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<?xml version="1.0"?>
<!-- This file was automatically generated from C sources - DO NOT EDIT!
To affect the contents of this file, edit the original C definitions,
and/or use gtk-doc annotations. -->
<repository version="1.2"
xmlns="http://www.gtk.org/introspection/core/1.0"
xmlns:c="http://www.gtk.org/introspection/c/1.0"
xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
<include name="Gst" version="0.10"/>
<include name="GstBase" version="0.10"/>
<package name="gstreamer-rtp-0.10"/>
<c:include name="gst/rtp/gstbasertpaudiopayload.h"/>
<c:include name="gst/rtp/gstbasertpdepayload.h"/>
<c:include name="gst/rtp/gstbasertppayload.h"/>
<c:include name="gst/rtp/gstrtcpbuffer.h"/>
<c:include name="gst/rtp/gstrtpbuffer.h"/>
<c:include name="gst/rtp/gstrtppayloads.h"/>
<namespace name="GstRtp"
version="0.10"
shared-library="libgstrtp-0.10.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
<class name="BaseRTPAudioPayload"
c:symbol-prefix="base_rtp_audio_payload"
c:type="GstBaseRTPAudioPayload"
parent="BaseRTPPayload"
glib:type-name="GstBaseRTPAudioPayload"
glib:get-type="gst_base_rtp_audio_payload_get_type"
glib:type-struct="BaseRTPAudioPayloadClass">
<doc xml:whitespace="preserve">Provides a base class for audio RTP payloaders for frame or sample based
audio codecs (constant bitrate)
This class derives from GstBaseRTPPayload. 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.
<refsect2>
<title>Usage</title>
<para>
To use this base class, your child element needs to call either
gst_base_rtp_audio_payload_set_frame_based() or
gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the
element's _init() function. Then, the child element must call either
gst_base_rtp_audio_payload_set_frame_options(),
gst_base_rtp_audio_payload_set_sample_options() or
gst_base_rtp_audio_payload_set_samplebits_options. Since
GstBaseRTPAudioPayload derives from GstBaseRTPPayload, 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 GstBaseRTPAudioPayload.
</para>
</refsect2></doc>
<method name="flush"
c:identifier="gst_base_rtp_audio_payload_flush"
version="0.10.25">
<doc xml:whitespace="preserve">Create an RTP buffer and store @payload_len bytes of the adapter as the
payload. Set the timestamp on the new buffer to @timestamp before pushing
the buffer downstream.
If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
-1, the timestamp will be calculated automatically.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="baseaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
<parameter name="payload_len" transfer-ownership="none">
<doc xml:whitespace="preserve">length of payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstClockTime</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="get_adapter"
c:identifier="gst_base_rtp_audio_payload_get_adapter"
version="0.10.13"
introspectable="0">
<doc xml:whitespace="preserve">Gets the internal adapter used by the depayloader.</doc>
<return-value>
<doc xml:whitespace="preserve">a #GstAdapter.</doc>
<type name="GstBase.Adapter" c:type="GstAdapter*"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPAudioPayload</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
</parameters>
</method>
<method name="push"
c:identifier="gst_base_rtp_audio_payload_push"
version="0.10.13">
<doc xml:whitespace="preserve">Create an RTP buffer and store @payload_len bytes of @data as the
payload. Set the timestamp on the new buffer to @timestamp before pushing
the buffer downstream.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="baseaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">data to set as payload</doc>
<type name="guint8" c:type="const guint8*"/>
</parameter>
<parameter name="payload_len" transfer-ownership="none">
<doc xml:whitespace="preserve">length of payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstClockTime</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="set_frame_based"
c:identifier="gst_base_rtp_audio_payload_set_frame_based">
<doc xml:whitespace="preserve">Tells #GstBaseRTPAudioPayload that the child element is for a frame based
audio codec</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the element.</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
</parameters>
</method>
<method name="set_frame_options"
c:identifier="gst_base_rtp_audio_payload_set_frame_options">
<doc xml:whitespace="preserve">Sets the options for frame based audio codecs.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the element.</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
<parameter name="frame_duration" transfer-ownership="none">
<doc xml:whitespace="preserve">The duraction of an audio frame in milliseconds.</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="frame_size" transfer-ownership="none">
<doc xml:whitespace="preserve">The size of an audio frame in bytes.</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</method>
<method name="set_sample_based"
c:identifier="gst_base_rtp_audio_payload_set_sample_based">
<doc xml:whitespace="preserve">Tells #GstBaseRTPAudioPayload that the child element is for a sample based
audio codec</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the element.</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
</parameters>
</method>
<method name="set_sample_options"
c:identifier="gst_base_rtp_audio_payload_set_sample_options">
<doc xml:whitespace="preserve">Sets the options for sample based audio codecs.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the element.</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
<parameter name="sample_size" transfer-ownership="none">
<doc xml:whitespace="preserve">Size per sample in bytes.</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</method>
<method name="set_samplebits_options"
c:identifier="gst_base_rtp_audio_payload_set_samplebits_options"
version="0.10.18">
<doc xml:whitespace="preserve">Sets the options for sample based audio codecs.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="basertpaudiopayload"
transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the element.</doc>
<type name="BaseRTPAudioPayload" c:type="GstBaseRTPAudioPayload*"/>
</instance-parameter>
<parameter name="sample_size" transfer-ownership="none">
<doc xml:whitespace="preserve">Size per sample in bits.</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</method>
<property name="buffer-list" writable="1" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
<field name="payload">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload"/>
</field>
<field name="priv">
<type name="BaseRTPAudioPayloadPrivate"
c:type="GstBaseRTPAudioPayloadPrivate*"/>
</field>
<field name="base_ts">
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</field>
<field name="frame_size">
<type name="gint" c:type="gint"/>
</field>
<field name="frame_duration">
<type name="gint" c:type="gint"/>
</field>
<field name="sample_size">
<type name="gint" c:type="gint"/>
</field>
<field name="_gst_reserved">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</class>
<record name="BaseRTPAudioPayloadClass"
c:type="GstBaseRTPAudioPayloadClass"
glib:is-gtype-struct-for="BaseRTPAudioPayload">
<doc xml:whitespace="preserve">Base class for audio RTP payloader.</doc>
<field name="parent_class">
<type name="BaseRTPPayloadClass" c:type="GstBaseRTPPayloadClass"/>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<record name="BaseRTPAudioPayloadPrivate"
c:type="GstBaseRTPAudioPayloadPrivate"
disguised="1">
</record>
<class name="BaseRTPDepayload"
c:symbol-prefix="base_rtp_depayload"
c:type="GstBaseRTPDepayload"
parent="Gst.Element"
glib:type-name="GstBaseRTPDepayload"
glib:get-type="gst_base_rtp_depayload_get_type"
glib:type-struct="BaseRTPDepayloadClass">
<doc xml:whitespace="preserve">Provides a base class for RTP depayloaders</doc>
<virtual-method name="add_to_queue">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="in" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="handle_event">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="packet_lost">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="process" introspectable="0">
<return-value>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<instance-parameter name="base" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="in" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="set_caps">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="caps" transfer-ownership="none">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="set_gst_timestamp">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="timestamp" transfer-ownership="none">
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="buf" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</virtual-method>
<method name="push" c:identifier="gst_base_rtp_depayload_push">
<doc xml:whitespace="preserve">Push @out_buf to the peer of @filter. This function takes ownership of
@out_buf.
Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
any timestamp on the outgoing buffer. Subclasses should therefore timestamp
outgoing buffers themselves.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPDepayload</doc>
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="out_buf" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<method name="push_list"
c:identifier="gst_base_rtp_depayload_push_list"
version="0.10.32">
<doc xml:whitespace="preserve">Push @out_list to the peer of @filter. This function takes ownership of
@out_list.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPDepayload</doc>
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="out_list" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBufferList</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</method>
<method name="push_ts" c:identifier="gst_base_rtp_depayload_push_ts">
<doc xml:whitespace="preserve">Push @out_buf to the peer of @filter. This function takes ownership of
@out_buf.
Unlike gst_base_rtp_depayload_push(), this function will by default apply
the last incomming timestamp on the outgoing buffer when it didn't have a
timestamp already. The set_get_timestamp vmethod can be overwritten to change
this behaviour (and take, for example, @timestamp into account).</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="filter" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPDepayload</doc>
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</instance-parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">an RTP timestamp to apply</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="out_buf" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<property name="queue-delay" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<field name="parent">
<type name="Gst.Element" c:type="GstElement"/>
</field>
<field name="sinkpad">
<type name="Gst.Pad" c:type="GstPad*"/>
</field>
<field name="srcpad">
<type name="Gst.Pad" c:type="GstPad*"/>
</field>
<field name="queuelock" introspectable="0">
<type c:type="GStaticRecMutex"/>
</field>
<field name="thread_running">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="thread">
<type name="GLib.Thread" c:type="GThread*"/>
</field>
<field name="clock_rate">
<type name="guint" c:type="guint"/>
</field>
<field name="queue_delay">
<type name="guint" c:type="guint"/>
</field>
<field name="queue">
<type name="GLib.Queue" c:type="GQueue*"/>
</field>
<field name="segment">
<type name="Gst.Segment" c:type="GstSegment"/>
</field>
<field name="need_newsegment">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="priv" readable="0" private="1">
<type name="BaseRTPDepayloadPrivate"
c:type="GstBaseRTPDepayloadPrivate*"/>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="3">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</class>
<record name="BaseRTPDepayloadClass"
c:type="GstBaseRTPDepayloadClass"
glib:is-gtype-struct-for="BaseRTPDepayload">
<doc xml:whitespace="preserve">Base class for audio RTP payloader.</doc>
<field name="parent_class">
<type name="Gst.ElementClass" c:type="GstElementClass"/>
</field>
<field name="set_caps">
<callback name="set_caps">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="caps" transfer-ownership="none">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="add_to_queue">
<callback name="add_to_queue">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="in" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="process" introspectable="0">
<callback name="process" introspectable="0">
<return-value>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="base" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="in" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="set_gst_timestamp">
<callback name="set_gst_timestamp">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="buf" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="packet_lost">
<callback name="packet_lost">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="handle_event">
<callback name="handle_event">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="filter" transfer-ownership="none">
<type name="BaseRTPDepayload" c:type="GstBaseRTPDepayload*"/>
</parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="2">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<record name="BaseRTPDepayloadPrivate"
c:type="GstBaseRTPDepayloadPrivate"
disguised="1">
</record>
<class name="BaseRTPPayload"
c:symbol-prefix="basertppayload"
c:type="GstBaseRTPPayload"
parent="Gst.Element"
abstract="1"
glib:type-name="GstBaseRTPPayload"
glib:get-type="gst_basertppayload_get_type"
glib:type-struct="BaseRTPPayloadClass">
<doc xml:whitespace="preserve">Provides a base class for RTP payloaders</doc>
<virtual-method name="get_caps">
<return-value transfer-ownership="full">
<type name="Gst.Caps" c:type="GstCaps*"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="pad" transfer-ownership="none">
<type name="Gst.Pad" c:type="GstPad*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="handle_buffer">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="buffer" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="set_caps">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="caps" transfer-ownership="none">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
</virtual-method>
<method name="is_filled" c:identifier="gst_basertppayload_is_filled">
<doc xml:whitespace="preserve">Check if the packet with @size and @duration would exceed the configured
maximum size.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the packet of @size and @duration would exceed the configured MTU or max_ptime.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the packet</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="duration" transfer-ownership="none">
<doc xml:whitespace="preserve">the duration of the packet</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="push" c:identifier="gst_basertppayload_push">
<doc xml:whitespace="preserve">Push @buffer to the peer element of the payloader. The SSRC, payload type,
seqnum and timestamp of the RTP buffer will be updated first.
This function takes ownership of @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<method name="push_list"
c:identifier="gst_basertppayload_push_list"
version="0.10.24">
<doc xml:whitespace="preserve">Push @list to the peer element of the payloader. The SSRC, payload type,
seqnum and timestamp of the RTP buffer will be updated first.
This function takes ownership of @list.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstFlowReturn.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBufferList</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</method>
<method name="set_options" c:identifier="gst_basertppayload_set_options">
<doc xml:whitespace="preserve">Set the rtp options of the payloader. These options will be set in the caps
of the payloader. Subclasses must call this method before calling
gst_basertppayload_push() or gst_basertppayload_set_outcaps().</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="media" transfer-ownership="none">
<doc xml:whitespace="preserve">the media type (typically "audio" or "video")</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="dynamic" transfer-ownership="none">
<doc xml:whitespace="preserve">if the payload type is dynamic</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
<parameter name="encoding_name" transfer-ownership="none">
<doc xml:whitespace="preserve">the encoding name</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="clock_rate" transfer-ownership="none">
<doc xml:whitespace="preserve">the clock rate of the media</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="set_outcaps"
c:identifier="gst_basertppayload_set_outcaps"
introspectable="0">
<doc xml:whitespace="preserve">Configure the output caps with the optional parameters.
Variable arguments should be in the form field name, field type
(as a GType), value(s). The last variable argument should be NULL.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the caps could be set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="payload" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBaseRTPPayload</doc>
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</instance-parameter>
<parameter name="fieldname" transfer-ownership="none">
<doc xml:whitespace="preserve">the first field name or %NULL</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="arg2" transfer-ownership="none">
<varargs>
</varargs>
</parameter>
</parameters>
</method>
<property name="max-ptime" writable="1" transfer-ownership="none">
<type name="gint64" c:type="gint64"/>
</property>
<property name="min-ptime" writable="1" transfer-ownership="none">
<type name="gint64" c:type="gint64"/>
</property>
<property name="mtu" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="perfect-rtptime" writable="1" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
<property name="pt" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="ptime-multiple" writable="1" transfer-ownership="none">
<type name="gint64" c:type="gint64"/>
</property>
<property name="seqnum" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="seqnum-offset" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="ssrc" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="timestamp" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="timestamp-offset" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<field name="element">
<type name="Gst.Element" c:type="GstElement"/>
</field>
<field name="sinkpad" readable="0" private="1">
<type name="Gst.Pad" c:type="GstPad*"/>
</field>
<field name="srcpad" readable="0" private="1">
<type name="Gst.Pad" c:type="GstPad*"/>
</field>
<field name="seq_rand" readable="0" private="1">
<type name="GLib.Rand" c:type="GRand*"/>
</field>
<field name="ssrc_rand" readable="0" private="1">
<type name="GLib.Rand" c:type="GRand*"/>
</field>
<field name="ts_rand" readable="0" private="1">
<type name="GLib.Rand" c:type="GRand*"/>
</field>
<field name="ts_base" readable="0" private="1">
<type name="guint32" c:type="guint32"/>
</field>
<field name="seqnum_base" readable="0" private="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="media" readable="0" private="1">
<type name="utf8" c:type="gchar*"/>
</field>
<field name="encoding_name" readable="0" private="1">
<type name="utf8" c:type="gchar*"/>
</field>
<field name="dynamic" readable="0" private="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="clock_rate" readable="0" private="1">
<type name="guint32" c:type="guint32"/>
</field>
<field name="ts_offset" readable="0" private="1">
<type name="gint32" c:type="gint32"/>
</field>
<field name="timestamp" readable="0" private="1">
<type name="guint32" c:type="guint32"/>
</field>
<field name="seqnum_offset" readable="0" private="1">
<type name="gint16" c:type="gint16"/>
</field>
<field name="seqnum" readable="0" private="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="max_ptime" readable="0" private="1">
<type name="gint64" c:type="gint64"/>
</field>
<field name="pt" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="ssrc" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="current_ssrc" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="mtu" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="segment" readable="0" private="1">
<type name="Gst.Segment" c:type="GstSegment"/>
</field>
<field name="min_ptime" readable="0" private="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="priv" readable="0" private="1">
<type name="BaseRTPPayloadPrivate" c:type="GstBaseRTPPayloadPrivate*"/>
</field>
<union name="abidata" c:type="abidata">
<record name="ABI" c:type="ABI">
<field name="ptime" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="ptime_multiple" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
</record>
<field name="_gst_reserved" writable="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="3">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</union>
</class>
<record name="BaseRTPPayloadClass"
c:type="GstBaseRTPPayloadClass"
glib:is-gtype-struct-for="BaseRTPPayload">
<doc xml:whitespace="preserve">Base class for audio RTP payloader.</doc>
<field name="parent_class">
<type name="Gst.ElementClass" c:type="GstElementClass"/>
</field>
<field name="set_caps">
<callback name="set_caps">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</parameter>
<parameter name="caps" transfer-ownership="none">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="handle_buffer">
<callback name="handle_buffer">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</parameter>
<parameter name="buffer" transfer-ownership="none">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="handle_event">
<callback name="handle_event">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="pad" transfer-ownership="none">
<type name="Gst.Pad" c:type="GstPad*"/>
</parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="get_caps">
<callback name="get_caps">
<return-value transfer-ownership="full">
<type name="Gst.Caps" c:type="GstCaps*"/>
</return-value>
<parameters>
<parameter name="payload" transfer-ownership="none">
<type name="BaseRTPPayload" c:type="GstBaseRTPPayload*"/>
</parameter>
<parameter name="pad" transfer-ownership="none">
<type name="Gst.Pad" c:type="GstPad*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="2">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<record name="BaseRTPPayloadPrivate"
c:type="GstBaseRTPPayloadPrivate"
disguised="1">
</record>
<enumeration name="RTCPFBType" version="0.10.23" c:type="GstRTCPFBType">
<doc xml:whitespace="preserve">Different types of feedback messages.</doc>
<member name="fb_type_invalid"
value="0"
c:identifier="GST_RTCP_FB_TYPE_INVALID">
<doc xml:whitespace="preserve">Invalid type</doc>
</member>
<member name="rtpfb_type_nack"
value="1"
c:identifier="GST_RTCP_RTPFB_TYPE_NACK">
<doc xml:whitespace="preserve">Generic NACK</doc>
</member>
<member name="rtpfb_type_tmmbr"
value="3"
c:identifier="GST_RTCP_RTPFB_TYPE_TMMBR">
<doc xml:whitespace="preserve">Temporary Maximum Media Stream Bit Rate Request</doc>
</member>
<member name="rtpfb_type_tmmbn"
value="4"
c:identifier="GST_RTCP_RTPFB_TYPE_TMMBN">
<doc xml:whitespace="preserve">Temporary Maximum Media Stream Bit Rate Notification</doc>
</member>
<member name="psfb_type_pli"
value="1"
c:identifier="GST_RTCP_PSFB_TYPE_PLI">
<doc xml:whitespace="preserve">Picture Loss Indication</doc>
</member>
<member name="psfb_type_sli"
value="2"
c:identifier="GST_RTCP_PSFB_TYPE_SLI">
<doc xml:whitespace="preserve">Slice Loss Indication</doc>
</member>
<member name="psfb_type_rpsi"
value="3"
c:identifier="GST_RTCP_PSFB_TYPE_RPSI">
<doc xml:whitespace="preserve">Reference Picture Selection Indication</doc>
</member>
<member name="psfb_type_afb"
value="15"
c:identifier="GST_RTCP_PSFB_TYPE_AFB">
<doc xml:whitespace="preserve">Application layer Feedback</doc>
</member>
<member name="psfb_type_fir"
value="4"
c:identifier="GST_RTCP_PSFB_TYPE_FIR">
<doc xml:whitespace="preserve">Full Intra Request Command</doc>
</member>
<member name="psfb_type_tstr"
value="5"
c:identifier="GST_RTCP_PSFB_TYPE_TSTR">
<doc xml:whitespace="preserve">Temporal-Spatial Trade-off Request</doc>
</member>
<member name="psfb_type_tstn"
value="6"
c:identifier="GST_RTCP_PSFB_TYPE_TSTN">
<doc xml:whitespace="preserve">Temporal-Spatial Trade-off Notification</doc>
</member>
<member name="psfb_type_vbcn"
value="7"
c:identifier="GST_RTCP_PSFB_TYPE_VBCN">
<doc xml:whitespace="preserve">Video Back Channel Message</doc>
</member>
</enumeration>
<record name="RTCPPacket" c:type="GstRTCPPacket">
<doc xml:whitespace="preserve">Data structure that points to a packet at @offset in @buffer.
