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Update gir-files to 1.16.0

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Sebastian Dröge 2019-04-23 12:05:44 +03:00
parent 2e4b6be986
commit 4e2c2d5774
16 changed files with 6468 additions and 3725 deletions
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904
gir-files/Gst-1.0.gir
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6
gir-files/GstApp-1.0.gir
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@ -1119,6 +1119,9 @@ buffers that the appsrc element will push to its source pad. Any
previous caps that were set on appsrc will be replaced by the caps
associated with the sample if not equal.
This function does not take ownership of the
sample so the sample needs to be unreffed after calling this function.
When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
@ -1351,6 +1354,9 @@ buffers that the appsrc element will push to its source pad. Any
previous caps that were set on appsrc will be replaced by the caps
associated with the sample if not equal.
This function does not take ownership of the
sample so the sample needs to be unreffed after calling this function.
When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">

562
gir-files/GstAudio-1.0.gir
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@ -250,6 +250,7 @@ transition band for the kaiser window. 0.087 is the default.</doc>
<class name="AudioAggregator"
c:symbol-prefix="audio_aggregator"
c:type="GstAudioAggregator"
version="1.14"
parent="GstBase.Aggregator"
abstract="1"
glib:type-name="GstAudioAggregator"
@ -265,7 +266,7 @@ on its sink pads, based on the format expected downstream: in order
to enable that behaviour, the GType of the sink pads must either be
a (subclass of) #GstAudioAggregatorConvertPad to use the default
#GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
implementing #GstAudioAggregatorPad.convert_buffer.
implementing #GstAudioAggregatorPadClass.convert_buffer.
To allow for the output caps to change, the mechanism is the same as
above, with the GType of the source pad.
@ -357,7 +358,6 @@ downstream specifies a range or a set of acceptable rates).</doc>
<type name="guint64" c:type="guint64"/>
</property>
<field name="parent">
<doc xml:space="preserve">The parent #GstAggregator</doc>
<type name="GstBase.Aggregator" c:type="GstAggregator"/>
</field>
<field name="current_caps">
@ -376,7 +376,8 @@ downstream specifies a range or a set of acceptable rates).</doc>
</class>
<record name="AudioAggregatorClass"
c:type="GstAudioAggregatorClass"
glib:is-gtype-struct-for="AudioAggregator">
glib:is-gtype-struct-for="AudioAggregator"
version="1.14">
<field name="parent_class">
<type name="GstBase.AggregatorClass" c:type="GstAggregatorClass"/>
</field>
@ -434,6 +435,7 @@ downstream specifies a range or a set of acceptable rates).</doc>
<class name="AudioAggregatorConvertPad"
c:symbol-prefix="audio_aggregator_convert_pad"
c:type="GstAudioAggregatorConvertPad"
version="1.14"
parent="AudioAggregatorPad"
glib:type-name="GstAudioAggregatorConvertPad"
glib:get-type="gst_audio_aggregator_convert_pad_get_type"
@ -444,8 +446,7 @@ See #GstAudioAggregator for more details.</doc>
<property name="converter-config" writable="1" transfer-ownership="none">
<type name="Gst.Structure"/>
</property>
<field name="parent">
<doc xml:space="preserve">The parent #GstAudioAggregatorPad</doc>
<field name="parent" readable="0" private="1">
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad"/>
</field>
<field name="priv" readable="0" private="1">
@ -460,7 +461,8 @@ See #GstAudioAggregator for more details.</doc>
</class>
<record name="AudioAggregatorConvertPadClass"
c:type="GstAudioAggregatorConvertPadClass"
glib:is-gtype-struct-for="AudioAggregatorConvertPad">
glib:is-gtype-struct-for="AudioAggregatorConvertPad"
version="1.14">
<field name="parent_class">
<type name="AudioAggregatorPadClass"
c:type="GstAudioAggregatorPadClass"/>
@ -478,6 +480,7 @@ See #GstAudioAggregator for more details.</doc>
<class name="AudioAggregatorPad"
c:symbol-prefix="audio_aggregator_pad"
c:type="GstAudioAggregatorPad"
version="1.14"
parent="GstBase.AggregatorPad"
glib:type-name="GstAudioAggregatorPad"
glib:get-type="gst_audio_aggregator_pad_get_type"
@ -513,7 +516,6 @@ See #GstAudioAggregator for more details.</doc>
</parameters>
</virtual-method>
<field name="parent">
<doc xml:space="preserve">The parent #GstAggregatorPad</doc>
<type name="GstBase.AggregatorPad" c:type="GstAggregatorPad"/>
</field>
<field name="info">
@ -532,7 +534,8 @@ See #GstAudioAggregator for more details.</doc>
</class>
<record name="AudioAggregatorPadClass"
c:type="GstAudioAggregatorPadClass"
glib:is-gtype-struct-for="AudioAggregatorPad">
glib:is-gtype-struct-for="AudioAggregatorPad"
version="1.14">
<field name="parent_class">
<type name="GstBase.AggregatorPadClass"
c:type="GstAggregatorPadClass"/>
@ -1326,6 +1329,231 @@ drifts too much.</doc>
<doc xml:space="preserve">No adjustment is done.</doc>
</member>
</enumeration>
<record name="AudioBuffer" c:type="GstAudioBuffer" version="1.16">
<doc xml:space="preserve">A structure containing the result of an audio buffer map operation,
which is executed with gst_audio_buffer_map(). For non-interleaved (planar)
buffers, the beginning of each channel in the buffer has its own pointer in
the @planes array. For interleaved buffers, the @planes array only contains
one item, which is the pointer to the beginning of the buffer, and @n_planes
equals 1.
The different channels in @planes are always in the GStreamer channel order.</doc>
<field name="info" writable="1">
<doc xml:space="preserve">a #GstAudioInfo describing the audio properties of this buffer</doc>
<type name="AudioInfo" c:type="GstAudioInfo"/>
</field>
<field name="n_samples" writable="1">
<doc xml:space="preserve">the size of the buffer in samples</doc>
<type name="gsize" c:type="gsize"/>
</field>
<field name="n_planes" writable="1">
<doc xml:space="preserve">the number of planes available</doc>
<type name="gint" c:type="gint"/>
</field>
<field name="planes" writable="1">
<doc xml:space="preserve">an array of @n_planes pointers pointing to the start of each
plane in the mapped buffer</doc>
<type name="gpointer" c:type="gpointer*"/>
</field>
<field name="buffer" writable="1">
<doc xml:space="preserve">the mapped buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</field>
<field name="map_infos" readable="0" private="1">
<type name="Gst.MapInfo" c:type="GstMapInfo*"/>
</field>
<field name="priv_planes_arr" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="8">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<field name="priv_map_infos_arr" readable="0" private="1">
<array zero-terminated="0" c:type="GstMapInfo" fixed-size="8">
<type name="Gst.MapInfo" c:type="GstMapInfo"/>
</array>
</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>
<method name="map" c:identifier="gst_audio_buffer_map" version="1.16">
<doc xml:space="preserve">Maps an audio @gstbuffer so that it can be read or written and stores the
result of the map operation in @buffer.
This is especially useful when the @gstbuffer is in non-interleaved (planar)
layout, in which case this function will use the information in the
@gstbuffer's attached #GstAudioMeta in order to map each channel in a
separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached
on the @gstbuffer, then it must be in interleaved layout.
If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked
against @info. Normally, they should be equal, but in case they are not,
a g_critical will be printed and the #GstAudioInfo from the meta will be
used.
In non-interleaved buffers, it is possible to have each channel on a separate
#GstMemory. In this case, each memory will be mapped separately to avoid
copying their contents in a larger memory area. Do note though that it is
not supported to have a single channel spanning over two or more different
#GstMemory objects. Although the map operation will likely succeed in this
case, it will be highly sub-optimal and it is recommended to merge all the
memories in the buffer before calling this function.
Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid
as long as it's mapped.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the map operation succeeded or %FALSE on failure</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="buffer" transfer-ownership="none">
<doc xml:space="preserve">pointer to a #GstAudioBuffer</doc>
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
</instance-parameter>
<parameter name="info" transfer-ownership="none">
<doc xml:space="preserve">the audio properties of the buffer</doc>
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
</parameter>
<parameter name="gstbuffer" transfer-ownership="none">
<doc xml:space="preserve">the #GstBuffer to be mapped</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<doc xml:space="preserve">the access mode for the memory</doc>
<type name="Gst.MapFlags" c:type="GstMapFlags"/>
</parameter>
</parameters>
</method>
<method name="unmap"
c:identifier="gst_audio_buffer_unmap"
version="1.16">
<doc xml:space="preserve">Unmaps an audio buffer that was previously mapped with
gst_audio_buffer_map().</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="buffer" transfer-ownership="none">
<doc xml:space="preserve">the #GstAudioBuffer to unmap</doc>
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
</instance-parameter>
</parameters>
</method>
<function name="clip" c:identifier="gst_audio_buffer_clip">
<doc xml:space="preserve">Clip the buffer to the given %GstSegment.
After calling this function the caller does not own a reference to
@buffer anymore.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">%NULL if the buffer is completely outside the configured segment,
otherwise the clipped buffer is returned.
If the buffer has no timestamp, it is assumed to be inside the segment and
is not clipped</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="full">
<doc xml:space="preserve">The buffer to clip.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="segment" transfer-ownership="none">
<doc xml:space="preserve">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
the buffer should be clipped.</doc>
<type name="Gst.Segment" c:type="const GstSegment*"/>
</parameter>
<parameter name="rate" transfer-ownership="none">
<doc xml:space="preserve">sample rate.</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="bpf" transfer-ownership="none">
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
number of channels.</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</function>
<function name="reorder_channels"
c:identifier="gst_audio_buffer_reorder_channels">
<doc xml:space="preserve">Reorders @buffer from the channel positions @from to the channel
positions @to. @from and @to must contain the same number of
positions and the same positions, only in a different order.
