forked from mirrors/gstreamer-rs
10671 lines
478 KiB
XML
10671 lines
478 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 xmlns="http://www.gtk.org/introspection/core/1.0" xmlns:c="http://www.gtk.org/introspection/c/1.0" xmlns:glib="http://www.gtk.org/introspection/glib/1.0" version="1.2">
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<include name="GLib" version="2.0"/>
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<include name="GModule" version="2.0"/>
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<include name="GObject" version="2.0"/>
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<include name="Gst" version="1.0"/>
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<include name="GstBase" version="1.0"/>
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<package name="gstreamer-audio-1.0"/>
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<c:include name="gst/audio/audio.h"/>
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<namespace name="GstAudio" version="1.0" shared-library="libgstaudio-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst">
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<function-macro name="AUDIO_AGGREGATOR" c:identifier="GST_AUDIO_AGGREGATOR" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_GET_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_PAD" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_PAD_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_AGGREGATOR_PAD_GET_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SINK" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SINK_CAST" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SINK_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SINK_CLOCK" introspectable="0">
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<doc xml:space="preserve">Get the #GstClock of @obj.</doc>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve">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" c:identifier="GST_AUDIO_BASE_SINK_GET_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SINK_PAD" introspectable="0">
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<doc xml:space="preserve">Get the sink #GstPad of @obj.</doc>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve">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" c:identifier="GST_AUDIO_BASE_SRC" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SRC_CAST" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SRC_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SRC_CLOCK" introspectable="0">
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<doc xml:space="preserve">Get the #GstClock of @obj.</doc>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve">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" c:identifier="GST_AUDIO_BASE_SRC_GET_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_BASE_SRC_PAD" introspectable="0">
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<doc xml:space="preserve">Get the source #GstPad of @obj.</doc>
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<parameters>
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<parameter name="obj">
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<doc xml:space="preserve">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" c:identifier="GST_AUDIO_BUFFER_BPF" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_BPS" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_CHANNELS" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_DEPTH" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_FORMAT" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_LAYOUT" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_N_PLANES" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_N_SAMPLES" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_PLANE_DATA" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_PLANE_SIZE" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_RATE" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_SAMPLE_STRIDE" introspectable="0">
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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" c:identifier="GST_AUDIO_BUFFER_WIDTH" introspectable="0">
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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" c:identifier="GST_AUDIO_CAPS_MAKE" introspectable="0">
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<doc xml:space="preserve">Generic caps string for audio, for use in pad templates.</doc>
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<parameters>
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<parameter name="format">
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<doc xml:space="preserve">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" c:identifier="GST_AUDIO_CD_SRC" introspectable="0">
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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" c:identifier="GST_AUDIO_CD_SRC_CLASS" introspectable="0">
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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" c:identifier="GST_AUDIO_CD_SRC_GET_CLASS" introspectable="0">
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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" value="(int) [ 1, max ]" c:type="GST_AUDIO_CHANNELS_RANGE">
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<doc xml:space="preserve">Maximum range of allowed channels, for use in template caps strings.</doc>
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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" c:identifier="GST_AUDIO_CHANNEL_POSITION_MASK" introspectable="0">
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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" c:identifier="GST_AUDIO_CLOCK" introspectable="0">
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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_CLOCK_CAST" c:identifier="GST_AUDIO_CLOCK_CAST" introspectable="0">
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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_CLOCK_CLASS" c:identifier="GST_AUDIO_CLOCK_CLASS" introspectable="0">
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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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<constant name="AUDIO_CONVERTER_OPT_DITHER_METHOD" value="GstAudioConverter.dither-method" c:type="GST_AUDIO_CONVERTER_OPT_DITHER_METHOD">
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<doc xml:space="preserve">#GstAudioDitherMethod, 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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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_MIX_MATRIX" value="GstAudioConverter.mix-matrix" c:type="GST_AUDIO_CONVERTER_OPT_MIX_MATRIX">
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<doc xml:space="preserve">#GST_TYPE_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 ]
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g_value_init (&v, GST_TYPE_ARRAY);
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gst_value_array_append_value (&v, &v2);
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g_value_unset (&v2);
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[ Repeat for as many v2's as your output channels - unset and reinit v2]
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g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
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g_value_unset (&v);
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]|</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD" value="GstAudioConverter.noise-shaping-method" c:type="GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD">
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<doc xml:space="preserve">#GstAudioNoiseShapingMethod, The noise shaping method to use
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to mask noise from quantization errors.
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Default is #GST_AUDIO_NOISE_SHAPING_NONE.</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_QUANTIZATION" value="GstAudioConverter.quantization" c:type="GST_AUDIO_CONVERTER_OPT_QUANTIZATION">
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<doc xml:space="preserve">#G_TYPE_UINT, The quantization amount. Components will be
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quantized to multiples of this value.
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Default is 1</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_RESAMPLER_METHOD" value="GstAudioConverter.resampler-method" c:type="GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD">
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<doc xml:space="preserve">#GstAudioResamplerMethod, The resampler method to use when
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changing sample rates.
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Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<function-macro name="AUDIO_DECODER" c:identifier="GST_AUDIO_DECODER" introspectable="0">
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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_DECODER_CAST" c:identifier="GST_AUDIO_DECODER_CAST" introspectable="0">
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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_DECODER_CLASS" c:identifier="GST_AUDIO_DECODER_CLASS" introspectable="0">
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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_DECODER_ERROR" c:identifier="GST_AUDIO_DECODER_ERROR" introspectable="0">
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<doc xml:space="preserve">Utility function that audio decoder elements can use in case they encountered
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a data processing error that may be fatal for the current "data unit" but
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need not prevent subsequent decoding. Such errors are counted and if there
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are too many, as configured in the context's max_errors, the pipeline will
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post an error message and the application will be requested to stop further
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media processing. Otherwise, it is considered a "glitch" and only a warning
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is logged. In either case, @ret is set to the proper value to
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return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).</doc>
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<parameters>
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<parameter name="el">
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<doc xml:space="preserve">the base audio decoder element that generates the error</doc>
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</parameter>
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<parameter name="weight">
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<doc xml:space="preserve">element defined weight of the error, added to error count</doc>
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</parameter>
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<parameter name="domain">
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<doc xml:space="preserve">like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)</doc>
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</parameter>
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<parameter name="code">
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<doc xml:space="preserve">error code defined for that domain (see #gstreamer-GstGError)</doc>
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</parameter>
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<parameter name="text">
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<doc xml:space="preserve">the message to display (format string and args enclosed in
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parentheses)</doc>
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</parameter>
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<parameter name="debug">
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<doc xml:space="preserve">debugging information for the message (format string and args
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enclosed in parentheses)</doc>
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</parameter>
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<parameter name="ret">
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<doc xml:space="preserve">variable to receive return value</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_DECODER_GET_CLASS" c:identifier="GST_AUDIO_DECODER_GET_CLASS" introspectable="0">
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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_DECODER_INPUT_SEGMENT" c:identifier="GST_AUDIO_DECODER_INPUT_SEGMENT" introspectable="0">
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<doc xml:space="preserve">Gives the input segment of the element.</doc>
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<parameters>
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<parameter name="obj">
|
|
<doc xml:space="preserve">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">Default maximum number of errors tolerated before signaling error.</doc>
|
|
|
|
<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">Gives the output segment of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">The name of the templates for the sink pad.</doc>
|
|
|
|
<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">Gives the pointer to the sink #GstPad object of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">The name of the templates for the source pad.</doc>
|
|
|
|
<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">Gives the pointer to the source #GstPad object of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<parameters>
|
|
<parameter name="dec">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_DECODER_STREAM_UNLOCK" c:identifier="GST_AUDIO_DECODER_STREAM_UNLOCK" introspectable="0">
|
|
|
|
<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">Standard number of channels used in consumer audio.</doc>
|
|
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_DEF_FORMAT" value="S16LE" c:type="GST_AUDIO_DEF_FORMAT">
|
|
<doc xml:space="preserve">Standard format used in consumer audio.</doc>
|
|
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_DEF_RATE" value="44100" c:type="GST_AUDIO_DEF_RATE">
|
|
<doc xml:space="preserve">Standard sampling rate used in consumer audio.</doc>
|
|
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_ENCODER" c:identifier="GST_AUDIO_ENCODER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_CAST" c:identifier="GST_AUDIO_ENCODER_CAST" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_CLASS" c:identifier="GST_AUDIO_ENCODER_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_GET_CLASS" c:identifier="GST_AUDIO_ENCODER_GET_CLASS" introspectable="0">
|
|
|
|
<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">Gives the input segment of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">Gives the output segment of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">the name of the templates for the sink pad</doc>
|
|
|
|
<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">Gives the pointer to the sink #GstPad object of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">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">the name of the templates for the source pad</doc>
|
|
|
|
<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">Gives the pointer to the source #GstPad object of the element.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
<doc xml:space="preserve">audio encoder instance</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_STREAM_LOCK" c:identifier="GST_AUDIO_ENCODER_STREAM_LOCK" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="enc">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_ENCODER_STREAM_UNLOCK" c:identifier="GST_AUDIO_ENCODER_STREAM_UNLOCK" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="enc">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER" c:identifier="GST_AUDIO_FILTER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_BPF" c:identifier="GST_AUDIO_FILTER_BPF" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_BPS" c:identifier="GST_AUDIO_FILTER_BPS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CAST" c:identifier="GST_AUDIO_FILTER_CAST" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CHANNELS" c:identifier="GST_AUDIO_FILTER_CHANNELS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CLASS" c:identifier="GST_AUDIO_FILTER_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_CLASS_CAST" c:identifier="GST_AUDIO_FILTER_CLASS_CAST" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_FORMAT" c:identifier="GST_AUDIO_FILTER_FORMAT" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_GET_CLASS" c:identifier="GST_AUDIO_FILTER_GET_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_INFO" c:identifier="GST_AUDIO_FILTER_INFO" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FILTER_RATE" c:identifier="GST_AUDIO_FILTER_RATE" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="filter">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="AUDIO_FORMATS_ALL" value="{ F64BE, F64LE, F32BE, F32LE, S32BE, S32LE, U32BE, U32LE, S24_32BE, S24_32LE, U24_32BE, U24_32LE, S24BE, S24LE, U24BE, U24LE, S20BE, S20LE, U20BE, U20LE, S18BE, S18LE, U18BE, U18LE, S16BE, S16LE, U16BE, U16LE, S8, U8 }" c:type="GST_AUDIO_FORMATS_ALL">
|
|
<doc xml:space="preserve">List of all audio formats, for use in template caps strings.
