Gst.RTP.BaseDepayload – gstreamer-rtp-1.0

Description:

[ CCode ( type_id = "gst_rtp_base_depayload_get_type ()" ) ]
[ GIR ( name = "RTPBaseDepayload" ) ]
public abstract class BaseDepayload : Element

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 gst_rtp_base_depayload_set_aggregate_hdrext_enabled. 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 gst_rtp_base_depayload_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 gst_rtp_base_depayload_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 gst_rtp_base_depayload_flush has been called.

If gst_rtp_base_depayload_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 push_list is used each buffer will see the same list of RTP header extensions.

Namespace: Gst.RTP

Content:

Properties:

public bool auto_header_extension { get; set; }

If enabled, the depayloader will automatically try to enable all the RTP header extensions provided in the sink caps, saving the application the need to handle these extensions manually using the GstRTPBaseDepayload:GstRTPBaseDepayload:request-extension: signal.

public int max_reorder { get; set; }

Max seqnum reorder before the sender is assumed to have restarted.

public bool source_info { get; set; }

Add RTP source information found in RTP header as meta to output buffer.

public Structure stats { owned get; }

Various depayloader statistics retrieved atomically (and are therefore synchroized with each other).

Creation methods:

Methods:

public virtual bool handle_event (Event event)

custom event handling

public bool is_source_info_enabled ()

Queries whether SourceMeta will be added to depayloaded buffers.

public virtual bool packet_lost (Event event)

signal the depayloader about packet loss

public virtual Buffer process (Buffer @in)

process incoming rtp packets.

public virtual Buffer process_rtp_packet (Buffer rtp_buffer)

Same as the process virtual function, but slightly more efficient, since it is passed the rtp buffer structure that has already been mapped (with GST_MAP_READ) by the base class and thus does not have to be mapped again by the subclass.

public FlowReturn push (Buffer out_buf)

Push out_buf to the peer of this.

public FlowReturn push_list (BufferList out_list)

Push out_list to the peer of this.

public virtual bool set_caps (Caps caps)

configure the depayloader

public void set_source_info_enabled (bool enable)

Enable or disable adding SourceMeta to depayloaded buffers.

Signals:

public signal void add_extension (owned HeaderExtension ext)

Add ext as an extension for reading part of an RTP header extension from incoming RTP packets.

public signal void clear_extensions ()

Clear all RTP header extensions used by this depayloader.

public signal HeaderExtension request_extension (uint ext_id, string? ext_uri)

The returned ext must be configured with the correct ext_id and with the necessary attributes as required by the extension implementation.

Fields:

Inherited Members:

BaseDepayload

Object Hierarchy: