diff options
| author | Thibault Saunier <tsaunier@igalia.com> | 2022-08-31 14:15:16 -0400 |
|---|---|---|
| committer | Thibault Saunier <tsaunier@igalia.com> | 2023-04-22 09:32:32 -0400 |
| commit | b14e675a2786c31aa614ed23591fbed03d05a946 (patch) | |
| tree | 734669cdb6c63a3fbe7ab27f796e044d47665094 /girs/GstAudio-1.0.gir | |
| parent | d4a910649955ecbee2db1ced67eed748995f6d58 (diff) | |
| download | gstreamer-b14e675a2786c31aa614ed23591fbed03d05a946.tar.gz | |
gir: Checkout all .gir files and check that they are updated on the CI
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3010>
Diffstat (limited to 'girs/GstAudio-1.0.gir')
| -rw-r--r-- | girs/GstAudio-1.0.gir | 10931 |
1 files changed, 10931 insertions, 0 deletions
diff --git a/girs/GstAudio-1.0.gir b/girs/GstAudio-1.0.gir new file mode 100644 index 0000000000..f967075685 --- /dev/null +++ b/girs/GstAudio-1.0.gir @@ -0,0 +1,10931 @@ +<!-- This file was automatically generated from C sources - DO NOT EDIT! +To affect the contents of this file, edit the original C definitions, +and/or use gtk-doc annotations. --> +<repository 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"> + <include name="GLib" version="2.0"/> + <include name="GModule" version="2.0"/> + <include name="GObject" version="2.0"/> + <include name="Gst" version="1.0"/> + <include name="GstBase" version="1.0"/> + <package name="gstreamer-audio-1.0"/> + <c:include name="gst/audio/audio.h"/> + <namespace name="GstAudio" version="1.0" shared-library="libgstaudio-1.0.so.0" c:identifier-prefixes="Gst" c:symbol-prefixes="gst"> + <function-macro name="AUDIO_AGGREGATOR" c:identifier="GST_AUDIO_AGGREGATOR" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_GET_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_PAD" c:identifier="GST_AUDIO_AGGREGATOR_PAD" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_PAD_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_PAD_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_AGGREGATOR_PAD_GET_CLASS" c:identifier="GST_AUDIO_AGGREGATOR_PAD_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK" c:identifier="GST_AUDIO_BASE_SINK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK_CAST" c:identifier="GST_AUDIO_BASE_SINK_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK_CLASS" c:identifier="GST_AUDIO_BASE_SINK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK_CLOCK" c:identifier="GST_AUDIO_BASE_SINK_CLOCK" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Get the #GstClock of @obj.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">a #GstAudioBaseSink</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK_GET_CLASS" c:identifier="GST_AUDIO_BASE_SINK_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SINK_PAD" c:identifier="GST_AUDIO_BASE_SINK_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Get the sink #GstPad of @obj.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">a #GstAudioBaseSink</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC" c:identifier="GST_AUDIO_BASE_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC_CAST" c:identifier="GST_AUDIO_BASE_SRC_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC_CLASS" c:identifier="GST_AUDIO_BASE_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC_CLOCK" c:identifier="GST_AUDIO_BASE_SRC_CLOCK" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Get the #GstClock of @obj.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">a #GstAudioBaseSrc</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC_GET_CLASS" c:identifier="GST_AUDIO_BASE_SRC_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BASE_SRC_PAD" c:identifier="GST_AUDIO_BASE_SRC_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Get the source #GstPad of @obj.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">a #GstAudioBaseSrc</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_BPF" c:identifier="GST_AUDIO_BUFFER_BPF" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_BPS" c:identifier="GST_AUDIO_BUFFER_BPS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_CHANNELS" c:identifier="GST_AUDIO_BUFFER_CHANNELS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_DEPTH" c:identifier="GST_AUDIO_BUFFER_DEPTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_FORMAT" c:identifier="GST_AUDIO_BUFFER_FORMAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_LAYOUT" c:identifier="GST_AUDIO_BUFFER_LAYOUT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_N_PLANES" c:identifier="GST_AUDIO_BUFFER_N_PLANES" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_N_SAMPLES" c:identifier="GST_AUDIO_BUFFER_N_SAMPLES" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_PLANE_DATA" c:identifier="GST_AUDIO_BUFFER_PLANE_DATA" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + <parameter name="p"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_PLANE_SIZE" c:identifier="GST_AUDIO_BUFFER_PLANE_SIZE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_RATE" c:identifier="GST_AUDIO_BUFFER_RATE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_SAMPLE_STRIDE" c:identifier="GST_AUDIO_BUFFER_SAMPLE_STRIDE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_BUFFER_WIDTH" c:identifier="GST_AUDIO_BUFFER_WIDTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CAPS_MAKE" c:identifier="GST_AUDIO_CAPS_MAKE" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Generic caps string for audio, for use in pad templates.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">string format that describes the sample layout, as string + (e.g. "S16LE", "S8", etc.)</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CD_SRC" c:identifier="GST_AUDIO_CD_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CD_SRC_CLASS" c:identifier="GST_AUDIO_CD_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CD_SRC_GET_CLASS" c:identifier="GST_AUDIO_CD_SRC_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_CHANNELS_RANGE" value="(int) [ 1, max ]" c:type="GST_AUDIO_CHANNELS_RANGE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Maximum range of allowed channels, for use in template caps strings.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_CHANNEL_POSITION_MASK" c:identifier="GST_AUDIO_CHANNEL_POSITION_MASK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <parameters> + <parameter name="pos"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CLOCK" c:identifier="GST_AUDIO_CLOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CLOCK_CAST" c:identifier="GST_AUDIO_CLOCK_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_CLOCK_CLASS" c:identifier="GST_AUDIO_CLOCK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_CONVERTER_OPT_DITHER_METHOD" value="GstAudioConverter.dither-method" c:type="GST_AUDIO_CONVERTER_OPT_DITHER_METHOD"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">#GstAudioDitherMethod, The dither method to use when +changing bit depth. +Default is #GST_AUDIO_DITHER_NONE.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_CONVERTER_OPT_DITHER_THRESHOLD" value="GstAudioConverter.dither-threshold" c:type="GST_AUDIO_CONVERTER_OPT_DITHER_THRESHOLD" version="1.22"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">Threshold for the output bit depth at/below which to apply dithering. + +Default is 20 bit.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_CONVERTER_OPT_MIX_MATRIX" value="GstAudioConverter.mix-matrix" c:type="GST_AUDIO_CONVERTER_OPT_MIX_MATRIX"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">#GST_TYPE_LIST, The channel mapping matrix. + +The matrix coefficients must be between -1 and 1: the number of rows is equal +to the number of output channels and the number of columns is equal to the +number of input channels. + +## Example matrix generation code +To generate the matrix using code: + +|[ +GValue v = G_VALUE_INIT; +GValue v2 = G_VALUE_INIT; +GValue v3 = G_VALUE_INIT; + +g_value_init (&v2, GST_TYPE_ARRAY); +g_value_init (&v3, G_TYPE_DOUBLE); +g_value_set_double (&v3, 1); +gst_value_array_append_value (&v2, &v3); +g_value_unset (&v3); +[ Repeat for as many double as your input channels - unset and reinit v3 ] +g_value_init (&v, GST_TYPE_ARRAY); +gst_value_array_append_value (&v, &v2); +g_value_unset (&v2); +[ Repeat for as many v2's as your output channels - unset and reinit v2] +g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v); +g_value_unset (&v); +]|</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD" value="GstAudioConverter.noise-shaping-method" c:type="GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">#GstAudioNoiseShapingMethod, The noise shaping method to use +to mask noise from quantization errors. +Default is #GST_AUDIO_NOISE_SHAPING_NONE.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_CONVERTER_OPT_QUANTIZATION" value="GstAudioConverter.quantization" c:type="GST_AUDIO_CONVERTER_OPT_QUANTIZATION"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">#G_TYPE_UINT, The quantization amount. Components will be +quantized to multiples of this value. +Default is 1</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_CONVERTER_OPT_RESAMPLER_METHOD" value="GstAudioConverter.resampler-method" c:type="GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">#GstAudioResamplerMethod, The resampler method to use when +changing sample rates. +Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_DECODER" c:identifier="GST_AUDIO_DECODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_CAST" c:identifier="GST_AUDIO_DECODER_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_CLASS" c:identifier="GST_AUDIO_DECODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_ERROR" c:identifier="GST_AUDIO_DECODER_ERROR" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Utility function that audio decoder elements can use in case they encountered +a data processing error that may be fatal for the current "data unit" but +need not prevent subsequent decoding. Such errors are counted and if there +are too many, as configured in the context's max_errors, the pipeline will +post an error message and the application will be requested to stop further +media processing. Otherwise, it is considered a "glitch" and only a warning +is logged. In either case, @ret is set to the proper value to +return to upstream/caller (indicating either GST_FLOW_ERROR or GST_FLOW_OK).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="el"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">the base audio decoder element that generates the error</doc> + </parameter> + <parameter name="weight"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">element defined weight of the error, added to error count</doc> + </parameter> + <parameter name="domain"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">like CORE, LIBRARY, RESOURCE or STREAM (see #gstreamer-GstGError)</doc> + </parameter> + <parameter name="code"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">error code defined for that domain (see #gstreamer-GstGError)</doc> + </parameter> + <parameter name="text"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">the message to display (format string and args enclosed in + parentheses)</doc> + </parameter> + <parameter name="debug"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">debugging information for the message (format string and args + enclosed in parentheses)</doc> + </parameter> + <parameter name="ret"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">variable to receive return value</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_GET_CLASS" c:identifier="GST_AUDIO_DECODER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_INPUT_SEGMENT" c:identifier="GST_AUDIO_DECODER_INPUT_SEGMENT" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Gives the input segment of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">audio decoder instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_DECODER_MAX_ERRORS" value="-1" c:type="GST_AUDIO_DECODER_MAX_ERRORS"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Default maximum number of errors tolerated before signaling error.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <type name="gint" c:type="gint"/> + </constant> + <function-macro name="AUDIO_DECODER_OUTPUT_SEGMENT" c:identifier="GST_AUDIO_DECODER_OUTPUT_SEGMENT" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Gives the output segment of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">audio decoder instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_DECODER_SINK_NAME" value="sink" c:type="GST_AUDIO_DECODER_SINK_NAME"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">The name of the templates for the sink pad.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_DECODER_SINK_PAD" c:identifier="GST_AUDIO_DECODER_SINK_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Gives the pointer to the sink #GstPad object of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">base audio codec instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_DECODER_SRC_NAME" value="src" c:type="GST_AUDIO_DECODER_SRC_NAME"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">The name of the templates for the source pad.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_DECODER_SRC_PAD" c:identifier="GST_AUDIO_DECODER_SRC_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Gives the pointer to the source #GstPad object of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">base audio codec instance</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_STREAM_LOCK" c:identifier="GST_AUDIO_DECODER_STREAM_LOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="dec"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_DECODER_STREAM_UNLOCK" c:identifier="GST_AUDIO_DECODER_STREAM_UNLOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="dec"> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_DEF_CHANNELS" value="2" c:type="GST_AUDIO_DEF_CHANNELS"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Standard number of channels used in consumer audio.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="gint" c:type="gint"/> + </constant> + <constant name="AUDIO_DEF_FORMAT" value="S16LE" c:type="GST_AUDIO_DEF_FORMAT"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Standard format used in consumer audio.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_DEF_RATE" value="44100" c:type="GST_AUDIO_DEF_RATE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Standard sampling rate used in consumer audio.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="gint" c:type="gint"/> + </constant> + <function-macro name="AUDIO_ENCODER" c:identifier="GST_AUDIO_ENCODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_CAST" c:identifier="GST_AUDIO_ENCODER_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_CLASS" c:identifier="GST_AUDIO_ENCODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_GET_CLASS" c:identifier="GST_AUDIO_ENCODER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_INPUT_SEGMENT" c:identifier="GST_AUDIO_ENCODER_INPUT_SEGMENT" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">Gives the input segment of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">base parse instance</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_OUTPUT_SEGMENT" c:identifier="GST_AUDIO_ENCODER_OUTPUT_SEGMENT" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">Gives the output segment of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">base parse instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_ENCODER_SINK_NAME" value="sink" c:type="GST_AUDIO_ENCODER_SINK_NAME"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">the name of the templates for the sink pad</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_ENCODER_SINK_PAD" c:identifier="GST_AUDIO_ENCODER_SINK_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">Gives the pointer to the sink #GstPad object of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">audio encoder instance</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_ENCODER_SRC_NAME" value="src" c:type="GST_AUDIO_ENCODER_SRC_NAME"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">the name of the templates for the source pad</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_ENCODER_SRC_PAD" c:identifier="GST_AUDIO_ENCODER_SRC_PAD" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">Gives the pointer to the source #GstPad object of the element.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">audio encoder instance</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_STREAM_LOCK" c:identifier="GST_AUDIO_ENCODER_STREAM_LOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="enc"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_ENCODER_STREAM_UNLOCK" c:identifier="GST_AUDIO_ENCODER_STREAM_UNLOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="enc"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER" c:identifier="GST_AUDIO_FILTER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_BPF" c:identifier="GST_AUDIO_FILTER_BPF" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="filter"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_BPS" c:identifier="GST_AUDIO_FILTER_BPS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="filter"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_CAST" c:identifier="GST_AUDIO_FILTER_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_CHANNELS" c:identifier="GST_AUDIO_FILTER_CHANNELS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="filter"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_CLASS" c:identifier="GST_AUDIO_FILTER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_CLASS_CAST" c:identifier="GST_AUDIO_FILTER_CLASS_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_FORMAT" c:identifier="GST_AUDIO_FILTER_FORMAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="filter"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_GET_CLASS" c:identifier="GST_AUDIO_FILTER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_INFO" c:identifier="GST_AUDIO_FILTER_INFO" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="filter"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FILTER_RATE" c:identifier="GST_AUDIO_FILTER_RATE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">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> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="AUDIO_FORMAT_INFO_DEPTH" c:identifier="GST_AUDIO_FORMAT_INFO_DEPTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_ENDIANNESS" c:identifier="GST_AUDIO_FORMAT_INFO_ENDIANNESS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_FLAGS" c:identifier="GST_AUDIO_FORMAT_INFO_FLAGS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_FORMAT" c:identifier="GST_AUDIO_FORMAT_INFO_FORMAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_BIG_ENDIAN" c:identifier="GST_AUDIO_FORMAT_INFO_IS_BIG_ENDIAN" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_FLOAT" c:identifier="GST_AUDIO_FORMAT_INFO_IS_FLOAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_INTEGER" c:identifier="GST_AUDIO_FORMAT_INFO_IS_INTEGER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN" c:identifier="GST_AUDIO_FORMAT_INFO_IS_LITTLE_ENDIAN" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_SIGNED" c:identifier="GST_AUDIO_FORMAT_INFO_IS_SIGNED" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_IS_VALID_RAW" c:identifier="GST_AUDIO_FORMAT_INFO_IS_VALID_RAW" version="1.22" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Tests that the given #GstAudioFormatInfo represents a valid un-encoded +format.