forked from mirrors/gstreamer-rs
14732 lines
619 KiB
XML
14732 lines
619 KiB
XML
<?xml version="1.0"?>
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<!-- This file was automatically generated from C sources - DO NOT EDIT!
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To affect the contents of this file, edit the original C definitions,
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and/or use gtk-doc annotations. -->
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<repository version="1.2"
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xmlns="http://www.gtk.org/introspection/core/1.0"
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xmlns:c="http://www.gtk.org/introspection/c/1.0"
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xmlns:glib="http://www.gtk.org/introspection/glib/1.0">
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<include name="Gst" version="1.0"/>
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<include name="GstBase" version="1.0"/>
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<include name="GstTag" version="1.0"/>
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<package name="gstreamer-audio-1.0"/>
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<c:include name="gst/audio/audio.h"/>
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<namespace name="GstAudio"
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version="1.0"
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shared-library="libgstaudio-1.0.so.0"
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c:identifier-prefixes="Gst"
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c:symbol-prefixes="gst">
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<function-macro name="AUDIO_AGGREGATOR"
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c:identifier="GST_AUDIO_AGGREGATOR"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="163"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="164"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD"
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c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="109"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="110"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="111"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_GET_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="165"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_PAD"
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c:identifier="GST_AUDIO_AGGREGATOR_PAD"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="46"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_PAD_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_PAD_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="47"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_AGGREGATOR_PAD_GET_CLASS"
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c:identifier="GST_AUDIO_AGGREGATOR_PAD_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudioaggregator.h" line="48"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK"
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c:identifier="GST_AUDIO_BASE_SINK"
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introspectable="0">
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<source-position filename="gstaudiobasesink.h" line="61"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK_CAST"
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c:identifier="GST_AUDIO_BASE_SINK_CAST"
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introspectable="0">
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<source-position filename="gstaudiobasesink.h" line="62"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK_CLASS"
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c:identifier="GST_AUDIO_BASE_SINK_CLASS"
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introspectable="0">
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<source-position filename="gstaudiobasesink.h" line="63"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK_CLOCK"
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c:identifier="GST_AUDIO_BASE_SINK_CLOCK"
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introspectable="0">
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<doc xml:space="preserve"
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filename="gstaudiobasesink.h"
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line="68">Get the #GstClock of @obj.</doc>
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<source-position filename="gstaudiobasesink.h" line="74"/>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve"
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filename="gstaudiobasesink.h"
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line="70">a #GstAudioBaseSink</doc>
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK_GET_CLASS"
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c:identifier="GST_AUDIO_BASE_SINK_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudiobasesink.h" line="64"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SINK_PAD"
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c:identifier="GST_AUDIO_BASE_SINK_PAD"
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introspectable="0">
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<doc xml:space="preserve"
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filename="gstaudiobasesink.h"
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line="75">Get the sink #GstPad of @obj.</doc>
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<source-position filename="gstaudiobasesink.h" line="81"/>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve"
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filename="gstaudiobasesink.h"
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line="77">a #GstAudioBaseSink</doc>
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC"
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c:identifier="GST_AUDIO_BASE_SRC"
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introspectable="0">
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<source-position filename="gstaudiobasesrc.h" line="39"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC_CAST"
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c:identifier="GST_AUDIO_BASE_SRC_CAST"
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introspectable="0">
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<source-position filename="gstaudiobasesrc.h" line="40"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC_CLASS"
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c:identifier="GST_AUDIO_BASE_SRC_CLASS"
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introspectable="0">
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<source-position filename="gstaudiobasesrc.h" line="41"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC_CLOCK"
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c:identifier="GST_AUDIO_BASE_SRC_CLOCK"
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introspectable="0">
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<doc xml:space="preserve"
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filename="gstaudiobasesrc.h"
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line="46">Get the #GstClock of @obj.</doc>
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<source-position filename="gstaudiobasesrc.h" line="52"/>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve"
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filename="gstaudiobasesrc.h"
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line="48">a #GstAudioBaseSrc</doc>
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC_GET_CLASS"
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c:identifier="GST_AUDIO_BASE_SRC_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudiobasesrc.h" line="42"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BASE_SRC_PAD"
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c:identifier="GST_AUDIO_BASE_SRC_PAD"
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introspectable="0">
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<doc xml:space="preserve"
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filename="gstaudiobasesrc.h"
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line="53">Get the source #GstPad of @obj.</doc>
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<source-position filename="gstaudiobasesrc.h" line="59"/>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve"
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filename="gstaudiobasesrc.h"
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line="55">a #GstAudioBaseSrc</doc>
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_BPF"
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c:identifier="GST_AUDIO_BUFFER_BPF"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="85"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_BPS"
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c:identifier="GST_AUDIO_BUFFER_BPS"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="84"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_CHANNELS"
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c:identifier="GST_AUDIO_BUFFER_CHANNELS"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="77"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_DEPTH"
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c:identifier="GST_AUDIO_BUFFER_DEPTH"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="82"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_FORMAT"
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c:identifier="GST_AUDIO_BUFFER_FORMAT"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="76"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_LAYOUT"
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c:identifier="GST_AUDIO_BUFFER_LAYOUT"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="78"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_N_PLANES"
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c:identifier="GST_AUDIO_BUFFER_N_PLANES"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="88"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_N_SAMPLES"
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c:identifier="GST_AUDIO_BUFFER_N_SAMPLES"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="87"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_PLANE_DATA"
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c:identifier="GST_AUDIO_BUFFER_PLANE_DATA"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="89"/>
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<parameters>
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<parameter name="b">
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</parameter>
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<parameter name="p">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_PLANE_SIZE"
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c:identifier="GST_AUDIO_BUFFER_PLANE_SIZE"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="92"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_RATE"
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c:identifier="GST_AUDIO_BUFFER_RATE"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="79"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_SAMPLE_STRIDE"
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c:identifier="GST_AUDIO_BUFFER_SAMPLE_STRIDE"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="83"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_BUFFER_WIDTH"
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c:identifier="GST_AUDIO_BUFFER_WIDTH"
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introspectable="0">
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<source-position filename="audio-buffer.h" line="81"/>
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<parameters>
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<parameter name="b">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_CAPS_MAKE"
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c:identifier="GST_AUDIO_CAPS_MAKE"
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introspectable="0">
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<doc xml:space="preserve"
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filename="audio-format.h"
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line="339">Generic caps string for audio, for use in pad templates.</doc>
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<source-position filename="audio-format.h" line="339"/>
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<parameters>
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<parameter name="format">
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<doc xml:space="preserve"
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filename="audio-format.h"
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line="341">string format that describes the sample layout, as string
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(e.g. "S16LE", "S8", etc.)</doc>
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_CD_SRC"
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c:identifier="GST_AUDIO_CD_SRC"
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introspectable="0">
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<source-position filename="gstaudiocdsrc.h" line="33"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_CD_SRC_CLASS"
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c:identifier="GST_AUDIO_CD_SRC_CLASS"
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introspectable="0">
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<source-position filename="gstaudiocdsrc.h" line="34"/>
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<parameters>
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<parameter name="klass">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_CD_SRC_GET_CLASS"
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c:identifier="GST_AUDIO_CD_SRC_GET_CLASS"
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introspectable="0">
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<source-position filename="gstaudiocdsrc.h" line="37"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
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<constant name="AUDIO_CHANNELS_RANGE"
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value="(int) [ 1, max ]"
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c:type="GST_AUDIO_CHANNELS_RANGE">
|
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<doc xml:space="preserve"
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filename="audio-format.h"
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line="298">Maximum range of allowed channels, for use in template caps strings.</doc>
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<source-position filename="audio-format.h" line="303"/>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<function-macro name="AUDIO_CHANNEL_POSITION_MASK"
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c:identifier="GST_AUDIO_CHANNEL_POSITION_MASK"
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introspectable="0">
|
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<source-position filename="audio-channels.h" line="132"/>
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<parameters>
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<parameter name="pos">
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</parameter>
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</parameters>
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</function-macro>
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<function-macro name="AUDIO_CLOCK"
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c:identifier="GST_AUDIO_CLOCK"
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introspectable="0">
|
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<source-position filename="gstaudioclock.h" line="36"/>
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<parameters>
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<parameter name="obj">
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</parameter>
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</parameters>
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</function-macro>
|
|
<function-macro name="AUDIO_CLOCK_CAST"
|
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c:identifier="GST_AUDIO_CLOCK_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioclock.h" line="44"/>
|
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<parameters>
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<parameter name="obj">
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</parameter>
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|
</parameters>
|
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</function-macro>
|
|
<function-macro name="AUDIO_CLOCK_CLASS"
|
|
c:identifier="GST_AUDIO_CLOCK_CLASS"
|
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introspectable="0">
|
|
<source-position filename="gstaudioclock.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="klass">
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</parameter>
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|
</parameters>
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|
</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"
|
|
filename="audio-converter.h"
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line="42">#GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
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changing bit depth.
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Default is #GST_AUDIO_DITHER_NONE.</doc>
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<source-position filename="audio-converter.h" line="49"/>
|
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_MIX_MATRIX"
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value="GstAudioConverter.mix-matrix"
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c:type="GST_AUDIO_CONVERTER_OPT_MIX_MATRIX">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
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line="69">#GST_TYPE_VALUE_LIST, The channel mapping matrix.
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The matrix coefficients must be between -1 and 1: the number of rows is equal
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to the number of output channels and the number of columns is equal to the
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number of input channels.
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## Example matrix generation code
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To generate the matrix using code:
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|[
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GValue v = G_VALUE_INIT;
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GValue v2 = G_VALUE_INIT;
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GValue v3 = G_VALUE_INIT;
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g_value_init (&v2, GST_TYPE_ARRAY);
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g_value_init (&v3, G_TYPE_DOUBLE);
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g_value_set_double (&v3, 1);
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gst_value_array_append_value (&v2, &v3);
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g_value_unset (&v3);
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[ Repeat for as many double as your input channels - unset and reinit v3 ]
|
|
g_value_init (&v, GST_TYPE_ARRAY);
|
|
gst_value_array_append_value (&v, &v2);
|
|
g_value_unset (&v2);
|
|
[ Repeat for as many v2's as your output channels - unset and reinit v2]
|
|
g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
|
|
g_value_unset (&v);
|
|
]|</doc>
|
|
<source-position filename="audio-converter.h" line="100"/>
|
|
<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"
|
|
filename="audio-converter.h"
|
|
line="51">#GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
|
|
to mask noise from quantization errors.
|
|
Default is #GST_AUDIO_NOISE_SHAPING_NONE.</doc>
|
|
<source-position filename="audio-converter.h" line="58"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_CONVERTER_OPT_QUANTIZATION"
|
|
value="GstAudioConverter.quantization"
|
|
c:type="GST_AUDIO_CONVERTER_OPT_QUANTIZATION">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
|
|
line="60">#G_TYPE_UINT, The quantization amount. Components will be
|
|
quantized to multiples of this value.
|
|
Default is 1</doc>
|
|
<source-position filename="audio-converter.h" line="67"/>
|
|
<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"
|
|
filename="audio-converter.h"
|
|
line="33">#GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
|
|
changing sample rates.
|
|
Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc>
|
|
<source-position filename="audio-converter.h" line="40"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_DECODER"
|
|
c:identifier="GST_AUDIO_DECODER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="37"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_CAST"
|
|
c:identifier="GST_AUDIO_DECODER_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="47"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_CLASS"
|
|
c:identifier="GST_AUDIO_DECODER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="39"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_ERROR"
|
|
c:identifier="GST_AUDIO_DECODER_ERROR"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="113">Utility function that audio decoder elements can use in case they encountered
|
|
a data processing error that may be fatal for the current "data unit" but
|
|
need not prevent subsequent decoding. Such errors are counted and if there
|
|
are too many, as configured in the context's max_errors, the pipeline will
|
|
post an error message and the application will be requested to stop further
|
|
media processing. Otherwise, it is considered a "glitch" and only a warning
|
|
is logged. In either case, @ret is set to the proper value to
|
|
return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="133"/>
|
|
<parameters>
|
|
<parameter name="el">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="115">the base audio decoder element that generates the error</doc>
|
|
</parameter>
|
|
<parameter name="weight">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="116">element defined weight of the error, added to error count</doc>
|
|
</parameter>
|
|
<parameter name="domain">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="117">like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)</doc>
|
|
</parameter>
|
|
<parameter name="code">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="118">error code defined for that domain (see #gstreamer-GstGError)</doc>
|
|
</parameter>
|
|
<parameter name="text">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="119">the message to display (format string and args enclosed in
|
|
parentheses)</doc>
|
|
</parameter>
|
|
<parameter name="debug">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="121">debugging information for the message (format string and args
|
|
enclosed in parentheses)</doc>
|
|
</parameter>
|
|
<parameter name="ret">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="123">variable to receive return value</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_GET_CLASS"
|
|
c:identifier="GST_AUDIO_DECODER_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="41"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_INPUT_SEGMENT"
|
|
c:identifier="GST_AUDIO_DECODER_INPUT_SEGMENT"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="83">Gives the input segment of the element.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="88"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="85">audio decoder instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_DECODER_MAX_ERRORS"
|
|
value="10"
|
|
c:type="GST_AUDIO_DECODER_MAX_ERRORS">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="145">Default maximum number of errors tolerated before signaling error.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="149"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_DECODER_OUTPUT_SEGMENT"
|
|
c:identifier="GST_AUDIO_DECODER_OUTPUT_SEGMENT"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="91">Gives the output segment of the element.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="96"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="93">audio decoder instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_DECODER_SINK_NAME"
|
|
value="sink"
|
|
c:type="GST_AUDIO_DECODER_SINK_NAME">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="51">The name of the templates for the sink pad.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="55"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_DECODER_SINK_PAD"
|
|
c:identifier="GST_AUDIO_DECODER_SINK_PAD"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="72">Gives the pointer to the sink #GstPad object of the element.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="77"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="74">base audio codec instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_DECODER_SRC_NAME"
|
|
value="src"
|
|
c:type="GST_AUDIO_DECODER_SRC_NAME">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="57">The name of the templates for the source pad.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="61"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_DECODER_SRC_PAD"
|
|
c:identifier="GST_AUDIO_DECODER_SRC_PAD"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="64">Gives the pointer to the source #GstPad object of the element.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="69"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="66">base audio codec instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_STREAM_LOCK"
|
|
c:identifier="GST_AUDIO_DECODER_STREAM_LOCK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="79"/>
|
|
<parameters>
|
|
<parameter name="dec">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_STREAM_UNLOCK"
|
|
c:identifier="GST_AUDIO_DECODER_STREAM_UNLOCK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="80"/>
|
|
<parameters>
|
|
<parameter name="dec">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_DEF_CHANNELS"
|
|
value="2"
|
|
c:type="GST_AUDIO_DEF_CHANNELS">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="358">Standard number of channels used in consumer audio.</doc>
|
|
<source-position filename="audio-format.h" line="356"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_DEF_FORMAT"
|
|
value="S16LE"
|
|
c:type="GST_AUDIO_DEF_FORMAT">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="364">Standard format used in consumer audio.</doc>
|
|
<source-position filename="audio-format.h" line="362"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_DEF_RATE" value="44100" c:type="GST_AUDIO_DEF_RATE">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="352">Standard sampling rate used in consumer audio.</doc>
|
|
<source-position filename="audio-format.h" line="350"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_ENCODER"
|
|
c:identifier="GST_AUDIO_ENCODER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="34"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_CAST"
|
|
c:identifier="GST_AUDIO_ENCODER_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="39"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_CLASS"
|
|
c:identifier="GST_AUDIO_ENCODER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="35"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_GET_CLASS"
|
|
c:identifier="GST_AUDIO_ENCODER_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_INPUT_SEGMENT"
|
|
c:identifier="GST_AUDIO_ENCODER_INPUT_SEGMENT"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="70">Gives the input segment of the element.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="76"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="72">base parse instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_OUTPUT_SEGMENT"
|
|
c:identifier="GST_AUDIO_ENCODER_OUTPUT_SEGMENT"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="78">Gives the output segment of the element.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="84"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="80">base parse instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_ENCODER_SINK_NAME"
|
|
value="sink"
|
|
c:type="GST_AUDIO_ENCODER_SINK_NAME">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="41">the name of the templates for the sink pad</doc>
|
|
<source-position filename="gstaudioencoder.h" line="46"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_ENCODER_SINK_PAD"
|
|
c:identifier="GST_AUDIO_ENCODER_SINK_PAD"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="62">Gives the pointer to the sink #GstPad object of the element.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="68"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="64">audio encoder instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_ENCODER_SRC_NAME"
|
|
value="src"
|
|
c:type="GST_AUDIO_ENCODER_SRC_NAME">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="47">the name of the templates for the source pad</doc>
|
|
<source-position filename="gstaudioencoder.h" line="52"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_ENCODER_SRC_PAD"
|
|
c:identifier="GST_AUDIO_ENCODER_SRC_PAD"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="54">Gives the pointer to the source #GstPad object of the element.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="60"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="56">audio encoder instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_STREAM_LOCK"
|
|
c:identifier="GST_AUDIO_ENCODER_STREAM_LOCK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="86"/>
|
|
<parameters>
|
|
<parameter name="enc">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_STREAM_UNLOCK"
|
|
c:identifier="GST_AUDIO_ENCODER_STREAM_UNLOCK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="87"/>
|
|
<parameters>
|
|
<parameter name="enc">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER"
|
|
c:identifier="GST_AUDIO_FILTER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_BPF"
|
|
c:identifier="GST_AUDIO_FILTER_BPF"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="71"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_BPS"
|
|
c:identifier="GST_AUDIO_FILTER_BPS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="72"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CAST"
|
|
c:identifier="GST_AUDIO_FILTER_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CHANNELS"
|
|
c:identifier="GST_AUDIO_FILTER_CHANNELS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="70"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CLASS"
|
|
c:identifier="GST_AUDIO_FILTER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="40"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CLASS_CAST"
|
|
c:identifier="GST_AUDIO_FILTER_CLASS_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="42"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_FORMAT"
|
|
c:identifier="GST_AUDIO_FILTER_FORMAT"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="68"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_GET_CLASS"
|
|
c:identifier="GST_AUDIO_FILTER_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="44"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_INFO"
|
|
c:identifier="GST_AUDIO_FILTER_INFO"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="66"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_RATE"
|
|
c:identifier="GST_AUDIO_FILTER_RATE"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="69"/>
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_FORMATS_ALL"
|
|
value=" { S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }"
|
|
c:type="GST_AUDIO_FORMATS_ALL">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="325">List of all audio formats, for use in template caps strings.</doc>
|
|
<source-position filename="audio-format.h" line="330"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_FORMAT_INFO_DEPTH"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_DEPTH"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="271"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_ENDIANNESS"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_ENDIANNESS"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="267"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_FLAGS"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_FLAGS"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="261"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_FORMAT"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_FORMAT"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="259"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_IS_BIG_ENDIAN"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="269"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_IS_FLOAT"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_IS_FLOAT"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="264"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_IS_INTEGER"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_IS_INTEGER"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="263"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="268"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_IS_SIGNED"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_IS_SIGNED"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="265"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_NAME"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_NAME"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="260"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_WIDTH"
|
|
c:identifier="GST_AUDIO_FORMAT_INFO_WIDTH"
|
|
introspectable="0">
|
|
<source-position filename="audio-format.h" line="270"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_BPF"
|
|
c:identifier="GST_AUDIO_INFO_BPF"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="113"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_BPS"
|
|
c:identifier="GST_AUDIO_INFO_BPS"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="97"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_CHANNELS"
|
|
c:identifier="GST_AUDIO_INFO_CHANNELS"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="112"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_DEPTH"
|
|
c:identifier="GST_AUDIO_INFO_DEPTH"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="96"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_ENDIANNESS"
|
|
c:identifier="GST_AUDIO_INFO_ENDIANNESS"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="103"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_FLAGS"
|
|
c:identifier="GST_AUDIO_INFO_FLAGS"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="107"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_FORMAT"
|
|
c:identifier="GST_AUDIO_INFO_FORMAT"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="93"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_BIG_ENDIAN"
|
|
c:identifier="GST_AUDIO_INFO_IS_BIG_ENDIAN"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="105"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_FLOAT"
|
|
c:identifier="GST_AUDIO_INFO_IS_FLOAT"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="100"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_INTEGER"
|
|
c:identifier="GST_AUDIO_INFO_IS_INTEGER"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="99"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_LITTLE_ENDIAN"
|
|
c:identifier="GST_AUDIO_INFO_IS_LITTLE_ENDIAN"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="104"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_SIGNED"
|
|
c:identifier="GST_AUDIO_INFO_IS_SIGNED"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="101"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_UNPOSITIONED"
|
|
c:identifier="GST_AUDIO_INFO_IS_UNPOSITIONED"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="108"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_VALID"
|
|
c:identifier="GST_AUDIO_INFO_IS_VALID"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="91"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_LAYOUT"
|
|
c:identifier="GST_AUDIO_INFO_LAYOUT"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="109"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_NAME"
|
|
c:identifier="GST_AUDIO_INFO_NAME"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="94"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_POSITION"
|
|
c:identifier="GST_AUDIO_INFO_POSITION"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="114"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
<parameter name="c">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_RATE"
|
|
c:identifier="GST_AUDIO_INFO_RATE"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="111"/>
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_WIDTH"
|
|
c:identifier="GST_AUDIO_INFO_WIDTH"
|
|
introspectable="0">
|
|
<source-position filename="audio-info.h" line="95"/>
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_NE"
|
|
c:identifier="GST_AUDIO_NE"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="305">Turns audio format string @s into the format string for native endianness.</doc>
|
|
<source-position filename="audio-format.h" line="318"/>
|
|
<parameters>
|
|
<parameter name="s">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="307">format string without endianness marker</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_OE"
|
|
c:identifier="GST_AUDIO_OE"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="311">Turns audio format string @s into the format string for other endianness.</doc>
|
|
<source-position filename="audio-format.h" line="319"/>
|
|
<parameters>
|
|
<parameter name="s">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="313">format string without endianness marker</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_RATE_RANGE"
|
|
value="(int) [ 1, max ]"
|
|
c:type="GST_AUDIO_RATE_RANGE">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="292">Maximum range of allowed sample rates, for use in template caps strings.</doc>
|
|
<source-position filename="audio-format.h" line="297"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_CUBIC_B"
|
|
value="GstAudioResampler.cubic-b"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_B">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="51">G_TYPE_DOUBLE, B parameter of the cubic filter.
|
|
Values between 0.0 and 2.0 are accepted. 1.0 is the default.
|
|
|
|
Below are some values of popular filters:
|
|
B C
|
|
Hermite 0.0 0.0
|
|
Spline 1.0 0.0
|
|
Catmull-Rom 0.0 1/2</doc>
|
|
<source-position filename="audio-resampler.h" line="63"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_CUBIC_C"
|
|
value="GstAudioResampler.cubic-c"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_C">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="64">G_TYPE_DOUBLE, C parameter of the cubic filter.
|
|
Values between 0.0 and 2.0 are accepted. 0.0 is the default.
|
|
|
|
See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values</doc>
|
|
<source-position filename="audio-resampler.h" line="72"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_CUTOFF"
|
|
value="GstAudioResampler.cutoff"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_CUTOFF">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="30">G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="35"/>
|
|
<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"
|
|
filename="audio-resampler.h"
|
|
line="131">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
|
|
interpolated.
|
|
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc>
|
|
<source-position filename="audio-resampler.h" line="138"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE"
|
|
value="GstAudioResampler.filter-mode"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="99">GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
|
|
constructed.
|
|
GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="106"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD"
|
|
value="GstAudioResampler.filter-mode-threshold"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="107">G_TYPE_UINT: the amount of memory to use for full filter tables before
|
|
switching to interpolated filter tables.
|
|
1048576 is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="114"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE"
|
|
value="GstAudioResampler.filter-oversample"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="139">G_TYPE_UINT, oversampling to use when interpolating filters
|
|
8 is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="145"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR"
|
|
value="GstAudioResampler.max-phase-error"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="147">G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
|
|
rates.
|
|
0.1 is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="154"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_N_TAPS"
|
|
value="GstAudioResampler.n-taps"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_N_TAPS">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="74">G_TYPE_INT: the number of taps to use for the filter.
|
|
0 is the default and selects the taps automatically.</doc>
|
|
<source-position filename="audio-resampler.h" line="80"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"
|
|
value="GstAudioResampler.stop-attenutation"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="36">G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
|
|
after the stopband for the kaiser window. 85 dB is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="42"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"
|
|
value="GstAudioResampler.transition-bandwidth"
|
|
c:type="GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="43">G_TYPE_DOUBLE, transition bandwidth. The width of the
|
|
transition band for the kaiser window. 0.087 is the default.</doc>
|
|
<source-position filename="audio-resampler.h" line="49"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_DEFAULT"
|
|
value="4"
|
|
c:type="GST_AUDIO_RESAMPLER_QUALITY_DEFAULT">
|
|
<source-position filename="audio-resampler.h" line="202"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_MAX"
|
|
value="10"
|
|
c:type="GST_AUDIO_RESAMPLER_QUALITY_MAX">
|
|
<source-position filename="audio-resampler.h" line="201"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_MIN"
|
|
value="0"
|
|
c:type="GST_AUDIO_RESAMPLER_QUALITY_MIN">
|
|
<source-position filename="audio-resampler.h" line="200"/>
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_RING_BUFFER"
|
|
c:identifier="GST_AUDIO_RING_BUFFER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="33"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_BROADCAST"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_BROADCAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="162"/>
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_CAST"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_CAST"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_CLASS"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="34"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_GET_CLASS"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="35"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_GET_COND"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_GET_COND"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="159"/>
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_SIGNAL"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_SIGNAL"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="161"/>
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_WAIT"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_WAIT"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="160"/>
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK"
|
|
c:identifier="GST_AUDIO_SINK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosink.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK_CLASS"
|
|
c:identifier="GST_AUDIO_SINK_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosink.h" line="37"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK_GET_CLASS"
|
|
c:identifier="GST_AUDIO_SINK_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosink.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC"
|
|
c:identifier="GST_AUDIO_SRC"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosrc.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC_CLASS"
|
|
c:identifier="GST_AUDIO_SRC_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosrc.h" line="37"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC_GET_CLASS"
|
|
c:identifier="GST_AUDIO_SRC_GET_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosrc.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<class name="AudioAggregator"
|
|
c:symbol-prefix="audio_aggregator"
|
|
c:type="GstAudioAggregator"
|
|
version="1.14"
|
|
parent="GstBase.Aggregator"
|
|
abstract="1"
|
|
glib:type-name="GstAudioAggregator"
|
|
glib:get-type="gst_audio_aggregator_get_type"
|
|
glib:type-struct="AudioAggregatorClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioaggregator.c"
|
|
line="26">Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
|
|
their source and sink pads,
|
|
gst_element_class_add_static_pad_template_with_gtype() is a convenient
|
|
helper.
|
|
|
|
#GstAudioAggregator can perform conversion on the data arriving
|
|
on its sink pads, based on the format expected downstream: in order
|
|
to enable that behaviour, the GType of the sink pads must either be
|
|
a (subclass of) #GstAudioAggregatorConvertPad to use the default
|
|
#GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
|
|
implementing #GstAudioAggregatorPadClass.convert_buffer.
|
|
|
|
To allow for the output caps to change, the mechanism is the same as
|
|
above, with the GType of the source pad.
|
|
|
|
See #GstAudioMixer for an example.
|
|
|
|
When conversion is enabled, #GstAudioAggregator will accept
|
|
any type of raw audio caps and perform conversion
|
|
on the data arriving on its sink pads, with whatever downstream
|
|
expects as the target format.
|
|
|
|
In case downstream caps are not fully fixated, it will use
|
|
the first configured sink pad to finish fixating its source pad
|
|
caps.
|
|
|
|
A notable exception for now is the sample rate, sink pads must
|
|
have the same sample rate as either the downstream requirement,
|
|
or the first configured pad, or a combination of both (when
|
|
downstream specifies a range or a set of acceptable rates).</doc>
|
|
<source-position filename="gstaudioaggregator.h" line="212"/>
|
|
<virtual-method name="aggregate_one_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="206"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="aagg" transfer-ownership="none">
|
|
<type name="AudioAggregator" c:type="GstAudioAggregator*"/>
|
|
</instance-parameter>
|
|
<parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="in_offset" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="out_offset" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="num_frames" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="create_output_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="204"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="aagg" transfer-ownership="none">
|
|
<type name="AudioAggregator" c:type="GstAudioAggregator*"/>
|
|
</instance-parameter>
|
|
<parameter name="num_frames" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="set_sink_caps"
|
|
c:identifier="gst_audio_aggregator_set_sink_caps">
|
|
<source-position filename="gstaudioaggregator.h" line="222"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="aagg" transfer-ownership="none">
|
|
<type name="AudioAggregator" c:type="GstAudioAggregator*"/>
|
|
</instance-parameter>
|
|
<parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="alignment-threshold"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="discont-wait" writable="1" transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="output-buffer-duration"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<field name="parent">
|
|
<type name="GstBase.Aggregator" c:type="GstAggregator"/>
|
|
</field>
|
|
<field name="current_caps">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioaggregator.h"
|
|
line="171">The caps set by the subclass</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioAggregatorPrivate"
|
|
c:type="GstAudioAggregatorPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioAggregatorClass"
|
|
c:type="GstAudioAggregatorClass"
|
|
glib:is-gtype-struct-for="AudioAggregator"
|
|
version="1.14">
|
|
<source-position filename="gstaudioaggregator.h" line="212"/>
|
|
<field name="parent_class">
|
|
<type name="GstBase.AggregatorClass" c:type="GstAggregatorClass"/>
|
|
</field>
|
|
<field name="create_output_buffer">
|
|
<callback name="create_output_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="204"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="aagg" transfer-ownership="none">
|
|
<type name="AudioAggregator" c:type="GstAudioAggregator*"/>
|
|
</parameter>
|
|
<parameter name="num_frames" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="aggregate_one_buffer">
|
|
<callback name="aggregate_one_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="206"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="aagg" transfer-ownership="none">
|
|
<type name="AudioAggregator" c:type="GstAudioAggregator*"/>
|
|
</parameter>
|
|
<parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="in_offset" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="out_offset" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="num_frames" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="20">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="AudioAggregatorConvertPad"
|
|
c:symbol-prefix="audio_aggregator_convert_pad"
|
|
c:type="GstAudioAggregatorConvertPad"
|
|
version="1.14"
|
|
parent="AudioAggregatorPad"
|
|
glib:type-name="GstAudioAggregatorConvertPad"
|
|
glib:get-type="gst_audio_aggregator_convert_pad_get_type"
|
|
glib:type-struct="AudioAggregatorConvertPadClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioaggregator.h"
|
|
line="123">An implementation of GstPad that can be used with #GstAudioAggregator.
|
|
|
|
See #GstAudioAggregator for more details.</doc>
|
|
<source-position filename="gstaudioaggregator.h" line="153"/>
|
|
<property name="converter-config" writable="1" transfer-ownership="none">
|
|
<type name="Gst.Structure"/>
|
|
</property>
|
|
<field name="parent" readable="0" private="1">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioAggregatorConvertPadPrivate"
|
|
c:type="GstAudioAggregatorConvertPadPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioAggregatorConvertPadClass"
|
|
c:type="GstAudioAggregatorConvertPadClass"
|
|
glib:is-gtype-struct-for="AudioAggregatorConvertPad"
|
|
version="1.14">
|
|
<source-position filename="gstaudioaggregator.h" line="153"/>
|
|
<field name="parent_class">
|
|
<type name="AudioAggregatorPadClass"
|
|
c:type="GstAudioAggregatorPadClass"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioAggregatorConvertPadPrivate"
|
|
c:type="GstAudioAggregatorConvertPadPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudioaggregator.h" line="121"/>
|
|
</record>
|
|
<class name="AudioAggregatorPad"
|
|
c:symbol-prefix="audio_aggregator_pad"
|
|
c:type="GstAudioAggregatorPad"
|
|
version="1.14"
|
|
parent="GstBase.AggregatorPad"
|
|
glib:type-name="GstAudioAggregatorPad"
|
|
glib:get-type="gst_audio_aggregator_pad_get_type"
|
|
glib:type-struct="AudioAggregatorPadClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioaggregator.h"
|
|
line="60">The default implementation of GstPad used with #GstAudioAggregator</doc>
|
|
<source-position filename="gstaudioaggregator.h" line="103"/>
|
|
<virtual-method name="convert_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="94"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="out_info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="update_conversion_info">
|
|
<source-position filename="gstaudioaggregator.h" line="99"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<field name="parent">
|
|
<type name="GstBase.AggregatorPad" c:type="GstAggregatorPad"/>
|
|
</field>
|
|
<field name="info">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioaggregator.h"
|
|
line="62">The audio info for this pad set from the incoming caps</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioAggregatorPadPrivate"
|
|
c:type="GstAudioAggregatorPadPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioAggregatorPadClass"
|
|
c:type="GstAudioAggregatorPadClass"
|
|
glib:is-gtype-struct-for="AudioAggregatorPad"
|
|
version="1.14">
|
|
<source-position filename="gstaudioaggregator.h" line="103"/>
|
|
<field name="parent_class">
|
|
<type name="GstBase.AggregatorPadClass"
|
|
c:type="GstAggregatorPadClass"/>
|
|
</field>
|
|
<field name="convert_buffer">
|
|
<callback name="convert_buffer">
|
|
<source-position filename="gstaudioaggregator.h" line="94"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</parameter>
|
|
<parameter name="in_info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="out_info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="update_conversion_info">
|
|
<callback name="update_conversion_info">
|
|
<source-position filename="gstaudioaggregator.h" line="99"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="pad" transfer-ownership="none">
|
|
<type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="20">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioAggregatorPadPrivate"
|
|
c:type="GstAudioAggregatorPadPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudioaggregator.h" line="58"/>
|
|
</record>
|
|
<record name="AudioAggregatorPrivate"
|
|
c:type="GstAudioAggregatorPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudioaggregator.h" line="37"/>
|
|
</record>
|
|
<class name="AudioBaseSink"
|
|
c:symbol-prefix="audio_base_sink"
|
|
c:type="GstAudioBaseSink"
|
|
parent="GstBase.BaseSink"
|
|
glib:type-name="GstAudioBaseSink"
|
|
glib:get-type="gst_audio_base_sink_get_type"
|
|
glib:type-struct="AudioBaseSinkClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="23">This is the base class for audio sinks. Subclasses need to implement the
|
|
::create_ringbuffer vmethod. This base class will then take care of
|
|
writing samples to the ringbuffer, synchronisation, clipping and flushing.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="221"/>
|
|
<virtual-method name="create_ringbuffer" invoker="create_ringbuffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2246">Create and return the #GstAudioRingBuffer for @sink. This function will
|
|
call the ::create_ringbuffer vmethod and will set @sink as the parent of
|
|
the returned buffer (see gst_object_set_parent()).</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="214"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2254">The new ringbuffer of @sink.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2248">a #GstAudioBaseSink.</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="payload">
|
|
<source-position filename="gstaudiobasesink.h" line="217"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="create_ringbuffer"
|
|
c:identifier="gst_audio_base_sink_create_ringbuffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2246">Create and return the #GstAudioRingBuffer for @sink. This function will
|
|
call the ::create_ringbuffer vmethod and will set @sink as the parent of
|
|
the returned buffer (see gst_object_set_parent()).</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="228"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2254">The new ringbuffer of @sink.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2248">a #GstAudioBaseSink.</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_alignment_threshold"
|
|
c:identifier="gst_audio_base_sink_get_alignment_threshold">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="694">Get the current alignment threshold, in nanoseconds, used by @sink.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="254"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="700">The current alignment threshold used by @sink.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="696">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_discont_wait"
|
|
c:identifier="gst_audio_base_sink_get_discont_wait">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="801">Get the current discont wait, in nanoseconds, used by @sink.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="261"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="807">The current discont wait used by @sink.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="803">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_drift_tolerance"
|
|
c:identifier="gst_audio_base_sink_get_drift_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="654">Get the current drift tolerance, in microseconds, used by @sink.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="247"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="660">The current drift tolerance used by @sink.</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="656">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_provide_clock"
|
|
c:identifier="gst_audio_base_sink_get_provide_clock">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="572">Queries whether @sink will provide a clock or not. See also
|
|
gst_audio_base_sink_set_provide_clock.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="234"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="579">%TRUE if @sink will provide a clock.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="574">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_slave_method"
|
|
c:identifier="gst_audio_base_sink_get_slave_method">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="613">Get the current slave method used by @sink.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="241"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="619">The current slave method used by @sink.</doc>
|
|
<type name="AudioBaseSinkSlaveMethod"
|
|
c:type="GstAudioBaseSinkSlaveMethod"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="615">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="report_device_failure"
|
|
c:identifier="gst_audio_base_sink_report_device_failure"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="778">Informs this base class that the audio output device has failed for
|
|
some reason, causing a discontinuity (for example, because the device
|
|
recovered from the error, but lost all contents of its ring buffer).
|
|
This function is typically called by derived classes, and is useful
|
|
for the custom slave method.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="271"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="780">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_alignment_threshold"
|
|
c:identifier="gst_audio_base_sink_set_alignment_threshold">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="676">Controls the sink's alignment threshold.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="250"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="678">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="679">the new alignment threshold in nanoseconds</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_custom_slaving_callback"
|
|
c:identifier="gst_audio_base_sink_set_custom_slaving_callback"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="734">Sets the custom slaving callback. This callback will
|
|
be invoked if the slave-method property is set to
|
|
GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink
|
|
receives and plays samples.
|
|
|
|
Setting the callback to NULL causes the sink to
|
|
behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE
|
|
method were used.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="265"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="736">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="callback"
|
|
transfer-ownership="none"
|
|
scope="notified"
|
|
closure="1"
|
|
destroy="2">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="737">a #GstAudioBaseSinkCustomSlavingCallback</doc>
|
|
<type name="AudioBaseSinkCustomSlavingCallback"
|
|
c:type="GstAudioBaseSinkCustomSlavingCallback"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="738">user data passed to the callback</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="notify" transfer-ownership="none" scope="async">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="739">called when user_data becomes unused</doc>
|
|
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_discont_wait"
|
|
c:identifier="gst_audio_base_sink_set_discont_wait">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="716">Controls how long the sink will wait before creating a discontinuity.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="257"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="718">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont_wait" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="719">the new discont wait in nanoseconds</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_drift_tolerance"
|
|
c:identifier="gst_audio_base_sink_set_drift_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="636">Controls the sink's drift tolerance.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="244"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="638">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="drift_tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="639">the new drift tolerance in microseconds</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_provide_clock"
|
|
c:identifier="gst_audio_base_sink_set_provide_clock">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="548">Controls whether @sink will provide a clock or not. If @provide is %TRUE,
|
|
gst_element_provide_clock() will return a clock that reflects the datarate
|
|
of @sink. If @provide is %FALSE, gst_element_provide_clock() will return
|
|
NULL.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="231"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="550">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="provide" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="551">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_slave_method"
|
|
c:identifier="gst_audio_base_sink_set_slave_method">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="595">Controls how clock slaving will be performed in @sink.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="237"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="597">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="598">the new slave method</doc>
|
|
<type name="AudioBaseSinkSlaveMethod"
|
|
c:type="GstAudioBaseSinkSlaveMethod"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="alignment-threshold"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="buffer-time" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="can-activate-pull"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="discont-wait" writable="1" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="237">A window of time in nanoseconds to wait before creating a discontinuity as
|
|
a result of breaching the drift-tolerance.</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</property>
|
|
<property name="drift-tolerance" writable="1" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="214">Controls the amount of time in microseconds that clocks are allowed
|
|
to drift before resynchronisation happens.</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="latency-time" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="provide-clock" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="slave-method" writable="1" transfer-ownership="none">
|
|
<type name="AudioBaseSinkSlaveMethod"/>
|
|
</property>
|
|
<field name="element">
|
|
<type name="GstBase.BaseSink" c:type="GstBaseSink"/>
|
|
</field>
|
|
<field name="ringbuffer">
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</field>
|
|
<field name="buffer_time">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="latency_time">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="next_sample">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="provided_clock">
|
|
<type name="Gst.Clock" c:type="GstClock*"/>
|
|
</field>
|
|
<field name="eos_rendering">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioBaseSinkPrivate" c:type="GstAudioBaseSinkPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioBaseSinkClass"
|
|
c:type="GstAudioBaseSinkClass"
|
|
glib:is-gtype-struct-for="AudioBaseSink">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="198">#GstAudioBaseSink class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="221"/>
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="200">the parent class.</doc>
|
|
<type name="GstBase.BaseSinkClass" c:type="GstBaseSinkClass"/>
|
|
</field>
|
|
<field name="create_ringbuffer">
|
|
<callback name="create_ringbuffer">
|
|
<source-position filename="gstaudiobasesink.h" line="214"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2254">The new ringbuffer of @sink.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.c"
|
|
line="2248">a #GstAudioBaseSink.</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="payload">
|
|
<callback name="payload">
|
|
<source-position filename="gstaudiobasesink.h" line="217"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<callback name="AudioBaseSinkCustomSlavingCallback"
|
|
c:type="GstAudioBaseSinkCustomSlavingCallback"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="130">This function is set with gst_audio_base_sink_set_custom_slaving_callback()
|
|
and is called during playback. It receives the current time of external and
|
|
internal clocks, which the callback can then use to apply any custom
|
|
slaving/synchronization schemes.
|
|
|
|
The external clock is the sink's element clock, the internal one is the
|
|
internal audio clock. The internal audio clock's calibration is applied to
|
|
the timestamps before they are passed to the callback. The difference between
|
|
etime and itime is the skew; how much internal and external clock lie apart
|
|
from each other. A skew of 0 means both clocks are perfectly in sync.
|
|
itime > etime means the external clock is going slower, while itime < etime
|
|
means it is going faster than the internal clock. etime and itime are always
|
|
valid timestamps, except for when a discontinuity happens.
|
|
|
|
requested_skew is an output value the callback can write to. It informs the
|
|
sink of whether or not it should move the playout pointer, and if so, by how
|
|
much. This pointer is only NULL if a discontinuity occurs; otherwise, it is
|
|
safe to write to *requested_skew. The default skew is 0.
|
|
|
|
The sink may experience discontinuities. If one happens, discont is TRUE,
|
|
itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
|
|
This makes it possible to reset custom clock slaving algorithms when a
|
|
discontinuity happens.</doc>
|
|
<source-position filename="gstaudiobasesink.h" line="165"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="132">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</parameter>
|
|
<parameter name="etime" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="133">external clock time</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="itime" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="134">internal clock time</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="requested_skew" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="135">skew amount requested by the callback</doc>
|
|
<type name="Gst.ClockTimeDiff" c:type="GstClockTimeDiff*"/>
|
|
</parameter>
|
|
<parameter name="discont_reason" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="136">reason for discontinuity (if any)</doc>
|
|
<type name="AudioBaseSinkDiscontReason"
|
|
c:type="GstAudioBaseSinkDiscontReason"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1"
|
|
closure="5">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="137">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<enumeration name="AudioBaseSinkDiscontReason"
|
|
version="1.6"
|
|
glib:type-name="GstAudioBaseSinkDiscontReason"
|
|
glib:get-type="gst_audio_base_sink_discont_reason_get_type"
|
|
c:type="GstAudioBaseSinkDiscontReason">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="106">Different possible reasons for discontinuities. This enum is useful for the custom
|
|
slave method.</doc>
|
|
<member name="no_discont"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT"
|
|
glib:nick="no-discont">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="108">No discontinuity occurred</doc>
|
|
</member>
|
|
<member name="new_caps"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS"
|
|
glib:nick="new-caps">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="109">New caps are set, causing renegotiotion</doc>
|
|
</member>
|
|
<member name="flush"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH"
|
|
glib:nick="flush">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="110">Samples have been flushed</doc>
|
|
</member>
|
|
<member name="sync_latency"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY"
|
|
glib:nick="sync-latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="111">Sink was synchronized to the estimated latency (occurs during initialization)</doc>
|
|
</member>
|
|
<member name="alignment"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT"
|
|
glib:nick="alignment">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="112">Aligning buffers failed because the timestamps are too discontinuous</doc>
|
|
</member>
|
|
<member name="device_failure"
|
|
value="5"
|
|
c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE"
|
|
glib:nick="device-failure">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="113">Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure())</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioBaseSinkPrivate"
|
|
c:type="GstAudioBaseSinkPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudiobasesink.h" line="104"/>
|
|
</record>
|
|
<enumeration name="AudioBaseSinkSlaveMethod"
|
|
glib:type-name="GstAudioBaseSinkSlaveMethod"
|
|
glib:get-type="gst_audio_base_sink_slave_method_get_type"
|
|
c:type="GstAudioBaseSinkSlaveMethod">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="83">Different possible clock slaving algorithms used when the internal audio
|
|
clock is not selected as the pipeline master clock.</doc>
|
|
<member name="resample"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE"
|
|
glib:nick="resample">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="85">Resample to match the master clock</doc>
|
|
</member>
|
|
<member name="skew"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_BASE_SINK_SLAVE_SKEW"
|
|
glib:nick="skew">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="86">Adjust playout pointer when master clock
|
|
drifts too much.</doc>
|
|
</member>
|
|
<member name="none"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_BASE_SINK_SLAVE_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="88">No adjustment is done.</doc>
|
|
</member>
|
|
<member name="custom"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_BASE_SINK_SLAVE_CUSTOM"
|
|
glib:nick="custom">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesink.h"
|
|
line="89">Use custom clock slaving algorithm (Since: 1.6)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="AudioBaseSrc"
|
|
c:symbol-prefix="audio_base_src"
|
|
c:type="GstAudioBaseSrc"
|
|
parent="GstBase.PushSrc"
|
|
glib:type-name="GstAudioBaseSrc"
|
|
glib:get-type="gst_audio_base_src_get_type"
|
|
glib:type-struct="AudioBaseSrcClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="23">This is the base class for audio sources. Subclasses need to implement the
|
|
::create_ringbuffer vmethod. This base class will then take care of
|
|
reading samples from the ringbuffer, synchronisation and flushing.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="133"/>
|
|
<virtual-method name="create_ringbuffer" invoker="create_ringbuffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1075">Create and return the #GstAudioRingBuffer for @src. This function will call
|
|
the ::create_ringbuffer vmethod and will set @src as the parent of the
|
|
returned buffer (see gst_object_set_parent()).</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="129"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1083">The new ringbuffer of @src.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1077">a #GstAudioBaseSrc.</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="create_ringbuffer"
|
|
c:identifier="gst_audio_base_src_create_ringbuffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1075">Create and return the #GstAudioRingBuffer for @src. This function will call
|
|
the ::create_ringbuffer vmethod and will set @src as the parent of the
|
|
returned buffer (see gst_object_set_parent()).</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="140"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1083">The new ringbuffer of @src.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1077">a #GstAudioBaseSrc.</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_provide_clock"
|
|
c:identifier="gst_audio_base_src_get_provide_clock">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="354">Queries whether @src will provide a clock or not. See also
|
|
gst_audio_base_src_set_provide_clock.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="146"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="361">%TRUE if @src will provide a clock.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="356">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_slave_method"
|
|
c:identifier="gst_audio_base_src_get_slave_method">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="395">Get the current slave method used by @src.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="153"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="401">The current slave method used by @src.</doc>
|
|
<type name="AudioBaseSrcSlaveMethod"
|
|
c:type="GstAudioBaseSrcSlaveMethod"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="397">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_provide_clock"
|
|
c:identifier="gst_audio_base_src_set_provide_clock">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="332">Controls whether @src will provide a clock or not. If @provide is %TRUE,
|
|
gst_element_provide_clock() will return a clock that reflects the datarate
|
|
of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="143"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="334">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="provide" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="335">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_slave_method"
|
|
c:identifier="gst_audio_base_src_set_slave_method">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="377">Controls how clock slaving will be performed in @src.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="149"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="379">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="380">the new slave method</doc>
|
|
<type name="AudioBaseSrcSlaveMethod"
|
|
c:type="GstAudioBaseSrcSlaveMethod"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="actual-buffer-time" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="159">Actual configured size of audio buffer in microseconds.</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="actual-latency-time" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="170">Actual configured audio latency in microseconds.</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="buffer-time" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="latency-time" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="provide-clock" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="slave-method" writable="1" transfer-ownership="none">
|
|
<type name="AudioBaseSrcSlaveMethod"/>
|
|
</property>
|
|
<field name="element">
|
|
<type name="GstBase.PushSrc" c:type="GstPushSrc"/>
|
|
</field>
|
|
<field name="ringbuffer">
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</field>
|
|
<field name="buffer_time">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</field>
|
|
<field name="latency_time">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</field>
|
|
<field name="next_sample">
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="clock">
|
|
<type name="Gst.Clock" c:type="GstClock*"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioBaseSrcPrivate" c:type="GstAudioBaseSrcPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioBaseSrcClass"
|
|
c:type="GstAudioBaseSrcClass"
|
|
glib:is-gtype-struct-for="AudioBaseSrc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="117">#GstAudioBaseSrc class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
<source-position filename="gstaudiobasesrc.h" line="133"/>
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="119">the parent class.</doc>
|
|
<type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/>
|
|
</field>
|
|
<field name="create_ringbuffer">
|
|
<callback name="create_ringbuffer">
|
|
<source-position filename="gstaudiobasesrc.h" line="129"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1083">The new ringbuffer of @src.</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.c"
|
|
line="1077">a #GstAudioBaseSrc.</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioBaseSrcPrivate"
|
|
c:type="GstAudioBaseSrcPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudiobasesrc.h" line="63"/>
|
|
</record>
|
|
<enumeration name="AudioBaseSrcSlaveMethod"
|
|
glib:type-name="GstAudioBaseSrcSlaveMethod"
|
|
glib:get-type="gst_audio_base_src_slave_method_get_type"
|
|
c:type="GstAudioBaseSrcSlaveMethod">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="67">Different possible clock slaving algorithms when the internal audio clock was
|
|
not selected as the pipeline clock.</doc>
|
|
<member name="resample"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE"
|
|
glib:nick="resample">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="69">Resample to match the master clock.</doc>
|
|
</member>
|
|
<member name="re_timestamp"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP"
|
|
glib:nick="re-timestamp">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="70">Retimestamp output buffers with master
|
|
clock time.</doc>
|
|
</member>
|
|
<member name="skew"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_BASE_SRC_SLAVE_SKEW"
|
|
glib:nick="skew">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="72">Adjust capture pointer when master clock
|
|
drifts too much.</doc>
|
|
</member>
|
|
<member name="none"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_BASE_SRC_SLAVE_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiobasesrc.h"
|
|
line="74">No adjustment is done.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioBuffer" c:type="GstAudioBuffer" version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="30">A structure containing the result of an audio buffer map operation,
|
|
which is executed with gst_audio_buffer_map(). For non-interleaved (planar)
|
|
buffers, the beginning of each channel in the buffer has its own pointer in
|
|
the @planes array. For interleaved buffers, the @planes array only contains
|
|
one item, which is the pointer to the beginning of the buffer, and @n_planes
|
|
equals 1.
|
|
|
|
The different channels in @planes are always in the GStreamer channel order.</doc>
|
|
<source-position filename="audio-buffer.h" line="65"/>
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="32">a #GstAudioInfo describing the audio properties of this buffer</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="n_samples" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="33">the size of the buffer in samples</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</field>
|
|
<field name="n_planes" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="34">the number of planes available</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="planes" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="35">an array of @n_planes pointers pointing to the start of each
|
|
plane in the mapped buffer</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</field>
|
|
<field name="buffer" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.h"
|
|
line="37">the mapped buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</field>
|
|
<field name="map_infos" readable="0" private="1">
|
|
<type name="Gst.MapInfo" c:type="GstMapInfo*"/>
|
|
</field>
|
|
<field name="priv_planes_arr" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="8">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<field name="priv_map_infos_arr" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="8">
|
|
<type name="Gst.MapInfo" c:type="GstMapInfo"/>
|
|
</array>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<method name="map" c:identifier="gst_audio_buffer_map" version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="56">Maps an audio @gstbuffer so that it can be read or written and stores the
|
|
result of the map operation in @buffer.
|
|
|
|
This is especially useful when the @gstbuffer is in non-interleaved (planar)
|
|
layout, in which case this function will use the information in the
|
|
@gstbuffer's attached #GstAudioMeta in order to map each channel in a
|
|
separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached
|
|
on the @gstbuffer, then it must be in interleaved layout.
|
|
|
|
If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked
|
|
against @info. Normally, they should be equal, but in case they are not,
|
|
a g_critical will be printed and the #GstAudioInfo from the meta will be
|
|
used.
|
|
|
|
In non-interleaved buffers, it is possible to have each channel on a separate
|
|
#GstMemory. In this case, each memory will be mapped separately to avoid
|
|
copying their contents in a larger memory area. Do note though that it is
|
|
not supported to have a single channel spanning over two or more different
|
|
#GstMemory objects. Although the map operation will likely succeed in this
|
|
case, it will be highly sub-optimal and it is recommended to merge all the
|
|
memories in the buffer before calling this function.
|
|
|
|
Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid
|
|
as long as it's mapped.</doc>
|
|
<source-position filename="audio-buffer.h" line="69"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="88">%TRUE if the map operation succeeded or %FALSE on failure</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="58">pointer to a #GstAudioBuffer</doc>
|
|
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="59">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="gstbuffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="60">the #GstBuffer to be mapped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="61">the access mode for the memory</doc>
|
|
<type name="Gst.MapFlags" c:type="GstMapFlags"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="unmap"
|
|
c:identifier="gst_audio_buffer_unmap"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="41">Unmaps an audio buffer that was previously mapped with
|
|
gst_audio_buffer_map().</doc>
|
|
<source-position filename="audio-buffer.h" line="73"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-buffer.c"
|
|
line="43">the #GstAudioBuffer to unmap</doc>
|
|
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<function name="clip" c:identifier="gst_audio_buffer_clip">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="58">Clip the buffer to the given %GstSegment.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
<source-position filename="audio.h" line="96"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="72">%NULL if the buffer is completely outside the configured segment,
|
|
otherwise the clipped buffer is returned.
|
|
|
|
If the buffer has no timestamp, it is assumed to be inside the segment and
|
|
is not clipped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="60">The buffer to clip.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="61">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
|
|
the buffer should be clipped.</doc>
|
|
<type name="Gst.Segment" c:type="const GstSegment*"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="63">sample rate.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="64">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="reorder_channels"
|
|
c:identifier="gst_audio_buffer_reorder_channels">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="278">Reorders @buffer from the channel positions @from to the channel
|
|
positions @to. @from and @to must contain the same number of
|
|
positions and the same positions, only in a different order.
|
|
@buffer must be writable.</doc>
|
|
<source-position filename="audio-channels.h" line="135"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="291">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="280">The buffer to reorder.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="281">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="282">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="283">The channel positions in the buffer.</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="284">The channel positions to convert to.</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="truncate"
|
|
c:identifier="gst_audio_buffer_truncate"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="247">Truncate the buffer to finally have @samples number of samples, removing
|
|
the necessary amount of samples from the end and @trim number of samples
|
|
from the beginning.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
<source-position filename="audio.h" line="101"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="264">the truncated buffer or %NULL if the arguments
|
|
were invalid</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="249">The buffer to truncate.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="250">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="trim" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="252">the number of samples to remove from the beginning of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="253">the final number of samples that should exist in this buffer or -1
|
|
to use all the remaining samples if you are only removing samples from the
|
|
beginning.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<class name="AudioCdSrc"
|
|
c:symbol-prefix="audio_cd_src"
|
|
c:type="GstAudioCdSrc"
|
|
parent="GstBase.PushSrc"
|
|
glib:type-name="GstAudioCdSrc"
|
|
glib:get-type="gst_audio_cd_src_get_type"
|
|
glib:type-struct="AudioCdSrcClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.c"
|
|
line="37">Provides a base class for CD digital audio (CDDA) sources, which handles
|
|
things like seeking, querying, discid calculation, tags, and buffer
|
|
timestamping.
|
|
|
|
## Using GstAudioCdSrc-based elements in applications
|
|
|
|
GstAudioCdSrc registers two #GstFormat<!-- -->s of its own, namely
|
|
the "track" format and the "sector" format. Applications will usually
|
|
only find the "track" format interesting. You can retrieve that #GstFormat
|
|
for use in seek events or queries with gst_format_get_by_nick("track").
|
|
|
|
In order to query the number of tracks, for example, an application would
|
|
set the CDDA source element to READY or PAUSED state and then query the
|
|
the number of tracks via gst_element_query_duration() using the track
|
|
format acquired above. Applications can query the currently playing track
|
|
in the same way.
|
|
|
|
Alternatively, applications may retrieve the currently playing track and
|
|
the total number of tracks from the taglist that will posted on the bus
|
|
whenever the CD is opened or the currently playing track changes. The
|
|
taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags.
|
|
|
|
Applications playing back CD audio using playbin and cdda://n URIs should
|
|
issue a seek command in track format to change between tracks, rather than
|
|
setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin
|
|
involves closing and re-opening the CD device, which is much much slower).
|
|
|
|
## Tags and meta-information
|
|
|
|
CDDA sources will automatically emit a number of tags, details about which
|
|
can be found in the libgsttag documentation. Those tags are:
|
|
#GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL,
|
|
#GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL,
|
|
among others.
|
|
|
|
## Tracks and Table of Contents (TOC)
|
|
|
|
Applications will be informed of the available tracks via a TOC message
|
|
on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for
|
|
each track, with information about each track. The duration for each
|
|
track can be retrieved via the #GST_TAG_DURATION tag from each entry's
|
|
tag list, or calculated via gst_toc_entry_get_start_stop_times().
|
|
The track entries in the TOC will be sorted by track number.</doc>
|
|
<source-position filename="gstaudiocdsrc.h" line="132"/>
|
|
<implements name="Gst.URIHandler"/>
|
|
<virtual-method name="close">
|
|
<source-position filename="gstaudiocdsrc.h" line="116"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
<source-position filename="gstaudiocdsrc.h" line="115"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="device" transfer-ownership="none">
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="read_sector">
|
|
<source-position filename="gstaudiocdsrc.h" line="119"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="sector" transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="add_track" c:identifier="gst_audio_cd_src_add_track">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.c"
|
|
line="1072">CDDA sources use this function from their start vfunc to announce the
|
|
available data and audio tracks to the base source class. The caller
|
|
should allocate @track on the stack, the base source will do a shallow
|
|
copy of the structure (and take ownership of the taglist if there is one).</doc>
|
|
<source-position filename="gstaudiocdsrc.h" line="138"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.c"
|
|
line="1082">FALSE on error, otherwise TRUE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.c"
|
|
line="1074">a #GstAudioCdSrc</doc>
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="track" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.c"
|
|
line="1075">address of #GstAudioCdSrcTrack to add</doc>
|
|
<type name="AudioCdSrcTrack" c:type="GstAudioCdSrcTrack*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="device" writable="1" transfer-ownership="none">
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</property>
|
|
<property name="mode" writable="1" transfer-ownership="none">
|
|
<type name="AudioCdSrcMode"/>
|
|
</property>
|
|
<property name="track" writable="1" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</property>
|
|
<field name="pushsrc">
|
|
<type name="GstBase.PushSrc" c:type="GstPushSrc"/>
|
|
</field>
|
|
<field name="tags">
|
|
<type name="Gst.TagList" c:type="GstTagList*"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioCdSrcPrivate" c:type="GstAudioCdSrcPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved1" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="2">
|
|
<type name="guint" c:type="guint"/>
|
|
</array>
|
|
</field>
|
|
<field name="_gst_reserved2" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="2">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioCdSrcClass"
|
|
c:type="GstAudioCdSrcClass"
|
|
glib:is-gtype-struct-for="AudioCdSrc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="100">Audio CD source base class.</doc>
|
|
<source-position filename="gstaudiocdsrc.h" line="132"/>
|
|
<field name="pushsrc_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="102">the parent class</doc>
|
|
<type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
<source-position filename="gstaudiocdsrc.h" line="115"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</parameter>
|
|
<parameter name="device" transfer-ownership="none">
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close">
|
|
<callback name="close">
|
|
<source-position filename="gstaudiocdsrc.h" line="116"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="read_sector">
|
|
<callback name="read_sector">
|
|
<source-position filename="gstaudiocdsrc.h" line="119"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</parameter>
|
|
<parameter name="sector" transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="20">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="AudioCdSrcMode"
|
|
glib:type-name="GstAudioCdSrcMode"
|
|
glib:get-type="gst_audio_cd_src_mode_get_type"
|
|
c:type="GstAudioCdSrcMode">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="44">Mode in which the CD audio source operates. Influences timestamping,
|
|
EOS handling and seeking.</doc>
|
|
<member name="normal"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_CD_SRC_MODE_NORMAL"
|
|
glib:nick="normal">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="46">each single track is a stream</doc>
|
|
</member>
|
|
<member name="continuous"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_CD_SRC_MODE_CONTINUOUS"
|
|
glib:nick="continuous">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="47">the entire disc is a single stream</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioCdSrcPrivate"
|
|
c:type="GstAudioCdSrcPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudiocdsrc.h" line="42"/>
|
|
</record>
|
|
<record name="AudioCdSrcTrack" c:type="GstAudioCdSrcTrack">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="57">CD track abstraction to communicate TOC entries to the base class.
|
|
|
|
This structure is only for use by sub-classed in connection with
|
|
gst_audio_cd_src_add_track().
|
|
|
|
Applications will be informed of the available tracks via a TOC message
|
|
on the pipeline's #GstBus instead.</doc>
|
|
<source-position filename="gstaudiocdsrc.h" line="84"/>
|
|
<field name="is_audio" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="59">Whether this is an audio track</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="num" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="60">Track number in TOC (usually starts from 1, but not always)</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="start" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="61">The first sector of this track (LBA)</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="end" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="62">The last sector of this track (LBA)</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="tags" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiocdsrc.h"
|
|
line="63">Track-specific tags (e.g. from cd-text information), or NULL</doc>
|
|
<type name="Gst.TagList" c:type="GstTagList*"/>
|
|
</field>
|
|
<field name="_gst_reserved1" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="2">
|
|
<type name="guint" c:type="guint"/>
|
|
</array>
|
|
</field>
|
|
<field name="_gst_reserved2" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="2">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioChannelMixer"
|
|
c:type="GstAudioChannelMixer"
|
|
disguised="1">
|
|
<source-position filename="audio-channel-mixer.h" line="29"/>
|
|
<method name="free" c:identifier="gst_audio_channel_mixer_free">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="76">Free memory allocated by @mix.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="65"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="mix" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="78">a #GstAudioChannelMixer</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_passthrough"
|
|
c:identifier="gst_audio_channel_mixer_is_passthrough">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1009">Check if @mix is in passthrough.
|
|
|
|
Only N x N mix identity matrices are considered passthrough,
|
|
this is determined by comparing the contents of the matrix
|
|
with 0.0 and 1.0.
|
|
|
|
As this is floating point comparisons, if the values have been
|
|
generated, they should be rounded up or down by explicit
|
|
assignment of 0.0 or 1.0 to values within a user-defined
|
|
epsilon, this code doesn't make assumptions as to what may
|
|
constitute an appropriate epsilon.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="72"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1025">%TRUE is @mix is passthrough.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="mix" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1011">a #GstAudioChannelMixer</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="samples" c:identifier="gst_audio_channel_mixer_samples">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1052">In case the samples are interleaved, @in and @out must point to an
|
|
array with a single element pointing to a block of interleaved samples.
|
|
|
|
If non-interleaved samples are used, @in and @out must point to an
|
|
array with pointers to memory blocks, one for each channel.
|
|
|
|
Perform channel mixing on @in_data and write the result to @out_data.
|
|
@in_data and @out_data need to be in @format and @layout.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="79"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="mix" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1054">a #GstAudioChannelMixer</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</instance-parameter>
|
|
<parameter name="in"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1055">input samples</doc>
|
|
<type name="gpointer" c:type="const gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1056">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="1057">number of samples</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<function name="new"
|
|
c:identifier="gst_audio_channel_mixer_new"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="973">Create a new channel mixer object for the given parameters.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="50"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="983">a new #GstAudioChannelMixer object, or %NULL if @format isn't supported.
|
|
Free with gst_audio_channel_mixer_free() after usage.</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="975">#GstAudioChannelMixerFlags</doc>
|
|
<type name="AudioChannelMixerFlags"
|
|
c:type="GstAudioChannelMixerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="in_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="976">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="977">positions of input channels</doc>
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="978">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="979">positions of output channels</doc>
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="new_with_matrix"
|
|
c:identifier="gst_audio_channel_mixer_new_with_matrix"
|
|
version="1.14"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="809">Create a new channel mixer object for the given parameters.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="58"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="820">a new #GstAudioChannelMixer object, or %NULL if @format isn't supported,
|
|
@matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
|
|
Free with gst_audio_channel_mixer_free() after usage.</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="811">#GstAudioChannelMixerFlags</doc>
|
|
<type name="AudioChannelMixerFlags"
|
|
c:type="GstAudioChannelMixerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="in_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="812">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="813">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="matrix"
|
|
transfer-ownership="full"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="814">channel conversion matrix, m[@in_channels][@out_channels].
|
|
If identity matrix, passthrough applies. If %NULL, a (potentially truncated)
|
|
identity matrix is generated.</doc>
|
|
<type name="gfloat" c:type="gfloat**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<bitfield name="AudioChannelMixerFlags"
|
|
glib:type-name="GstAudioChannelMixerFlags"
|
|
glib:get-type="gst_audio_channel_mixer_flags_get_type"
|
|
c:type="GstAudioChannelMixerFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="31">Flags passed to gst_audio_channel_mixer_new()</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="33">no flag</doc>
|
|
</member>
|
|
<member name="non_interleaved_in"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN"
|
|
glib:nick="non-interleaved-in">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="34">input channels are not interleaved</doc>
|
|
</member>
|
|
<member name="non_interleaved_out"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT"
|
|
glib:nick="non-interleaved-out">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="35">output channels are not interleaved</doc>
|
|
</member>
|
|
<member name="unpositioned_in"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN"
|
|
glib:nick="unpositioned-in">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="36">input channels are explicitly unpositioned</doc>
|
|
</member>
|
|
<member name="unpositioned_out"
|
|
value="8"
|
|
c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT"
|
|
glib:nick="unpositioned-out">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.h"
|
|
line="37">output channels are explicitly unpositioned</doc>
|
|
</member>
|
|
</bitfield>
|
|
<enumeration name="AudioChannelPosition"
|
|
glib:type-name="GstAudioChannelPosition"
|
|
glib:get-type="gst_audio_channel_position_get_type"
|
|
c:type="GstAudioChannelPosition">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="31">Audio channel positions.
|
|
|
|
These are the channels defined in SMPTE 2036-2-2008
|
|
Table 1 for 22.2 audio systems with the Surround and Wide channels from
|
|
DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the
|
|
actual channel layout is expressed with a channel count and a channel mask,
|
|
which describes the existing channels. The positions in the bit mask correspond
|
|
to the enum values.
|
|
For negotiation it is allowed to have more bits set in the channel mask than
|
|
the number of channels to specify the allowed channel positions but this is
|
|
not allowed in negotiated caps. It is not allowed in any situation other
|
|
than the one mentioned below to have less bits set in the channel mask than
|
|
the number of channels.
|
|
|
|
@GST_AUDIO_CHANNEL_POSITION_MONO can only be used with a single mono channel that
|
|
has no direction information and would be mixed into all directional channels.
|
|
This is expressed in caps by having a single channel and no channel mask.
|
|
|
|
@GST_AUDIO_CHANNEL_POSITION_NONE can only be used if all channels have this position.
|
|
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
|
|
channel.</doc>
|
|
<member name="none"
|
|
value="-3"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="63">used for position-less channels, e.g.
|
|
from a sound card that records 1024 channels; mutually exclusive with
|
|
any other channel position</doc>
|
|
</member>
|
|
<member name="mono"
|
|
value="-2"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_MONO"
|
|
glib:nick="mono">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="33">Mono without direction;
|
|
can only be used with 1 channel</doc>
|
|
</member>
|
|
<member name="invalid"
|
|
value="-1"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_INVALID"
|
|
glib:nick="invalid">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="66">invalid position</doc>
|
|
</member>
|
|
<member name="front_left"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT"
|
|
glib:nick="front-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="35">Front left</doc>
|
|
</member>
|
|
<member name="front_right"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT"
|
|
glib:nick="front-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="36">Front right</doc>
|
|
</member>
|
|
<member name="front_center"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER"
|
|
glib:nick="front-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="37">Front center</doc>
|
|
</member>
|
|
<member name="lfe1"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE1"
|
|
glib:nick="lfe1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="38">Low-frequency effects 1 (subwoofer)</doc>
|
|
</member>
|
|
<member name="rear_left"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_LEFT"
|
|
glib:nick="rear-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="39">Rear left</doc>
|
|
</member>
|
|
<member name="rear_right"
|
|
value="5"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT"
|
|
glib:nick="rear-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="40">Rear right</doc>
|
|
</member>
|
|
<member name="front_left_of_center"
|
|
value="6"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER"
|
|
glib:nick="front-left-of-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="41">Front left of center</doc>
|
|
</member>
|
|
<member name="front_right_of_center"
|
|
value="7"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER"
|
|
glib:nick="front-right-of-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="42">Front right of center</doc>
|
|
</member>
|
|
<member name="rear_center"
|
|
value="8"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_CENTER"
|
|
glib:nick="rear-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="43">Rear center</doc>
|
|
</member>
|
|
<member name="lfe2"
|
|
value="9"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE2"
|
|
glib:nick="lfe2">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="44">Low-frequency effects 2 (subwoofer)</doc>
|
|
</member>
|
|
<member name="side_left"
|
|
value="10"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT"
|
|
glib:nick="side-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="45">Side left</doc>
|
|
</member>
|
|
<member name="side_right"
|
|
value="11"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT"
|
|
glib:nick="side-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="46">Side right</doc>
|
|
</member>
|
|
<member name="top_front_left"
|
|
value="12"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT"
|
|
glib:nick="top-front-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="47">Top front left</doc>
|
|
</member>
|
|
<member name="top_front_right"
|
|
value="13"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT"
|
|
glib:nick="top-front-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="48">Top front right</doc>
|
|
</member>
|
|
<member name="top_front_center"
|
|
value="14"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER"
|
|
glib:nick="top-front-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="49">Top front center</doc>
|
|
</member>
|
|
<member name="top_center"
|
|
value="15"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_CENTER"
|
|
glib:nick="top-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="50">Top center</doc>
|
|
</member>
|
|
<member name="top_rear_left"
|
|
value="16"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT"
|
|
glib:nick="top-rear-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="51">Top rear left</doc>
|
|
</member>
|
|
<member name="top_rear_right"
|
|
value="17"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT"
|
|
glib:nick="top-rear-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="52">Top rear right</doc>
|
|
</member>
|
|
<member name="top_side_left"
|
|
value="18"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT"
|
|
glib:nick="top-side-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="53">Top side right</doc>
|
|
</member>
|
|
<member name="top_side_right"
|
|
value="19"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT"
|
|
glib:nick="top-side-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="54">Top rear right</doc>
|
|
</member>
|
|
<member name="top_rear_center"
|
|
value="20"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER"
|
|
glib:nick="top-rear-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="55">Top rear center</doc>
|
|
</member>
|
|
<member name="bottom_front_center"
|
|
value="21"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER"
|
|
glib:nick="bottom-front-center">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="56">Bottom front center</doc>
|
|
</member>
|
|
<member name="bottom_front_left"
|
|
value="22"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT"
|
|
glib:nick="bottom-front-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="57">Bottom front left</doc>
|
|
</member>
|
|
<member name="bottom_front_right"
|
|
value="23"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT"
|
|
glib:nick="bottom-front-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="58">Bottom front right</doc>
|
|
</member>
|
|
<member name="wide_left"
|
|
value="24"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT"
|
|
glib:nick="wide-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="59">Wide left (between front left and side left)</doc>
|
|
</member>
|
|
<member name="wide_right"
|
|
value="25"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT"
|
|
glib:nick="wide-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="60">Wide right (between front right and side right)</doc>
|
|
</member>
|
|
<member name="surround_left"
|
|
value="26"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT"
|
|
glib:nick="surround-left">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="61">Surround left (between rear left and side left)</doc>
|
|
</member>
|
|
<member name="surround_right"
|
|
value="27"
|
|
c:identifier="GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT"
|
|
glib:nick="surround-right">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.h"
|
|
line="62">Surround right (between rear right and side right)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioClippingMeta"
|
|
c:type="GstAudioClippingMeta"
|
|
version="1.8">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="85">Extra buffer metadata describing how much audio has to be clipped from
|
|
the start or end of a buffer. This is used for compressed formats, where
|
|
the first frame usually has some additional samples due to encoder and
|
|
decoder delays, and the last frame usually has some additional samples to
|
|
be able to fill the complete last frame.
|
|
|
|
This is used to ensure that decoded data in the end has the same amount of
|
|
samples, and multiply decoded streams can be gaplessly concatenated.
|
|
|
|
Note: If clipping of the start is done by adjusting the segment, this meta
|
|
has to be dropped from buffers as otherwise clipping could happen twice.</doc>
|
|
<source-position filename="gstaudiometa.h" line="112"/>
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="87">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="format" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="88">GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</field>
|
|
<field name="start" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="89">Amount of audio to clip from start of buffer</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="end" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="90">Amount of to clip from end of buffer</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<function name="get_info"
|
|
c:identifier="gst_audio_clipping_meta_get_info">
|
|
<source-position filename="gstaudiometa.h" line="118"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
</record>
|
|
<class name="AudioClock"
|
|
c:symbol-prefix="audio_clock"
|
|
c:type="GstAudioClock"
|
|
parent="Gst.SystemClock"
|
|
glib:type-name="GstAudioClock"
|
|
glib:get-type="gst_audio_clock_get_type"
|
|
glib:type-struct="AudioClockClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="23">#GstAudioClock makes it easy for elements to implement a #GstClock, they
|
|
simply need to provide a function that returns the current clock time.
|
|
|
|
This object is internally used to implement the clock in #GstAudioBaseSink.</doc>
|
|
<source-position filename="gstaudioclock.h" line="90"/>
|
|
<constructor name="new" c:identifier="gst_audio_clock_new">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="89">Create a new #GstAudioClock instance. Whenever the clock time should be
|
|
calculated it will call @func with @user_data. When @func returns
|
|
#GST_CLOCK_TIME_NONE, the clock will return the last reported time.</doc>
|
|
<source-position filename="gstaudioclock.h" line="96"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="100">a new #GstAudioClock casted to a #GstClock.</doc>
|
|
<type name="Gst.Clock" c:type="GstClock*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="name" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="91">the name of the clock</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
<parameter name="func"
|
|
transfer-ownership="none"
|
|
scope="notified"
|
|
closure="2"
|
|
destroy="3">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="92">a function</doc>
|
|
<type name="AudioClockGetTimeFunc"
|
|
c:type="GstAudioClockGetTimeFunc"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="93">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="destroy_notify"
|
|
transfer-ownership="none"
|
|
scope="async">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="94">#GDestroyNotify for @user_data</doc>
|
|
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="adjust" c:identifier="gst_audio_clock_adjust">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="207">Adjust @time with the internal offset of the audio clock.</doc>
|
|
<source-position filename="gstaudioclock.h" line="106"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="214">@time adjusted with the internal offset.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="clock" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="209">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
<parameter name="time" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="210">a #GstClockTime</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_time" c:identifier="gst_audio_clock_get_time">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="180">Report the time as returned by the #GstAudioClockGetTimeFunc without applying
|
|
any offsets.</doc>
|
|
<source-position filename="gstaudioclock.h" line="103"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="187">the time as reported by the time function of the audio clock</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="clock" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="182">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="invalidate" c:identifier="gst_audio_clock_invalidate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="226">Invalidate the clock function. Call this function when the provided
|
|
#GstAudioClockGetTimeFunc cannot be called anymore, for example, when the
|
|
user_data becomes invalid.
|
|
|
|
After calling this function, @clock will return the last returned time for
|
|
the rest of its lifetime.</doc>
|
|
<source-position filename="gstaudioclock.h" line="109"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="clock" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="228">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="reset" c:identifier="gst_audio_clock_reset">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="119">Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values
|
|
starting from @time. The clock will update an internal offset to make sure that
|
|
future calls to internal_time will return an increasing result as required by
|
|
the #GstClock object.</doc>
|
|
<source-position filename="gstaudioclock.h" line="100"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="clock" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="121">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
<parameter name="time" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.c"
|
|
line="122">a #GstClockTime</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="clock">
|
|
<type name="Gst.SystemClock" c:type="GstSystemClock"/>
|
|
</field>
|
|
<field name="func">
|
|
<type name="AudioClockGetTimeFunc" c:type="GstAudioClockGetTimeFunc"/>
|
|
</field>
|
|
<field name="user_data">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</field>
|
|
<field name="destroy_notify">
|
|
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
|
|
</field>
|
|
<field name="last_time" readable="0" private="1">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</field>
|
|
<field name="time_offset" readable="0" private="1">
|
|
<type name="Gst.ClockTimeDiff" c:type="GstClockTimeDiff"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioClockClass"
|
|
c:type="GstAudioClockClass"
|
|
glib:is-gtype-struct-for="AudioClock">
|
|
<source-position filename="gstaudioclock.h" line="90"/>
|
|
<field name="parent_class">
|
|
<type name="Gst.SystemClockClass" c:type="GstSystemClockClass"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<callback name="AudioClockGetTimeFunc" c:type="GstAudioClockGetTimeFunc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.h"
|
|
line="51">This function will be called whenever the current clock time needs to be
|
|
calculated. If this function returns #GST_CLOCK_TIME_NONE, the last reported
|
|
time will be returned by the clock.</doc>
|
|
<source-position filename="gstaudioclock.h" line="63"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.h"
|
|
line="60">the current time or #GST_CLOCK_TIME_NONE if the previous time should
|
|
be used.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="clock" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.h"
|
|
line="53">the #GstAudioClock</doc>
|
|
<type name="Gst.Clock" c:type="GstClock*"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1"
|
|
closure="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioclock.h"
|
|
line="54">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<record name="AudioConverter"
|
|
c:type="GstAudioConverter"
|
|
glib:type-name="GstAudioConverter"
|
|
glib:get-type="gst_audio_converter_get_type"
|
|
c:symbol-prefix="audio_converter">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="33">This object is used to convert audio samples from one format to another.
|
|
The object can perform conversion of:
|
|
|
|
* audio format with optional dithering and noise shaping
|
|
|
|
* audio samplerate
|
|
|
|
* audio channels and channel layout</doc>
|
|
<source-position filename="audio-converter.h" line="31"/>
|
|
<constructor name="new" c:identifier="gst_audio_converter_new">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1306">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_*
|
|
parameters for details about the options and values.</doc>
|
|
<source-position filename="audio-converter.h" line="119"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1319">a #GstAudioConverter or %NULL if conversion is not possible.</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1308">extra #GstAudioConverterFlags</doc>
|
|
<type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
|
|
</parameter>
|
|
<parameter name="in_info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1309">a source #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="out_info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1310">a destination #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="config"
|
|
transfer-ownership="full"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1311">a #GstStructure with configuration options</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="convert"
|
|
c:identifier="gst_audio_converter_convert"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1599">Convenience wrapper around gst_audio_converter_samples(), which will
|
|
perform allocation of the output buffer based on the result from
|
|
gst_audio_converter_get_out_frames().</doc>
|
|
<source-position filename="audio-converter.h" line="166"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1613">%TRUE is the conversion could be performed.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1601">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1602">extra #GstAudioConverterFlags</doc>
|
|
<type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
|
|
</parameter>
|
|
<parameter name="in" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1603">input data</doc>
|
|
<array length="2" zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="in_size" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1604">size of @in</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="out"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1605">a pointer where
|
|
the output data will be written</doc>
|
|
<array length="4" zero-terminated="0" c:type="gpointer*">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="out_size"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1607">a pointer where the size of @out will be written</doc>
|
|
<type name="gsize" c:type="gsize*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free" c:identifier="gst_audio_converter_free">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1447">Free a previously allocated @convert instance.</doc>
|
|
<source-position filename="audio-converter.h" line="128"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1449">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_config" c:identifier="gst_audio_converter_get_config">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="375">Get the current configuration of @convert.</doc>
|
|
<source-position filename="audio-converter.h" line="139"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="383">
|
|
a #GstStructure that remains valid for as long as @convert is valid
|
|
or until gst_audio_converter_update_config() is called.</doc>
|
|
<type name="Gst.Structure" c:type="const GstStructure*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="377">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_rate"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="378">result input rate</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
<parameter name="out_rate"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="379">result output rate</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_in_frames"
|
|
c:identifier="gst_audio_converter_get_in_frames">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1501">Calculate how many input frames are currently needed by @convert to produce
|
|
@out_frames of output frames.</doc>
|
|
<source-position filename="audio-converter.h" line="147"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1509">the number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1503">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1504">number of output frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_max_latency"
|
|
c:identifier="gst_audio_converter_get_max_latency">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1521">Get the maximum number of input frames that the converter would
|
|
need before producing output.</doc>
|
|
<source-position filename="audio-converter.h" line="151"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1528">the latency of @convert as expressed in the number of
|
|
frames.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1523">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_out_frames"
|
|
c:identifier="gst_audio_converter_get_out_frames">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1481">Calculate how many output frames can be produced when @in_frames input
|
|
frames are given to @convert.</doc>
|
|
<source-position filename="audio-converter.h" line="143"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1489">the number of output frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1483">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1484">number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_passthrough"
|
|
c:identifier="gst_audio_converter_is_passthrough"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1653">Returns whether the audio converter will operate in passthrough mode.
|
|
The return value would be typically input to gst_base_transform_set_passthrough()</doc>
|
|
<source-position filename="audio-converter.h" line="163"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1659">%TRUE when no conversion will actually occur.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="reset" c:identifier="gst_audio_converter_reset">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1540">Reset @convert to the state it was when it was first created, clearing
|
|
any history it might currently have.</doc>
|
|
<source-position filename="audio-converter.h" line="131"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1542">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="samples" c:identifier="gst_audio_converter_samples">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1556">Perform the conversion with @in_frames in @in to @out_frames in @out
|
|
using @convert.
|
|
|
|
In case the samples are interleaved, @in and @out must point to an
|
|
array with a single element pointing to a block of interleaved samples.
|
|
|
|
If non-interleaved samples are used, @in and @out must point to an
|
|
array with pointers to memory blocks, one for each channel.
|
|
|
|
@in may be %NULL, in which case @in_frames of silence samples are processed
|
|
by the converter.
|
|
|
|
This function always produces @out_frames of output and consumes @in_frames of
|
|
input. Use gst_audio_converter_get_out_frames() and
|
|
gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames
|
|
are matching and @in and @out point to enough memory.</doc>
|
|
<source-position filename="audio-converter.h" line="154"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1582">%TRUE is the conversion could be performed.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1558">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1559">extra #GstAudioConverterFlags</doc>
|
|
<type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
|
|
</parameter>
|
|
<parameter name="in"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1560">input frames</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1561">number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="out"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1562">output frames</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1563">number of output frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="supports_inplace"
|
|
c:identifier="gst_audio_converter_supports_inplace">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1638">Returns whether the audio converter can perform the conversion in-place.
|
|
The return value would be typically input to gst_base_transform_set_in_place()</doc>
|
|
<source-position filename="audio-converter.h" line="160"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1645">%TRUE when the conversion can be done in place.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="1640">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="update_config"
|
|
c:identifier="gst_audio_converter_update_config">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="322">Set @in_rate, @out_rate and @config as extra configuration for @convert.
|
|
|
|
@in_rate and @out_rate specify the new sample rates of input and output
|
|
formats. A value of 0 leaves the sample rate unchanged.
|
|
|
|
@config can be %NULL, in which case, the current configuration is not
|
|
changed.
|
|
|
|
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
|
|
option and values.</doc>
|
|
<source-position filename="audio-converter.h" line="134"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="344">%TRUE when the new parameters could be set</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="324">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="325">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="326">output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="config"
|
|
transfer-ownership="full"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.c"
|
|
line="327">a #GstStructure or %NULL</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<bitfield name="AudioConverterFlags"
|
|
glib:type-name="GstAudioConverterFlags"
|
|
glib:get-type="gst_audio_converter_flags_get_type"
|
|
c:type="GstAudioConverterFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
|
|
line="102">Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_CONVERTER_FLAG_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
|
|
line="104">no flag</doc>
|
|
</member>
|
|
<member name="in_writable"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE"
|
|
glib:nick="in-writable">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
|
|
line="105">the input sample arrays are writable and can be
|
|
used as temporary storage during conversion.</doc>
|
|
</member>
|
|
<member name="variable_rate"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE"
|
|
glib:nick="variable-rate">
|
|
<doc xml:space="preserve"
|
|
filename="audio-converter.h"
|
|
line="107">allow arbitrary rate updates with
|
|
gst_audio_converter_update_config().</doc>
|
|
</member>
|
|
</bitfield>
|
|
<class name="AudioDecoder"
|
|
c:symbol-prefix="audio_decoder"
|
|
c:type="GstAudioDecoder"
|
|
parent="Gst.Element"
|
|
abstract="1"
|
|
glib:type-name="GstAudioDecoder"
|
|
glib:get-type="gst_audio_decoder_get_type"
|
|
glib:type-struct="AudioDecoderClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="24">This base class is for audio decoders turning encoded data into
|
|
raw audio samples.
|
|
|
|
GstAudioDecoder and subclass should cooperate as follows.
|
|
|
|
## Configuration
|
|
|
|
* Initially, GstAudioDecoder calls @start when the decoder element
|
|
is activated, which allows subclass to perform any global setup.
|
|
Base class (context) parameters can already be set according to subclass
|
|
capabilities (or possibly upon receive more information in subsequent
|
|
@set_format).
|
|
* GstAudioDecoder calls @set_format to inform subclass of the format
|
|
of input audio data that it is about to receive.
|
|
While unlikely, it might be called more than once, if changing input
|
|
parameters require reconfiguration.
|
|
* GstAudioDecoder calls @stop at end of all processing.
|
|
|
|
As of configuration stage, and throughout processing, GstAudioDecoder
|
|
provides various (context) parameters, e.g. describing the format of
|
|
output audio data (valid when output caps have been set) or current parsing state.
|
|
Conversely, subclass can and should configure context to inform
|
|
base class of its expectation w.r.t. buffer handling.
|
|
|
|
## Data processing
|
|
* Base class gathers input data, and optionally allows subclass
|
|
to parse this into subsequently manageable (as defined by subclass)
|
|
chunks. Such chunks are subsequently referred to as 'frames',
|
|
though they may or may not correspond to 1 (or more) audio format frame.
|
|
* Input frame is provided to subclass' @handle_frame.
|
|
* If codec processing results in decoded data, subclass should call
|
|
@gst_audio_decoder_finish_frame to have decoded data pushed
|
|
downstream.
|
|
* Just prior to actually pushing a buffer downstream,
|
|
it is passed to @pre_push. Subclass should either use this callback
|
|
to arrange for additional downstream pushing or otherwise ensure such
|
|
custom pushing occurs after at least a method call has finished since
|
|
setting src pad caps.
|
|
* During the parsing process GstAudioDecoderClass will handle both
|
|
srcpad and sinkpad events. Sink events will be passed to subclass
|
|
if @event callback has been provided.
|
|
|
|
## Shutdown phase
|
|
|
|
* GstAudioDecoder class calls @stop to inform the subclass that data
|
|
parsing will be stopped.
|
|
|
|
Subclass is responsible for providing pad template caps for
|
|
source and sink pads. The pads need to be named "sink" and "src". It also
|
|
needs to set the fixed caps on srcpad, when the format is ensured. This
|
|
is typically when base class calls subclass' @set_format function, though
|
|
it might be delayed until calling @gst_audio_decoder_finish_frame.
|
|
|
|
In summary, above process should have subclass concentrating on
|
|
codec data processing while leaving other matters to base class,
|
|
such as most notably timestamp handling. While it may exert more control
|
|
in this area (see e.g. @pre_push), it is very much not recommended.
|
|
|
|
In particular, base class will try to arrange for perfect output timestamps
|
|
as much as possible while tracking upstream timestamps.
|
|
To this end, if deviation between the next ideal expected perfect timestamp
|
|
and upstream exceeds #GstAudioDecoder:tolerance, then resync to upstream
|
|
occurs (which would happen always if the tolerance mechanism is disabled).
|
|
|
|
In non-live pipelines, baseclass can also (configurably) arrange for
|
|
output buffer aggregation which may help to redue large(r) numbers of
|
|
small(er) buffers being pushed and processed downstream. Note that this
|
|
feature is only available if the buffer layout is interleaved. For planar
|
|
buffers, the decoder implementation is fully responsible for the output
|
|
buffer size.
|
|
|
|
On the other hand, it should be noted that baseclass only provides limited
|
|
seeking support (upon explicit subclass request), as full-fledged support
|
|
should rather be left to upstream demuxer, parser or alike. This simple
|
|
approach caters for seeking and duration reporting using estimated input
|
|
bitrates.
|
|
|
|
Things that subclass need to take care of:
|
|
|
|
* Provide pad templates
|
|
* Set source pad caps when appropriate
|
|
* Set user-configurable properties to sane defaults for format and
|
|
implementing codec at hand, and convey some subclass capabilities and
|
|
expectations in context.
|
|
|
|
* Accept data in @handle_frame and provide encoded results to
|
|
@gst_audio_decoder_finish_frame. If it is prepared to perform
|
|
PLC, it should also accept NULL data in @handle_frame and provide for
|
|
data for indicated duration.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="315"/>
|
|
<virtual-method name="close">
|
|
<source-position filename="gstaudiodecoder.h" line="294"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="decide_allocation">
|
|
<source-position filename="gstaudiodecoder.h" line="298"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="flush">
|
|
<source-position filename="gstaudiodecoder.h" line="282"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="hard" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="getcaps">
|
|
<source-position filename="gstaudiodecoder.h" line="307"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="filter" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="handle_frame">
|
|
<source-position filename="gstaudiodecoder.h" line="279"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="negotiate" invoker="negotiate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="732">Negotiate with downstream elements to currently configured #GstAudioInfo.
|
|
Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
|
|
negotiate fails.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="296"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="740">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="734">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
<source-position filename="gstaudiodecoder.h" line="292"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="parse">
|
|
<source-position filename="gstaudiodecoder.h" line="275"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="adapter" transfer-ownership="none">
|
|
<type name="GstBase.Adapter" c:type="GstAdapter*"/>
|
|
</parameter>
|
|
<parameter name="offset" transfer-ownership="none">
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="pre_push">
|
|
<source-position filename="gstaudiodecoder.h" line="284"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="propose_allocation">
|
|
<source-position filename="gstaudiodecoder.h" line="300"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="set_format">
|
|
<source-position filename="gstaudiodecoder.h" line="272"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="sink_event">
|
|
<source-position filename="gstaudiodecoder.h" line="287"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="sink_query">
|
|
<source-position filename="gstaudiodecoder.h" line="303"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="src_event">
|
|
<source-position filename="gstaudiodecoder.h" line="289"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="src_query">
|
|
<source-position filename="gstaudiodecoder.h" line="305"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="start">
|
|
<source-position filename="gstaudiodecoder.h" line="268"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="stop">
|
|
<source-position filename="gstaudiodecoder.h" line="270"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="transform_meta">
|
|
<source-position filename="gstaudiodecoder.h" line="310"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="meta" transfer-ownership="none">
|
|
<type name="Gst.Meta" c:type="GstMeta*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="allocate_output_buffer"
|
|
c:identifier="gst_audio_decoder_allocate_output_buffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3669">Helper function that allocates a buffer to hold an audio frame
|
|
for @dec's current output format.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="344"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3677">allocated buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3671">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="size" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3672">size of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="finish_frame"
|
|
c:identifier="gst_audio_decoder_finish_frame">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1268">Collects decoded data and pushes it downstream.
|
|
|
|
@buf may be NULL in which case the indicated number of frames
|
|
are discarded and considered to have produced no output
|
|
(e.g. lead-in or setup frames).
|
|
Otherwise, source pad caps must be set when it is called with valid
|
|
data in @buf.
|
|
|
|
Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
|
|
invalidated by a call to this function.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="340"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1285">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1270">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1271">decoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1272">number of decoded frames represented by decoded data</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="finish_subframe"
|
|
c:identifier="gst_audio_decoder_finish_subframe"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1234">Collects decoded data and pushes it downstream. This function may be called
|
|
multiple times for a given input frame.
|
|
|
|
@buf may be NULL in which case it is assumed that the current input frame is
|
|
finished. This is equivalent to calling gst_audio_decoder_finish_subframe()
|
|
with a NULL buffer and frames=1 after having pushed out all decoded audio
|
|
subframes using this function.
|
|
|
|
When called with valid data in @buf the source pad caps must have been set
|
|
already.
|
|
|
|
Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
|
|
invalidated by a call to this function.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="336"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1253">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1236">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="1237">decoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_allocator"
|
|
c:identifier="gst_audio_decoder_get_allocator">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3720">Lets #GstAudioDecoder sub-classes to know the memory @allocator
|
|
used by the base class and its @params.
|
|
|
|
Unref the @allocator after use it.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="433"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3722">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="allocator"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3723">the #GstAllocator
|
|
used</doc>
|
|
<type name="Gst.Allocator" c:type="GstAllocator**"/>
|
|
</parameter>
|
|
<parameter name="params"
|
|
direction="out"
|
|
caller-allocates="1"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3725">the
|
|
#GstAllocationParams of @allocator</doc>
|
|
<type name="Gst.AllocationParams" c:type="GstAllocationParams*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_audio_info"
|
|
c:identifier="gst_audio_decoder_get_audio_info">
|
|
<source-position filename="gstaudiodecoder.h" line="350"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3200">a #GstAudioInfo describing the input audio format</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3198">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_delay" c:identifier="gst_audio_decoder_get_delay">
|
|
<source-position filename="gstaudiodecoder.h" line="367"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3272">currently configured decoder delay</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3270">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_drainable"
|
|
c:identifier="gst_audio_decoder_get_drainable">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3558">Queries decoder drain handling.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="423"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3564">TRUE if drainable handling is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3560">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_estimate_rate"
|
|
c:identifier="gst_audio_decoder_get_estimate_rate">
|
|
<source-position filename="gstaudiodecoder.h" line="364"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3258">currently configured byte to time conversion setting</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3256">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_latency" c:identifier="gst_audio_decoder_get_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3340">Sets the variables pointed to by @min and @max to the currently configured
|
|
latency.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="382"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3342">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3343">a pointer to storage to hold minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
<parameter name="max"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3344">a pointer to storage to hold maximum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_max_errors"
|
|
c:identifier="gst_audio_decoder_get_max_errors">
|
|
<source-position filename="gstaudiodecoder.h" line="374"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3304">currently configured decoder tolerated error count.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3302">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_min_latency"
|
|
c:identifier="gst_audio_decoder_get_min_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3469">Queries decoder's latency aggregation.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="409"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3475">aggregation latency.
|
|
|
|
MT safe.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3471">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_needs_format"
|
|
c:identifier="gst_audio_decoder_get_needs_format">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3606">Queries decoder required format handling.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="430"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3612">TRUE if required format handling is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3608">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_parse_state"
|
|
c:identifier="gst_audio_decoder_get_parse_state">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3363">Return current parsing (sync and eos) state.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="387"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3365">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="sync"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3366">a pointer to a variable to hold the current sync state</doc>
|
|
<type name="gboolean" c:type="gboolean*"/>
|
|
</parameter>
|
|
<parameter name="eos"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3367">a pointer to a variable to hold the current eos state</doc>
|
|
<type name="gboolean" c:type="gboolean*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_plc" c:identifier="gst_audio_decoder_get_plc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3426">Queries decoder packet loss concealment handling.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="402"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3432">TRUE if packet loss concealment is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3428">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_plc_aware"
|
|
c:identifier="gst_audio_decoder_get_plc_aware">
|
|
<source-position filename="gstaudiodecoder.h" line="357"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3229">currently configured plc handling</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3227">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_tolerance"
|
|
c:identifier="gst_audio_decoder_get_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3512">Queries current audio jitter tolerance threshold.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="416"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3518">decoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3514">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="merge_tags" c:identifier="gst_audio_decoder_merge_tags">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3630">Sets the audio decoder tags and how they should be merged with any
|
|
upstream stream tags. This will override any tags previously-set
|
|
with gst_audio_decoder_merge_tags().
|
|
|
|
Note that this is provided for convenience, and the subclass is
|
|
not required to use this and can still do tag handling on its own.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="438"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3632">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tags"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3633">a #GstTagList to merge, or NULL</doc>
|
|
<type name="Gst.TagList" c:type="const GstTagList*"/>
|
|
</parameter>
|
|
<parameter name="mode" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3634">the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE</doc>
|
|
<type name="Gst.TagMergeMode" c:type="GstTagMergeMode"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="negotiate" c:identifier="gst_audio_decoder_negotiate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="732">Negotiate with downstream elements to currently configured #GstAudioInfo.
|
|
Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
|
|
negotiate fails.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="333"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="740">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="734">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="proxy_getcaps"
|
|
c:identifier="gst_audio_decoder_proxy_getcaps"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="2694">Returns caps that express @caps (or sink template caps if @caps == NULL)
|
|
restricted to rate/channels/... combinations supported by downstream
|
|
elements.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="328"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="2704">a #GstCaps owned by caller</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="decoder" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="2696">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="2697">initial caps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
<parameter name="filter"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="2698">filter caps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_allocation_caps"
|
|
c:identifier="gst_audio_decoder_set_allocation_caps"
|
|
version="1.10">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3383">Sets a caps in allocation query which are different from the set
|
|
pad's caps. Use this function before calling
|
|
gst_audio_decoder_negotiate(). Setting to %NULL the allocation
|
|
query will use the caps from the pad.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="392"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3385">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="allocation_caps"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3386">a #GstCaps or %NULL</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_drainable"
|
|
c:identifier="gst_audio_decoder_set_drainable">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3536">Configures decoder drain handling. If drainable, subclass might
|
|
be handed a NULL buffer to have it return any leftover decoded data.
|
|
Otherwise, it is not considered so capable and will only ever be passed
|
|
real data.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="419"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3538">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3539">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_estimate_rate"
|
|
c:identifier="gst_audio_decoder_set_estimate_rate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3239">Allows baseclass to perform byte to time estimated conversion.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="360"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3241">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3242">whether to enable byte to time conversion</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_latency" c:identifier="gst_audio_decoder_set_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3314">Sets decoder latency.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="377"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3316">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3317">minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="max" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3318">maximum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_max_errors"
|
|
c:identifier="gst_audio_decoder_set_max_errors">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3282">Sets numbers of tolerated decoder errors, where a tolerated one is then only
|
|
warned about, but more than tolerated will lead to fatal error. You can set
|
|
-1 for never returning fatal errors. Default is set to
|
|
GST_AUDIO_DECODER_MAX_ERRORS.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="370"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3284">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3285">max tolerated errors</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_min_latency"
|
|
c:identifier="gst_audio_decoder_set_min_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3450">Sets decoder minimum aggregation latency.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="405"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3452">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3453">new minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_needs_format"
|
|
c:identifier="gst_audio_decoder_set_needs_format">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3582">Configures decoder format needs. If enabled, subclass needs to be
|
|
negotiated with format caps before it can process any data. It will then
|
|
never be handed any data before it has been configured.
|
|
Otherwise, it might be handed data without having been configured and
|
|
is then expected being able to do so either by default
|
|
or based on the input data.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="426"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3584">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3585">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_output_caps"
|
|
c:identifier="gst_audio_decoder_set_output_caps"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="797">Configure output caps on the srcpad of @dec. Similar to
|
|
gst_audio_decoder_set_output_format(), but allows subclasses to specify
|
|
output caps that can't be expressed via #GstAudioInfo e.g. caps that have
|
|
caps features.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="325"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="807">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="799">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="800">(fixed) #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_output_format"
|
|
c:identifier="gst_audio_decoder_set_output_format">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="764">Configure output info on the srcpad of @dec.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="321"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="771">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="766">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="767">#GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_plc" c:identifier="gst_audio_decoder_set_plc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3404">Enable or disable decoder packet loss concealment, provided subclass
|
|
and codec are capable and allow handling plc.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="398"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3406">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3407">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_plc_aware"
|
|
c:identifier="gst_audio_decoder_set_plc_aware">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3210">Indicates whether or not subclass handles packet loss concealment (plc).</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="353"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3212">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="plc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3213">new plc state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_tolerance"
|
|
c:identifier="gst_audio_decoder_set_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3493">Configures decoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="412"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3495">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3496">new tolerance</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_use_default_pad_acceptcaps"
|
|
c:identifier="gst_audio_decoder_set_use_default_pad_acceptcaps"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3747">Lets #GstAudioDecoder sub-classes decide if they want the sink pad
|
|
to use the default pad query handler to reply to accept-caps queries.
|
|
|
|
By setting this to true it is possible to further customize the default
|
|
handler with %GST_PAD_SET_ACCEPT_INTERSECT and
|
|
%GST_PAD_SET_ACCEPT_TEMPLATE</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="442"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="decoder" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3749">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="use" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="3750">if the default pad accept-caps query handling should be used</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="min-latency" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<property name="plc" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="tolerance" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<field name="element">
|
|
<type name="Gst.Element" c:type="GstElement"/>
|
|
</field>
|
|
<field name="sinkpad">
|
|
<type name="Gst.Pad" c:type="GstPad*"/>
|
|
</field>
|
|
<field name="srcpad">
|
|
<type name="Gst.Pad" c:type="GstPad*"/>
|
|
</field>
|
|
<field name="stream_lock">
|
|
<type name="GLib.RecMutex" c:type="GRecMutex"/>
|
|
</field>
|
|
<field name="input_segment">
|
|
<type name="Gst.Segment" c:type="GstSegment"/>
|
|
</field>
|
|
<field name="output_segment">
|
|
<type name="Gst.Segment" c:type="GstSegment"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioDecoderPrivate" c:type="GstAudioDecoderPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="20">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioDecoderClass"
|
|
c:type="GstAudioDecoderClass"
|
|
glib:is-gtype-struct-for="AudioDecoder">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="181">Subclasses can override any of the available virtual methods or not, as
|
|
needed. At minimum @handle_frame (and likely @set_format) needs to be
|
|
overridden.</doc>
|
|
<source-position filename="gstaudiodecoder.h" line="315"/>
|
|
<field name="element_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.h"
|
|
line="183">The parent class structure</doc>
|
|
<type name="Gst.ElementClass" c:type="GstElementClass"/>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
<source-position filename="gstaudiodecoder.h" line="268"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="stop">
|
|
<callback name="stop">
|
|
<source-position filename="gstaudiodecoder.h" line="270"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="set_format">
|
|
<callback name="set_format">
|
|
<source-position filename="gstaudiodecoder.h" line="272"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="parse">
|
|
<callback name="parse">
|
|
<source-position filename="gstaudiodecoder.h" line="275"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="adapter" transfer-ownership="none">
|
|
<type name="GstBase.Adapter" c:type="GstAdapter*"/>
|
|
</parameter>
|
|
<parameter name="offset" transfer-ownership="none">
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="handle_frame">
|
|
<callback name="handle_frame">
|
|
<source-position filename="gstaudiodecoder.h" line="279"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="flush">
|
|
<callback name="flush">
|
|
<source-position filename="gstaudiodecoder.h" line="282"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="hard" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="pre_push">
|
|
<callback name="pre_push">
|
|
<source-position filename="gstaudiodecoder.h" line="284"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="sink_event">
|
|
<callback name="sink_event">
|
|
<source-position filename="gstaudiodecoder.h" line="287"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="src_event">
|
|
<callback name="src_event">
|
|
<source-position filename="gstaudiodecoder.h" line="289"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
<source-position filename="gstaudiodecoder.h" line="292"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close">
|
|
<callback name="close">
|
|
<source-position filename="gstaudiodecoder.h" line="294"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="negotiate">
|
|
<callback name="negotiate">
|
|
<source-position filename="gstaudiodecoder.h" line="296"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="740">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiodecoder.c"
|
|
line="734">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="decide_allocation">
|
|
<callback name="decide_allocation">
|
|
<source-position filename="gstaudiodecoder.h" line="298"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="propose_allocation">
|
|
<callback name="propose_allocation">
|
|
<source-position filename="gstaudiodecoder.h" line="300"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="sink_query">
|
|
<callback name="sink_query">
|
|
<source-position filename="gstaudiodecoder.h" line="303"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="src_query">
|
|
<callback name="src_query">
|
|
<source-position filename="gstaudiodecoder.h" line="305"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="getcaps">
|
|
<callback name="getcaps">
|
|
<source-position filename="gstaudiodecoder.h" line="307"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="filter" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="transform_meta">
|
|
<callback name="transform_meta">
|
|
<source-position filename="gstaudiodecoder.h" line="310"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="meta" transfer-ownership="none">
|
|
<type name="Gst.Meta" c:type="GstMeta*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="16">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioDecoderPrivate"
|
|
c:type="GstAudioDecoderPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudiodecoder.h" line="102"/>
|
|
</record>
|
|
<enumeration name="AudioDitherMethod"
|
|
glib:type-name="GstAudioDitherMethod"
|
|
glib:get-type="gst_audio_dither_method_get_type"
|
|
c:type="GstAudioDitherMethod">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="32">Set of available dithering methods.</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_DITHER_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="34">No dithering</doc>
|
|
</member>
|
|
<member name="rpdf"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_DITHER_RPDF"
|
|
glib:nick="rpdf">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="35">Rectangular dithering</doc>
|
|
</member>
|
|
<member name="tpdf"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_DITHER_TPDF"
|
|
glib:nick="tpdf">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="36">Triangular dithering (default)</doc>
|
|
</member>
|
|
<member name="tpdf_hf"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_DITHER_TPDF_HF"
|
|
glib:nick="tpdf-hf">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="37">High frequency triangular dithering</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioDownmixMeta" c:type="GstAudioDownmixMeta">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="32">Extra buffer metadata describing audio downmixing matrix. This metadata is
|
|
attached to audio buffers and contains a matrix to downmix the buffer number
|
|
of channels to @channels.
|
|
|
|
@matrix is an two-dimensional array of @to_channels times @from_channels
|
|
coefficients, i.e. the i-th output channels is constructed by multiplicating
|
|
the input channels with the coefficients in @matrix[i] and taking the sum
|
|
of the results.</doc>
|
|
<source-position filename="gstaudiometa.h" line="57"/>
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="34">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="from_position" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="35">the channel positions of the source</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</field>
|
|
<field name="to_position" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="36">the channel positions of the destination</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</field>
|
|
<field name="from_channels" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="37">the number of channels of the source</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="to_channels" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="38">the number of channels of the destination</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="matrix" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="39">the matrix coefficients.</doc>
|
|
<type name="gfloat" c:type="gfloat**"/>
|
|
</field>
|
|
<function name="get_info" c:identifier="gst_audio_downmix_meta_get_info">
|
|
<source-position filename="gstaudiometa.h" line="63"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
</record>
|
|
<class name="AudioEncoder"
|
|
c:symbol-prefix="audio_encoder"
|
|
c:type="GstAudioEncoder"
|
|
parent="Gst.Element"
|
|
abstract="1"
|
|
glib:type-name="GstAudioEncoder"
|
|
glib:get-type="gst_audio_encoder_get_type"
|
|
glib:type-struct="AudioEncoderClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="22">This base class is for audio encoders turning raw audio samples into
|
|
encoded audio data.
|
|
|
|
GstAudioEncoder and subclass should cooperate as follows.
|
|
|
|
## Configuration
|
|
|
|
* Initially, GstAudioEncoder calls @start when the encoder element
|
|
is activated, which allows subclass to perform any global setup.
|
|
|
|
* GstAudioEncoder calls @set_format to inform subclass of the format
|
|
of input audio data that it is about to receive. Subclass should
|
|
setup for encoding and configure various base class parameters
|
|
appropriately, notably those directing desired input data handling.
|
|
While unlikely, it might be called more than once, if changing input
|
|
parameters require reconfiguration.
|
|
|
|
* GstAudioEncoder calls @stop at end of all processing.
|
|
|
|
As of configuration stage, and throughout processing, GstAudioEncoder
|
|
maintains various parameters that provide required context,
|
|
e.g. describing the format of input audio data.
|
|
Conversely, subclass can and should configure these context parameters
|
|
to inform base class of its expectation w.r.t. buffer handling.
|
|
|
|
## Data processing
|
|
|
|
* Base class gathers input sample data (as directed by the context's
|
|
frame_samples and frame_max) and provides this to subclass' @handle_frame.
|
|
* If codec processing results in encoded data, subclass should call
|
|
gst_audio_encoder_finish_frame() to have encoded data pushed
|
|
downstream. Alternatively, it might also call
|
|
gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
|
|
dropped samples) to indicate dropped (non-encoded) samples.
|
|
* Just prior to actually pushing a buffer downstream,
|
|
it is passed to @pre_push.
|
|
* During the parsing process GstAudioEncoderClass will handle both
|
|
srcpad and sinkpad events. Sink events will be passed to subclass
|
|
if @event callback has been provided.
|
|
|
|
## Shutdown phase
|
|
|
|
* GstAudioEncoder class calls @stop to inform the subclass that data
|
|
parsing will be stopped.
|
|
|
|
Subclass is responsible for providing pad template caps for
|
|
source and sink pads. The pads need to be named "sink" and "src". It also
|
|
needs to set the fixed caps on srcpad, when the format is ensured. This
|
|
is typically when base class calls subclass' @set_format function, though
|
|
it might be delayed until calling @gst_audio_encoder_finish_frame.
|
|
|
|
In summary, above process should have subclass concentrating on
|
|
codec data processing while leaving other matters to base class,
|
|
such as most notably timestamp handling. While it may exert more control
|
|
in this area (see e.g. @pre_push), it is very much not recommended.
|
|
|
|
In particular, base class will either favor tracking upstream timestamps
|
|
(at the possible expense of jitter) or aim to arrange for a perfect stream of
|
|
output timestamps, depending on #GstAudioEncoder:perfect-timestamp.
|
|
However, in the latter case, the input may not be so perfect or ideal, which
|
|
is handled as follows. An input timestamp is compared with the expected
|
|
timestamp as dictated by input sample stream and if the deviation is less
|
|
than #GstAudioEncoder:tolerance, the deviation is discarded.
|
|
Otherwise, it is considered a discontuinity and subsequent output timestamp
|
|
is resynced to the new position after performing configured discontinuity
|
|
processing. In the non-perfect-timestamp case, an upstream variation
|
|
exceeding tolerance only leads to marking DISCONT on subsequent outgoing
|
|
(while timestamps are adjusted to upstream regardless of variation).
|
|
While DISCONT is also marked in the perfect-timestamp case, this one
|
|
optionally (see #GstAudioEncoder:hard-resync)
|
|
performs some additional steps, such as clipping of (early) input samples
|
|
or draining all currently remaining input data, depending on the direction
|
|
of the discontuinity.
|
|
|
|
If perfect timestamps are arranged, it is also possible to request baseclass
|
|
(usually set by subclass) to provide additional buffer metadata (in OFFSET
|
|
and OFFSET_END) fields according to granule defined semantics currently
|
|
needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
|
|
including buffer) and OFFSET_END to corresponding timestamp (as determined
|
|
by same sample count and sample rate).
|
|
|
|
Things that subclass need to take care of:
|
|
|
|
* Provide pad templates
|
|
* Set source pad caps when appropriate
|
|
* Inform base class of buffer processing needs using context's
|
|
frame_samples and frame_bytes.
|
|
* Set user-configurable properties to sane defaults for format and
|
|
implementing codec at hand, e.g. those controlling timestamp behaviour
|
|
and discontinuity processing.
|
|
* Accept data in @handle_frame and provide encoded results to
|
|
gst_audio_encoder_finish_frame().</doc>
|
|
<source-position filename="gstaudioencoder.h" line="248"/>
|
|
<implements name="Gst.Preset"/>
|
|
<virtual-method name="close">
|
|
<source-position filename="gstaudioencoder.h" line="227"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="decide_allocation">
|
|
<source-position filename="gstaudioencoder.h" line="231"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="flush">
|
|
<source-position filename="gstaudioencoder.h" line="212"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="getcaps">
|
|
<source-position filename="gstaudioencoder.h" line="223"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="filter" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="handle_frame">
|
|
<source-position filename="gstaudioencoder.h" line="209"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="negotiate" invoker="negotiate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2803">Negotiate with downstream elements to currently configured #GstCaps.
|
|
Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
|
|
negotiate fails.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="229"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2811">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2805">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
<source-position filename="gstaudioencoder.h" line="225"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="pre_push">
|
|
<source-position filename="gstaudioencoder.h" line="214"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="propose_allocation">
|
|
<source-position filename="gstaudioencoder.h" line="233"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="set_format">
|
|
<source-position filename="gstaudioencoder.h" line="206"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="sink_event">
|
|
<source-position filename="gstaudioencoder.h" line="217"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="sink_query">
|
|
<source-position filename="gstaudioencoder.h" line="239"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="encoder" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="src_event">
|
|
<source-position filename="gstaudioencoder.h" line="220"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="src_query">
|
|
<source-position filename="gstaudioencoder.h" line="242"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="encoder" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="start">
|
|
<source-position filename="gstaudioencoder.h" line="202"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="stop">
|
|
<source-position filename="gstaudioencoder.h" line="204"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="transform_meta">
|
|
<source-position filename="gstaudioencoder.h" line="236"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="meta" transfer-ownership="none">
|
|
<type name="Gst.Meta" c:type="GstMeta*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="allocate_output_buffer"
|
|
c:identifier="gst_audio_encoder_allocate_output_buffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2881">Helper function that allocates a buffer to hold an encoded audio frame
|
|
for @enc's current output format.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="271"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2889">allocated buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2883">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="size" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2884">size of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="finish_frame"
|
|
c:identifier="gst_audio_encoder_finish_frame">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="717">Collects encoded data and pushes encoded data downstream.
|
|
Source pad caps must be set when this is called.
|
|
|
|
If @samples < 0, then best estimate is all samples provided to encoder
|
|
(subclass) so far. @buf may be NULL, in which case next number of @samples
|
|
are considered discarded, e.g. as a result of discontinuous transmission,
|
|
and a discontinuity is marked.
|
|
|
|
Note that samples received in #GstAudioEncoderClass.handle_frame()
|
|
may be invalidated by a call to this function.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="254"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="734">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="719">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="720">encoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="721">number of samples (per channel) represented by encoded data</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_allocator"
|
|
c:identifier="gst_audio_encoder_get_allocator">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2932">Lets #GstAudioEncoder sub-classes to know the memory @allocator
|
|
used by the base class and its @params.
|
|
|
|
Unref the @allocator after use it.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="366"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2934">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="allocator"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2935">the #GstAllocator
|
|
used</doc>
|
|
<type name="Gst.Allocator" c:type="GstAllocator**"/>
|
|
</parameter>
|
|
<parameter name="params"
|
|
direction="out"
|
|
caller-allocates="1"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2937">the
|
|
#GstAllocationParams of @allocator</doc>
|
|
<type name="Gst.AllocationParams" c:type="GstAllocationParams*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_audio_info"
|
|
c:identifier="gst_audio_encoder_get_audio_info">
|
|
<source-position filename="gstaudioencoder.h" line="277"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2137">a #GstAudioInfo describing the input audio format</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2135">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_drainable"
|
|
c:identifier="gst_audio_encoder_get_drainable">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2631">Queries encoder drain handling.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="363"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2637">TRUE if drainable handling is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2633">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_frame_max"
|
|
c:identifier="gst_audio_encoder_get_frame_max">
|
|
<source-position filename="gstaudioencoder.h" line="292"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2245">currently configured maximum handled frames</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2243">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_frame_samples_max"
|
|
c:identifier="gst_audio_encoder_get_frame_samples_max">
|
|
<source-position filename="gstaudioencoder.h" line="286"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2211">currently maximum requested samples per frame</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2209">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_frame_samples_min"
|
|
c:identifier="gst_audio_encoder_get_frame_samples_min">
|
|
<source-position filename="gstaudioencoder.h" line="280"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2174">currently minimum requested samples per frame</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2172">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_hard_min"
|
|
c:identifier="gst_audio_encoder_get_hard_min">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2585">Queries encoder hard minimum handling.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="356"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2591">TRUE if hard minimum handling is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2587">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_hard_resync"
|
|
c:identifier="gst_audio_encoder_get_hard_resync">
|
|
<source-position filename="gstaudioencoder.h" line="342"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_latency" c:identifier="gst_audio_encoder_get_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2317">Sets the variables pointed to by @min and @max to the currently configured
|
|
latency.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="304"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2319">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2320">a pointer to storage to hold minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
<parameter name="max"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full"
|
|
optional="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2321">a pointer to storage to hold maximum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_lookahead"
|
|
c:identifier="gst_audio_encoder_get_lookahead">
|
|
<source-position filename="gstaudioencoder.h" line="298"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2278">currently configured encoder lookahead</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2276">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_mark_granule"
|
|
c:identifier="gst_audio_encoder_get_mark_granule">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2403">Queries if the encoder will handle granule marking.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="328"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2409">TRUE if granule marking is enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2405">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_perfect_timestamp"
|
|
c:identifier="gst_audio_encoder_get_perfect_timestamp">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2449">Queries encoder perfect timestamp behaviour.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="335"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2455">TRUE if perfect timestamp setting enabled.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2451">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_tolerance"
|
|
c:identifier="gst_audio_encoder_get_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2539">Queries current audio jitter tolerance threshold.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="349"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2545">encoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2541">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="merge_tags" c:identifier="gst_audio_encoder_merge_tags">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2655">Sets the audio encoder tags and how they should be merged with any
|
|
upstream stream tags. This will override any tags previously-set
|
|
with gst_audio_encoder_merge_tags().
|
|
|
|
Note that this is provided for convenience, and the subclass is
|
|
not required to use this and can still do tag handling on its own.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="371"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2657">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tags"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2658">a #GstTagList to merge, or NULL to unset
|
|
previously-set tags</doc>
|
|
<type name="Gst.TagList" c:type="const GstTagList*"/>
|
|
</parameter>
|
|
<parameter name="mode" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2660">the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE</doc>
|
|
<type name="Gst.TagMergeMode" c:type="GstTagMergeMode"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="negotiate" c:identifier="gst_audio_encoder_negotiate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2803">Negotiate with downstream elements to currently configured #GstCaps.
|
|
Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
|
|
negotiate fails.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="268"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2811">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2805">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="proxy_getcaps"
|
|
c:identifier="gst_audio_encoder_proxy_getcaps">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="1482">Returns caps that express @caps (or sink template caps if @caps == NULL)
|
|
restricted to channel/rate combinations supported by downstream elements
|
|
(e.g. muxers).</doc>
|
|
<source-position filename="gstaudioencoder.h" line="259"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="1492">a #GstCaps owned by caller</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="1484">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="1485">initial caps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
<parameter name="filter"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="1486">filter caps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_allocation_caps"
|
|
c:identifier="gst_audio_encoder_set_allocation_caps"
|
|
version="1.10">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2361">Sets a caps in allocation query which are different from the set
|
|
pad's caps. Use this function before calling
|
|
gst_audio_encoder_negotiate(). Setting to %NULL the allocation
|
|
query will use the caps from the pad.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="318"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2363">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="allocation_caps"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2364">a #GstCaps or %NULL</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_drainable"
|
|
c:identifier="gst_audio_encoder_set_drainable">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2609">Configures encoder drain handling. If drainable, subclass might
|
|
be handed a NULL buffer to have it return any leftover encoded data.
|
|
Otherwise, it is not considered so capable and will only ever be passed
|
|
real data.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="359"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2611">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2612">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_frame_max"
|
|
c:identifier="gst_audio_encoder_set_frame_max">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2221">Sets max number of frames accepted at once (assumed minimally 1).
|
|
Requires @frame_samples_min and @frame_samples_max to be the equal.
|
|
|
|
Note: This value will be reset to 0 every time before
|
|
#GstAudioEncoderClass.set_format() is called.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="295"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2223">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2224">number of frames</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_frame_samples_max"
|
|
c:identifier="gst_audio_encoder_set_frame_samples_max">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2184">Sets number of samples (per channel) subclass needs to be handed,
|
|
at most or will be handed all available if 0.
|
|
|
|
If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
|
|
must be called with the same number.
|
|
|
|
Note: This value will be reset to 0 every time before
|
|
#GstAudioEncoderClass.set_format() is called.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="289"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2186">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2187">number of samples per frame</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_frame_samples_min"
|
|
c:identifier="gst_audio_encoder_set_frame_samples_min">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2147">Sets number of samples (per channel) subclass needs to be handed,
|
|
at least or will be handed all available if 0.
|
|
|
|
If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
|
|
must be called with the same number.
|
|
|
|
Note: This value will be reset to 0 every time before
|
|
#GstAudioEncoderClass.set_format() is called.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="283"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2149">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2150">number of samples per frame</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_hard_min"
|
|
c:identifier="gst_audio_encoder_set_hard_min">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2563">Configures encoder hard minimum handling. If enabled, subclass
|
|
will never be handed less samples than it configured, which otherwise
|
|
might occur near end-of-data handling. Instead, the leftover samples
|
|
will simply be discarded.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="352"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2565">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2566">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_hard_resync"
|
|
c:identifier="gst_audio_encoder_set_hard_resync">
|
|
<source-position filename="gstaudioencoder.h" line="338"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_headers" c:identifier="gst_audio_encoder_set_headers">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2340">Set the codec headers to be sent downstream whenever requested.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="314"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2342">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="headers" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2343">a list of
|
|
#GstBuffer containing the codec header</doc>
|
|
<type name="GLib.List" c:type="GList*">
|
|
<type name="Gst.Buffer"/>
|
|
</type>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_latency" c:identifier="gst_audio_encoder_set_latency">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2288">Sets encoder latency.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="309"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2290">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2291">minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="max" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2292">maximum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_lookahead"
|
|
c:identifier="gst_audio_encoder_set_lookahead">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2255">Sets encoder lookahead (in units of input rate samples)
|
|
|
|
Note: This value will be reset to 0 every time before
|
|
#GstAudioEncoderClass.set_format() is called.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="301"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2257">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2258">lookahead</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_mark_granule"
|
|
c:identifier="gst_audio_encoder_set_mark_granule">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2382">Enable or disable encoder granule handling.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="324"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2384">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2385">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_output_format"
|
|
c:identifier="gst_audio_encoder_set_output_format">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2835">Configure output caps on the srcpad of @enc.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="264"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2842">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2837">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2838">#GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_perfect_timestamp"
|
|
c:identifier="gst_audio_encoder_set_perfect_timestamp">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2427">Enable or disable encoder perfect output timestamp preference.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="331"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2429">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2430">new state</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_tolerance"
|
|
c:identifier="gst_audio_encoder_set_tolerance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2518">Configures encoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="345"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2520">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2521">new tolerance</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="hard-resync" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="mark-granule" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="perfect-timestamp"
|
|
writable="1"
|
|
transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="tolerance" writable="1" transfer-ownership="none">
|
|
<type name="gint64" c:type="gint64"/>
|
|
</property>
|
|
<field name="element">
|
|
<type name="Gst.Element" c:type="GstElement"/>
|
|
</field>
|
|
<field name="sinkpad">
|
|
<type name="Gst.Pad" c:type="GstPad*"/>
|
|
</field>
|
|
<field name="srcpad">
|
|
<type name="Gst.Pad" c:type="GstPad*"/>
|
|
</field>
|
|
<field name="stream_lock">
|
|
<type name="GLib.RecMutex" c:type="GRecMutex"/>
|
|
</field>
|
|
<field name="input_segment">
|
|
<type name="Gst.Segment" c:type="GstSegment"/>
|
|
</field>
|
|
<field name="output_segment">
|
|
<type name="Gst.Segment" c:type="GstSegment"/>
|
|
</field>
|
|
<field name="priv" readable="0" private="1">
|
|
<type name="AudioEncoderPrivate" c:type="GstAudioEncoderPrivate*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="20">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioEncoderClass"
|
|
c:type="GstAudioEncoderClass"
|
|
glib:is-gtype-struct-for="AudioEncoder">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="122">Subclasses can override any of the available virtual methods or not, as
|
|
needed. At minimum @set_format and @handle_frame needs to be overridden.</doc>
|
|
<source-position filename="gstaudioencoder.h" line="248"/>
|
|
<field name="element_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.h"
|
|
line="124">The parent class structure</doc>
|
|
<type name="Gst.ElementClass" c:type="GstElementClass"/>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
<source-position filename="gstaudioencoder.h" line="202"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="stop">
|
|
<callback name="stop">
|
|
<source-position filename="gstaudioencoder.h" line="204"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="set_format">
|
|
<callback name="set_format">
|
|
<source-position filename="gstaudioencoder.h" line="206"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="handle_frame">
|
|
<callback name="handle_frame">
|
|
<source-position filename="gstaudioencoder.h" line="209"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="flush">
|
|
<callback name="flush">
|
|
<source-position filename="gstaudioencoder.h" line="212"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="pre_push">
|
|
<callback name="pre_push">
|
|
<source-position filename="gstaudioencoder.h" line="214"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="sink_event">
|
|
<callback name="sink_event">
|
|
<source-position filename="gstaudioencoder.h" line="217"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="src_event">
|
|
<callback name="src_event">
|
|
<source-position filename="gstaudioencoder.h" line="220"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="event" transfer-ownership="none">
|
|
<type name="Gst.Event" c:type="GstEvent*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="getcaps">
|
|
<callback name="getcaps">
|
|
<source-position filename="gstaudioencoder.h" line="223"/>
|
|
<return-value transfer-ownership="full">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="filter" transfer-ownership="none">
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
<source-position filename="gstaudioencoder.h" line="225"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close">
|
|
<callback name="close">
|
|
<source-position filename="gstaudioencoder.h" line="227"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="negotiate">
|
|
<callback name="negotiate">
|
|
<source-position filename="gstaudioencoder.h" line="229"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2811">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioencoder.c"
|
|
line="2805">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="decide_allocation">
|
|
<callback name="decide_allocation">
|
|
<source-position filename="gstaudioencoder.h" line="231"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="propose_allocation">
|
|
<callback name="propose_allocation">
|
|
<source-position filename="gstaudioencoder.h" line="233"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="transform_meta">
|
|
<callback name="transform_meta">
|
|
<source-position filename="gstaudioencoder.h" line="236"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="outbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="meta" transfer-ownership="none">
|
|
<type name="Gst.Meta" c:type="GstMeta*"/>
|
|
</parameter>
|
|
<parameter name="inbuf" transfer-ownership="none">
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="sink_query">
|
|
<callback name="sink_query">
|
|
<source-position filename="gstaudioencoder.h" line="239"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="encoder" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
<parameter name="query" transfer-ownership="none">
|
|
<type name="Gst.Query" c:type="GstQuery*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="src_query">
|
|
<callback name="src_query">
|
|
<source-position filename="gstaudioencoder.h" line="242"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="encoder" transfer-ownership="none">
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</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="17">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioEncoderPrivate"
|
|
c:type="GstAudioEncoderPrivate"
|
|
disguised="1">
|
|
<source-position filename="gstaudioencoder.h" line="92"/>
|
|
</record>
|
|
<class name="AudioFilter"
|
|
c:symbol-prefix="audio_filter"
|
|
c:type="GstAudioFilter"
|
|
parent="GstBase.BaseTransform"
|
|
abstract="1"
|
|
glib:type-name="GstAudioFilter"
|
|
glib:get-type="gst_audio_filter_get_type"
|
|
glib:type-struct="AudioFilterClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.c"
|
|
line="22">#GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio
|
|
filters, ie. those that output the same format that they get as input.
|
|
|
|
#GstAudioFilter will parse the input format for you (with error checking)
|
|
before calling your setup function. Also, elements deriving from
|
|
#GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
|
|
their class_init function to easily configure the set of caps/formats that
|
|
the element is able to handle.
|
|
|
|
Derived classes should override the #GstAudioFilterClass.setup() and
|
|
#GstBaseTransformClass.transform_ip() and/or
|
|
#GstBaseTransformClass.transform()
|
|
virtual functions in their class_init function.</doc>
|
|
<source-position filename="gstaudiofilter.h" line="93"/>
|
|
<virtual-method name="setup">
|
|
<source-position filename="gstaudiofilter.h" line="89"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="filter" transfer-ownership="none">
|
|
<type name="AudioFilter" c:type="GstAudioFilter*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<field name="basetransform">
|
|
<type name="GstBase.BaseTransform" c:type="GstBaseTransform"/>
|
|
</field>
|
|
<field name="info">
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioFilterClass"
|
|
c:type="GstAudioFilterClass"
|
|
glib:is-gtype-struct-for="AudioFilter">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.h"
|
|
line="75">In addition to the @setup virtual function, you should also override the
|
|
GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual
|
|
function.</doc>
|
|
<source-position filename="gstaudiofilter.h" line="93"/>
|
|
<field name="basetransformclass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.h"
|
|
line="77">parent class</doc>
|
|
<type name="GstBase.BaseTransformClass"
|
|
c:type="GstBaseTransformClass"/>
|
|
</field>
|
|
<field name="setup">
|
|
<callback name="setup">
|
|
<source-position filename="gstaudiofilter.h" line="89"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="filter" transfer-ownership="none">
|
|
<type name="AudioFilter" c:type="GstAudioFilter*"/>
|
|
</parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<method name="add_pad_templates"
|
|
c:identifier="gst_audio_filter_class_add_pad_templates">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.c"
|
|
line="206">Convenience function to add pad templates to this element class, with
|
|
@allowed_caps as the caps that can be handled.
|
|
|
|
This function is usually used from within a GObject class_init function.</doc>
|
|
<source-position filename="gstaudiofilter.h" line="99"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="klass" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.c"
|
|
line="208">an #GstAudioFilterClass</doc>
|
|
<type name="AudioFilterClass" c:type="GstAudioFilterClass*"/>
|
|
</instance-parameter>
|
|
<parameter name="allowed_caps" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiofilter.c"
|
|
line="209">what formats the filter can handle, as #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<bitfield name="AudioFlags"
|
|
glib:type-name="GstAudioFlags"
|
|
glib:get-type="gst_audio_flags_get_type"
|
|
c:type="GstAudioFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="33">Extra audio flags</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_FLAG_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="35">no valid flag</doc>
|
|
</member>
|
|
<member name="unpositioned"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_FLAG_UNPOSITIONED"
|
|
glib:nick="unpositioned">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="36">the position array explicitly
|
|
contains unpositioned channels.</doc>
|
|
</member>
|
|
</bitfield>
|
|
<enumeration name="AudioFormat"
|
|
glib:type-name="GstAudioFormat"
|
|
glib:get-type="gst_audio_format_get_type"
|
|
c:type="GstAudioFormat">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="39">Enum value describing the most common audio formats.</doc>
|
|
<member name="unknown"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_FORMAT_UNKNOWN"
|
|
glib:nick="unknown">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="41">unknown or unset audio format</doc>
|
|
</member>
|
|
<member name="encoded"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_FORMAT_ENCODED"
|
|
glib:nick="encoded">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="42">encoded audio format</doc>
|
|
</member>
|
|
<member name="s8"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_FORMAT_S8"
|
|
glib:nick="s8">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="43">8 bits in 8 bits, signed</doc>
|
|
</member>
|
|
<member name="u8"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_FORMAT_U8"
|
|
glib:nick="u8">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="44">8 bits in 8 bits, unsigned</doc>
|
|
</member>
|
|
<member name="s16le"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_FORMAT_S16LE"
|
|
glib:nick="s16le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="45">16 bits in 16 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s16be"
|
|
value="5"
|
|
c:identifier="GST_AUDIO_FORMAT_S16BE"
|
|
glib:nick="s16be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="46">16 bits in 16 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u16le"
|
|
value="6"
|
|
c:identifier="GST_AUDIO_FORMAT_U16LE"
|
|
glib:nick="u16le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="47">16 bits in 16 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u16be"
|
|
value="7"
|
|
c:identifier="GST_AUDIO_FORMAT_U16BE"
|
|
glib:nick="u16be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="48">16 bits in 16 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="s24_32le"
|
|
value="8"
|
|
c:identifier="GST_AUDIO_FORMAT_S24_32LE"
|
|
glib:nick="s24-32le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="49">24 bits in 32 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s24_32be"
|
|
value="9"
|
|
c:identifier="GST_AUDIO_FORMAT_S24_32BE"
|
|
glib:nick="s24-32be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="50">24 bits in 32 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u24_32le"
|
|
value="10"
|
|
c:identifier="GST_AUDIO_FORMAT_U24_32LE"
|
|
glib:nick="u24-32le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="51">24 bits in 32 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u24_32be"
|
|
value="11"
|
|
c:identifier="GST_AUDIO_FORMAT_U24_32BE"
|
|
glib:nick="u24-32be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="52">24 bits in 32 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="s32le"
|
|
value="12"
|
|
c:identifier="GST_AUDIO_FORMAT_S32LE"
|
|
glib:nick="s32le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="53">32 bits in 32 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s32be"
|
|
value="13"
|
|
c:identifier="GST_AUDIO_FORMAT_S32BE"
|
|
glib:nick="s32be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="54">32 bits in 32 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u32le"
|
|
value="14"
|
|
c:identifier="GST_AUDIO_FORMAT_U32LE"
|
|
glib:nick="u32le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="55">32 bits in 32 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u32be"
|
|
value="15"
|
|
c:identifier="GST_AUDIO_FORMAT_U32BE"
|
|
glib:nick="u32be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="56">32 bits in 32 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="s24le"
|
|
value="16"
|
|
c:identifier="GST_AUDIO_FORMAT_S24LE"
|
|
glib:nick="s24le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="57">24 bits in 24 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s24be"
|
|
value="17"
|
|
c:identifier="GST_AUDIO_FORMAT_S24BE"
|
|
glib:nick="s24be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="58">24 bits in 24 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u24le"
|
|
value="18"
|
|
c:identifier="GST_AUDIO_FORMAT_U24LE"
|
|
glib:nick="u24le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="59">24 bits in 24 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u24be"
|
|
value="19"
|
|
c:identifier="GST_AUDIO_FORMAT_U24BE"
|
|
glib:nick="u24be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="60">24 bits in 24 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="s20le"
|
|
value="20"
|
|
c:identifier="GST_AUDIO_FORMAT_S20LE"
|
|
glib:nick="s20le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="61">20 bits in 24 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s20be"
|
|
value="21"
|
|
c:identifier="GST_AUDIO_FORMAT_S20BE"
|
|
glib:nick="s20be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="62">20 bits in 24 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u20le"
|
|
value="22"
|
|
c:identifier="GST_AUDIO_FORMAT_U20LE"
|
|
glib:nick="u20le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="63">20 bits in 24 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u20be"
|
|
value="23"
|
|
c:identifier="GST_AUDIO_FORMAT_U20BE"
|
|
glib:nick="u20be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="64">20 bits in 24 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="s18le"
|
|
value="24"
|
|
c:identifier="GST_AUDIO_FORMAT_S18LE"
|
|
glib:nick="s18le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="65">18 bits in 24 bits, signed, little endian</doc>
|
|
</member>
|
|
<member name="s18be"
|
|
value="25"
|
|
c:identifier="GST_AUDIO_FORMAT_S18BE"
|
|
glib:nick="s18be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="66">18 bits in 24 bits, signed, big endian</doc>
|
|
</member>
|
|
<member name="u18le"
|
|
value="26"
|
|
c:identifier="GST_AUDIO_FORMAT_U18LE"
|
|
glib:nick="u18le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="67">18 bits in 24 bits, unsigned, little endian</doc>
|
|
</member>
|
|
<member name="u18be"
|
|
value="27"
|
|
c:identifier="GST_AUDIO_FORMAT_U18BE"
|
|
glib:nick="u18be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="68">18 bits in 24 bits, unsigned, big endian</doc>
|
|
</member>
|
|
<member name="f32le"
|
|
value="28"
|
|
c:identifier="GST_AUDIO_FORMAT_F32LE"
|
|
glib:nick="f32le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="69">32-bit floating point samples, little endian</doc>
|
|
</member>
|
|
<member name="f32be"
|
|
value="29"
|
|
c:identifier="GST_AUDIO_FORMAT_F32BE"
|
|
glib:nick="f32be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="70">32-bit floating point samples, big endian</doc>
|
|
</member>
|
|
<member name="f64le"
|
|
value="30"
|
|
c:identifier="GST_AUDIO_FORMAT_F64LE"
|
|
glib:nick="f64le">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="71">64-bit floating point samples, little endian</doc>
|
|
</member>
|
|
<member name="f64be"
|
|
value="31"
|
|
c:identifier="GST_AUDIO_FORMAT_F64BE"
|
|
glib:nick="f64be">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="72">64-bit floating point samples, big endian</doc>
|
|
</member>
|
|
<member name="s16"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_FORMAT_S16"
|
|
glib:nick="s16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="73">16 bits in 16 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u16"
|
|
value="6"
|
|
c:identifier="GST_AUDIO_FORMAT_U16"
|
|
glib:nick="u16">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="74">16 bits in 16 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="s24_32"
|
|
value="8"
|
|
c:identifier="GST_AUDIO_FORMAT_S24_32"
|
|
glib:nick="s24-32">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="75">24 bits in 32 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u24_32"
|
|
value="10"
|
|
c:identifier="GST_AUDIO_FORMAT_U24_32"
|
|
glib:nick="u24-32">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="76">24 bits in 32 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="s32"
|
|
value="12"
|
|
c:identifier="GST_AUDIO_FORMAT_S32"
|
|
glib:nick="s32">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="77">32 bits in 32 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u32"
|
|
value="14"
|
|
c:identifier="GST_AUDIO_FORMAT_U32"
|
|
glib:nick="u32">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="78">32 bits in 32 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="s24"
|
|
value="16"
|
|
c:identifier="GST_AUDIO_FORMAT_S24"
|
|
glib:nick="s24">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="79">24 bits in 24 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u24"
|
|
value="18"
|
|
c:identifier="GST_AUDIO_FORMAT_U24"
|
|
glib:nick="u24">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="80">24 bits in 24 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="s20"
|
|
value="20"
|
|
c:identifier="GST_AUDIO_FORMAT_S20"
|
|
glib:nick="s20">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="81">20 bits in 24 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u20"
|
|
value="22"
|
|
c:identifier="GST_AUDIO_FORMAT_U20"
|
|
glib:nick="u20">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="82">20 bits in 24 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="s18"
|
|
value="24"
|
|
c:identifier="GST_AUDIO_FORMAT_S18"
|
|
glib:nick="s18">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="83">18 bits in 24 bits, signed, native endianness</doc>
|
|
</member>
|
|
<member name="u18"
|
|
value="26"
|
|
c:identifier="GST_AUDIO_FORMAT_U18"
|
|
glib:nick="u18">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="84">18 bits in 24 bits, unsigned, native endianness</doc>
|
|
</member>
|
|
<member name="f32"
|
|
value="28"
|
|
c:identifier="GST_AUDIO_FORMAT_F32"
|
|
glib:nick="f32">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="85">32-bit floating point samples, native endianness</doc>
|
|
</member>
|
|
<member name="f64"
|
|
value="30"
|
|
c:identifier="GST_AUDIO_FORMAT_F64"
|
|
glib:nick="f64">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="86">64-bit floating point samples, native endianness</doc>
|
|
</member>
|
|
<function name="build_integer"
|
|
c:identifier="gst_audio_format_build_integer">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="358">Construct a #GstAudioFormat with given parameters.</doc>
|
|
<source-position filename="audio-format.h" line="275"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="367">a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
|
|
exists with the given parameters.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sign" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="360">signed or unsigned format</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="361">G_LITTLE_ENDIAN or G_BIG_ENDIAN</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="width" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="362">amount of bits used per sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="depth" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="363">amount of used bits in @width</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="fill_silence"
|
|
c:identifier="gst_audio_format_fill_silence">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="454">Fill @length bytes in @dest with silence samples for @info.</doc>
|
|
<source-position filename="audio-format.h" line="289"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="456">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="457">a destination
|
|
to fill</doc>
|
|
<array length="2" zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="459">the length to fill</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="from_string" c:identifier="gst_audio_format_from_string">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="404">Convert the @format string to its #GstAudioFormat.</doc>
|
|
<source-position filename="audio-format.h" line="279"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="410">the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
|
|
string is not a known format.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="406">a format string</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="get_info" c:identifier="gst_audio_format_get_info">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="438">Get the #GstAudioFormatInfo for @format</doc>
|
|
<source-position filename="audio-format.h" line="286"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="444">The #GstAudioFormatInfo for @format.</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="440">a #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="to_string" c:identifier="gst_audio_format_to_string">
|
|
<source-position filename="audio-format.h" line="282"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</enumeration>
|
|
<bitfield name="AudioFormatFlags"
|
|
glib:type-name="GstAudioFormatFlags"
|
|
glib:get-type="gst_audio_format_flags_get_type"
|
|
c:type="GstAudioFormatFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="151">The different audio flags that a format info can have.</doc>
|
|
<member name="integer"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_FORMAT_FLAG_INTEGER"
|
|
glib:nick="integer">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="153">integer samples</doc>
|
|
</member>
|
|
<member name="float"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_FORMAT_FLAG_FLOAT"
|
|
glib:nick="float">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="154">float samples</doc>
|
|
</member>
|
|
<member name="signed"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_FORMAT_FLAG_SIGNED"
|
|
glib:nick="signed">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="155">signed samples</doc>
|
|
</member>
|
|
<member name="complex"
|
|
value="16"
|
|
c:identifier="GST_AUDIO_FORMAT_FLAG_COMPLEX"
|
|
glib:nick="complex">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="156">complex layout</doc>
|
|
</member>
|
|
<member name="unpack"
|
|
value="32"
|
|
c:identifier="GST_AUDIO_FORMAT_FLAG_UNPACK"
|
|
glib:nick="unpack">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="157">the format can be used in
|
|
#GstAudioFormatUnpack and #GstAudioFormatPack functions</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioFormatInfo" c:type="GstAudioFormatInfo">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="221">Information for an audio format.</doc>
|
|
<source-position filename="audio-format.h" line="254"/>
|
|
<field name="format" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="223">#GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</field>
|
|
<field name="name" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="224">string representation of the format</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</field>
|
|
<field name="description" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="225">user readable description of the format</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</field>
|
|
<field name="flags" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="226">#GstAudioFormatFlags</doc>
|
|
<type name="AudioFormatFlags" c:type="GstAudioFormatFlags"/>
|
|
</field>
|
|
<field name="endianness" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="227">the endianness</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="width" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="228">amount of bits used for one sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="depth" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="229">amount of valid bits in @width</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="silence" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="230">@width/8 bytes with 1 silent sample</doc>
|
|
<array zero-terminated="0" fixed-size="8">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</field>
|
|
<field name="unpack_format" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="231">the format of the unpacked samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</field>
|
|
<field name="unpack_func" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="232">function to unpack samples</doc>
|
|
<type name="AudioFormatUnpack" c:type="GstAudioFormatUnpack"/>
|
|
</field>
|
|
<field name="pack_func" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="233">function to pack samples</doc>
|
|
<type name="AudioFormatPack" c:type="GstAudioFormatPack"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<callback name="AudioFormatPack" c:type="GstAudioFormatPack">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="204">Packs @length samples from @src to the data array in format @info.
|
|
The samples from source have each channel interleaved
|
|
and will be packed into @data.</doc>
|
|
<source-position filename="audio-format.h" line="217"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="206">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="207">#GstAudioPackFlags</doc>
|
|
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
|
|
</parameter>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="208">a source array</doc>
|
|
<array zero-terminated="0" c:type="gconstpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="209">pointer to the destination
|
|
data</doc>
|
|
<array zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="211">the amount of samples to pack.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<callback name="AudioFormatUnpack" c:type="GstAudioFormatUnpack">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="188">Unpacks @length samples from the given data of format @info.
|
|
The samples will be unpacked into @dest which each channel
|
|
interleaved. @dest should at least be big enough to hold @length *
|
|
channels * size(unpack_format) bytes.</doc>
|
|
<source-position filename="audio-format.h" line="201"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="190">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="191">#GstAudioPackFlags</doc>
|
|
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="192">a destination array</doc>
|
|
<array zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="193">pointer to the audio data</doc>
|
|
<array zero-terminated="0" c:type="gconstpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="194">the amount of samples to unpack.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<record name="AudioInfo"
|
|
c:type="GstAudioInfo"
|
|
glib:type-name="GstAudioInfo"
|
|
glib:get-type="gst_audio_info_get_type"
|
|
c:symbol-prefix="audio_info">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="58">Information describing audio properties. This information can be filled
|
|
in from GstCaps with gst_audio_info_from_caps().
|
|
|
|
Use the provided macros to access the info in this structure.</doc>
|
|
<source-position filename="audio-info.h" line="85"/>
|
|
<field name="finfo" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="60">the format info of the audio</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</field>
|
|
<field name="flags" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="61">additional audio flags</doc>
|
|
<type name="AudioFlags" c:type="GstAudioFlags"/>
|
|
</field>
|
|
<field name="layout" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="62">audio layout</doc>
|
|
<type name="AudioLayout" c:type="GstAudioLayout"/>
|
|
</field>
|
|
<field name="rate" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="63">the audio sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="channels" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="64">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="bpf" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="65">the number of bytes for one frame, this is the size of one
|
|
sample * @channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="position" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="67">the positions for each channel</doc>
|
|
<array zero-terminated="0" fixed-size="64">
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<constructor name="new" c:identifier="gst_audio_info_new">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="83">Allocate a new #GstAudioInfo that is also initialized with
|
|
gst_audio_info_init().</doc>
|
|
<source-position filename="audio-info.h" line="117"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="89">a new #GstAudioInfo. free with gst_audio_info_free().</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</return-value>
|
|
</constructor>
|
|
<method name="convert" c:identifier="gst_audio_info_convert">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="396">Converts among various #GstFormat types. This function handles
|
|
GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
|
|
raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
|
|
function can be used to handle pad queries of the type GST_QUERY_CONVERT.</doc>
|
|
<source-position filename="audio-info.h" line="140"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="409">TRUE if the conversion was successful.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="398">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="src_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="399">#GstFormat of the @src_val</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="src_val" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="400">value to convert</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</parameter>
|
|
<parameter name="dest_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="401">#GstFormat of the @dest_val</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="dest_val"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="402">pointer to destination value</doc>
|
|
<type name="gint64" c:type="gint64*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="copy" c:identifier="gst_audio_info_copy">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="53">Copy a GstAudioInfo structure.</doc>
|
|
<source-position filename="audio-info.h" line="123"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="59">a new #GstAudioInfo. free with gst_audio_info_free.</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="55">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free" c:identifier="gst_audio_info_free">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="67">Free a GstAudioInfo structure previously allocated with gst_audio_info_new()
|
|
or gst_audio_info_copy().</doc>
|
|
<source-position filename="audio-info.h" line="126"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="69">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="from_caps" c:identifier="gst_audio_info_from_caps">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="182">Parse @caps and update @info.</doc>
|
|
<source-position filename="audio-info.h" line="134"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="189">TRUE if @caps could be parsed</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="184">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="185">a #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="const GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="init" c:identifier="gst_audio_info_init">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="102">Initialize @info with default values.</doc>
|
|
<source-position filename="audio-info.h" line="120"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="104">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_equal"
|
|
c:identifier="gst_audio_info_is_equal"
|
|
version="1.2">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="489">Compares two #GstAudioInfo and returns whether they are equal or not</doc>
|
|
<source-position filename="audio-info.h" line="145"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="496">%TRUE if @info and @other are equal, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="491">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="other" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="492">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_format" c:identifier="gst_audio_info_set_format">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="118">Set the default info for the audio info of @format and @rate and @channels.
|
|
|
|
Note: This initializes @info first, no values are preserved.</doc>
|
|
<source-position filename="audio-info.h" line="129"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="120">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="121">the format</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="122">the samplerate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="123">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="position"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="124">the channel positions</doc>
|
|
<array zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*"
|
|
fixed-size="64">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="to_caps" c:identifier="gst_audio_info_to_caps">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="315">Convert the values of @info into a #GstCaps.</doc>
|
|
<source-position filename="audio-info.h" line="137"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="321">the new #GstCaps containing the
|
|
info of @info.</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.c"
|
|
line="317">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<enumeration name="AudioLayout"
|
|
glib:type-name="GstAudioLayout"
|
|
glib:get-type="gst_audio_layout_get_type"
|
|
c:type="GstAudioLayout">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="46">Layout of the audio samples for the different channels.</doc>
|
|
<member name="interleaved"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_LAYOUT_INTERLEAVED"
|
|
glib:nick="interleaved">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="48">interleaved audio</doc>
|
|
</member>
|
|
<member name="non_interleaved"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_LAYOUT_NON_INTERLEAVED"
|
|
glib:nick="non-interleaved">
|
|
<doc xml:space="preserve"
|
|
filename="audio-info.h"
|
|
line="49">non-interleaved audio</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioMeta" c:type="GstAudioMeta" version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="20">#GstAudioDownmixMeta defines an audio downmix matrix to be send along with
|
|
audio buffers. These functions in this module help to create and attach the
|
|
meta as well as extracting it.</doc>
|
|
<source-position filename="gstaudiometa.h" line="185"/>
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="136">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="137">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="samples" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="138">the number of valid samples in the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</field>
|
|
<field name="offsets" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.h"
|
|
line="139">the offsets (in bytes) where each channel plane starts in the
|
|
buffer or %NULL if the buffer has interleaved layout; if not %NULL, this
|
|
is guaranteed to be an array of @info.channels elements</doc>
|
|
<type name="gsize" c:type="gsize*"/>
|
|
</field>
|
|
<field name="priv_offsets_arr" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="8">
|
|
<type name="gsize" c:type="gsize"/>
|
|
</array>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
<function name="get_info" c:identifier="gst_audio_meta_get_info">
|
|
<source-position filename="gstaudiometa.h" line="191"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
</record>
|
|
<enumeration name="AudioNoiseShapingMethod"
|
|
glib:type-name="GstAudioNoiseShapingMethod"
|
|
glib:get-type="gst_audio_noise_shaping_method_get_type"
|
|
c:type="GstAudioNoiseShapingMethod">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="49">Set of available noise shaping methods</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_NOISE_SHAPING_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="51">No noise shaping (default)</doc>
|
|
</member>
|
|
<member name="error_feedback"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK"
|
|
glib:nick="error-feedback">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="52">Error feedback</doc>
|
|
</member>
|
|
<member name="simple"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_NOISE_SHAPING_SIMPLE"
|
|
glib:nick="simple">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="53">Simple 2-pole noise shaping</doc>
|
|
</member>
|
|
<member name="medium"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_NOISE_SHAPING_MEDIUM"
|
|
glib:nick="medium">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="54">Medium 5-pole noise shaping</doc>
|
|
</member>
|
|
<member name="high"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_NOISE_SHAPING_HIGH"
|
|
glib:nick="high">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="55">High 8-pole noise shaping</doc>
|
|
</member>
|
|
</enumeration>
|
|
<bitfield name="AudioPackFlags"
|
|
glib:type-name="GstAudioPackFlags"
|
|
glib:get-type="gst_audio_pack_flags_get_type"
|
|
c:type="GstAudioPackFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="171">The different flags that can be used when packing and unpacking.</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_PACK_FLAG_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="173">No flag</doc>
|
|
</member>
|
|
<member name="truncate_range"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE"
|
|
glib:nick="truncate-range">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.h"
|
|
line="174">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
|
|
is not specified, the most significant bits of the source are duplicated
|
|
in the least significant bits of the destination.</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioQuantize" c:type="GstAudioQuantize" disguised="1">
|
|
<source-position filename="audio-quantize.h" line="82"/>
|
|
<method name="free" c:identifier="gst_audio_quantize_free">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="470">Free a #GstAudioQuantize.</doc>
|
|
<source-position filename="audio-quantize.h" line="93"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="quant" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="472">a #GstAudioQuantize</doc>
|
|
<type name="AudioQuantize" c:type="GstAudioQuantize*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="reset" c:identifier="gst_audio_quantize_reset">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="489">Reset @quant to the state is was when created, clearing any
|
|
history it might have.</doc>
|
|
<source-position filename="audio-quantize.h" line="96"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="quant" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="491">a #GstAudioQuantize</doc>
|
|
<type name="AudioQuantize" c:type="GstAudioQuantize*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="samples" c:identifier="gst_audio_quantize_samples">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="504">Perform quantization on @samples in @in and write the result to @out.
|
|
|
|
In case the samples are interleaved, @in and @out must point to an
|
|
array with a single element pointing to a block of interleaved samples.
|
|
|
|
If non-interleaved samples are used, @in and @out must point to an
|
|
array with pointers to memory blocks, one for each channel.
|
|
|
|
@in and @out may point to the same memory location, in which case samples will be
|
|
modified in-place.</doc>
|
|
<source-position filename="audio-quantize.h" line="99"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="quant" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="506">a #GstAudioQuantize</doc>
|
|
<type name="AudioQuantize" c:type="GstAudioQuantize*"/>
|
|
</instance-parameter>
|
|
<parameter name="in"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="507">input samples</doc>
|
|
<type name="gpointer" c:type="const gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="508">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="509">number of samples</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<function name="new"
|
|
c:identifier="gst_audio_quantize_new"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="413">Create a new quantizer object with the given parameters.
|
|
|
|
Output samples will be quantized to a multiple of @quantizer. Better
|
|
performance is achieved when @quantizer is a power of 2.
|
|
|
|
Dithering and noise-shaping can be performed during quantization with
|
|
the @dither and @ns parameters.</doc>
|
|
<source-position filename="audio-quantize.h" line="85"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="430">a new #GstAudioQuantize. Free with gst_audio_quantize_free().</doc>
|
|
<type name="AudioQuantize" c:type="GstAudioQuantize*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dither" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="415">a #GstAudioDitherMethod</doc>
|
|
<type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/>
|
|
</parameter>
|
|
<parameter name="ns" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="416">a #GstAudioNoiseShapingMethod</doc>
|
|
<type name="AudioNoiseShapingMethod"
|
|
c:type="GstAudioNoiseShapingMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="417">#GstAudioQuantizeFlags</doc>
|
|
<type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="418">the #GstAudioFormat of the samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="419">the amount of channels in the samples</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="quantizer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="420">the quantizer to use</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<bitfield name="AudioQuantizeFlags"
|
|
glib:type-name="GstAudioQuantizeFlags"
|
|
glib:get-type="gst_audio_quantize_flags_get_type"
|
|
c:type="GstAudioQuantizeFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="68">Extra flags that can be passed to gst_audio_quantize_new()</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_QUANTIZE_FLAG_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="70">no flags</doc>
|
|
</member>
|
|
<member name="non_interleaved"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED"
|
|
glib:nick="non-interleaved">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.h"
|
|
line="71">samples are non-interleaved</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioResampler" c:type="GstAudioResampler" disguised="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="43">#GstAudioResampler is a structure which holds the information
|
|
required to perform various kinds of resampling filtering.</doc>
|
|
<source-position filename="audio-resampler.h" line="28"/>
|
|
<method name="free"
|
|
c:identifier="gst_audio_resampler_free"
|
|
version="1.6">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1620">Free a previously allocated #GstAudioResampler @resampler.</doc>
|
|
<source-position filename="audio-resampler.h" line="218"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1622">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_in_frames"
|
|
c:identifier="gst_audio_resampler_get_in_frames">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1682">Get the number of input frames that would currently be needed
|
|
to produce @out_frames from @resampler.</doc>
|
|
<source-position filename="audio-resampler.h" line="233"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1690">The number of input frames needed for producing
|
|
@out_frames of data from @resampler.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1684">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1685">number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_max_latency"
|
|
c:identifier="gst_audio_resampler_get_max_latency">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1709">Get the maximum number of input samples that the resampler would
|
|
need before producing output.</doc>
|
|
<source-position filename="audio-resampler.h" line="237"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1716">the latency of @resampler as expressed in the number of
|
|
frames.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1711">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_out_frames"
|
|
c:identifier="gst_audio_resampler_get_out_frames">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1643">Get the number of output frames that would be currently available when
|
|
@in_frames are given to @resampler.</doc>
|
|
<source-position filename="audio-resampler.h" line="229"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1651">The number of frames that would be availabe after giving
|
|
@in_frames as input to @resampler.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1645">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1646">number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="resample" c:identifier="gst_audio_resampler_resample">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1727">Perform resampling on @in_frames frames in @in and write @out_frames to @out.
|
|
|
|
In case the samples are interleaved, @in and @out must point to an
|
|
array with a single element pointing to a block of interleaved samples.
|
|
|
|
If non-interleaved samples are used, @in and @out must point to an
|
|
array with pointers to memory blocks, one for each channel.
|
|
|
|
@in may be %NULL, in which case @in_frames of silence samples are pushed
|
|
into the resampler.
|
|
|
|
This function always produces @out_frames of output and consumes @in_frames of
|
|
input. Use gst_audio_resampler_get_out_frames() and
|
|
gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
|
|
are matching and @in and @out point to enough memory.</doc>
|
|
<source-position filename="audio-resampler.h" line="240"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1729">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="in"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1730">input samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1731">number of input frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="out"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1732">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1733">number of output frames</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="reset" c:identifier="gst_audio_resampler_reset">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1462">Reset @resampler to the state it was when it was first created, discarding
|
|
all sample history.</doc>
|
|
<source-position filename="audio-resampler.h" line="221"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1464">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="update" c:identifier="gst_audio_resampler_update">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1490">Update the resampler parameters for @resampler. This function should
|
|
not be called concurrently with any other function on @resampler.
|
|
|
|
When @in_rate or @out_rate is 0, its value is unchanged.
|
|
|
|
When @options is %NULL, the previously configured options are reused.</doc>
|
|
<source-position filename="audio-resampler.h" line="224"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1504">%TRUE if the new parameters could be set</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1492">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1493">new input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1494">new output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1495">new options or %NULL</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<function name="new" c:identifier="gst_audio_resampler_new">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1333">Make a new resampler.</doc>
|
|
<source-position filename="audio-resampler.h" line="211"/>
|
|
<return-value transfer-ownership="full" skip="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1345">The new #GstAudioResampler, or
|
|
%NULL on failure.</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1335">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod"
|
|
c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1336">#GstAudioResamplerFlags</doc>
|
|
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1337">the #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1338">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1339">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1340">output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1341">extra options</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="options_set_quality"
|
|
c:identifier="gst_audio_resampler_options_set_quality">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1269">Set the parameters for resampling from @in_rate to @out_rate using @method
|
|
for @quality in @options.</doc>
|
|
<source-position filename="audio-resampler.h" line="205"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1271">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod"
|
|
c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="quality" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1272">the quality</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1273">the input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1274">the output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1275">a #GstStructure</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<enumeration name="AudioResamplerFilterInterpolation"
|
|
glib:type-name="GstAudioResamplerFilterInterpolation"
|
|
glib:get-type="gst_audio_resampler_filter_interpolation_get_type"
|
|
c:type="GstAudioResamplerFilterInterpolation">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="116">The different filter interpolation methods.</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="118">no interpolation</doc>
|
|
</member>
|
|
<member name="linear"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR"
|
|
glib:nick="linear">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="119">linear interpolation of the
|
|
filter coeficients.</doc>
|
|
</member>
|
|
<member name="cubic"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC"
|
|
glib:nick="cubic">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="121">cubic interpolation of the
|
|
filter coeficients.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="AudioResamplerFilterMode"
|
|
glib:type-name="GstAudioResamplerFilterMode"
|
|
glib:get-type="gst_audio_resampler_filter_mode_get_type"
|
|
c:type="GstAudioResamplerFilterMode">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="82">Select for the filter tables should be set up.</doc>
|
|
<member name="interpolated"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED"
|
|
glib:nick="interpolated">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="84">Use interpolated filter tables. This
|
|
uses less memory but more CPU and is slightly less accurate but it allows for more
|
|
efficient variable rate resampling with gst_audio_resampler_update().</doc>
|
|
</member>
|
|
<member name="full"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_FULL"
|
|
glib:nick="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="87">Use full filter table. This uses more memory
|
|
but less CPU.</doc>
|
|
</member>
|
|
<member name="auto"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO"
|
|
glib:nick="auto">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="89">Automatically choose between interpolated
|
|
and full filter tables.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<bitfield name="AudioResamplerFlags"
|
|
glib:type-name="GstAudioResamplerFlags"
|
|
glib:get-type="gst_audio_resampler_flags_get_type"
|
|
c:type="GstAudioResamplerFlags">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="178">Different resampler flags.</doc>
|
|
<member name="none"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FLAG_NONE"
|
|
glib:nick="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="180">no flags</doc>
|
|
</member>
|
|
<member name="non_interleaved_in"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN"
|
|
glib:nick="non-interleaved-in">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="181">input samples are non-interleaved.
|
|
an array of blocks of samples, one for each channel, should be passed to the
|
|
resample function.</doc>
|
|
</member>
|
|
<member name="non_interleaved_out"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT"
|
|
glib:nick="non-interleaved-out">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="184">output samples are non-interleaved.
|
|
an array of blocks of samples, one for each channel, should be passed to the
|
|
resample function.</doc>
|
|
</member>
|
|
<member name="variable_rate"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE"
|
|
glib:nick="variable-rate">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="187">optimize for dynamic updates of the sample
|
|
rates with gst_audio_resampler_update(). This will select an interpolating filter
|
|
when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.</doc>
|
|
</member>
|
|
</bitfield>
|
|
<enumeration name="AudioResamplerMethod"
|
|
version="1.6"
|
|
glib:type-name="GstAudioResamplerMethod"
|
|
glib:get-type="gst_audio_resampler_method_get_type"
|
|
c:type="GstAudioResamplerMethod">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="156">Different subsampling and upsampling methods</doc>
|
|
<member name="nearest"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RESAMPLER_METHOD_NEAREST"
|
|
glib:nick="nearest">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="158">Duplicates the samples when
|
|
upsampling and drops when downsampling</doc>
|
|
</member>
|
|
<member name="linear"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RESAMPLER_METHOD_LINEAR"
|
|
glib:nick="linear">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="160">Uses linear interpolation to reconstruct
|
|
missing samples and averaging to downsample</doc>
|
|
</member>
|
|
<member name="cubic"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RESAMPLER_METHOD_CUBIC"
|
|
glib:nick="cubic">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="162">Uses cubic interpolation</doc>
|
|
</member>
|
|
<member name="blackman_nuttall"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL"
|
|
glib:nick="blackman-nuttall">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="163">Uses Blackman-Nuttall windowed sinc interpolation</doc>
|
|
</member>
|
|
<member name="kaiser"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_RESAMPLER_METHOD_KAISER"
|
|
glib:nick="kaiser">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.h"
|
|
line="164">Uses Kaiser windowed sinc interpolation</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="AudioRingBuffer"
|
|
c:symbol-prefix="audio_ring_buffer"
|
|
c:type="GstAudioRingBuffer"
|
|
parent="Gst.Object"
|
|
abstract="1"
|
|
glib:type-name="GstAudioRingBuffer"
|
|
glib:get-type="gst_audio_ring_buffer_get_type"
|
|
glib:type-struct="AudioRingBufferClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="20">This object is the base class for audio ringbuffers used by the base
|
|
audio source and sink classes.
|
|
|
|
The ringbuffer abstracts a circular buffer of data. One reader and
|
|
one writer can operate on the data from different threads in a lockfree
|
|
manner. The base class is sufficiently flexible to be used as an
|
|
abstraction for DMA based ringbuffers as well as a pure software
|
|
implementations.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="268"/>
|
|
<function name="debug_spec_buff"
|
|
c:identifier="gst_audio_ring_buffer_debug_spec_buff">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="169">Print debug info about the buffer sized in @spec to the debug log.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="293"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="171">the spec to debug</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="debug_spec_caps"
|
|
c:identifier="gst_audio_ring_buffer_debug_spec_caps">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="139">Print debug info about the parsed caps in @spec to the debug log.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="290"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="141">the spec to debug</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="parse_caps"
|
|
c:identifier="gst_audio_ring_buffer_parse_caps">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="193">Parse @caps into @spec.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="287"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="200">TRUE if the caps could be parsed.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="195">a spec</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="196">a #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<virtual-method name="acquire" invoker="acquire">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="588">Allocate the resources for the ringbuffer. This function fills
|
|
in the data pointer of the ring buffer with a valid #GstBuffer
|
|
to which samples can be written.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="246"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="597">TRUE if the device could be acquired, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="590">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="591">the specs of the buffer</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="activate" invoker="activate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="805">Activate @buf to start or stop pulling data.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="258"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="814">TRUE if the device could be activated in the requested mode,
|
|
FALSE on error.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="807">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="808">the new mode</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="clear_all" invoker="clear_all">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1310">Fill the ringbuffer with silence.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="264"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1312">the #GstAudioRingBuffer to clear</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="close_device" invoker="close_device">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="497">Close the audio device associated with the ring buffer. The ring buffer
|
|
should already have been released via gst_audio_ring_buffer_release().</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="248"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="504">TRUE if the device could be closed, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="499">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="commit" invoker="commit">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1669">Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
|
|
|
|
@in_samples and @out_samples define the rate conversion to perform on the
|
|
samples in @data. For negative rates, @out_samples must be negative and
|
|
@in_samples positive.
|
|
|
|
When @out_samples is positive, the first sample will be written at position @sample
|
|
in the ringbuffer. When @out_samples is negative, the last sample will be written to
|
|
@sample in reverse order.
|
|
|
|
@out_samples does not need to be a multiple of the segment size of the ringbuffer
|
|
although it is recommended for optimal performance.
|
|
|
|
@accum will hold a temporary accumulator used in rate conversion and should be
|
|
set to 0 when this function is first called. In case the commit operation is
|
|
interrupted, one can resume the processing by passing the previously returned
|
|
@accum value back to this function.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="260"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1698">The number of samples written to the ringbuffer or -1 on error. The
|
|
number of samples written can be less than @out_samples when @buf was interrupted
|
|
with a flush or stop.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1671">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1672">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1673">the data to commit</doc>
|
|
<array length="2" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="in_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1674">the number of samples in the data to commit</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1675">the number of samples to write to the ringbuffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="accum"
|
|
direction="inout"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1676">accumulator for rate conversion.</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="delay" invoker="delay">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1183">Get the number of samples queued in the audio device. This is
|
|
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
|
|
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>
|
|
<source-position filename="gstaudioringbuffer.h" line="255"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1198">The number of samples queued in the audio device.
|
|
|
|
MT safe.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1185">the #GstAudioRingBuffer to query</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open_device" invoker="open_device">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="436">Open the audio device associated with the ring buffer. Does not perform any
|
|
setup on the device. You must open the device before acquiring the ring
|
|
buffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="245"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="444">TRUE if the device could be opened, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="438">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="pause" invoker="pause">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1079">Pause processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="251"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1085">TRUE if the device could be paused, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1081">the #GstAudioRingBuffer to pause</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="release" invoker="release">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="703">Free the resources of the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="247"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="709">TRUE if the device could be released, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="705">the #GstAudioRingBuffer to release</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="resume">
|
|
<source-position filename="gstaudioringbuffer.h" line="252"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="start" invoker="start">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="947">Start processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="250"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="953">TRUE if the device could be started, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="949">the #GstAudioRingBuffer to start</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="stop" invoker="stop">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1123">Stop processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="253"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1129">TRUE if the device could be stopped, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1125">the #GstAudioRingBuffer to stop</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<method name="acquire" c:identifier="gst_audio_ring_buffer_acquire">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="588">Allocate the resources for the ringbuffer. This function fills
|
|
in the data pointer of the ring buffer with a valid #GstBuffer
|
|
to which samples can be written.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="314"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="597">TRUE if the device could be acquired, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="590">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="591">the specs of the buffer</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="activate" c:identifier="gst_audio_ring_buffer_activate">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="805">Activate @buf to start or stop pulling data.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="330"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="814">TRUE if the device could be activated in the requested mode,
|
|
FALSE on error.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="807">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="808">the new mode</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="advance" c:identifier="gst_audio_ring_buffer_advance">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1915">Subclasses should call this function to notify the fact that
|
|
@advance segments are now processed by the device.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="402"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1917">the #GstAudioRingBuffer to advance</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="advance" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1918">the number of segments written</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="clear" c:identifier="gst_audio_ring_buffer_clear">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1944">Clear the given segment of the buffer with silence samples.
|
|
This function is used by subclasses.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="399"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1946">the #GstAudioRingBuffer to clear</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1947">the segment to clear</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="clear_all" c:identifier="gst_audio_ring_buffer_clear_all">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1310">Fill the ringbuffer with silence.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="368"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1312">the #GstAudioRingBuffer to clear</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="close_device"
|
|
c:identifier="gst_audio_ring_buffer_close_device">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="497">Close the audio device associated with the ring buffer. The ring buffer
|
|
should already have been released via gst_audio_ring_buffer_release().</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="306"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="504">TRUE if the device could be closed, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="499">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="commit" c:identifier="gst_audio_ring_buffer_commit">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1669">Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
|
|
|
|
@in_samples and @out_samples define the rate conversion to perform on the
|
|
samples in @data. For negative rates, @out_samples must be negative and
|
|
@in_samples positive.
|
|
|
|
When @out_samples is positive, the first sample will be written at position @sample
|
|
in the ringbuffer. When @out_samples is negative, the last sample will be written to
|
|
@sample in reverse order.
|
|
|
|
@out_samples does not need to be a multiple of the segment size of the ringbuffer
|
|
although it is recommended for optimal performance.
|
|
|
|
@accum will hold a temporary accumulator used in rate conversion and should be
|
|
set to 0 when this function is first called. In case the commit operation is
|
|
interrupted, one can resume the processing by passing the previously returned
|
|
@accum value back to this function.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="373"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1698">The number of samples written to the ringbuffer or -1 on error. The
|
|
number of samples written can be less than @out_samples when @buf was interrupted
|
|
with a flush or stop.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1671">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1672">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1673">the data to commit</doc>
|
|
<array length="2" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="in_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1674">the number of samples in the data to commit</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1675">the number of samples to write to the ringbuffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="accum"
|
|
direction="inout"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1676">accumulator for rate conversion.</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="convert" c:identifier="gst_audio_ring_buffer_convert">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="352">Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result
|
|
will be put in @dest_val.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="296"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="363">TRUE if the conversion succeeded.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="354">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="src_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="355">the source format</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="src_val" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="356">the source value</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</parameter>
|
|
<parameter name="dest_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="357">the destination format</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="dest_val"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="358">a location to store the converted value</doc>
|
|
<type name="gint64" c:type="gint64*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="delay" c:identifier="gst_audio_ring_buffer_delay">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1183">Get the number of samples queued in the audio device. This is
|
|
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
|
|
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>
|
|
<source-position filename="gstaudioringbuffer.h" line="357"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1198">The number of samples queued in the audio device.
|
|
|
|
MT safe.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1185">the #GstAudioRingBuffer to query</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="device_is_open"
|
|
c:identifier="gst_audio_ring_buffer_device_is_open">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="564">Checks the status of the device associated with the ring buffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="309"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="570">TRUE if the device was open, FALSE if it was closed.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="566">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_acquired"
|
|
c:identifier="gst_audio_ring_buffer_is_acquired">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="781">Check if the ringbuffer is acquired and ready to use.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="320"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="787">TRUE if the ringbuffer is acquired, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="783">the #GstAudioRingBuffer to check</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_active" c:identifier="gst_audio_ring_buffer_is_active">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="873">Check if @buf is activated.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="333"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="881">TRUE if the device is active.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="875">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="is_flushing"
|
|
c:identifier="gst_audio_ring_buffer_is_flushing">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="923">Check if @buf is flushing.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="341"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="931">TRUE if the device is flushing.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="925">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="may_start" c:identifier="gst_audio_ring_buffer_may_start">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1979">Tell the ringbuffer that it is allowed to start playback when
|
|
the ringbuffer is filled with samples.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="405"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1981">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="allowed" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1982">the new value</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="open_device"
|
|
c:identifier="gst_audio_ring_buffer_open_device">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="436">Open the audio device associated with the ring buffer. Does not perform any
|
|
setup on the device. You must open the device before acquiring the ring
|
|
buffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="303"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="444">TRUE if the device could be opened, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="438">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="pause" c:identifier="gst_audio_ring_buffer_pause">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1079">Pause processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="349"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1085">TRUE if the device could be paused, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1081">the #GstAudioRingBuffer to pause</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="prepare_read"
|
|
c:identifier="gst_audio_ring_buffer_prepare_read">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1860">Returns a pointer to memory where the data from segment @segment
|
|
can be found. This function is mostly used by subclasses.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="395"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1871">FALSE if the buffer is not started.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1862">the #GstAudioRingBuffer to read from</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="segment"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1863">the segment to read</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
<parameter name="readptr"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1864">
|
|
the pointer to the memory where samples can be read</doc>
|
|
<array length="2" zero-terminated="0" c:type="guint8**">
|
|
<type name="guint8" c:type="guint8*"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="len"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1866">the number of bytes to read</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="read" c:identifier="gst_audio_ring_buffer_read">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1722">Read @len samples from the ringbuffer into the memory pointed
|
|
to by @data.
|
|
The first sample should be read from position @sample in
|
|
the ringbuffer.
|
|
|
|
@len should not be a multiple of the segment size of the ringbuffer
|
|
although it is recommended.
|
|
|
|
@timestamp will return the timestamp associated with the data returned.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="380"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1740">The number of samples read from the ringbuffer or -1 on
|
|
error.
|
|
|
|
MT safe.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1724">the #GstAudioRingBuffer to read from</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1725">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1726">where the data should be read</doc>
|
|
<array length="2" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="len" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1727">the number of samples in data to read</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="timestamp"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1728">where the timestamp is returned</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="release" c:identifier="gst_audio_ring_buffer_release">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="703">Free the resources of the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="317"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="709">TRUE if the device could be released, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="705">the #GstAudioRingBuffer to release</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="samples_done"
|
|
c:identifier="gst_audio_ring_buffer_samples_done">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1229">Get the number of samples that were processed by the ringbuffer
|
|
since it was last started. This does not include the number of samples not
|
|
yet processed (see gst_audio_ring_buffer_delay()).</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="360"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1237">The number of samples processed by the ringbuffer.
|
|
|
|
MT safe.</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1231">the #GstAudioRingBuffer to query</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_callback"
|
|
c:identifier="gst_audio_ring_buffer_set_callback"
|
|
shadowed-by="set_callback_full"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="380">Sets the given callback function on the buffer. This function
|
|
will be called every time a segment has been written to a device.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="276"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="382">the #GstAudioRingBuffer to set the callback on</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="cb"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1"
|
|
closure="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="383">the callback to set</doc>
|
|
<type name="AudioRingBufferCallback"
|
|
c:type="GstAudioRingBufferCallback"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="384">user data passed to the callback</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_callback_full"
|
|
c:identifier="gst_audio_ring_buffer_set_callback_full"
|
|
shadows="set_callback"
|
|
version="1.12">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="398">Sets the given callback function on the buffer. This function
|
|
will be called every time a segment has been written to a device.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="281"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="400">the #GstAudioRingBuffer to set the callback on</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="cb"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1"
|
|
scope="notified"
|
|
closure="1"
|
|
destroy="2">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="401">the callback to set</doc>
|
|
<type name="AudioRingBufferCallback"
|
|
c:type="GstAudioRingBufferCallback"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="402">user data passed to the callback</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="notify" transfer-ownership="none" scope="async">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="403">function to be called when @user_data is no longer needed</doc>
|
|
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_channel_positions"
|
|
c:identifier="gst_audio_ring_buffer_set_channel_positions">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="2015">Tell the ringbuffer about the device's channel positions. This must
|
|
be called in when the ringbuffer is acquired.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="325"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="2017">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="2018">the device channel positions</doc>
|
|
<array zero-terminated="0" c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_flushing"
|
|
c:identifier="gst_audio_ring_buffer_set_flushing">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="898">Set the ringbuffer to flushing mode or normal mode.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="338"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="900">the #GstAudioRingBuffer to flush</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="flushing" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="901">the new mode</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_sample"
|
|
c:identifier="gst_audio_ring_buffer_set_sample">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1258">Make sure that the next sample written to the device is
|
|
accounted for as being the @sample sample written to the
|
|
device. This value will be used in reporting the current
|
|
sample position of the ringbuffer.
|
|
|
|
This function will also clear the buffer with silence.
|
|
|
|
MT safe.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="363"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1260">the #GstAudioRingBuffer to use</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1261">the sample number to set</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_timestamp"
|
|
c:identifier="gst_audio_ring_buffer_set_timestamp">
|
|
<source-position filename="gstaudioringbuffer.h" line="386"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="readseg" transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="timestamp" transfer-ownership="none">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="start" c:identifier="gst_audio_ring_buffer_start">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="947">Start processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="346"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="953">TRUE if the device could be started, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="949">the #GstAudioRingBuffer to start</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="stop" c:identifier="gst_audio_ring_buffer_stop">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1123">Stop processing samples from the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="352"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1129">TRUE if the device could be stopped, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1125">the #GstAudioRingBuffer to stop</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<field name="object">
|
|
<type name="Gst.Object" c:type="GstObject"/>
|
|
</field>
|
|
<field name="cond">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="166">used to signal start/stop/pause/resume actions</doc>
|
|
<type name="GLib.Cond" c:type="GCond"/>
|
|
</field>
|
|
<field name="open">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="167">boolean indicating that the ringbuffer is open</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="acquired">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="168">boolean indicating that the ringbuffer is acquired</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="memory">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="169">data in the ringbuffer</doc>
|
|
<type name="guint8" c:type="guint8*"/>
|
|
</field>
|
|
<field name="size">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="170">size of data in the ringbuffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</field>
|
|
<field name="timestamps" readable="0" private="1">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</field>
|
|
<field name="spec">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="171">format and layout of the ringbuffer data</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec"/>
|
|
</field>
|
|
<field name="samples_per_seg">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="172">number of samples in one segment</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="empty_seg">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="173">pointer to memory holding one segment of silence samples</doc>
|
|
<type name="guint8" c:type="guint8*"/>
|
|
</field>
|
|
<field name="state">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="174">state of the buffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segdone">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="175">readpointer in the ringbuffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segbase">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="176">segment corresponding to segment 0 (unused)</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="waiting">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="177">is a reader or writer waiting for a free segment</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="callback" readable="0" private="1">
|
|
<type name="AudioRingBufferCallback"
|
|
c:type="GstAudioRingBufferCallback"/>
|
|
</field>
|
|
<field name="cb_data" readable="0" private="1">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</field>
|
|
<field name="need_reorder" readable="0" private="1">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="channel_reorder_map" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="64">
|
|
<type name="gint" c:type="gint"/>
|
|
</array>
|
|
</field>
|
|
<field name="flushing" readable="0" private="1">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="may_start" readable="0" private="1">
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="active" readable="0" private="1">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="cb_data_notify" readable="0" private="1">
|
|
<type name="GLib.DestroyNotify" c:type="GDestroyNotify"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="3">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<callback name="AudioRingBufferCallback"
|
|
c:type="GstAudioRingBufferCallback">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="44">This function is set with gst_audio_ring_buffer_set_callback() and is
|
|
called to fill the memory at @data with @len bytes of samples.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="54"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="rbuf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="46">a #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="47">target to fill</doc>
|
|
<array length="2" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="len" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="48">amount to fill</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="user_data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1"
|
|
closure="3">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="49">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<record name="AudioRingBufferClass"
|
|
c:type="GstAudioRingBufferClass"
|
|
glib:is-gtype-struct-for="AudioRingBuffer">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="222">The vmethods that subclasses can override to implement the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="268"/>
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="224">parent class</doc>
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="open_device">
|
|
<callback name="open_device">
|
|
<source-position filename="gstaudioringbuffer.h" line="245"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="444">TRUE if the device could be opened, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="438">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="acquire">
|
|
<callback name="acquire">
|
|
<source-position filename="gstaudioringbuffer.h" line="246"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="597">TRUE if the device could be acquired, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="590">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="591">the specs of the buffer</doc>
|
|
<type name="AudioRingBufferSpec"
|
|
c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="release">
|
|
<callback name="release">
|
|
<source-position filename="gstaudioringbuffer.h" line="247"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="709">TRUE if the device could be released, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="705">the #GstAudioRingBuffer to release</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close_device">
|
|
<callback name="close_device">
|
|
<source-position filename="gstaudioringbuffer.h" line="248"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="504">TRUE if the device could be closed, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="499">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
<source-position filename="gstaudioringbuffer.h" line="250"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="953">TRUE if the device could be started, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="949">the #GstAudioRingBuffer to start</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="pause">
|
|
<callback name="pause">
|
|
<source-position filename="gstaudioringbuffer.h" line="251"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1085">TRUE if the device could be paused, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1081">the #GstAudioRingBuffer to pause</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="resume">
|
|
<callback name="resume">
|
|
<source-position filename="gstaudioringbuffer.h" line="252"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="stop">
|
|
<callback name="stop">
|
|
<source-position filename="gstaudioringbuffer.h" line="253"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1129">TRUE if the device could be stopped, FALSE on error.
|
|
|
|
MT safe.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1125">the #GstAudioRingBuffer to stop</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="delay">
|
|
<callback name="delay">
|
|
<source-position filename="gstaudioringbuffer.h" line="255"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1198">The number of samples queued in the audio device.
|
|
|
|
MT safe.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1185">the #GstAudioRingBuffer to query</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="activate">
|
|
<callback name="activate">
|
|
<source-position filename="gstaudioringbuffer.h" line="258"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="814">TRUE if the device could be activated in the requested mode,
|
|
FALSE on error.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="807">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="808">the new mode</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="commit">
|
|
<callback name="commit">
|
|
<source-position filename="gstaudioringbuffer.h" line="260"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1698">The number of samples written to the ringbuffer or -1 on error. The
|
|
number of samples written can be less than @out_samples when @buf was interrupted
|
|
with a flush or stop.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1671">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1672">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1673">the data to commit</doc>
|
|
<array length="3" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="in_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1674">the number of samples in the data to commit</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1675">the number of samples to write to the ringbuffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="accum"
|
|
direction="inout"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1676">accumulator for rate conversion.</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="clear_all">
|
|
<callback name="clear_all">
|
|
<source-position filename="gstaudioringbuffer.h" line="264"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.c"
|
|
line="1312">the #GstAudioRingBuffer to clear</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="AudioRingBufferFormatType"
|
|
glib:type-name="GstAudioRingBufferFormatType"
|
|
glib:get-type="gst_audio_ring_buffer_format_type_get_type"
|
|
c:type="GstAudioRingBufferFormatType">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="74">The format of the samples in the ringbuffer.</doc>
|
|
<member name="raw"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW"
|
|
glib:nick="raw">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="76">samples in linear or float</doc>
|
|
</member>
|
|
<member name="mu_law"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW"
|
|
glib:nick="mu-law">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="77">samples in mulaw</doc>
|
|
</member>
|
|
<member name="a_law"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW"
|
|
glib:nick="a-law">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="78">samples in alaw</doc>
|
|
</member>
|
|
<member name="ima_adpcm"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM"
|
|
glib:nick="ima-adpcm">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="79">samples in ima adpcm</doc>
|
|
</member>
|
|
<member name="mpeg"
|
|
value="4"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG"
|
|
glib:nick="mpeg">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="80">samples in mpeg audio (but not AAC) format</doc>
|
|
</member>
|
|
<member name="gsm"
|
|
value="5"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM"
|
|
glib:nick="gsm">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="81">samples in gsm format</doc>
|
|
</member>
|
|
<member name="iec958"
|
|
value="6"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958"
|
|
glib:nick="iec958">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="82">samples in IEC958 frames (e.g. AC3)</doc>
|
|
</member>
|
|
<member name="ac3"
|
|
value="7"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3"
|
|
glib:nick="ac3">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="83">samples in AC3 format</doc>
|
|
</member>
|
|
<member name="eac3"
|
|
value="8"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3"
|
|
glib:nick="eac3">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="84">samples in EAC3 format</doc>
|
|
</member>
|
|
<member name="dts"
|
|
value="9"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS"
|
|
glib:nick="dts">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="85">samples in DTS format</doc>
|
|
</member>
|
|
<member name="mpeg2_aac"
|
|
value="10"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC"
|
|
glib:nick="mpeg2-aac">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="86">samples in MPEG-2 AAC ADTS format</doc>
|
|
</member>
|
|
<member name="mpeg4_aac"
|
|
value="11"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC"
|
|
glib:nick="mpeg4-aac">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="87">samples in MPEG-4 AAC ADTS format</doc>
|
|
</member>
|
|
<member name="mpeg2_aac_raw"
|
|
value="12"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW"
|
|
glib:nick="mpeg2-aac-raw">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="88">samples in MPEG-2 AAC raw format (Since: 1.12)</doc>
|
|
</member>
|
|
<member name="mpeg4_aac_raw"
|
|
value="13"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW"
|
|
glib:nick="mpeg4-aac-raw">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="89">samples in MPEG-4 AAC raw format (Since: 1.12)</doc>
|
|
</member>
|
|
<member name="flac"
|
|
value="14"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC"
|
|
glib:nick="flac">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="90">samples in FLAC format (Since: 1.12)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="113">The structure containing the format specification of the ringbuffer.</doc>
|
|
<source-position filename="gstaudioringbuffer.h" line="157"/>
|
|
<field name="caps" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="115">The caps that generated the Spec.</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</field>
|
|
<field name="type" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="116">the sample type</doc>
|
|
<type name="AudioRingBufferFormatType"
|
|
c:type="GstAudioRingBufferFormatType"/>
|
|
</field>
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="117">the #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="latency_time" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="118">the latency in microseconds</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="buffer_time" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="119">the total buffer size in microseconds</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="segsize" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="120">the size of one segment in bytes</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segtotal" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="121">the total number of segments</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="seglatency" writable="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="122">number of segments queued in the lower level device,
|
|
defaults to segtotal</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<enumeration name="AudioRingBufferState"
|
|
glib:type-name="GstAudioRingBufferState"
|
|
glib:get-type="gst_audio_ring_buffer_state_get_type"
|
|
c:type="GstAudioRingBufferState">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="56">The state of the ringbuffer.</doc>
|
|
<member name="stopped"
|
|
value="0"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_STATE_STOPPED"
|
|
glib:nick="stopped">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="58">The ringbuffer is stopped</doc>
|
|
</member>
|
|
<member name="paused"
|
|
value="1"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_STATE_PAUSED"
|
|
glib:nick="paused">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="59">The ringbuffer is paused</doc>
|
|
</member>
|
|
<member name="started"
|
|
value="2"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_STATE_STARTED"
|
|
glib:nick="started">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="60">The ringbuffer is started</doc>
|
|
</member>
|
|
<member name="error"
|
|
value="3"
|
|
c:identifier="GST_AUDIO_RING_BUFFER_STATE_ERROR"
|
|
glib:nick="error">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioringbuffer.h"
|
|
line="61">The ringbuffer has encountered an
|
|
error after it has been started, e.g. because the device was
|
|
disconnected (Since: 1.2)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<class name="AudioSink"
|
|
c:symbol-prefix="audio_sink"
|
|
c:type="GstAudioSink"
|
|
parent="AudioBaseSink"
|
|
glib:type-name="GstAudioSink"
|
|
glib:get-type="gst_audio_sink_get_type"
|
|
glib:type-struct="AudioSinkClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosink.c"
|
|
line="23">This is the most simple base class for audio sinks that only requires
|
|
subclasses to implement a set of simple functions:
|
|
|
|
* `open()` :Open the device.
|
|
|
|
* `prepare()` :Configure the device with the specified format.
|
|
|
|
* `write()` :Write samples to the device.
|
|
|
|
* `reset()` :Unblock writes and flush the device.
|
|
|
|
* `delay()` :Get the number of samples written but not yet played
|
|
by the device.
|
|
|
|
* `unprepare()` :Undo operations done by prepare.
|
|
|
|
* `close()` :Close the device.
|
|
|
|
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>
|
|
<source-position filename="gstaudiosink.h" line="98"/>
|
|
<virtual-method name="close">
|
|
<source-position filename="gstaudiosink.h" line="88"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="delay">
|
|
<source-position filename="gstaudiosink.h" line="92"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
<source-position filename="gstaudiosink.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</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">
|
|
<source-position filename="gstaudiosink.h" line="84"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="reset">
|
|
<source-position filename="gstaudiosink.h" line="94"/>
|
|
<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">
|
|
<source-position filename="gstaudiosink.h" line="86"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="write">
|
|
<source-position filename="gstaudiosink.h" line="90"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<field name="element">
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink"/>
|
|
</field>
|
|
<field name="thread" readable="0" private="1">
|
|
<type name="GLib.Thread" c:type="GThread*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioSinkClass"
|
|
c:type="GstAudioSinkClass"
|
|
glib:is-gtype-struct-for="AudioSink">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosink.h"
|
|
line="60">#GstAudioSink class. Override the vmethods to implement functionality.</doc>
|
|
<source-position filename="gstaudiosink.h" line="98"/>
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosink.h"
|
|
line="62">the parent class structure.</doc>
|
|
<type name="AudioBaseSinkClass" c:type="GstAudioBaseSinkClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
<source-position filename="gstaudiosink.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="prepare">
|
|
<callback name="prepare">
|
|
<source-position filename="gstaudiosink.h" line="84"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<type name="AudioRingBufferSpec"
|
|
c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="unprepare">
|
|
<callback name="unprepare">
|
|
<source-position filename="gstaudiosink.h" line="86"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close">
|
|
<callback name="close">
|
|
<source-position filename="gstaudiosink.h" line="88"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="write">
|
|
<callback name="write">
|
|
<source-position filename="gstaudiosink.h" line="90"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="delay">
|
|
<callback name="delay">
|
|
<source-position filename="gstaudiosink.h" line="92"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="reset">
|
|
<callback name="reset">
|
|
<source-position filename="gstaudiosink.h" line="94"/>
|
|
<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">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<class name="AudioSrc"
|
|
c:symbol-prefix="audio_src"
|
|
c:type="GstAudioSrc"
|
|
parent="AudioBaseSrc"
|
|
glib:type-name="GstAudioSrc"
|
|
glib:get-type="gst_audio_src_get_type"
|
|
glib:type-struct="AudioSrcClass">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosrc.c"
|
|
line="23">This is the most simple base class for audio sources that only requires
|
|
subclasses to implement a set of simple functions:
|
|
|
|
* `open()` :Open the device.
|
|
* `prepare()` :Configure the device with the specified format.
|
|
* `read()` :Read samples from the device.
|
|
* `reset()` :Unblock reads and flush the device.
|
|
* `delay()` :Get the number of samples in the device but not yet read.
|
|
* `unprepare()` :Undo operations done by prepare.
|
|
* `close()` :Close the device.
|
|
|
|
All scheduling of samples and timestamps is done in this base class
|
|
together with #GstAudioBaseSrc using a default implementation of a
|
|
#GstAudioRingBuffer that uses threads.</doc>
|
|
<source-position filename="gstaudiosrc.h" line="97"/>
|
|
<virtual-method name="close">
|
|
<source-position filename="gstaudiosrc.h" line="86"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="delay">
|
|
<source-position filename="gstaudiosrc.h" line="91"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
<source-position filename="gstaudiosrc.h" line="80"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="prepare">
|
|
<source-position filename="gstaudiosrc.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="read">
|
|
<source-position filename="gstaudiosrc.h" line="88"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="timestamp" transfer-ownership="none">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="reset">
|
|
<source-position filename="gstaudiosrc.h" line="93"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="unprepare">
|
|
<source-position filename="gstaudiosrc.h" line="84"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<field name="element">
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc"/>
|
|
</field>
|
|
<field name="thread" readable="0" private="1">
|
|
<type name="GLib.Thread" c:type="GThread*"/>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</class>
|
|
<record name="AudioSrcClass"
|
|
c:type="GstAudioSrcClass"
|
|
glib:is-gtype-struct-for="AudioSrc">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosrc.h"
|
|
line="60">#GstAudioSrc class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
<source-position filename="gstaudiosrc.h" line="97"/>
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiosrc.h"
|
|
line="62">the parent class.</doc>
|
|
<type name="AudioBaseSrcClass" c:type="GstAudioBaseSrcClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
<source-position filename="gstaudiosrc.h" line="80"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="prepare">
|
|
<callback name="prepare">
|
|
<source-position filename="gstaudiosrc.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<type name="AudioRingBufferSpec"
|
|
c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="unprepare">
|
|
<callback name="unprepare">
|
|
<source-position filename="gstaudiosrc.h" line="84"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close">
|
|
<callback name="close">
|
|
<source-position filename="gstaudiosrc.h" line="86"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="read">
|
|
<callback name="read">
|
|
<source-position filename="gstaudiosrc.h" line="88"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
<parameter name="data"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="timestamp" transfer-ownership="none">
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="delay">
|
|
<callback name="delay">
|
|
<source-position filename="gstaudiosrc.h" line="91"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="reset">
|
|
<callback name="reset">
|
|
<source-position filename="gstaudiosrc.h" line="93"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<type name="AudioSrc" c:type="GstAudioSrc*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="_gst_reserved" readable="0" private="1">
|
|
<array zero-terminated="0" fixed-size="4">
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</array>
|
|
</field>
|
|
</record>
|
|
<record name="AudioStreamAlign"
|
|
c:type="GstAudioStreamAlign"
|
|
version="1.14"
|
|
glib:type-name="GstAudioStreamAlign"
|
|
glib:get-type="gst_audio_stream_align_get_type"
|
|
c:symbol-prefix="audio_stream_align">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="28">#GstAudioStreamAlign provides a helper object that helps tracking audio
|
|
stream alignment and discontinuities, and detects discontinuities if
|
|
possible.
|
|
|
|
See gst_audio_stream_align_new() for a description of its parameters and
|
|
gst_audio_stream_align_process() for the details of the processing.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="39"/>
|
|
<constructor name="new"
|
|
c:identifier="gst_audio_stream_align_new"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="60">Allocate a new #GstAudioStreamAlign with the given configuration. All
|
|
processing happens according to sample rate @rate, until
|
|
gst_audio_stream_align_set_rate() is called with a new @rate.
|
|
A negative rate can be used for reverse playback.
|
|
|
|
@alignment_threshold gives the tolerance in nanoseconds after which a
|
|
timestamp difference is considered a discontinuity. Once detected,
|
|
@discont_wait nanoseconds have to pass without going below the threshold
|
|
again until the output buffer is marked as a discontinuity. These can later
|
|
be re-configured with gst_audio_stream_align_set_alignment_threshold() and
|
|
gst_audio_stream_align_set_discont_wait().</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="45"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="78">a new #GstAudioStreamAlign. free with gst_audio_stream_align_free().</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="62">a sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="63">a alignment threshold in nanoseconds</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="discont_wait" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="64">discont wait in nanoseconds</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</constructor>
|
|
<method name="copy"
|
|
c:identifier="gst_audio_stream_align_copy"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="102">Copy a GstAudioStreamAlign structure.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="49"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="108">a new #GstAudioStreamAlign. free with gst_audio_stream_align_free.</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="104">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="free"
|
|
c:identifier="gst_audio_stream_align_free"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="125">Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new()
|
|
or gst_audio_stream_align_copy().</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="51"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="127">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_alignment_threshold"
|
|
c:identifier="gst_audio_stream_align_get_alignment_threshold"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="200">Gets the currently configured alignment threshold.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="63"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="206">The currently configured alignment threshold</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="202">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_discont_wait"
|
|
c:identifier="gst_audio_stream_align_get_discont_wait"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="236">Gets the currently configured discont wait.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="69"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="242">The currently configured discont wait</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="238">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_rate"
|
|
c:identifier="gst_audio_stream_align_get_rate"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="164">Gets the currently configured sample rate.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="57"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="170">The currently configured sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="166">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_samples_since_discont"
|
|
c:identifier="gst_audio_stream_align_get_samples_since_discont"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="290">Returns the number of samples that were processed since the last
|
|
discontinuity was detected.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="79"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="297">The number of samples processed since the last discontinuity.</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="292">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_timestamp_at_discont"
|
|
c:identifier="gst_audio_stream_align_get_timestamp_at_discont"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="271">Timestamp that was passed when a discontinuity was detected, i.e. the first
|
|
timestamp after the discontinuity.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="76"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="278">The last timestamp at when a discontinuity was detected</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="273">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="mark_discont"
|
|
c:identifier="gst_audio_stream_align_mark_discont"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="254">Marks the next buffer as discontinuous and resets timestamp tracking.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="73"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="256">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="process"
|
|
c:identifier="gst_audio_stream_align_process"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="309">Processes data with @timestamp and @n_samples, and returns the output
|
|
timestamp, duration and sample position together with a boolean to signal
|
|
whether a discontinuity was detected or not. All non-discontinuous data
|
|
will have perfect timestamps and durations.
|
|
|
|
A discontinuity is detected once the difference between the actual
|
|
timestamp and the timestamp calculated from the sample count since the last
|
|
discontinuity differs by more than the alignment threshold for a duration
|
|
longer than discont wait.