@buffer must be writable.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the reordering was possible.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:space="preserve">The buffer to reorder.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="format" transfer-ownership="none">
<doc xml:space="preserve">The %GstAudioFormat of the buffer.</doc>
<type name="AudioFormat" c:type="GstAudioFormat"/>
</parameter>
<parameter name="channels" transfer-ownership="none">
<doc xml:space="preserve">The number of channels.</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="from" transfer-ownership="none">
<doc xml:space="preserve">The channel positions in the buffer.</doc>
<array length="2"
zero-terminated="0"
c:type="const GstAudioChannelPosition*">
<type name="AudioChannelPosition"
c:type="GstAudioChannelPosition"/>
</array>
</parameter>
<parameter name="to" transfer-ownership="none">
<doc xml:space="preserve">The channel positions to convert to.</doc>
<array length="2"
zero-terminated="0"
c:type="const GstAudioChannelPosition*">
<type name="AudioChannelPosition"
c:type="GstAudioChannelPosition"/>
</array>
</parameter>
</parameters>
</function>
<function name="truncate"
c:identifier="gst_audio_buffer_truncate"
version="1.16">
<doc xml:space="preserve">Truncate the buffer to finally have @samples number of samples, removing
the necessary amount of samples from the end and @trim number of samples
from the beginning.
After calling this function the caller does not own a reference to
@buffer anymore.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the truncated buffer or %NULL if the arguments
were invalid</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="full">
<doc xml:space="preserve">The buffer to truncate.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="bpf" transfer-ownership="none">
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
number of channels.</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="trim" transfer-ownership="none">
<doc xml:space="preserve">the number of samples to remove from the beginning of the buffer</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
<parameter name="samples" transfer-ownership="none">
<doc xml:space="preserve">the final number of samples that should exist in this buffer or -1
to use all the remaining samples if you are only removing samples from the
beginning.</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
</parameters>
</function>
</record>
<class name="AudioCdSrc"
c:symbol-prefix="audio_cd_src"
c:type="GstAudioCdSrc"
@ -2205,11 +2433,19 @@ be used.</doc>
glib:type-name="GstAudioConverter"
glib:get-type="gst_audio_converter_get_type"
c:symbol-prefix="audio_converter">
<doc xml:space="preserve">This object is used to convert audio samples from one format to another.
The object can perform conversion of:
* audio format with optional dithering and noise shaping
* audio samplerate
* audio channels and channel layout</doc>
<constructor name="new" c:identifier="gst_audio_converter_new">
<doc xml:space="preserve">Create a new #GstAudioConverter that is able to convert between @in and @out
audio formats.
@config contains extra configuration options, see #GST_VIDEO_CONVERTER_OPT_*
@config contains extra configuration options, see #GST_AUDIO_CONVERTER_OPT_*
parameters for details about the options and values.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a #GstAudioConverter or %NULL if conversion is not possible.</doc>
@ -2249,6 +2485,7 @@ gst_audio_converter_get_out_frames().</doc>
</return-value>
<parameters>
<instance-parameter name="convert" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioConverter</doc>
<type name="AudioConverter" c:type="GstAudioConverter*"/>
</instance-parameter>
<parameter name="flags" transfer-ownership="none">
@ -2383,6 +2620,21 @@ frames are given to @convert.</doc>
</parameter>
</parameters>
</method>
<method name="is_passthrough"
c:identifier="gst_audio_converter_is_passthrough"
version="1.16">
<doc xml:space="preserve">Returns whether the audio converter will operate in passthrough mode.
The return value would be typically input to gst_base_transform_set_passthrough()</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE when no conversion will actually occur.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="convert" transfer-ownership="none">
<type name="AudioConverter" c:type="GstAudioConverter*"/>
</instance-parameter>
</parameters>
</method>
<method name="reset" c:identifier="gst_audio_converter_reset">
<doc xml:space="preserve">Reset @convert to the state it was when it was first created, clearing
any history it might currently have.</doc>
@ -2608,7 +2860,10 @@ occurs (which would happen always if the tolerance mechanism is disabled).
In non-live pipelines, baseclass can also (configurably) arrange for
output buffer aggregation which may help to redue large(r) numbers of
small(er) buffers being pushed and processed downstream.
small(er) buffers being pushed and processed downstream. Note that this
feature is only available if the buffer layout is interleaved. For planar
buffers, the decoder implementation is fully responsible for the output
buffer size.
On the other hand, it should be noted that baseclass only provides limited
seeking support (upon explicit subclass request), as full-fledged support
@ -2893,7 +3148,7 @@ are discarded and considered to have produced no output
Otherwise, source pad caps must be set when it is called with valid
data in @buf.
Note that a frame received in gst_audio_decoder_handle_frame() may be
Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
invalidated by a call to this function.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
@ -2914,6 +3169,37 @@ invalidated by a call to this function.</doc>
</parameter>
</parameters>
</method>
<method name="finish_subframe"
c:identifier="gst_audio_decoder_finish_subframe"
version="1.16">
<doc xml:space="preserve">Collects decoded data and pushes it downstream. This function may be called
multiple times for a given input frame.
@buf may be NULL in which case it is assumed that the current input frame is
finished. This is equivalent to calling gst_audio_decoder_finish_subframe()
with a NULL buffer and frames=1 after having pushed out all decoded audio
subframes using this function.
When called with valid data in @buf the source pad caps must have been set
already.
Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
invalidated by a call to this function.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="dec" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioDecoder</doc>
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
</instance-parameter>
<parameter name="buf" transfer-ownership="none">
<doc xml:space="preserve">decoded data</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<method name="get_allocator"
c:identifier="gst_audio_decoder_get_allocator">
<doc xml:space="preserve">Lets #GstAudioDecoder sub-classes to know the memory @allocator
@ -2945,7 +3231,7 @@ used</doc>
optional="1"
allow-none="1">
<doc xml:space="preserve">the
#GstAllocatorParams of @allocator</doc>
#GstAllocationParams of @allocator</doc>
<type name="Gst.AllocationParams" c:type="GstAllocationParams*"/>
</parameter>
</parameters>
@ -3375,6 +3661,28 @@ MT safe.</doc>
</parameter>
</parameters>
</method>
<method name="set_output_caps"
c:identifier="gst_audio_decoder_set_output_caps"
version="1.16">
<doc xml:space="preserve">Configure output caps on the srcpad of @dec. Similar to
gst_audio_decoder_set_output_format(), but allows subclasses to specify
output caps that can't be expressed via #GstAudioInfo e.g. caps that have
caps features.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE on success.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="dec" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioDecoder</doc>
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
</instance-parameter>
<parameter name="caps" transfer-ownership="none">
<doc xml:space="preserve">(fixed) #GstCaps</doc>
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
</method>
<method name="set_output_format"
c:identifier="gst_audio_decoder_set_output_format">
<doc xml:space="preserve">Configure output info on the srcpad of @dec.</doc>
@ -4209,7 +4517,7 @@ If @samples &lt; 0, then best estimate is all samples provided to encoder
are considered discarded, e.g. as a result of discontinuous transmission,
and a discontinuity is marked.
Note that samples received in gst_audio_encoder_handle_frame()
Note that samples received in #GstAudioEncoderClass.handle_frame()
may be invalidated by a call to this function.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
@ -4261,7 +4569,7 @@ used</doc>
optional="1"
allow-none="1">
<doc xml:space="preserve">the
#GstAllocatorParams of @allocator</doc>
#GstAllocationParams of @allocator</doc>
<type name="Gst.AllocationParams" c:type="GstAllocationParams*"/>
</parameter>
</parameters>
@ -4581,7 +4889,7 @@ MT safe.</doc>
Requires @frame_samples_min and @frame_samples_max to be the equal.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.</doc>
#GstAudioEncoderClass.set_format() is called.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -4605,7 +4913,7 @@ If an exact number of samples is required, gst_audio_encoder_set_frame_samples_m
must be called with the same number.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.</doc>
#GstAudioEncoderClass.set_format() is called.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -4629,7 +4937,7 @@ If an exact number of samples is required, gst_audio_encoder_set_frame_samples_m
must be called with the same number.
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.</doc>
#GstAudioEncoderClass.set_format() is called.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -4724,7 +5032,7 @@ MT safe.</doc>
<doc xml:space="preserve">Sets encoder lookahead (in units of input rate samples)
Note: This value will be reset to 0 every time before
GstAudioEncoder::set_format() is called.</doc>
#GstAudioEncoderClass.set_format() is called.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -5710,6 +6018,7 @@ and will be packed into @data.</doc>
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<doc xml:space="preserve">#GstAudioPackFlags</doc>
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
</parameter>
<parameter name="src" transfer-ownership="none">
@ -5745,6 +6054,7 @@ channels * size(unpack_format) bytes.</doc>
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<doc xml:space="preserve">#GstAudioPackFlags</doc>
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
</parameter>
<parameter name="dest" transfer-ownership="none">
@ -6000,6 +6310,44 @@ Note: This initializes @info first, no values are preserved.</doc>
<doc xml:space="preserve">non-interleaved audio</doc>
</member>
</enumeration>
<record name="AudioMeta" c:type="GstAudioMeta" version="1.16">
<doc xml:space="preserve">#GstAudioDownmixMeta defines an audio downmix matrix to be send along with
audio buffers. These functions in this module help to create and attach the
meta as well as extracting it.</doc>
<field name="meta" writable="1">
<doc xml:space="preserve">parent #GstMeta</doc>
<type name="Gst.Meta" c:type="GstMeta"/>
</field>
<field name="info" writable="1">
<doc xml:space="preserve">the audio properties of the buffer</doc>
<type name="AudioInfo" c:type="GstAudioInfo"/>
</field>
<field name="samples" writable="1">
<doc xml:space="preserve">the number of valid samples in the buffer</doc>
<type name="gsize" c:type="gsize"/>
</field>
<field name="offsets" writable="1">
<doc xml:space="preserve">the offsets (in bytes) where each channel plane starts in the
buffer or %NULL if the buffer has interleaved layout; if not %NULL, this
is guaranteed to be an array of @info.channels elements</doc>
<type name="gsize" c:type="gsize*"/>
</field>
<field name="priv_offsets_arr" readable="0" private="1">
<array zero-terminated="0" c:type="gsize" fixed-size="8">
<type name="gsize" c:type="gsize"/>
</array>
</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>
<function name="get_info" c:identifier="gst_audio_meta_get_info">
<return-value transfer-ownership="none">
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
</return-value>
</function>
</record>
<enumeration name="AudioNoiseShapingMethod"
glib:type-name="GstAudioNoiseShapingMethod"
glib:get-type="gst_audio_noise_shaping_method_get_type"
@ -6351,7 +6699,8 @@ When @options is %NULL, the previously configured options are reused.</doc>
<function name="new" c:identifier="gst_audio_resampler_new">
<doc xml:space="preserve">Make a new resampler.</doc>
<return-value transfer-ownership="full" skip="1">
<doc xml:space="preserve">%TRUE on success</doc>
<doc xml:space="preserve">The new #GstAudioResampler, or
%NULL on failure.</doc>
<type name="AudioResampler" c:type="GstAudioResampler*"/>
</return-value>
<parameters>
@ -6365,9 +6714,11 @@ When @options is %NULL, the previously configured options are reused.</doc>
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
</parameter>
<parameter name="format" transfer-ownership="none">
<doc xml:space="preserve">the #GstAudioFormat</doc>
<type name="AudioFormat" c:type="GstAudioFormat"/>
</parameter>
<parameter name="channels" transfer-ownership="none">
<doc xml:space="preserve">the number of channels</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="in_rate" transfer-ownership="none">
@ -7512,7 +7863,7 @@ MT safe.</doc>
<doc xml:space="preserve">size of data in the ringbuffer</doc>
<type name="gsize" c:type="gsize"/>
</field>
<field name="timestamps">
<field name="timestamps" readable="0" private="1">
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
</field>
<field name="spec">
@ -7921,19 +8272,19 @@ with a flush or stop.</doc>
value="12"
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW"
glib:nick="mpeg2-aac-raw">
<doc xml:space="preserve">samples in MPEG-2 AAC raw format (Since 1.12)</doc>
<doc xml:space="preserve">samples in MPEG-2 AAC raw format (Since: 1.12)</doc>
</member>
<member name="mpeg4_aac_raw"
value="13"
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW"
glib:nick="mpeg4-aac-raw">
<doc xml:space="preserve">samples in MPEG-4 AAC raw format (Since 1.12)</doc>
<doc xml:space="preserve">samples in MPEG-4 AAC raw format (Since: 1.12)</doc>
</member>
<member name="flac"
value="14"
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC"
glib:nick="flac">
<doc xml:space="preserve">samples in FLAC format (Since 1.12)</doc>
<doc xml:space="preserve">samples in FLAC format (Since: 1.12)</doc>
</member>
</enumeration>
<record name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec">
@ -8007,7 +8358,7 @@ with a flush or stop.</doc>
glib:nick="error">
<doc xml:space="preserve">The ringbuffer has encountered an
error after it has been started, e.g. because the device was
disconnected (Since 1.2)</doc>
disconnected (Since: 1.2)</doc>
</member>
</enumeration>
<class name="AudioSink"
@ -8493,7 +8844,7 @@ gst_audio_stream_align_process() for the details of the processing.</doc>
version="1.14">
<doc xml:space="preserve">Allocate a new #GstAudioStreamAlign with the given configuration. All
processing happens according to sample rate @rate, until
gst_audio_discont_wait_set_rate() is called with a new @rate.