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_NAME" c:identifier="GST_AUDIO_FORMAT_INFO_NAME" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_FORMAT_INFO_WIDTH" c:identifier="GST_AUDIO_FORMAT_INFO_WIDTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_BPF" c:identifier="GST_AUDIO_INFO_BPF" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_BPS" c:identifier="GST_AUDIO_INFO_BPS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_CHANNELS" c:identifier="GST_AUDIO_INFO_CHANNELS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_DEPTH" c:identifier="GST_AUDIO_INFO_DEPTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_ENDIANNESS" c:identifier="GST_AUDIO_INFO_ENDIANNESS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_FLAGS" c:identifier="GST_AUDIO_INFO_FLAGS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_FORMAT" c:identifier="GST_AUDIO_INFO_FORMAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_BIG_ENDIAN" c:identifier="GST_AUDIO_INFO_IS_BIG_ENDIAN" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_FLOAT" c:identifier="GST_AUDIO_INFO_IS_FLOAT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_INTEGER" c:identifier="GST_AUDIO_INFO_IS_INTEGER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_LITTLE_ENDIAN" c:identifier="GST_AUDIO_INFO_IS_LITTLE_ENDIAN" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_SIGNED" c:identifier="GST_AUDIO_INFO_IS_SIGNED" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_UNPOSITIONED" c:identifier="GST_AUDIO_INFO_IS_UNPOSITIONED" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_IS_VALID" c:identifier="GST_AUDIO_INFO_IS_VALID" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_LAYOUT" c:identifier="GST_AUDIO_INFO_LAYOUT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_NAME" c:identifier="GST_AUDIO_INFO_NAME" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_POSITION" c:identifier="GST_AUDIO_INFO_POSITION" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + <parameter name="c"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_RATE" c:identifier="GST_AUDIO_INFO_RATE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="info"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_INFO_WIDTH" c:identifier="GST_AUDIO_INFO_WIDTH" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <parameters> + <parameter name="i"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_NE" c:identifier="GST_AUDIO_NE" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Turns audio format string @s into the format string for native endianness.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="s"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">format string without endianness marker</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_OE" c:identifier="GST_AUDIO_OE" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Turns audio format string @s into the format string for other endianness.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <parameters> + <parameter name="s"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">format string without endianness marker</doc> + </parameter> + </parameters> + </function-macro> + <constant name="AUDIO_RATE_RANGE" value="(int) [ 1, max ]" c:type="GST_AUDIO_RATE_RANGE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Maximum range of allowed sample rates, for use in template caps strings.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_CUBIC_B" value="GstAudioResampler.cubic-b" c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_B"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE, B parameter of the cubic filter. +Values between 0.0 and 2.0 are accepted. 1.0 is the default. + +Below are some values of popular filters: + B C +Hermite 0.0 0.0 +Spline 1.0 0.0 +Catmull-Rom 0.0 1/2</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_CUBIC_C" value="GstAudioResampler.cubic-c" c:type="GST_AUDIO_RESAMPLER_OPT_CUBIC_C"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE, C parameter of the cubic filter. +Values between 0.0 and 2.0 are accepted. 0.0 is the default. + +See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_CUTOFF" value="GstAudioResampler.cutoff" c:type="GST_AUDIO_RESAMPLER_OPT_CUTOFF"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION" value="GstAudioResampler.filter-interpolation" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be + interpolated. +GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE" value="GstAudioResampler.filter-mode" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be +constructed. +GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD" value="GstAudioResampler.filter-mode-threshold" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_UINT: the amount of memory to use for full filter tables before +switching to interpolated filter tables. +1048576 is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE" value="GstAudioResampler.filter-oversample" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_UINT, oversampling to use when interpolating filters +8 is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR" value="GstAudioResampler.max-phase-error" c:type="GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE: The maximum allowed phase error when switching sample +rates. +0.1 is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_N_TAPS" value="GstAudioResampler.n-taps" c:type="GST_AUDIO_RESAMPLER_OPT_N_TAPS"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_INT: the number of taps to use for the filter. +0 is the default and selects the taps automatically.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_STOP_ATTENUATION" value="GstAudioResampler.stop-attenutation" c:type="GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation +after the stopband for the kaiser window. 85 dB is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH" value="GstAudioResampler.transition-bandwidth" c:type="GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">G_TYPE_DOUBLE, transition bandwidth. The width of the +transition band for the kaiser window. 0.087 is the default.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="AUDIO_RESAMPLER_QUALITY_DEFAULT" value="4" c:type="GST_AUDIO_RESAMPLER_QUALITY_DEFAULT"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="gint" c:type="gint"/> + </constant> + <constant name="AUDIO_RESAMPLER_QUALITY_MAX" value="10" c:type="GST_AUDIO_RESAMPLER_QUALITY_MAX"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="gint" c:type="gint"/> + </constant> + <constant name="AUDIO_RESAMPLER_QUALITY_MIN" value="0" c:type="GST_AUDIO_RESAMPLER_QUALITY_MIN"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <type name="gint" c:type="gint"/> + </constant> + <function-macro name="AUDIO_RING_BUFFER" c:identifier="GST_AUDIO_RING_BUFFER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_BROADCAST" c:identifier="GST_AUDIO_RING_BUFFER_BROADCAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="buf"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_CAST" c:identifier="GST_AUDIO_RING_BUFFER_CAST" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_CLASS" c:identifier="GST_AUDIO_RING_BUFFER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_GET_CLASS" c:identifier="GST_AUDIO_RING_BUFFER_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_GET_COND" c:identifier="GST_AUDIO_RING_BUFFER_GET_COND" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="buf"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_SIGNAL" c:identifier="GST_AUDIO_RING_BUFFER_SIGNAL" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="buf"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_RING_BUFFER_WAIT" c:identifier="GST_AUDIO_RING_BUFFER_WAIT" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="buf"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SINK" c:identifier="GST_AUDIO_SINK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SINK_CLASS" c:identifier="GST_AUDIO_SINK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SINK_GET_CLASS" c:identifier="GST_AUDIO_SINK_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SRC" c:identifier="GST_AUDIO_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SRC_CLASS" c:identifier="GST_AUDIO_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="AUDIO_SRC_GET_CLASS" c:identifier="GST_AUDIO_SRC_GET_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <class name="AudioAggregator" c:symbol-prefix="audio_aggregator" c:type="GstAudioAggregator" version="1.14" parent="GstBase.Aggregator" abstract="1" glib:type-name="GstAudioAggregator" glib:get-type="gst_audio_aggregator_get_type" glib:type-struct="AudioAggregatorClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.c">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> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <virtual-method name="aggregate_one_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="aagg" transfer-ownership="none"> + <type name="AudioAggregator" c:type="GstAudioAggregator*"/> + </instance-parameter> + <parameter name="pad" transfer-ownership="none"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="in_offset" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="out_offset" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="num_frames" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="create_output_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="aagg" transfer-ownership="none"> + <type name="AudioAggregator" c:type="GstAudioAggregator*"/> + </instance-parameter> + <parameter name="num_frames" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </virtual-method> + <method name="set_sink_caps" c:identifier="gst_audio_aggregator_set_sink_caps"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <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="force-live" version="1.22" writable="1" construct-only="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.c">Causes the element to aggregate on a timeout even when no live source is +connected to its sinks. See #GstAggregator:min-upstream-latency for a +companion property: in the vast majority of cases where you plan to plug in +live sources with a non-zero latency, you should set it to a non-zero value.</doc> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="ignore-inactive-pads" version="1.20" writable="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.c">Don't wait for inactive pads when live. An inactive pad +is a pad that hasn't yet received a buffer, but that has +been waited on at least once. + +The purpose of this property is to avoid aggregating on +timeout when new pads are requested in advance of receiving +data flow, for example the user may decide to connect it later, +but wants to configure it already.</doc> + <type name="gboolean" c:type="gboolean"/> + </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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h">The caps set by the subclass</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioAggregatorPrivate" c:type="GstAudioAggregatorPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioAggregatorClass" c:type="GstAudioAggregatorClass" glib:is-gtype-struct-for="AudioAggregator" version="1.14"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <field name="parent_class"> + <type name="GstBase.AggregatorClass" c:type="GstAggregatorClass"/> + </field> + <field name="create_output_buffer"> + <callback name="create_output_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="aagg" transfer-ownership="none"> + <type name="AudioAggregator" c:type="GstAudioAggregator*"/> + </parameter> + <parameter name="num_frames" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="aggregate_one_buffer"> + <callback name="aggregate_one_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="aagg" transfer-ownership="none"> + <type name="AudioAggregator" c:type="GstAudioAggregator*"/> + </parameter> + <parameter name="pad" transfer-ownership="none"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="in_offset" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="out_offset" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="num_frames" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="20"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <class name="AudioAggregatorConvertPad" c:symbol-prefix="audio_aggregator_convert_pad" c:type="GstAudioAggregatorConvertPad" version="1.14" parent="AudioAggregatorPad" glib:type-name="GstAudioAggregatorConvertPad" glib:get-type="gst_audio_aggregator_convert_pad_get_type" glib:type-struct="AudioAggregatorConvertPadClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h">An implementation of GstPad that can be used with #GstAudioAggregator. + +See #GstAudioAggregator for more details.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <property name="converter-config" writable="1" transfer-ownership="none"> + <type name="Gst.Structure"/> + </property> + <field name="parent" readable="0" private="1"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioAggregatorConvertPadPrivate" c:type="GstAudioAggregatorConvertPadPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioAggregatorConvertPadClass" c:type="GstAudioAggregatorConvertPadClass" glib:is-gtype-struct-for="AudioAggregatorConvertPad" version="1.14"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <field name="parent_class"> + <type name="AudioAggregatorPadClass" c:type="GstAudioAggregatorPadClass"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioAggregatorConvertPadPrivate" c:type="GstAudioAggregatorConvertPadPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + </record> + <class name="AudioAggregatorPad" c:symbol-prefix="audio_aggregator_pad" c:type="GstAudioAggregatorPad" version="1.14" parent="GstBase.AggregatorPad" glib:type-name="GstAudioAggregatorPad" glib:get-type="gst_audio_aggregator_pad_get_type" glib:type-struct="AudioAggregatorPadClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h">The default implementation of GstPad used with #GstAudioAggregator</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <virtual-method name="convert_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="pad" transfer-ownership="none"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/> + </instance-parameter> + <parameter name="in_info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="out_info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="update_conversion_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <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> + <property name="qos-messages" version="1.20" writable="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.c">Emit QoS messages when dropping buffers.</doc> + <type name="gboolean" c:type="gboolean"/> + </property> + <field name="parent"> + <type name="GstBase.AggregatorPad" c:type="GstAggregatorPad"/> + </field> + <field name="info"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h">The audio info for this pad set from the incoming caps</doc> + <type name="AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioAggregatorPadPrivate" c:type="GstAudioAggregatorPadPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioAggregatorPadClass" c:type="GstAudioAggregatorPadClass" glib:is-gtype-struct-for="AudioAggregatorPad" version="1.14"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <field name="parent_class"> + <type name="GstBase.AggregatorPadClass" c:type="GstAggregatorPadClass"/> + </field> + <field name="convert_buffer"> + <callback name="convert_buffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="pad" transfer-ownership="none"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/> + </parameter> + <parameter name="in_info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="out_info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="update_conversion_info"> + <callback name="update_conversion_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="pad" transfer-ownership="none"> + <type name="AudioAggregatorPad" c:type="GstAudioAggregatorPad*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="20"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioAggregatorPadPrivate" c:type="GstAudioAggregatorPadPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + </record> + <record name="AudioAggregatorPrivate" c:type="GstAudioAggregatorPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + </record> + <class name="AudioBaseSink" c:symbol-prefix="audio_base_sink" c:type="GstAudioBaseSink" parent="GstBase.BaseSink" glib:type-name="GstAudioBaseSink" glib:get-type="gst_audio_base_sink_get_type" glib:type-struct="AudioBaseSinkClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">This is the base class for audio sinks. Subclasses need to implement the +::create_ringbuffer vmethod. This base class will then take care of +writing samples to the ringbuffer, synchronisation, clipping and flushing.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <virtual-method name="create_ringbuffer" invoker="create_ringbuffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Create and return the #GstAudioRingBuffer for @sink. This function will +call the ::create_ringbuffer vmethod and will set @sink as the parent of +the returned buffer (see gst_object_set_parent()).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The new ringbuffer of @sink.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink.</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="payload"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <method name="create_ringbuffer" c:identifier="gst_audio_base_sink_create_ringbuffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Create and return the #GstAudioRingBuffer for @sink. This function will +call the ::create_ringbuffer vmethod and will set @sink as the parent of +the returned buffer (see gst_object_set_parent()).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The new ringbuffer of @sink.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink.</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_alignment_threshold" c:identifier="gst_audio_base_sink_get_alignment_threshold"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Get the current alignment threshold, in nanoseconds, used by @sink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The current alignment threshold used by @sink.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_discont_wait" c:identifier="gst_audio_base_sink_get_discont_wait"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Get the current discont wait, in nanoseconds, used by @sink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The current discont wait used by @sink.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_drift_tolerance" c:identifier="gst_audio_base_sink_get_drift_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Get the current drift tolerance, in microseconds, used by @sink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The current drift tolerance used by @sink.</doc> + <type name="gint64" c:type="gint64"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_provide_clock" c:identifier="gst_audio_base_sink_get_provide_clock"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Queries whether @sink will provide a clock or not. See also +gst_audio_base_sink_set_provide_clock.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">%TRUE if @sink will provide a clock.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_slave_method" c:identifier="gst_audio_base_sink_get_slave_method"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Get the current slave method used by @sink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The current slave method used by @sink.</doc> + <type name="AudioBaseSinkSlaveMethod" c:type="GstAudioBaseSinkSlaveMethod"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="report_device_failure" c:identifier="gst_audio_base_sink_report_device_failure" version="1.6"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Informs this base class that the audio output device has failed for +some reason, causing a discontinuity (for example, because the device +recovered from the error, but lost all contents of its ring buffer). +This function is typically called by derived classes, and is useful +for the custom slave method.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + </parameters> + </method> + <method name="set_alignment_threshold" c:identifier="gst_audio_base_sink_set_alignment_threshold"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls the sink's alignment threshold.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="alignment_threshold" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">the new alignment threshold in nanoseconds</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_custom_slaving_callback" c:identifier="gst_audio_base_sink_set_custom_slaving_callback" version="1.6"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Sets the custom slaving callback. This callback will +be invoked if the slave-method property is set to +GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink +receives and plays samples. + +Setting the callback to NULL causes the sink to +behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE +method were used.