The size of the structure is made public to allow stack allocations.</doc>
<field name="buffer" writable="1">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</field>
<field name="offset" writable="1">
<type name="guint" c:type="guint"/>
</field>
<field name="padding" readable="0" private="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="count" readable="0" private="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="type" readable="0" private="1">
<type name="RTCPType" c:type="GstRTCPType"/>
</field>
<field name="length" readable="0" private="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="item_offset" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="item_count" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="entry_offset" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<method name="add_rb" c:identifier="gst_rtcp_packet_add_rb">
<doc xml:whitespace="preserve">Add a new report block to @packet with the given values.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the packet was created. This function can return %FALSE if the max MTU is exceeded or the number of report blocks is greater than #GST_RTCP_MAX_RB_COUNT.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR or RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">data source being reported</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="fractionlost" transfer-ownership="none">
<doc xml:whitespace="preserve">fraction lost since last SR/RR</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="packetslost" transfer-ownership="none">
<doc xml:whitespace="preserve">the cumululative number of packets lost</doc>
<type name="gint32" c:type="gint32"/>
</parameter>
<parameter name="exthighestseq" transfer-ownership="none">
<doc xml:whitespace="preserve">the extended last sequence number received</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="jitter" transfer-ownership="none">
<doc xml:whitespace="preserve">the interarrival jitter</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="lsr" transfer-ownership="none">
<doc xml:whitespace="preserve">the last SR packet from this source</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="dlsr" transfer-ownership="none">
<doc xml:whitespace="preserve">the delay since last SR packet</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="bye_add_ssrc" c:identifier="gst_rtcp_packet_bye_add_ssrc">
<doc xml:whitespace="preserve">Add @ssrc to the BYE @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the ssrc was added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">an SSRC to add</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="bye_add_ssrcs"
c:identifier="gst_rtcp_packet_bye_add_ssrcs">
<doc xml:whitespace="preserve">Adds @len SSRCs in @ssrc to BYE @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the all the SSRCs were added. This function can return %FALSE if the max MTU is exceeded or the number of sources blocks is greater than #GST_RTCP_MAX_BYE_SSRC_COUNT.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">an array of SSRCs to add</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">number of elements in @ssrc</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</method>
<method name="bye_get_nth_ssrc"
c:identifier="gst_rtcp_packet_bye_get_nth_ssrc">
<doc xml:whitespace="preserve">Get the @nth SSRC of the BYE @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The @nth SSRC of @packet.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">the nth SSRC to get</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</method>
<method name="bye_get_reason"
c:identifier="gst_rtcp_packet_bye_get_reason">
<doc xml:whitespace="preserve">Get the reason in @packet.</doc>
<return-value transfer-ownership="full">
<doc xml:whitespace="preserve">The reason for the BYE @packet or NULL if the packet did not contain a reason string. The string must be freed with g_free() after usage.</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="bye_get_reason_len"
c:identifier="gst_rtcp_packet_bye_get_reason_len">
<doc xml:whitespace="preserve">Get the length of the reason string.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of the reason string or 0 when there is no reason string present.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="bye_get_ssrc_count"
c:identifier="gst_rtcp_packet_bye_get_ssrc_count">
<doc xml:whitespace="preserve">Get the number of SSRC fields in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The number of SSRC fields in @packet.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="bye_set_reason"
c:identifier="gst_rtcp_packet_bye_set_reason">
<doc xml:whitespace="preserve">Set the reason string to @reason in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if the string could be set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid BYE #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="reason" transfer-ownership="none">
<doc xml:whitespace="preserve">a reason string</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</method>
<method name="fb_get_fci"
c:identifier="gst_rtcp_packet_fb_get_fci"
version="0.10.31">
<doc xml:whitespace="preserve">Get the Feedback Control Information attached to a RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a pointer to the FCI</doc>
<type name="guint8" c:type="guint8*"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="fb_get_fci_length"
c:identifier="gst_rtcp_packet_fb_get_fci_length"
version="0.10.31">
<doc xml:whitespace="preserve">Get the length of the Feedback Control Information attached to a
RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of the FCI in 32-bit words.</doc>
<type name="guint16" c:type="guint16"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="fb_get_media_ssrc"
c:identifier="gst_rtcp_packet_fb_get_media_ssrc"
version="0.10.23">
<doc xml:whitespace="preserve">Get the media SSRC field of the RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the media SSRC.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="fb_get_sender_ssrc"
c:identifier="gst_rtcp_packet_fb_get_sender_ssrc"
version="0.10.23">
<doc xml:whitespace="preserve">Get the sender SSRC field of the RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the sender SSRC.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="fb_get_type"
c:identifier="gst_rtcp_packet_fb_get_type"
version="0.10.23">
<doc xml:whitespace="preserve">Get the feedback message type of the FB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The feedback message type.</doc>
<type name="RTCPFBType" c:type="GstRTCPFBType"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="fb_set_fci_length"
c:identifier="gst_rtcp_packet_fb_set_fci_length"
version="0.10.31">
<doc xml:whitespace="preserve">Set the length of the Feedback Control Information attached to a
RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if there was enough space in the packet to add this much FCI</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="wordlen" transfer-ownership="none">
<doc xml:whitespace="preserve">Length of the FCI in 32-bit words</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</method>
<method name="fb_set_media_ssrc"
c:identifier="gst_rtcp_packet_fb_set_media_ssrc"
version="0.10.23">
<doc xml:whitespace="preserve">Set the media SSRC field of the RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">a media SSRC</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="fb_set_sender_ssrc"
c:identifier="gst_rtcp_packet_fb_set_sender_ssrc"
version="0.10.23">
<doc xml:whitespace="preserve">Set the sender SSRC field of the RTPFB or PSFB @packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">a sender SSRC</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="fb_set_type"
c:identifier="gst_rtcp_packet_fb_set_type"
version="0.10.23">
<doc xml:whitespace="preserve">Set the feedback message type of the FB @packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTPFB or PSFB #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">the #GstRTCPFBType to set</doc>
<type name="RTCPFBType" c:type="GstRTCPFBType"/>
</parameter>
</parameters>
</method>
<method name="get_count" c:identifier="gst_rtcp_packet_get_count">
<doc xml:whitespace="preserve">Get the count field in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The count field in @packet or -1 if @packet does not point to a valid packet.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_length" c:identifier="gst_rtcp_packet_get_length">
<doc xml:whitespace="preserve">Get the length field of @packet. This is the length of the packet in
32-bit words minus one.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length field of @packet.</doc>
<type name="guint16" c:type="guint16"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_padding" c:identifier="gst_rtcp_packet_get_padding">
<doc xml:whitespace="preserve">Get the packet padding of the packet pointed to by @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">If the packet has the padding bit set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_rb" c:identifier="gst_rtcp_packet_get_rb">
<doc xml:whitespace="preserve">Parse the values of the @nth report block in @packet and store the result in
the values.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR or RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">the nth report block in @packet</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">result for data source being reported</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="fractionlost" transfer-ownership="none">
<doc xml:whitespace="preserve">result for fraction lost since last SR/RR</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="packetslost" transfer-ownership="none">
<doc xml:whitespace="preserve">result for the cumululative number of packets lost</doc>
<type name="gint32" c:type="gint32*"/>
</parameter>
<parameter name="exthighestseq" transfer-ownership="none">
<doc xml:whitespace="preserve">result for the extended last sequence number received</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="jitter" transfer-ownership="none">
<doc xml:whitespace="preserve">result for the interarrival jitter</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="lsr" transfer-ownership="none">
<doc xml:whitespace="preserve">result for the last SR packet from this source</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="dlsr" transfer-ownership="none">
<doc xml:whitespace="preserve">result for the delay since last SR packet</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
</parameters>
</method>
<method name="get_rb_count" c:identifier="gst_rtcp_packet_get_rb_count">
<doc xml:whitespace="preserve">Get the number of report blocks in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The number of report blocks in @packet.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR or RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_type" c:identifier="gst_rtcp_packet_get_type">
<doc xml:whitespace="preserve">Get the packet type of the packet pointed to by @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The packet type or GST_RTCP_TYPE_INVALID when @packet is not pointing to a valid packet.</doc>
<type name="RTCPType" c:type="GstRTCPType"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="move_to_next" c:identifier="gst_rtcp_packet_move_to_next">
<doc xml:whitespace="preserve">Move the packet pointer @packet to the next packet in the payload.