|
|
|
|
Formats are sorted by decreasing "quality", using these criteria by priority:
|
|
- depth
|
|
- width
|
|
- Float > Signed > Unsigned
|
|
- native endianness preferred</doc>
|
|
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_FORMAT_INFO_DEPTH" c:identifier="GST_AUDIO_FORMAT_INFO_DEPTH" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_ENDIANNESS" c:identifier="GST_AUDIO_FORMAT_INFO_ENDIANNESS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_FLAGS" c:identifier="GST_AUDIO_FORMAT_INFO_FLAGS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_FORMAT" c:identifier="GST_AUDIO_FORMAT_INFO_FORMAT" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_NAME" c:identifier="GST_AUDIO_FORMAT_INFO_NAME" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_FORMAT_INFO_WIDTH" c:identifier="GST_AUDIO_FORMAT_INFO_WIDTH" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_BPF" c:identifier="GST_AUDIO_INFO_BPF" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_BPS" c:identifier="GST_AUDIO_INFO_BPS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_CHANNELS" c:identifier="GST_AUDIO_INFO_CHANNELS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_DEPTH" c:identifier="GST_AUDIO_INFO_DEPTH" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_ENDIANNESS" c:identifier="GST_AUDIO_INFO_ENDIANNESS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_FLAGS" c:identifier="GST_AUDIO_INFO_FLAGS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_FORMAT" c:identifier="GST_AUDIO_INFO_FORMAT" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_FLOAT" c:identifier="GST_AUDIO_INFO_IS_FLOAT" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_INTEGER" c:identifier="GST_AUDIO_INFO_IS_INTEGER" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_SIGNED" c:identifier="GST_AUDIO_INFO_IS_SIGNED" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_UNPOSITIONED" c:identifier="GST_AUDIO_INFO_IS_UNPOSITIONED" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_IS_VALID" c:identifier="GST_AUDIO_INFO_IS_VALID" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_LAYOUT" c:identifier="GST_AUDIO_INFO_LAYOUT" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_NAME" c:identifier="GST_AUDIO_INFO_NAME" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_POSITION" c:identifier="GST_AUDIO_INFO_POSITION" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="info">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_INFO_WIDTH" c:identifier="GST_AUDIO_INFO_WIDTH" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="i">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_NE" c:identifier="GST_AUDIO_NE" introspectable="0">
|
|
<doc xml:space="preserve">Turns audio format string @s into the format string for native endianness.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="s">
|
|
<doc xml:space="preserve">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">Turns audio format string @s into the format string for other endianness.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="s">
|
|
<doc xml:space="preserve">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">Maximum range of allowed sample rates, for use in template caps strings.</doc>
|
|
|
|
<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">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>
|
|
|
|
<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">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>
|
|
|
|
<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">G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.</doc>
|
|
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION" value="GstAudioResampler.filter-interpolation" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION">
|
|
<doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be
|
|
interpolated.
|
|
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc>
|
|
|
|
<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">GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
|
|
constructed.
|
|
GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.</doc>
|
|
|
|
<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">G_TYPE_UINT: the amount of memory to use for full filter tables before
|
|
switching to interpolated filter tables.
|
|
1048576 is the default.</doc>
|
|
|
|
<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">G_TYPE_UINT, oversampling to use when interpolating filters
|
|
8 is the default.</doc>
|
|
|
|
<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">G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
|
|
rates.
|
|
0.1 is the default.</doc>
|
|
|
|
<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">G_TYPE_INT: the number of taps to use for the filter.
|
|
0 is the default and selects the taps automatically.</doc>
|
|
|
|
<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">G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
|
|
after the stopband for the kaiser window. 85 dB is the default.</doc>
|
|
|
|
<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">G_TYPE_DOUBLE, transition bandwidth. The width of the
|
|
transition band for the kaiser window. 0.087 is the default.</doc>
|
|
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_DEFAULT" value="4" c:type="GST_AUDIO_RESAMPLER_QUALITY_DEFAULT">
|
|
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_MAX" value="10" c:type="GST_AUDIO_RESAMPLER_QUALITY_MAX">
|
|
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<constant name="AUDIO_RESAMPLER_QUALITY_MIN" value="0" c:type="GST_AUDIO_RESAMPLER_QUALITY_MIN">
|
|
|
|
<type name="gint" c:type="gint"/>
|
|
</constant>
|
|
<function-macro name="AUDIO_RING_BUFFER" c:identifier="GST_AUDIO_RING_BUFFER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_BROADCAST" c:identifier="GST_AUDIO_RING_BUFFER_BROADCAST" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_CAST" c:identifier="GST_AUDIO_RING_BUFFER_CAST" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_CLASS" c:identifier="GST_AUDIO_RING_BUFFER_CLASS" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_SIGNAL" c:identifier="GST_AUDIO_RING_BUFFER_SIGNAL" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_RING_BUFFER_WAIT" c:identifier="GST_AUDIO_RING_BUFFER_WAIT" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="buf">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK" c:identifier="GST_AUDIO_SINK" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK_CLASS" c:identifier="GST_AUDIO_SINK_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SINK_GET_CLASS" c:identifier="GST_AUDIO_SINK_GET_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC" c:identifier="GST_AUDIO_SRC" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC_CLASS" c:identifier="GST_AUDIO_SRC_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="AUDIO_SRC_GET_CLASS" c:identifier="GST_AUDIO_SRC_GET_CLASS" introspectable="0">
|
|
|
|
<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">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).
|
|
|
|
The #GstAggregator::samples-selected signal is provided with some
|
|
additional information about the output buffer:
|
|
- "offset" G_TYPE_UINT64 Offset in samples since segment start
|
|
for the position that is next to be filled in the output buffer.
|
|
- "frames" G_TYPE_UINT Number of frames per output buffer.
|
|
|
|
In addition the gst_aggregator_peek_next_sample() function returns
|
|
additional information in the info #GstStructure of the returned sample:
|
|
- "output-offset" G_TYPE_UINT64 Sample offset in output segment relative to
|
|
the output segment's start where the current position of this input
|
|
buffer would be placed
|
|
- "position" G_TYPE_UINT current position in the input buffer in samples
|
|
- "size" G_TYPE_UINT size of the input buffer in samples</doc>
|
|
|
|
<virtual-method name="aggregate_one_buffer">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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>
|
|
<property name="output-buffer-duration-fraction" version="1.18" writable="1" transfer-ownership="none">
|
|
<doc xml:space="preserve">Output block size in nanoseconds, expressed as a fraction.</doc>
|
|
<type name="Gst.Fraction"/>
|
|
</property>
|
|
<field name="parent">
|
|
<type name="GstBase.Aggregator" c:type="GstAggregator"/>
|
|
</field>
|
|
<field name="current_caps">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<field name="parent_class">
|
|
<type name="GstBase.AggregatorClass" c:type="GstAggregatorClass"/>
|
|
</field>
|
|
<field name="create_output_buffer">
|
|
<callback name="create_output_buffer">
|
|
|
|
<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">
|
|
|
|
<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">An implementation of GstPad that can be used with #GstAudioAggregator.
