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gir-files: Update to gstreamer master

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Vivia Nikolaidou 2020-04-30 17:31:25 +03:00 committed by Sebastian Dröge
parent 20972ae65f
commit 7cefac11e5
18 changed files with 9581 additions and 4330 deletions
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8161
gir-files/GES-1.0.gir
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gir-files/Gst-1.0.gir
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46
gir-files/GstApp-1.0.gir
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@ -294,7 +294,7 @@ filled.</doc>
<doc xml:space="preserve">Check if appsink will emit the "new-preroll" and "new-sample" signals.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if @appsink is emiting the "new-preroll" and "new-sample"
<doc xml:space="preserve">%TRUE if @appsink is emitting the "new-preroll" and "new-sample"
signals.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
@ -1032,7 +1032,7 @@ mode when implementing various network protocols or hardware devices.
The pull mode, in which the need-data signal triggers the next push-buffer call.
This mode is typically used in the "random-access" stream-type. Use this
mode for file access or other randomly accessable sources. In this mode, a
mode for file access or other randomly accessible sources. In this mode, a
buffer of exactly the amount of bytes given by the need-data signal should be
pushed into appsrc.
@ -1052,7 +1052,7 @@ occurs or the state of the appsrc has gone through READY.</doc>
element is the last buffer of the stream.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
@ -1096,9 +1096,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1121,9 +1121,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1150,9 +1150,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1186,7 +1186,7 @@ extracted</doc>
element is the last buffer of the stream.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
@ -1328,9 +1328,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1353,9 +1353,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1382,9 +1382,9 @@ When the block property is TRUE, this function can block until free
space becomes available in the queue.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1843,9 +1843,9 @@ gst_app_src_set_callbacks().</doc>
<callback name="push_buffer">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1864,7 +1864,7 @@ gst_app_src_set_callbacks().</doc>
<callback name="end_of_stream">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
@ -1880,9 +1880,9 @@ gst_app_src_set_callbacks().</doc>
<callback name="push_sample">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
@ -1902,9 +1902,9 @@ extracted</doc>
<callback name="push_buffer_list">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
#GST_FLOW_EOS when EOS occured.</doc>
#GST_FLOW_EOS when EOS occurred.</doc>
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>

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gir-files/GstAudio-1.0.gir
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@ -324,14 +324,14 @@ and/or use gtk-doc annotations. -->
</parameters>
</function-macro>
<constant name="AUDIO_CONVERTER_OPT_DITHER_METHOD" value="GstAudioConverter.dither-method" c:type="GST_AUDIO_CONVERTER_OPT_DITHER_METHOD">
<doc xml:space="preserve">#GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
<doc xml:space="preserve">#GstAudioDitherMethod, The dither method to use when
changing bit depth.
Default is #GST_AUDIO_DITHER_NONE.</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="AUDIO_CONVERTER_OPT_MIX_MATRIX" value="GstAudioConverter.mix-matrix" c:type="GST_AUDIO_CONVERTER_OPT_MIX_MATRIX">
<doc xml:space="preserve">#GST_TYPE_VALUE_LIST, The channel mapping matrix.
<doc xml:space="preserve">#GST_TYPE_LIST, The channel mapping matrix.
The matrix coefficients must be between -1 and 1: the number of rows is equal
to the number of output channels and the number of columns is equal to the
@ -362,7 +362,7 @@ g_value_unset (&amp;v);
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD" value="GstAudioConverter.noise-shaping-method" c:type="GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD">
<doc xml:space="preserve">#GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
<doc xml:space="preserve">#GstAudioNoiseShapingMethod, The noise shaping method to use
to mask noise from quantization errors.
Default is #GST_AUDIO_NOISE_SHAPING_NONE.</doc>
@ -376,7 +376,7 @@ Default is 1</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="AUDIO_CONVERTER_OPT_RESAMPLER_METHOD" value="GstAudioConverter.resampler-method" c:type="GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD">
<doc xml:space="preserve">#GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
<doc xml:space="preserve">#GstAudioResamplerMethod, The resampler method to use when
changing sample rates.
Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc>
@ -957,7 +957,7 @@ See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION" value="GstAudioResampler.filter-interpolation" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION">
<doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
<doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be
interpolated.
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc>
@ -1219,6 +1219,10 @@ downstream specifies a range or a set of acceptable rates).</doc>
<property name="output-buffer-duration" writable="1" transfer-ownership="none">
<type name="guint64" c:type="guint64"/>
</property>
<property name="output-buffer-duration-fraction" version="1.18" writable="1" transfer-ownership="none">
<doc xml:space="preserve">Output block size in nanoseconds, expressed as a fraction.</doc>
<type name="Gst.Fraction"/>
</property>
<field name="parent">
<type name="GstBase.Aggregator" c:type="GstAggregator"/>
</field>
@ -2733,7 +2737,7 @@ This is expressed in caps by having a single channel and no channel mask.
This is expressed in caps by having a channel mask with no bits set.
As another special case it is allowed to have two channels without a channel mask.
This implicitely means that this is a stereo stream with a front left and front right
This implicitly means that this is a stereo stream with a front left and front right
channel.</doc>
<member name="none" value="-3" c:identifier="GST_AUDIO_CHANNEL_POSITION_NONE" glib:nick="none">
<doc xml:space="preserve">used for position-less channels, e.g.
@ -3043,7 +3047,7 @@ The object can perform conversion of:
<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_AUDIO_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">
@ -3306,7 +3310,7 @@ If the parameters in @config can not be set exactly, this function returns
%FALSE and will try to update as much state as possible. The new state can
then be retrieved and refined with gst_audio_converter_get_config().
Look at the #GST_AUDIO_CONVERTER_OPT_* fields to check valid configuration
Look at the `GST_AUDIO_CONVERTER_OPT_*` fields to check valid configuration
option and values.</doc>
<return-value transfer-ownership="none">
@ -3732,7 +3736,7 @@ invalidated by a call to this function.</doc>
<doc xml:space="preserve">a #GstAudioDecoder</doc>
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
</instance-parameter>
<parameter name="buf" transfer-ownership="none">
<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
<doc xml:space="preserve">decoded data</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
@ -3766,7 +3770,7 @@ invalidated by a call to this function.</doc>
<doc xml:space="preserve">a #GstAudioDecoder</doc>
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
</instance-parameter>
<parameter name="buf" transfer-ownership="none">
<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
<doc xml:space="preserve">decoded data</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
@ -5075,7 +5079,7 @@ may be invalidated by a call to this function.</doc>
<doc xml:space="preserve">a #GstAudioEncoder</doc>
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
</instance-parameter>
<parameter name="buffer" transfer-ownership="none">
<parameter name="buffer" transfer-ownership="full" nullable="1" allow-none="1">
<doc xml:space="preserve">encoded data</doc>
<type name="Gst.Buffer" c:type="GstBuffer*"/>
</parameter>
@ -6724,7 +6728,7 @@ meta as well as extracting it.</doc>
<member name="truncate_range" value="1" c:identifier="GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE" glib:nick="truncate-range">
<doc xml:space="preserve">When the source has a smaller depth
than the target format, set the least significant bits of the target
to 0. This is likely sightly faster but less accurate. When this flag
to 0. This is likely slightly faster but less accurate. When this flag
is not specified, the most significant bits of the source are duplicated
in the least significant bits of the destination.</doc>
</member>
@ -6900,7 +6904,7 @@ frames.</doc>
@in_frames are given to @resampler.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The number of frames that would be availabe after giving
<doc xml:space="preserve">The number of frames that would be available after giving
@in_frames as input to @resampler.</doc>
<type name="gsize" c:type="gsize"/>
</return-value>
@ -7080,11 +7084,11 @@ for @quality in @options.</doc>
</member>
<member name="linear" value="1" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR" glib:nick="linear">
<doc xml:space="preserve">linear interpolation of the
filter coeficients.</doc>
filter coefficients.</doc>
</member>
<member name="cubic" value="2" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC" glib:nick="cubic">
<doc xml:space="preserve">cubic interpolation of the
filter coeficients.</doc>
filter coefficients.</doc>
</member>
</enumeration>
<enumeration name="AudioResamplerFilterMode" glib:type-name="GstAudioResamplerFilterMode" glib:get-type="gst_audio_resampler_filter_mode_get_type" c:type="GstAudioResamplerFilterMode">
@ -7242,7 +7246,7 @@ FALSE on error.</doc>
</parameters>
</virtual-method>
<virtual-method name="clear_all" invoker="clear_all">
<doc xml:space="preserve">Fill the ringbuffer with silence.
<doc xml:space="preserve">Clear all samples from the ringbuffer.
MT safe.</doc>
@ -7335,11 +7339,11 @@ usually less than the segment size but can be bigger when the
implementation uses another internal buffer between the audio
device.
For playback ringbuffers this is the amount of samples transfered from the
For playback ringbuffers this is the amount of samples transferred from the
ringbuffer to the device but still not played.
For capture ringbuffers this is the amount of samples in the device that are
not yet transfered to the ringbuffer.</doc>
not yet transferred to the ringbuffer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The number of samples queued in the audio device.
@ -7531,7 +7535,7 @@ MT safe.</doc>
</parameters>
</method>
<method name="clear_all" c:identifier="gst_audio_ring_buffer_clear_all">
<doc xml:space="preserve">Fill the ringbuffer with silence.
<doc xml:space="preserve">Clear all samples from the ringbuffer.
MT safe.</doc>
@ -7655,11 +7659,11 @@ usually less than the segment size but can be bigger when the
implementation uses another internal buffer between the audio
device.
For playback ringbuffers this is the amount of samples transfered from the
For playback ringbuffers this is the amount of samples transferred from the
ringbuffer to the device but still not played.
For capture ringbuffers this is the amount of samples in the device that are
not yet transfered to the ringbuffer.</doc>
not yet transferred to the ringbuffer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The number of samples queued in the audio device.
@ -8548,6 +8552,17 @@ All scheduling of samples and timestamps is done in this base class
together with #GstAudioBaseSink using a default implementation of a
#GstAudioRingBuffer that uses threads.</doc>
<virtual-method name="clear_all">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="close">
<return-value transfer-ownership="none">
@ -8581,6 +8596,17 @@ together with #GstAudioBaseSink using a default implementation of a
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="pause">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="prepare">
<return-value transfer-ownership="none">
@ -8606,6 +8632,28 @@ together with #GstAudioBaseSink using a default implementation of a
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="resume">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="stop">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="unprepare">
<return-value transfer-ownership="none">
@ -8647,7 +8695,6 @@ together with #GstAudioBaseSink using a default implementation of a
</field>
</class>
<record name="AudioSinkClass" c:type="GstAudioSinkClass" glib:is-gtype-struct-for="AudioSink">
<doc xml:space="preserve">#GstAudioSink class. Override the vmethods to implement functionality.</doc>
<field name="parent_class">
<doc xml:space="preserve">the parent class structure.</doc>
@ -8753,8 +8800,60 @@ together with #GstAudioBaseSink using a default implementation of a
</parameters>
</callback>
</field>
<field name="pause">
<callback name="pause">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="resume">
<callback name="resume">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="stop">
<callback name="stop">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="clear_all">
<callback name="clear_all">
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="sink" transfer-ownership="none">
<type name="AudioSink" c:type="GstAudioSink*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" fixed-size="4">
<array zero-terminated="0" fixed-size="0">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -9235,7 +9334,7 @@ of the current one.</doc>
</method>
<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>
rate differs this implicitly marks the next data as discontinuous.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
@ -9458,13 +9557,6 @@ rate differs this implicitely marks the next data as discontinuous.</doc>
</parameter>
</parameters>
</function-macro>
<function-macro name="IS_STREAM_VOLUME" c:identifier="GST_IS_STREAM_VOLUME" introspectable="0">
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<constant name="META_TAG_AUDIO_CHANNELS_STR" value="channels" c:type="GST_META_TAG_AUDIO_CHANNELS_STR" version="1.2">
<doc xml:space="preserve">This metadata stays relevant as long as channels are unchanged.</doc>
@ -9480,17 +9572,10 @@ rate differs this implicitely marks the next data as discontinuous.</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<function-macro name="STREAM_VOLUME" c:identifier="GST_STREAM_VOLUME" introspectable="0">
<parameters>
<parameter name="obj">
</parameter>
</parameters>
</function-macro>
<function-macro name="STREAM_VOLUME_GET_INTERFACE" c:identifier="GST_STREAM_VOLUME_GET_INTERFACE" introspectable="0">
<parameters>
<parameter name="inst">
<parameter name="obj">
</parameter>
</parameters>
</function-macro>