|
|
|
|
Note: In reverse playback, every buffer is considered discontinuous in the
|
|
context of buffer flags because the last sample of the previous buffer is
|
|
discontinuous with the first sample of the current one. However for this
|
|
function they are only considered discontinuous in reverse playback if the
|
|
first sample of the previous buffer is discontinuous with the last sample
|
|
of the current one.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="336">%TRUE if a discontinuity was detected, %FALSE otherwise.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="311">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="312">if this data is considered to be discontinuous</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="timestamp" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="313">a #GstClockTime of the start of the data</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="n_samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="314">number of samples to process</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="out_timestamp"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="315">output timestamp of the data</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
<parameter name="out_duration"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="316">output duration of the data</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
|
</parameter>
|
|
<parameter name="out_sample_position"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="317">output sample position of the start of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_alignment_threshold"
|
|
c:identifier="gst_audio_stream_align_set_alignment_threshold"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="182">Sets @alignment_treshold as new alignment threshold for the following processing.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="60"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="184">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="185">a new alignment threshold</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_discont_wait"
|
|
c:identifier="gst_audio_stream_align_set_discont_wait"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="218">Sets @alignment_treshold as new discont wait for the following processing.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="66"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="220">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont_wait" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="221">a new discont wait</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_rate"
|
|
c:identifier="gst_audio_stream_align_set_rate"
|
|
version="1.14">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="141">Sets @rate as new sample rate for the following processing. If the sample
|
|
rate differs this implicitely marks the next data as discontinuous.</doc>
|
|
<source-position filename="gstaudiostreamalign.h" line="54"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="143">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiostreamalign.c"
|
|
line="144">a new sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<function-macro name="CLOCK_TIME_TO_FRAMES"
|
|
c:identifier="GST_CLOCK_TIME_TO_FRAMES"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="52">Calculate frames from @clocktime and sample @rate.</doc>
|
|
<source-position filename="audio.h" line="59"/>
|
|
<parameters>
|
|
<parameter name="clocktime">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="54">clock time</doc>
|
|
</parameter>
|
|
<parameter name="rate">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="55">sampling rate</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="FRAMES_TO_CLOCK_TIME"
|
|
c:identifier="GST_FRAMES_TO_CLOCK_TIME"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="42">Calculate clocktime from sample @frames and @rate.</doc>
|
|
<source-position filename="audio.h" line="49"/>
|
|
<parameters>
|
|
<parameter name="frames">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="44">sample frames</doc>
|
|
</parameter>
|
|
<parameter name="rate">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="45">sampling rate</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="166"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_CLASS"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="167"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_CONVERT_PAD"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="112"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="113"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_PAD"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR_PAD"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="49"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_PAD_CLASS"
|
|
c:identifier="GST_IS_AUDIO_AGGREGATOR_PAD_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioaggregator.h" line="50"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SINK"
|
|
c:identifier="GST_IS_AUDIO_BASE_SINK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiobasesink.h" line="65"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SINK_CLASS"
|
|
c:identifier="GST_IS_AUDIO_BASE_SINK_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiobasesink.h" line="66"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SRC"
|
|
c:identifier="GST_IS_AUDIO_BASE_SRC"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiobasesrc.h" line="43"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SRC_CLASS"
|
|
c:identifier="GST_IS_AUDIO_BASE_SRC_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiobasesrc.h" line="44"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CD_SRC"
|
|
c:identifier="GST_IS_AUDIO_CD_SRC"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiocdsrc.h" line="35"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CD_SRC_CLASS"
|
|
c:identifier="GST_IS_AUDIO_CD_SRC_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiocdsrc.h" line="36"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CLOCK"
|
|
c:identifier="GST_IS_AUDIO_CLOCK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioclock.h" line="40"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CLOCK_CLASS"
|
|
c:identifier="GST_IS_AUDIO_CLOCK_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioclock.h" line="42"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_DECODER"
|
|
c:identifier="GST_IS_AUDIO_DECODER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="43"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_DECODER_CLASS"
|
|
c:identifier="GST_IS_AUDIO_DECODER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiodecoder.h" line="45"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_ENCODER"
|
|
c:identifier="GST_IS_AUDIO_ENCODER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="37"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_ENCODER_CLASS"
|
|
c:identifier="GST_IS_AUDIO_ENCODER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioencoder.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_FILTER"
|
|
c:identifier="GST_IS_AUDIO_FILTER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="46"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_FILTER_CLASS"
|
|
c:identifier="GST_IS_AUDIO_FILTER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiofilter.h" line="48"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_RING_BUFFER"
|
|
c:identifier="GST_IS_AUDIO_RING_BUFFER"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="37"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_RING_BUFFER_CLASS"
|
|
c:identifier="GST_IS_AUDIO_RING_BUFFER_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudioringbuffer.h" line="38"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SINK"
|
|
c:identifier="GST_IS_AUDIO_SINK"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosink.h" line="39"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SINK_CLASS"
|
|
c:identifier="GST_IS_AUDIO_SINK_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosink.h" line="40"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SRC"
|
|
c:identifier="GST_IS_AUDIO_SRC"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosrc.h" line="39"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SRC_CLASS"
|
|
c:identifier="GST_IS_AUDIO_SRC_CLASS"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiosrc.h" line="40"/>
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_STREAM_VOLUME"
|
|
c:identifier="GST_IS_STREAM_VOLUME"
|
|
introspectable="0">
|
|
<source-position filename="streamvolume.h" line="31"/>
|
|
<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"
|
|
filename="audio.h"
|
|
line="72">This metadata stays relevant as long as channels are unchanged.</doc>
|
|
<source-position filename="audio.h" line="79"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="META_TAG_AUDIO_RATE_STR"
|
|
value="rate"
|
|
c:type="GST_META_TAG_AUDIO_RATE_STR"
|
|
version="1.8">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="81">This metadata stays relevant as long as sample rate is unchanged.</doc>
|
|
<source-position filename="audio.h" line="88"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="META_TAG_AUDIO_STR"
|
|
value="audio"
|
|
c:type="GST_META_TAG_AUDIO_STR"
|
|
version="1.2">
|
|
<doc xml:space="preserve"
|
|
filename="audio.h"
|
|
line="64">This metadata is relevant for audio streams.</doc>
|
|
<source-position filename="audio.h" line="71"/>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="STREAM_VOLUME"
|
|
c:identifier="GST_STREAM_VOLUME"
|
|
introspectable="0">
|
|
<source-position filename="streamvolume.h" line="29"/>
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="STREAM_VOLUME_GET_INTERFACE"
|
|
c:identifier="GST_STREAM_VOLUME_GET_INTERFACE"
|
|
introspectable="0">
|
|
<source-position filename="streamvolume.h" line="33"/>
|
|
<parameters>
|
|
<parameter name="inst">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<interface name="StreamVolume"
|
|
c:symbol-prefix="stream_volume"
|
|
c:type="GstStreamVolume"
|
|
glib:type-name="GstStreamVolume"
|
|
glib:get-type="gst_stream_volume_get_type"
|
|
glib:type-struct="StreamVolumeInterface">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="20">This interface is implemented by elements that provide a stream volume. Examples for
|
|
such elements are #volume and #playbin.
|
|
|
|
Applications can use this interface to get or set the current stream volume. For this
|
|
the "volume" #GObject property can be used or the helper functions gst_stream_volume_set_volume()
|
|
and gst_stream_volume_get_volume(). This volume is always a linear factor, i.e. 0.0 is muted
|
|
1.0 is 100%. For showing the volume in a GUI it might make sense to convert it to
|
|
a different format by using gst_stream_volume_convert_volume(). Volume sliders should usually
|
|
use a cubic volume.
|
|
|
|
Separate from the volume the stream can also be muted by the "mute" #GObject property or
|
|
gst_stream_volume_set_mute() and gst_stream_volume_get_mute().
|
|
|
|
Elements that provide some kind of stream volume should implement the "volume" and
|
|
"mute" #GObject properties and handle setting and getting of them properly.
|
|
The volume property is defined to be a linear volume factor.</doc>
|
|
<source-position filename="streamvolume.h" line="42"/>
|
|
<function name="convert_volume"
|
|
c:identifier="gst_stream_volume_convert_volume">
|
|
<source-position filename="streamvolume.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="170">the converted volume</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="166">#GstStreamVolumeFormat to convert from</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="167">#GstStreamVolumeFormat to convert to</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="168">Volume in @from format that should be converted</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<method name="get_mute" c:identifier="gst_stream_volume_get_mute">
|
|
<source-position filename="streamvolume.h" line="79"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="138">Returns %TRUE if the stream is muted</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="volume" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="136">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_volume" c:identifier="gst_stream_volume_get_volume">
|
|
<source-position filename="streamvolume.h" line="71"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="97">The current stream volume as linear factor</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="volume" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="94">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="95">#GstStreamVolumeFormat which should be returned</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_mute" c:identifier="gst_stream_volume_set_mute">
|
|
<source-position filename="streamvolume.h" line="75"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="volume" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="153">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="mute" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="154">Mute state that should be set</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_volume" c:identifier="gst_stream_volume_set_volume">
|
|
<source-position filename="streamvolume.h" line="66"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="volume" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="117">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="118">#GstStreamVolumeFormat of @val</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="119">Linear volume factor that should be set</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="mute" writable="1" transfer-ownership="none">
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</property>
|
|
<property name="volume" writable="1" transfer-ownership="none">
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</property>
|
|
</interface>
|
|
<enumeration name="StreamVolumeFormat" c:type="GstStreamVolumeFormat">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.h"
|
|
line="44">Different representations of a stream volume. gst_stream_volume_convert_volume()
|
|
allows to convert between the different representations.
|
|
|
|
Formulas to convert from a linear to a cubic or dB volume are
|
|
cbrt(val) and 20 * log10 (val).</doc>
|
|
<source-position filename="streamvolume.h" line="60"/>
|
|
<member name="linear"
|
|
value="0"
|
|
c:identifier="GST_STREAM_VOLUME_FORMAT_LINEAR">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.h"
|
|
line="46">Linear scale factor, 1.0 = 100%</doc>
|
|
</member>
|
|
<member name="cubic"
|
|
value="1"
|
|
c:identifier="GST_STREAM_VOLUME_FORMAT_CUBIC">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.h"
|
|
line="47">Cubic volume scale</doc>
|
|
</member>
|
|
<member name="db" value="2" c:identifier="GST_STREAM_VOLUME_FORMAT_DB">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.h"
|
|
line="48">Logarithmic volume scale (dB, amplitude not power)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="StreamVolumeInterface"
|
|
c:type="GstStreamVolumeInterface"
|
|
glib:is-gtype-struct-for="StreamVolume">
|
|
<source-position filename="streamvolume.h" line="42"/>
|
|
<field name="iface">
|
|
<type name="GObject.TypeInterface" c:type="GTypeInterface"/>
|
|
</field>
|
|
</record>
|
|
<function name="audio_buffer_clip"
|
|
c:identifier="gst_audio_buffer_clip"
|
|
moved-to="AudioBuffer.clip">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="58">Clip the buffer to the given %GstSegment.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
<source-position filename="audio.h" line="96"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="72">%NULL if the buffer is completely outside the configured segment,
|
|
otherwise the clipped buffer is returned.
|
|
|
|
If the buffer has no timestamp, it is assumed to be inside the segment and
|
|
is not clipped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="60">The buffer to clip.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="61">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
|
|
the buffer should be clipped.</doc>
|
|
<type name="Gst.Segment" c:type="const GstSegment*"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="63">sample rate.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="64">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_buffer_reorder_channels"
|
|
c:identifier="gst_audio_buffer_reorder_channels"
|
|
moved-to="AudioBuffer.reorder_channels">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="278">Reorders @buffer from the channel positions @from to the channel
|
|
positions @to. @from and @to must contain the same number of
|
|
positions and the same positions, only in a different order.
|
|
@buffer must be writable.</doc>
|
|
<source-position filename="audio-channels.h" line="135"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="291">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="280">The buffer to reorder.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="281">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="282">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="283">The channel positions in the buffer.</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="284">The channel positions to convert to.</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_buffer_truncate"
|
|
c:identifier="gst_audio_buffer_truncate"
|
|
moved-to="AudioBuffer.truncate"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="247">Truncate the buffer to finally have @samples number of samples, removing
|
|
the necessary amount of samples from the end and @trim number of samples
|
|
from the beginning.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
<source-position filename="audio.h" line="101"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="264">the truncated buffer or %NULL if the arguments
|
|
were invalid</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="249">The buffer to truncate.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="250">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="trim" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="252">the number of samples to remove from the beginning of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio.c"
|
|
line="253">the final number of samples that should exist in this buffer or -1
|
|
to use all the remaining samples if you are only removing samples from the
|
|
beginning.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_get_fallback_mask"
|
|
c:identifier="gst_audio_channel_get_fallback_mask"
|
|
version="1.8">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="568">Get the fallback channel-mask for the given number of channels.
|
|
|
|
This function returns a reasonable fallback channel-mask and should be
|
|
called as a last resort when the specific channel map is unknown.</doc>
|
|
<source-position filename="audio-channels.h" line="172"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="577">a fallback channel-mask for @channels or 0 when there is no
|
|
mask and mono.</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="570">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_mixer_new"
|
|
c:identifier="gst_audio_channel_mixer_new"
|
|
moved-to="AudioChannelMixer.new"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="973">Create a new channel mixer object for the given parameters.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="50"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="983">a new #GstAudioChannelMixer object, or %NULL if @format isn't supported.
|
|
Free with gst_audio_channel_mixer_free() after usage.</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="975">#GstAudioChannelMixerFlags</doc>
|
|
<type name="AudioChannelMixerFlags"
|
|
c:type="GstAudioChannelMixerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="in_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="976">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="977">positions of input channels</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="978">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="979">positions of output channels</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_mixer_new_with_matrix"
|
|
c:identifier="gst_audio_channel_mixer_new_with_matrix"
|
|
moved-to="AudioChannelMixer.new_with_matrix"
|
|
version="1.14"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="809">Create a new channel mixer object for the given parameters.</doc>
|
|
<source-position filename="audio-channel-mixer.h" line="58"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="820">a new #GstAudioChannelMixer object, or %NULL if @format isn't supported,
|
|
@matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
|
|
Free with gst_audio_channel_mixer_free() after usage.</doc>
|
|
<type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="811">#GstAudioChannelMixerFlags</doc>
|
|
<type name="AudioChannelMixerFlags"
|
|
c:type="GstAudioChannelMixerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="in_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="812">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="813">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="matrix"
|
|
transfer-ownership="full"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channel-mixer.c"
|
|
line="814">channel conversion matrix, m[@in_channels][@out_channels].
|
|
If identity matrix, passthrough applies. If %NULL, a (potentially truncated)
|
|
identity matrix is generated.</doc>
|
|
<type name="gfloat" c:type="gfloat**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_positions_from_mask"
|
|
c:identifier="gst_audio_channel_positions_from_mask">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="367">Convert the @channels present in @channel_mask to a @position array
|
|
(which should have at least @channels entries ensured by caller).
|
|
If @channel_mask is set to 0, it is considered as 'not present' for purpose
|
|
of conversion.
|
|
A partially valid @channel_mask with less bits set than the number
|
|
of channels is considered valid.</doc>
|
|
<source-position filename="audio-channels.h" line="162"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="381">%TRUE if channel and channel mask are valid and could be converted</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="369">The number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="channel_mask" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="370">The input channel_mask</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="371">The
|
|
%GstAudioChannelPosition<!-- -->s</doc>
|
|
<array length="0"
|
|
zero-terminated="0"
|
|
c:type="GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_positions_to_mask"
|
|
c:identifier="gst_audio_channel_positions_to_mask">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="345">Convert the @position array of @channels channels to a bitmask.
|
|
|
|
If @force_order is %TRUE it additionally checks if the channels are
|
|
in the order required by GStreamer.</doc>
|
|
<source-position filename="audio-channels.h" line="157"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="357">%TRUE if the channel positions are valid and could be converted.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="347">The %GstAudioChannelPositions</doc>
|
|
<array length="1"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="348">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="force_order" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="349">Only consider the GStreamer channel order.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="channel_mask"
|
|
direction="out"
|
|
caller-allocates="0"
|
|
transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="350">the output channel mask</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_positions_to_string"
|
|
c:identifier="gst_audio_channel_positions_to_string"
|
|
version="1.10">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="666">Converts @position to a human-readable string representation for
|
|
debugging purposes.</doc>
|
|
<source-position filename="audio-channels.h" line="175"/>
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="675">a newly allocated string representing
|
|
@position</doc>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="668">The %GstAudioChannelPositions
|
|
to convert.</doc>
|
|
<array length="1"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="670">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_positions_to_valid_order"
|
|
c:identifier="gst_audio_channel_positions_to_valid_order">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="491">Reorders the channel positions in @position from any order to
|
|
the GStreamer channel order.</doc>
|
|
<source-position filename="audio-channels.h" line="149"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="500">%TRUE if the channel positions are valid and reordering
|
|
was successful.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="493">The channel positions to
|
|
reorder to.</doc>
|
|
<array length="1"
|
|
zero-terminated="0"
|
|
c:type="GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="495">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_check_valid_channel_positions"
|
|
c:identifier="gst_audio_check_valid_channel_positions">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="325">Checks if @position contains valid channel positions for
|
|
@channels channels. If @force_order is %TRUE it additionally
|
|
checks if the channels are in the order required by GStreamer.</doc>
|
|
<source-position filename="audio-channels.h" line="153"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="336">%TRUE if the channel positions are valid.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="327">The %GstAudioChannelPositions
|
|
to check.</doc>
|
|
<array length="1"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="329">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="force_order" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="330">Only consider the GStreamer channel order.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_clipping_meta_api_get_type"
|
|
c:identifier="gst_audio_clipping_meta_api_get_type">
|
|
<source-position filename="gstaudiometa.h" line="115"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="GType" c:type="GType"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_clipping_meta_get_info"
|
|
c:identifier="gst_audio_clipping_meta_get_info"
|
|
moved-to="AudioClippingMeta.get_info">
|
|
<source-position filename="gstaudiometa.h" line="118"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_downmix_meta_api_get_type"
|
|
c:identifier="gst_audio_downmix_meta_api_get_type">
|
|
<source-position filename="gstaudiometa.h" line="60"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="GType" c:type="GType"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_downmix_meta_get_info"
|
|
c:identifier="gst_audio_downmix_meta_get_info"
|
|
moved-to="AudioDownmixMeta.get_info">
|
|
<source-position filename="gstaudiometa.h" line="63"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_format_build_integer"
|
|
c:identifier="gst_audio_format_build_integer"
|
|
moved-to="AudioFormat.build_integer">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="358">Construct a #GstAudioFormat with given parameters.</doc>
|
|
<source-position filename="audio-format.h" line="275"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="367">a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
|
|
exists with the given parameters.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sign" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="360">signed or unsigned format</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="361">G_LITTLE_ENDIAN or G_BIG_ENDIAN</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="width" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="362">amount of bits used per sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="depth" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="363">amount of used bits in @width</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_format_fill_silence"
|
|
c:identifier="gst_audio_format_fill_silence"
|
|
moved-to="AudioFormat.fill_silence">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="454">Fill @length bytes in @dest with silence samples for @info.</doc>
|
|
<source-position filename="audio-format.h" line="289"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="456">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="457">a destination
|
|
to fill</doc>
|
|
<array length="2" zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="length" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="459">the length to fill</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_format_from_string"
|
|
c:identifier="gst_audio_format_from_string"
|
|
moved-to="AudioFormat.from_string">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="404">Convert the @format string to its #GstAudioFormat.</doc>
|
|
<source-position filename="audio-format.h" line="279"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="410">the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
|
|
string is not a known format.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="406">a format string</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_format_get_info"
|
|
c:identifier="gst_audio_format_get_info"
|
|
moved-to="AudioFormat.get_info">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="438">Get the #GstAudioFormatInfo for @format</doc>
|
|
<source-position filename="audio-format.h" line="286"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="444">The #GstAudioFormatInfo for @format.</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-format.c"
|
|
line="440">a #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_format_info_get_type"
|
|
c:identifier="gst_audio_format_info_get_type">
|
|
<source-position filename="audio-format.h" line="257"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="GType" c:type="GType"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_format_to_string"
|
|
c:identifier="gst_audio_format_to_string"
|
|
moved-to="AudioFormat.to_string">
|
|
<source-position filename="audio-format.h" line="282"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_get_channel_reorder_map"
|
|
c:identifier="gst_audio_get_channel_reorder_map">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="429">Returns a reorder map for @from to @to that can be used in
|
|
custom channel reordering code, e.g. to convert from or to the
|
|
GStreamer channel order. @from and @to must contain the same
|
|
number of positions and the same positions, only in a
|
|
different order.
|
|
|
|
The resulting @reorder_map can be used for reordering by assigning
|
|
channel i of the input to channel reorder_map[i] of the output.</doc>
|
|
<source-position filename="audio-channels.h" line="166"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="445">%TRUE if the channel positions are valid and reordering
|
|
is possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="431">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="432">The channel positions to reorder from.</doc>
|
|
<array length="0"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="433">The channel positions to reorder to.</doc>
|
|
<array length="0"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="reorder_map" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="434">Pointer to the reorder map.</doc>
|
|
<array length="0" zero-terminated="0" c:type="gint*">
|
|
<type name="gint" c:type="gint"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_iec61937_frame_size"
|
|
c:identifier="gst_audio_iec61937_frame_size">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="65">Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
|
|
payloading type from @spec.</doc>
|
|
<source-position filename="gstaudioiec61937.h" line="28"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="72">the size or 0 if the given @type is not supported or cannot be
|
|
payloaded.</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="67">the ringbufer spec</doc>
|
|
<type name="AudioRingBufferSpec"
|
|
c:type="const GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_iec61937_payload"
|
|
c:identifier="gst_audio_iec61937_payload">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="142">Payloads @src in the form specified by IEC 61937 for the type from @spec and
|
|
stores the result in @dst. @src must contain exactly one frame of data and
|
|
the frame is not checked for errors.</doc>
|
|
<source-position filename="gstaudioiec61937.h" line="31"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="156">transfer-full: %TRUE if the payloading was successful, %FALSE
|
|
otherwise.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="144">a buffer containing the data to payload</doc>
|
|
<array length="1" zero-terminated="0" c:type="const guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="src_n" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="145">size of @src in bytes</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="dst" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="146">the destination buffer to store the
|
|
payloaded contents in. Should not overlap with @src</doc>
|
|
<array length="3" zero-terminated="0" c:type="guint8*">
|
|
<type name="guint8" c:type="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="dst_n" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="148">size of @dst in bytes</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="149">the ringbufer spec for @src</doc>
|
|
<type name="AudioRingBufferSpec"
|
|
c:type="const GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudioiec61937.c"
|
|
line="150">the expected byte order of the payloaded data</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_meta_api_get_type"
|
|
c:identifier="gst_audio_meta_api_get_type">
|
|
<source-position filename="gstaudiometa.h" line="188"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="GType" c:type="GType"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_meta_get_info"
|
|
c:identifier="gst_audio_meta_get_info"
|
|
moved-to="AudioMeta.get_info">
|
|
<source-position filename="gstaudiometa.h" line="191"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_quantize_new"
|
|
c:identifier="gst_audio_quantize_new"
|
|
moved-to="AudioQuantize.new"
|
|
introspectable="0">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="413">Create a new quantizer object with the given parameters.
|
|
|
|
Output samples will be quantized to a multiple of @quantizer. Better
|
|
performance is achieved when @quantizer is a power of 2.
|
|
|
|
Dithering and noise-shaping can be performed during quantization with
|
|
the @dither and @ns parameters.</doc>
|
|
<source-position filename="audio-quantize.h" line="85"/>
|
|
<return-value>
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="430">a new #GstAudioQuantize. Free with gst_audio_quantize_free().</doc>
|
|
<type name="AudioQuantize" c:type="GstAudioQuantize*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dither" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="415">a #GstAudioDitherMethod</doc>
|
|
<type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/>
|
|
</parameter>
|
|
<parameter name="ns" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="416">a #GstAudioNoiseShapingMethod</doc>
|
|
<type name="AudioNoiseShapingMethod"
|
|
c:type="GstAudioNoiseShapingMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="417">#GstAudioQuantizeFlags</doc>
|
|
<type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="418">the #GstAudioFormat of the samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="419">the amount of channels in the samples</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="quantizer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-quantize.c"
|
|
line="420">the quantizer to use</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_reorder_channels"
|
|
c:identifier="gst_audio_reorder_channels">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="185">Reorders @data from the channel positions @from to the channel
|
|
positions @to. @from and @to must contain the same number of
|
|
positions and the same positions, only in a different order.
|
|
|
|
Note: this function assumes the audio data is in interleaved layout</doc>
|
|
<source-position filename="audio-channels.h" line="142"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="201">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="187">The pointer to
|
|
the memory.</doc>
|
|
<array length="1" zero-terminated="0" c:type="gpointer">
|
|
<type name="guint8"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="size" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="189">The size of the memory.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="190">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="191">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="192">The channel positions in the buffer.</doc>
|
|
<array length="3"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-channels.c"
|
|
line="193">The channel positions to convert to.</doc>
|
|
<array length="3"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_resampler_new"
|
|
c:identifier="gst_audio_resampler_new"
|
|
moved-to="AudioResampler.new">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1333">Make a new resampler.</doc>
|
|
<source-position filename="audio-resampler.h" line="211"/>
|
|
<return-value transfer-ownership="full" skip="1">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1345">The new #GstAudioResampler, or
|
|
%NULL on failure.</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1335">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1336">#GstAudioResamplerFlags</doc>
|
|
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1337">the #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1338">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1339">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1340">output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1341">extra options</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_resampler_options_set_quality"
|
|
c:identifier="gst_audio_resampler_options_set_quality"
|
|
moved-to="AudioResampler.options_set_quality">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1269">Set the parameters for resampling from @in_rate to @out_rate using @method
|
|
for @quality in @options.</doc>
|
|
<source-position filename="audio-resampler.h" line="205"/>
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1271">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="quality" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1272">the quality</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1273">the input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1274">the output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="audio-resampler.c"
|
|
line="1275">a #GstStructure</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="buffer_add_audio_clipping_meta"
|
|
c:identifier="gst_buffer_add_audio_clipping_meta"
|
|
version="1.8">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="248">Attaches #GstAudioClippingMeta metadata to @buffer with the given parameters.</doc>
|
|
<source-position filename="gstaudiometa.h" line="123"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="257">the #GstAudioClippingMeta on @buffer.</doc>
|
|
<type name="AudioClippingMeta" c:type="GstAudioClippingMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="250">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="251">GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="start" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="252">Amount of audio to clip from start of buffer</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
<parameter name="end" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="253">Amount of to clip from end of buffer</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="buffer_add_audio_downmix_meta"
|
|
c:identifier="gst_buffer_add_audio_downmix_meta">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="116">Attaches #GstAudioDownmixMeta metadata to @buffer with the given parameters.
|
|
|
|
@matrix is an two-dimensional array of @to_channels times @from_channels
|
|
coefficients, i.e. the i-th output channels is constructed by multiplicating
|
|
the input channels with the coefficients in @matrix[i] and taking the sum
|
|
of the results.</doc>
|
|
<source-position filename="gstaudiometa.h" line="72"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="134">the #GstAudioDownmixMeta on @buffer.</doc>
|
|
<type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="118">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="from_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="119">the channel positions
|
|
of the source</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="from_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="121">The number of channels of the source</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="to_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="122">the channel positions of
|
|
the destination</doc>
|
|
<array length="4"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="124">The number of channels of the destination</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="matrix" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="125">The matrix coefficients.</doc>
|
|
<type name="gfloat" c:type="const gfloat**"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="buffer_add_audio_meta"
|
|
c:identifier="gst_buffer_add_audio_meta"
|
|
version="1.16">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="354">Allocates and attaches a #GstAudioMeta on @buffer, which must be writable
|
|
for that purpose. The fields of the #GstAudioMeta are directly populated
|
|
from the arguments of this function.
|
|
|
|
When @info->layout is %GST_AUDIO_LAYOUT_NON_INTERLEAVED and @offsets is
|
|
%NULL, the offsets are calculated with a formula that assumes the planes are
|
|
tightly packed and in sequence:
|
|
offsets[channel] = channel * @samples * sample_stride
|
|
|
|
It is not allowed for channels to overlap in memory,
|
|
i.e. for each i in [0, channels), the range
|
|
[@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap
|
|
with any other such range. This function will assert if the parameters
|
|
specified cause this restriction to be violated.
|
|
|
|
It is, obviously, also not allowed to specify parameters that would cause
|
|
out-of-bounds memory access on @buffer. This is also checked, which means
|
|
that you must add enough memory on the @buffer before adding this meta.</doc>
|
|
<source-position filename="gstaudiometa.h" line="197"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="382">the #GstAudioMeta that was attached on the @buffer</doc>
|
|
<type name="AudioMeta" c:type="GstAudioMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="356">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="357">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="358">the number of valid samples in the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="offsets"
|
|
transfer-ownership="none"
|
|
nullable="1"
|
|
allow-none="1">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="359">the offsets (in bytes) where each channel plane starts
|
|
in the buffer or %NULL to calculate it (see below); must be %NULL also
|
|
when @info->layout is %GST_AUDIO_LAYOUT_INTERLEAVED</doc>
|
|
<type name="gsize" c:type="gsize*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function-macro name="buffer_get_audio_clipping_meta"
|
|
c:identifier="gst_buffer_get_audio_clipping_meta"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiometa.h" line="120"/>
|
|
<parameters>
|
|
<parameter name="b">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="buffer_get_audio_downmix_meta"
|
|
c:identifier="gst_buffer_get_audio_downmix_meta"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiometa.h" line="65"/>
|
|
<parameters>
|
|
<parameter name="b">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function name="buffer_get_audio_downmix_meta_for_channels"
|
|
c:identifier="gst_buffer_get_audio_downmix_meta_for_channels">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="84">Find the #GstAudioDownmixMeta on @buffer for the given destination
|
|
channel positions.</doc>
|
|
<source-position filename="gstaudiometa.h" line="67"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="94">the #GstAudioDownmixMeta on @buffer.</doc>
|
|
<type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="86">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="to_position" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="87">the channel positions of
|
|
the destination</doc>
|
|
<array length="2"
|
|
zero-terminated="0"
|
|
c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition"
|
|
c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="gstaudiometa.c"
|
|
line="89">The number of channels of the destination</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function-macro name="buffer_get_audio_meta"
|
|
c:identifier="gst_buffer_get_audio_meta"
|
|
introspectable="0">
|
|
<source-position filename="gstaudiometa.h" line="193"/>
|
|
<parameters>
|
|
<parameter name="b">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function name="stream_volume_convert_volume"
|
|
c:identifier="gst_stream_volume_convert_volume"
|
|
moved-to="StreamVolume.convert_volume">
|
|
<source-position filename="streamvolume.h" line="82"/>
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="170">the converted volume</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="166">#GstStreamVolumeFormat to convert from</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="167">#GstStreamVolumeFormat to convert to</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve"
|
|
filename="streamvolume.c"
|
|
line="168">Volume in @from format that should be converted</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|