|
|
|
|
See #GstAudioAggregator for more details.</doc>
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
</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">The default implementation of GstPad used with #GstAudioAggregator</doc>
|
|
|
|
<virtual-method name="convert_buffer">
|
|
|
|
<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">
|
|
|
|
<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">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">
|
|
|
|
<field name="parent_class">
|
|
<type name="GstBase.AggregatorPadClass" c:type="GstAggregatorPadClass"/>
|
|
</field>
|
|
<field name="convert_buffer">
|
|
<callback name="convert_buffer">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
</record>
|
|
<record name="AudioAggregatorPrivate" c:type="GstAudioAggregatorPrivate" disguised="1">
|
|
|
|
</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">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>
|
|
|
|
<virtual-method name="create_ringbuffer" invoker="create_ringbuffer">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioBaseSink.</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="payload">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the current alignment threshold, in nanoseconds, used by @sink.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the current discont wait, in nanoseconds, used by @sink.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the current drift tolerance, in microseconds, used by @sink.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries whether @sink will provide a clock or not. See also
|
|
gst_audio_base_sink_set_provide_clock.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">Get the current slave method used by @sink.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">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">Controls the sink's alignment threshold.</doc>
|
|
|
|
<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">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">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">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">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">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">Controls how long the sink will wait before creating a discontinuity.</doc>
|
|
|
|
<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">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont_wait" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Controls the sink's drift tolerance.</doc>
|
|
|
|
<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">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="drift_tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="provide" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Controls how clock slaving will be performed in @sink.</doc>
|
|
|
|
<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">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</instance-parameter>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">#GstAudioBaseSink class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve">the parent class.</doc>
|
|
<type name="GstBase.BaseSinkClass" c:type="GstBaseSinkClass"/>
|
|
</field>
|
|
<field name="create_ringbuffer">
|
|
<callback name="create_ringbuffer">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioBaseSink.</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="payload">
|
|
<callback name="payload">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioBaseSink</doc>
|
|
<type name="AudioBaseSink" c:type="GstAudioBaseSink*"/>
|
|
</parameter>
|
|
<parameter name="etime" transfer-ownership="none">
|
|
<doc xml:space="preserve">external clock time</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="itime" transfer-ownership="none">
|
|
<doc xml:space="preserve">internal clock time</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="requested_skew" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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">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">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">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">
|
|
|
|
</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">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">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">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">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">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">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>
|
|
|
|
<virtual-method name="create_ringbuffer" invoker="create_ringbuffer">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries whether @src will provide a clock or not. See also
|
|
gst_audio_base_src_set_provide_clock.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">Get the current slave method used by @src.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="provide" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Controls how clock slaving will be performed in @src.</doc>
|
|
|
|
<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">a #GstAudioBaseSrc</doc>
|
|
<type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">#GstAudioBaseSrc class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve">the parent class.</doc>
|
|
<type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/>
|
|
</field>
|
|
<field name="create_ringbuffer">
|
|
<callback name="create_ringbuffer">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
</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">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">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">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">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">No adjustment is done.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioBuffer" c:type="GstAudioBuffer" version="1.16">
|
|
<doc xml:space="preserve">A structure containing the result of an audio buffer map operation,
|
|
which is executed with gst_audio_buffer_map(). For non-interleaved (planar)
|
|
buffers, the beginning of each channel in the buffer has its own pointer in
|
|
the @planes array. For interleaved buffers, the @planes array only contains
|
|
one item, which is the pointer to the beginning of the buffer, and @n_planes
|
|
equals 1.
|
|
|
|
The different channels in @planes are always in the GStreamer channel order.</doc>
|
|
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve">a #GstAudioInfo describing the audio properties of this buffer</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="n_samples" writable="1">
|
|
<doc xml:space="preserve">the size of the buffer in samples</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</field>
|
|
<field name="n_planes" writable="1">
|
|
<doc xml:space="preserve">the number of planes available</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="planes" writable="1">
|
|
<doc xml:space="preserve">an array of @n_planes pointers pointing to the start of each
|
|
plane in the mapped buffer</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</field>
|
|
<field name="buffer" writable="1">
|
|
<doc xml:space="preserve">the mapped buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</field>
|
|
<field name="map_infos" readable="0" private="1">
|
|
<type name="Gst.MapInfo" c:type="GstMapInfo*"/>
|
|
</field>
|
|
<field name="priv_planes_arr" readable="0" private="1">
|
|
<array zero-terminated="0" 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">Maps an audio @gstbuffer so that it can be read or written and stores the
|
|
result of the map operation in @buffer.
|
|
|
|
This is especially useful when the @gstbuffer is in non-interleaved (planar)
|
|
layout, in which case this function will use the information in the
|
|
@gstbuffer's attached #GstAudioMeta in order to map each channel in a
|
|
separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached
|
|
on the @gstbuffer, then it must be in interleaved layout.
|
|
|
|
If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked
|
|
against @info. Normally, they should be equal, but in case they are not,
|
|
a g_critical will be printed and the #GstAudioInfo from the meta will be
|
|
used.
|
|
|
|
In non-interleaved buffers, it is possible to have each channel on a separate
|
|
#GstMemory. In this case, each memory will be mapped separately to avoid
|
|
copying their contents in a larger memory area. Do note though that it is
|
|
not supported to have a single channel spanning over two or more different
|
|
#GstMemory objects. Although the map operation will likely succeed in this
|
|
case, it will be highly sub-optimal and it is recommended to merge all the
|
|
memories in the buffer before calling this function.
|
|
|
|
Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid
|
|
as long as it's mapped.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the map operation succeeded or %FALSE on failure</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">pointer to a #GstAudioBuffer</doc>
|
|
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="gstbuffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstBuffer to be mapped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">the access mode for the memory</doc>
|
|
<type name="Gst.MapFlags" c:type="GstMapFlags"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="unmap" c:identifier="gst_audio_buffer_unmap" version="1.16">
|
|
<doc xml:space="preserve">Unmaps an audio buffer that was previously mapped with
|
|
gst_audio_buffer_map().</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioBuffer to unmap</doc>
|
|
<type name="AudioBuffer" c:type="GstAudioBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<function name="clip" c:identifier="gst_audio_buffer_clip">
|
|
<doc xml:space="preserve">Clip the buffer to the given %GstSegment.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">%NULL if the buffer is completely outside the configured segment,
|
|
otherwise the clipped buffer is returned.
|
|
|
|
If the buffer has no timestamp, it is assumed to be inside the segment and
|
|
is not clipped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve">The buffer to clip.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
|
|
the buffer should be clipped.</doc>
|
|
<type name="Gst.Segment" c:type="const GstSegment*"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">sample rate.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="reorder_channels" c:identifier="gst_audio_buffer_reorder_channels">
|
|
<doc xml:space="preserve">Reorders @buffer from the channel positions @from to the channel
|
|
positions @to. @from and @to must contain the same number of
|
|
positions and the same positions, only in a different order.
|
|
@buffer must be writable.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">The buffer to reorder.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions in the buffer.</doc>
|
|
<array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions to convert to.</doc>
|
|
<array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="truncate" c:identifier="gst_audio_buffer_truncate" version="1.16">
|
|
<doc xml:space="preserve">Truncate the buffer to finally have @samples number of samples, removing
|
|
the necessary amount of samples from the end and @trim number of samples
|
|
from the beginning.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">the truncated buffer or %NULL if the arguments
|
|
were invalid</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve">The buffer to truncate.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="trim" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of samples to remove from the beginning of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">the final number of samples that should exist in this buffer or -1
|
|
to use all the remaining samples if you are only removing samples from the
|
|
beginning.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<class name="AudioCdSrc" c:symbol-prefix="audio_cd_src" c:type="GstAudioCdSrc" parent="GstBase.PushSrc" glib:type-name="GstAudioCdSrc" glib:get-type="gst_audio_cd_src_get_type" glib:type-struct="AudioCdSrcClass">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<implements name="Gst.URIHandler"/>
|
|
<virtual-method name="close">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioCdSrc</doc>
|
|
<type name="AudioCdSrc" c:type="GstAudioCdSrc*"/>
|
|
</instance-parameter>
|
|
<parameter name="track" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Audio CD source base class.</doc>
|
|
|
|
<field name="pushsrc_class">
|
|
<doc xml:space="preserve">the parent class</doc>
|
|
<type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">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">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">the entire disc is a single stream</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioCdSrcPrivate" c:type="GstAudioCdSrcPrivate" disguised="1">
|
|
|
|
</record>
|
|
<record name="AudioCdSrcTrack" c:type="GstAudioCdSrcTrack">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<field name="is_audio" writable="1">
|
|
<doc xml:space="preserve">Whether this is an audio track</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="num" writable="1">
|
|
<doc xml:space="preserve">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">The first sector of this track (LBA)</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="end" writable="1">
|
|
<doc xml:space="preserve">The last sector of this track (LBA)</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</field>
|
|
<field name="tags" writable="1">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<method name="free" c:identifier="gst_audio_channel_mixer_free">
|
|
<doc xml:space="preserve">Free memory allocated by @mix.</doc>
|
|
|
|
<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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">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>
|
|
|
|
<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">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">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">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Create a new channel mixer object for the given parameters.</doc>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">#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">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">positions of input channels</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Create a new channel mixer object for the given parameters.</doc>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">#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">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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">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">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 implicitly means that this is a stereo stream with a front left and front right
|
|
channel.</doc>
|
|
<member name="none" value="-3" c:identifier="GST_AUDIO_CHANNEL_POSITION_NONE" glib:nick="none">
|
|
<doc xml:space="preserve">used for position-less channels, e.g.
|
|
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">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">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">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">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">Front center</doc>
|
|
</member>
|
|
<member name="lfe1" value="3" c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE1" glib:nick="lfe1">
|
|
<doc xml:space="preserve">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">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">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">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">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">Rear center</doc>
|
|
</member>
|
|
<member name="lfe2" value="9" c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE2" glib:nick="lfe2">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">Surround right (between rear right and side right)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioClippingMeta" c:type="GstAudioClippingMeta" version="1.8">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="format" writable="1">
|
|
<doc xml:space="preserve">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">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">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">
|
|
|
|
<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">#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>
|
|
|
|
<constructor name="new" c:identifier="gst_audio_clock_new">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">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">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
<parameter name="destroy_notify" transfer-ownership="none" scope="async">
|
|
<doc xml:space="preserve">#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">Adjust @time with the internal offset of the audio clock.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">@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">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
<parameter name="time" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Report the time as returned by the #GstAudioClockGetTimeFunc without applying
|
|
any offsets.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">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">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>
|
|
|
|
<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">a #GstAudioClock</doc>
|
|
<type name="AudioClock" c:type="GstAudioClock*"/>
|
|
</instance-parameter>
|
|
<parameter name="time" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">user data</doc>
|
|
<type name="gpointer" c:type="gpointer"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
<record name="AudioConverter" c:type="GstAudioConverter" version="1.8" glib:type-name="GstAudioConverter" glib:get-type="gst_audio_converter_get_type" c:symbol-prefix="audio_converter">
|
|
<doc xml:space="preserve">This object is used to convert audio samples from one format to another.