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="callback" transfer-ownership="none" scope="notified" closure="1" destroy="2"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSinkCustomSlavingCallback</doc> + <type name="AudioBaseSinkCustomSlavingCallback" c:type="GstAudioBaseSinkCustomSlavingCallback"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">user data passed to the callback</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + <parameter name="notify" transfer-ownership="none" scope="async"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">called when user_data becomes unused</doc> + <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/> + </parameter> + </parameters> + </method> + <method name="set_discont_wait" c:identifier="gst_audio_base_sink_set_discont_wait"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls how long the sink will wait before creating a discontinuity.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="discont_wait" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">the new discont wait in nanoseconds</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_drift_tolerance" c:identifier="gst_audio_base_sink_set_drift_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls the sink's drift tolerance.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="drift_tolerance" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">the new drift tolerance in microseconds</doc> + <type name="gint64" c:type="gint64"/> + </parameter> + </parameters> + </method> + <method name="set_provide_clock" c:identifier="gst_audio_base_sink_set_provide_clock"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls whether @sink will provide a clock or not. If @provide is %TRUE, +gst_element_provide_clock() will return a clock that reflects the datarate +of @sink. If @provide is %FALSE, gst_element_provide_clock() will return +NULL.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="provide" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_slave_method" c:identifier="gst_audio_base_sink_set_slave_method"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls how clock slaving will be performed in @sink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </instance-parameter> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">the new slave method</doc> + <type name="AudioBaseSinkSlaveMethod" c:type="GstAudioBaseSinkSlaveMethod"/> + </parameter> + </parameters> + </method> + <property name="alignment-threshold" writable="1" transfer-ownership="none"> + <type name="guint64" c:type="guint64"/> + </property> + <property name="buffer-time" writable="1" transfer-ownership="none"> + <type name="gint64" c:type="gint64"/> + </property> + <property name="can-activate-pull" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="discont-wait" writable="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">A window of time in nanoseconds to wait before creating a discontinuity as +a result of breaching the drift-tolerance.</doc> + <type name="guint64" c:type="guint64"/> + </property> + <property name="drift-tolerance" writable="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">Controls the amount of time in microseconds that clocks are allowed +to drift before resynchronisation happens.</doc> + <type name="gint64" c:type="gint64"/> + </property> + <property name="latency-time" writable="1" transfer-ownership="none"> + <type name="gint64" c:type="gint64"/> + </property> + <property name="provide-clock" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="slave-method" writable="1" transfer-ownership="none"> + <type name="AudioBaseSinkSlaveMethod"/> + </property> + <field name="element"> + <type name="GstBase.BaseSink" c:type="GstBaseSink"/> + </field> + <field name="ringbuffer"> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </field> + <field name="buffer_time"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="latency_time"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="next_sample"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="provided_clock"> + <type name="Gst.Clock" c:type="GstClock*"/> + </field> + <field name="eos_rendering"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioBaseSinkPrivate" c:type="GstAudioBaseSinkPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioBaseSinkClass" c:type="GstAudioBaseSinkClass" glib:is-gtype-struct-for="AudioBaseSink"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">#GstAudioBaseSink class. Override the vmethod to implement +functionality.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <field name="parent_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">the parent class.</doc> + <type name="GstBase.BaseSinkClass" c:type="GstBaseSinkClass"/> + </field> + <field name="create_ringbuffer"> + <callback name="create_ringbuffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">The new ringbuffer of @sink.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.c">a #GstAudioBaseSink.</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="payload"> + <callback name="payload"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <callback name="AudioBaseSinkCustomSlavingCallback" c:type="GstAudioBaseSinkCustomSlavingCallback" version="1.6"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">This function is set with gst_audio_base_sink_set_custom_slaving_callback() +and is called during playback. It receives the current time of external and +internal clocks, which the callback can then use to apply any custom +slaving/synchronization schemes. + +The external clock is the sink's element clock, the internal one is the +internal audio clock. The internal audio clock's calibration is applied to +the timestamps before they are passed to the callback. The difference between +etime and itime is the skew; how much internal and external clock lie apart +from each other. A skew of 0 means both clocks are perfectly in sync. +itime > etime means the external clock is going slower, while itime < etime +means it is going faster than the internal clock. etime and itime are always +valid timestamps, except for when a discontinuity happens. + +requested_skew is an output value the callback can write to. It informs the +sink of whether or not it should move the playout pointer, and if so, by how +much. This pointer is only NULL if a discontinuity occurs; otherwise, it is +safe to write to *requested_skew. The default skew is 0. + +The sink may experience discontinuities. If one happens, discont is TRUE, +itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL. +This makes it possible to reset custom clock slaving algorithms when a +discontinuity happens.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">a #GstAudioBaseSink</doc> + <type name="AudioBaseSink" c:type="GstAudioBaseSink*"/> + </parameter> + <parameter name="etime" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">external clock time</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="itime" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">internal clock time</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="requested_skew" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">skew amount requested by the callback</doc> + <type name="Gst.ClockTimeDiff" c:type="GstClockTimeDiff*"/> + </parameter> + <parameter name="discont_reason" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">reason for discontinuity (if any)</doc> + <type name="AudioBaseSinkDiscontReason" c:type="GstAudioBaseSinkDiscontReason"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="5"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">user data</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + </parameters> + </callback> + <enumeration name="AudioBaseSinkDiscontReason" version="1.6" glib:type-name="GstAudioBaseSinkDiscontReason" glib:get-type="gst_audio_base_sink_discont_reason_get_type" c:type="GstAudioBaseSinkDiscontReason"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Different possible reasons for discontinuities. This enum is useful for the custom +slave method.</doc> + <member name="no_discont" value="0" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT" glib:nick="no-discont"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">No discontinuity occurred</doc> + </member> + <member name="new_caps" value="1" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS" glib:nick="new-caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">New caps are set, causing renegotiotion</doc> + </member> + <member name="flush" value="2" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH" glib:nick="flush"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Samples have been flushed</doc> + </member> + <member name="sync_latency" value="3" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY" glib:nick="sync-latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Sink was synchronized to the estimated latency (occurs during initialization)</doc> + </member> + <member name="alignment" value="4" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT" glib:nick="alignment"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Aligning buffers failed because the timestamps are too discontinuous</doc> + </member> + <member name="device_failure" value="5" c:identifier="GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE" glib:nick="device-failure"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Audio output device experienced and recovered from an error but introduced latency in the process (see also gst_audio_base_sink_report_device_failure())</doc> + </member> + </enumeration> + <record name="AudioBaseSinkPrivate" c:type="GstAudioBaseSinkPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + </record> + <enumeration name="AudioBaseSinkSlaveMethod" glib:type-name="GstAudioBaseSinkSlaveMethod" glib:get-type="gst_audio_base_sink_slave_method_get_type" c:type="GstAudioBaseSinkSlaveMethod"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Different possible clock slaving algorithms used when the internal audio +clock is not selected as the pipeline master clock.</doc> + <member name="resample" value="0" c:identifier="GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE" glib:nick="resample"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Resample to match the master clock</doc> + </member> + <member name="skew" value="1" c:identifier="GST_AUDIO_BASE_SINK_SLAVE_SKEW" glib:nick="skew"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Adjust playout pointer when master clock +drifts too much.</doc> + </member> + <member name="none" value="2" c:identifier="GST_AUDIO_BASE_SINK_SLAVE_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">No adjustment is done.</doc> + </member> + <member name="custom" value="3" c:identifier="GST_AUDIO_BASE_SINK_SLAVE_CUSTOM" glib:nick="custom"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h">Use custom clock slaving algorithm (Since: 1.6)</doc> + </member> + </enumeration> + <class name="AudioBaseSrc" c:symbol-prefix="audio_base_src" c:type="GstAudioBaseSrc" parent="GstBase.PushSrc" glib:type-name="GstAudioBaseSrc" glib:get-type="gst_audio_base_src_get_type" glib:type-struct="AudioBaseSrcClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">This is the base class for audio sources. Subclasses need to implement the +::create_ringbuffer vmethod. This base class will then take care of +reading samples from the ringbuffer, synchronisation and flushing.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <virtual-method name="create_ringbuffer" invoker="create_ringbuffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Create and return the #GstAudioRingBuffer for @src. This function will call +the ::create_ringbuffer vmethod and will set @src as the parent of the +returned buffer (see gst_object_set_parent()).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">The new ringbuffer of @src.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc.</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <method name="create_ringbuffer" c:identifier="gst_audio_base_src_create_ringbuffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Create and return the #GstAudioRingBuffer for @src. This function will call +the ::create_ringbuffer vmethod and will set @src as the parent of the +returned buffer (see gst_object_set_parent()).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">The new ringbuffer of @src.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc.</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_provide_clock" c:identifier="gst_audio_base_src_get_provide_clock"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Queries whether @src will provide a clock or not. See also +gst_audio_base_src_set_provide_clock.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">%TRUE if @src will provide a clock.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_slave_method" c:identifier="gst_audio_base_src_get_slave_method"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Get the current slave method used by @src.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">The current slave method used by @src.</doc> + <type name="AudioBaseSrcSlaveMethod" c:type="GstAudioBaseSrcSlaveMethod"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + </parameters> + </method> + <method name="set_provide_clock" c:identifier="gst_audio_base_src_set_provide_clock"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Controls whether @src will provide a clock or not. If @provide is %TRUE, +gst_element_provide_clock() will return a clock that reflects the datarate +of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + <parameter name="provide" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_slave_method" c:identifier="gst_audio_base_src_set_slave_method"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Controls how clock slaving will be performed in @src.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </instance-parameter> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">the new slave method</doc> + <type name="AudioBaseSrcSlaveMethod" c:type="GstAudioBaseSrcSlaveMethod"/> + </parameter> + </parameters> + </method> + <property name="actual-buffer-time" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Actual configured size of audio buffer in microseconds.</doc> + <type name="gint64" c:type="gint64"/> + </property> + <property name="actual-latency-time" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">Actual configured audio latency in microseconds.</doc> + <type name="gint64" c:type="gint64"/> + </property> + <property name="buffer-time" writable="1" transfer-ownership="none"> + <type name="gint64" c:type="gint64"/> + </property> + <property name="latency-time" writable="1" transfer-ownership="none"> + <type name="gint64" c:type="gint64"/> + </property> + <property name="provide-clock" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="slave-method" writable="1" transfer-ownership="none"> + <type name="AudioBaseSrcSlaveMethod"/> + </property> + <field name="element"> + <type name="GstBase.PushSrc" c:type="GstPushSrc"/> + </field> + <field name="ringbuffer"> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </field> + <field name="buffer_time"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </field> + <field name="latency_time"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </field> + <field name="next_sample"> + <type name="guint64" c:type="guint64"/> + </field> + <field name="clock"> + <type name="Gst.Clock" c:type="GstClock*"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioBaseSrcPrivate" c:type="GstAudioBaseSrcPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioBaseSrcClass" c:type="GstAudioBaseSrcClass" glib:is-gtype-struct-for="AudioBaseSrc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">#GstAudioBaseSrc class. Override the vmethod to implement +functionality.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <field name="parent_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">the parent class.</doc> + <type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/> + </field> + <field name="create_ringbuffer"> + <callback name="create_ringbuffer"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">The new ringbuffer of @src.</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.c">a #GstAudioBaseSrc.</doc> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioBaseSrcPrivate" c:type="GstAudioBaseSrcPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + </record> + <enumeration name="AudioBaseSrcSlaveMethod" glib:type-name="GstAudioBaseSrcSlaveMethod" glib:get-type="gst_audio_base_src_slave_method_get_type" c:type="GstAudioBaseSrcSlaveMethod"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Different possible clock slaving algorithms when the internal audio clock was +not selected as the pipeline clock.</doc> + <member name="resample" value="0" c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE" glib:nick="resample"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Resample to match the master clock.</doc> + </member> + <member name="re_timestamp" value="1" c:identifier="GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP" glib:nick="re-timestamp"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Retimestamp output buffers with master +clock time.</doc> + </member> + <member name="skew" value="2" c:identifier="GST_AUDIO_BASE_SRC_SLAVE_SKEW" glib:nick="skew"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">Adjust capture pointer when master clock +drifts too much.</doc> + </member> + <member name="none" value="3" c:identifier="GST_AUDIO_BASE_SRC_SLAVE_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h">No adjustment is done.</doc> + </member> + </enumeration> + <record name="AudioBuffer" c:type="GstAudioBuffer" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">A structure containing the result of an audio buffer map operation, +which is executed with gst_audio_buffer_map(). For non-interleaved (planar) +buffers, the beginning of each channel in the buffer has its own pointer in +the @planes array. For interleaved buffers, the @planes array only contains +one item, which is the pointer to the beginning of the buffer, and @n_planes +equals 1. + +The different channels in @planes are always in the GStreamer channel order.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <field name="info" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">a #GstAudioInfo describing the audio properties of this buffer</doc> + <type name="AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="n_samples" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">the size of the buffer in samples</doc> + <type name="gsize" c:type="gsize"/> + </field> + <field name="n_planes" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">the number of planes available</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="planes" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">an array of @n_planes pointers pointing to the start of each + plane in the mapped buffer</doc> + <type name="gpointer" c:type="gpointer*"/> + </field> + <field name="buffer" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h">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="unmap" c:identifier="gst_audio_buffer_unmap" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">Unmaps an audio buffer that was previously mapped with +gst_audio_buffer_map().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the #GstAudioBuffer to unmap</doc> + <type name="AudioBuffer" c:type="GstAudioBuffer*"/> + </instance-parameter> + </parameters> + </method> + <function name="clip" c:identifier="gst_audio_buffer_clip"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">Clip the buffer to the given %GstSegment. + +After calling this function the caller does not own a reference to +@buffer anymore.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">%NULL if the buffer is completely outside the configured segment, +otherwise the clipped buffer is returned. + +If the buffer has no timestamp, it is assumed to be inside the segment and +is not clipped</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">The buffer to clip.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="segment" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which + the buffer should be clipped.