Use gst_rtcp_buffer_get_first_packet() to initialize @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @packet is pointing to a valid packet after calling this function.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="remove" c:identifier="gst_rtcp_packet_remove">
<doc xml:whitespace="preserve">Removes the packet pointed to by @packet and moves pointer to the next one</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @packet is pointing to a valid packet after calling this function.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="rr_get_ssrc" c:identifier="gst_rtcp_packet_rr_get_ssrc">
<doc xml:whitespace="preserve">Get the ssrc field of the RR @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the ssrc.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="rr_set_ssrc" c:identifier="gst_rtcp_packet_rr_set_ssrc">
<doc xml:whitespace="preserve">Set the ssrc field of the RR @packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC to set</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="sdes_add_entry"
c:identifier="gst_rtcp_packet_sdes_add_entry">
<doc xml:whitespace="preserve">Add a new SDES entry to the current item in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the item could be added, %FALSE if the MTU has been reached.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">the #GstRTCPSDESType of the SDES entry</doc>
<type name="RTCPSDESType" c:type="GstRTCPSDESType"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the data length</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">the data</doc>
<type name="guint8" c:type="const guint8*"/>
</parameter>
</parameters>
</method>
<method name="sdes_add_item"
c:identifier="gst_rtcp_packet_sdes_add_item">
<doc xml:whitespace="preserve">Add a new SDES item for @ssrc to @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the item could be added, %FALSE if the maximum amount of items has been exceeded for the SDES packet or the MTU has been reached.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC of the new item to add</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="sdes_copy_entry"
c:identifier="gst_rtcp_packet_sdes_copy_entry">
<doc xml:whitespace="preserve">This function is like gst_rtcp_packet_sdes_get_entry() but it returns a
null-terminated copy of the data instead. use g_free() after usage.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if there was valid data.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">result of the entry type</doc>
<type name="RTCPSDESType" c:type="GstRTCPSDESType*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">result length of the entry data</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">result entry data</doc>
<type name="guint8" c:type="guint8**"/>
</parameter>
</parameters>
</method>
<method name="sdes_first_entry"
c:identifier="gst_rtcp_packet_sdes_first_entry">
<doc xml:whitespace="preserve">Move to the first SDES entry in the current item.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if there was a first entry.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="sdes_first_item"
c:identifier="gst_rtcp_packet_sdes_first_item">
<doc xml:whitespace="preserve">Move to the first SDES item in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if there was a first item.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="sdes_get_entry"
c:identifier="gst_rtcp_packet_sdes_get_entry">
<doc xml:whitespace="preserve">Get the data of the current SDES item entry. @type (when not NULL) will
contain the type of the entry. @data (when not NULL) will point to @len
bytes.
When @type refers to a text item, @data will point to a UTF8 string. Note
that this UTF8 string is NOT null-terminated. Use
gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if there was valid data.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">result of the entry type</doc>
<type name="RTCPSDESType" c:type="GstRTCPSDESType*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">result length of the entry data</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">result entry data</doc>
<type name="guint8" c:type="guint8**"/>
</parameter>
</parameters>
</method>
<method name="sdes_get_item_count"
c:identifier="gst_rtcp_packet_sdes_get_item_count">
<doc xml:whitespace="preserve">Get the number of items in the SDES packet @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The number of items in @packet.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="sdes_get_ssrc"
c:identifier="gst_rtcp_packet_sdes_get_ssrc">
<doc xml:whitespace="preserve">Get the SSRC of the current SDES item.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC of the current item.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="sdes_next_entry"
c:identifier="gst_rtcp_packet_sdes_next_entry">
<doc xml:whitespace="preserve">Move to the next SDES entry in the current item.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if there was a next entry.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="sdes_next_item"
c:identifier="gst_rtcp_packet_sdes_next_item">
<doc xml:whitespace="preserve">Move to the next SDES item in @packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if there was a next item.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SDES #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
</parameters>
</method>
<method name="set_rb" c:identifier="gst_rtcp_packet_set_rb">
<doc xml:whitespace="preserve">Set the @nth new report block in @packet with the given values.
Note: Not implemented.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR or RR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">the nth report block to set</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">data source being reported</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="fractionlost" transfer-ownership="none">
<doc xml:whitespace="preserve">fraction lost since last SR/RR</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="packetslost" transfer-ownership="none">
<doc xml:whitespace="preserve">the cumululative number of packets lost</doc>
<type name="gint32" c:type="gint32"/>
</parameter>
<parameter name="exthighestseq" transfer-ownership="none">
<doc xml:whitespace="preserve">the extended last sequence number received</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="jitter" transfer-ownership="none">
<doc xml:whitespace="preserve">the interarrival jitter</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="lsr" transfer-ownership="none">
<doc xml:whitespace="preserve">the last SR packet from this source</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="dlsr" transfer-ownership="none">
<doc xml:whitespace="preserve">the delay since last SR packet</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
<method name="sr_get_sender_info"
c:identifier="gst_rtcp_packet_sr_get_sender_info">
<doc xml:whitespace="preserve">Parse the SR sender info and store the values.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">result SSRC</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="ntptime" transfer-ownership="none">
<doc xml:whitespace="preserve">result NTP time</doc>
<type name="guint64" c:type="guint64*"/>
</parameter>
<parameter name="rtptime" transfer-ownership="none">
<doc xml:whitespace="preserve">result RTP time</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="packet_count" transfer-ownership="none">
<doc xml:whitespace="preserve">result packet count</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
<parameter name="octet_count" transfer-ownership="none">
<doc xml:whitespace="preserve">result octet count</doc>
<type name="guint32" c:type="guint32*"/>
</parameter>
</parameters>
</method>
<method name="sr_set_sender_info"
c:identifier="gst_rtcp_packet_sr_set_sender_info">
<doc xml:whitespace="preserve">Set the given values in the SR packet @packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid SR #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</instance-parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="ntptime" transfer-ownership="none">
<doc xml:whitespace="preserve">the NTP time</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
<parameter name="rtptime" transfer-ownership="none">
<doc xml:whitespace="preserve">the RTP time</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="packet_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the packet count</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="octet_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the octet count</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</method>
</record>
<enumeration name="RTCPSDESType" c:type="GstRTCPSDESType">
<doc xml:whitespace="preserve">Different types of SDES content.</doc>
<member name="invalid" value="-1" c:identifier="GST_RTCP_SDES_INVALID">
<doc xml:whitespace="preserve">Invalid SDES entry</doc>
</member>
<member name="end" value="0" c:identifier="GST_RTCP_SDES_END">
<doc xml:whitespace="preserve">End of SDES list</doc>
</member>
<member name="cname" value="1" c:identifier="GST_RTCP_SDES_CNAME">
<doc xml:whitespace="preserve">Canonical name</doc>
</member>
<member name="name" value="2" c:identifier="GST_RTCP_SDES_NAME">
<doc xml:whitespace="preserve">User name</doc>
</member>
<member name="email" value="3" c:identifier="GST_RTCP_SDES_EMAIL">
<doc xml:whitespace="preserve">User's electronic mail address</doc>
</member>
<member name="phone" value="4" c:identifier="GST_RTCP_SDES_PHONE">
<doc xml:whitespace="preserve">User's phone number</doc>
</member>
<member name="loc" value="5" c:identifier="GST_RTCP_SDES_LOC">
<doc xml:whitespace="preserve">Geographic user location</doc>
</member>
<member name="tool" value="6" c:identifier="GST_RTCP_SDES_TOOL">
<doc xml:whitespace="preserve">Name of application or tool</doc>
</member>
<member name="note" value="7" c:identifier="GST_RTCP_SDES_NOTE">
<doc xml:whitespace="preserve">Notice about the source</doc>
</member>
<member name="priv" value="8" c:identifier="GST_RTCP_SDES_PRIV">
<doc xml:whitespace="preserve">Private extensions</doc>
</member>
</enumeration>
<enumeration name="RTCPType" c:type="GstRTCPType">
<doc xml:whitespace="preserve">Different RTCP packet types.</doc>
<member name="invalid" value="0" c:identifier="GST_RTCP_TYPE_INVALID">
<doc xml:whitespace="preserve">Invalid type</doc>
</member>
<member name="sr" value="200" c:identifier="GST_RTCP_TYPE_SR">
<doc xml:whitespace="preserve">Sender report</doc>
</member>
<member name="rr" value="201" c:identifier="GST_RTCP_TYPE_RR">
<doc xml:whitespace="preserve">Receiver report</doc>
</member>
<member name="sdes" value="202" c:identifier="GST_RTCP_TYPE_SDES">
<doc xml:whitespace="preserve">Source description</doc>
</member>
<member name="bye" value="203" c:identifier="GST_RTCP_TYPE_BYE">
<doc xml:whitespace="preserve">Goodbye</doc>
</member>
<member name="app" value="204" c:identifier="GST_RTCP_TYPE_APP">
<doc xml:whitespace="preserve">Application defined</doc>
</member>
<member name="rtpfb" value="205" c:identifier="GST_RTCP_TYPE_RTPFB">
<doc xml:whitespace="preserve">Transport layer feedback. Since: 0.10.23</doc>
</member>
<member name="psfb" value="206" c:identifier="GST_RTCP_TYPE_PSFB">
<doc xml:whitespace="preserve">Payload-specific feedback. Since: 0.10.23</doc>
</member>
</enumeration>
<constant name="RTCP_MAX_BYE_SSRC_COUNT"
value="31"
c:type="GST_RTCP_MAX_BYE_SSRC_COUNT">
<doc xml:whitespace="preserve">The maximum amount of SSRCs in a BYE packet.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_MAX_RB_COUNT"
value="31"
c:type="GST_RTCP_MAX_RB_COUNT">
<doc xml:whitespace="preserve">The maximum amount of Receiver report blocks in RR and SR messages.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_MAX_SDES" value="255" c:type="GST_RTCP_MAX_SDES">
<doc xml:whitespace="preserve">The maximum text length for an SDES item.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_MAX_SDES_ITEM_COUNT"
value="31"
c:type="GST_RTCP_MAX_SDES_ITEM_COUNT">
<doc xml:whitespace="preserve">The maximum amount of SDES items.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_VALID_MASK"
value="57598"
c:type="GST_RTCP_VALID_MASK">
<doc xml:whitespace="preserve">Mask for version, padding bit and packet type pair</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_VALID_VALUE" value="0" c:type="GST_RTCP_VALID_VALUE">
<doc xml:whitespace="preserve">Valid value for the first two bytes of an RTCP packet after applying
#GST_RTCP_VALID_MASK to them.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTCP_VERSION" value="2" c:type="GST_RTCP_VERSION">
<doc xml:whitespace="preserve">The supported RTCP version 2.</doc>
<type name="gint" c:type="gint"/>
</constant>
<enumeration name="RTPPayload" c:type="GstRTPPayload">
<doc xml:whitespace="preserve">Standard predefined fixed payload types.