|
|
The object can perform conversion of:
|
|
|
|
* audio format with optional dithering and noise shaping
|
|
|
|
* audio samplerate
|
|
|
|
* audio channels and channel layout</doc>
|
|
|
|
<constructor name="new" c:identifier="gst_audio_converter_new">
|
|
<doc xml:space="preserve">Create a new #GstAudioConverter that is able to convert between @in and @out
|
|
audio formats.
|
|
|
|
@config contains extra configuration options, see `GST_AUDIO_CONVERTER_OPT_*`
|
|
parameters for details about the options and values.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a #GstAudioConverter or %NULL if conversion is not possible.</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">extra #GstAudioConverterFlags</doc>
|
|
<type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
|
|
</parameter>
|
|
<parameter name="in_info" transfer-ownership="none">
|
|
<doc xml:space="preserve">a source #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="out_info" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">extra #GstAudioConverterFlags</doc>
|
|
<type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/>
|
|
</parameter>
|
|
<parameter name="in" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">Free a previously allocated @convert instance.</doc>
|
|
|
|
<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">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">Get the current configuration of @convert.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">
|
|
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">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">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">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">Calculate how many input frames are currently needed by @convert to produce
|
|
@out_frames of output frames.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Get the maximum number of input frames that the converter would
|
|
need before producing output.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Calculate how many output frames can be produced when @in_frames input
|
|
frames are given to @convert.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Returns whether the audio converter will operate in passthrough mode.
|
|
The return value would be typically input to gst_base_transform_set_passthrough()</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE when no conversion will actually occur.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="convert" transfer-ownership="none">
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="reset" c:identifier="gst_audio_converter_reset">
|
|
<doc xml:space="preserve">Reset @convert to the state it was when it was first created, clearing
|
|
any history it might currently have.</doc>
|
|
|
|
<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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">input frames</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">output frames</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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" version="1.12">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioConverter</doc>
|
|
<type name="AudioConverter" c:type="GstAudioConverter*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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>
|
|
|
|
<virtual-method name="close">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">Helper function that allocates a buffer to hold an audio frame
|
|
for @dec's current output format.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">allocated buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="size" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">decoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Collects decoded data and pushes it downstream. This function may be called
|
|
multiple times for a given input frame.
|
|
|
|
@buf may be NULL in which case it is assumed that the current input frame is
|
|
finished. This is equivalent to calling gst_audio_decoder_finish_subframe()
|
|
with a NULL buffer and frames=1 after having pushed out all decoded audio
|
|
subframes using this function.
|
|
|
|
When called with valid data in @buf the source pad caps must have been set
|
|
already.
|
|
|
|
Note that a frame received in #GstAudioDecoderClass.handle_frame() may be
|
|
invalidated by a call to this function.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">decoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_allocator" c:identifier="gst_audio_decoder_get_allocator">
|
|
<doc xml:space="preserve">Lets #GstAudioDecoder sub-classes to know the memory @allocator
|
|
used by the base class and its @params.
|
|
|
|
Unref the @allocator after use it.</doc>
|
|
|
|
<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">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">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">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">
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_delay" c:identifier="gst_audio_decoder_get_delay">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries decoder drain handling.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Sets the variables pointed to by @min and @max to the currently configured
|
|
latency.</doc>
|
|
|
|
<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">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">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries decoder's latency aggregation.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries decoder required format handling.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Return current parsing (sync and eos) state.</doc>
|
|
|
|
<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">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">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">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">Queries decoder packet loss concealment handling.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries current audio jitter tolerance threshold.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Returns caps that express @caps (or sink template caps if @caps == NULL)
|
|
restricted to rate/channels/... combinations supported by downstream
|
|
elements.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">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">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">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>
|
|
|
|
<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">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">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">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>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Allows baseclass to perform byte to time estimated conversion.</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Sets decoder latency.</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min" transfer-ownership="none">
|
|
<doc xml:space="preserve">minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="max" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Sets decoder minimum aggregation latency.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Configure output caps on the srcpad of @dec. Similar to
|
|
gst_audio_decoder_set_output_format(), but allows subclasses to specify
|
|
output caps that can't be expressed via #GstAudioInfo e.g. caps that have
|
|
caps features.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve">(fixed) #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_output_format" c:identifier="gst_audio_decoder_set_output_format">
|
|
<doc xml:space="preserve">Configure output info on the srcpad of @dec.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">#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">Enable or disable decoder packet loss concealment, provided subclass
|
|
and codec are capable and allow handling plc.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Indicates whether or not subclass handles packet loss concealment (plc).</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="plc" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Configures decoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="use" transfer-ownership="none">
|
|
<doc xml:space="preserve">if the default pad accept-caps query handling should be used</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<property name="max-errors" version="1.18" writable="1" transfer-ownership="none">
|
|
<doc xml:space="preserve">Maximum number of tolerated consecutive decode errors. See
|
|
gst_audio_decoder_set_max_errors() for more details.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</property>
|
|
<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">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>
|
|
|
|
<field name="element_class">
|
|
<doc xml:space="preserve">The parent class structure</doc>
|
|
<type name="Gst.ElementClass" c:type="GstElementClass"/>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="dec" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioDecoder</doc>
|
|
<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="decide_allocation">
|
|
<callback name="decide_allocation">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
</record>
|
|
<enumeration name="AudioDitherMethod" glib:type-name="GstAudioDitherMethod" glib:get-type="gst_audio_dither_method_get_type" c:type="GstAudioDitherMethod">
|
|
<doc xml:space="preserve">Set of available dithering methods.</doc>
|
|
<member name="none" value="0" c:identifier="GST_AUDIO_DITHER_NONE" glib:nick="none">
|
|
<doc xml:space="preserve">No dithering</doc>
|
|
</member>
|
|
<member name="rpdf" value="1" c:identifier="GST_AUDIO_DITHER_RPDF" glib:nick="rpdf">
|
|
<doc xml:space="preserve">Rectangular dithering</doc>
|
|
</member>
|
|
<member name="tpdf" value="2" c:identifier="GST_AUDIO_DITHER_TPDF" glib:nick="tpdf">
|
|
<doc xml:space="preserve">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">High frequency triangular dithering</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioDownmixMeta" c:type="GstAudioDownmixMeta">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="from_position" writable="1">
|
|
<doc xml:space="preserve">the channel positions of the source</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</field>
|
|
<field name="to_position" writable="1">
|
|
<doc xml:space="preserve">the channel positions of the destination</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</field>
|
|
<field name="from_channels" writable="1">
|
|
<doc xml:space="preserve">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">the number of channels of the destination</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="matrix" writable="1">
|
|
<doc xml:space="preserve">the matrix coefficients.</doc>
|
|
<type name="gfloat" c:type="gfloat**"/>
|
|
</field>
|
|
<function name="get_info" c:identifier="gst_audio_downmix_meta_get_info">
|
|
|
|
<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">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>
|
|
|
|
<implements name="Gst.Preset"/>
|
|
<virtual-method name="close">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="open">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">Helper function that allocates a buffer to hold an encoded audio frame
|
|
for @enc's current output format.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">allocated buffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="size" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstFlowReturn that should be escalated to caller (of caller)</doc>
|
|
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="buffer" transfer-ownership="full" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">encoded data</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Lets #GstAudioEncoder sub-classes to know the memory @allocator
|
|
used by the base class and its @params.