gst_audio_stream_align_set_rate() is called with a new @rate.
A negative rate can be used for reverse playback.
@alignment_threshold gives the tolerance in nanoseconds after which a
@ -8552,33 +8903,46 @@ or gst_audio_stream_align_copy().</doc>
</parameters>
</method>
<method name="get_alignment_threshold"
c:identifier="gst_audio_stream_align_get_alignment_threshold">
c:identifier="gst_audio_stream_align_get_alignment_threshold"
version="1.14">
<doc xml:space="preserve">Gets the currently configured alignment threshold.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The currently configured alignment threshold</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_discont_wait"
c:identifier="gst_audio_stream_align_get_discont_wait">
c:identifier="gst_audio_stream_align_get_discont_wait"
version="1.14">
<doc xml:space="preserve">Gets the currently configured discont wait.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The currently configured discont wait</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_rate" c:identifier="gst_audio_stream_align_get_rate">
<method name="get_rate"
c:identifier="gst_audio_stream_align_get_rate"
version="1.14">
<doc xml:space="preserve">Gets the currently configured sample rate.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The currently configured sample rate</doc>
<type name="gint" c:type="gint"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
</parameters>
@ -8693,42 +9057,56 @@ of the current one.</doc>
</parameters>
</method>
<method name="set_alignment_threshold"
c:identifier="gst_audio_stream_align_set_alignment_threshold">
c:identifier="gst_audio_stream_align_set_alignment_threshold"
version="1.14">
<doc xml:space="preserve">Sets @alignment_treshold as new alignment threshold for the following processing.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
<parameter name="alignment_threshold" transfer-ownership="none">
<doc xml:space="preserve">a new alignment threshold</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="set_discont_wait"
c:identifier="gst_audio_stream_align_set_discont_wait">
c:identifier="gst_audio_stream_align_set_discont_wait"
version="1.14">
<doc xml:space="preserve">Sets @alignment_treshold as new discont wait for the following processing.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
<parameter name="discont_wait" transfer-ownership="none">
<doc xml:space="preserve">a new discont wait</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="set_rate" c:identifier="gst_audio_stream_align_set_rate">
<method name="set_rate"
c:identifier="gst_audio_stream_align_set_rate"
version="1.14">
<doc xml:space="preserve">Sets @rate as new sample rate for the following processing. If the sample
rate differs this implicitely marks the next data as discontinuous.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="align" transfer-ownership="none">
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
</instance-parameter>
<parameter name="rate" transfer-ownership="none">
<doc xml:space="preserve">a new sample rate</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
@ -8894,7 +9272,9 @@ cbrt(val) and 20 * log10 (val).</doc>
<type name="GObject.TypeInterface" c:type="GTypeInterface"/>
</field>
</record>
<function name="audio_buffer_clip" c:identifier="gst_audio_buffer_clip">
<function name="audio_buffer_clip"
c:identifier="gst_audio_buffer_clip"
moved-to="AudioBuffer.clip">
<doc xml:space="preserve">Clip the buffer to the given %GstSegment.
After calling this function the caller does not own a reference to
@ -8929,7 +9309,8 @@ number of channels.</doc>
</parameters>
</function>
<function name="audio_buffer_reorder_channels"
c:identifier="gst_audio_buffer_reorder_channels">
c:identifier="gst_audio_buffer_reorder_channels"
moved-to="AudioBuffer.reorder_channels">
<doc xml:space="preserve">Reorders @buffer from the channel positions @from to the channel
positions @to. @from and @to must contain the same number of
positions and the same positions, only in a different order.
@ -8971,6 +9352,43 @@ positions and the same positions, only in a different order.
</parameter>
</parameters>
</function>
<function name="audio_buffer_truncate"
c:identifier="gst_audio_buffer_truncate"
moved-to="AudioBuffer.truncate"
version="1.16">
<doc xml:space="preserve">Truncate the buffer to finally have @samples number of samples, removing
the necessary amount of samples from the end and @trim number of samples
from the beginning.
After calling this function the caller does not own a reference to
@buffer anymore.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the truncated buffer or %NULL if the arguments
were invalid</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="full">
<doc xml:space="preserve">The buffer to truncate.</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="bpf" transfer-ownership="none">
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
number of channels.</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="trim" transfer-ownership="none">
<doc xml:space="preserve">the number of samples to remove from the beginning of the buffer</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
<parameter name="samples" transfer-ownership="none">
<doc xml:space="preserve">the final number of samples that should exist in this buffer or -1
to use all the remaining samples if you are only removing samples from the
beginning.</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
</parameters>
</function>
<function name="audio_channel_get_fallback_mask"
c:identifier="gst_audio_channel_get_fallback_mask"
version="1.8">
@ -9138,14 +9556,13 @@ in the order required by GStreamer.</doc>
</parameters>
</function>
<function name="audio_channel_positions_to_string"
c:identifier="gst_audio_channel_positions_to_string">
c:identifier="gst_audio_channel_positions_to_string"
version="1.10">
<doc xml:space="preserve">Converts @position to a human-readable string representation for
debugging purposes.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a newly allocated string representing
@position
Since 1.10</doc>
@position</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>
@ -9455,6 +9872,19 @@ otherwise.</doc>
</parameter>
</parameters>
</function>
<function name="audio_meta_api_get_type"
c:identifier="gst_audio_meta_api_get_type">
<return-value transfer-ownership="none">
<type name="GType" c:type="GType"/>
</return-value>
</function>
<function name="audio_meta_get_info"
c:identifier="gst_audio_meta_get_info"
moved-to="AudioMeta.get_info">
<return-value transfer-ownership="none">
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
</return-value>
</function>
<function name="audio_quantize_new"
c:identifier="gst_audio_quantize_new"
moved-to="AudioQuantize.new"
@ -9502,7 +9932,9 @@ the @dither and @ns parameters.</doc>
c:identifier="gst_audio_reorder_channels">
<doc xml:space="preserve">Reorders @data from the channel positions @from to the channel
positions @to. @from and @to must contain the same number of
positions and the same positions, only in a different order.</doc>
positions and the same positions, only in a different order.
Note: this function assumes the audio data is in interleaved layout</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the reordering was possible.</doc>
<type name="gboolean" c:type="gboolean"/>
@ -9552,7 +9984,8 @@ positions and the same positions, only in a different order.</doc>
moved-to="AudioResampler.new">
<doc xml:space="preserve">Make a new resampler.</doc>
<return-value transfer-ownership="full" skip="1">
<doc xml:space="preserve">%TRUE on success</doc>
<doc xml:space="preserve">The new #GstAudioResampler, or
%NULL on failure.</doc>
<type name="AudioResampler" c:type="GstAudioResampler*"/>
</return-value>
<parameters>
@ -9565,9 +9998,11 @@ positions and the same positions, only in a different order.</doc>
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
</parameter>
<parameter name="format" transfer-ownership="none">
<doc xml:space="preserve">the #GstAudioFormat</doc>
<type name="AudioFormat" c:type="GstAudioFormat"/>
</parameter>
<parameter name="channels" transfer-ownership="none">
<doc xml:space="preserve">the number of channels</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="in_rate" transfer-ownership="none">
@ -9693,6 +10128,55 @@ of the results.</doc>
</parameter>
</parameters>
</function>
<function name="buffer_add_audio_meta"
c:identifier="gst_buffer_add_audio_meta"
version="1.16">
<doc xml:space="preserve">Allocates and attaches a #GstAudioMeta on @buffer, which must be writable
for that purpose. The fields of the #GstAudioMeta are directly populated
from the arguments of this function.
When @info-&gt;layout is %GST_AUDIO_LAYOUT_NON_INTERLEAVED and @offsets is
%NULL, the offsets are calculated with a formula that assumes the planes are
tightly packed and in sequence:
offsets[channel] = channel * @samples * sample_stride
It is not allowed for channels to overlap in memory,
i.e. for each i in [0, channels), the range
[@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap
with any other such range. This function will assert if the parameters
specified cause this restriction to be violated.
It is, obviously, also not allowed to specify parameters that would cause
out-of-bounds memory access on @buffer. This is also checked, which means
that you must add enough memory on the @buffer before adding this meta.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the #GstAudioMeta that was attached on the @buffer</doc>
<type name="AudioMeta" c:type="GstAudioMeta*"/>
</return-value>
<parameters>
<parameter name="buffer" transfer-ownership="none">
<doc xml:space="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
<parameter name="info" transfer-ownership="none">
<doc xml:space="preserve">the audio properties of the buffer</doc>
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
</parameter>
<parameter name="samples" transfer-ownership="none">
<doc xml:space="preserve">the number of valid samples in the buffer</doc>
<type name="gsize" c:type="gsize"/>
</parameter>
<parameter name="offsets"
transfer-ownership="none"
nullable="1"
allow-none="1">
<doc xml:space="preserve">the offsets (in bytes) where each channel plane starts
in the buffer or %NULL to calculate it (see below); must be %NULL also
when @info-&gt;layout is %GST_AUDIO_LAYOUT_INTERLEAVED</doc>
<type name="gsize" c:type="gsize*"/>
</parameter>
</parameters>
</function>
<function name="buffer_get_audio_downmix_meta_for_channels"
c:identifier="gst_buffer_get_audio_downmix_meta_for_channels">
<doc xml:space="preserve">Find the #GstAudioDownmixMeta on @buffer for the given destination

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@ -1713,7 +1713,7 @@ reached.
MT safe.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a @gboolean %TRUE if the waits have been registered, %FALSE if not.
(Could be that it timed out waiting or that more waits then waits was found)</doc>
(Could be that it timed out waiting or that more waits than waits was found)</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -2314,6 +2314,39 @@ MT safe.</doc>
</parameter>
</parameters>
</method>
<method name="timed_wait_for_multiple_pending_ids" c:identifier="gst_test_clock_timed_wait_for_multiple_pending_ids" version="1.16">
<doc xml:space="preserve">Blocks until at least @count clock notifications have been requested from
@test_clock, or the timeout expires.
MT safe.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a @gboolean %TRUE if the waits have been registered, %FALSE if not.
(Could be that it timed out waiting or that more waits than waits was found)</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="test_clock" transfer-ownership="none">
<doc xml:space="preserve">#GstTestClock for which to await having enough pending clock</doc>
<type name="TestClock" c:type="GstTestClock*"/>
</instance-parameter>
<parameter name="count" transfer-ownership="none">
<doc xml:space="preserve">the number of pending clock notifications to wait for</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timeout_ms" transfer-ownership="none">
<doc xml:space="preserve">the timeout in milliseconds</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="pending_list" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1">
<doc xml:space="preserve">Address
of a #GList pointer variable to store the list of pending #GstClockIDs
that expired, or %NULL</doc>
<type name="GLib.List" c:type="GList**">
<type name="Gst.ClockID"/>
</type>
</parameter>
</parameters>
</method>
<method name="wait_for_multiple_pending_ids" c:identifier="gst_test_clock_wait_for_multiple_pending_ids" version="1.4">
<doc xml:space="preserve">Blocks until at least @count clock notifications have been requested from
@test_clock. There is no timeout for this wait, see the main description of

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@ -108,9 +108,11 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_eit">
<doc xml:space="preserve">Event Information Table (ATSC)</doc>
<field name="source_id" writable="1">
<doc xml:space="preserve">The source id</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="protocol_version" writable="1">
<doc xml:space="preserve">The protocol version</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="events" writable="1">
@ -127,15 +129,19 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_eit_event">
<doc xml:space="preserve">An ATSC EIT Event</doc>
<field name="event_id" writable="1">
<doc xml:space="preserve">The event id</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="start_time" writable="1">
<doc xml:space="preserve">The start time</doc>
<type name="guint32" c:type="guint32"/>
</field>
<field name="etm_location" writable="1">
<doc xml:space="preserve">The etm location</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="length_in_seconds" writable="1">
<doc xml:space="preserve">The length in seconds</doc>
<type name="guint32" c:type="guint32"/>
</field>
<field name="titles" writable="1">
@ -161,9 +167,11 @@ Consult the relevant specifications for more details.</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="protocol_version" writable="1">
<doc xml:space="preserve">The protocol version</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="etm_id" writable="1">
<doc xml:space="preserve">The etm id</doc>
<type name="guint32" c:type="guint32"/>
</field>
<field name="messages" writable="1">
@ -180,9 +188,11 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_mgt">
<doc xml:space="preserve">Master Guide Table (A65)</doc>
<field name="protocol_version" writable="1">
<doc xml:space="preserve">The protocol version</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="tables_defined" writable="1">
<doc xml:space="preserve">The numbers of subtables</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="tables" writable="1">
@ -205,12 +215,15 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_mgt_table">
<doc xml:space="preserve">Source from a @GstMpegtsAtscMGT</doc>
<field name="table_type" writable="1">
<doc xml:space="preserve">#GstMpegtsAtscMGTTableType</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="pid" writable="1">
<doc xml:space="preserve">The packet ID</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="version_number" writable="1">
<doc xml:space="preserve">The version number</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="number_bytes" writable="1">
@ -247,6 +260,7 @@ Consult the relevant specifications for more details.</doc>
glib:get-type="gst_mpegts_atsc_mult_string_get_type"
c:symbol-prefix="atsc_mult_string">
<field name="iso_639_langcode" writable="1">
<doc xml:space="preserve">The ISO639 language code</doc>
<array zero-terminated="0" c:type="gchar" fixed-size="4">
<type name="gchar" c:type="gchar"/>
</array>
@ -264,21 +278,26 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_stt">
<doc xml:space="preserve">System Time Table (A65)</doc>
<field name="protocol_version" writable="1">
<doc xml:space="preserve">The protocol version</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="system_time" writable="1">
<doc xml:space="preserve">The system time</doc>
<type name="guint32" c:type="guint32"/>
</field>
<field name="gps_utc_offset" writable="1">
<doc xml:space="preserve">The GPS to UTC offset</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="ds_status" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="ds_dayofmonth" writable="1">
<doc xml:space="preserve">The day of month</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="ds_hour" writable="1">
<doc xml:space="preserve">The hour</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="descriptors" writable="1">
@ -288,6 +307,7 @@ Consult the relevant specifications for more details.</doc>
</array>
</field>
<field name="utc_datetime" writable="1">
<doc xml:space="preserve">The UTC date and time</doc>
<type name="Gst.DateTime" c:type="GstDateTime*"/>
</field>
<method name="get_datetime_utc"
@ -307,16 +327,21 @@ Consult the relevant specifications for more details.</doc>
glib:type-name="GstMpegtsAtscStringSegment"
glib:get-type="gst_mpegts_atsc_string_segment_get_type"
c:symbol-prefix="atsc_string_segment">
<doc xml:space="preserve">A string segment</doc>
<field name="compression_type" writable="1">
<doc xml:space="preserve">The compression type</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="mode" writable="1">
<doc xml:space="preserve">The mode</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="compressed_data_size" writable="1">
<doc xml:space="preserve">The size of compressed data</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="compressed_data" writable="1">
<doc xml:space="preserve">The compressed data</doc>
<type name="guint8" c:type="guint8*"/>
</field>
<field name="cached_string" writable="1">
@ -344,9 +369,11 @@ Consult the relevant specifications for more details.</doc>
Terrestrial Virtual Channel Table (A65)
Cable Virtual Channel Table (A65)</doc>
<field name="transport_stream_id" writable="1">
<doc xml:space="preserve">The transport stream</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="protocol_version" writable="1">
<doc xml:space="preserve">The protocol version</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="sources" writable="1">
@ -369,48 +396,63 @@ Consult the relevant specifications for more details.</doc>
c:symbol-prefix="atsc_vct_source">
<doc xml:space="preserve">Source from a @GstMpegtsAtscVCT, can be used both for TVCT and CVCT tables</doc>
<field name="short_name" writable="1">
<doc xml:space="preserve">The short name of a source</doc>
<type name="utf8" c:type="gchar*"/>
</field>
<field name="major_channel_number" writable="1">
<doc xml:space="preserve">The major channel number</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="minor_channel_number" writable="1">
<doc xml:space="preserve">The minor channel number</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="modulation_mode" writable="1">
<doc xml:space="preserve">The modulation mode</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="carrier_frequency" writable="1">
<doc xml:space="preserve">The carrier frequency</doc>
<type name="guint32" c:type="guint32"/>
</field>
<field name="channel_TSID" writable="1">
<doc xml:space="preserve">The transport stream ID</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="program_number" writable="1">
<doc xml:space="preserve">The program number (see #GstMpegtsPatProgram)</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="ETM_location" writable="1">
<doc xml:space="preserve">The ETM location</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="access_controlled" writable="1">
<doc xml:space="preserve">is access controlled</doc>
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="hidden" writable="1">
<doc xml:space="preserve">is hidden</doc>
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="path_select" writable="1">
<doc xml:space="preserve">is path select, CVCT only</doc>
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="out_of_band" writable="1">
<doc xml:space="preserve">is out of band, CVCT only</doc>
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="hide_guide" writable="1">
<doc xml:space="preserve">is hide guide</doc>
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="service_type" writable="1">
<doc xml:space="preserve">The service type</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="source_id" writable="1">
<doc xml:space="preserve">The source id</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="descriptors" writable="1">

2
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@ -791,7 +791,7 @@ parameters if it wasn't called before.</doc>
version="1.6">
<doc xml:space="preserve">Check if the GStreamer PTP clock subsystem is initialized.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the GStreamer PTP clock subsystem is intialized.</doc>
<doc xml:space="preserve">%TRUE if the GStreamer PTP clock subsystem is initialized.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
</function>

89
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@ -123,7 +123,7 @@ new frame.</doc>
<record name="AudioVisualizerClass"
c:type="GstAudioVisualizerClass"
glib:is-gtype-struct-for="AudioVisualizer">
<field name="parent_class">
<field name="parent_class" readable="0" private="1">
<type name="Gst.ElementClass" c:type="GstElementClass"/>
</field>
<field name="setup">
@ -433,6 +433,12 @@ If the discovery of a URI times out, the %GST_DISCOVERER_TIMEOUT will be
set on the result flags.</doc>
<type name="guint64" c:type="guint64"/>
</property>
<property name="use-cache"
writable="1"
construct="1"
transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
<field name="parent">
<type name="GObject.Object" c:type="GObject"/>
</field>
@ -1717,12 +1723,12 @@ subtitles), are currently ignored.</doc>
</parameter>
</parameters>
</function>
<method name="copy" c:identifier="gst_encoding_profile_copy">
<method name="copy"
c:identifier="gst_encoding_profile_copy"
version="1.12">
<doc xml:space="preserve">Makes a deep copy of @self</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">The copy of @self
Since 1.12</doc>
<doc xml:space="preserve">The copy of @self</doc>
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
</return-value>
<parameters>
@ -1951,10 +1957,9 @@ during the encoding</doc>
</parameters>
</method>
<method name="set_enabled"
c:identifier="gst_encoding_profile_set_enabled">
<doc xml:space="preserve">Set whether the profile should be used or not.
Since 1.6</doc>
c:identifier="gst_encoding_profile_set_enabled"
version="1.6">
<doc xml:space="preserve">Set whether the profile should be used or not.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -2747,13 +2752,13 @@ in debugging.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MICRO"
value="4"
value="0"
c:type="GST_PLUGINS_BASE_VERSION_MICRO">
<doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MINOR"
value="14"
value="16"
c:type="GST_PLUGINS_BASE_VERSION_MINOR">
<doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
<type name="gint" c:type="gint"/>
@ -2796,12 +2801,11 @@ If mpegversion is 4, the "base-profile" field is also set in @caps.</doc>
</parameters>
</function>
<function name="codec_utils_aac_get_channels"
c:identifier="gst_codec_utils_aac_get_channels">
c:identifier="gst_codec_utils_aac_get_channels"
version="1.10">
<doc xml:space="preserve">Returns the channels of the given AAC stream.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The channels or 0 if the channel could not be determined.
Since 1.10</doc>
<doc xml:space="preserve">The channels or 0 if the channel could not be determined.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
@ -2843,13 +2847,12 @@ Main, LTP, SSR and others, the Main profile is used.
The @audio_config parameter follows the following format, starting from the
most significant bit of the first byte:
* Bit 0:4 contains the AudioObjectType
* Bit 0:4 contains the AudioObjectType (if this is 0x5, then the
real AudioObjectType is carried after the rate and channel data)
* Bit 5:8 contains the sample frequency index (if this is 0xf, then the
next 24 bits define the actual sample frequency, and subsequent
fields are appropriately shifted).
* Bit 9:12 contains the channel configuration
&gt; HE-AAC support has not yet been implemented.</doc>
* Bit 9:12 contains the channel configuration</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The level as a const string and %NULL if the level could not be
determined.</doc>
@ -2873,10 +2876,8 @@ determined.</doc>
<function name="codec_utils_aac_get_profile"
c:identifier="gst_codec_utils_aac_get_profile">
<doc xml:space="preserve">Returns the profile of the given AAC stream as a string. The profile is
determined using the AudioObjectType field which is in the first 5 bits of
@audio_config.
&gt; HE-AAC support has not yet been implemented.</doc>
normally determined using the AudioObjectType field which is in the first
5 bits of @audio_config</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The profile as a const string and %NULL if the profile could not be
determined.</doc>
@ -2898,13 +2899,12 @@ determined.</doc>
</parameters>
</function>
<function name="codec_utils_aac_get_sample_rate"
c:identifier="gst_codec_utils_aac_get_sample_rate">
c:identifier="gst_codec_utils_aac_get_sample_rate"
version="1.10">
<doc xml:space="preserve">Translates the sample rate index found in AAC headers to the actual sample
rate.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The sample rate if sr_idx is valid, 0 otherwise.
Since 1.10</doc>
<doc xml:space="preserve">The sample rate if sr_idx is valid, 0 otherwise.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
@ -3034,15 +3034,14 @@ byte.
</parameters>
</function>
<function name="codec_utils_h265_caps_set_level_tier_and_profile"
c:identifier="gst_codec_utils_h265_caps_set_level_tier_and_profile">
c:identifier="gst_codec_utils_h265_caps_set_level_tier_and_profile"
version="1.4">
<doc xml:space="preserve">Sets the level, tier and profile in @caps if it can be determined from
@profile_tier_level. See gst_codec_utils_h265_get_level(),
gst_codec_utils_h265_get_tier() and gst_codec_utils_h265_get_profile()
for more details on the parameters.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the level, tier, profile could be set, %FALSE otherwise.
Since 1.4</doc>
<doc xml:space="preserve">%TRUE if the level, tier, profile could be set, %FALSE otherwise.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -3064,14 +3063,13 @@ Since 1.4</doc>
</parameters>
</function>
<function name="codec_utils_h265_get_level"
c:identifier="gst_codec_utils_h265_get_level">
c:identifier="gst_codec_utils_h265_get_level"
version="1.4">
<doc xml:space="preserve">Converts the level indication (general_level_idc) in the stream's
profile_tier_level structure into a string. The profiel_tier_level is
expected to have the same format as for gst_codec_utils_h264_get_profile().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The level as a const string, or %NULL if there is an error.
Since 1.4</doc>
<doc xml:space="preserve">The level as a const string, or %NULL if there is an error.</doc>
<type name="utf8" c:type="const gchar*"/>
</return-value>
<parameters>
@ -3089,12 +3087,11 @@ Since 1.4</doc>
</parameters>
</function>
<function name="codec_utils_h265_get_level_idc"
c:identifier="gst_codec_utils_h265_get_level_idc">
c:identifier="gst_codec_utils_h265_get_level_idc"
version="1.4">
<doc xml:space="preserve">Transform a level string from the caps into the level_idc</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the level_idc or 0 if the level is unknown
Since 1.4</doc>
<doc xml:space="preserve">the level_idc or 0 if the level is unknown</doc>
<type name="guint8" c:type="guint8"/>
</return-value>
<parameters>
@ -3105,7 +3102,8 @@ Since 1.4</doc>
</parameters>
</function>
<function name="codec_utils_h265_get_profile"
c:identifier="gst_codec_utils_h265_get_profile">
c:identifier="gst_codec_utils_h265_get_profile"
version="1.4">
<doc xml:space="preserve">Converts the profile indication (general_profile_idc) in the stream's
profile_level_tier structure into a string. The profile_tier_level is
expected to have the following format, as defined in the H.265
@ -3123,9 +3121,7 @@ with bit 0 being the most significant bit of the first byte.
* Bit 44:87 - general_reserved_zero_44bits
* Bit 88:95 - general_level_idc</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The profile as a const string, or %NULL if there is an error.
Since 1.4</doc>
<doc xml:space="preserve">The profile as a const string, or %NULL if there is an error.</doc>
<type name="utf8" c:type="const gchar*"/>
</return-value>
<parameters>
@ -3143,14 +3139,13 @@ Since 1.4</doc>
</parameters>
</function>
<function name="codec_utils_h265_get_tier"
c:identifier="gst_codec_utils_h265_get_tier">
c:identifier="gst_codec_utils_h265_get_tier"
version="1.4">
<doc xml:space="preserve">Converts the tier indication (general_tier_flag) in the stream's
profile_tier_level structure into a string. The profile_tier_level
is expected to have the same format as for gst_codec_utils_h264_get_profile().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The tier as a const string, or %NULL if there is an error.
Since 1.4</doc>
<doc xml:space="preserve">The tier as a const string, or %NULL if there is an error.</doc>
<type name="utf8" c:type="const gchar*"/>
</return-value>
<parameters>

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@ -53,11 +53,10 @@ performed.</doc>
</parameters>
</constructor>
<function name="config_get_position_update_interval"
c:identifier="gst_player_config_get_position_update_interval">
c:identifier="gst_player_config_get_position_update_interval"
version="1.10">
<return-value transfer-ownership="none">
<doc xml:space="preserve">current position update interval in milliseconds
Since 1.10</doc>
<doc xml:space="preserve">current position update interval in milliseconds</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
@ -68,11 +67,10 @@ Since 1.10</doc>
</parameters>
</function>
<function name="config_get_seek_accurate"
c:identifier="gst_player_config_get_seek_accurate">
c:identifier="gst_player_config_get_seek_accurate"
version="1.12">
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if accurate seeking is enabled
Since 1.12</doc>
<doc xml:space="preserve">%TRUE if accurate seeking is enabled</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -83,12 +81,12 @@ Since 1.12</doc>
</parameters>
</function>
<function name="config_get_user_agent"
c:identifier="gst_player_config_get_user_agent">
c:identifier="gst_player_config_get_user_agent"
version="1.10">
<doc xml:space="preserve">Return the user agent which has been configured using
gst_player_config_set_user_agent() if any.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the configured agent, or %NULL
Since 1.10</doc>
<doc xml:space="preserve">the configured agent, or %NULL</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>
@ -99,10 +97,10 @@ Since 1.10</doc>
</parameters>
</function>
<function name="config_set_position_update_interval"
c:identifier="gst_player_config_set_position_update_interval">
c:identifier="gst_player_config_set_position_update_interval"
version="1.10">
<doc xml:space="preserve">set interval in milliseconds between two position-updated signals.
pass 0 to stop updating the position.
Since 1.10</doc>
pass 0 to stop updating the position.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -144,12 +142,11 @@ Accurate seeking is disabled by default.</doc>
</parameters>
</function>
<function name="config_set_user_agent"
c:identifier="gst_player_config_set_user_agent">
c:identifier="gst_player_config_set_user_agent"
version="1.10">
<doc xml:space="preserve">Set the user agent to pass to the server if @player needs to connect
to a server during playback. This is typically used when playing HTTP
or RTSP streams.
Since 1.10</doc>
or RTSP streams.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -239,12 +236,11 @@ matching #GstPlayerVideoInfo.</doc>
</return-value>
</function>
<method name="get_audio_video_offset"
c:identifier="gst_player_get_audio_video_offset">
c:identifier="gst_player_get_audio_video_offset"
version="1.10">
<doc xml:space="preserve">Retrieve the current value of audio-video-offset property</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The current value of audio-video-offset in nanoseconds
Since 1.10</doc>
<doc xml:space="preserve">The current value of audio-video-offset in nanoseconds</doc>
<type name="gint64" c:type="gint64"/>
</return-value>
<parameters>
@ -274,15 +270,15 @@ Since 1.10</doc>
</parameter>
</parameters>
</method>
<method name="get_config" c:identifier="gst_player_get_config">
<method name="get_config"
c:identifier="gst_player_get_config"
version="1.10">
<doc xml:space="preserve">Get a copy of the current configuration of the player. This configuration
can either be modified and used for the gst_player_set_config() call
or it must be freed after usage.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a copy of the current configuration of @player. Use
gst_structure_free() after usage or gst_player_set_config().
Since 1.10</doc>
gst_structure_free() after usage or gst_player_set_config().</doc>
<type name="Gst.Structure" c:type="GstStructure*"/>
</return-value>
<parameters>
@ -479,6 +475,21 @@ currently-playing stream.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_subtitle_video_offset"
c:identifier="gst_player_get_subtitle_video_offset"
version="1.16">
<doc xml:space="preserve">Retrieve the current value of subtitle-video-offset property</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The current value of subtitle-video-offset in nanoseconds</doc>
<type name="gint64" c:type="gint64"/>
</return-value>
<parameters>
<instance-parameter name="player" transfer-ownership="none">
<doc xml:space="preserve">#GstPlayer instance</doc>
<type name="Player" c:type="GstPlayer*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_uri" c:identifier="gst_player_get_uri">
<doc xml:space="preserve">Gets the URI of the currently-playing stream.</doc>
<return-value transfer-ownership="full">
@ -494,7 +505,8 @@ currently-playing stream. g_free() after usage.</doc>
</parameters>
</method>
<method name="get_video_snapshot"
c:identifier="gst_player_get_video_snapshot">
c:identifier="gst_player_get_video_snapshot"
version="1.12">
<doc xml:space="preserve">Get a snapshot of the currently selected video stream, if any. The format can be
selected with @format and optional configuration is possible with @config
Currently supported settings are:
@ -502,9 +514,7 @@ Currently supported settings are:
- pixel-aspect-ratio of type GST_TYPE_FRACTION
Except for GST_PLAYER_THUMBNAIL_RAW_NATIVE format, if no config is set, pixel-aspect-ratio would be 1/1</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">Current video snapshot sample or %NULL on failure
Since 1.12</doc>
<doc xml:space="preserve">Current video snapshot sample or %NULL on failure</doc>
<type name="Gst.Sample" c:type="GstSample*"/>
</return-value>
<parameters>
@ -631,10 +641,9 @@ Sets the audio track @stream_idex.</doc>
</parameters>
</method>
<method name="set_audio_video_offset"
c:identifier="gst_player_set_audio_video_offset">
<doc xml:space="preserve">Sets audio-video-offset property by value of @offset
Since 1.10</doc>
c:identifier="gst_player_set_audio_video_offset"
version="1.10">
<doc xml:space="preserve">Sets audio-video-offset property by value of @offset</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
@ -672,7 +681,9 @@ value.</doc>
</parameter>
</parameters>
</method>
<method name="set_config" c:identifier="gst_player_set_config">
<method name="set_config"
c:identifier="gst_player_set_config"
version="1.10">
<doc xml:space="preserve">Set the configuration of the player. If the player is already configured, and
the configuration haven't change, this function will return %TRUE. If the
player is not in the GST_PLAYER_STATE_STOPPED, this method will return %FALSE
@ -683,8 +694,7 @@ the player.
This function takes ownership of @config.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE when the configuration could be set.
Since 1.10</doc>
<doc xml:space="preserve">%TRUE when the configuration could be set.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -825,6 +835,24 @@ rendered.</doc>
</parameter>
</parameters>
</method>
<method name="set_subtitle_video_offset"
c:identifier="gst_player_set_subtitle_video_offset"
version="1.16">
<doc xml:space="preserve">Sets subtitle-video-offset property by value of @offset</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="player" transfer-ownership="none">
<doc xml:space="preserve">#GstPlayer instance</doc>
<type name="Player" c:type="GstPlayer*"/>
</instance-parameter>
<parameter name="offset" transfer-ownership="none">
<doc xml:space="preserve">#gint64 in nanoseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="set_uri" c:identifier="gst_player_set_uri">
<doc xml:space="preserve">Sets the next URI to play.</doc>
<return-value transfer-ownership="none">
@ -980,6 +1008,11 @@ in the stream.</doc>
transfer-ownership="none">
<type name="PlayerSignalDispatcher"/>
</property>
<property name="subtitle-video-offset"
writable="1"
transfer-ownership="none">
<type name="gint64" c:type="gint64"/>
</property>
<property name="suburi" writable="1" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</property>
@ -1986,11 +2019,9 @@ of the given @info (ex: "audio", "video", "subtitle")</doc>
</parameters>
</function>
<function name="new_with_sink"
c:identifier="gst_player_video_overlay_video_renderer_new_with_sink">
c:identifier="gst_player_video_overlay_video_renderer_new_with_sink"
version="1.12">
<return-value transfer-ownership="full">
<doc xml:space="preserve">
Since 1.12</doc>
<type name="PlayerVideoRenderer" c:type="GstPlayerVideoRenderer*"/>
</return-value>
<parameters>

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gir-files/GstRtsp-1.0.gir
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@ -8,6 +8,7 @@ and/or use gtk-doc annotations. -->
xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
<include name="Gio" version="2.0"/>
<include name="Gst" version="1.0"/>
<include name="GstBase" version="1.0"/>
<include name="GstSdp" version="1.0"/>
<package name="gstreamer-rtsp-1.0"/>
<c:include name="gst/rtsp/rtsp.h"/>
@ -21,13 +22,17 @@ and/or use gtk-doc annotations. -->
glib:type-name="GstRTSPAuthCredential"
glib:get-type="gst_rtsp_auth_credential_get_type"
c:symbol-prefix="rtsp_auth_credential">
<doc xml:space="preserve">RTSP Authentication credentials</doc>
<field name="scheme" writable="1">
<doc xml:space="preserve">a #GstRTSPAuthMethod</doc>
<type name="RTSPAuthMethod" c:type="GstRTSPAuthMethod"/>
</field>
<field name="params" writable="1">
<doc xml:space="preserve">A NULL-terminated array of #GstRTSPAuthParam</doc>
<type name="RTSPAuthParam" c:type="GstRTSPAuthParam**"/>
</field>
<field name="authorization" writable="1">
<doc xml:space="preserve">The authorization for the basic schem</doc>
<type name="utf8" c:type="gchar*"/>
</field>
</record>
@ -60,10 +65,13 @@ and/or use gtk-doc annotations. -->
glib:type-name="GstRTSPAuthParam"
glib:get-type="gst_rtsp_auth_param_get_type"
c:symbol-prefix="rtsp_auth_param">
<doc xml:space="preserve">RTSP Authentication parameter</doc>
<field name="name" writable="1">
<doc xml:space="preserve">The name of the parameter</doc>
<type name="utf8" c:type="gchar*"/>
</field>
<field name="value" writable="1">
<doc xml:space="preserve">The value of the parameter</doc>
<type name="utf8" c:type="gchar*"/>
</field>
<method name="copy" c:identifier="gst_rtsp_auth_param_copy">
@ -140,7 +148,8 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameters>
</method>
<method name="connect_with_response"
c:identifier="gst_rtsp_connection_connect_with_response">
c:identifier="gst_rtsp_connection_connect_with_response"
version="1.8">
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
gst_rtsp_connection_create(). If @timeout is %NULL this function can block
forever. If @timeout contains a valid timeout, this function will return
@ -149,9 +158,7 @@ forever. If @timeout contains a valid timeout, this function will return
This function can be cancelled with gst_rtsp_connection_flush().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK when a connection could be made.
Since 1.8</doc>
<doc xml:space="preserve">#GST_RTSP_OK when a connection could be made.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
@ -545,6 +552,39 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="send_messages"
c:identifier="gst_rtsp_connection_send_messages"
version="1.16">
<doc xml:space="preserve">Attempt to send @messages to the connected @conn, blocking up to
the specified @timeout. @timeout can be %NULL, in which case this function
might block forever.
This function can be cancelled with gst_rtsp_connection_flush().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="conn" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPConnection</doc>
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
</instance-parameter>
<parameter name="messages" transfer-ownership="none">
<doc xml:space="preserve">the messages to send</doc>
<array length="1" zero-terminated="0" c:type="GstRTSPMessage*">
<type name="RTSPMessage" c:type="GstRTSPMessage"/>
</array>
</parameter>
<parameter name="n_messages" transfer-ownership="none">
<doc xml:space="preserve">the number of messages to send</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout value or %NULL</doc>
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
</parameter>
</parameters>
</method>
<method name="set_accept_certificate_func"
c:identifier="gst_rtsp_connection_set_accept_certificate_func"
version="1.14">
@ -1290,6 +1330,7 @@ read from @socket which should be used before starting to read new data.</doc>
<record name="RTSPExtensionInterface"
c:type="GstRTSPExtensionInterface"
glib:is-gtype-struct-for="RTSPExtension">
<doc xml:space="preserve">An interface representing RTSP extensions.</doc>
<field name="parent">
<type name="GObject.TypeInterface" c:type="GTypeInterface"/>
</field>
@ -1925,8 +1966,18 @@ read from @socket which should be used before starting to read new data.</doc>
c:identifier="GST_RTSP_HDR_ACCEPT_RANGES"
glib:nick="accept-ranges">
</member>
<member name="last"
<member name="frames"
value="87"
c:identifier="GST_RTSP_HDR_FRAMES"
glib:nick="frames">
</member>
<member name="rate_control"
value="88"
c:identifier="GST_RTSP_HDR_RATE_CONTROL"
glib:nick="rate-control">
</member>
<member name="last"
value="89"
c:identifier="GST_RTSP_HDR_LAST"
glib:nick="last">
</member>
@ -2023,8 +2074,11 @@ read from @socket which should be used before starting to read new data.</doc>
<field name="body_size" readable="0" private="1">
<type name="guint" c:type="guint"/>
</field>
<field name="body_buffer" readable="0" private="1">
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" c:type="gpointer" fixed-size="4">
<array zero-terminated="0" c:type="gpointer" fixed-size="3">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -2142,7 +2196,11 @@ for transmission.</doc>
</method>
<method name="get_body" c:identifier="gst_rtsp_message_get_body">
<doc xml:space="preserve">Get the body of @msg. @data remains valid for as long as @msg is valid and
unchanged.</doc>
unchanged.
If the message body was set as a #GstBuffer before this will cause the data
to be copied and stored in the message. The #GstBuffer will no longer be
kept in the message.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
@ -2170,6 +2228,33 @@ unchanged.</doc>
</parameter>
</parameters>
</method>
<method name="get_body_buffer"
c:identifier="gst_rtsp_message_get_body_buffer"
version="1.16">
<doc xml:space="preserve">Get the body of @msg. @buffer remains valid for as long as @msg is valid and
unchanged.