263
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@ -180,8 +180,8 @@ gst_adapter_offset_at_discont(). The number of bytes that were consumed
since then can be queried with gst_adapter_distance_from_discont().
A last thing to note is that while #GstAdapter is pretty optimized,
merging buffers still might be an operation that requires a malloc() and
memcpy() operation, and these operations are not the fastest. Because of
merging buffers still might be an operation that requires a `malloc()` and
`memcpy()` operation, and these operations are not the fastest. Because of
this, some functions like gst_adapter_available_fast() are provided to help
speed up such cases should you want to. To avoid repeated memory allocations,
gst_adapter_copy() can be used to copy data into a (statically allocated)
@ -1120,6 +1120,22 @@ sent before pushing the buffer.</doc>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="negotiate" invoker="negotiate" version="1.18">
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if #GstAggregatorClass.negotiate() fails.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="self" transfer-ownership="none">
<doc xml:space="preserve">a #GstAggregator</doc>
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="negotiated_src_caps">
<return-value transfer-ownership="none">
@ -1171,6 +1187,23 @@ sent before pushing the buffer.</doc>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="sink_event_pre_queue">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<instance-parameter name="aggregator" transfer-ownership="none">
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
<parameter name="aggregator_pad" transfer-ownership="none">
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
</parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="sink_query">
<return-value transfer-ownership="none">
@ -1188,6 +1221,23 @@ sent before pushing the buffer.</doc>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="sink_query_pre_queue">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="aggregator" transfer-ownership="none">
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
<parameter name="aggregator_pad" transfer-ownership="none">
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
</parameter>
<parameter name="query" transfer-ownership="none">
<type name="Gst.Query" c:type="GstQuery*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="src_activate">
<return-value transfer-ownership="none">
@ -1349,6 +1399,22 @@ Typically only called by subclasses.</doc>
</instance-parameter>
</parameters>
</method>
<method name="negotiate" c:identifier="gst_aggregator_negotiate" version="1.18">
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if #GstAggregatorClass.negotiate() fails.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="self" transfer-ownership="none">
<doc xml:space="preserve">a #GstAggregator</doc>
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
</parameters>
</method>
<method name="set_latency" c:identifier="gst_aggregator_set_latency">
<doc xml:space="preserve">Lets #GstAggregator sub-classes tell the baseclass what their internal
latency is. Will also post a LATENCY message on the bus so the pipeline
@ -1390,7 +1456,7 @@ can reconfigure its global latency.</doc>
</parameters>
</method>
<method name="simple_get_next_time" c:identifier="gst_aggregator_simple_get_next_time" version="1.16">
<doc xml:space="preserve">This is a simple #GstAggregator::get_next_time implementation that
<doc xml:space="preserve">This is a simple #GstAggregatorClass.get_next_time() implementation that
just looks at the #GstSegment on the srcpad of the aggregator and bases
the next time on the running time there.
@ -1408,6 +1474,23 @@ and you have a dead line based aggregator subclass.</doc>
</instance-parameter>
</parameters>
</method>
<method name="update_segment" c:identifier="gst_aggregator_update_segment" version="1.18">
<doc xml:space="preserve">Subclasses should use this to update the segment on their
source pad, instead of directly pushing new segment events
downstream.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="self" transfer-ownership="none">
<type name="Aggregator" c:type="GstAggregator*"/>
</instance-parameter>
<parameter name="segment" transfer-ownership="none">
<type name="Gst.Segment" c:type="GstSegment*"/>
</parameter>
</parameters>
</method>
<property name="latency" writable="1" transfer-ownership="none">
<type name="guint64" c:type="guint64"/>
</property>
@ -1760,14 +1843,67 @@ _finish_buffer from inside that function.</doc>
</parameters>
</callback>
</field>
<field name="negotiate">
<callback name="negotiate">
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="self" transfer-ownership="none">
<doc xml:space="preserve">a #GstAggregator</doc>
<type name="Aggregator" c:type="GstAggregator*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="sink_event_pre_queue">
<callback name="sink_event_pre_queue">
<return-value transfer-ownership="none">
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
</return-value>
<parameters>
<parameter name="aggregator" transfer-ownership="none">
<type name="Aggregator" c:type="GstAggregator*"/>
</parameter>
<parameter name="aggregator_pad" transfer-ownership="none">
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
</parameter>
<parameter name="event" transfer-ownership="none">
<type name="Gst.Event" c:type="GstEvent*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="sink_query_pre_queue">
<callback name="sink_query_pre_queue">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="aggregator" transfer-ownership="none">
<type name="Aggregator" c:type="GstAggregator*"/>
</parameter>
<parameter name="aggregator_pad" transfer-ownership="none">
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
</parameter>
<parameter name="query" transfer-ownership="none">
<type name="Gst.Query" c:type="GstQuery*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" fixed-size="20">
<array zero-terminated="0" fixed-size="17">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
</record>
<class name="AggregatorPad" c:symbol-prefix="aggregator_pad" c:type="GstAggregatorPad" version="1.14" parent="Gst.Pad" glib:type-name="GstAggregatorPad" glib:get-type="gst_aggregator_pad_get_type" glib:type-struct="AggregatorPadClass">
<doc xml:space="preserve">Pads managed by a #GstAggregor subclass.
<doc xml:space="preserve">Pads managed by a #GstAggregator subclass.
This class used to live in gst-plugins-bad and was moved to core.</doc>
@ -2396,7 +2532,7 @@ a parser and share a lot of rather complex code.
* During the parsing process #GstBaseParseClass will handle both srcpad
and sinkpad events. They will be passed to subclass if
#GstBaseParseClass.event() or #GstBaseParseClass.src_event()
#GstBaseParseClass.sink_event() or #GstBaseParseClass.src_event()
implementations have been provided.
## Shutdown phase
@ -3950,6 +4086,25 @@ information about the render delay.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_stats" c:identifier="gst_base_sink_get_stats" version="1.18">
<doc xml:space="preserve">Return various #GstBaseSink statistics. This function returns a #GstStructure
with name `application/x-gst-base-sink-stats` with the following fields:
- "average-rate" G_TYPE_DOUBLE average frame rate
- "dropped" G_TYPE_UINT64 Number of dropped frames
- "rendered" G_TYPE_UINT64 Number of rendered frames</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">pointer to #GstStructure</doc>
<type name="Gst.Structure" c:type="GstStructure*"/>
</return-value>
<parameters>
<instance-parameter name="sink" transfer-ownership="none">
<doc xml:space="preserve">#GstBaseSink</doc>
<type name="BaseSink" c:type="GstBaseSink*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_sync" c:identifier="gst_base_sink_get_sync">
<doc xml:space="preserve">Checks if @sink is currently configured to synchronize against the
clock.</doc>
@ -4460,6 +4615,15 @@ media. This property will add additional latency to the device in order to
make other sinks compensate for the delay.</doc>
<type name="guint64" c:type="guint64"/>
</property>
<property name="stats" version="1.18" transfer-ownership="none">
<doc xml:space="preserve">Various #GstBaseSink statistics. This property returns a #GstStructure
with name `application/x-gst-base-sink-stats` with the following fields:
- "average-rate" G_TYPE_DOUBLE average frame rate
- "dropped" G_TYPE_UINT64 Number of dropped frames
- "rendered" G_TYPE_UINT64 Number of rendered frames</doc>
<type name="Gst.Structure"/>
</property>
<property name="sync" writable="1" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</property>
@ -5074,6 +5238,7 @@ implementation will call alloc and fill.</doc>
</parameters>
</virtual-method>
<virtual-method name="get_caps">
<doc xml:space="preserve">Called to get the caps to report.</doc>
<return-value transfer-ownership="full">
<type name="Gst.Caps" c:type="GstCaps*"/>
@ -5082,7 +5247,7 @@ implementation will call alloc and fill.</doc>
<instance-parameter name="src" transfer-ownership="none">
<type name="BaseSrc" c:type="GstBaseSrc*"/>
</instance-parameter>
<parameter name="filter" transfer-ownership="none">
<parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
@ -5134,13 +5299,22 @@ out. The base class will sync on the clock using these times.</doc>
</instance-parameter>
</parameters>
</virtual-method>
<virtual-method name="negotiate">
<virtual-method name="negotiate" invoker="negotiate" version="1.18">
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if #GstBaseSrcClass.negotiate() fails.
Do not call this in the #GstBaseSrcClass.fill() vmethod. Call this in
#GstBaseSrcClass.create() or in #GstBaseSrcClass.alloc(), _before_ any
buffer is allocated.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="src" transfer-ownership="none">
<doc xml:space="preserve">base source instance</doc>
<type name="BaseSrc" c:type="GstBaseSrc*"/>
</instance-parameter>
</parameters>
@ -5334,9 +5508,29 @@ by the src; unref it after usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="negotiate" c:identifier="gst_base_src_negotiate" version="1.18">
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if #GstBaseSrcClass.negotiate() fails.
Do not call this in the #GstBaseSrcClass.fill() vmethod. Call this in
#GstBaseSrcClass.create() or in #GstBaseSrcClass.alloc(), _before_ any
buffer is allocated.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="src" transfer-ownership="none">
<doc xml:space="preserve">base source instance</doc>
<type name="BaseSrc" c:type="GstBaseSrc*"/>
</instance-parameter>
</parameters>
</method>
<method name="new_seamless_segment" c:identifier="gst_base_src_new_seamless_segment">
<doc xml:space="preserve">Prepare a new seamless segment for emission downstream. This function must
only be called by derived sub-classes, and only from the create() function,
only be called by derived sub-classes, and only from the #GstBaseSrcClass::create function,
as the stream-lock needs to be held.
The format for the new segment will be the current format of the source, as
@ -5425,7 +5619,7 @@ when trying to read more should set this to %FALSE.
When @src operates in %GST_FORMAT_TIME, #GstBaseSrc will send an EOS
when a buffer outside of the currently configured segment is pushed if
@automatic_eos is %TRUE. Since 1.16, if @automatic_eos is %FALSE an
EOS will be pushed only when the #GstBaseSrc.create implementation
EOS will be pushed only when the #GstBaseSrcClass.create() implementation
returns %GST_FLOW_EOS.</doc>
<return-value transfer-ownership="none">
@ -5742,7 +5936,7 @@ buffers.</doc>
<parameter name="src" transfer-ownership="none">
<type name="BaseSrc" c:type="GstBaseSrc*"/>
</parameter>
<parameter name="filter" transfer-ownership="none">
<parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1">
<type name="Gst.Caps" c:type="GstCaps*"/>
</parameter>
</parameters>
@ -5752,10 +5946,12 @@ buffers.</doc>
<callback name="negotiate">
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="src" transfer-ownership="none">
<doc xml:space="preserve">base source instance</doc>
<type name="BaseSrc" c:type="GstBaseSrc*"/>
</parameter>
</parameters>
@ -6542,10 +6738,10 @@ It provides for:
</parameters>
</virtual-method>
<method name="get_allocator" c:identifier="gst_base_transform_get_allocator">
<doc xml:space="preserve">Lets #GstBaseTransform sub-classes to know the memory @allocator
<doc xml:space="preserve">Lets #GstBaseTransform sub-classes know the memory @allocator
used by the base class and its @params.
Unref the @allocator after use it.</doc>
Unref the @allocator after use.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
@ -6571,7 +6767,7 @@ used</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the instance of the #GstBufferPool used
by @trans; free it after use it</doc>
by @trans; free it after use</doc>
<type name="Gst.BufferPool" c:type="GstBufferPool*"/>
</return-value>
<parameters>
@ -6585,7 +6781,7 @@ by @trans; free it after use it</doc>
<doc xml:space="preserve">See if @trans is configured as a in_place transform.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE is the transform is configured in in_place mode.
<doc xml:space="preserve">%TRUE if the transform is configured in in_place mode.
MT safe.</doc>
<type name="gboolean" c:type="gboolean"/>
@ -6601,7 +6797,7 @@ MT safe.</doc>
<doc xml:space="preserve">See if @trans is configured as a passthrough transform.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE is the transform is configured in passthrough mode.
<doc xml:space="preserve">%TRUE if the transform is configured in passthrough mode.
MT safe.</doc>
<type name="gboolean" c:type="gboolean"/>
@ -6629,6 +6825,33 @@ MT safe.</doc>
</instance-parameter>
</parameters>
</method>
<method name="reconfigure" c:identifier="gst_base_transform_reconfigure" version="1.18">
<doc xml:space="preserve">Negotiates src pad caps with downstream elements if the source pad is
marked as needing reconfiguring. Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in
any case. But marks it again if negotiation fails.
Do not call this in the #GstBaseTransformClass.transform() or
#GstBaseTransformClass.transform_ip() vmethod. Call this in
#GstBaseTransformClass.submit_input_buffer(),
#GstBaseTransformClass.prepare_output_buffer() or in
#GstBaseTransformClass.generate_output() _before_ any output buffer is
allocated.
It will be default be called when handling an ALLOCATION query or at the
very beginning of the default #GstBaseTransformClass.submit_input_buffer()
implementation.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="trans" transfer-ownership="none">
<doc xml:space="preserve">the #GstBaseTransform to set</doc>
<type name="BaseTransform" c:type="GstBaseTransform*"/>
</instance-parameter>
</parameters>
</method>
<method name="reconfigure_sink" c:identifier="gst_base_transform_reconfigure_sink">
<doc xml:space="preserve">Instructs @trans to request renegotiation upstream. This function is
typically called after properties on the transform were set that
@ -6807,14 +7030,14 @@ running_time.</doc>
</parameters>
</method>
<method name="update_src_caps" c:identifier="gst_base_transform_update_src_caps" version="1.6">
<doc xml:space="preserve">Updates the srcpad caps and send the caps downstream. This function
<doc xml:space="preserve">Updates the srcpad caps and sends the caps downstream. This function
can be used by subclasses when they have already negotiated their caps
but found a change in them (or computed new information). This way,
they can notify downstream about that change without losing any
buffer.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the caps could be send downstream %FALSE otherwise</doc>
<doc xml:space="preserve">%TRUE if the caps could be sent downstream %FALSE otherwise</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -8047,7 +8270,7 @@ Free-function: gst_bit_writer_free</doc>
<function name="new_with_data" c:identifier="gst_bit_writer_new_with_data" introspectable="0">
<doc xml:space="preserve">Creates a new #GstBitWriter instance with the given memory area. If
@initialized is %TRUE it is possible to read @size bits from the
#GstBitWriter from the beginnig.
#GstBitWriter from the beginning.
Free-function: gst_bit_writer_free</doc>
@ -12855,7 +13078,7 @@ Free-function: gst_bit_writer_free</doc>
<function name="bit_writer_new_with_data" c:identifier="gst_bit_writer_new_with_data" moved-to="BitWriter.new_with_data" introspectable="0">
<doc xml:space="preserve">Creates a new #GstBitWriter instance with the given memory area. If
@initialized is %TRUE it is possible to read @size bits from the
#GstBitWriter from the beginnig.
#GstBitWriter from the beginning.
Free-function: gst_bit_writer_free</doc>

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@ -787,6 +787,27 @@ MT safe.</doc>
</instance-parameter>
</parameters>
</method>
<method name="pull_until_eos" c:identifier="gst_harness_pull_until_eos" version="1.18">
<doc xml:space="preserve">Pulls a #GstBuffer from the #GAsyncQueue on the #GstHarness sinkpad. The pull
will block until an EOS event is received, or timeout in 60 seconds.
MT safe.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE on success, %FALSE on timeout.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="h" transfer-ownership="none">
<doc xml:space="preserve">a #GstHarness</doc>
<type name="Harness" c:type="GstHarness*"/>
</instance-parameter>
<parameter name="buf" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">A #GstBuffer, or %NULL if EOS or timeout occures
first.</doc>
<type name="Gst.Buffer" c:type="GstBuffer**"/>
</parameter>
</parameters>
</method>
<method name="pull_upstream_event" c:identifier="gst_harness_pull_upstream_event" version="1.6">
<doc xml:space="preserve">Pulls an #GstEvent from the #GAsyncQueue on the #GstHarness srcpad.
Timeouts after 60 seconds similar to gst_harness_pull.
@ -2325,7 +2346,8 @@ MT safe.</doc>
<method name="crank" c:identifier="gst_test_clock_crank" version="1.8">
<doc xml:space="preserve">A "crank" consists of three steps:
1: Wait for a #GstClockID to be registered with the #GstTestClock.
2: Advance the #GstTestClock to the time the #GstClockID is waiting for.
2: Advance the #GstTestClock to the time the #GstClockID is waiting, unless
the clock time is already passed the clock id (Since 1.18).
3: Release the #GstClockID wait.
A "crank" can be though of as the notion of
manually driving the clock forward to its next logical step.</doc>
@ -2423,6 +2445,25 @@ notification to look for</doc>
</parameter>
</parameters>
</method>
<method name="process_id" c:identifier="gst_test_clock_process_id" version="1.18">
<doc xml:space="preserve">Processes and releases the pending ID.
MT safe.</doc>
<return-value transfer-ownership="none">
<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 process the pending IDs</doc>
<type name="TestClock" c:type="GstTestClock*"/>
</instance-parameter>
<parameter name="pending_id" transfer-ownership="full">
<doc xml:space="preserve">#GstClockID</doc>
<type name="Gst.ClockID" c:type="GstClockID"/>
</parameter>
</parameters>
</method>
<method name="process_id_list" c:identifier="gst_test_clock_process_id_list" version="1.4">
<doc xml:space="preserve">Processes and releases the pending IDs in the list.