</doc> + <type name="Gst.Segment" c:type="const GstSegment*"/> + </parameter> + <parameter name="rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">sample rate.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="bpf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">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="map" c:identifier="gst_audio_buffer_map" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">Maps an audio @gstbuffer so that it can be read or written and stores the +result of the map operation in @buffer. + +This is especially useful when the @gstbuffer is in non-interleaved (planar) +layout, in which case this function will use the information in the +@gstbuffer's attached #GstAudioMeta in order to map each channel in a +separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached +on the @gstbuffer, then it must be in interleaved layout. + +If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked +against @info. Normally, they should be equal, but in case they are not, +a g_critical will be printed and the #GstAudioInfo from the meta will be +used. + +In non-interleaved buffers, it is possible to have each channel on a separate +#GstMemory. In this case, each memory will be mapped separately to avoid +copying their contents in a larger memory area. Do note though that it is +not supported to have a single channel spanning over two or more different +#GstMemory objects. Although the map operation will likely succeed in this +case, it will be highly sub-optimal and it is recommended to merge all the +memories in the buffer before calling this function. + +Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid +as long as it's mapped.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">%TRUE if the map operation succeeded or %FALSE on failure</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buffer" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">pointer to a #GstAudioBuffer</doc> + <type name="AudioBuffer" c:type="GstAudioBuffer*"/> + </parameter> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the audio properties of the buffer</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + <parameter name="gstbuffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the #GstBuffer to be mapped</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the access mode for the memory</doc> + <type name="Gst.MapFlags" c:type="GstMapFlags"/> + </parameter> + </parameters> + </function> + <function name="reorder_channels" c:identifier="gst_audio_buffer_reorder_channels"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Reorders @buffer from the channel positions @from to the channel +positions @to. @from and @to must contain the same number of +positions and the same positions, only in a different order. +@buffer must be writable.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the reordering was possible.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The buffer to reorder.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioFormat of the buffer.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions in the buffer.</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to convert to.</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </function> + <function name="truncate" c:identifier="gst_audio_buffer_truncate" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">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. + +This function does not know the audio rate, therefore the caller is +responsible for re-setting the correct timestamp and duration to the +buffer. However, timestamp will be preserved if trim == 0, and duration +will also be preserved if there is no trimming to be done. Offset and +offset end will be preserved / updated. + +After calling this function the caller does not own a reference to +@buffer anymore.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the truncated buffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">The buffer to truncate.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="bpf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">size of one audio frame in bytes. This is the size of one sample * +number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="trim" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the number of samples to remove from the beginning of the buffer</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the final number of samples that should exist in this buffer or -1 +to use all the remaining samples if you are only removing samples from the +beginning.</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </function> + </record> + <class name="AudioCdSrc" c:symbol-prefix="audio_cd_src" c:type="GstAudioCdSrc" parent="GstBase.PushSrc" glib:type-name="GstAudioCdSrc" glib:get-type="gst_audio_cd_src_get_type" glib:type-struct="AudioCdSrcClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.c">Provides a base class for CD digital audio (CDDA) sources, which handles +things like seeking, querying, discid calculation, tags, and buffer +timestamping. + +## Using GstAudioCdSrc-based elements in applications + +GstAudioCdSrc registers two #GstFormat<!-- -->s of its own, namely +the "track" format and the "sector" format. Applications will usually +only find the "track" format interesting. You can retrieve that #GstFormat +for use in seek events or queries with gst_format_get_by_nick("track"). + +In order to query the number of tracks, for example, an application would +set the CDDA source element to READY or PAUSED state and then query the +the number of tracks via gst_element_query_duration() using the track +format acquired above. Applications can query the currently playing track +in the same way. + +Alternatively, applications may retrieve the currently playing track and +the total number of tracks from the taglist that will posted on the bus +whenever the CD is opened or the currently playing track changes. The +taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags. + +Applications playing back CD audio using playbin and cdda://n URIs should +issue a seek command in track format to change between tracks, rather than +setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin +involves closing and re-opening the CD device, which is much much slower). + +## Tags and meta-information + +CDDA sources will automatically emit a number of tags, details about which +can be found in the libgsttag documentation. Those tags are: +#GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL, +#GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL, +among others. + +## Tracks and Table of Contents (TOC) + +Applications will be informed of the available tracks via a TOC message +on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for +each track, with information about each track. The duration for each +track can be retrieved via the #GST_TAG_DURATION tag from each entry's +tag list, or calculated via gst_toc_entry_get_start_stop_times(). +The track entries in the TOC will be sorted by track number.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <implements name="Gst.URIHandler"/> + <virtual-method name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </instance-parameter> + <parameter name="device" transfer-ownership="none"> + <type name="utf8" c:type="const gchar*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="read_sector"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </instance-parameter> + <parameter name="sector" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </virtual-method> + <method name="add_track" c:identifier="gst_audio_cd_src_add_track"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.c">CDDA sources use this function from their start vfunc to announce the +available data and audio tracks to the base source class. The caller +should allocate @track on the stack, the base source will do a shallow +copy of the structure (and take ownership of the taglist if there is one).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.c">FALSE on error, otherwise TRUE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.c">a #GstAudioCdSrc</doc> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </instance-parameter> + <parameter name="track" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.c">address of #GstAudioCdSrcTrack to add</doc> + <type name="AudioCdSrcTrack" c:type="GstAudioCdSrcTrack*"/> + </parameter> + </parameters> + </method> + <property name="device" writable="1" transfer-ownership="none"> + <type name="utf8" c:type="gchar*"/> + </property> + <property name="mode" writable="1" transfer-ownership="none"> + <type name="AudioCdSrcMode"/> + </property> + <property name="track" writable="1" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </property> + <field name="pushsrc"> + <type name="GstBase.PushSrc" c:type="GstPushSrc"/> + </field> + <field name="tags"> + <type name="Gst.TagList" c:type="GstTagList*"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioCdSrcPrivate" c:type="GstAudioCdSrcPrivate*"/> + </field> + <field name="_gst_reserved1" readable="0" private="1"> + <array zero-terminated="0" fixed-size="2"> + <type name="guint" c:type="guint"/> + </array> + </field> + <field name="_gst_reserved2" readable="0" private="1"> + <array zero-terminated="0" fixed-size="2"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioCdSrcClass" c:type="GstAudioCdSrcClass" glib:is-gtype-struct-for="AudioCdSrc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">Audio CD source base class.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <field name="pushsrc_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">the parent class</doc> + <type name="GstBase.PushSrcClass" c:type="GstPushSrcClass"/> + </field> + <field name="open"> + <callback name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </parameter> + <parameter name="device" transfer-ownership="none"> + <type name="utf8" c:type="const gchar*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close"> + <callback name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="read_sector"> + <callback name="read_sector"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioCdSrc" c:type="GstAudioCdSrc*"/> + </parameter> + <parameter name="sector" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="20"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <enumeration name="AudioCdSrcMode" glib:type-name="GstAudioCdSrcMode" glib:get-type="gst_audio_cd_src_mode_get_type" c:type="GstAudioCdSrcMode"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">Mode in which the CD audio source operates. Influences timestamping, +EOS handling and seeking.</doc> + <member name="normal" value="0" c:identifier="GST_AUDIO_CD_SRC_MODE_NORMAL" glib:nick="normal"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">each single track is a stream</doc> + </member> + <member name="continuous" value="1" c:identifier="GST_AUDIO_CD_SRC_MODE_CONTINUOUS" glib:nick="continuous"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">the entire disc is a single stream</doc> + </member> + </enumeration> + <record name="AudioCdSrcPrivate" c:type="GstAudioCdSrcPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + </record> + <record name="AudioCdSrcTrack" c:type="GstAudioCdSrcTrack"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">CD track abstraction to communicate TOC entries to the base class. + +This structure is only for use by sub-classed in connection with +gst_audio_cd_src_add_track(). + +Applications will be informed of the available tracks via a TOC message +on the pipeline's #GstBus instead.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <field name="is_audio" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">Whether this is an audio track</doc> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="num" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">Track number in TOC (usually starts from 1, but not always)</doc> + <type name="guint" c:type="guint"/> + </field> + <field name="start" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">The first sector of this track (LBA)</doc> + <type name="guint" c:type="guint"/> + </field> + <field name="end" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">The last sector of this track (LBA)</doc> + <type name="guint" c:type="guint"/> + </field> + <field name="tags" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h">Track-specific tags (e.g. from cd-text information), or NULL</doc> + <type name="Gst.TagList" c:type="GstTagList*"/> + </field> + <field name="_gst_reserved1" readable="0" private="1"> + <array zero-terminated="0" fixed-size="2"> + <type name="guint" c:type="guint"/> + </array> + </field> + <field name="_gst_reserved2" readable="0" private="1"> + <array zero-terminated="0" fixed-size="2"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioChannelMixer" c:type="GstAudioChannelMixer" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <method name="free" c:identifier="gst_audio_channel_mixer_free"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Free memory allocated by @mix.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="mix" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a #GstAudioChannelMixer</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </instance-parameter> + </parameters> + </method> + <method name="is_passthrough" c:identifier="gst_audio_channel_mixer_is_passthrough"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Check if @mix is in passthrough. + +Only N x N mix identity matrices are considered passthrough, +this is determined by comparing the contents of the matrix +with 0.0 and 1.0. + +As this is floating point comparisons, if the values have been +generated, they should be rounded up or down by explicit +assignment of 0.0 or 1.0 to values within a user-defined +epsilon, this code doesn't make assumptions as to what may +constitute an appropriate epsilon.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">%TRUE is @mix is passthrough.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="mix" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a #GstAudioChannelMixer</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </instance-parameter> + </parameters> + </method> + <method name="samples" c:identifier="gst_audio_channel_mixer_samples"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">In case the samples are interleaved, @in and @out must point to an +array with a single element pointing to a block of interleaved samples. + +If non-interleaved samples are used, @in and @out must point to an +array with pointers to memory blocks, one for each channel. + +Perform channel mixing on @in_data and write the result to @out_data. +@in_data and @out_data need to be in @format and @layout.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="mix" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a #GstAudioChannelMixer</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </instance-parameter> + <parameter name="in" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">input samples</doc> + <type name="gpointer" c:type="const gpointer*"/> + </parameter> + <parameter name="out" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">output samples</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of samples</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <function name="new" c:identifier="gst_audio_channel_mixer_new" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Create a new channel mixer object for the given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a new #GstAudioChannelMixer object. + Free with gst_audio_channel_mixer_free() after usage.</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </return-value> + <parameters> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">#GstAudioChannelMixerFlags</doc> + <type name="AudioChannelMixerFlags" c:type="GstAudioChannelMixerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="in_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of input channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="in_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">positions of input channels</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </parameter> + <parameter name="out_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of output channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">positions of output channels</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </parameter> + </parameters> + </function> + <function name="new_with_matrix" c:identifier="gst_audio_channel_mixer_new_with_matrix" version="1.14" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Create a new channel mixer object for the given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a new #GstAudioChannelMixer object. + Free with gst_audio_channel_mixer_free() after usage.</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </return-value> + <parameters> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">#GstAudioChannelMixerFlags</doc> + <type name="AudioChannelMixerFlags" c:type="GstAudioChannelMixerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="in_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of input channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of output channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="matrix" transfer-ownership="full" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">channel conversion matrix, m[@in_channels][@out_channels]. + If identity matrix, passthrough applies. If %NULL, a (potentially truncated) + identity matrix is generated.