|
|
|
|
Unref the @allocator after use it.</doc>
|
|
|
|
<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">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">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">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">
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">Queries encoder drain handling.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries encoder hard minimum handling.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<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">Sets the variables pointed to by @min and @max to the currently configured
|
|
latency.</doc>
|
|
|
|
<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">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">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">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries if the encoder will handle granule marking.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries encoder perfect timestamp behaviour.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Queries current audio jitter tolerance threshold.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">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">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">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>
|
|
|
|
<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">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">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<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">Set the codec headers to be sent downstream whenever requested.</doc>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="headers" transfer-ownership="full">
|
|
<doc xml:space="preserve">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">Sets encoder latency.</doc>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="min" transfer-ownership="none">
|
|
<doc xml:space="preserve">minimum latency</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
<parameter name="max" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="num" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Enable or disable encoder granule handling.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Configure output caps on the srcpad of @enc.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE on success.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve">#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">Enable or disable encoder perfect output timestamp preference.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="enabled" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Configures encoder audio jitter tolerance threshold.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</instance-parameter>
|
|
<parameter name="tolerance" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<field name="element_class">
|
|
<doc xml:space="preserve">The parent class structure</doc>
|
|
<type name="Gst.ElementClass" c:type="GstElementClass"/>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="enc" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioEncoder</doc>
|
|
<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="decide_allocation">
|
|
<callback name="decide_allocation">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
</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">#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>
|
|
|
|
<virtual-method name="setup">
|
|
|
|
<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">In addition to the @setup virtual function, you should also override the
|
|
GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual
|
|
function.</doc>
|
|
|
|
<field name="basetransformclass">
|
|
<doc xml:space="preserve">parent class</doc>
|
|
<type name="GstBase.BaseTransformClass" c:type="GstBaseTransformClass"/>
|
|
</field>
|
|
<field name="setup">
|
|
<callback name="setup">
|
|
|
|
<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">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>
|
|
|
|
<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">an #GstAudioFilterClass</doc>
|
|
<type name="AudioFilterClass" c:type="GstAudioFilterClass*"/>
|
|
</instance-parameter>
|
|
<parameter name="allowed_caps" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Extra audio flags</doc>
|
|
<member name="none" value="0" c:identifier="GST_AUDIO_FLAG_NONE" glib:nick="none">
|
|
<doc xml:space="preserve">no valid flag</doc>
|
|
</member>
|
|
<member name="unpositioned" value="1" c:identifier="GST_AUDIO_FLAG_UNPOSITIONED" glib:nick="unpositioned">
|
|
<doc xml:space="preserve">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">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">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">encoded audio format</doc>
|
|
</member>
|
|
<member name="s8" value="2" c:identifier="GST_AUDIO_FORMAT_S8" glib:nick="s8">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">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">64-bit floating point samples, native endianness</doc>
|
|
</member>
|
|
<function name="build_integer" c:identifier="gst_audio_format_build_integer">
|
|
<doc xml:space="preserve">Construct a #GstAudioFormat with given parameters.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">signed or unsigned format</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">amount of bits used per sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="depth" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Fill @length bytes in @dest with silence samples for @info.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Convert the @format string to its #GstAudioFormat.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the #GstAudioFormatInfo for @format</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="to_string" c:identifier="gst_audio_format_to_string">
|
|
|
|
<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">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">integer samples</doc>
|
|
</member>
|
|
<member name="float" value="2" c:identifier="GST_AUDIO_FORMAT_FLAG_FLOAT" glib:nick="float">
|
|
<doc xml:space="preserve">float samples</doc>
|
|
</member>
|
|
<member name="signed" value="4" c:identifier="GST_AUDIO_FORMAT_FLAG_SIGNED" glib:nick="signed">
|
|
<doc xml:space="preserve">signed samples</doc>
|
|
</member>
|
|
<member name="complex" value="16" c:identifier="GST_AUDIO_FORMAT_FLAG_COMPLEX" glib:nick="complex">
|
|
<doc xml:space="preserve">complex layout</doc>
|
|
</member>
|
|
<member name="unpack" value="32" c:identifier="GST_AUDIO_FORMAT_FLAG_UNPACK" glib:nick="unpack">
|
|
<doc xml:space="preserve">the format can be used in
|
|
#GstAudioFormatUnpack and #GstAudioFormatPack functions</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioFormatInfo" c:type="GstAudioFormatInfo">
|
|
<doc xml:space="preserve">Information for an audio format.</doc>
|
|
|
|
<field name="format" writable="1">
|
|
<doc xml:space="preserve">#GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</field>
|
|
<field name="name" writable="1">
|
|
<doc xml:space="preserve">string representation of the format</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</field>
|
|
<field name="description" writable="1">
|
|
<doc xml:space="preserve">user readable description of the format</doc>
|
|
<type name="utf8" c:type="const gchar*"/>
|
|
</field>
|
|
<field name="flags" writable="1">
|
|
<doc xml:space="preserve">#GstAudioFormatFlags</doc>
|
|
<type name="AudioFormatFlags" c:type="GstAudioFormatFlags"/>
|
|
</field>
|
|
<field name="endianness" writable="1">
|
|
<doc xml:space="preserve">the endianness</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="width" writable="1">
|
|
<doc xml:space="preserve">amount of bits used for one sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="depth" writable="1">
|
|
<doc xml:space="preserve">amount of valid bits in @width</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="silence" writable="1">
|
|
<doc xml:space="preserve">@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">the format of the unpacked samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</field>
|
|
<field name="unpack_func" writable="1">
|
|
<doc xml:space="preserve">function to unpack samples</doc>
|
|
<type name="AudioFormatUnpack" c:type="GstAudioFormatUnpack"/>
|
|
</field>
|
|
<field name="pack_func" writable="1">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioPackFlags</doc>
|
|
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
|
|
</parameter>
|
|
<parameter name="src" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioPackFlags</doc>
|
|
<type name="AudioPackFlags" c:type="GstAudioPackFlags"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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>
|
|
|
|
<field name="finfo" writable="1">
|
|
<doc xml:space="preserve">the format info of the audio</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</field>
|
|
<field name="flags" writable="1">
|
|
<doc xml:space="preserve">additional audio flags</doc>
|
|
<type name="AudioFlags" c:type="GstAudioFlags"/>
|
|
</field>
|
|
<field name="layout" writable="1">
|
|
<doc xml:space="preserve">audio layout</doc>
|
|
<type name="AudioLayout" c:type="GstAudioLayout"/>
|
|
</field>
|
|
<field name="rate" writable="1">
|
|
<doc xml:space="preserve">the audio sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="channels" writable="1">
|
|
<doc xml:space="preserve">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="bpf" writable="1">
|
|
<doc xml:space="preserve">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">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">Allocate a new #GstAudioInfo that is also initialized with
|
|
gst_audio_info_init().</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="src_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve">#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">value to convert</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</parameter>
|
|
<parameter name="dest_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve">#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">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">Copy a GstAudioInfo structure.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">Free a GstAudioInfo structure previously allocated with gst_audio_info_new()
|
|
or gst_audio_info_copy().</doc>
|
|
|
|
<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">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">Parse @caps and update @info.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Initialize @info with default values.</doc>
|
|
|
|
<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">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">Compares two #GstAudioInfo and returns whether they are equal or not</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="other" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Set the default info for the audio info of @format and @rate and @channels.
|
|
|
|
Note: This initializes @info first, no values are preserved.</doc>
|
|
|
|
<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">a #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">the format</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">the samplerate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="position" transfer-ownership="none" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">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">Convert the values of @info into a #GstCaps.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">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">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">non-interleaved audio</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioMeta" c:type="GstAudioMeta" version="1.16">
|
|
<doc xml:space="preserve">#GstAudioDownmixMeta defines an audio downmix matrix to be send along with
|
|
audio buffers. These functions in this module help to create and attach the
|
|
meta as well as extracting it.</doc>
|
|
|
|
<field name="meta" writable="1">
|
|
<doc xml:space="preserve">parent #GstMeta</doc>
|
|
<type name="Gst.Meta" c:type="GstMeta"/>
|
|
</field>
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="samples" writable="1">
|
|
<doc xml:space="preserve">the number of valid samples in the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</field>
|
|
<field name="offsets" writable="1">
|
|
<doc xml:space="preserve">the offsets (in bytes) where each channel plane starts in the
|
|
buffer or %NULL if the buffer has interleaved layout; if not %NULL, this
|
|
is guaranteed to be an array of @info.channels elements</doc>
|
|
<type name="gsize" c:type="gsize*"/>
|
|
</field>
|
|
<field name="priv_offsets_arr" readable="0" private="1">
|
|
<array zero-terminated="0" 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">
|
|
|
|
<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">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">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">Error feedback</doc>
|
|
</member>
|
|
<member name="simple" value="2" c:identifier="GST_AUDIO_NOISE_SHAPING_SIMPLE" glib:nick="simple">
|
|
<doc xml:space="preserve">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">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">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">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">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">When the source has a smaller depth
|
|
than the target format, set the least significant bits of the target
|
|
to 0. This is likely slightly faster but less accurate. When this flag
|
|
is not specified, the most significant bits of the source are duplicated
|
|
in the least significant bits of the destination.</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioQuantize" c:type="GstAudioQuantize" disguised="1">
|
|
|
|
<method name="free" c:identifier="gst_audio_quantize_free">
|
|
<doc xml:space="preserve">Free a #GstAudioQuantize.</doc>
|
|
|
|
<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">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">Reset @quant to the state is was when created, clearing any
|
|
history it might have.</doc>
|
|
|
|
<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">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">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>
|
|
|
|
<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">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">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">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">a #GstAudioDitherMethod</doc>
|
|
<type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/>
|
|
</parameter>
|
|
<parameter name="ns" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioNoiseShapingMethod</doc>
|
|
<type name="AudioNoiseShapingMethod" c:type="GstAudioNoiseShapingMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioQuantizeFlags</doc>
|
|
<type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioFormat of the samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">samples are non-interleaved</doc>
|
|
</member>
|
|
</bitfield>
|
|