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@ -152,6 +152,12 @@ Consult the relevant specifications for more details.</doc>
<type name="Descriptor"/>
</array>
</field>
<constructor name="new" c:identifier="gst_mpegts_atsc_mgt_new">
<return-value transfer-ownership="full">
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
</return-value>
</constructor>
</record>
<record name="AtscMGTTable" c:type="GstMpegtsAtscMGTTable" glib:type-name="GstMpegtsAtscMGTTable" glib:get-type="gst_mpegts_atsc_mgt_table_get_type" c:symbol-prefix="atsc_mgt_table">
<doc xml:space="preserve">Source from a @GstMpegtsAtscMGT</doc>
@ -203,6 +209,88 @@ Consult the relevant specifications for more details.</doc>
</array>
</field>
</record>
<record name="AtscRRT" c:type="GstMpegtsAtscRRT" version="1.18" glib:type-name="GstMpegtsAtscRRT" glib:get-type="gst_mpegts_atsc_rrt_get_type" c:symbol-prefix="atsc_rrt">
<doc xml:space="preserve">Region Rating 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="names" writable="1">
<doc xml:space="preserve">the names</doc>
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="AtscMultString"/>
</array>
</field>
<field name="dimensions_defined" writable="1">
<doc xml:space="preserve">the number of dimensions defined for this rating table</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="dimensions" writable="1">
<doc xml:space="preserve">A set of dimensions</doc>
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="AtscRRTDimension"/>
</array>
</field>
<field name="descriptors" writable="1">
<doc xml:space="preserve">descriptors</doc>
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_new">
<return-value transfer-ownership="full">
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
</return-value>
</constructor>
</record>
<record name="AtscRRTDimension" c:type="GstMpegtsAtscRRTDimension" glib:type-name="GstMpegtsAtscRRTDimension" glib:get-type="gst_mpegts_atsc_rrt_dimension_get_type" c:symbol-prefix="atsc_rrt_dimension">
<field name="names" writable="1">
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<field name="graduated_scale" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="values_defined" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="values" writable="1">
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_dimension_new">
<return-value transfer-ownership="full">
<type name="AtscRRTDimension" c:type="GstMpegtsAtscRRTDimension*"/>
</return-value>
</constructor>
</record>
<record name="AtscRRTDimensionValue" c:type="GstMpegtsAtscRRTDimensionValue" version="1.18" glib:type-name="GstMpegtsAtscRRTDimensionValue" glib:get-type="gst_mpegts_atsc_rrt_dimension_value_get_type" c:symbol-prefix="atsc_rrt_dimension_value">
<field name="abbrev_ratings" writable="1">
<doc xml:space="preserve">the abbreviated ratings</doc>
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="AtscMultString"/>
</array>
</field>
<field name="ratings" writable="1">
<doc xml:space="preserve">the ratings</doc>
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="AtscMultString"/>
</array>
</field>
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_dimension_value_new">
<return-value transfer-ownership="full">
<type name="AtscRRTDimensionValue" c:type="GstMpegtsAtscRRTDimensionValue*"/>
</return-value>
</constructor>
</record>
<record name="AtscSTT" c:type="GstMpegtsAtscSTT" glib:type-name="GstMpegtsAtscSTT" glib:get-type="gst_mpegts_atsc_stt_get_type" c:symbol-prefix="atsc_stt">
<doc xml:space="preserve">System Time Table (A65)</doc>
@ -239,6 +327,12 @@ Consult the relevant specifications for more details.</doc>
<doc xml:space="preserve">The UTC date and time</doc>
<type name="Gst.DateTime" c:type="GstDateTime*"/>
</field>
<constructor name="new" c:identifier="gst_mpegts_atsc_stt_new">
<return-value transfer-ownership="full">
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
</return-value>
</constructor>
<method name="get_datetime_utc" c:identifier="gst_mpegts_atsc_stt_get_datetime_utc">
<return-value transfer-ownership="full">
@ -284,6 +378,26 @@ Consult the relevant specifications for more details.</doc>
</instance-parameter>
</parameters>
</method>
<method name="set_string" c:identifier="gst_mpegts_atsc_string_segment_set_string">
<return-value transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="seg" transfer-ownership="none">
<type name="AtscStringSegment" c:type="GstMpegtsAtscStringSegment*"/>
</instance-parameter>
<parameter name="string" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</parameter>
<parameter name="compression_type" transfer-ownership="none">
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="mode" transfer-ownership="none">
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</method>
</record>
<record name="AtscVCT" c:type="GstMpegtsAtscVCT" glib:type-name="GstMpegtsAtscVCT" glib:get-type="gst_mpegts_atsc_vct_get_type" c:symbol-prefix="atsc_vct">
<doc xml:space="preserve">Represents both:
@ -739,6 +853,8 @@ Consult the relevant specifications for more details.</doc>
</member>
<member name="uri_linkage" value="19" c:identifier="GST_MTS_DESC_EXT_DVB_URI_LINKAGE">
</member>
<member name="ac4" value="21" c:identifier="GST_MTS_DESC_EXT_DVB_AC4">
</member>
</enumeration>
<record name="DVBLinkageDescriptor" c:type="GstMpegtsDVBLinkageDescriptor" glib:type-name="GstMpegtsDVBLinkageDescriptor" glib:get-type="gst_mpegts_dvb_linkage_descriptor_get_type" c:symbol-prefix="dvb_linkage_descriptor">
@ -755,14 +871,14 @@ Consult the relevant specifications for more details.</doc>
<type name="guint16" c:type="guint16"/>
</field>
<field name="linkage_type" writable="1">
<doc xml:space="preserve">the type which %linkage_data has</doc>
<doc xml:space="preserve">the type which @linkage_data has</doc>
<type name="DVBLinkageType" c:type="GstMpegtsDVBLinkageType"/>
</field>
<field name="linkage_data" readable="0" private="1">
<type name="gpointer" c:type="gpointer"/>
</field>
<field name="private_data_length" writable="1">
<doc xml:space="preserve">the length for %private_data_bytes</doc>
<doc xml:space="preserve">the length for @private_data_bytes</doc>
<type name="guint8" c:type="guint8"/>
</field>
<field name="private_data_bytes" writable="1">
@ -1090,7 +1206,7 @@ As specified in Table 100 of ETSI EN 300 468 v1.13.1</doc>
<doc xml:space="preserve">These are the base descriptor types and methods.
For more details, refer to the ITU H.222.0 or ISO/IEC 13818-1 specifications
and other specifications mentionned in the documentation.</doc>
and other specifications mentioned in the documentation.</doc>
<field name="tag" writable="1">
<doc xml:space="preserve">the type of descriptor</doc>
@ -1295,7 +1411,7 @@ are found in http://www.dvbservices.com</doc>
</array>
</parameter>
<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">the length of #id_selector_bytes</doc>
<doc xml:space="preserve">the length of @id_selector_bytes</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
</parameters>
@ -1336,7 +1452,7 @@ are found in http://www.dvbservices.com</doc>
</parameter>
<parameter name="list" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">a list of all frequencies in Hz/kHz
depending on %offset</doc>
depending on @offset</doc>
<array name="GLib.Array" c:type="GArray**">
<type name="guint32"/>
</array>
@ -1516,7 +1632,7 @@ registered by http://www.dvbservices.com/</doc>
</array>
</parameter>
<parameter name="length" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1">
<doc xml:space="preserve">length of %private_data</doc>
<doc xml:space="preserve">length of @private_data</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
</parameters>
@ -1620,7 +1736,7 @@ the list of services</doc>
<doc xml:space="preserve">Extracts the component tag from @descriptor.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if the parsing happended correctly, else %FALSE.</doc>
<doc xml:space="preserve">%TRUE if the parsing happened correctly, else %FALSE.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -2643,6 +2759,135 @@ Corresponds to table 6 of ETSI EN 300 468 (v1.13.0)</doc>
<member name="off_air" value="5" c:identifier="GST_MPEGTS_RUNNING_STATUS_OFF_AIR">
</member>
</enumeration>
<record name="SCTESIT" c:type="GstMpegtsSCTESIT" glib:type-name="GstMpegtsSCTESIT" glib:get-type="gst_mpegts_scte_sit_get_type" c:symbol-prefix="scte_sit">
<field name="encrypted_packet" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="encryption_algorithm" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="pts_adjustment" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="cw_index" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="tier" writable="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="splice_command_length" writable="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="splice_command_type" writable="1">
<type name="SCTESpliceCommandType" c:type="GstMpegtsSCTESpliceCommandType"/>
</field>
<field name="splice_time_specified" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="splice_time" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="splices" writable="1">
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<field name="descriptors" writable="1">
<array name="GLib.PtrArray" c:type="GPtrArray*">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
<constructor name="new" c:identifier="gst_mpegts_scte_sit_new">
<doc xml:space="preserve">Allocates and initializes a #GstMpegtsSCTESIT.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</return-value>
</constructor>
</record>
<enumeration name="SCTESpliceCommandType" c:type="GstMpegtsSCTESpliceCommandType">
<member name="null" value="0" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_NULL">
</member>
<member name="schedule" value="4" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_SCHEDULE">
</member>
<member name="insert" value="5" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_INSERT">
</member>
<member name="time" value="6" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_TIME">
</member>
<member name="bandwidth" value="7" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_BANDWIDTH">
</member>
<member name="private" value="255" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_PRIVATE">
</member>
</enumeration>
<enumeration name="SCTESpliceDescriptor" c:type="GstMpegtsSCTESpliceDescriptor">
<member name="avail" value="0" c:identifier="GST_MTS_SCTE_DESC_AVAIL">
</member>
<member name="dtmf" value="1" c:identifier="GST_MTS_SCTE_DESC_DTMF">
</member>
<member name="segmentation" value="2" c:identifier="GST_MTS_SCTE_DESC_SEGMENTATION">
</member>
<member name="time" value="3" c:identifier="GST_MTS_SCTE_DESC_TIME">
</member>
<member name="audio" value="4" c:identifier="GST_MTS_SCTE_DESC_AUDIO">
</member>
</enumeration>
<record name="SCTESpliceEvent" c:type="GstMpegtsSCTESpliceEvent" glib:type-name="GstMpegtsSCTESpliceEvent" glib:get-type="gst_mpegts_scte_splice_event_get_type" c:symbol-prefix="scte_splice_event">
<field name="insert_event" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="splice_event_id" writable="1">
<type name="guint32" c:type="guint32"/>
</field>
<field name="splice_event_cancel_indicator" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="out_of_network_indicator" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="program_splice_flag" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="duration_flag" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="splice_immediate_flag" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="program_splice_time_specified" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="program_splice_time" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="break_duration_auto_return" writable="1">
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="break_duration" writable="1">
<type name="guint64" c:type="guint64"/>
</field>
<field name="unique_program_id" writable="1">
<type name="guint16" c:type="guint16"/>
</field>
<field name="avail_num" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<field name="avails_expected" writable="1">
<type name="guint8" c:type="guint8"/>
</field>
<constructor name="new" c:identifier="gst_mpegts_scte_splice_event_new">
<doc xml:space="preserve">Allocates and initializes a #GstMpegtsSCTESpliceEvent.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESpliceEvent</doc>
<type name="SCTESpliceEvent" c:type="GstMpegtsSCTESpliceEvent*"/>
</return-value>
</constructor>
</record>
<record name="SDT" c:type="GstMpegtsSDT" glib:type-name="GstMpegtsSDT" glib:get-type="gst_mpegts_sdt_get_type" c:symbol-prefix="sdt">
<doc xml:space="preserve">Service Description Table (EN 300 468)</doc>
@ -2798,6 +3043,9 @@ else in the western part.</doc>
<member name="isoch_data" value="131" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_ISOCH_DATA">
<doc xml:space="preserve">SCTE-19 Isochronous data</doc>
</member>
<member name="sit" value="134" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_SIT">
<doc xml:space="preserve">SCTE-35 Splice Information Table</doc>
</member>
<member name="dst_nrt" value="149" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_DST_NRT">
<doc xml:space="preserve">SCTE-07 Data Service or
Network Resource Table</doc>
@ -2980,6 +3228,21 @@ happened.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_atsc_rrt" c:identifier="gst_mpegts_section_get_atsc_rrt" version="1.18">
<doc xml:space="preserve">Returns the #GstMpegtsAtscRRT contained in the @section.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The #GstMpegtsAtscRRT contained in the section, or %NULL if an error
happened.</doc>
<type name="AtscRRT" c:type="const GstMpegtsAtscRRT*"/>
</return-value>
<parameters>
<instance-parameter name="section" transfer-ownership="none">
<doc xml:space="preserve">a #GstMpegtsSection of type %GST_MPEGTS_SECTION_ATSC_RRT</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_atsc_stt" c:identifier="gst_mpegts_section_get_atsc_stt">
<doc xml:space="preserve">Returns the #GstMpegtsAtscSTT contained in the @section.</doc>
@ -3126,6 +3389,21 @@ happened.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_scte_sit" c:identifier="gst_mpegts_section_get_scte_sit">
<doc xml:space="preserve">Returns the #GstMpegtsSCTESIT contained in the @section.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">The #GstMpegtsSCTESIT contained in the section, or %NULL if an error
happened.</doc>
<type name="SCTESIT" c:type="const GstMpegtsSCTESIT*"/>
</return-value>
<parameters>
<instance-parameter name="section" transfer-ownership="none">
<doc xml:space="preserve">a #GstMpegtsSection of type %GST_MPEGTS_SECTION_SCTE_SIT</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_sdt" c:identifier="gst_mpegts_section_get_sdt">
<doc xml:space="preserve">Returns the #GstMpegtsSDT contained in the @section.</doc>
@ -3227,6 +3505,41 @@ The #GstEvent is sent to the @element #GstElement.</doc>
</parameter>
</parameters>
</method>
<function name="from_atsc_mgt" c:identifier="gst_mpegts_section_from_atsc_mgt" version="1.18">
<return-value transfer-ownership="full">
<doc xml:space="preserve">the #GstMpegtsSection</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="mgt" transfer-ownership="full">
<doc xml:space="preserve">a #GstMpegtsAtscMGT to create the #GstMpegtsSection from</doc>
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
</parameter>
</parameters>
</function>
<function name="from_atsc_rrt" c:identifier="gst_mpegts_section_from_atsc_rrt">
<return-value transfer-ownership="full">
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="rrt" transfer-ownership="none">
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
</parameter>
</parameters>
</function>
<function name="from_atsc_stt" c:identifier="gst_mpegts_section_from_atsc_stt">
<return-value transfer-ownership="full">
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="stt" transfer-ownership="none">
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
</parameter>
</parameters>
</function>
<function name="from_nit" c:identifier="gst_mpegts_section_from_nit">
<doc xml:space="preserve">Ownership of @nit is taken. The data in @nit is managed by the #GstMpegtsSection</doc>
@ -3279,6 +3592,23 @@ The #GstEvent is sent to the @element #GstElement.</doc>
</parameter>
</parameters>
</function>
<function name="from_scte_sit" c:identifier="gst_mpegts_section_from_scte_sit">
<doc xml:space="preserve">Ownership of @sit is taken. The data in @sit is managed by the #GstMpegtsSection</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the #GstMpegtsSection</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="sit" transfer-ownership="full">
<doc xml:space="preserve">a #GstMpegtsSCTESIT to create the #GstMpegtsSection from</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</parameter>
<parameter name="pid" transfer-ownership="none">
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="from_sdt" c:identifier="gst_mpegts_section_from_sdt">
<doc xml:space="preserve">Ownership of @sdt is taken. The data in @sdt is managed by the #GstMpegtsSection</doc>
@ -3544,6 +3874,11 @@ see also #GstMpegtsSectionATSCTableID, #GstMpegtsSectionDVBTableID, and
<member name="atsc_stt" value="16" c:identifier="GST_MPEGTS_SECTION_ATSC_STT">
<doc xml:space="preserve">ATSC System Time Table (A65)</doc>
</member>
<member name="atsc_rrt" value="17" c:identifier="GST_MPEGTS_SECTION_ATSC_RRT">
</member>
<member name="scte_sit" value="18" c:identifier="GST_MPEGTS_SECTION_SCTE_SIT">
<doc xml:space="preserve">SCTE Splice Information Table (SCTE-35)</doc>
</member>
</enumeration>
<enumeration name="StreamType" c:type="GstMpegtsStreamType">
<doc xml:space="preserve">Type of mpeg-ts stream type.
@ -3803,7 +4138,7 @@ profiles defined in Annex A for service-compatible stereoscopic 3D services</doc
<type name="gboolean" c:type="gboolean"/>
</field>
<field name="constellation" writable="1">
<doc xml:space="preserve">the constallation</doc>
<doc xml:space="preserve">the constellation</doc>
<type name="ModulationType" c:type="GstMpegtsModulationType"/>
</field>
<field name="hierarchy" writable="1">
@ -4073,7 +4408,7 @@ Note: To look for descriptors that can be present more than once in an
array of descriptors, iterate the #GArray manually.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the first descriptor matchin @tag, else %NULL.</doc>
<doc xml:space="preserve">the first descriptor matching @tag, else %NULL.</doc>
<type name="Descriptor" c:type="const GstMpegtsDescriptor*"/>
</return-value>
<parameters>
@ -4166,6 +4501,115 @@ Release with #g_array_unref when done with it.</doc>
</array>
</return-value>
</function>
<function name="scte_cancel_new" c:identifier="gst_mpegts_scte_cancel_new">
<doc xml:space="preserve">Allocates and initializes a new INSERT command #GstMpegtsSCTESIT
setup to cancel the specified @event_id.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</return-value>
<parameters>
<parameter name="event_id" transfer-ownership="none">
<doc xml:space="preserve">The event ID to cancel.</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
</parameters>
</function>
<function name="scte_null_new" c:identifier="gst_mpegts_scte_null_new">
<doc xml:space="preserve">Allocates and initializes a NULL command #GstMpegtsSCTESIT.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</return-value>
</function>
<function name="scte_splice_in_new" c:identifier="gst_mpegts_scte_splice_in_new">
<doc xml:space="preserve">Allocates and initializes a new "Splice In" INSERT command
#GstMpegtsSCTESIT for the given @event_id and @splice_time.
If the @splice_time is #G_MAXUINT64 then the event will be
immediate as opposed to for the target @splice_time.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</return-value>
<parameters>
<parameter name="event_id" transfer-ownership="none">
<doc xml:space="preserve">The event ID.</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="splice_time" transfer-ownership="none">
<doc xml:space="preserve">The PCR time for the splice event</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
</parameters>
</function>
<function name="scte_splice_out_new" c:identifier="gst_mpegts_scte_splice_out_new">
<doc xml:space="preserve">Allocates and initializes a new "Splice Out" INSERT command
#GstMpegtsSCTESIT for the given @event_id, @splice_time and
duration.
If the @splice_time is #G_MAXUINT64 then the event will be
immediate as opposed to for the target @splice_time.
If the @duration is 0 it won't be specified in the event.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</return-value>
<parameters>
<parameter name="event_id" transfer-ownership="none">
<doc xml:space="preserve">The event ID.</doc>
<type name="guint32" c:type="guint32"/>
</parameter>
<parameter name="splice_time" transfer-ownership="none">
<doc xml:space="preserve">The PCR time for the splice event</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
<parameter name="duration" transfer-ownership="none">
<doc xml:space="preserve">The optional duration.</doc>
<type name="guint64" c:type="guint64"/>
</parameter>
</parameters>
</function>
<function name="section_from_atsc_mgt" c:identifier="gst_mpegts_section_from_atsc_mgt" moved-to="Section.from_atsc_mgt" version="1.18">
<return-value transfer-ownership="full">
<doc xml:space="preserve">the #GstMpegtsSection</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="mgt" transfer-ownership="full">
<doc xml:space="preserve">a #GstMpegtsAtscMGT to create the #GstMpegtsSection from</doc>
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
</parameter>
</parameters>
</function>
<function name="section_from_atsc_rrt" c:identifier="gst_mpegts_section_from_atsc_rrt" moved-to="Section.from_atsc_rrt">
<return-value transfer-ownership="full">
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="rrt" transfer-ownership="none">
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
</parameter>
</parameters>
</function>
<function name="section_from_atsc_stt" c:identifier="gst_mpegts_section_from_atsc_stt" moved-to="Section.from_atsc_stt">
<return-value transfer-ownership="full">
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="stt" transfer-ownership="none">
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
</parameter>
</parameters>
</function>
<function name="section_from_nit" c:identifier="gst_mpegts_section_from_nit" moved-to="Section.from_nit">
<doc xml:space="preserve">Ownership of @nit is taken. The data in @nit is managed by the #GstMpegtsSection</doc>
@ -4218,6 +4662,23 @@ Release with #g_array_unref when done with it.</doc>
</parameter>
</parameters>
</function>
<function name="section_from_scte_sit" c:identifier="gst_mpegts_section_from_scte_sit" moved-to="Section.from_scte_sit">
<doc xml:space="preserve">Ownership of @sit is taken. The data in @sit is managed by the #GstMpegtsSection</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the #GstMpegtsSection</doc>
<type name="Section" c:type="GstMpegtsSection*"/>
</return-value>
<parameters>
<parameter name="sit" transfer-ownership="full">
<doc xml:space="preserve">a #GstMpegtsSCTESIT to create the #GstMpegtsSection from</doc>
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
</parameter>
<parameter name="pid" transfer-ownership="none">
<type name="guint16" c:type="guint16"/>
</parameter>
</parameters>
</function>
<function name="section_from_sdt" c:identifier="gst_mpegts_section_from_sdt" moved-to="Section.from_sdt">
<doc xml:space="preserve">Ownership of @sdt is taken. The data in @sdt is managed by the #GstMpegtsSection</doc>

20
gir-files/GstNet-1.0.gir
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@ -159,7 +159,7 @@ send @GST_MESSAGE_ELEMENT messages with an attached #GstStructure containing
statistics about clock accuracy and network traffic.</doc>
<constructor name="new" c:identifier="gst_net_client_clock_new">
<doc xml:space="preserve">Create a new #GstNetClientInternalClock that will report the time
<doc xml:space="preserve">Create a new #GstNetClientClock that will report the time
provided by the #GstNetTimeProvider on @remote_address and
@remote_port.</doc>
@ -799,6 +799,24 @@ calls, but otherwise returns NULL on error.</doc>
</parameter>
</parameters>
</function>
<function name="net_utils_set_socket_tos" c:identifier="gst_net_utils_set_socket_tos" version="1.18">
<doc xml:space="preserve">Configures IP_TOS value of socket, i.e. sets QoS DSCP.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">TRUE if successful, FALSE in case an error occurred.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="socket" transfer-ownership="none">
<doc xml:space="preserve">Socket to configure</doc>
<type name="Gio.Socket" c:type="GSocket*"/>
</parameter>
<parameter name="qos_dscp" transfer-ownership="none">
<doc xml:space="preserve">QoS DSCP value</doc>
<type name="gint" c:type="gint"/>
</parameter>
</parameters>
</function>
<function name="ptp_deinit" c:identifier="gst_ptp_deinit" version="1.6">
<doc xml:space="preserve">Deinitialize the GStreamer PTP subsystem and stop the PTP clock. If there
are any remaining GstPtpClock instances, they won't be further synchronized

56
gir-files/GstPbutils-1.0.gir
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@ -65,7 +65,7 @@ audio-rate to video-rate and handles renegotiation (downstream video size
changes).
It also provides several background shading effects. These effects are
applied to a previous picture before the render() implementation can draw a
applied to a previous picture before the `render()` implementation can draw a
new frame.</doc>
<virtual-method name="decide_allocation">
@ -849,7 +849,7 @@ gst_discoverer_stream_info_list_free().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">An array of strings
containing informations about how to install the various missing elements
containing information about how to install the various missing elements
for @info to be usable. If you wish to use the strings after the life-time
of @info, you will need to copy them.</doc>
<array c:type="const gchar**">
@ -1807,6 +1807,21 @@ Can be %NULL. Unref after usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_single_segment" c:identifier="gst_encoding_profile_get_single_segment" version="1.18">
<return-value transfer-ownership="none">
<doc xml:space="preserve">#TRUE if the stream represented by @profile should use a single
segment before the encoder, #FALSE otherwise. This means that buffers will be retimestamped
and segments will be eat so as to appear as one segment.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="profile" transfer-ownership="none">
<doc xml:space="preserve">a #GstEncodingProfile</doc>
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_type_nick" c:identifier="gst_encoding_profile_get_type_nick">
<return-value transfer-ownership="none">
@ -1898,7 +1913,7 @@ during the encoding</doc>
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
</instance-parameter>
<parameter name="enabled" transfer-ownership="none">
<doc xml:space="preserve">%FALSE to disable #profile, %TRUE to enable it</doc>
<doc xml:space="preserve">%FALSE to disable @profile, %TRUE to enable it</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
@ -2011,6 +2026,25 @@ for more about restrictions. Does not apply to #GstEncodingContainerProfile.</do
</parameter>
</parameters>
</method>
<method name="set_single_segment" c:identifier="gst_encoding_profile_set_single_segment" version="1.18">
<doc xml:space="preserve">If using a single segment, buffers will be retimestamped
and segments will be eat so as to appear as one segment.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="profile" transfer-ownership="none">
<doc xml:space="preserve">a #GstEncodingProfile</doc>
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
</instance-parameter>
<parameter name="single_segment" transfer-ownership="none">
<doc xml:space="preserve">#TRUE if the stream represented by @profile should use a single
segment before the encoder #FALSE otherwise.</doc>
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</method>
<property name="restriction-caps" writable="1" transfer-ownership="none">
<type name="Gst.Caps"/>
</property>
@ -2030,9 +2064,9 @@ The name and category can only consist of lowercase ASCII letters for the
first character, followed by either lowercase ASCII letters, digits or
hyphens ('-').
The @category &lt;emphasis&gt;should&lt;/emphasis&gt; be one of the existing
The @category *should* be one of the existing
well-defined categories, like #GST_ENCODING_CATEGORY_DEVICE, but it
&lt;emphasis&gt;can&lt;/emphasis&gt; be a application or user specific category if
*can* be a application or user specific category if
needed.</doc>
<return-value transfer-ownership="full">
@ -2641,7 +2675,7 @@ program and to what extent the requested plugins could be installed.</doc>
some (but not all) of the requested plugins could be installed.</doc>
</member>
<member name="error" value="2" c:identifier="GST_INSTALL_PLUGINS_ERROR" glib:nick="error">
<doc xml:space="preserve">an error occured during the installation. If
<doc xml:space="preserve">an error occurred during the installation. If
this happens, the user has already seen an error message and another
one should not be displayed</doc>
</member>
@ -2666,7 +2700,7 @@ program and to what extent the requested plugins could be installed.</doc>
</member>
<member name="internal_failure" value="201" c:identifier="GST_INSTALL_PLUGINS_INTERNAL_FAILURE" glib:nick="internal-failure">
<doc xml:space="preserve">some internal failure has
occured when trying to start the installer</doc>
occurred when trying to start the installer</doc>
</member>
<member name="helper_missing" value="202" c:identifier="GST_INSTALL_PLUGINS_HELPER_MISSING" glib:nick="helper-missing">
<doc xml:space="preserve">the helper script to call the
@ -2706,17 +2740,17 @@ in debugging.</doc>
<type name="gint" c:type="gint"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_MICRO" value="2" c:type="GST_PLUGINS_BASE_VERSION_MICRO">
<constant name="PLUGINS_BASE_VERSION_MICRO" 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="16" c:type="GST_PLUGINS_BASE_VERSION_MINOR">
<constant name="PLUGINS_BASE_VERSION_MINOR" value="17" 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"/>
</constant>
<constant name="PLUGINS_BASE_VERSION_NANO" value="0" c:type="GST_PLUGINS_BASE_VERSION_NANO">
<constant name="PLUGINS_BASE_VERSION_NANO" value="1" c:type="GST_PLUGINS_BASE_VERSION_NANO">
<doc xml:space="preserve">The nano version of GStreamer's gst-plugins-base libraries at compile time.
Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
@ -3063,7 +3097,7 @@ with bit 0 being the most significant bit of the first byte.
* Bit 41 - general_interlaced_source_flag
* Bit 42 - general_non_packed_constraint_flag
* Bit 43 - general_frame_only_constraint_flag
* Bit 44:87 - general_reserved_zero_44bits
* Bit 44:87 - See below
* Bit 88:95 - general_level_idc</doc>
<return-value transfer-ownership="none">

8
gir-files/GstPlayer-1.0.gir
View file

@ -1073,7 +1073,7 @@ value.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE or %FALSE
Sets the subtitle strack @stream_index.</doc>
Sets the subtitle stack @stream_index.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
@ -1567,7 +1567,7 @@ gain.</doc>
<implements name="PlayerSignalDispatcher"/>
<function name="new" c:identifier="gst_player_g_main_context_signal_dispatcher_new">
<doc xml:space="preserve">Creates a new GstPlayerSignalDispatcher that uses @application_context,
or the thread default one if %NULL is used. See gst_player_new_full().</doc>
or the thread default one if %NULL is used. See gst_player_new().</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">the new GstPlayerSignalDispatcher</doc>
@ -1634,7 +1634,7 @@ matching #GstPlayerAudioInfo.</doc>
</method>
<method name="get_image_sample" c:identifier="gst_player_media_info_get_image_sample">
<doc xml:space="preserve">Function to get the image (or preview-image) stored in taglist.
Application can use gst_sample_*_() API's to get caps, buffer etc.</doc>
Application can use `gst_sample_*_()` API's to get caps, buffer etc.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">GstSample or NULL.</doc>
@ -1927,7 +1927,7 @@ stream.</doc>
</function>
</enumeration>
<class name="PlayerStreamInfo" c:symbol-prefix="player_stream_info" c:type="GstPlayerStreamInfo" parent="GObject.Object" abstract="1" glib:type-name="GstPlayerStreamInfo" glib:get-type="gst_player_stream_info_get_type" glib:type-struct="PlayerStreamInfoClass">
<doc xml:space="preserve">Base structure for information concering a media stream. Depending on
<doc xml:space="preserve">Base structure for information concerning a media stream. Depending on
the stream type, one can find more media-specific information in
#GstPlayerVideoInfo, #GstPlayerAudioInfo, #GstPlayerSubtitleInfo.</doc>

122
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@ -321,6 +321,8 @@ gst_rtcp_buffer_validate_reduced().</doc>
<member name="rtpfb_type_rtcp_sr_req" value="5" c:identifier="GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ" glib:nick="rtpfb-type-rtcp-sr-req">
<doc xml:space="preserve">Request an SR packet for early
synchronization</doc>
</member>
<member name="rtpfb_type_twcc" value="15" c:identifier="GST_RTCP_RTPFB_TYPE_TWCC" glib:nick="rtpfb-type-twcc">
</member>
<member name="psfb_type_pli" value="1" c:identifier="GST_RTCP_PSFB_TYPE_PLI" glib:nick="psfb-type-pli">
<doc xml:space="preserve">Picture Loss Indication</doc>
@ -2096,7 +2098,7 @@ RTP packets always contain full frames.
To use this base class, your child element needs to call either
gst_rtp_base_audio_payload_set_frame_based() or
gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
element's _init() function. Then, the child element must call either
element's `_init()` function. Then, the child element must call either
gst_rtp_base_audio_payload_set_frame_options(),
gst_rtp_base_audio_payload_set_sample_options() or
gst_rtp_base_audio_payload_set_samplebits_options. Since
@ -2393,7 +2395,7 @@ audio codec</doc>
<doc xml:space="preserve">Push @out_buf to the peer of @filter. This function takes ownership of
@out_buf.
This function will by default apply the last incomming timestamp on
This function will by default apply the last incoming timestamp on
the outgoing buffer when it didn't have a timestamp already.</doc>
<return-value transfer-ownership="none">
@ -2447,6 +2449,13 @@ the outgoing buffer when it didn't have a timestamp already.</doc>
</parameter>
</parameters>
</method>
<property name="max-reorder" version="1.18" writable="1" transfer-ownership="none">
<doc xml:space="preserve">Max seqnum reorder before the sender is assumed to have restarted.
When max-reorder is set to 0 all reordered/duplicate packets are
considered coming from a restarted sender.</doc>
<type name="gint" c:type="gint"/>
</property>
<property name="source-info" version="1.16" writable="1" transfer-ownership="none">
<doc xml:space="preserve">Add RTP source information found in RTP header as meta to output buffer.</doc>
<type name="gboolean" c:type="gboolean"/>
@ -2931,6 +2940,17 @@ timestamps for audio streams.</doc>
<doc xml:space="preserve">Force buffers to be multiples of this duration in ns (0 disables)</doc>
<type name="gint64" c:type="gint64"/>
</property>
<property name="scale-rtptime" version="1.18" writable="1" transfer-ownership="none">
<doc xml:space="preserve">Make the RTP packets' timestamps be scaled with the segment's rate
(corresponding to RTSP speed parameter). Disabling this property means
the timestamps will not be affected by the set delivery speed (RTSP speed).
Example: A server wants to allow streaming a recorded video in double
speed but still have the timestamps correspond to the position in the
video. This is achieved by the client setting RTSP Speed to 2 while the
server has this property disabled.</doc>
<type name="gboolean" c:type="gboolean"/>
</property>
<property name="seqnum" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
@ -2968,6 +2988,16 @@ the last processed buffer and current state of the stream being payloaded:
<property name="timestamp-offset" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="twcc-ext-id" version="1.18" writable="1" transfer-ownership="none">
<doc xml:space="preserve">The RTP header-extension ID used for tagging buffers with Transport-Wide
Congestion Control sequence-numbers.
To use this across multiple bundled streams (transport wide), the
GstRTPFunnel can mux TWCC sequence-numbers together.
This is experimental, as it is still a draft and not yet a standard.</doc>
<type name="guint" c:type="guint"/>
</property>
<field name="element">
<type name="Gst.Element" c:type="GstElement"/>
</field>
@ -3197,7 +3227,7 @@ the last processed buffer and current state of the stream being payloaded:
RTP header. If there is already a RFC 5285 header extension with a one byte
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
the mechanism described in RFC 5285 or if there is a header extension with
a two bytes header as described in RFC 5285. In that case, use
gst_rtp_buffer_add_extension_twobytes_header()</doc>
@ -3231,7 +3261,7 @@ gst_rtp_buffer_add_extension_twobytes_header()</doc>
RTP header. If there is already a RFC 5285 header extension with a two bytes
header, the new extension will be appended.
It will not work if there is already a header extension that does not follow
the mecanism described in RFC 5285 or if there is a header extension with
the mechanism described in RFC 5285 or if there is a header extension with
a one byte header as described in RFC 5285. In that case, use
gst_rtp_buffer_add_extension_onebyte_header()</doc>
@ -4039,6 +4069,46 @@ into account:
</parameter>
</parameters>
</function>
<function name="get_extension_onebyte_header_from_bytes" c:identifier="gst_rtp_buffer_get_extension_onebyte_header_from_bytes" version="1.18">
<doc xml:space="preserve">Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
Parses RFC 5285 style header extensions with a one byte header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">TRUE if @bytes had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="bytes" transfer-ownership="none">
<doc xml:space="preserve">#GBytes</doc>
<type name="GLib.Bytes" c:type="GBytes*"/>
</parameter>
<parameter name="bit_pattern" transfer-ownership="none">
<doc xml:space="preserve">The bit-pattern. Anything but 0xBEDE is rejected.</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:space="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:space="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" direction="out" caller-allocates="0" transfer-ownership="none">
<doc xml:space="preserve">
location for data</doc>
<array length="5" zero-terminated="0" c:type="gpointer*">
<type name="guint8"/>
</array>
</parameter>
<parameter name="size" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="map" c:identifier="gst_rtp_buffer_map">
<doc xml:space="preserve">Map the contents of @buffer into @rtp.</doc>
@ -4320,7 +4390,7 @@ channels. NULL = not applicable.</doc>
mostly used to get the default clock-rate and bandwidth for dynamic payload
types specified with @media and @encoding name.
The search for @encoding_name will be performed in a case insensitve way.</doc>
The search for @encoding_name will be performed in a case insensitive way.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
@ -5180,6 +5250,46 @@ into account:
</parameter>
</parameters>
</function>
<function name="rtp_buffer_get_extension_onebyte_header_from_bytes" c:identifier="gst_rtp_buffer_get_extension_onebyte_header_from_bytes" moved-to="RTPBuffer.get_extension_onebyte_header_from_bytes" version="1.18">
<doc xml:space="preserve">Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
Parses RFC 5285 style header extensions with a one byte header. It will
return the nth extension with the requested id.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">TRUE if @bytes had the requested header extension</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<parameter name="bytes" transfer-ownership="none">
<doc xml:space="preserve">#GBytes</doc>
<type name="GLib.Bytes" c:type="GBytes*"/>
</parameter>
<parameter name="bit_pattern" transfer-ownership="none">
<doc xml:space="preserve">The bit-pattern. Anything but 0xBEDE is rejected.</doc>
<type name="guint16" c:type="guint16"/>
</parameter>
<parameter name="id" transfer-ownership="none">
<doc xml:space="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
<type name="guint8" c:type="guint8"/>
</parameter>
<parameter name="nth" transfer-ownership="none">
<doc xml:space="preserve">Read the nth extension packet with the requested ID</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="data" direction="out" caller-allocates="0" transfer-ownership="none">
<doc xml:space="preserve">
location for data</doc>
<array length="5" zero-terminated="0" c:type="gpointer*">
<type name="guint8"/>
</array>
</parameter>
<parameter name="size" direction="out" caller-allocates="0" transfer-ownership="full">
<doc xml:space="preserve">the size of the data in bytes</doc>
<type name="guint" c:type="guint*"/>
</parameter>
</parameters>
</function>
<function name="rtp_buffer_map" c:identifier="gst_rtp_buffer_map" moved-to="RTPBuffer.map">
<doc xml:space="preserve">Map the contents of @buffer into @rtp.</doc>
@ -5399,7 +5509,7 @@ extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes.</doc>
mostly used to get the default clock-rate and bandwidth for dynamic payload
types specified with @media and @encoding name.
The search for @encoding_name will be performed in a case insensitve way.</doc>
The search for @encoding_name will be performed in a case insensitive way.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>

353
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@ -111,7 +111,7 @@ state as when it was first created.</doc>
</instance-parameter>
</parameters>
</method>
<method name="connect" c:identifier="gst_rtsp_connection_connect">
<method name="connect" c:identifier="gst_rtsp_connection_connect" deprecated="1" deprecated-version="1.18">
<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
@ -129,12 +129,35 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
</instance-parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a #GTimeVal timeout</doc>
<doc xml:space="preserve">a GTimeVal timeout</doc>
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
</parameter>
</parameters>
</method>
<method name="connect_with_response" c:identifier="gst_rtsp_connection_connect_with_response" version="1.8">
<method name="connect_usec" c:identifier="gst_rtsp_connection_connect_usec" version="1.18">
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
gst_rtsp_connection_create(). If @timeout is 0 this function can block
forever. If @timeout contains a valid timeout, this function will return
#GST_RTSP_ETIMEOUT after the timeout expired.
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.</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="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="connect_with_response" c:identifier="gst_rtsp_connection_connect_with_response" version="1.8" deprecated="1" deprecated-version="1.18">
<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
@ -153,7 +176,7 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
</instance-parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a #GTimeVal timeout</doc>
<doc xml:space="preserve">a GTimeVal timeout</doc>
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
</parameter>
<parameter name="response" transfer-ownership="none">
@ -162,6 +185,34 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="connect_with_response_usec" c:identifier="gst_rtsp_connection_connect_with_response_usec" version="1.18">
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
gst_rtsp_connection_create(). If @timeout is 0 this function can block
forever. If @timeout contains a valid timeout, this function will return
#GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
@response will contain a response to the tunneling request messages.
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.</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="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
<parameter name="response" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMessage</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</parameter>
</parameters>
</method>
<method name="do_tunnel" c:identifier="gst_rtsp_connection_do_tunnel">
<doc xml:space="preserve">If @conn received the first tunnel connection and @conn2 received
the second tunnel connection, link the two connections together so that
@ -395,7 +446,7 @@ error. The file descriptor remains valid until the connection is closed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="next_timeout" c:identifier="gst_rtsp_connection_next_timeout">
<method name="next_timeout" c:identifier="gst_rtsp_connection_next_timeout" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Calculate the next timeout for @conn, storing the result in @timeout.</doc>
<return-value transfer-ownership="none">
@ -413,7 +464,21 @@ error. The file descriptor remains valid until the connection is closed.</doc>
</parameter>
</parameters>
</method>
<method name="poll" c:identifier="gst_rtsp_connection_poll">
<method name="next_timeout_usec" c:identifier="gst_rtsp_connection_next_timeout_usec" version="1.18">
<doc xml:space="preserve">Calculate the next timeout for @conn</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">#the next timeout in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</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>
</parameters>
</method>
<method name="poll" c:identifier="gst_rtsp_connection_poll" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Wait up to the specified @timeout for the connection to become available for
at least one of the operations specified in @events. When the function returns
with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
@ -446,7 +511,40 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="read" c:identifier="gst_rtsp_connection_read">
<method name="poll_usec" c:identifier="gst_rtsp_connection_poll_usec" version="1.18">
<doc xml:space="preserve">Wait up to the specified @timeout for the connection to become available for
at least one of the operations specified in @events. When the function returns
with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
@conn.
@timeout can be 0, 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="events" transfer-ownership="none">
<doc xml:space="preserve">a bitmask of #GstRTSPEvent flags to check</doc>
<type name="RTSPEvent" c:type="GstRTSPEvent"/>
</parameter>
<parameter name="revents" transfer-ownership="none">
<doc xml:space="preserve">location for result flags</doc>
<type name="RTSPEvent" c:type="GstRTSPEvent*"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="read" c:identifier="gst_rtsp_connection_read" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Attempt to read @size bytes into @data from the connected @conn, blocking up to
the specified @timeout. @timeout can be %NULL, in which case this function
might block forever.
@ -476,7 +574,37 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="receive" c:identifier="gst_rtsp_connection_receive">
<method name="read_usec" c:identifier="gst_rtsp_connection_read_usec" version="1.18">
<doc xml:space="preserve">Attempt to read @size bytes into @data from the connected @conn, blocking up to
the specified @timeout. @timeout can be 0, 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="data" transfer-ownership="none">
<doc xml:space="preserve">the data to read</doc>
<type name="guint8" c:type="guint8*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:space="preserve">the size of @data</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout value in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="receive" c:identifier="gst_rtsp_connection_receive" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Attempt to read into @message from the connected @conn, blocking up to
the specified @timeout. @timeout can be %NULL, in which case this function
might block forever.
@ -502,6 +630,32 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="receive_usec" c:identifier="gst_rtsp_connection_receive_usec" version="1.18">
<doc xml:space="preserve">Attempt to read into @message from the connected @conn, blocking up to
the specified @timeout. @timeout can be 0, 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="message" transfer-ownership="none">
<doc xml:space="preserve">the message to read</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout value or 0</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="reset_timeout" c:identifier="gst_rtsp_connection_reset_timeout">
<doc xml:space="preserve">Reset the timeout of @conn.</doc>
@ -516,7 +670,7 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</instance-parameter>
</parameters>
</method>
<method name="send" c:identifier="gst_rtsp_connection_send">
<method name="send" c:identifier="gst_rtsp_connection_send" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Attempt to send @message to the connected @conn, blocking up to
the specified @timeout. @timeout can be %NULL, in which case this function
might block forever.
@ -542,7 +696,7 @@ 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">
<method name="send_messages" c:identifier="gst_rtsp_connection_send_messages" version="1.16" deprecated="1" deprecated-version="1.18">
<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.
@ -574,6 +728,64 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="send_messages_usec" c:identifier="gst_rtsp_connection_send_messages_usec" version="1.18">
<doc xml:space="preserve">Attempt to send @messages to the connected @conn, blocking up to
the specified @timeout. @timeout can be 0, 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 Since.</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 in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="send_usec" c:identifier="gst_rtsp_connection_send_usec" version="1.18">
<doc xml:space="preserve">Attempt to send @message to the connected @conn, blocking up to
the specified @timeout. @timeout can be 0, 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="message" transfer-ownership="none">
<doc xml:space="preserve">the message to send</doc>
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout value in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="set_accept_certificate_func" c:identifier="gst_rtsp_connection_set_accept_certificate_func" version="1.14">
<doc xml:space="preserve">Sets a custom accept-certificate function for checking certificates for
validity. This will directly map to #GTlsConnection 's "accept-certificate"
@ -632,7 +844,7 @@ user and password respectively.</doc>
</parameters>
</method>
<method name="set_auth_param" c:identifier="gst_rtsp_connection_set_auth_param">
<doc xml:space="preserve">Setup @conn with authentication directives. This is not necesary for
<doc xml:space="preserve">Setup @conn with authentication directives. This is not necessary for
methods #GST_RTSP_AUTH_NONE and #GST_RTSP_AUTH_BASIC. For
#GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge
in the WWW-Authenticate response header and can include realm, domain,
@ -656,6 +868,25 @@ nonce, opaque, stale, algorithm, qop as per RFC2617.</doc>
</parameter>
</parameters>
</method>
<method name="set_content_length_limit" c:identifier="gst_rtsp_connection_set_content_length_limit" version="1.18">
<doc xml:space="preserve">Configure @conn to use the specified Content-Length limit.
Both requests and responses are validated. If content-length is
exceeded, ENOMEM error will be returned.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</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="limit" transfer-ownership="none">
<doc xml:space="preserve">Content-Length limit</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</method>
<method name="set_http_mode" c:identifier="gst_rtsp_connection_set_http_mode">
<doc xml:space="preserve">By setting the HTTP mode to %TRUE the message parsing will support HTTP
messages in addition to the RTSP messages. It will also disable the
@ -828,7 +1059,7 @@ the @conn is connected.</doc>
</parameter>
</parameters>
</method>
<method name="write" c:identifier="gst_rtsp_connection_write">
<method name="write" c:identifier="gst_rtsp_connection_write" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Attempt to write @size bytes of @data to the connected @conn, blocking up to
the specified @timeout. @timeout can be %NULL, in which case this function
might block forever.
@ -858,6 +1089,36 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
</parameter>
</parameters>
</method>
<method name="write_usec" c:identifier="gst_rtsp_connection_write_usec" version="1.18">
<doc xml:space="preserve">Attempt to write @size bytes of @data to the connected @conn, blocking up to
the specified @timeout. @timeout can be 0, 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="data" transfer-ownership="none">
<doc xml:space="preserve">the data to write</doc>
<type name="guint8" c:type="const guint8*"/>
</parameter>
<parameter name="size" transfer-ownership="none">
<doc xml:space="preserve">the size of @data</doc>
<type name="guint" c:type="guint"/>
</parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout value or 0</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<function name="accept" c:identifier="gst_rtsp_connection_accept">
<doc xml:space="preserve">Accept a new connection on @socket and create a new #GstRTSPConnection for
handling communication on new socket.</doc>
@ -2654,7 +2915,7 @@ UTC times will be converted to nanoseconds since 1900.</doc>
<doc xml:space="preserve">no error</doc>
</member>
<member name="error" value="-1" c:identifier="GST_RTSP_ERROR" glib:nick="error">
<doc xml:space="preserve">some unspecified error occured</doc>
<doc xml:space="preserve">some unspecified error occurred</doc>
</member>
<member name="einval" value="-2" c:identifier="GST_RTSP_EINVAL" glib:nick="einval">
<doc xml:space="preserve">invalid arguments were provided to a function</doc>
@ -2666,16 +2927,16 @@ UTC times will be converted to nanoseconds since 1900.</doc>
<doc xml:space="preserve">no memory was available for the operation</doc>
</member>
<member name="eresolv" value="-5" c:identifier="GST_RTSP_ERESOLV" glib:nick="eresolv">
<doc xml:space="preserve">a host resolve error occured</doc>
<doc xml:space="preserve">a host resolve error occurred</doc>
</member>
<member name="enotimpl" value="-6" c:identifier="GST_RTSP_ENOTIMPL" glib:nick="enotimpl">
<doc xml:space="preserve">function not implemented</doc>
</member>
<member name="esys" value="-7" c:identifier="GST_RTSP_ESYS" glib:nick="esys">
<doc xml:space="preserve">a system error occured, errno contains more details</doc>
<doc xml:space="preserve">a system error occurred, errno contains more details</doc>
</member>
<member name="eparse" value="-8" c:identifier="GST_RTSP_EPARSE" glib:nick="eparse">
<doc xml:space="preserve">a parsing error occured</doc>
<doc xml:space="preserve">a parsing error occurred</doc>
</member>
<member name="ewsastart" value="-9" c:identifier="GST_RTSP_EWSASTART" glib:nick="ewsastart">
<doc xml:space="preserve">windows networking could not start</doc>
@ -2687,13 +2948,13 @@ UTC times will be converted to nanoseconds since 1900.</doc>
<doc xml:space="preserve">end-of-file was reached</doc>
</member>
<member name="enet" value="-12" c:identifier="GST_RTSP_ENET" glib:nick="enet">
<doc xml:space="preserve">a network problem occured, h_errno contains more details</doc>
<doc xml:space="preserve">a network problem occurred, h_errno contains more details</doc>
</member>
<member name="enotip" value="-13" c:identifier="GST_RTSP_ENOTIP" glib:nick="enotip">
<doc xml:space="preserve">the host is not an IP host</doc>
</member>
<member name="etimeout" value="-14" c:identifier="GST_RTSP_ETIMEOUT" glib:nick="etimeout">
<doc xml:space="preserve">a timeout occured</doc>
<doc xml:space="preserve">a timeout occurred</doc>
</member>
<member name="etget" value="-15" c:identifier="GST_RTSP_ETGET" glib:nick="etget">
<doc xml:space="preserve">the tunnel GET request has been performed</doc>
@ -3249,6 +3510,28 @@ g_strfreev() when no longer needed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_request_uri_with_control" c:identifier="gst_rtsp_url_get_request_uri_with_control">
<doc xml:space="preserve">Get a newly allocated string describing the request URI for @url
combined with the control path for @control_path</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a string with the request URI combined with the control path.
g_free() after usage.
Since 1.18</doc>
<type name="utf8" c:type="gchar*"/>
</return-value>
<parameters>
<instance-parameter name="url" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPUrl</doc>
<type name="RTSPUrl" c:type="const GstRTSPUrl*"/>
</instance-parameter>
<parameter name="control_path" transfer-ownership="none">
<doc xml:space="preserve">an RTSP aggregate control path</doc>
<type name="utf8" c:type="const gchar*"/>
</parameter>
</parameters>
</method>
<method name="set_port" c:identifier="gst_rtsp_url_set_port">
<doc xml:space="preserve">Set the port number in @url to @port.</doc>
@ -3493,7 +3776,7 @@ count is zero the watch and associated memory will be destroyed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="wait_backlog" c:identifier="gst_rtsp_watch_wait_backlog" version="1.4">
<method name="wait_backlog" c:identifier="gst_rtsp_watch_wait_backlog" version="1.4" deprecated="1" deprecated-version="1.18">
<doc xml:space="preserve">Wait until there is place in the backlog queue, @timeout is reached
or @watch is set to flushing.
@ -3518,11 +3801,41 @@ free space in the backlog queue and try again.</doc>
<type name="RTSPWatch" c:type="GstRTSPWatch*"/>
</instance-parameter>
<parameter name="timeout" transfer-ownership="none">
<doc xml:space="preserve">a #GTimeVal timeout</doc>
<doc xml:space="preserve">a GTimeVal timeout</doc>
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
</parameter>
</parameters>
</method>
<method name="wait_backlog_usec" c:identifier="gst_rtsp_watch_wait_backlog_usec" version="1.18">
<doc xml:space="preserve">Wait until there is place in the backlog queue, @timeout is reached
or @watch is set to flushing.
If @timeout is 0 this function can block forever. If @timeout
contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
after the timeout expired.
The typically use of this function is when gst_rtsp_watch_write_data
returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
free space in the backlog queue and try again.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%GST_RTSP_OK when if there is room in queue.
%GST_RTSP_ETIMEOUT when @timeout was reached.
%GST_RTSP_EINTR when @watch is flushing
%GST_RTSP_EINVAL when called with invalid parameters.</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="timeout" transfer-ownership="none">
<doc xml:space="preserve">a timeout in microseconds</doc>
<type name="gint64" c:type="gint64"/>
</parameter>
</parameters>
</method>
<method name="write_data" c:identifier="gst_rtsp_watch_write_data">
<doc xml:space="preserve">Write @data using the connection of the @watch. If it cannot be sent
immediately, it will be queued for transmission in @watch. The contents of

447
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@ -1113,6 +1113,49 @@ no one else overrides it.</doc>
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</return-value>
</constructor>
<virtual-method name="adjust_play_mode">
<return-value transfer-ownership="none">
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
</return-value>
<parameters>
<instance-parameter name="client" transfer-ownership="none">
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</instance-parameter>
<parameter name="context" transfer-ownership="none">
<type name="RTSPContext" c:type="GstRTSPContext*"/>
</parameter>
<parameter name="range" transfer-ownership="none">
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange**"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<type name="Gst.SeekFlags" c:type="GstSeekFlags*"/>
</parameter>
<parameter name="rate" transfer-ownership="none">
<type name="gdouble" c:type="gdouble*"/>
</parameter>
<parameter name="trickmode_interval" transfer-ownership="none">
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
</parameter>
<parameter name="enable_rate_control" transfer-ownership="none">
<type name="gboolean" c:type="gboolean*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="adjust_play_response">
<return-value transfer-ownership="none">
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
</return-value>
<parameters>
<instance-parameter name="client" transfer-ownership="none">
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</instance-parameter>
<parameter name="context" transfer-ownership="none">
<type name="RTSPContext" c:type="GstRTSPContext*"/>
</parameter>
</parameters>
</virtual-method>
<virtual-method name="announce_request">
<return-value transfer-ownership="none">
@ -1662,6 +1705,20 @@ The connection object returned remains valid until the client is freed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_content_length_limit" c:identifier="gst_rtsp_client_get_content_length_limit" version="1.18">
<doc xml:space="preserve">Get the Content-Length limit of @client.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the Content-Length limit.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<instance-parameter name="client" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPClient</doc>
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_mount_points" c:identifier="gst_rtsp_client_get_mount_points">
<doc xml:space="preserve">Get the #GstRTSPMountPoints object that @client uses to manage its sessions.</doc>
@ -1690,6 +1747,26 @@ The connection object returned remains valid until the client is freed.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_stream_transport" c:identifier="gst_rtsp_client_get_stream_transport" version="1.18">
<doc xml:space="preserve">This is useful when providing a send function through
gst_rtsp_client_set_send_func() when doing RTSP over TCP:
the send function must call gst_rtsp_stream_transport_message_sent ()
on the appropriate transport when data has been received for streaming
to continue.</doc>
<return-value transfer-ownership="none" nullable="1">
<doc xml:space="preserve">the #GstRTSPStreamTransport associated with @channel.</doc>
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
</return-value>
<parameters>
<instance-parameter name="client" transfer-ownership="none">
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</instance-parameter>
<parameter name="channel" transfer-ownership="none">
<type name="guint8" c:type="guint8"/>
</parameter>
</parameters>
</method>
<method name="get_thread_pool" c:identifier="gst_rtsp_client_get_thread_pool">
<doc xml:space="preserve">Get the #GstRTSPThreadPool used as the thread pool of @client.</doc>
@ -1821,6 +1898,26 @@ element in the #GList should be unreffed before the list is freed.</doc>
</parameter>
</parameters>
</method>
<method name="set_content_length_limit" c:identifier="gst_rtsp_client_set_content_length_limit" version="1.18">
<doc xml:space="preserve">Configure @client to use the specified Content-Length limit.
Define an appropriate request size limit and reject requests exceeding the
limit with response status 413 Request Entity Too Large</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="client" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPClient</doc>
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</instance-parameter>
<parameter name="limit" transfer-ownership="none">
<doc xml:space="preserve">Content-Length limit</doc>
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</method>
<method name="set_mount_points" c:identifier="gst_rtsp_client_set_mount_points">
<doc xml:space="preserve">Set @mounts as the mount points for @client which it will use to map urls
to media streams. These mount points are usually inherited from the server that
@ -1948,6 +2045,9 @@ that created the client but can be overridden later.</doc>
<property name="mount-points" writable="1" transfer-ownership="none">
<type name="RTSPMountPoints"/>
</property>
<property name="post-session-timeout" writable="1" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="session-pool" writable="1" transfer-ownership="none">
<type name="RTSPSessionPool"/>
</property>
@ -2375,6 +2475,53 @@ that created the client but can be overridden later.</doc>
</parameters>
</callback>
</field>
<field name="adjust_play_mode">
<callback name="adjust_play_mode">
<return-value transfer-ownership="none">
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
</return-value>
<parameters>
<parameter name="client" transfer-ownership="none">
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</parameter>
<parameter name="context" transfer-ownership="none">
<type name="RTSPContext" c:type="GstRTSPContext*"/>
</parameter>
<parameter name="range" transfer-ownership="none">
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange**"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<type name="Gst.SeekFlags" c:type="GstSeekFlags*"/>
</parameter>
<parameter name="rate" transfer-ownership="none">
<type name="gdouble" c:type="gdouble*"/>
</parameter>
<parameter name="trickmode_interval" transfer-ownership="none">
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
</parameter>
<parameter name="enable_rate_control" transfer-ownership="none">
<type name="gboolean" c:type="gboolean*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="adjust_play_response">
<callback name="adjust_play_response">
<return-value transfer-ownership="none">
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
</return-value>
<parameters>
<parameter name="client" transfer-ownership="none">
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</parameter>
<parameter name="context" transfer-ownership="none">
<type name="RTSPContext" c:type="GstRTSPContext*"/>
</parameter>
</parameters>
</callback>
</field>
<field name="closed">
<callback name="closed">
@ -2820,7 +2967,7 @@ that created the client but can be overridden later.</doc>
</callback>
</field>
<field name="_gst_reserved" readable="0" private="1">
<array zero-terminated="0" fixed-size="4">
<array zero-terminated="0" fixed-size="2">
<type name="gpointer" c:type="gpointer"/>
</array>
</field>
@ -3009,7 +3156,7 @@ context can then be received using gst_rtsp_context_get_current().</doc>
</return-value>
<parameters>
<instance-parameter name="ctx" transfer-ownership="none">
<doc xml:space="preserve">a ##GstRTSPContext</doc>
<doc xml:space="preserve">a #GstRTSPContext</doc>
<type name="RTSPContext" c:type="GstRTSPContext*"/>
</instance-parameter>
</parameters>
@ -3646,6 +3793,43 @@ gst_rtsp_media_prepare ().</doc>
</parameter>
</parameters>
</method>
<method name="get_rate_control" c:identifier="gst_rtsp_media_get_rate_control" version="1.18">
<return-value transfer-ownership="none">
<doc xml:space="preserve">whether @media will follow the Rate-Control=no behaviour as specified
in the ONVIF replay spec.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_rates" c:identifier="gst_rtsp_media_get_rates" version="1.18">
<doc xml:space="preserve">Get the rate and applied_rate of the current segment.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%FALSE if looking up the rate and applied rate failed. Otherwise
%TRUE is returned and @rate and @applied_rate are set to the rate and
applied_rate of the current segment.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia
@rate (allow-none): the rate of the current segment
@applied_rate (allow-none): the applied_rate of the current segment</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
<parameter name="rate" transfer-ownership="none">
<type name="gdouble" c:type="gdouble*"/>
</parameter>
<parameter name="applied_rate" transfer-ownership="none">
<type name="gdouble" c:type="gdouble*"/>
</parameter>
</parameters>
</method>
<method name="get_retransmission_time" c:identifier="gst_rtsp_media_get_retransmission_time">
<doc xml:space="preserve">Get the amount of time to store retransmission data.</doc>
@ -3763,6 +3947,19 @@ will listen on @address and @port for client time requests.</doc>
</parameter>
</parameters>
</method>
<method name="has_completed_sender" c:identifier="gst_rtsp_media_has_completed_sender" version="1.18">
<doc xml:space="preserve">See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">whether @media has at least one complete sender stream.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="is_bind_mcast_address" c:identifier="gst_rtsp_media_is_bind_mcast_address" version="1.16">
<doc xml:space="preserve">Check if multicast sockets are configured to be bound to multicast addresses.</doc>
@ -3792,6 +3989,18 @@ unpreparing.</doc>
</instance-parameter>
</parameters>
</method>
<method name="is_receive_only" c:identifier="gst_rtsp_media_is_receive_only" version="1.18">
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if @media is receive-only, %FALSE otherwise.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="is_reusable" c:identifier="gst_rtsp_media_is_reusable">
<doc xml:space="preserve">Check if the pipeline for @media can be reused after an unprepare.</doc>
@ -3852,6 +4061,27 @@ Use gst_rtsp_media_get_time_provider() to get the network clock.</doc>
</instance-parameter>
</parameters>
</method>
<method name="lock" c:identifier="gst_rtsp_media_lock" version="1.18">
<doc xml:space="preserve">Lock the entire media. This is needed by callers such as rtsp_client to
protect the media when it is shared by many clients.
The lock prevents that concurrent clients alters the shared media,
while one client already is working with it.
Typically the lock is taken in external RTSP API calls that uses shared media
such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.
As best practice take the lock as soon as the function get hold of a shared
media object. Release the lock right before the function returns.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="n_streams" c:identifier="gst_rtsp_media_n_streams">
<doc xml:space="preserve">Get the number of streams in this media.</doc>
@ -3909,10 +4139,9 @@ gst_rtsp_media_prepare().</doc>
</parameter>
</parameters>
</method>
<method name="seek_full" c:identifier="gst_rtsp_media_seek_full" version="1.14">
<doc xml:space="preserve">Seek the pipeline of @media to @range. @media must be prepared with
gst_rtsp_media_prepare(). In order to perform the seek operation,
the pipeline must contain all needed transport parts (transport sinks).</doc>
<method name="seek_full" c:identifier="gst_rtsp_media_seek_full" version="1.18">
<doc xml:space="preserve">Seek the pipeline of @media to @range with the given @flags.
@media must be prepared with gst_rtsp_media_prepare().</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE on success.</doc>
@ -3933,6 +4162,40 @@ the pipeline must contain all needed transport parts (transport sinks).</doc>
</parameter>
</parameters>
</method>
<method name="seek_trickmode" c:identifier="gst_rtsp_media_seek_trickmode" version="1.18">
<doc xml:space="preserve">Seek the pipeline of @media to @range with the given @flags and @rate,
and @trickmode_interval.
@media must be prepared with gst_rtsp_media_prepare().
In order to perform the seek operation, the pipeline must contain all
needed transport parts (transport sinks).</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="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
<parameter name="range" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPTimeRange</doc>
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange*"/>
</parameter>
<parameter name="flags" transfer-ownership="none">
<doc xml:space="preserve">The minimal set of #GstSeekFlags to use</doc>
<type name="Gst.SeekFlags" c:type="GstSeekFlags"/>
</parameter>
<parameter name="rate" transfer-ownership="none">
<doc xml:space="preserve">the rate to use in the seek</doc>
<type name="gdouble" c:type="gdouble"/>
</parameter>
<parameter name="trickmode_interval" transfer-ownership="none">
<doc xml:space="preserve">The trickmode interval to use for KEY_UNITS trick mode</doc>
<type name="Gst.ClockTime" c:type="GstClockTime"/>
</parameter>
</parameters>
</method>
<method name="seekable" c:identifier="gst_rtsp_media_seekable" version="1.14">
<doc xml:space="preserve">Check if the pipeline for @media seek and up to what point in time,
it can seek.</doc>
@ -4189,6 +4452,22 @@ it is unprepared.</doc>
</parameter>
</parameters>
</method>
<method name="set_rate_control" c:identifier="gst_rtsp_media_set_rate_control" version="1.18">
<doc xml:space="preserve">Define whether @media will follow the Rate-Control=no behaviour as specified
in the ONVIF replay spec.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
<parameter name="enabled" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</method>
<method name="set_retransmission_time" c:identifier="gst_rtsp_media_set_retransmission_time">
<doc xml:space="preserve">Set the amount of time to store retransmission packets.</doc>
@ -4378,12 +4657,25 @@ taken of @pipeline.</doc>
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
<parameter name="pipeline" transfer-ownership="full">
<parameter name="pipeline" transfer-ownership="none">
<doc xml:space="preserve">a #GstPipeline</doc>
<type name="Gst.Pipeline" c:type="GstPipeline*"/>
</parameter>
</parameters>
</method>
<method name="unlock" c:identifier="gst_rtsp_media_unlock" version="1.18">
<doc xml:space="preserve">Unlock the media.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="media" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPMedia</doc>
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
</instance-parameter>
</parameters>
</method>
<method name="unprepare" c:identifier="gst_rtsp_media_unprepare">
<doc xml:space="preserve">Unprepare @media. After this call, the media should be prepared again before
it can be used again. If the media is set to be non-reusable, a new instance
@ -5909,7 +6201,7 @@ when a client disconnects without sending TEARDOWN.</doc>
</record>
<class name="RTSPMediaFactoryURI" c:symbol-prefix="rtsp_media_factory_uri" c:type="GstRTSPMediaFactoryURI" parent="RTSPMediaFactory" glib:type-name="GstRTSPMediaFactoryURI" glib:get-type="gst_rtsp_media_factory_uri_get_type" glib:type-struct="RTSPMediaFactoryURIClass">
<doc xml:space="preserve">A media factory that creates a pipeline to play and uri.</doc>
<doc xml:space="preserve">A media factory that creates a pipeline to play any uri.</doc>
<constructor name="new" c:identifier="gst_rtsp_media_factory_uri_new">
<doc xml:space="preserve">Create a new #GstRTSPMediaFactoryURI instance.</doc>
@ -6017,7 +6309,10 @@ and called when a message has been sent on the transport.</doc>
<type name="none" c:type="void"/>
</return-value>
<parameters>
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="0">
<parameter name="trans" transfer-ownership="none">
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
</parameter>
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="1">
<doc xml:space="preserve">user data</doc>
<type name="gpointer" c:type="gpointer"/>
</parameter>
@ -6187,6 +6482,14 @@ g_object_unref() after usage.</doc>
</record>
<class name="RTSPOnvifClient" c:symbol-prefix="rtsp_onvif_client" c:type="GstRTSPOnvifClient" version="1.14" parent="RTSPClient" glib:type-name="GstRTSPOnvifClient" glib:get-type="gst_rtsp_onvif_client_get_type" glib:type-struct="RTSPOnvifClientClass">
<constructor name="new" c:identifier="gst_rtsp_onvif_client_new" version="1.18">
<doc xml:space="preserve">Create a new #GstRTSPOnvifClient instance.</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a new #GstRTSPOnvifClient</doc>
<type name="RTSPClient" c:type="GstRTSPClient*"/>
</return-value>
</constructor>
<field name="parent">
<type name="RTSPClient" c:type="GstRTSPClient"/>
</field>
@ -6370,6 +6673,18 @@ usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="has_replay_support" c:identifier="gst_rtsp_onvif_media_factory_has_replay_support" version="1.18">
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if ONVIF replay is supported by the media factory.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="factory" transfer-ownership="none">
<type name="RTSPOnvifMediaFactory" c:type="GstRTSPOnvifMediaFactory*"/>
</instance-parameter>
</parameters>
</method>
<method name="set_backchannel_bandwidth" c:identifier="gst_rtsp_onvif_media_factory_set_backchannel_bandwidth" version="1.14">
<doc xml:space="preserve">Set the configured/supported bandwidth of the ONVIF backchannel pipeline in
bits per second.</doc>
@ -6420,6 +6735,21 @@ prepare.</doc>
</parameter>
</parameters>
</method>
<method name="set_replay_support" c:identifier="gst_rtsp_onvif_media_factory_set_replay_support" version="1.18">
<doc xml:space="preserve">Set to %TRUE if ONVIF replay is supported by the media factory.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="factory" transfer-ownership="none">
<type name="RTSPOnvifMediaFactory" c:type="GstRTSPOnvifMediaFactory*"/>
</instance-parameter>
<parameter name="has_replay_support" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</method>
<field name="parent">
<type name="RTSPMediaFactory" c:type="GstRTSPMediaFactory"/>
</field>
@ -6974,6 +7304,20 @@ usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_content_length_limit" c:identifier="gst_rtsp_server_get_content_length_limit" version="1.18">
<doc xml:space="preserve">Get the Content-Length limit of @server.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the Content-Length limit.</doc>
<type name="guint" c:type="guint"/>
</return-value>
<parameters>
<instance-parameter name="server" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPServer</doc>
<type name="RTSPServer" c:type="GstRTSPServer*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_mount_points" c:identifier="gst_rtsp_server_get_mount_points">
<doc xml:space="preserve">Get the #GstRTSPMountPoints used as the mount points of @server.</doc>
@ -7089,6 +7433,24 @@ This function must be called before the server is bound.</doc>
</parameter>
</parameters>
</method>
<method name="set_content_length_limit" c:identifier="gst_rtsp_server_set_content_length_limit" version="1.18">
<doc xml:space="preserve">Define an appropriate request size limit and reject requests exceeding the
limit.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="server" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPServer
Configure @server to use the specified Content-Length limit.</doc>
<type name="RTSPServer" c:type="GstRTSPServer*"/>
</instance-parameter>
<parameter name="limit" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</parameter>
</parameters>
</method>
<method name="set_mount_points" c:identifier="gst_rtsp_server_set_mount_points">
<doc xml:space="preserve">configure @mounts to be used as the mount points of @server.</doc>
@ -7207,6 +7569,9 @@ that the HTTP server read from the socket while parsing the HTTP header.</doc>
<property name="bound-port" transfer-ownership="none">
<type name="gint" c:type="gint"/>
</property>
<property name="content-length-limit" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="mount-points" writable="1" transfer-ownership="none">
<type name="RTSPMountPoints"/>
</property>
@ -7614,6 +7979,9 @@ cleaned up when there is no activity for @timeout seconds.</doc>
</instance-parameter>
</parameters>
</method>
<property name="extra-timeout" writable="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
<property name="sessionid" writable="1" construct-only="1" transfer-ownership="none">
<type name="utf8" c:type="gchar*"/>
</property>
@ -8024,8 +8392,7 @@ what happens to the session. @func will be called with the session pool
locked so no further actions on @pool can be performed from @func.
If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the
expired state with gst_rtsp_session_set_expired() and removed from
@pool.
expired state and removed from @pool.
If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool.
@ -8344,7 +8711,8 @@ allocated.</doc>
</method>
<method name="add_transport" c:identifier="gst_rtsp_stream_add_transport">
<doc xml:space="preserve">Add the transport in @trans to @stream. The media of @stream will
then also be send to the values configured in @trans.
then also be send to the values configured in @trans. Adding the
same transport twice will not add it a second time.
@stream must be joined to a bin.
@ -8656,6 +9024,41 @@ g_free() after usage.</doc>
</instance-parameter>
</parameters>
</method>
<method name="get_rate_control" c:identifier="gst_rtsp_stream_get_rate_control" version="1.18">
<return-value transfer-ownership="none">
<doc xml:space="preserve">whether @stream will follow the Rate-Control=no behaviour as specified
in the ONVIF replay spec.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="stream" transfer-ownership="none">
<type name="RTSPStream" c:type="GstRTSPStream*"/>
</instance-parameter>
</parameters>
</method>
<method name="get_rates" c:identifier="gst_rtsp_stream_get_rates" version="1.18">
<doc xml:space="preserve">Retrieve the current rate and/or applied_rate.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">%TRUE if rate and/or applied_rate could be determined.</doc>
<type name="gboolean" c:type="gboolean"/>
</return-value>
<parameters>
<instance-parameter name="stream" transfer-ownership="none">
<doc xml:space="preserve">a #GstRTSPStream</doc>
<type name="RTSPStream" c:type="GstRTSPStream*"/>
</instance-parameter>
<parameter name="rate" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">the configured rate</doc>
<type name="gdouble" c:type="gdouble*"/>
</parameter>
<parameter name="applied_rate" transfer-ownership="none" nullable="1" allow-none="1">
<doc xml:space="preserve">the configured applied_rate</doc>
<type name="gdouble" c:type="gdouble*"/>
</parameter>
</parameters>
</method>
<method name="get_retransmission_pt" c:identifier="gst_rtsp_stream_get_retransmission_pt">
<doc xml:space="preserve">Get the payload-type used for retransmission of this stream</doc>
@ -9591,6 +9994,22 @@ an RTSP connection.</doc>
</parameter>
</parameters>
</method>
<method name="set_rate_control" c:identifier="gst_rtsp_stream_set_rate_control" version="1.18">
<doc xml:space="preserve">Define whether @stream will follow the Rate-Control=no behaviour as specified
in the ONVIF replay spec.</doc>
<return-value transfer-ownership="none">
<type name="none" c:type="void"/>
</return-value>
<parameters>
<instance-parameter name="stream" transfer-ownership="none">
<type name="RTSPStream" c:type="GstRTSPStream*"/>
</instance-parameter>
<parameter name="enabled" transfer-ownership="none">
<type name="gboolean" c:type="gboolean"/>
</parameter>
</parameters>
</method>
<method name="set_retransmission_pt" c:identifier="gst_rtsp_stream_set_retransmission_pt">
<doc xml:space="preserve">Set the payload type (pt) for retransmission of this stream.</doc>
@ -11203,6 +11622,10 @@ port pair in multicast. No response is sent when the check returns
</parameter>
</parameters>
</function-macro>
<constant name="RTSP_ONVIF_REPLAY_REQUIREMENT" value="onvif-replay" c:type="GST_RTSP_ONVIF_REPLAY_REQUIREMENT">
<type name="utf8" c:type="gchar*"/>
</constant>
<function-macro name="RTSP_ONVIF_SERVER" c:identifier="GST_RTSP_ONVIF_SERVER" introspectable="0">
<parameters>