</doc> + <type name="gfloat" c:type="gfloat**"/> + </parameter> + </parameters> + </function> + </record> + <bitfield name="AudioChannelMixerFlags" glib:type-name="GstAudioChannelMixerFlags" glib:get-type="gst_audio_channel_mixer_flags_get_type" c:type="GstAudioChannelMixerFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">Flags passed to gst_audio_channel_mixer_new()</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">no flag</doc> + </member> + <member name="non_interleaved_in" value="1" c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_IN" glib:nick="non-interleaved-in"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">input channels are not interleaved</doc> + </member> + <member name="non_interleaved_out" value="2" c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_NON_INTERLEAVED_OUT" glib:nick="non-interleaved-out"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">output channels are not interleaved</doc> + </member> + <member name="unpositioned_in" value="4" c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_IN" glib:nick="unpositioned-in"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">input channels are explicitly unpositioned</doc> + </member> + <member name="unpositioned_out" value="8" c:identifier="GST_AUDIO_CHANNEL_MIXER_FLAGS_UNPOSITIONED_OUT" glib:nick="unpositioned-out"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h">output channels are explicitly unpositioned</doc> + </member> + </bitfield> + <enumeration name="AudioChannelPosition" glib:type-name="GstAudioChannelPosition" glib:get-type="gst_audio_channel_position_get_type" c:type="GstAudioChannelPosition"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">used for position-less channels, e.g. + from a sound card that records 1024 channels; mutually exclusive with + any other channel position</doc> + </member> + <member name="mono" value="-2" c:identifier="GST_AUDIO_CHANNEL_POSITION_MONO" glib:nick="mono"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Mono without direction; + can only be used with 1 channel</doc> + </member> + <member name="invalid" value="-1" c:identifier="GST_AUDIO_CHANNEL_POSITION_INVALID" glib:nick="invalid"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">invalid position</doc> + </member> + <member name="front_left" value="0" c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT" glib:nick="front-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Front left</doc> + </member> + <member name="front_right" value="1" c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT" glib:nick="front-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Front right</doc> + </member> + <member name="front_center" value="2" c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER" glib:nick="front-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Front center</doc> + </member> + <member name="lfe1" value="3" c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE1" glib:nick="lfe1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Low-frequency effects 1 (subwoofer)</doc> + </member> + <member name="rear_left" value="4" c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_LEFT" glib:nick="rear-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Rear left</doc> + </member> + <member name="rear_right" value="5" c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT" glib:nick="rear-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Rear right</doc> + </member> + <member name="front_left_of_center" value="6" c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER" glib:nick="front-left-of-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Front left of center</doc> + </member> + <member name="front_right_of_center" value="7" c:identifier="GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER" glib:nick="front-right-of-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Front right of center</doc> + </member> + <member name="rear_center" value="8" c:identifier="GST_AUDIO_CHANNEL_POSITION_REAR_CENTER" glib:nick="rear-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Rear center</doc> + </member> + <member name="lfe2" value="9" c:identifier="GST_AUDIO_CHANNEL_POSITION_LFE2" glib:nick="lfe2"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Low-frequency effects 2 (subwoofer)</doc> + </member> + <member name="side_left" value="10" c:identifier="GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT" glib:nick="side-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Side left</doc> + </member> + <member name="side_right" value="11" c:identifier="GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT" glib:nick="side-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Side right</doc> + </member> + <member name="top_front_left" value="12" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT" glib:nick="top-front-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top front left</doc> + </member> + <member name="top_front_right" value="13" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT" glib:nick="top-front-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top front right</doc> + </member> + <member name="top_front_center" value="14" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER" glib:nick="top-front-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top front center</doc> + </member> + <member name="top_center" value="15" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_CENTER" glib:nick="top-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top center</doc> + </member> + <member name="top_rear_left" value="16" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT" glib:nick="top-rear-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top rear left</doc> + </member> + <member name="top_rear_right" value="17" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT" glib:nick="top-rear-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top rear right</doc> + </member> + <member name="top_side_left" value="18" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT" glib:nick="top-side-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top side right</doc> + </member> + <member name="top_side_right" value="19" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT" glib:nick="top-side-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top rear right</doc> + </member> + <member name="top_rear_center" value="20" c:identifier="GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER" glib:nick="top-rear-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Top rear center</doc> + </member> + <member name="bottom_front_center" value="21" c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER" glib:nick="bottom-front-center"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Bottom front center</doc> + </member> + <member name="bottom_front_left" value="22" c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT" glib:nick="bottom-front-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Bottom front left</doc> + </member> + <member name="bottom_front_right" value="23" c:identifier="GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT" glib:nick="bottom-front-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Bottom front right</doc> + </member> + <member name="wide_left" value="24" c:identifier="GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT" glib:nick="wide-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Wide left (between front left and side left)</doc> + </member> + <member name="wide_right" value="25" c:identifier="GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT" glib:nick="wide-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Wide right (between front right and side right)</doc> + </member> + <member name="surround_left" value="26" c:identifier="GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT" glib:nick="surround-left"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Surround left (between rear left and side left)</doc> + </member> + <member name="surround_right" value="27" c:identifier="GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT" glib:nick="surround-right"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h">Surround right (between rear right and side right)</doc> + </member> + </enumeration> + <record name="AudioClippingMeta" c:type="GstAudioClippingMeta" version="1.8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">Extra buffer metadata describing how much audio has to be clipped from +the start or end of a buffer. This is used for compressed formats, where +the first frame usually has some additional samples due to encoder and +decoder delays, and the last frame usually has some additional samples to +be able to fill the complete last frame. + +This is used to ensure that decoded data in the end has the same amount of +samples, and multiply decoded streams can be gaplessly concatenated. + +Note: If clipping of the start is done by adjusting the segment, this meta +has to be dropped from buffers as otherwise clipping could happen twice.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <field name="meta" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">parent #GstMeta</doc> + <type name="Gst.Meta" c:type="GstMeta"/> + </field> + <field name="format" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </field> + <field name="start" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">Amount of audio to clip from start of buffer</doc> + <type name="guint64" c:type="guint64"/> + </field> + <field name="end" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">Amount of to clip from end of buffer</doc> + <type name="guint64" c:type="guint64"/> + </field> + <function name="get_info" c:identifier="gst_audio_clipping_meta_get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + </record> + <class name="AudioClock" c:symbol-prefix="audio_clock" c:type="GstAudioClock" parent="Gst.SystemClock" glib:type-name="GstAudioClock" glib:get-type="gst_audio_clock_get_type" glib:type-struct="AudioClockClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">#GstAudioClock makes it easy for elements to implement a #GstClock, they +simply need to provide a function that returns the current clock time. + +This object is internally used to implement the clock in #GstAudioBaseSink.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <constructor name="new" c:identifier="gst_audio_clock_new"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">Create a new #GstAudioClock instance. Whenever the clock time should be +calculated it will call @func with @user_data. When @func returns +#GST_CLOCK_TIME_NONE, the clock will return the last reported time.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a new #GstAudioClock casted to a #GstClock.</doc> + <type name="Gst.Clock" c:type="GstClock*"/> + </return-value> + <parameters> + <parameter name="name" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">the name of the clock</doc> + <type name="utf8" c:type="const gchar*"/> + </parameter> + <parameter name="func" transfer-ownership="none" scope="notified" closure="2" destroy="3"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a function</doc> + <type name="AudioClockGetTimeFunc" c:type="GstAudioClockGetTimeFunc"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">user data</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + <parameter name="destroy_notify" transfer-ownership="none" scope="async"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">#GDestroyNotify for @user_data</doc> + <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/> + </parameter> + </parameters> + </constructor> + <method name="adjust" c:identifier="gst_audio_clock_adjust"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">Adjust @time with the internal offset of the audio clock.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">@time adjusted with the internal offset.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="clock" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstAudioClock</doc> + <type name="AudioClock" c:type="GstAudioClock*"/> + </instance-parameter> + <parameter name="time" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstClockTime</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="get_time" c:identifier="gst_audio_clock_get_time"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">Report the time as returned by the #GstAudioClockGetTimeFunc without applying +any offsets.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">the time as reported by the time function of the audio clock</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="clock" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstAudioClock</doc> + <type name="AudioClock" c:type="GstAudioClock*"/> + </instance-parameter> + </parameters> + </method> + <method name="invalidate" c:identifier="gst_audio_clock_invalidate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">Invalidate the clock function. Call this function when the provided +#GstAudioClockGetTimeFunc cannot be called anymore, for example, when the +user_data becomes invalid. + +After calling this function, @clock will return the last returned time for +the rest of its lifetime.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="clock" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstAudioClock</doc> + <type name="AudioClock" c:type="GstAudioClock*"/> + </instance-parameter> + </parameters> + </method> + <method name="reset" c:identifier="gst_audio_clock_reset"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values +starting from @time. The clock will update an internal offset to make sure that +future calls to internal_time will return an increasing result as required by +the #GstClock object.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="clock" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstAudioClock</doc> + <type name="AudioClock" c:type="GstAudioClock*"/> + </instance-parameter> + <parameter name="time" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.c">a #GstClockTime</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <field name="clock"> + <type name="Gst.SystemClock" c:type="GstSystemClock"/> + </field> + <field name="func"> + <type name="AudioClockGetTimeFunc" c:type="GstAudioClockGetTimeFunc"/> + </field> + <field name="user_data"> + <type name="gpointer" c:type="gpointer"/> + </field> + <field name="destroy_notify"> + <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/> + </field> + <field name="last_time" readable="0" private="1"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </field> + <field name="time_offset" readable="0" private="1"> + <type name="Gst.ClockTimeDiff" c:type="GstClockTimeDiff"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioClockClass" c:type="GstAudioClockClass" glib:is-gtype-struct-for="AudioClock"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <field name="parent_class"> + <type name="Gst.SystemClockClass" c:type="GstSystemClockClass"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <callback name="AudioClockGetTimeFunc" c:type="GstAudioClockGetTimeFunc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h">This function will be called whenever the current clock time needs to be +calculated. If this function returns #GST_CLOCK_TIME_NONE, the last reported +time will be returned by the clock.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h">the current time or #GST_CLOCK_TIME_NONE if the previous time should +be used.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <parameter name="clock" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h">the #GstAudioClock</doc> + <type name="Gst.Clock" c:type="GstClock*"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">This object is used to convert audio samples from one format to another. +The object can perform conversion of: + + * audio format with optional dithering and noise shaping + + * audio samplerate + + * audio channels and channel layout</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <constructor name="new" c:identifier="gst_audio_converter_new"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Create a new #GstAudioConverter that is able to convert between @in and @out +audio formats. + +@config contains extra configuration options, see `GST_AUDIO_CONVERTER_OPT_*` +parameters for details about the options and values.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter or %NULL if conversion is not possible.</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </return-value> + <parameters> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">extra #GstAudioConverterFlags</doc> + <type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/> + </parameter> + <parameter name="in_info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a source #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="out_info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a destination #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="config" transfer-ownership="full" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstStructure with configuration options</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </constructor> + <method name="convert" c:identifier="gst_audio_converter_convert" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Convenience wrapper around gst_audio_converter_samples(), which will +perform allocation of the output buffer based on the result from +gst_audio_converter_get_out_frames().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">%TRUE is the conversion could be performed.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">extra #GstAudioConverterFlags</doc> + <type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/> + </parameter> + <parameter name="in" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">input data</doc> + <array length="2" zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="in_size" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">size of @in</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="out" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a pointer where + the output data will be written</doc> + <array length="4" zero-terminated="0" c:type="gpointer*"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="out_size" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a pointer where the size of @out will be written</doc> + <type name="gsize" c:type="gsize*"/> + </parameter> + </parameters> + </method> + <method name="free" c:identifier="gst_audio_converter_free"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Free a previously allocated @convert instance.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_config" c:identifier="gst_audio_converter_get_config"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Get the current configuration of @convert.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c"> + a #GstStructure that remains valid for as long as @convert is valid + or until gst_audio_converter_update_config() is called.</doc> + <type name="Gst.Structure" c:type="const GstStructure*"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="in_rate" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">result input rate</doc> + <type name="gint" c:type="gint*"/> + </parameter> + <parameter name="out_rate" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">result output rate</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </method> + <method name="get_in_frames" c:identifier="gst_audio_converter_get_in_frames"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Calculate how many input frames are currently needed by @convert to produce +@out_frames of output frames.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">the number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="out_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">number of output frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="get_max_latency" c:identifier="gst_audio_converter_get_max_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Get the maximum number of input frames that the converter would +need before producing output.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">the latency of @convert as expressed in the number of +frames.</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_out_frames" c:identifier="gst_audio_converter_get_out_frames"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Calculate how many output frames can be produced when @in_frames input +frames are given to @convert.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">the number of output frames</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="in_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="is_passthrough" c:identifier="gst_audio_converter_is_passthrough" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Returns whether the audio converter will operate in passthrough mode. +The return value would be typically input to gst_base_transform_set_passthrough()</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">%TRUE when no conversion will actually occur.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + </parameters> + </method> + <method name="reset" c:identifier="gst_audio_converter_reset"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Reset @convert to the state it was when it was first created, clearing +any history it might currently have.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + </parameters> + </method> + <method name="samples" c:identifier="gst_audio_converter_samples"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Perform the conversion with @in_frames in @in to @out_frames in @out +using @convert. + +In case the samples are interleaved, @in and @out must point to an +array with a single element pointing to a block of interleaved samples. + +If non-interleaved samples are used, @in and @out must point to an +array with pointers to memory blocks, one for each channel. + +@in may be %NULL, in which case @in_frames of silence samples are processed +by the converter. + +This function always produces @out_frames of output and consumes @in_frames of +input. Use gst_audio_converter_get_out_frames() and +gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames +are matching and @in and @out point to enough memory.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">%TRUE is the conversion could be performed.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">extra #GstAudioConverterFlags</doc> + <type name="AudioConverterFlags" c:type="GstAudioConverterFlags"/> + </parameter> + <parameter name="in" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">input frames</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="in_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="out" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">output frames</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="out_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Returns whether the audio converter can perform the conversion in-place. +The return value would be typically input to gst_base_transform_set_in_place()</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">%TRUE when the conversion can be done in place.