieve the supported flagssize_t function(GstRTPHeaderExtension * ext, const(GstBuffer) * inputMeta) getMaxSizeretrieve the maximum size for this extension based on the information available from input_meta. Implementations should attempt to provide as accurate information as possible as the returned value...ptrdiff_t function(GstRTPHeaderExtension * ext, const(GstBuffer) * inputMeta, GstRTPHeaderExtensionFlags writeFlags, GstBuffer * output, ubyte * data, size_t size) writewrite into @data the information for this extension. Various information is provided to help writing extensions in particular cases.gboolean function(GstRTPHeaderExtension * ext, GstRTPHeaderExtensionFlags readFlags, const(ubyte) * data, size_t size, GstBuffer * buffer) readread from a rtp payloaded buffer and extract the extension information, optionally adding some meta onto the output buffer.gboolean function(GstRTPHeaderExtension * ext, const(GstCaps) * caps) setNonRtpSinkCapsread any information from sink caps that the header extension needs for its function.gboolean function(GstRTPHeaderExtension * ext, GstCaps * caps) updateNonRtpSrcCapsupdate depayloader non-RTP (depayloaded) caps with the information parsed from RTP header.gboolean function(GstRTPHeaderExtension * ext, GstRTPHeaderExtensionDirection direction, const(char) * attributes) setAttributesset the necessary attributes that may be signaled e.g. with an SDP.gboolean function(GstRTPHeaderExtension * ext, GstCaps * caps) setCapsFromAttributeswrite the necessary caps field/s for the configured attributes e.g. as signalled with SDP.void *[20] GstReservedStructure holding default payload type information.
ubyte payloadTypepayload type, -1 means dynamicconst(char) * mediathe media type(s), usually "audio", "video", "application", "text", "message".const(char) * encodingNamethe encoding name of @ptuint clockRatedefault clock rate, 0 = unknown/variableconst(char) * encodingParametersencoding parameters. For audio this is the number of channels. NULL = not applicable.uint bitratethe bitrate of the media. 0 = unknown/variable.void *[4] GstReservedMeta describing the source(s) of the buffer.
GstMeta metaparent #GstMetauint ssrcthe SSRCgboolean ssrcValidwhether @ssrc is set and validuint[15] csrcpointer to the CSRCsuint csrcCountnumber of elements in @csrc