18
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@ -379,7 +379,7 @@ in NTP-UTC format.</doc>
</parameters>
</method>
<method name="find_payload" c:identifier="gst_mikey_message_find_payload" version="1.4">
<doc xml:space="preserve">Find the @nth occurence of the payload with @type in @msg.</doc>
<doc xml:space="preserve">Find the @nth occurrence of the payload with @type in @msg.</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">the @nth #GstMIKEYPayload of @type.</doc>
@ -1218,7 +1218,7 @@ specific security protocol</doc>
</field>
</record>
<record name="MIKEYPayloadSPParam" c:type="GstMIKEYPayloadSPParam">
<doc xml:space="preserve">A Type/Length/Value field for security paramaters</doc>
<doc xml:space="preserve">A Type/Length/Value field for security parameters</doc>
<field name="type" writable="1">
<doc xml:space="preserve">specifies the type of the parameter</doc>
@ -1280,7 +1280,7 @@ specific security protocol</doc>
<doc xml:space="preserve">Cert hash payload</doc>
</member>
<member name="v" value="9" c:identifier="GST_MIKEY_PT_V">
<doc xml:space="preserve">Verfication message payload</doc>
<doc xml:space="preserve">Verification message payload</doc>
</member>
<member name="sp" value="10" c:identifier="GST_MIKEY_PT_SP">
<doc xml:space="preserve">Security Policy payload</doc>
@ -1935,7 +1935,9 @@ a=rtpmap:(payload) (encoding_name)/(clock_rate)[/(encoding_params)]
a=framesize:(payload) (width)-(height)
a=fmtp:(payload) (param)[=(value)];...</doc>
a=fmtp:(payload) (param)[=(value)];...
Note that the extmap attribute is set only by gst_sdp_media_attributes_to_caps().</doc>
<return-value transfer-ownership="full">
<doc xml:space="preserve">a #GstCaps, or %NULL if an error happened</doc>
@ -2499,7 +2501,9 @@ a=framesize:(payload) (width)-(height)
a=fmtp:(payload) (param)[=(value)];...
a=rtcp-fb:(payload) (param1) [param2]...</doc>
a=rtcp-fb:(payload) (param1) [param2]...
a=extmap:(id)[/direction] (extensionname) (extensionattributes)</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstSDPResult.</doc>
@ -4158,7 +4162,9 @@ a=framesize:(payload) (width)-(height)
a=fmtp:(payload) (param)[=(value)];...
a=rtcp-fb:(payload) (param1) [param2]...</doc>
a=rtcp-fb:(payload) (param1) [param2]...
a=extmap:(id)[/direction] (extensionname) (extensionattributes)</doc>
<return-value transfer-ownership="none">
<doc xml:space="preserve">a #GstSDPResult.</doc>

26
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@ -43,6 +43,16 @@ and/or use gtk-doc annotations. -->
</parameter>
</parameters>
</function-macro>
<constant name="TAG_ACOUSTID_FINGERPRINT" value="chromaprint-fingerprint" c:type="GST_TAG_ACOUSTID_FINGERPRINT" version="1.18">
<doc xml:space="preserve">AcoustID Fingerprint (Chromaprint)</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="TAG_ACOUSTID_ID" value="acoustid-id" c:type="GST_TAG_ACOUSTID_ID" version="1.18">
<doc xml:space="preserve">AcoustID Identifier</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="TAG_CAPTURING_CONTRAST" value="capturing-contrast" c:type="GST_TAG_CAPTURING_CONTRAST">
<doc xml:space="preserve">Direction of contrast processing applied when capturing an image. (string)
@ -311,6 +321,16 @@ keys.</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="TAG_MUSICBRAINZ_RELEASEGROUPID" value="musicbrainz-releasegroupid" c:type="GST_TAG_MUSICBRAINZ_RELEASEGROUPID" version="1.18">
<doc xml:space="preserve">MusicBrainz Release Group ID</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="TAG_MUSICBRAINZ_RELEASETRACKID" value="musicbrainz-releasetrackid" c:type="GST_TAG_MUSICBRAINZ_RELEASETRACKID" version="1.18">
<doc xml:space="preserve">MusicBrainz Release Track ID</doc>
<type name="utf8" c:type="gchar*"/>
</constant>
<constant name="TAG_MUSICBRAINZ_TRACKID" value="musicbrainz-trackid" c:type="GST_TAG_MUSICBRAINZ_TRACKID">
<doc xml:space="preserve">MusicBrainz track ID</doc>
@ -1062,7 +1082,7 @@ code (both are accepted for convenience).
The "bibliographic" code is derived from the English name of the language
(e.g. "ger" for German instead of "de" or "deu"). In most scenarios, the
"terminological" codes are prefered.
"terminological" codes are preferred.
Language codes are case-sensitive and expected to be lower case.</doc>
@ -1086,7 +1106,7 @@ code (both are accepted for convenience).
The "terminological" code is derived from the local name of the language
(e.g. "deu" for German instead of "ger"). In most scenarios, the
"terminological" codes are prefered over the "bibliographic" ones.
"terminological" codes are preferred over the "bibliographic" ones.
Language codes are case-sensitive and expected to be lower case.</doc>
@ -1296,7 +1316,7 @@ rather than the image itself.
In GStreamer, image tags are #GstSample&lt;!-- --&gt;s containing the raw image
data, with the sample caps describing the content type of the image
(e.g. image/jpeg, image/png, text/uri-list). The sample info may contain
an additional 'image-type' field of #GST_TYPE_TAG_IMAGE_TYPE to describe
an additional 'image-type' field of #GstTagImageType to describe
the type of image (front cover, back cover etc.). #GST_TAG_PREVIEW_IMAGE
tags should not carry an image type, their type is already indicated via
the special tag name.