<record name="AudioResampler" c:type="GstAudioResampler" disguised="1" version="1.10">
|
|
<doc xml:space="preserve">#GstAudioResampler is a structure which holds the information
|
|
required to perform various kinds of resampling filtering.</doc>
|
|
|
|
<method name="free" c:identifier="gst_audio_resampler_free">
|
|
<doc xml:space="preserve">Free a previously allocated #GstAudioResampler @resampler.</doc>
|
|
|
|
<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">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">Get the number of input frames that would currently be needed
|
|
to produce @out_frames from @resampler.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Get the maximum number of input samples that the resampler would
|
|
need before producing output.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the number of output frames that would be currently available when
|
|
@in_frames are given to @resampler.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">The number of frames that would be available after giving
|
|
@in_frames as input to @resampler.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="resampler" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">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">input samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="in_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">output samples</doc>
|
|
<type name="gpointer" c:type="gpointer*"/>
|
|
</parameter>
|
|
<parameter name="out_frames" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Reset @resampler to the state it was when it was first created, discarding
|
|
all sample history.</doc>
|
|
|
|
<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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioResampler</doc>
|
|
<type name="AudioResampler" c:type="GstAudioResampler*"/>
|
|
</instance-parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">new input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">new output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Make a new resampler.</doc>
|
|
|
|
<return-value transfer-ownership="full" skip="1">
|
|
<doc xml:space="preserve">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">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioResamplerFlags</doc>
|
|
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Set the parameters for resampling from @in_rate to @out_rate using @method
|
|
for @quality in @options.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="quality" transfer-ownership="none">
|
|
<doc xml:space="preserve">the quality</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">the input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">the output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstStructure</doc>
|
|
<type name="Gst.Structure" c:type="GstStructure*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</record>
|
|
<enumeration name="AudioResamplerFilterInterpolation" version="1.10" glib:type-name="GstAudioResamplerFilterInterpolation" glib:get-type="gst_audio_resampler_filter_interpolation_get_type" c:type="GstAudioResamplerFilterInterpolation">
|
|
<doc xml:space="preserve">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">no interpolation</doc>
|
|
</member>
|
|
<member name="linear" value="1" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR" glib:nick="linear">
|
|
<doc xml:space="preserve">linear interpolation of the
|
|
filter coefficients.</doc>
|
|
</member>
|
|
<member name="cubic" value="2" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC" glib:nick="cubic">
|
|
<doc xml:space="preserve">cubic interpolation of the
|
|
filter coefficients.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<enumeration name="AudioResamplerFilterMode" version="1.10" glib:type-name="GstAudioResamplerFilterMode" glib:get-type="gst_audio_resampler_filter_mode_get_type" c:type="GstAudioResamplerFilterMode">
|
|
<doc xml:space="preserve">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">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">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">Automatically choose between interpolated
|
|
and full filter tables.</doc>
|
|
</member>
|
|
</enumeration>
|
|
<bitfield name="AudioResamplerFlags" version="1.10" glib:type-name="GstAudioResamplerFlags" glib:get-type="gst_audio_resampler_flags_get_type" c:type="GstAudioResamplerFlags">
|
|
<doc xml:space="preserve">Different resampler flags.</doc>
|
|
<member name="none" value="0" c:identifier="GST_AUDIO_RESAMPLER_FLAG_NONE" glib:nick="none">
|
|
<doc xml:space="preserve">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">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">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">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="1.10" glib:type-name="GstAudioResamplerMethod" glib:get-type="gst_audio_resampler_method_get_type" c:type="GstAudioResamplerMethod">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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>
|
|
|
|
<function name="debug_spec_buff" c:identifier="gst_audio_ring_buffer_debug_spec_buff">
|
|
<doc xml:space="preserve">Print debug info about the buffer sized in @spec to the debug log.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Print debug info about the parsed caps in @spec to the debug log.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Parse @caps into @spec.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">a spec</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
<parameter name="caps" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<virtual-method name="acquire" invoker="acquire">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Activate @buf to start or stop pulling data.
|
|
|
|
MT safe.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Clear all samples from the ringbuffer.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">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">Close the audio device associated with the ring buffer. The ring buffer
|
|
should already have been released via gst_audio_ring_buffer_release().</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="commit" invoker="commit">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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 transferred from the
|
|
ringbuffer to the device but still not played.
|
|
|
|
For capture ringbuffers this is the amount of samples in the device that are
|
|
not yet transferred to the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="pause" invoker="pause">
|
|
<doc xml:space="preserve">Pause processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Free the resources of the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to release</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="resume">
|
|
|
|
<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">Start processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Stop processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Activate @buf to start or stop pulling data.
|
|
|
|
MT safe.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Subclasses should call this function to notify the fact that
|
|
@advance segments are now processed by the device.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">the #GstAudioRingBuffer to advance</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="advance" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Clear the given segment of the buffer with silence samples.
|
|
This function is used by subclasses.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">the #GstAudioRingBuffer to clear</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Clear all samples from the ringbuffer.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">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">Close the audio device associated with the ring buffer. The ring buffer
|
|
should already have been released via gst_audio_ring_buffer_release().</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result
|
|
will be put in @dest_val.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="src_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve">the source format</doc>
|
|
<type name="Gst.Format" c:type="GstFormat"/>
|
|
</parameter>
|
|
<parameter name="src_val" transfer-ownership="none">
|
|
<doc xml:space="preserve">the source value</doc>
|
|
<type name="gint64" c:type="gint64"/>
|
|
</parameter>
|
|
<parameter name="dest_fmt" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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 transferred from the
|
|
ringbuffer to the device but still not played.
|
|
|
|
For capture ringbuffers this is the amount of samples in the device that are
|
|
not yet transferred to the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Checks the status of the device associated with the ring buffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Check if the ringbuffer is acquired and ready to use.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Check if @buf is activated.
|
|
|
|
MT safe.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Check if @buf is flushing.
|
|
|
|
MT safe.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Tell the ringbuffer that it is allowed to start playback when
|
|
the ringbuffer is filled with samples.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="allowed" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Pause processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Returns a pointer to memory where the data from segment @segment
|
|
can be found. This function is mostly used by subclasses.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">
|
|
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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to read from</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">Free the resources of the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<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">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">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">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">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>
|
|
|
|
<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">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">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">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">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">Tell the ringbuffer about the device's channel positions. This must
|
|
be called in when the ringbuffer is acquired.</doc>
|
|
|
|
<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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Set the ringbuffer to flushing mode or normal mode.
|
|
|
|
MT safe.</doc>
|
|
|
|
<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">the #GstAudioRingBuffer to flush</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="flushing" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<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">the #GstAudioRingBuffer to use</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</instance-parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<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">Start processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Stop processing samples from the ringbuffer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">used to signal start/stop/pause/resume actions</doc>
|
|
<type name="GLib.Cond" c:type="GCond"/>
|
|
</field>
|
|
<field name="open">
|
|
<doc xml:space="preserve">boolean indicating that the ringbuffer is open</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="acquired">
|
|
<doc xml:space="preserve">boolean indicating that the ringbuffer is acquired</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</field>
|
|
<field name="memory">
|
|
<doc xml:space="preserve">data in the ringbuffer</doc>
|
|
<type name="guint8" c:type="guint8*"/>
|
|
</field>
|
|
<field name="size">
|
|
<doc xml:space="preserve">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">format and layout of the ringbuffer data</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec"/>
|
|
</field>
|
|
<field name="samples_per_seg">
|
|
<doc xml:space="preserve">number of samples in one segment</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="empty_seg">
|
|
<doc xml:space="preserve">pointer to memory holding one segment of silence samples</doc>
|
|
<type name="guint8" c:type="guint8*"/>
|
|
</field>
|
|
<field name="state">
|
|
<doc xml:space="preserve">state of the buffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segdone">
|
|
<doc xml:space="preserve">readpointer in the ringbuffer</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segbase">
|
|
<doc xml:space="preserve">segment corresponding to segment 0 (unused)</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="waiting">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="rbuf" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">The vmethods that subclasses can override to implement the ringbuffer.</doc>
|
|
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve">parent class</doc>
|
|
<type name="Gst.ObjectClass" c:type="GstObjectClass"/>
|
|
</field>
|
|
<field name="open_device">
|
|
<callback name="open_device">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="acquire">
|
|
<callback name="acquire">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to acquire</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">the specs of the buffer</doc>
|
|
<type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="release">
|
|
<callback name="release">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to release</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="close_device">
|
|
<callback name="close_device">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="start">
|
|
<callback name="start">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to start</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="pause">
|
|
<callback name="pause">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to pause</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="resume">
|
|
<callback name="resume">
|
|
|
|
<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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to stop</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="delay">
|
|
<callback name="delay">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to query</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="activate">
|
|
<callback name="activate">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to activate</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="active" transfer-ownership="none">
|
|
<doc xml:space="preserve">the new mode</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="commit">
|
|
<callback name="commit">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">the #GstAudioRingBuffer to commit</doc>
|
|
<type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/>
|
|
</parameter>
|
|
<parameter name="sample" transfer-ownership="none">
|
|
<doc xml:space="preserve">the sample position of the data</doc>
|
|