If body data was set from raw memory instead of a #GstBuffer this function
will always return %NULL. The caller can check if there is a body buffer by
calling gst_rtsp_message_has_body_buffer().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="msg" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="const GstRTSPMessage*"/>
</instance-parameter>
<parameter name="buffer"
direction="out"
caller-allocates="0"
transfer-ownership="none">
<doc xml:space="preserve">location for the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer**"/>
</parameter>
</parameters>
</method>
<method name="get_header" c:identifier="gst_rtsp_message_get_header">
<doc xml:space="preserve">Get the @indx header value with key @field from @msg. The result in @value
stays valid as long as it remains present in @msg.</doc>
@ -2245,6 +2330,22 @@ was not found.</doc>
</instance-parameter>
</parameters>
</method>
<method name="has_body_buffer"
c:identifier="gst_rtsp_message_has_body_buffer"
version="1.16">
<doc xml:space="preserve">Checks if @msg has a body and the body is stored as #GstBuffer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if @msg has a body and it's stored as #GstBuffer, %FALSE
otherwise.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="msg" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="const GstRTSPMessage*"/>
</instance-parameter>
</parameters>
</method>
<method name="init" c:identifier="gst_rtsp_message_init">
<doc xml:space="preserve">Initialize @msg. This function is mostly used when @msg is allocated on the
stack. The reverse operation of this is gst_rtsp_message_unset().</doc>
@ -2515,7 +2616,8 @@ all matching headers will be removed.</doc>
</parameters>
</method>
<method name="set_body" c:identifier="gst_rtsp_message_set_body">
<doc xml:space="preserve">Set the body of @msg to a copy of @data.</doc>
<doc xml:space="preserve">Set the body of @msg to a copy of @data. Any existing body or body buffer
will be replaced by the new body.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
@ -2537,6 +2639,26 @@ all matching headers will be removed.</doc>
</parameter>
</parameters>
</method>
<method name="set_body_buffer"
c:identifier="gst_rtsp_message_set_body_buffer"
version="1.16">
<doc xml:space="preserve">Set the body of @msg to @buffer. Any existing body or body buffer
will be replaced by the new body.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="msg" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</instance-parameter>
<parameter name="buffer" transfer-ownership="none">
<doc xml:space="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<method name="steal_body" c:identifier="gst_rtsp_message_steal_body">
<doc xml:space="preserve">Take the body of @msg and store it in @data and @size. After this method,
the body and size of @msg will be set to %NULL and 0 respectively.</doc>
@ -2567,9 +2689,36 @@ the body and size of @msg will be set to %NULL and 0 respectively.</doc>
</parameter>
</parameters>
</method>
<method name="steal_body_buffer"
c:identifier="gst_rtsp_message_steal_body_buffer"
version="1.16">
<doc xml:space="preserve">Take the body of @msg and store it in @buffer. After this method,
the body and size of @msg will be set to %NULL and 0 respectively.
If body data was set from raw memory instead of a #GstBuffer this function
will always return %NULL. The caller can check if there is a body buffer by
calling gst_rtsp_message_has_body_buffer().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="msg" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</instance-parameter>
<parameter name="buffer"
direction="out"
caller-allocates="0"
transfer-ownership="full">
<doc xml:space="preserve">location for the buffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer**"/>
</parameter>
</parameters>
</method>
<method name="take_body" c:identifier="gst_rtsp_message_take_body">
<doc xml:space="preserve">Set the body of @msg to @data and @size. This method takes ownership of
@data.</doc>
@data. Any existing body or body buffer will be replaced by the new body.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
@ -2591,6 +2740,26 @@ the body and size of @msg will be set to %NULL and 0 respectively.</doc>
</parameter>
</parameters>
</method>
<method name="take_body_buffer"
c:identifier="gst_rtsp_message_take_body_buffer"
version="1.16">
<doc xml:space="preserve">Set the body of @msg to @buffer. This method takes ownership of @buffer.
Any existing body or body buffer will be replaced by the new body.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="msg" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</instance-parameter>
<parameter name="buffer" transfer-ownership="full">
<doc xml:space="preserve">a #GstBuffer</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
</parameters>
</method>
<method name="take_header" c:identifier="gst_rtsp_message_take_header">
<doc xml:space="preserve">Add a header with key @field and @value to @msg. This function takes
ownership of @value.</doc>
@ -4011,6 +4180,46 @@ callback.</doc>
</parameter>
</parameters>
</method>
<method name="send_messages"
c:identifier="gst_rtsp_watch_send_messages"
version="1.16">
<doc xml:space="preserve">Sends @messages using the connection of the @watch. If they cannot be sent
immediately, they will be queued for transmission in @watch. The contents of
@messages will then be serialized and transmitted when the connection of the
@watch becomes writable. In case the @messages are queued, the ID returned in
@id will be non-zero and used as the ID argument in the message_sent
callback once the last message is sent. The callback will only be called
once for the last message.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
<type name="RTSPResult" c:type="GstRTSPResult"/>
</return-value>
<parameters>
<instance-parameter name="watch" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPWatch</doc>
<type name="RTSPWatch" c:type="GstRTSPWatch*"/>
</instance-parameter>
<parameter name="messages" transfer-ownership="none">
<doc xml:space="preserve">the messages to send</doc>
<array length="1" zero-terminated="0" c:type="GstRTSPMessage*">
<type name="RTSPMessage" c:type="GstRTSPMessage"/>
</array>
</parameter>
<parameter name="n_messages" transfer-ownership="none">
<doc xml:space="preserve">the number of messages to send</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="id"
direction="out"
caller-allocates="0"
transfer-ownership="full"
optional="1"
allow-none="1">
<doc xml:space="preserve">location for a message ID or %NULL</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</method>
<method name="set_flushing"
c:identifier="gst_rtsp_watch_set_flushing"
version="1.4">
@ -4573,6 +4782,46 @@ Currently only supported algorithm "md5".</doc>
</parameter>
</parameters>
</function>
<function name="rtsp_generate_digest_auth_response_from_md5"
c:identifier="gst_rtsp_generate_digest_auth_response_from_md5"
version="1.16">
<doc xml:space="preserve">Calculates the digest auth response from the values given by the server and
the md5sum. See RFC2069 for details.
This function is useful when the passwords are not stored in clear text,
but instead in the same format as the .htdigest file.
Currently only supported algorithm "md5".</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">Authentication response or %NULL if unsupported</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>
<parameter name="algorithm"
transfer-ownership="none"
nullable="1"
allow-none="1">
<doc xml:space="preserve">Hash algorithm to use, or %NULL for MD5</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="method" transfer-ownership="none">
<doc xml:space="preserve">Request method, e.g. PLAY</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="md5" transfer-ownership="none">
<doc xml:space="preserve">The md5 sum of username:realm:password</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="uri" transfer-ownership="none">
<doc xml:space="preserve">Original request URI</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="nonce" transfer-ownership="none">
<doc xml:space="preserve">Nonce</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</function>
<function name="rtsp_header_allow_multiple"
c:identifier="gst_rtsp_header_allow_multiple">
<doc xml:space="preserve">Check whether @field may appear multiple times in a message.</doc>

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gir-files/GstSdp-1.0.gir
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@ -44,6 +44,11 @@ and/or use gtk-doc annotations. -->
c:identifier="GST_MIKEY_ENC_AES_KW_128">
<doc xml:space="preserve">AES Key Wrap using a 128-bit key</doc>
</member>
<member name="aes_gcm_128"
value="6"
c:identifier="GST_MIKEY_ENC_AES_GCM_128">
<doc xml:space="preserve">AES-GCM using a 128-bit key (Since: 1.16)</doc>
</member>
</enumeration>
<record name="MIKEYEncryptInfo" c:type="GstMIKEYEncryptInfo" disguised="1">
</record>
@ -100,6 +105,7 @@ protocol sessions.</doc>
<member name="mikey_map_type_srtp"
value="0"
c:identifier="GST_MIKEY_MAP_TYPE_SRTP">
<doc xml:space="preserve">SRTP</doc>
</member>
</enumeration>
<record name="MIKEYMessage"
@ -656,7 +662,7 @@ will be appended to @msg.</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="map_type" transfer-ownership="none">
<doc xml:space="preserve">the #GstMIKEYCSIDMapType</doc>
<doc xml:space="preserve">the #GstMIKEYMapType</doc>
<type name="MIKEYMapType" c:type="GstMIKEYMapType"/>
</parameter>
</parameters>
@ -1177,6 +1183,7 @@ payload to the KEMAC.</doc>
<type name="MIKEYPayload" c:type="GstMIKEYPayload"/>
</field>
<field name="key_type" writable="1">
<doc xml:space="preserve">the #GstMIKEYKeyDataType of @key_data</doc>
<type name="MIKEYKeyDataType" c:type="GstMIKEYKeyDataType"/>
</field>
<field name="key_len" writable="1">
@ -1184,6 +1191,7 @@ payload to the KEMAC.</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="key_data" writable="1">
<doc xml:space="preserve">the key data</doc>
<type name="guint8" c:type="guint8*"/>
</field>
<field name="salt_len" writable="1">
@ -1264,7 +1272,7 @@ specific security protocol</doc>
<type name="MIKEYSecProto" c:type="GstMIKEYSecProto"/>
</field>
<field name="params" writable="1">
<doc xml:space="preserve">array of #GstMIKEYPayloadPSParam</doc>
<doc xml:space="preserve">array of #GstMIKEYPayloadSPParam</doc>
<array name="GLib.Array" c:type="GArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
@ -1353,6 +1361,7 @@ specific security protocol</doc>
<member name="mikey_sec_proto_srtp"
value="0"
c:identifier="GST_MIKEY_SEC_PROTO_SRTP">
<doc xml:space="preserve">SRTP</doc>