The official list is at:
http://www.iana.org/assignments/rtp-parameters
Audio:
reserved: 19
unassigned: 20-23,
Video:
unassigned: 24, 27, 29, 30, 35-71, 77-95
Reserved for RTCP conflict avoidance: 72-76</doc>
<member name="pcmu" value="0" c:identifier="GST_RTP_PAYLOAD_PCMU">
<doc xml:whitespace="preserve">ITU-T G.711. mu-law audio (RFC 3551)</doc>
</member>
<member name="1016" value="1" c:identifier="GST_RTP_PAYLOAD_1016">
<doc xml:whitespace="preserve">RFC 3551 says reserved</doc>
</member>
<member name="g721" value="2" c:identifier="GST_RTP_PAYLOAD_G721">
<doc xml:whitespace="preserve">RFC 3551 says reserved</doc>
</member>
<member name="gsm" value="3" c:identifier="GST_RTP_PAYLOAD_GSM">
<doc xml:whitespace="preserve">GSM audio</doc>
</member>
<member name="g723" value="4" c:identifier="GST_RTP_PAYLOAD_G723">
<doc xml:whitespace="preserve">ITU G.723.1 audio</doc>
</member>
<member name="dvi4_8000"
value="5"
c:identifier="GST_RTP_PAYLOAD_DVI4_8000">
<doc xml:whitespace="preserve">IMA ADPCM wave type (RFC 3551)</doc>
</member>
<member name="dvi4_16000"
value="6"
c:identifier="GST_RTP_PAYLOAD_DVI4_16000">
<doc xml:whitespace="preserve">IMA ADPCM wave type (RFC 3551)</doc>
</member>
<member name="lpc" value="7" c:identifier="GST_RTP_PAYLOAD_LPC">
<doc xml:whitespace="preserve">experimental linear predictive encoding</doc>
</member>
<member name="pcma" value="8" c:identifier="GST_RTP_PAYLOAD_PCMA">
<doc xml:whitespace="preserve">ITU-T G.711 A-law audio (RFC 3551)</doc>
</member>
<member name="g722" value="9" c:identifier="GST_RTP_PAYLOAD_G722">
<doc xml:whitespace="preserve">ITU-T G.722 (RFC 3551)</doc>
</member>
<member name="l16_stereo"
value="10"
c:identifier="GST_RTP_PAYLOAD_L16_STEREO">
<doc xml:whitespace="preserve">stereo PCM</doc>
</member>
<member name="l16_mono"
value="11"
c:identifier="GST_RTP_PAYLOAD_L16_MONO">
<doc xml:whitespace="preserve">mono PCM</doc>
</member>
<member name="qcelp" value="12" c:identifier="GST_RTP_PAYLOAD_QCELP">
<doc xml:whitespace="preserve">EIA & TIA standard IS-733</doc>
</member>
<member name="cn" value="13" c:identifier="GST_RTP_PAYLOAD_CN">
<doc xml:whitespace="preserve">Comfort Noise (RFC 3389)</doc>
</member>
<member name="mpa" value="14" c:identifier="GST_RTP_PAYLOAD_MPA">
<doc xml:whitespace="preserve">Audio MPEG 1-3.</doc>
</member>
<member name="g728" value="15" c:identifier="GST_RTP_PAYLOAD_G728">
<doc xml:whitespace="preserve">ITU-T G.728 Speech coder (RFC 3551)</doc>
</member>
<member name="dvi4_11025"
value="16"
c:identifier="GST_RTP_PAYLOAD_DVI4_11025">
<doc xml:whitespace="preserve">IMA ADPCM wave type (RFC 3551)</doc>
</member>
<member name="dvi4_22050"
value="17"
c:identifier="GST_RTP_PAYLOAD_DVI4_22050">
<doc xml:whitespace="preserve">IMA ADPCM wave type (RFC 3551)</doc>
</member>
<member name="g729" value="18" c:identifier="GST_RTP_PAYLOAD_G729">
<doc xml:whitespace="preserve">ITU-T G.729 Speech coder (RFC 3551)</doc>
</member>
<member name="cellb" value="25" c:identifier="GST_RTP_PAYLOAD_CELLB">
<doc xml:whitespace="preserve">See RFC 2029</doc>
</member>
<member name="jpeg" value="26" c:identifier="GST_RTP_PAYLOAD_JPEG">
<doc xml:whitespace="preserve">ISO Standards 10918-1 and 10918-2 (RFC 2435)</doc>
</member>
<member name="nv" value="28" c:identifier="GST_RTP_PAYLOAD_NV">
<doc xml:whitespace="preserve">nv encoding by Ron Frederick</doc>
</member>
<member name="h261" value="31" c:identifier="GST_RTP_PAYLOAD_H261">
<doc xml:whitespace="preserve">ITU-T Recommendation H.261 (RFC 2032)</doc>
</member>
<member name="mpv" value="32" c:identifier="GST_RTP_PAYLOAD_MPV">
<doc xml:whitespace="preserve">Video MPEG 1 & 2 (RFC 2250)</doc>
</member>
<member name="mp2t" value="33" c:identifier="GST_RTP_PAYLOAD_MP2T">
<doc xml:whitespace="preserve">MPEG-2 transport stream (RFC 2250)</doc>
</member>
<member name="h263" value="34" c:identifier="GST_RTP_PAYLOAD_H263">
<doc xml:whitespace="preserve">Video H263 (RFC 2190)</doc>
</member>
</enumeration>
<record name="RTPPayloadInfo" c:type="GstRTPPayloadInfo">
<doc xml:whitespace="preserve">Structure holding default payload type information.</doc>
<field name="payload_type" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="media" writable="1">
<type name="utf8" c:type="const gchar*"/>
</field>
<field name="encoding_name" writable="1">
<type name="utf8" c:type="const gchar*"/>
</field>
<field name="clock_rate" writable="1">
<type name="guint" c:type="guint"/>
</field>
<field name="encoding_parameters" writable="1">
<type name="utf8" c:type="const gchar*"/>
</field>
<field name="bitrate" writable="1">
<type name="guint" c:type="guint"/>
</field>
<function name="for_name" c:identifier="gst_rtp_payload_info_for_name">
<doc xml:whitespace="preserve">Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is
mostly used to get the default clock-rate and bandwidth for dynamic payload
types specified with @media and @encoding name.
The search for @encoding_name will be performed in a case insensitve way.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
<type name="RTPPayloadInfo" c:type="const GstRTPPayloadInfo*"/>
</return-value>
<parameters>
<parameter name="media" transfer-ownership="none">
<doc xml:whitespace="preserve">the media to find</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="encoding_name" transfer-ownership="none">
<doc xml:whitespace="preserve">the encoding name to find</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</function>
<function name="for_pt" c:identifier="gst_rtp_payload_info_for_pt">
<doc xml:whitespace="preserve">Get the #GstRTPPayloadInfo for @payload_type. This function is
mostly used to get the default clock-rate and bandwidth for static payload
types specified with @payload_type.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
<type name="RTPPayloadInfo" c:type="const GstRTPPayloadInfo*"/>
</return-value>
<parameters>
<parameter name="payload_type" transfer-ownership="none">
<doc xml:whitespace="preserve">the payload_type to find</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
</record>
<constant name="RTP_PAYLOAD_1016_STRING"
value="1"
c:type="GST_RTP_PAYLOAD_1016_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_CELLB_STRING"
value="25"
c:type="GST_RTP_PAYLOAD_CELLB_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_CN_STRING"
value="13"
c:type="GST_RTP_PAYLOAD_CN_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_DVI4_11025_STRING"
value="16"
c:type="GST_RTP_PAYLOAD_DVI4_11025_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_DVI4_16000_STRING"
value="6"
c:type="GST_RTP_PAYLOAD_DVI4_16000_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_DVI4_22050_STRING"
value="17"
c:type="GST_RTP_PAYLOAD_DVI4_22050_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_DVI4_8000_STRING"
value="5"
c:type="GST_RTP_PAYLOAD_DVI4_8000_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_DYNAMIC_STRING"
value="[96, 127]"
c:type="GST_RTP_PAYLOAD_DYNAMIC_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G721_STRING"
value="2"
c:type="GST_RTP_PAYLOAD_G721_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G722_STRING"
value="9"
c:type="GST_RTP_PAYLOAD_G722_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G723_53"
value="17"
c:type="GST_RTP_PAYLOAD_G723_53">
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTP_PAYLOAD_G723_53_STRING"
value="17"
c:type="GST_RTP_PAYLOAD_G723_53_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G723_63"
value="16"
c:type="GST_RTP_PAYLOAD_G723_63">
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTP_PAYLOAD_G723_63_STRING"
value="16"
c:type="GST_RTP_PAYLOAD_G723_63_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G723_STRING"
value="4"
c:type="GST_RTP_PAYLOAD_G723_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G728_STRING"
value="15"
c:type="GST_RTP_PAYLOAD_G728_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_G729_STRING"
value="18"
c:type="GST_RTP_PAYLOAD_G729_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_GSM_STRING"
value="3"
c:type="GST_RTP_PAYLOAD_GSM_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_H261_STRING"
value="31"
c:type="GST_RTP_PAYLOAD_H261_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_H263_STRING"
value="34"
c:type="GST_RTP_PAYLOAD_H263_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_JPEG_STRING"
value="26"
c:type="GST_RTP_PAYLOAD_JPEG_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_L16_MONO_STRING"
value="11"
c:type="GST_RTP_PAYLOAD_L16_MONO_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_L16_STEREO_STRING"
value="10"
c:type="GST_RTP_PAYLOAD_L16_STEREO_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_LPC_STRING"
value="7"
c:type="GST_RTP_PAYLOAD_LPC_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_MP2T_STRING"
value="33"
c:type="GST_RTP_PAYLOAD_MP2T_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_MPA_STRING"
value="14"
c:type="GST_RTP_PAYLOAD_MPA_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_MPV_STRING"
value="32"
c:type="GST_RTP_PAYLOAD_MPV_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_NV_STRING"
value="28"
c:type="GST_RTP_PAYLOAD_NV_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_PCMA_STRING"
value="8"
c:type="GST_RTP_PAYLOAD_PCMA_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_PCMU_STRING"
value="0"
c:type="GST_RTP_PAYLOAD_PCMU_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_QCELP_STRING"
value="12"
c:type="GST_RTP_PAYLOAD_QCELP_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_TS41" value="19" c:type="GST_RTP_PAYLOAD_TS41">
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTP_PAYLOAD_TS41_STRING"
value="19"
c:type="GST_RTP_PAYLOAD_TS41_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_PAYLOAD_TS48" value="18" c:type="GST_RTP_PAYLOAD_TS48">
<type name="gint" c:type="gint"/>
</constant>
<constant name="RTP_PAYLOAD_TS48_STRING"
value="18"
c:type="GST_RTP_PAYLOAD_TS48_STRING">
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="RTP_VERSION" value="2" c:type="GST_RTP_VERSION">
<doc xml:whitespace="preserve">The supported RTP version 2.</doc>
<type name="gint" c:type="gint"/>
</constant>
<function name="rtcp_buffer_add_packet"
c:identifier="gst_rtcp_buffer_add_packet">
<doc xml:whitespace="preserve">Add a new packet of @type to @buffer. @packet will point to the newly created
packet.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if the packet could be created. This function returns %FALSE if the max mtu is exceeded for the buffer.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTCP buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">the #GstRTCPType of the new packet</doc>
<type name="RTCPType" c:type="GstRTCPType"/>
</parameter>
<parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">pointer to new packet</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_end" c:identifier="gst_rtcp_buffer_end">
<doc xml:whitespace="preserve">Finish @buffer after being constructured. This function is usually called
after gst_rtcp_buffer_new() and after adding the RTCP items to the new buffer.