<type name="guint64" c:type="guint64*"/>
|
|
</parameter>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">accumulator for rate conversion.</doc>
|
|
<type name="gint" c:type="gint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="clear_all">
|
|
<callback name="clear_all">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buf" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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">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">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">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">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">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">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">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">samples in MPEG-2 AAC raw format (Since: 1.12)</doc>
|
|
</member>
|
|
<member name="mpeg4_aac_raw" value="13" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW" glib:nick="mpeg4-aac-raw">
|
|
<doc xml:space="preserve">samples in MPEG-4 AAC raw format (Since: 1.12)</doc>
|
|
</member>
|
|
<member name="flac" value="14" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC" glib:nick="flac">
|
|
<doc xml:space="preserve">samples in FLAC format (Since: 1.12)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec">
|
|
<doc xml:space="preserve">The structure containing the format specification of the ringbuffer.</doc>
|
|
|
|
<field name="caps" writable="1">
|
|
<doc xml:space="preserve">The caps that generated the Spec.</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</field>
|
|
<field name="type" writable="1">
|
|
<doc xml:space="preserve">the sample type</doc>
|
|
<type name="AudioRingBufferFormatType" c:type="GstAudioRingBufferFormatType"/>
|
|
</field>
|
|
<field name="info" writable="1">
|
|
<doc xml:space="preserve">the #GstAudioInfo</doc>
|
|
<type name="AudioInfo" c:type="GstAudioInfo"/>
|
|
</field>
|
|
<field name="latency_time" writable="1">
|
|
<doc xml:space="preserve">the latency in microseconds</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="buffer_time" writable="1">
|
|
<doc xml:space="preserve">the total buffer size in microseconds</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</field>
|
|
<field name="segsize" writable="1">
|
|
<doc xml:space="preserve">the size of one segment in bytes</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="segtotal" writable="1">
|
|
<doc xml:space="preserve">the total number of segments</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</field>
|
|
<field name="seglatency" writable="1">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">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>
|
|
|
|
<virtual-method name="close">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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="pause">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="prepare">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<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">
|
|
|
|
<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="resume">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="stop">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</virtual-method>
|
|
<virtual-method name="unprepare">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve">the parent class structure.</doc>
|
|
<type name="AudioBaseSinkClass" c:type="GstAudioBaseSinkClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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="pause">
|
|
<callback name="pause">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="resume">
|
|
<callback name="resume">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="stop">
|
|
<callback name="stop">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
<field name="extension">
|
|
<doc xml:space="preserve">class extension structure. Since: 1.18</doc>
|
|
<type name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension*"/>
|
|
</field>
|
|
</record>
|
|
<record name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension">
|
|
|
|
<field name="clear_all">
|
|
<callback name="clear_all">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="sink" transfer-ownership="none">
|
|
<type name="AudioSink" c:type="GstAudioSink*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</callback>
|
|
</field>
|
|
</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">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>
|
|
|
|
<virtual-method name="close">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">#GstAudioSrc class. Override the vmethod to implement
|
|
functionality.</doc>
|
|
|
|
<field name="parent_class">
|
|
<doc xml:space="preserve">the parent class.</doc>
|
|
<type name="AudioBaseSrcClass" c:type="GstAudioBaseSrcClass"/>
|
|
</field>
|
|
<field name="open">
|
|
<callback name="open">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">#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>
|
|
|
|
<constructor name="new" c:identifier="gst_audio_stream_align_new" version="1.14">
|
|
<doc xml:space="preserve">Allocate a new #GstAudioStreamAlign with the given configuration. All
|
|
processing happens according to sample rate @rate, until
|
|
gst_audio_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>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">a sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Copy a GstAudioStreamAlign structure.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">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">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">Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new()
|
|
or gst_audio_stream_align_copy().</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_alignment_threshold" c:identifier="gst_audio_stream_align_get_alignment_threshold" version="1.14">
|
|
<doc xml:space="preserve">Gets the currently configured alignment threshold.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">The currently configured alignment threshold</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_discont_wait" c:identifier="gst_audio_stream_align_get_discont_wait" version="1.14">
|
|
<doc xml:space="preserve">Gets the currently configured discont wait.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">The currently configured discont wait</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_rate" c:identifier="gst_audio_stream_align_get_rate" version="1.14">
|
|
<doc xml:space="preserve">Gets the currently configured sample rate.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">The currently configured sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="get_samples_since_discont" c:identifier="gst_audio_stream_align_get_samples_since_discont" version="1.14">
|
|
<doc xml:space="preserve">Returns the number of samples that were processed since the last
|
|
discontinuity was detected.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Timestamp that was passed when a discontinuity was detected, i.e. the first
|
|
timestamp after the discontinuity.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Marks the next buffer as discontinuous and resets timestamp tracking.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="process" c:identifier="gst_audio_stream_align_process" version="1.14">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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">Sets @alignment_treshold as new alignment threshold for the following processing.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="alignment_threshold" transfer-ownership="none">
|
|
<doc xml:space="preserve">a new alignment threshold</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_discont_wait" c:identifier="gst_audio_stream_align_set_discont_wait" version="1.14">
|
|
<doc xml:space="preserve">Sets @alignment_treshold as new discont wait for the following processing.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="discont_wait" transfer-ownership="none">
|
|
<doc xml:space="preserve">a new discont wait</doc>
|
|
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
<method name="set_rate" c:identifier="gst_audio_stream_align_set_rate" version="1.14">
|
|
<doc xml:space="preserve">Sets @rate as new sample rate for the following processing. If the sample
|
|
rate differs this implicitly marks the next data as discontinuous.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<instance-parameter name="align" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioStreamAlign</doc>
|
|
<type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/>
|
|
</instance-parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">a new sample rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</method>
|
|
</record>
|
|
<function-macro name="CLOCK_TIME_TO_FRAMES" c:identifier="GST_CLOCK_TIME_TO_FRAMES" introspectable="0">
|
|
<doc xml:space="preserve">Calculate frames from @clocktime and sample @rate.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="clocktime">
|
|
<doc xml:space="preserve">clock time</doc>
|
|
</parameter>
|
|
<parameter name="rate">
|
|
<doc xml:space="preserve">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">Calculate clocktime from sample @frames and @rate.</doc>
|
|
|
|
<parameters>
|
|
<parameter name="frames">
|
|
<doc xml:space="preserve">sample frames</doc>
|
|
</parameter>
|
|
<parameter name="rate">
|
|
<doc xml:space="preserve">sampling rate</doc>
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR" c:identifier="GST_IS_AUDIO_AGGREGATOR" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_CLASS" c:identifier="GST_IS_AUDIO_AGGREGATOR_CLASS" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_AGGREGATOR_PAD" c:identifier="GST_IS_AUDIO_AGGREGATOR_PAD" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SINK" c:identifier="GST_IS_AUDIO_BASE_SINK" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_BASE_SRC" c:identifier="GST_IS_AUDIO_BASE_SRC" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CD_SRC" c:identifier="GST_IS_AUDIO_CD_SRC" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CLOCK" c:identifier="GST_IS_AUDIO_CLOCK" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_CLOCK_CLASS" c:identifier="GST_IS_AUDIO_CLOCK_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_DECODER" c:identifier="GST_IS_AUDIO_DECODER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_DECODER_CLASS" c:identifier="GST_IS_AUDIO_DECODER_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_ENCODER" c:identifier="GST_IS_AUDIO_ENCODER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_ENCODER_CLASS" c:identifier="GST_IS_AUDIO_ENCODER_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_FILTER" c:identifier="GST_IS_AUDIO_FILTER" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_FILTER_CLASS" c:identifier="GST_IS_AUDIO_FILTER_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_RING_BUFFER" c:identifier="GST_IS_AUDIO_RING_BUFFER" introspectable="0">
|
|
|
|
<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">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SINK" c:identifier="GST_IS_AUDIO_SINK" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SINK_CLASS" c:identifier="GST_IS_AUDIO_SINK_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SRC" c:identifier="GST_IS_AUDIO_SRC" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<function-macro name="IS_AUDIO_SRC_CLASS" c:identifier="GST_IS_AUDIO_SRC_CLASS" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="klass">
|
|
</parameter>
|
|
</parameters>
|
|
</function-macro>
|
|
<constant name="META_TAG_AUDIO_CHANNELS_STR" value="channels" c:type="GST_META_TAG_AUDIO_CHANNELS_STR" version="1.2">
|
|
<doc xml:space="preserve">This metadata stays relevant as long as channels are unchanged.</doc>
|
|
|
|
<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">This metadata stays relevant as long as sample rate is unchanged.</doc>
|
|
|
|
<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">This metadata is relevant for audio streams.</doc>
|
|
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</constant>
|
|
<function-macro name="STREAM_VOLUME_GET_INTERFACE" c:identifier="GST_STREAM_VOLUME_GET_INTERFACE" introspectable="0">
|
|
|
|
<parameters>
|
|
<parameter name="obj">
|
|
</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">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>
|
|
|
|
<function name="convert_volume" c:identifier="gst_stream_volume_convert_volume">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the converted volume</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstStreamVolumeFormat to convert from</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstStreamVolumeFormat to convert to</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">#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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">#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">
|
|
|
|
<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">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="mute" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<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">#GstStreamVolume that should be used</doc>
|
|
<type name="StreamVolume" c:type="GstStreamVolume*"/>
|
|
</instance-parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstStreamVolumeFormat of @val</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<member name="linear" value="0" c:identifier="GST_STREAM_VOLUME_FORMAT_LINEAR">
|
|
<doc xml:space="preserve">Linear scale factor, 1.0 = 100%</doc>
|
|
</member>
|
|
<member name="cubic" value="1" c:identifier="GST_STREAM_VOLUME_FORMAT_CUBIC">
|
|
<doc xml:space="preserve">Cubic volume scale</doc>
|
|
</member>
|
|
<member name="db" value="2" c:identifier="GST_STREAM_VOLUME_FORMAT_DB">
|
|
<doc xml:space="preserve">Logarithmic volume scale (dB, amplitude not power)</doc>
|
|
</member>
|
|
</enumeration>
|
|
<record name="StreamVolumeInterface" c:type="GstStreamVolumeInterface" glib:is-gtype-struct-for="StreamVolume">
|
|
|
|
<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">Clip the buffer to the given %GstSegment.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">%NULL if the buffer is completely outside the configured segment,
|
|
otherwise the clipped buffer is returned.
|
|
|
|
If the buffer has no timestamp, it is assumed to be inside the segment and
|
|
is not clipped</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve">The buffer to clip.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="segment" transfer-ownership="none">
|
|
<doc xml:space="preserve">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
|
|
the buffer should be clipped.</doc>
|
|
<type name="Gst.Segment" c:type="const GstSegment*"/>
|
|
</parameter>
|
|
<parameter name="rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">sample rate.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_buffer_reorder_channels" c:identifier="gst_audio_buffer_reorder_channels" moved-to="AudioBuffer.reorder_channels">
|
|
<doc xml:space="preserve">Reorders @buffer from the channel positions @from to the channel
|
|
positions @to. @from and @to must contain the same number of
|
|
positions and the same positions, only in a different order.