</member>
</enumeration>
<enumeration name="MIKEYSecSRTP" c:type="GstMIKEYSecSRTP">
@ -1420,6 +1429,11 @@ specific security protocol</doc>
c:identifier="GST_MIKEY_SP_SRTP_SRTP_PREFIX_LEN">
<doc xml:space="preserve">SRTP prefix length</doc>
</member>
<member name="aead_auth_tag_len"
value="20"
c:identifier="GST_MIKEY_SP_SRTP_AEAD_AUTH_TAG_LEN">
<doc xml:space="preserve">AEAD authentication tag length (Since: 1.16)</doc>
</member>
</enumeration>
<enumeration name="MIKEYTSType" c:type="GstMIKEYTSType">
<doc xml:space="preserve">Specifies the timestamp type.</doc>
@ -3355,11 +3369,11 @@ When -1 is given as @idx, the zone is inserted at the end.</doc>
<type name="SDPMessage" c:type="GstSDPMessage*"/>
</instance-parameter>
<parameter name="idx" transfer-ownership="none">
<doc xml:space="preserve">an index
@zone a #GstSDPZone</doc>
<doc xml:space="preserve">an index</doc>
<type name="gint" c:type="gint"/>
</parameter>
<parameter name="zone" transfer-ownership="none">
<doc xml:space="preserve">a #GstSDPZone</doc>
<type name="SDPZone" c:type="GstSDPZone*"/>
</parameter>
</parameters>
@ -3906,6 +3920,28 @@ stack and initialized with gst_sdp_message_init().</doc>
</parameter>
</parameters>
</function>
<function name="new_from_text"
c:identifier="gst_sdp_message_new_from_text"
version="1.16">
<doc xml:space="preserve">Parse @text and create a new SDPMessage from these.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstSDPResult.</doc>
<type name="SDPResult" c:type="GstSDPResult"/>
</return-value>
<parameters>
<parameter name="text" transfer-ownership="none">
<doc xml:space="preserve">A dynamically allocated string representing the SDP description</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="msg"
direction="out"
caller-allocates="0"
transfer-ownership="full">
<doc xml:space="preserve">pointer to new #GstSDPMessage</doc>
<type name="SDPMessage" c:type="GstSDPMessage**"/>
</parameter>
</parameters>
</function>
<function name="parse_buffer"
c:identifier="gst_sdp_message_parse_buffer">
<doc xml:space="preserve">Parse the contents of @size bytes pointed to by @data and store the result in
@ -4261,6 +4297,29 @@ a=rtcp-fb:(payload) (param1) [param2]...</doc>
</parameter>
</parameters>
</function>
<function name="sdp_message_new_from_text"
c:identifier="gst_sdp_message_new_from_text"
moved-to="SDPMessage.new_from_text"
version="1.16">
<doc xml:space="preserve">Parse @text and create a new SDPMessage from these.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstSDPResult.</doc>
<type name="SDPResult" c:type="GstSDPResult"/>
</return-value>
<parameters>
<parameter name="text" transfer-ownership="none">
<doc xml:space="preserve">A dynamically allocated string representing the SDP description</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
<parameter name="msg"
direction="out"
caller-allocates="0"
transfer-ownership="full">
<doc xml:space="preserve">pointer to new #GstSDPMessage</doc>
<type name="SDPMessage" c:type="GstSDPMessage**"/>
</parameter>
</parameters>
</function>
<function name="sdp_message_parse_buffer"
c:identifier="gst_sdp_message_parse_buffer"
moved-to="SDPMessage.parse_buffer">

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gir-files/GstWebRTC-1.0.gir
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@ -15,6 +15,37 @@ and/or use gtk-doc annotations. -->
shared-library="libgstwebrtc-1.0.so.0"
c:identifier-prefixes="Gst"
c:symbol-prefixes="gst">
<enumeration name="WebRTCBundlePolicy"
glib:type-name="GstWebRTCBundlePolicy"
glib:get-type="gst_webrtc_bundle_policy_get_type"
c:type="GstWebRTCBundlePolicy">
<doc xml:space="preserve">GST_WEBRTC_BUNDLE_POLICY_NONE: none
GST_WEBRTC_BUNDLE_POLICY_BALANCED: balanced
GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT: max-compat
GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE: max-bundle
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.</doc>
<member name="none"
value="0"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_NONE"
glib:nick="none">
</member>
<member name="balanced"
value="1"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_BALANCED"
glib:nick="balanced">
</member>
<member name="max_compat"
value="2"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT"
glib:nick="max-compat">
</member>
<member name="max_bundle"
value="3"
c:identifier="GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE"
glib:nick="max-bundle">
</member>
</enumeration>
<enumeration name="WebRTCDTLSSetup"
glib:type-name="GstWebRTCDTLSSetup"
glib:get-type="gst_webrtc_dtls_setup_get_type"
@ -183,6 +214,42 @@ GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
glib:nick="connected">
</member>
</enumeration>
<enumeration name="WebRTCDataChannelState"
glib:type-name="GstWebRTCDataChannelState"
glib:get-type="gst_webrtc_data_channel_state_get_type"
c:type="GstWebRTCDataChannelState">
<doc xml:space="preserve">GST_WEBRTC_DATA_CHANNEL_STATE_NEW: new
GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate&lt;/ulink&gt;</doc>
<member name="new"
value="0"
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW"
glib:nick="new">
</member>
<member name="connecting"
value="1"
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING"
glib:nick="connecting">
</member>
<member name="open"
value="2"
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_OPEN"
glib:nick="open">
</member>
<member name="closing"
value="3"
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING"
glib:nick="closing">
</member>
<member name="closed"
value="4"
c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED"
glib:nick="closed">
</member>
</enumeration>
<enumeration name="WebRTCFECType"
glib:type-name="GstWebRTCFECType"
glib:get-type="gst_webrtc_fec_type_get_type"
@ -465,6 +532,25 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
</array>
</field>
</record>
<enumeration name="WebRTCICETransportPolicy"
glib:type-name="GstWebRTCICETransportPolicy"
glib:get-type="gst_webrtc_ice_transport_policy_get_type"
c:type="GstWebRTCICETransportPolicy">
<doc xml:space="preserve">GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL: all
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY: relay
See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
for more information.</doc>
<member name="all"
value="0"
c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL"
glib:nick="all">
</member>
<member name="relay"
value="1"
c:identifier="GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY"
glib:nick="relay">
</member>
</enumeration>
<enumeration name="WebRTCPeerConnectionState"
glib:type-name="GstWebRTCPeerConnectionState"
glib:get-type="gst_webrtc_peer_connection_state_get_type"
@ -507,6 +593,36 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
glib:nick="closed">
</member>
</enumeration>
<enumeration name="WebRTCPriorityType"
glib:type-name="GstWebRTCPriorityType"
glib:get-type="gst_webrtc_priority_type_get_type"
c:type="GstWebRTCPriorityType">
<doc xml:space="preserve">GST_WEBRTC_PRIORITY_TYPE_VERY_LOW: very-low
GST_WEBRTC_PRIORITY_TYPE_LOW: low
GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
GST_WEBRTC_PRIORITY_TYPE_HIGH: high
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype&lt;/ulink&gt;</doc>
<member name="very_low"
value="1"
c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW"
glib:nick="very-low">
</member>
<member name="low"
value="2"
c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW"
glib:nick="low">
</member>
<member name="medium"
value="3"
c:identifier="GST_WEBRTC_PRIORITY_TYPE_MEDIUM"
glib:nick="medium">
</member>
<member name="high"
value="4"
c:identifier="GST_WEBRTC_PRIORITY_TYPE_HIGH"
glib:nick="high">
</member>
</enumeration>
<class name="WebRTCRTPReceiver"
c:symbol-prefix="webrtc_rtp_receiver"
c:type="GstWebRTCRTPReceiver"
@ -749,6 +865,36 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
glib:nick="sendrecv">
</member>
</enumeration>
<enumeration name="WebRTCSCTPTransportState"
glib:type-name="GstWebRTCSCTPTransportState"
glib:get-type="gst_webrtc_sctp_transport_state_get_type"
c:type="GstWebRTCSCTPTransportState">
<doc xml:space="preserve">GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW: new
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate&lt;/ulink&gt;</doc>
<member name="new"
value="0"
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW"
glib:nick="new">
</member>
<member name="connecting"
value="1"
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING"
glib:nick="connecting">
</member>
<member name="connected"
value="2"
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED"
glib:nick="connected">
</member>
<member name="closed"
value="3"
c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED"
glib:nick="closed">
</member>
</enumeration>
<enumeration name="WebRTCSDPType"
glib:type-name="GstWebRTCSDPType"
glib:get-type="gst_webrtc_sdp_type_get_type"

34
gir-files/fix.sh
View file

@ -1,32 +1,6 @@
#!/bin/bash
set -x -e
# Remove GLFuncs record
# commit 5765641
xmlstarlet ed --pf --inplace --delete '//_:record[@name="GLFuncs"]' GstGL-1.0.gir
# Add a disguised GFuncs record (two steps)
xmlstarlet ed --pf --inplace \
--subnode '//_:namespace' --type elem -n 'recordTMP' --value ' ' \
GstGL-1.0.gir
xmlstarlet ed --pf --inplace \
--insert '//_:recordTMP' -t attr -n 'name' --value 'GLFuncs' \
--insert '//_:recordTMP' -t attr -n 'c:type' --value 'GstGLFuncs' \
--insert '//_:recordTMP' -t attr -n 'disguised' --value '1' \
--rename '//_:recordTMP' --value 'record' \
GstGL-1.0.gir
# incorrect GIR due bug #797144
xmlstarlet ed --pf --inplace \
--update '//*[@c:identifier="Dubois optimised Green-Magenta anaglyph"]/@c:identifier' \
--value GST_GL_STEREO_DOWNMIX_ANAGLYPH_GREEN_MAGENTA_DUBOIS \
--update '//*[@c:identifier="Dubois optimised Red-Cyan anaglyph"]/@c:identifier' \
--value GST_GL_STEREO_DOWNMIX_ANAGLYPH_RED_CYAN_DUBOIS \
--update '//*[@c:identifier="Dubois optimised Amber-Blue anaglyph"]/@c:identifier' \
--value GST_GL_STEREO_DOWNMIX_ANAGLYPH_AMBER_BLUE_DUBOIS \
GstGL-1.0.gir
# replace wayland structures to gpointers
xmlstarlet ed --pf --inplace \
--update '//*[@c:type="wl_display*"]/@c:type' \
@ -71,3 +45,11 @@ xmlstarlet ed --pf --inplace \
--delete '//_:callback[starts-with(@name, "Check")]' \
--delete '//_:record[starts-with(@name, "Check")]' \
GstCheck-1.0.gir
# Change GstVideoAncillary.data to a fixed-size 256 byte array
xmlstarlet ed --pf --inplace \
--delete '//_:record[@name="VideoAncillary"]/_:field[@name="data"]/_:array/@length' \
--insert '//_:record[@name="VideoAncillary"]/_:field[@name="data"]/_:array' \
--type attr --name 'fixed-size' --value '256' \
GstVideo-1.0.gir