The function adjusts the size of @buffer with the total length of all the
added packets.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a buffer with an RTCP packet</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_get_first_packet"
c:identifier="gst_rtcp_buffer_get_first_packet">
<doc xml:whitespace="preserve">Initialize a new #GstRTCPPacket pointer that points to the first packet in
@buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if the packet existed in @buffer.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTCP buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="packet" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTCPPacket</doc>
<type name="RTCPPacket" c:type="GstRTCPPacket*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_get_packet_count"
c:identifier="gst_rtcp_buffer_get_packet_count">
<doc xml:whitespace="preserve">Get the number of RTCP packets in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the number of RTCP packets in @buffer.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a valid RTCP buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_new"
c:identifier="gst_rtcp_buffer_new"
introspectable="0">
<doc xml:whitespace="preserve">Create a new buffer for constructing RTCP packets. The packet will have a
maximum size of @mtu.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="mtu" transfer-ownership="none">
<doc xml:whitespace="preserve">the maximum mtu size.</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_new_copy_data"
c:identifier="gst_rtcp_buffer_new_copy_data"
introspectable="0">
<doc xml:whitespace="preserve">Create a new buffer and set the data to a copy of @len
bytes of @data and the size to @len. The data will be freed when the buffer
is freed.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer with a copy of @data and of size @len.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">data for the new buffer</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of data</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_new_take_data"
c:identifier="gst_rtcp_buffer_new_take_data"
introspectable="0">
<doc xml:whitespace="preserve">Create a new buffer and set the data and size of the buffer to @data and @len
respectively. @data will be freed when the buffer is unreffed, so this
function transfers ownership of @data to the new buffer.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer with @data and of size @len.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">data for the new buffer</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of data</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_validate"
c:identifier="gst_rtcp_buffer_validate">
<doc xml:whitespace="preserve">Check if the data pointed to by @buffer is a valid RTCP packet using
gst_rtcp_buffer_validate_data().</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer is a valid RTCP packet.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer to validate</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_buffer_validate_data"
c:identifier="gst_rtcp_buffer_validate_data">
<doc xml:whitespace="preserve">Check if the @data and @size point to the data of a valid RTCP (compound)
packet.
Use this function to validate a packet before using the other functions in
this module.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if the data points to a valid RTCP packet.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">the data to validate</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of @data to validate</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtcp_ntp_to_unix" c:identifier="gst_rtcp_ntp_to_unix">
<doc xml:whitespace="preserve">Converts an NTP time to UNIX nanoseconds. @ntptime can typically be
the NTP time of an SR RTCP message and contains, in the upper 32 bits, the
number of seconds since 1900 and, in the lower 32 bits, the fractional
seconds. The resulting value will be the number of nanoseconds since 1970.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the UNIX time for @ntptime in nanoseconds.</doc>
<type name="guint64" c:type="guint64"/>
</return-value>
<parameters>
<parameter name="ntptime" transfer-ownership="none">
<doc xml:whitespace="preserve">an NTP timestamp</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
</parameters>
</function>
<function name="rtcp_sdes_name_to_type"
c:identifier="gst_rtcp_sdes_name_to_type"
version="0.10.26">
<doc xml:whitespace="preserve">Convert @name into a @GstRTCPSDESType. @name is typically a key in a
#GstStructure containing SDES items.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name is a private sdes item.</doc>
<type name="RTCPSDESType" c:type="GstRTCPSDESType"/>
</return-value>
<parameters>
<parameter name="name" transfer-ownership="none">
<doc xml:whitespace="preserve">a SDES name</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</function>
<function name="rtcp_sdes_type_to_name"
c:identifier="gst_rtcp_sdes_type_to_name"
version="0.10.26">
<doc xml:whitespace="preserve">Converts @type to the string equivalent. The string is typically used as a
key in a #GstStructure containing SDES items.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the string equivalent of @type</doc>
<type name="utf8" c:type="const gchar*"/>
</return-value>
<parameters>
<parameter name="type" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTCPSDESType</doc>
<type name="RTCPSDESType" c:type="GstRTCPSDESType"/>
</parameter>
</parameters>
</function>
<function name="rtcp_unix_to_ntp" c:identifier="gst_rtcp_unix_to_ntp">
<doc xml:whitespace="preserve">Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should
pass a value with nanoseconds since 1970. The NTP time will, in the upper
32 bits, contain the number of seconds since 1900 and, in the lower 32
bits, the fractional seconds. The resulting value can be used as an ntptime
for constructing SR RTCP packets.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the NTP time for @unixtime.</doc>
<type name="guint64" c:type="guint64"/>
</return-value>
<parameters>
<parameter name="unixtime" transfer-ownership="none">
<doc xml:whitespace="preserve">an UNIX timestamp in nanoseconds</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_add_extension_onebyte_header"
c:identifier="gst_rtp_buffer_add_extension_onebyte_header"
version="0.10.31">
<doc xml:whitespace="preserve">Adds a RFC 5285 header extension with a one byte header to the end of the
RTP header. If there is already a RFC 5285 header extension with a one byte
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
a two bytes header as described in RFC 5285. In that case, use
gst_rtp_buffer_add_extension_twobytes_header()</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if header extension could be added</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_add_extension_twobytes_header"
c:identifier="gst_rtp_buffer_add_extension_twobytes_header"
version="0.10.31">
<doc xml:whitespace="preserve">Adds a RFC 5285 header extension with a two bytes header to the end of the
RTP header. If there is already a RFC 5285 header extension with a two bytes
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
a one byte header as described in RFC 5285. In that case, use
gst_rtp_buffer_add_extension_onebyte_header()</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if header extension could be added</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="appbits" transfer-ownership="none">
<doc xml:whitespace="preserve">Application specific bits</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_allocate_data"
c:identifier="gst_rtp_buffer_allocate_data">
<doc xml:whitespace="preserve">Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs,
a payload length of @payload_len and padding of @pad_len.
MALLOCDATA of @buffer will be overwritten and will not be freed.
All other RTP header fields will be set to 0/FALSE.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="payload_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of the payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pad_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the amount of padding</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_calc_header_len"
c:identifier="gst_rtp_buffer_calc_header_len">
<doc xml:whitespace="preserve">Calculate the header length of an RTP packet with @csrc_count CSRC entries.
An RTP packet can have at most 15 CSRC entries.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of an RTP header with @csrc_count CSRC entries.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_calc_packet_len"
c:identifier="gst_rtp_buffer_calc_packet_len">
<doc xml:whitespace="preserve">Calculate the total length of an RTP packet with a payload size of @payload_len,
a padding of @pad_len and a @csrc_count CSRC entries.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The total length of an RTP header with given parameters.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="payload_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of the payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pad_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the amount of padding</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_calc_payload_len"
c:identifier="gst_rtp_buffer_calc_payload_len">
<doc xml:whitespace="preserve">Calculate the length of the payload of an RTP packet with size @packet_len,
a padding of @pad_len and a @csrc_count CSRC entries.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of the payload of an RTP packet with given parameters.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="packet_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of the total RTP packet</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pad_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the amount of padding</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_compare_seqnum"
c:identifier="gst_rtp_buffer_compare_seqnum"
version="0.10.15">
<doc xml:whitespace="preserve">Compare two sequence numbers, taking care of wraparounds. This function
returns the difference between @seqnum1 and @seqnum2.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a negative value if @seqnum1 is bigger than @seqnum2, 0 if they are equal or a positive value if @seqnum1 is smaller than @segnum2.</doc>
<type name="gint" c:type="gint"/>
</return-value>
<parameters>
<parameter name="seqnum1" transfer-ownership="none">
<doc xml:whitespace="preserve">a sequence number</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
<parameter name="seqnum2" transfer-ownership="none">
<doc xml:whitespace="preserve">a sequence number</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_default_clock_rate"
c:identifier="gst_rtp_buffer_default_clock_rate"
version="0.10.13">
<doc xml:whitespace="preserve">Get the default clock-rate for the static payload type @payload_type.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the default clock rate or -1 if the payload type is not static or the clock-rate is undefined.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="payload_type" transfer-ownership="none">
<doc xml:whitespace="preserve">the static payload type</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_ext_timestamp"
c:identifier="gst_rtp_buffer_ext_timestamp"
version="0.10.15">
<doc xml:whitespace="preserve">Update the @exttimestamp field with @timestamp. For the first call of the
method, @exttimestamp should point to a location with a value of -1.