|
|
@buffer must be writable.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">The buffer to reorder.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions in the buffer.</doc>
|
|
<array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions to convert to.</doc>
|
|
<array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*">
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/>
|
|
</array>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_buffer_truncate" c:identifier="gst_audio_buffer_truncate" moved-to="AudioBuffer.truncate" version="1.16">
|
|
<doc xml:space="preserve">Truncate the buffer to finally have @samples number of samples, removing
|
|
the necessary amount of samples from the end and @trim number of samples
|
|
from the beginning.
|
|
|
|
After calling this function the caller does not own a reference to
|
|
@buffer anymore.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">the truncated buffer or %NULL if the arguments
|
|
were invalid</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="full">
|
|
<doc xml:space="preserve">The buffer to truncate.</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="bpf" transfer-ownership="none">
|
|
<doc xml:space="preserve">size of one audio frame in bytes. This is the size of one sample *
|
|
number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="trim" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of samples to remove from the beginning of the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">the final number of samples that should exist in this buffer or -1
|
|
to use all the remaining samples if you are only removing samples from the
|
|
beginning.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_channel_get_fallback_mask" c:identifier="gst_audio_channel_get_fallback_mask" version="1.8">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Create a new channel mixer object for the given parameters.</doc>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">#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">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">positions of input channels</doc>
|
|
<type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">number of output channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Create a new channel mixer object for the given parameters.</doc>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">#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">number of input channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">The number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="channel_mask" transfer-ownership="none">
|
|
<doc xml:space="preserve">The input channel_mask</doc>
|
|
<type name="guint64" c:type="guint64"/>
|
|
</parameter>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="force_order" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Converts @position to a human-readable string representation for
|
|
debugging purposes.</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">a newly allocated string representing
|
|
@position</doc>
|
|
<type name="utf8" c:type="gchar*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Reorders the channel positions in @position from any order to
|
|
the GStreamer channel order.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">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">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="force_order" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">
|
|
|
|
<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">Construct a #GstAudioFormat with given parameters.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">signed or unsigned format</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">amount of bits used per sample</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="depth" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Fill @length bytes in @dest with silence samples for @info.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioFormatInfo</doc>
|
|
<type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/>
|
|
</parameter>
|
|
<parameter name="dest" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Convert the @format string to its #GstAudioFormat.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">Get the #GstAudioFormatInfo for @format</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<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">
|
|
|
|
<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_formats_raw" c:identifier="gst_audio_formats_raw" version="1.18">
|
|
<doc xml:space="preserve">Return all the raw audio formats supported by GStreamer.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">an array of #GstAudioFormat</doc>
|
|
<array length="0" zero-terminated="0" c:type="const GstAudioFormat*">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</array>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
|
|
<doc xml:space="preserve">the number of elements in the returned array</doc>
|
|
<type name="guint" c:type="guint*"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_get_channel_reorder_map" c:identifier="gst_audio_get_channel_reorder_map">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%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">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions 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">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">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">Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
|
|
payloading type from @spec.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">size of @src in bytes</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="dst" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">size of @dst in bytes</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="spec" transfer-ownership="none">
|
|
<doc xml:space="preserve">the ringbufer spec for @src</doc>
|
|
<type name="AudioRingBufferSpec" c:type="const GstAudioRingBufferSpec*"/>
|
|
</parameter>
|
|
<parameter name="endianness" transfer-ownership="none">
|
|
<doc xml:space="preserve">the expected byte order of the payloaded data</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_make_raw_caps" c:identifier="gst_audio_make_raw_caps" version="1.18">
|
|
<doc xml:space="preserve">Return a generic raw audio caps for formats defined in @formats.
|
|
If @formats is %NULL returns a caps for all the supported raw audio formats,
|
|
see gst_audio_formats_raw().</doc>
|
|
|
|
<return-value transfer-ownership="full">
|
|
<doc xml:space="preserve">an audio @GstCaps</doc>
|
|
<type name="Gst.Caps" c:type="GstCaps*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="formats" transfer-ownership="none" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">an array of raw #GstAudioFormat, or %NULL</doc>
|
|
<array length="1" zero-terminated="0" c:type="const GstAudioFormat*">
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</array>
|
|
</parameter>
|
|
<parameter name="len" transfer-ownership="none">
|
|
<doc xml:space="preserve">the size of @formats</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="layout" transfer-ownership="none">
|
|
<doc xml:space="preserve">the layout of audio samples</doc>
|
|
<type name="AudioLayout" c:type="GstAudioLayout"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
<function name="audio_meta_api_get_type" c:identifier="gst_audio_meta_api_get_type">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="GType" c:type="GType"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_meta_get_info" c:identifier="gst_audio_meta_get_info" moved-to="AudioMeta.get_info">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/>
|
|
</return-value>
|
|
</function>
|
|
<function name="audio_quantize_new" c:identifier="gst_audio_quantize_new" moved-to="AudioQuantize.new" introspectable="0">
|
|
<doc xml:space="preserve">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>
|
|
|
|
<return-value>
|
|
<doc xml:space="preserve">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">a #GstAudioDitherMethod</doc>
|
|
<type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/>
|
|
</parameter>
|
|
<parameter name="ns" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioNoiseShapingMethod</doc>
|
|
<type name="AudioNoiseShapingMethod" c:type="GstAudioNoiseShapingMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioQuantizeFlags</doc>
|
|
<type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioFormat of the samples</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">%TRUE if the reordering was possible.</doc>
|
|
<type name="gboolean" c:type="gboolean"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="data" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">The size of the memory.</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">The %GstAudioFormat of the buffer.</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">The number of channels.</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">The channel positions in the buffer.</doc>
|
|
<array length="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">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">Make a new resampler.</doc>
|
|
|
|
<return-value transfer-ownership="full" skip="1">
|
|
<doc xml:space="preserve">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">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="flags" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstAudioResamplerFlags</doc>
|
|
<type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioFormat</doc>
|
|
<type name="AudioFormat" c:type="GstAudioFormat"/>
|
|
</parameter>
|
|
<parameter name="channels" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of channels</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Set the parameters for resampling from @in_rate to @out_rate using @method
|
|
for @quality in @options.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<type name="none" c:type="void"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="method" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstAudioResamplerMethod</doc>
|
|
<type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/>
|
|
</parameter>
|
|
<parameter name="quality" transfer-ownership="none">
|
|
<doc xml:space="preserve">the quality</doc>
|
|
<type name="guint" c:type="guint"/>
|
|
</parameter>
|
|
<parameter name="in_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">the input rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="out_rate" transfer-ownership="none">
|
|
<doc xml:space="preserve">the output rate</doc>
|
|
<type name="gint" c:type="gint"/>
|
|
</parameter>
|
|
<parameter name="options" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">Attaches #GstAudioClippingMeta metadata to @buffer with the given parameters.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioClippingMeta on @buffer.</doc>
|
|
<type name="AudioClippingMeta" c:type="GstAudioClippingMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="format" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioDownmixMeta on @buffer.</doc>
|
|
<type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="from_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">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">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">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">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>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioMeta that was attached on the @buffer</doc>
|
|
<type name="AudioMeta" c:type="GstAudioMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="info" transfer-ownership="none">
|
|
<doc xml:space="preserve">the audio properties of the buffer</doc>
|
|
<type name="AudioInfo" c:type="const GstAudioInfo*"/>
|
|
</parameter>
|
|
<parameter name="samples" transfer-ownership="none">
|
|
<doc xml:space="preserve">the number of valid samples in the buffer</doc>
|
|
<type name="gsize" c:type="gsize"/>
|
|
</parameter>
|
|
<parameter name="offsets" transfer-ownership="none" nullable="1" allow-none="1">
|
|
<doc xml:space="preserve">the offsets (in bytes) where each channel plane starts
|
|
in the buffer or %NULL to calculate it (see below); must be %NULL also
|
|
when @info->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">
|
|
|
|
<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">
|
|
|
|
<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">Find the #GstAudioDownmixMeta on @buffer for the given destination
|
|
channel positions.</doc>
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the #GstAudioDownmixMeta on @buffer.</doc>
|
|
<type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="buffer" transfer-ownership="none">
|
|
<doc xml:space="preserve">a #GstBuffer</doc>
|
|
<type name="Gst.Buffer" c:type="GstBuffer*"/>
|
|
</parameter>
|
|
<parameter name="to_position" transfer-ownership="none">
|
|
<doc xml:space="preserve">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">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">
|
|
|
|
<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">
|
|
|
|
<return-value transfer-ownership="none">
|
|
<doc xml:space="preserve">the converted volume</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</return-value>
|
|
<parameters>
|
|
<parameter name="from" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstStreamVolumeFormat to convert from</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="to" transfer-ownership="none">
|
|
<doc xml:space="preserve">#GstStreamVolumeFormat to convert to</doc>
|
|
<type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/>
|
|
</parameter>
|
|
<parameter name="val" transfer-ownership="none">
|
|
<doc xml:space="preserve">Volume in @from format that should be converted</doc>
|
|
<type name="gdouble" c:type="gdouble"/>
|
|
</parameter>
|
|
</parameters>
|
|
</function>
|
|
</namespace>
|
|
</repository>
|