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@ -200,17 +200,17 @@ for more information.</doc>
</member>
</enumeration>
<enumeration name="WebRTCDTLSSetup" glib:type-name="GstWebRTCDTLSSetup" glib:get-type="gst_webrtc_dtls_setup_get_type" c:type="GstWebRTCDTLSSetup">
<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
<member name="none" value="0" c:identifier="GST_WEBRTC_DTLS_SETUP_NONE" glib:nick="none">
<doc xml:space="preserve">none</doc>
</member>
<member name="actpass" value="1" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS" glib:nick="actpass">
<doc xml:space="preserve">actpass</doc>
</member>
<member name="active" value="2" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE" glib:nick="active">
<doc xml:space="preserve">sendonly</doc>
</member>
<member name="passive" value="3" c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE" glib:nick="passive">
<doc xml:space="preserve">recvonly</doc>
</member>
</enumeration>
<class name="WebRTCDTLSTransport" c:symbol-prefix="webrtc_dtls_transport" c:type="GstWebRTCDTLSTransport" parent="Gst.Object" glib:type-name="GstWebRTCDTLSTransport" glib:get-type="gst_webrtc_dtls_transport_get_type" glib:type-struct="WebRTCDTLSTransportClass">
@ -306,20 +306,20 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
</field>
</record>
<enumeration name="WebRTCDTLSTransportState" glib:type-name="GstWebRTCDTLSTransportState" glib:get-type="gst_webrtc_dtls_transport_state_get_type" c:type="GstWebRTCDTLSTransportState">
<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
<member name="new" value="0" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" glib:nick="new">
<doc xml:space="preserve">new</doc>
</member>
<member name="closed" value="1" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" glib:nick="closed">
<doc xml:space="preserve">closed</doc>
</member>
<member name="failed" value="2" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" glib:nick="failed">
<doc xml:space="preserve">failed</doc>
</member>
<member name="connecting" value="3" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" glib:nick="connecting">
<doc xml:space="preserve">connecting</doc>
</member>
<member name="connected" value="4" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" glib:nick="connected">
<doc xml:space="preserve">connected</doc>
</member>
</enumeration>
<enumeration name="WebRTCDataChannelState" version="1.16" glib:type-name="GstWebRTCDataChannelState" glib:get-type="gst_webrtc_data_channel_state_get_type" c:type="GstWebRTCDataChannelState">
@ -328,7 +328,7 @@ 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>
See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate&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">
@ -349,55 +349,55 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate"&gt;h
</member>
</enumeration>
<enumeration name="WebRTCICEComponent" glib:type-name="GstWebRTCICEComponent" glib:get-type="gst_webrtc_ice_component_get_type" c:type="GstWebRTCICEComponent">
<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP" glib:nick="rtp">
<doc xml:space="preserve">RTP component</doc>
</member>
<member name="rtcp" value="1" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP" glib:nick="rtcp">
<doc xml:space="preserve">RTCP component</doc>
</member>
</enumeration>
<enumeration name="WebRTCICEConnectionState" glib:type-name="GstWebRTCICEConnectionState" glib:get-type="gst_webrtc_ice_connection_state_get_type" c:type="GstWebRTCICEConnectionState">
<doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate&gt;</doc>
<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" glib:nick="new">
<doc xml:space="preserve">new</doc>
</member>
<member name="checking" value="1" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" glib:nick="checking">
<doc xml:space="preserve">checking</doc>
</member>
<member name="connected" value="2" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" glib:nick="connected">
<doc xml:space="preserve">connected</doc>
</member>
<member name="completed" value="3" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" glib:nick="completed">
<doc xml:space="preserve">completed</doc>
</member>
<member name="failed" value="4" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" glib:nick="failed">
<doc xml:space="preserve">failed</doc>
</member>
<member name="disconnected" value="5" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
<doc xml:space="preserve">disconnected</doc>
</member>
<member name="closed" value="6" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" glib:nick="closed">
<doc xml:space="preserve">closed</doc>
</member>
</enumeration>
<enumeration name="WebRTCICEGatheringState" glib:type-name="GstWebRTCICEGatheringState" glib:get-type="gst_webrtc_ice_gathering_state_get_type" c:type="GstWebRTCICEGatheringState">
<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate&gt;</doc>
<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW" glib:nick="new">
<doc xml:space="preserve">new</doc>
</member>
<member name="gathering" value="1" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" glib:nick="gathering">
<doc xml:space="preserve">gathering</doc>
</member>
<member name="complete" value="2" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" glib:nick="complete">
<doc xml:space="preserve">complete</doc>
</member>
</enumeration>
<enumeration name="WebRTCICERole" glib:type-name="GstWebRTCICERole" glib:get-type="gst_webrtc_ice_role_get_type" c:type="GstWebRTCICERole">
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
<member name="controlled" value="0" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED" glib:nick="controlled">
<doc xml:space="preserve">controlled</doc>
</member>
<member name="controlling" value="1" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING" glib:nick="controlling">
<doc xml:space="preserve">controlling</doc>
</member>
</enumeration>
<class name="WebRTCICETransport" c:symbol-prefix="webrtc_ice_transport" c:type="GstWebRTCICETransport" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCICETransport" glib:get-type="gst_webrtc_ice_transport_get_type" glib:type-struct="WebRTCICETransportClass">
@ -558,24 +558,24 @@ for more information.</doc>
</member>
</enumeration>
<enumeration name="WebRTCPeerConnectionState" glib:type-name="GstWebRTCPeerConnectionState" glib:get-type="gst_webrtc_peer_connection_state_get_type" c:type="GstWebRTCPeerConnectionState">
<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate&gt;</doc>
<member name="new" value="0" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" glib:nick="new">
<doc xml:space="preserve">new</doc>
</member>
<member name="connecting" value="1" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" glib:nick="connecting">
<doc xml:space="preserve">connecting</doc>
</member>
<member name="connected" value="2" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" glib:nick="connected">
<doc xml:space="preserve">connected</doc>
</member>
<member name="disconnected" value="3" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
<doc xml:space="preserve">disconnected</doc>
</member>
<member name="failed" value="4" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" glib:nick="failed">
<doc xml:space="preserve">failed</doc>
</member>
<member name="closed" value="5" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" glib:nick="closed">
<doc xml:space="preserve">closed</doc>
</member>
</enumeration>
<enumeration name="WebRTCPriorityType" version="1.16" glib:type-name="GstWebRTCPriorityType" glib:get-type="gst_webrtc_priority_type_get_type" c:type="GstWebRTCPriorityType">
@ -583,7 +583,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
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>
See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype&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">
@ -724,6 +724,10 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
</record>
<class name="WebRTCRTPTransceiver" c:symbol-prefix="webrtc_rtp_transceiver" c:type="GstWebRTCRTPTransceiver" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCRTPTransceiver" glib:get-type="gst_webrtc_rtp_transceiver_get_type" glib:type-struct="WebRTCRTPTransceiverClass">
<property name="direction" version="1.18" writable="1" transfer-ownership="none">
<doc xml:space="preserve">Direction of the transceiver.</doc>
<type name="WebRTCRTPTransceiverDirection"/>
</property>
<property name="mlineindex" writable="1" construct-only="1" transfer-ownership="none">
<type name="guint" c:type="guint"/>
</property>
@ -779,14 +783,19 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
</record>
<enumeration name="WebRTCRTPTransceiverDirection" glib:type-name="GstWebRTCRTPTransceiverDirection" glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type" c:type="GstWebRTCRTPTransceiverDirection">
<member name="none" value="0" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" glib:nick="none">
<doc xml:space="preserve">none</doc>
</member>
<member name="inactive" value="1" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" glib:nick="inactive">
<doc xml:space="preserve">inactive</doc>
</member>
<member name="sendonly" value="2" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" glib:nick="sendonly">
<doc xml:space="preserve">sendonly</doc>
</member>
<member name="recvonly" value="3" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" glib:nick="recvonly">
<doc xml:space="preserve">recvonly</doc>
</member>
<member name="sendrecv" value="4" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" glib:nick="sendrecv">
<doc xml:space="preserve">sendrecv</doc>
</member>
</enumeration>
<enumeration name="WebRTCSCTPTransportState" version="1.16" glib:type-name="GstWebRTCSCTPTransportState" glib:get-type="gst_webrtc_sctp_transport_state_get_type" c:type="GstWebRTCSCTPTransportState">
@ -794,7 +803,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype"&gt;http:
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>
See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate&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">
@ -805,18 +814,18 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate"&gt
</member>
</enumeration>
<enumeration name="WebRTCSDPType" glib:type-name="GstWebRTCSDPType" glib:get-type="gst_webrtc_sdp_type_get_type" c:type="GstWebRTCSDPType">
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
GST_WEBRTC_SDP_TYPE_ANSWER: answer
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.github.io/webrtc-pc/#rtcsdptype&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;http://w3c.github.io/webrtc-pc/#rtcsdptype&gt;</doc>
<member name="offer" value="1" c:identifier="GST_WEBRTC_SDP_TYPE_OFFER" glib:nick="offer">
<doc xml:space="preserve">offer</doc>
</member>
<member name="pranswer" value="2" c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER" glib:nick="pranswer">
<doc xml:space="preserve">pranswer</doc>
</member>
<member name="answer" value="3" c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER" glib:nick="answer">
<doc xml:space="preserve">answer</doc>
</member>
<member name="rollback" value="4" c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK" glib:nick="rollback">
<doc xml:space="preserve">rollback</doc>
</member>
<function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
@ -834,7 +843,7 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.git
</function>
</enumeration>
<record name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription" glib:type-name="GstWebRTCSessionDescription" glib:get-type="gst_webrtc_session_description_get_type" c:symbol-prefix="webrtc_session_description">
<doc xml:space="preserve">See &lt;ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class"&gt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;https://www.w3.org/TR/webrtc/#rtcsessiondescription-class&gt;</doc>
<field name="type" writable="1">
<doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
@ -890,68 +899,68 @@ See &lt;ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype"&gt;http://w3c.git
</method>
</record>
<enumeration name="WebRTCSignalingState" glib:type-name="GstWebRTCSignalingState" glib:get-type="gst_webrtc_signaling_state_get_type" c:type="GstWebRTCSignalingState">
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
See &lt;ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate"&gt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&lt;/ulink&gt;</doc>
<doc xml:space="preserve">See &lt;http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate&gt;</doc>
<member name="stable" value="0" c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE" glib:nick="stable">
<doc xml:space="preserve">stable</doc>
</member>
<member name="closed" value="1" c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED" glib:nick="closed">
<doc xml:space="preserve">closed</doc>
</member>
<member name="have_local_offer" value="2" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" glib:nick="have-local-offer">
<doc xml:space="preserve">have-local-offer</doc>
</member>
<member name="have_remote_offer" value="3" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" glib:nick="have-remote-offer">
<doc xml:space="preserve">have-remote-offer</doc>
</member>
<member name="have_local_pranswer" value="4" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" glib:nick="have-local-pranswer">
<doc xml:space="preserve">have-local-pranswer</doc>
</member>
<member name="have_remote_pranswer" value="5" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" glib:nick="have-remote-pranswer">
<doc xml:space="preserve">have-remote-pranswer</doc>
</member>
</enumeration>
<enumeration name="WebRTCStatsType" glib:type-name="GstWebRTCStatsType" glib:get-type="gst_webrtc_stats_type_get_type" c:type="GstWebRTCStatsType">
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
GST_WEBRTC_STATS_CSRC: csrc
GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
GST_WEBRTC_STATS_STREAM: stream
GST_WEBRTC_STATS_TRANSPORT: transport
GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
<member name="codec" value="1" c:identifier="GST_WEBRTC_STATS_CODEC" glib:nick="codec">
<doc xml:space="preserve">codec</doc>
</member>
<member name="inbound_rtp" value="2" c:identifier="GST_WEBRTC_STATS_INBOUND_RTP" glib:nick="inbound-rtp">
<doc xml:space="preserve">inbound-rtp</doc>
</member>
<member name="outbound_rtp" value="3" c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP" glib:nick="outbound-rtp">
<doc xml:space="preserve">outbound-rtp</doc>
</member>
<member name="remote_inbound_rtp" value="4" c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" glib:nick="remote-inbound-rtp">
<doc xml:space="preserve">remote-inbound-rtp</doc>
</member>
<member name="remote_outbound_rtp" value="5" c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" glib:nick="remote-outbound-rtp">
<doc xml:space="preserve">remote-outbound-rtp</doc>
</member>
<member name="csrc" value="6" c:identifier="GST_WEBRTC_STATS_CSRC" glib:nick="csrc">
<doc xml:space="preserve">csrc</doc>
</member>
<member name="peer_connection" value="7" c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION" glib:nick="peer-connection">
<doc xml:space="preserve">peer-connectiion</doc>
</member>
<member name="data_channel" value="8" c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL" glib:nick="data-channel">
<doc xml:space="preserve">data-channel</doc>
</member>
<member name="stream" value="9" c:identifier="GST_WEBRTC_STATS_STREAM" glib:nick="stream">
<doc xml:space="preserve">stream</doc>
</member>
<member name="transport" value="10" c:identifier="GST_WEBRTC_STATS_TRANSPORT" glib:nick="transport">
<doc xml:space="preserve">transport</doc>
</member>
<member name="candidate_pair" value="11" c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR" glib:nick="candidate-pair">
<doc xml:space="preserve">candidate-pair</doc>
</member>
<member name="local_candidate" value="12" c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE" glib:nick="local-candidate">
<doc xml:space="preserve">local-candidate</doc>
</member>
<member name="remote_candidate" value="13" c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE" glib:nick="remote-candidate">
<doc xml:space="preserve">remote-candidate</doc>
</member>
<member name="certificate" value="14" c:identifier="GST_WEBRTC_STATS_CERTIFICATE" glib:nick="certificate">
<doc xml:space="preserve">certificate</doc>
</member>
</enumeration>
<function name="webrtc_sdp_type_to_string" c:identifier="gst_webrtc_sdp_type_to_string" moved-to="WebRTCSDPType.to_string">