This function makes sure that the returned value is a constantly increasing
value even in the case where there is a timestamp wraparound.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The extended timestamp of @timestamp.</doc>
<type name="guint64" c:type="guint64"/>
</return-value>
<parameters>
<parameter name="exttimestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">a previous extended timestamp</doc>
<type name="guint64" c:type="guint64*"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">a new timestamp</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_csrc"
c:identifier="gst_rtp_buffer_get_csrc">
<doc xml:whitespace="preserve">Get the CSRC at index @idx in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the CSRC at index @idx in host order.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="idx" transfer-ownership="none">
<doc xml:whitespace="preserve">the index of the CSRC to get</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_csrc_count"
c:identifier="gst_rtp_buffer_get_csrc_count">
<doc xml:whitespace="preserve">Get the CSRC count of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the CSRC count of @buffer.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_extension"
c:identifier="gst_rtp_buffer_get_extension">
<doc xml:whitespace="preserve">Check if the extension bit is set on the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer has the extension bit set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_extension_data"
c:identifier="gst_rtp_buffer_get_extension_data"
version="0.10.15">
<doc xml:whitespace="preserve">Get the extension data. @bits will contain the extension 16 bits of custom
data. @data will point to the data in the extension and @wordlen will contain
the length of @data in 32 bits words.
If @buffer did not contain an extension, this function will return %FALSE
with @bits, @data and @wordlen unchanged.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer had the extension bit set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="bits" transfer-ownership="none">
<doc xml:whitespace="preserve">location for result bits</doc>
<type name="guint16" c:type="guint16*"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer*"/>
</parameter>
<parameter name="wordlen" transfer-ownership="none">
<doc xml:whitespace="preserve">location for length of @data in 32 bits words</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_extension_onebyte_header"
c:identifier="gst_rtp_buffer_get_extension_onebyte_header"
version="0.10.31">
<doc xml:whitespace="preserve">Parses RFC 5285 style header extensions with a one byte header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_extension_twobytes_header"
c:identifier="gst_rtp_buffer_get_extension_twobytes_header"
version="0.10.31">
<doc xml:whitespace="preserve">Parses RFC 5285 style header extensions with a two bytes header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="appbits" transfer-ownership="none">
<doc xml:whitespace="preserve">Application specific bits</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_header_len"
c:identifier="gst_rtp_buffer_get_header_len">
<doc xml:whitespace="preserve">Return the total length of the header in @buffer. This include the length of
the fixed header, the CSRC list and the extension header.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The total length of the header in @buffer.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_marker"
c:identifier="gst_rtp_buffer_get_marker">
<doc xml:whitespace="preserve">Check if the marker bit is set on the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer has the marker bit set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_packet_len"
c:identifier="gst_rtp_buffer_get_packet_len">
<doc xml:whitespace="preserve">Return the total length of the packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The total length of the packet in @buffer.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_padding"
c:identifier="gst_rtp_buffer_get_padding">
<doc xml:whitespace="preserve">Check if the padding bit is set on the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer has the padding bit set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_payload"
c:identifier="gst_rtp_buffer_get_payload"
introspectable="0">
<doc xml:whitespace="preserve">Get a pointer to the payload data in @buffer. This pointer is valid as long
as a reference to @buffer is held.</doc>
<return-value>
<doc xml:whitespace="preserve">A pointer to the payload data in @buffer.</doc>
<type name="gpointer" c:type="gpointer"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_payload_buffer"
c:identifier="gst_rtp_buffer_get_payload_buffer"
introspectable="0">
<doc xml:whitespace="preserve">Create a buffer of the payload of the RTP packet in @buffer. This function
will internally create a subbuffer of @buffer so that a memcpy can be
avoided.</doc>
<return-value>
<doc xml:whitespace="preserve">A new buffer with the data of the payload.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_payload_len"
c:identifier="gst_rtp_buffer_get_payload_len">
<doc xml:whitespace="preserve">Get the length of the payload of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of the payload in @buffer.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_payload_subbuffer"
c:identifier="gst_rtp_buffer_get_payload_subbuffer"
version="0.10.10"
introspectable="0">
<doc xml:whitespace="preserve">Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes
are skipped in the payload and the subbuffer will be of size @len.
If @len is -1 the total payload starting from @offset if subbuffered.</doc>
<return-value>
<doc xml:whitespace="preserve">A new buffer with the specified data of the payload.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="offset" transfer-ownership="none">
<doc xml:whitespace="preserve">the offset in the payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length in the payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_payload_type"
c:identifier="gst_rtp_buffer_get_payload_type">
<doc xml:whitespace="preserve">Get the payload type of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The payload type.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_seq" c:identifier="gst_rtp_buffer_get_seq">
<doc xml:whitespace="preserve">Get the sequence number of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The sequence number in host order.</doc>
<type name="guint16" c:type="guint16"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_ssrc"
c:identifier="gst_rtp_buffer_get_ssrc">
<doc xml:whitespace="preserve">Get the SSRC of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC of @buffer in host order.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_timestamp"
c:identifier="gst_rtp_buffer_get_timestamp">
<doc xml:whitespace="preserve">Get the timestamp of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The timestamp in host order.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_version"
c:identifier="gst_rtp_buffer_get_version">
<doc xml:whitespace="preserve">Get the version number of the RTP packet in @buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The version of @buffer.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_add_extension_onebyte_header"
c:identifier="gst_rtp_buffer_list_add_extension_onebyte_header"
version="0.10.31">
<doc xml:whitespace="preserve">Adds a RFC 5285 header extension with a one byte header to the end of the
RTP header. If there is already a RFC 5285 header extension with a one byte
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
a two bytes header as described in RFC 5285. In that case, use
gst_rtp_buffer_list_add_extension_twobytes_header()
This function will not modify the data section of the RTP buffer, only
the header.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if header extension could be added</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="it" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBufferListIterator pointing right after the #GstBuffer where the header extension should be added</doc>
<type name="Gst.BufferListIterator" c:type="GstBufferListIterator*"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_add_extension_twobytes_header"
c:identifier="gst_rtp_buffer_list_add_extension_twobytes_header"
version="0.10.31">
<doc xml:whitespace="preserve">Adds a RFC 5285 header extension with a two bytes header to the end of the
RTP header. If there is already a RFC 5285 header extension with a two bytes
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
a one byte header as described in RFC 5285. In that case, use
gst_rtp_buffer_add_extension_onebyte_header()
This function will not modify the data section of the RTP buffer, only
the header.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">%TRUE if header extension could be added</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="it" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBufferListIterator pointing right after the #GstBuffer where the header extension should be added</doc>
<type name="Gst.BufferListIterator" c:type="GstBufferListIterator*"/>
</parameter>
<parameter name="appbits" transfer-ownership="none">
<doc xml:whitespace="preserve">Application specific bits</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_from_buffer"
c:identifier="gst_rtp_buffer_list_from_buffer"
introspectable="0">
<doc xml:whitespace="preserve">Splits a #GstBuffer into a #GstBufferList containing separate
buffers for the header and data sections.</doc>
<return-value>
<doc xml:whitespace="preserve">a #GstBufferList</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstBuffer containing a RTP packet</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_extension_onebyte_header"
c:identifier="gst_rtp_buffer_list_get_extension_onebyte_header"
version="0.10.31">
<doc xml:whitespace="preserve">Parses RFC 5285 style header extensions with a one byte header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="bufferlist" transfer-ownership="none">
<doc xml:whitespace="preserve">the bufferlist</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="group_idx" transfer-ownership="none">
<doc xml:whitespace="preserve">The index of the group in the #GstBufferList</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_extension_twobytes_header"
c:identifier="gst_rtp_buffer_list_get_extension_twobytes_header"
version="0.10.31">
<doc xml:whitespace="preserve">Parses RFC 5285 style header extensions with a two bytes header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="bufferlist" transfer-ownership="none">
<doc xml:whitespace="preserve">the bufferlist</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="group_idx" transfer-ownership="none">
<doc xml:whitespace="preserve">The index of the group in the #GstBufferList</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="appbits" transfer-ownership="none">
<doc xml:whitespace="preserve">Application specific bits</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:whitespace="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:whitespace="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">location for data</doc>
<type name="gpointer" c:type="gpointer*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:whitespace="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_payload_len"
c:identifier="gst_rtp_buffer_list_get_payload_len"
version="0.10.24">
<doc xml:whitespace="preserve">Get the length of the payload of the RTP packet in @list.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The length of the payload in @list.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_payload_type"
c:identifier="gst_rtp_buffer_list_get_payload_type"
version="0.10.24">
<doc xml:whitespace="preserve">Get the payload type of the first RTP packet in @list.
All packets in @list should have the same payload type.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The payload type.</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_seq"
c:identifier="gst_rtp_buffer_list_get_seq"
version="0.10.24">
<doc xml:whitespace="preserve">Get the sequence number of the first RTP packet in @list.
All packets within @list have the same sequence number.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The seq number</doc>
<type name="guint16" c:type="guint16"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_ssrc"
c:identifier="gst_rtp_buffer_list_get_ssrc"
version="0.10.24">
<doc xml:whitespace="preserve">Get the SSRC of the first RTP packet in @list.
All RTP packets within @list have the same SSRC.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">the SSRC of @list in host order.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_get_timestamp"
c:identifier="gst_rtp_buffer_list_get_timestamp"
version="0.10.24">
<doc xml:whitespace="preserve">Get the timestamp of the first RTP packet in @list.
All packets within @list have the same timestamp.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The timestamp in host order.</doc>
<type name="guint32" c:type="guint32"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_set_payload_type"
c:identifier="gst_rtp_buffer_list_set_payload_type"
version="0.10.24">
<doc xml:whitespace="preserve">Set the payload type of each RTP packet in @list to @payload_type.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="payload_type" transfer-ownership="none">
<doc xml:whitespace="preserve">the new type</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_set_seq"
c:identifier="gst_rtp_buffer_list_set_seq"
version="0.10.24">
<doc xml:whitespace="preserve">Set the sequence number of each RTP packet in @list to @seq.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">The seq number of the last packet in the list + 1.</doc>
<type name="guint16" c:type="guint16"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="seq" transfer-ownership="none">
<doc xml:whitespace="preserve">the new sequence number</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_set_ssrc"
c:identifier="gst_rtp_buffer_list_set_ssrc"
version="0.10.24">
<doc xml:whitespace="preserve">Set the SSRC on each RTP packet in @list to @ssrc.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the new SSRC</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_set_timestamp"
c:identifier="gst_rtp_buffer_list_set_timestamp"
version="0.10.24">
<doc xml:whitespace="preserve">Set the timestamp of each RTP packet in @list to @timestamp.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">the new timestamp</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_list_validate"
c:identifier="gst_rtp_buffer_list_validate"
version="0.10.24">
<doc xml:whitespace="preserve">Check if all RTP packets in the @list are valid using gst_rtp_buffer_validate_data().
Use this function to validate an list before using the other functions in
this module.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @list consists only of valid RTP packets.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="list" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer list to validate</doc>
<type name="Gst.BufferList" c:type="GstBufferList*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_new_allocate"
c:identifier="gst_rtp_buffer_new_allocate"
introspectable="0">
<doc xml:whitespace="preserve">Allocate a new #GstBuffer with enough data to hold an RTP packet with
@csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len.
All other RTP header fields will be set to 0/FALSE.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer that can hold an RTP packet with given parameters.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="payload_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of the payload</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pad_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the amount of padding</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_new_allocate_len"
c:identifier="gst_rtp_buffer_new_allocate_len"
introspectable="0">
<doc xml:whitespace="preserve">Create a new #GstBuffer that can hold an RTP packet that is exactly
@packet_len long. The length of the payload depends on @pad_len and
@csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len().
All RTP header fields will be set to 0/FALSE.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer that can hold an RTP packet of @packet_len.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="packet_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the total length of the packet</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pad_len" transfer-ownership="none">
<doc xml:whitespace="preserve">the amount of padding</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc_count" transfer-ownership="none">
<doc xml:whitespace="preserve">the number of CSRC entries</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_new_copy_data"
c:identifier="gst_rtp_buffer_new_copy_data"
introspectable="0">
<doc xml:whitespace="preserve">Create a new buffer and set the data to a copy of @len
bytes of @data and the size to @len. The data will be freed when the buffer
is freed.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer with a copy of @data and of size @len.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">data for the new buffer</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of data</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_new_take_data"
c:identifier="gst_rtp_buffer_new_take_data"
introspectable="0">
<doc xml:whitespace="preserve">Create a new buffer and set the data and size of the buffer to @data and @len
respectively. @data will be freed when the buffer is unreffed, so this
function transfers ownership of @data to the new buffer.</doc>
<return-value>
<doc xml:whitespace="preserve">A newly allocated buffer with @data and of size @len.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">data for the new buffer</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of data</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_pad_to" c:identifier="gst_rtp_buffer_pad_to">
<doc xml:whitespace="preserve">Set the amount of padding in the RTP packet in @buffer to
@len. If @len is 0, the padding is removed.
NOTE: This function does not work correctly.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the new amount of padding</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_csrc"
c:identifier="gst_rtp_buffer_set_csrc">
<doc xml:whitespace="preserve">Modify the CSRC at index @idx in @buffer to @csrc.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="idx" transfer-ownership="none">
<doc xml:whitespace="preserve">the CSRC index to set</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="csrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the CSRC in host order to set at @idx</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_extension"
c:identifier="gst_rtp_buffer_set_extension">
<doc xml:whitespace="preserve">Set the extension bit on the RTP packet in @buffer to @extension.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="extension" transfer-ownership="none">
<doc xml:whitespace="preserve">the new extension</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_extension_data"
c:identifier="gst_rtp_buffer_set_extension_data"
version="0.10.18">
<doc xml:whitespace="preserve">Set the extension bit of the rtp buffer and fill in the @bits and @length of the
extension header. It will refuse to set the extension data if the buffer is not
large enough.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">True if done.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="bits" transfer-ownership="none">
<doc xml:whitespace="preserve">the bits specific for the extension</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
<parameter name="length" transfer-ownership="none">
<doc xml:whitespace="preserve">the length that counts the number of 32-bit words in the extension, excluding the extension header ( therefore zero is a valid length)</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_marker"
c:identifier="gst_rtp_buffer_set_marker">
<doc xml:whitespace="preserve">Set the marker bit on the RTP packet in @buffer to @marker.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="marker" transfer-ownership="none">
<doc xml:whitespace="preserve">the new marker</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_packet_len"
c:identifier="gst_rtp_buffer_set_packet_len">
<doc xml:whitespace="preserve">Set the total @buffer size to @len. The data in the buffer will be made
larger if needed. Any padding will be removed from the packet.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the new packet length</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_padding"
c:identifier="gst_rtp_buffer_set_padding">
<doc xml:whitespace="preserve">Set the padding bit on the RTP packet in @buffer to @padding.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="padding" transfer-ownership="none">
<doc xml:whitespace="preserve">the new padding</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_payload_type"
c:identifier="gst_rtp_buffer_set_payload_type">
<doc xml:whitespace="preserve">Set the payload type of the RTP packet in @buffer to @payload_type.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="payload_type" transfer-ownership="none">
<doc xml:whitespace="preserve">the new type</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_seq" c:identifier="gst_rtp_buffer_set_seq">
<doc xml:whitespace="preserve">Set the sequence number of the RTP packet in @buffer to @seq.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="seq" transfer-ownership="none">
<doc xml:whitespace="preserve">the new sequence number</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_ssrc"
c:identifier="gst_rtp_buffer_set_ssrc">
<doc xml:whitespace="preserve">Set the SSRC on the RTP packet in @buffer to @ssrc.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="ssrc" transfer-ownership="none">
<doc xml:whitespace="preserve">the new SSRC</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_timestamp"
c:identifier="gst_rtp_buffer_set_timestamp">
<doc xml:whitespace="preserve">Set the timestamp of the RTP packet in @buffer to @timestamp.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="timestamp" transfer-ownership="none">
<doc xml:whitespace="preserve">the new timestamp</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_set_version"
c:identifier="gst_rtp_buffer_set_version">
<doc xml:whitespace="preserve">Set the version of the RTP packet in @buffer to @version.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="version" transfer-ownership="none">
<doc xml:whitespace="preserve">the new version</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_validate"
c:identifier="gst_rtp_buffer_validate">
<doc xml:whitespace="preserve">Check if the data pointed to by @buffer is a valid RTP packet using
gst_rtp_buffer_validate_data().
Use this function to validate a packet before using the other functions in
this module.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if @buffer is a valid RTP packet.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:whitespace="preserve">the buffer to validate</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_validate_data"
c:identifier="gst_rtp_buffer_validate_data">
<doc xml:whitespace="preserve">Check if the @data and @size point to the data of a valid RTP packet.
This function checks the length, version and padding of the packet data.
Use this function to validate a packet before using the other functions in
this module.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">TRUE if the data points to a valid RTP packet.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="data" transfer-ownership="none">
<doc xml:whitespace="preserve">the data to validate</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="len" transfer-ownership="none">
<doc xml:whitespace="preserve">the length of @data to validate</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</function>
<function name="rtp_payload_info_for_name"
c:identifier="gst_rtp_payload_info_for_name"
moved-to="RTPPayloadInfo.for_name">
<doc xml:whitespace="preserve">Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is
mostly used to get the default clock-rate and bandwidth for dynamic payload
types specified with @media and @encoding name.
The search for @encoding_name will be performed in a case insensitve way.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
<type name="RTPPayloadInfo" c:type="const GstRTPPayloadInfo*"/>
</return-value>
<parameters>
<parameter name="media" transfer-ownership="none">
<doc xml:whitespace="preserve">the media to find</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="encoding_name" transfer-ownership="none">
<doc xml:whitespace="preserve">the encoding name to find</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</function>
<function name="rtp_payload_info_for_pt"
c:identifier="gst_rtp_payload_info_for_pt"
moved-to="RTPPayloadInfo.for_pt">
<doc xml:whitespace="preserve">Get the #GstRTPPayloadInfo for @payload_type. This function is
mostly used to get the default clock-rate and bandwidth for static payload
types specified with @payload_type.</doc>
<return-value transfer-ownership="none">
<doc xml:whitespace="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
<type name="RTPPayloadInfo" c:type="const GstRTPPayloadInfo*"/>
</return-value>
<parameters>
<parameter name="payload_type" transfer-ownership="none">
<doc xml:whitespace="preserve">the payload_type to find</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</function>
</namespace>
</repository>