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + </parameters> + </method> + <method name="update_config" c:identifier="gst_audio_converter_update_config"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">Set @in_rate, @out_rate and @config as extra configuration for @convert. + +@in_rate and @out_rate specify the new sample rates of input and output +formats. A value of 0 leaves the sample rate unchanged. + +@config can be %NULL, in which case, the current configuration is not +changed. + +If the parameters in @config can not be set exactly, this function returns +%FALSE and will try to update as much state as possible. The new state can +then be retrieved and refined with gst_audio_converter_get_config(). + +Look at the `GST_AUDIO_CONVERTER_OPT_*` fields to check valid configuration +option and values.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">%TRUE when the new parameters could be set</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="convert" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstAudioConverter</doc> + <type name="AudioConverter" c:type="GstAudioConverter*"/> + </instance-parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="config" transfer-ownership="full" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.c">a #GstStructure or %NULL</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </method> + </record> + <bitfield name="AudioConverterFlags" glib:type-name="GstAudioConverterFlags" glib:get-type="gst_audio_converter_flags_get_type" c:type="GstAudioConverterFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_CONVERTER_FLAG_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">no flag</doc> + </member> + <member name="in_writable" value="1" c:identifier="GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE" glib:nick="in-writable"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">the input sample arrays are writable and can be + used as temporary storage during conversion.</doc> + </member> + <member name="variable_rate" value="2" c:identifier="GST_AUDIO_CONVERTER_FLAG_VARIABLE_RATE" glib:nick="variable-rate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-converter.h">allow arbitrary rate updates with + gst_audio_converter_update_config().</doc> + </member> + </bitfield> + <class name="AudioDecoder" c:symbol-prefix="audio_decoder" c:type="GstAudioDecoder" parent="Gst.Element" abstract="1" glib:type-name="GstAudioDecoder" glib:get-type="gst_audio_decoder_get_type" glib:type-struct="AudioDecoderClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">This base class is for audio decoders turning encoded data into +raw audio samples. + +GstAudioDecoder and subclass should cooperate as follows. + +## Configuration + + * Initially, GstAudioDecoder calls @start when the decoder element + is activated, which allows subclass to perform any global setup. + Base class (context) parameters can already be set according to subclass + capabilities (or possibly upon receive more information in subsequent + @set_format). + * GstAudioDecoder calls @set_format to inform subclass of the format + of input audio data that it is about to receive. + While unlikely, it might be called more than once, if changing input + parameters require reconfiguration. + * GstAudioDecoder calls @stop at end of all processing. + +As of configuration stage, and throughout processing, GstAudioDecoder +provides various (context) parameters, e.g. describing the format of +output audio data (valid when output caps have been set) or current parsing state. +Conversely, subclass can and should configure context to inform +base class of its expectation w.r.t. buffer handling. + +## Data processing + * Base class gathers input data, and optionally allows subclass + to parse this into subsequently manageable (as defined by subclass) + chunks. Such chunks are subsequently referred to as 'frames', + though they may or may not correspond to 1 (or more) audio format frame. + * Input frame is provided to subclass' @handle_frame. + * If codec processing results in decoded data, subclass should call + @gst_audio_decoder_finish_frame to have decoded data pushed + downstream. + * Just prior to actually pushing a buffer downstream, + it is passed to @pre_push. Subclass should either use this callback + to arrange for additional downstream pushing or otherwise ensure such + custom pushing occurs after at least a method call has finished since + setting src pad caps. + * During the parsing process GstAudioDecoderClass will handle both + srcpad and sinkpad events. Sink events will be passed to subclass + if @event callback has been provided. + +## Shutdown phase + + * GstAudioDecoder class calls @stop to inform the subclass that data + parsing will be stopped. + +Subclass is responsible for providing pad template caps for +source and sink pads. The pads need to be named "sink" and "src". It also +needs to set the fixed caps on srcpad, when the format is ensured. This +is typically when base class calls subclass' @set_format function, though +it might be delayed until calling @gst_audio_decoder_finish_frame. + +In summary, above process should have subclass concentrating on +codec data processing while leaving other matters to base class, +such as most notably timestamp handling. While it may exert more control +in this area (see e.g. @pre_push), it is very much not recommended. + +In particular, base class will try to arrange for perfect output timestamps +as much as possible while tracking upstream timestamps. +To this end, if deviation between the next ideal expected perfect timestamp +and upstream exceeds #GstAudioDecoder:tolerance, then resync to upstream +occurs (which would happen always if the tolerance mechanism is disabled). + +In non-live pipelines, baseclass can also (configurably) arrange for +output buffer aggregation which may help to redue large(r) numbers of +small(er) buffers being pushed and processed downstream. Note that this +feature is only available if the buffer layout is interleaved. For planar +buffers, the decoder implementation is fully responsible for the output +buffer size. + +On the other hand, it should be noted that baseclass only provides limited +seeking support (upon explicit subclass request), as full-fledged support +should rather be left to upstream demuxer, parser or alike. This simple +approach caters for seeking and duration reporting using estimated input +bitrates. + +Things that subclass need to take care of: + + * Provide pad templates + * Set source pad caps when appropriate + * Set user-configurable properties to sane defaults for format and + implementing codec at hand, and convey some subclass capabilities and + expectations in context. + + * Accept data in @handle_frame and provide encoded results to + @gst_audio_decoder_finish_frame. If it is prepared to perform + PLC, it should also accept NULL data in @handle_frame and provide for + data for indicated duration.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <virtual-method name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="flush"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="hard" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="getcaps"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="filter" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="handle_frame"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="negotiate" invoker="negotiate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Negotiate with downstream elements to currently configured #GstAudioInfo. +Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if +negotiate fails.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="parse"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <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" direction="out" caller-allocates="0" transfer-ownership="full"> + <type name="gint" c:type="gint*"/> + </parameter> + <parameter name="length" direction="out" caller-allocates="0" transfer-ownership="full"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="pre_push"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_format"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="caps" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="sink_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="sink_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="src_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="src_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="start"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="stop"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="transform_meta"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="meta" transfer-ownership="none"> + <type name="Gst.Meta" c:type="GstMeta*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <method name="allocate_output_buffer" c:identifier="gst_audio_decoder_allocate_output_buffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Helper function that allocates a buffer to hold an audio frame +for @dec's current output format.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">allocated buffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="size" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">size of the buffer</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="finish_frame" c:identifier="gst_audio_decoder_finish_frame"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Collects decoded data and pushes it downstream. + +@buf may be NULL in which case the indicated number of frames +are discarded and considered to have produced no output +(e.g. lead-in or setup frames). +Otherwise, source pad caps must be set when it is called with valid +data in @buf. + +Note that a frame received in #GstAudioDecoderClass.handle_frame() may be +invalidated by a call to this function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstFlowReturn that should be escalated to caller (of caller)</doc> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">decoded data</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">number of decoded frames represented by decoded data</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="finish_subframe" c:identifier="gst_audio_decoder_finish_subframe" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Collects decoded data and pushes it downstream. This function may be called +multiple times for a given input frame. + +@buf may be NULL in which case it is assumed that the current input frame is +finished. This is equivalent to calling gst_audio_decoder_finish_subframe() +with a NULL buffer and frames=1 after having pushed out all decoded audio +subframes using this function. + +When called with valid data in @buf the source pad caps must have been set +already. + +Note that a frame received in #GstAudioDecoderClass.handle_frame() may be +invalidated by a call to this function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstFlowReturn that should be escalated to caller (of caller)</doc> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">decoded data</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </method> + <method name="get_allocator" c:identifier="gst_audio_decoder_get_allocator"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Lets #GstAudioDecoder sub-classes to know the memory @allocator +used by the base class and its @params. + +Unref the @allocator after use it.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="allocator" direction="out" caller-allocates="0" transfer-ownership="full" nullable="1" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">the #GstAllocator +used</doc> + <type name="Gst.Allocator" c:type="GstAllocator**"/> + </parameter> + <parameter name="params" direction="out" caller-allocates="1" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">the +#GstAllocationParams of @allocator</doc> + <type name="Gst.AllocationParams" c:type="GstAllocationParams*"/> + </parameter> + </parameters> + </method> + <method name="get_audio_info" c:identifier="gst_audio_decoder_get_audio_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioInfo describing the input audio format</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_delay" c:identifier="gst_audio_decoder_get_delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">currently configured decoder delay</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_drainable" c:identifier="gst_audio_decoder_get_drainable"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Queries decoder drain handling.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">TRUE if drainable handling is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_estimate_rate" c:identifier="gst_audio_decoder_get_estimate_rate"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">currently configured byte to time conversion setting</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_latency" c:identifier="gst_audio_decoder_get_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets the variables pointed to by @min and @max to the currently configured +latency.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="min" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a pointer to storage to hold minimum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="max" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a pointer to storage to hold maximum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </method> + <method name="get_max_errors" c:identifier="gst_audio_decoder_get_max_errors"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">currently configured decoder tolerated error count.</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_min_latency" c:identifier="gst_audio_decoder_get_min_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Queries decoder's latency aggregation.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">aggregation latency. + +MT safe.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_needs_format" c:identifier="gst_audio_decoder_get_needs_format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Queries decoder required format handling.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">TRUE if required format handling is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_parse_state" c:identifier="gst_audio_decoder_get_parse_state"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Return current parsing (sync and eos) state.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="sync" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a pointer to a variable to hold the current sync state</doc> + <type name="gboolean" c:type="gboolean*"/> + </parameter> + <parameter name="eos" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a pointer to a variable to hold the current eos state</doc> + <type name="gboolean" c:type="gboolean*"/> + </parameter> + </parameters> + </method> + <method name="get_plc" c:identifier="gst_audio_decoder_get_plc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Queries decoder packet loss concealment handling.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">TRUE if packet loss concealment is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_plc_aware" c:identifier="gst_audio_decoder_get_plc_aware"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">currently configured plc handling</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_tolerance" c:identifier="gst_audio_decoder_get_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Queries current audio jitter tolerance threshold.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">decoder audio jitter tolerance threshold. + +MT safe.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="merge_tags" c:identifier="gst_audio_decoder_merge_tags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets the audio decoder tags and how they should be merged with any +upstream stream tags. This will override any tags previously-set +with gst_audio_decoder_merge_tags(). + +Note that this is provided for convenience, and the subclass is +not required to use this and can still do tag handling on its own.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="tags" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstTagList to merge, or NULL</doc> + <type name="Gst.TagList" c:type="const GstTagList*"/> + </parameter> + <parameter name="mode" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE</doc> + <type name="Gst.TagMergeMode" c:type="GstTagMergeMode"/> + </parameter> + </parameters> + </method> + <method name="negotiate" c:identifier="gst_audio_decoder_negotiate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Negotiate with downstream elements to currently configured #GstAudioInfo. +Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if +negotiate fails.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="proxy_getcaps" c:identifier="gst_audio_decoder_proxy_getcaps" version="1.6"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Returns caps that express @caps (or sink template caps if @caps == NULL) +restricted to rate/channels/... combinations supported by downstream +elements.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstCaps owned by caller</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <instance-parameter name="decoder" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="caps" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">initial caps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + <parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">filter caps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_allocation_caps" c:identifier="gst_audio_decoder_set_allocation_caps" version="1.10"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets a caps in allocation query which are different from the set +pad's caps. Use this function before calling +gst_audio_decoder_negotiate(). Setting to %NULL the allocation +query will use the caps from the pad.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="allocation_caps" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstCaps or %NULL</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_drainable" c:identifier="gst_audio_decoder_set_drainable"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Configures decoder drain handling. If drainable, subclass might +be handed a NULL buffer to have it return any leftover decoded data. +Otherwise, it is not considered so capable and will only ever be passed +real data. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_estimate_rate" c:identifier="gst_audio_decoder_set_estimate_rate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Allows baseclass to perform byte to time estimated conversion.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">whether to enable byte to time conversion</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_latency" c:identifier="gst_audio_decoder_set_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets decoder latency. If the provided values changed from +previously provided ones, this will also post a LATENCY message on the bus +so the pipeline can reconfigure its global latency.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="min" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">minimum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="max" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">maximum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_max_errors" c:identifier="gst_audio_decoder_set_max_errors"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets numbers of tolerated decoder errors, where a tolerated one is then only +warned about, but more than tolerated will lead to fatal error. You can set +-1 for never returning fatal errors. Default is set to +GST_AUDIO_DECODER_MAX_ERRORS.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">max tolerated errors</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="set_min_latency" c:identifier="gst_audio_decoder_set_min_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Sets decoder minimum aggregation latency. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new minimum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_needs_format" c:identifier="gst_audio_decoder_set_needs_format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Configures decoder format needs. If enabled, subclass needs to be +negotiated with format caps before it can process any data. It will then +never be handed any data before it has been configured. +Otherwise, it might be handed data without having been configured and +is then expected being able to do so either by default +or based on the input data. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_output_caps" c:identifier="gst_audio_decoder_set_output_caps" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Configure output caps on the srcpad of @dec. Similar to +gst_audio_decoder_set_output_format(), but allows subclasses to specify +output caps that can't be expressed via #GstAudioInfo e.g. caps that have +caps features.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">%TRUE on success.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">(fixed) #GstCaps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_output_format" c:identifier="gst_audio_decoder_set_output_format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Configure output info on the srcpad of @dec.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">%TRUE on success.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">#GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </method> + <method name="set_plc" c:identifier="gst_audio_decoder_set_plc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Enable or disable decoder packet loss concealment, provided subclass +and codec are capable and allow handling plc. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_plc_aware" c:identifier="gst_audio_decoder_set_plc_aware"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Indicates whether or not subclass handles packet loss concealment (plc).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="plc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new plc state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_tolerance" c:identifier="gst_audio_decoder_set_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Configures decoder audio jitter tolerance threshold. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="tolerance" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">new tolerance</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_use_default_pad_acceptcaps" c:identifier="gst_audio_decoder_set_use_default_pad_acceptcaps" version="1.6"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">Lets #GstAudioDecoder sub-classes decide if they want the sink pad +to use the default pad query handler to reply to accept-caps queries. + +By setting this to true it is possible to further customize the default +handler with %GST_PAD_SET_ACCEPT_INTERSECT and +%GST_PAD_SET_ACCEPT_TEMPLATE</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="decoder" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </instance-parameter> + <parameter name="use" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">Subclasses can override any of the available virtual methods or not, as +needed. At minimum @handle_frame (and likely @set_format) needs to be +overridden.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <field name="element_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h">The parent class structure</doc> + <type name="Gst.ElementClass" c:type="GstElementClass"/> + </field> + <field name="start"> + <callback name="start"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="stop"> + <callback name="stop"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_format"> + <callback name="set_format"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="caps" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="parse"> + <callback name="parse"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <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" direction="out" caller-allocates="0" transfer-ownership="full"> + <type name="gint" c:type="gint*"/> + </parameter> + <parameter name="length" direction="out" caller-allocates="0" transfer-ownership="full"> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="handle_frame"> + <callback name="handle_frame"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="flush"> + <callback name="flush"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="hard" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </callback> + </field> + <field name="pre_push"> + <callback name="pre_push"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + </parameters> + </callback> + </field> + <field name="sink_event"> + <callback name="sink_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="src_event"> + <callback name="src_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="open"> + <callback name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close"> + <callback name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="negotiate"> + <callback name="negotiate"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c">a #GstAudioDecoder</doc> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="decide_allocation"> + <callback name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="propose_allocation"> + <callback name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="sink_query"> + <callback name="sink_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="src_query"> + <callback name="src_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="getcaps"> + <callback name="getcaps"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <parameter name="dec" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="filter" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="transform_meta"> + <callback name="transform_meta"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioDecoder" c:type="GstAudioDecoder*"/> + </parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="meta" transfer-ownership="none"> + <type name="Gst.Meta" c:type="GstMeta*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="16"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioDecoderPrivate" c:type="GstAudioDecoderPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + </record> + <enumeration name="AudioDitherMethod" glib:type-name="GstAudioDitherMethod" glib:get-type="gst_audio_dither_method_get_type" c:type="GstAudioDitherMethod"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Set of available dithering methods.</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_DITHER_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">No dithering</doc> + </member> + <member name="rpdf" value="1" c:identifier="GST_AUDIO_DITHER_RPDF" glib:nick="rpdf"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Rectangular dithering</doc> + </member> + <member name="tpdf" value="2" c:identifier="GST_AUDIO_DITHER_TPDF" glib:nick="tpdf"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Triangular dithering (default)</doc> + </member> + <member name="tpdf_hf" value="3" c:identifier="GST_AUDIO_DITHER_TPDF_HF" glib:nick="tpdf-hf"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">High frequency triangular dithering</doc> + </member> + </enumeration> + <record name="AudioDownmixMeta" c:type="GstAudioDownmixMeta"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">Extra buffer metadata describing audio downmixing matrix. This metadata is +attached to audio buffers and contains a matrix to downmix the buffer number +of channels to @channels. + +@matrix is an two-dimensional array of @to_channels times @from_channels +coefficients, i.e. the i-th output channels is constructed by multiplicating +the input channels with the coefficients in @matrix[i] and taking the sum +of the results.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <field name="meta" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">parent #GstMeta</doc> + <type name="Gst.Meta" c:type="GstMeta"/> + </field> + <field name="from_position" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the channel positions of the source</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </field> + <field name="to_position" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the channel positions of the destination</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </field> + <field name="from_channels" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the number of channels of the source</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="to_channels" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the number of channels of the destination</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="matrix" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the matrix coefficients.</doc> + <type name="gfloat" c:type="gfloat**"/> + </field> + <function name="get_info" c:identifier="gst_audio_downmix_meta_get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + </record> + <class name="AudioEncoder" c:symbol-prefix="audio_encoder" c:type="GstAudioEncoder" parent="Gst.Element" abstract="1" glib:type-name="GstAudioEncoder" glib:get-type="gst_audio_encoder_get_type" glib:type-struct="AudioEncoderClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">This base class is for audio encoders turning raw audio samples into +encoded audio data. + +GstAudioEncoder and subclass should cooperate as follows. + +## Configuration + + * Initially, GstAudioEncoder calls @start when the encoder element + is activated, which allows subclass to perform any global setup. + + * GstAudioEncoder calls @set_format to inform subclass of the format + of input audio data that it is about to receive. Subclass should + setup for encoding and configure various base class parameters + appropriately, notably those directing desired input data handling. + While unlikely, it might be called more than once, if changing input + parameters require reconfiguration. + + * GstAudioEncoder calls @stop at end of all processing. + +As of configuration stage, and throughout processing, GstAudioEncoder +maintains various parameters that provide required context, +e.g. describing the format of input audio data. +Conversely, subclass can and should configure these context parameters +to inform base class of its expectation w.r.t. buffer handling. + +## Data processing + + * Base class gathers input sample data (as directed by the context's + frame_samples and frame_max) and provides this to subclass' @handle_frame. + * If codec processing results in encoded data, subclass should call + gst_audio_encoder_finish_frame() to have encoded data pushed + downstream. Alternatively, it might also call + gst_audio_encoder_finish_frame() (with a NULL buffer and some number of + dropped samples) to indicate dropped (non-encoded) samples. + * Just prior to actually pushing a buffer downstream, + it is passed to @pre_push. + * During the parsing process GstAudioEncoderClass will handle both + srcpad and sinkpad events. Sink events will be passed to subclass + if @event callback has been provided. + +## Shutdown phase + + * GstAudioEncoder class calls @stop to inform the subclass that data + parsing will be stopped. + +Subclass is responsible for providing pad template caps for +source and sink pads. The pads need to be named "sink" and "src". It also +needs to set the fixed caps on srcpad, when the format is ensured. This +is typically when base class calls subclass' @set_format function, though +it might be delayed until calling @gst_audio_encoder_finish_frame. + +In summary, above process should have subclass concentrating on +codec data processing while leaving other matters to base class, +such as most notably timestamp handling. While it may exert more control +in this area (see e.g. @pre_push), it is very much not recommended. + +In particular, base class will either favor tracking upstream timestamps +(at the possible expense of jitter) or aim to arrange for a perfect stream of +output timestamps, depending on #GstAudioEncoder:perfect-timestamp. +However, in the latter case, the input may not be so perfect or ideal, which +is handled as follows. An input timestamp is compared with the expected +timestamp as dictated by input sample stream and if the deviation is less +than #GstAudioEncoder:tolerance, the deviation is discarded. +Otherwise, it is considered a discontuinity and subsequent output timestamp +is resynced to the new position after performing configured discontinuity +processing. In the non-perfect-timestamp case, an upstream variation +exceeding tolerance only leads to marking DISCONT on subsequent outgoing +(while timestamps are adjusted to upstream regardless of variation). +While DISCONT is also marked in the perfect-timestamp case, this one +optionally (see #GstAudioEncoder:hard-resync) +performs some additional steps, such as clipping of (early) input samples +or draining all currently remaining input data, depending on the direction +of the discontuinity. + +If perfect timestamps are arranged, it is also possible to request baseclass +(usually set by subclass) to provide additional buffer metadata (in OFFSET +and OFFSET_END) fields according to granule defined semantics currently +needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count +including buffer) and OFFSET_END to corresponding timestamp (as determined +by same sample count and sample rate). + +Things that subclass need to take care of: + + * Provide pad templates + * Set source pad caps when appropriate + * Inform base class of buffer processing needs using context's + frame_samples and frame_bytes. + * Set user-configurable properties to sane defaults for format and + implementing codec at hand, e.g. those controlling timestamp behaviour + and discontinuity processing. + * Accept data in @handle_frame and provide encoded results to + gst_audio_encoder_finish_frame().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <implements name="Gst.Preset"/> + <virtual-method name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="flush"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="getcaps"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="filter" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="handle_frame"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="negotiate" invoker="negotiate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Negotiate with downstream elements to currently configured #GstCaps. +Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if +negotiate fails.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="pre_push"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="set_format"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="sink_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="sink_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="encoder" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="src_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="src_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="encoder" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="start"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="stop"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="transform_meta"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="meta" transfer-ownership="none"> + <type name="Gst.Meta" c:type="GstMeta*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </virtual-method> + <method name="allocate_output_buffer" c:identifier="gst_audio_encoder_allocate_output_buffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Helper function that allocates a buffer to hold an encoded audio frame +for @enc's current output format.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">allocated buffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="size" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">size of the buffer</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="finish_frame" c:identifier="gst_audio_encoder_finish_frame"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Collects encoded data and pushes encoded data downstream. +Source pad caps must be set when this is called. + +If @samples < 0, then best estimate is all samples provided to encoder +(subclass) so far. @buf may be NULL, in which case next number of @samples +are considered discarded, e.g. as a result of discontinuous transmission, +and a discontinuity is marked. + +Note that samples received in #GstAudioEncoderClass.handle_frame() +may be invalidated by a call to this function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstFlowReturn that should be escalated to caller (of caller)</doc> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">encoded data</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">number of samples (per channel) represented by encoded data</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="get_allocator" c:identifier="gst_audio_encoder_get_allocator"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Lets #GstAudioEncoder sub-classes to know the memory @allocator +used by the base class and its @params. + +Unref the @allocator after use it.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="allocator" direction="out" caller-allocates="0" transfer-ownership="full" nullable="1" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">the #GstAllocator +used</doc> + <type name="Gst.Allocator" c:type="GstAllocator**"/> + </parameter> + <parameter name="params" direction="out" caller-allocates="1" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">the +#GstAllocationParams of @allocator</doc> + <type name="Gst.AllocationParams" c:type="GstAllocationParams*"/> + </parameter> + </parameters> + </method> + <method name="get_audio_info" c:identifier="gst_audio_encoder_get_audio_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioInfo describing the input audio format</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_drainable" c:identifier="gst_audio_encoder_get_drainable"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Queries encoder drain handling.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">TRUE if drainable handling is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_frame_max" c:identifier="gst_audio_encoder_get_frame_max"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">currently configured maximum handled frames</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_frame_samples_max" c:identifier="gst_audio_encoder_get_frame_samples_max"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">currently maximum requested samples per frame</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_frame_samples_min" c:identifier="gst_audio_encoder_get_frame_samples_min"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">currently minimum requested samples per frame</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_hard_min" c:identifier="gst_audio_encoder_get_hard_min"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Queries encoder hard minimum handling.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">TRUE if hard minimum handling is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_hard_resync" c:identifier="gst_audio_encoder_get_hard_resync"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_latency" c:identifier="gst_audio_encoder_get_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets the variables pointed to by @min and @max to the currently configured +latency.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="min" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a pointer to storage to hold minimum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="max" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a pointer to storage to hold maximum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </method> + <method name="get_lookahead" c:identifier="gst_audio_encoder_get_lookahead"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">currently configured encoder lookahead</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_mark_granule" c:identifier="gst_audio_encoder_get_mark_granule"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Queries if the encoder will handle granule marking.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">TRUE if granule marking is enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_perfect_timestamp" c:identifier="gst_audio_encoder_get_perfect_timestamp"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Queries encoder perfect timestamp behaviour.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">TRUE if perfect timestamp setting enabled. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_tolerance" c:identifier="gst_audio_encoder_get_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Queries current audio jitter tolerance threshold.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">encoder audio jitter tolerance threshold. + +MT safe.</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="merge_tags" c:identifier="gst_audio_encoder_merge_tags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets the audio encoder tags and how they should be merged with any +upstream stream tags. This will override any tags previously-set +with gst_audio_encoder_merge_tags(). + +Note that this is provided for convenience, and the subclass is +not required to use this and can still do tag handling on its own. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="tags" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstTagList to merge, or NULL to unset + previously-set tags</doc> + <type name="Gst.TagList" c:type="const GstTagList*"/> + </parameter> + <parameter name="mode" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE</doc> + <type name="Gst.TagMergeMode" c:type="GstTagMergeMode"/> + </parameter> + </parameters> + </method> + <method name="negotiate" c:identifier="gst_audio_encoder_negotiate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Negotiate with downstream elements to currently configured #GstCaps. +Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if +negotiate fails.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + </parameters> + </method> + <method name="proxy_getcaps" c:identifier="gst_audio_encoder_proxy_getcaps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Returns caps that express @caps (or sink template caps if @caps == NULL) +restricted to channel/rate combinations supported by downstream elements +(e.g. muxers).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstCaps owned by caller</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="caps" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">initial caps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + <parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">filter caps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_allocation_caps" c:identifier="gst_audio_encoder_set_allocation_caps" version="1.10"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets a caps in allocation query which are different from the set +pad's caps. Use this function before calling +gst_audio_encoder_negotiate(). Setting to %NULL the allocation +query will use the caps from the pad.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="allocation_caps" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstCaps or %NULL</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_drainable" c:identifier="gst_audio_encoder_set_drainable"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Configures encoder drain handling. If drainable, subclass might +be handed a NULL buffer to have it return any leftover encoded data. +Otherwise, it is not considered so capable and will only ever be passed +real data. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_frame_max" c:identifier="gst_audio_encoder_set_frame_max"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets max number of frames accepted at once (assumed minimally 1). +Requires @frame_samples_min and @frame_samples_max to be the equal. + +Note: This value will be reset to 0 every time before +#GstAudioEncoderClass.set_format() is called.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">number of frames</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="set_frame_samples_max" c:identifier="gst_audio_encoder_set_frame_samples_max"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets number of samples (per channel) subclass needs to be handed, +at most or will be handed all available if 0. + +If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min() +must be called with the same number. + +Note: This value will be reset to 0 every time before +#GstAudioEncoderClass.set_format() is called.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">number of samples per frame</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="set_frame_samples_min" c:identifier="gst_audio_encoder_set_frame_samples_min"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets number of samples (per channel) subclass needs to be handed, +at least or will be handed all available if 0. + +If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max() +must be called with the same number. + +Note: This value will be reset to 0 every time before +#GstAudioEncoderClass.set_format() is called.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">number of samples per frame</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="set_hard_min" c:identifier="gst_audio_encoder_set_hard_min"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Configures encoder hard minimum handling. If enabled, subclass +will never be handed less samples than it configured, which otherwise +might occur near end-of-data handling. Instead, the leftover samples +will simply be discarded. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_hard_resync" c:identifier="gst_audio_encoder_set_hard_resync"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_headers" c:identifier="gst_audio_encoder_set_headers"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Set the codec headers to be sent downstream whenever requested.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="headers" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a list of + #GstBuffer containing the codec header</doc> + <type name="GLib.List" c:type="GList*"> + <type name="Gst.Buffer"/> + </type> + </parameter> + </parameters> + </method> + <method name="set_latency" c:identifier="gst_audio_encoder_set_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets encoder latency. If the provided values changed from +previously provided ones, this will also post a LATENCY message on the bus +so the pipeline can reconfigure its global latency.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="min" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">minimum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="max" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">maximum latency</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_lookahead" c:identifier="gst_audio_encoder_set_lookahead"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Sets encoder lookahead (in units of input rate samples) + +Note: This value will be reset to 0 every time before +#GstAudioEncoderClass.set_format() is called.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="num" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">lookahead</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="set_mark_granule" c:identifier="gst_audio_encoder_set_mark_granule"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Enable or disable encoder granule handling. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_output_format" c:identifier="gst_audio_encoder_set_output_format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Configure output caps on the srcpad of @enc.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">%TRUE on success.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">#GstCaps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + <method name="set_perfect_timestamp" c:identifier="gst_audio_encoder_set_perfect_timestamp"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Enable or disable encoder perfect output timestamp preference. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="enabled" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">new state</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_tolerance" c:identifier="gst_audio_encoder_set_tolerance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">Configures encoder audio jitter tolerance threshold. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </instance-parameter> + <parameter name="tolerance" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">new tolerance</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <property name="hard-resync" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="mark-granule" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="perfect-timestamp" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="tolerance" writable="1" transfer-ownership="none"> + <type name="gint64" c:type="gint64"/> + </property> + <field name="element"> + <type name="Gst.Element" c:type="GstElement"/> + </field> + <field name="sinkpad"> + <type name="Gst.Pad" c:type="GstPad*"/> + </field> + <field name="srcpad"> + <type name="Gst.Pad" c:type="GstPad*"/> + </field> + <field name="stream_lock"> + <type name="GLib.RecMutex" c:type="GRecMutex"/> + </field> + <field name="input_segment"> + <type name="Gst.Segment" c:type="GstSegment"/> + </field> + <field name="output_segment"> + <type name="Gst.Segment" c:type="GstSegment"/> + </field> + <field name="priv" readable="0" private="1"> + <type name="AudioEncoderPrivate" c:type="GstAudioEncoderPrivate*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="20"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioEncoderClass" c:type="GstAudioEncoderClass" glib:is-gtype-struct-for="AudioEncoder"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">Subclasses can override any of the available virtual methods or not, as +needed. At minimum @set_format and @handle_frame needs to be overridden.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <field name="element_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h">The parent class structure</doc> + <type name="Gst.ElementClass" c:type="GstElementClass"/> + </field> + <field name="start"> + <callback name="start"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="stop"> + <callback name="stop"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="set_format"> + <callback name="set_format"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="info" transfer-ownership="none"> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="handle_frame"> + <callback name="handle_frame"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="flush"> + <callback name="flush"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="pre_push"> + <callback name="pre_push"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.FlowReturn" c:type="GstFlowReturn"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="buffer" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer**"/> + </parameter> + </parameters> + </callback> + </field> + <field name="sink_event"> + <callback name="sink_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="src_event"> + <callback name="src_event"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="event" transfer-ownership="none"> + <type name="Gst.Event" c:type="GstEvent*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="getcaps"> + <callback name="getcaps"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="full"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="filter" transfer-ownership="none"> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="open"> + <callback name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close"> + <callback name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="negotiate"> + <callback name="negotiate"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">%TRUE if the negotiation succeeded, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c">a #GstAudioEncoder</doc> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="decide_allocation"> + <callback name="decide_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="propose_allocation"> + <callback name="propose_allocation"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="transform_meta"> + <callback name="transform_meta"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="enc" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="outbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="meta" transfer-ownership="none"> + <type name="Gst.Meta" c:type="GstMeta*"/> + </parameter> + <parameter name="inbuf" transfer-ownership="none"> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="sink_query"> + <callback name="sink_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="encoder" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="src_query"> + <callback name="src_query"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="encoder" transfer-ownership="none"> + <type name="AudioEncoder" c:type="GstAudioEncoder*"/> + </parameter> + <parameter name="query" transfer-ownership="none"> + <type name="Gst.Query" c:type="GstQuery*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="17"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioEncoderPrivate" c:type="GstAudioEncoderPrivate" disguised="1"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + </record> + <class name="AudioFilter" c:symbol-prefix="audio_filter" c:type="GstAudioFilter" parent="GstBase.BaseTransform" abstract="1" glib:type-name="GstAudioFilter" glib:get-type="gst_audio_filter_get_type" glib:type-struct="AudioFilterClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.c">#GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio +filters, ie. those that output the same format that they get as input. + +#GstAudioFilter will parse the input format for you (with error checking) +before calling your setup function. Also, elements deriving from +#GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from +their class_init function to easily configure the set of caps/formats that +the element is able to handle. + +Derived classes should override the #GstAudioFilterClass.setup() and +#GstBaseTransformClass.transform_ip() and/or +#GstBaseTransformClass.transform() +virtual functions in their class_init function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <virtual-method name="setup"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="filter" transfer-ownership="none"> + <type name="AudioFilter" c:type="GstAudioFilter*"/> + </instance-parameter> + <parameter name="info" transfer-ownership="none"> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </virtual-method> + <field name="basetransform"> + <type name="GstBase.BaseTransform" c:type="GstBaseTransform"/> + </field> + <field name="info"> + <type name="AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioFilterClass" c:type="GstAudioFilterClass" glib:is-gtype-struct-for="AudioFilter"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h">In addition to the @setup virtual function, you should also override the +GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual +function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <field name="basetransformclass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h">parent class</doc> + <type name="GstBase.BaseTransformClass" c:type="GstBaseTransformClass"/> + </field> + <field name="setup"> + <callback name="setup"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="filter" transfer-ownership="none"> + <type name="AudioFilter" c:type="GstAudioFilter*"/> + </parameter> + <parameter name="info" transfer-ownership="none"> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + <method name="add_pad_templates" c:identifier="gst_audio_filter_class_add_pad_templates"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.c">Convenience function to add pad templates to this element class, with +@allowed_caps as the caps that can be handled. + +This function is usually used from within a GObject class_init function.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="klass" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.c">an #GstAudioFilterClass</doc> + <type name="AudioFilterClass" c:type="GstAudioFilterClass*"/> + </instance-parameter> + <parameter name="allowed_caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.c">what formats the filter can handle, as #GstCaps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </method> + </record> + <bitfield name="AudioFlags" glib:type-name="GstAudioFlags" glib:get-type="gst_audio_flags_get_type" c:type="GstAudioFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">Extra audio flags</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_FLAG_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">no valid flag</doc> + </member> + <member name="unpositioned" value="1" c:identifier="GST_AUDIO_FLAG_UNPOSITIONED" glib:nick="unpositioned"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the position array explicitly + contains unpositioned channels.</doc> + </member> + </bitfield> + <enumeration name="AudioFormat" glib:type-name="GstAudioFormat" glib:get-type="gst_audio_format_get_type" c:type="GstAudioFormat"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Enum value describing the most common audio formats.</doc> + <member name="unknown" value="0" c:identifier="GST_AUDIO_FORMAT_UNKNOWN" glib:nick="unknown"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">unknown or unset audio format</doc> + </member> + <member name="encoded" value="1" c:identifier="GST_AUDIO_FORMAT_ENCODED" glib:nick="encoded"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">encoded audio format</doc> + </member> + <member name="s8" value="2" c:identifier="GST_AUDIO_FORMAT_S8" glib:nick="s8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">8 bits in 8 bits, signed</doc> + </member> + <member name="u8" value="3" c:identifier="GST_AUDIO_FORMAT_U8" glib:nick="u8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">8 bits in 8 bits, unsigned</doc> + </member> + <member name="s16le" value="4" c:identifier="GST_AUDIO_FORMAT_S16LE" glib:nick="s16le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, signed, little endian</doc> + </member> + <member name="s16be" value="5" c:identifier="GST_AUDIO_FORMAT_S16BE" glib:nick="s16be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, signed, big endian</doc> + </member> + <member name="u16le" value="6" c:identifier="GST_AUDIO_FORMAT_U16LE" glib:nick="u16le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, unsigned, little endian</doc> + </member> + <member name="u16be" value="7" c:identifier="GST_AUDIO_FORMAT_U16BE" glib:nick="u16be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, unsigned, big endian</doc> + </member> + <member name="s24_32le" value="8" c:identifier="GST_AUDIO_FORMAT_S24_32LE" glib:nick="s24-32le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, signed, little endian</doc> + </member> + <member name="s24_32be" value="9" c:identifier="GST_AUDIO_FORMAT_S24_32BE" glib:nick="s24-32be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, signed, big endian</doc> + </member> + <member name="u24_32le" value="10" c:identifier="GST_AUDIO_FORMAT_U24_32LE" glib:nick="u24-32le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, unsigned, little endian</doc> + </member> + <member name="u24_32be" value="11" c:identifier="GST_AUDIO_FORMAT_U24_32BE" glib:nick="u24-32be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, unsigned, big endian</doc> + </member> + <member name="s32le" value="12" c:identifier="GST_AUDIO_FORMAT_S32LE" glib:nick="s32le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, signed, little endian</doc> + </member> + <member name="s32be" value="13" c:identifier="GST_AUDIO_FORMAT_S32BE" glib:nick="s32be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, signed, big endian</doc> + </member> + <member name="u32le" value="14" c:identifier="GST_AUDIO_FORMAT_U32LE" glib:nick="u32le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, unsigned, little endian</doc> + </member> + <member name="u32be" value="15" c:identifier="GST_AUDIO_FORMAT_U32BE" glib:nick="u32be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, unsigned, big endian</doc> + </member> + <member name="s24le" value="16" c:identifier="GST_AUDIO_FORMAT_S24LE" glib:nick="s24le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, signed, little endian</doc> + </member> + <member name="s24be" value="17" c:identifier="GST_AUDIO_FORMAT_S24BE" glib:nick="s24be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, signed, big endian</doc> + </member> + <member name="u24le" value="18" c:identifier="GST_AUDIO_FORMAT_U24LE" glib:nick="u24le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, unsigned, little endian</doc> + </member> + <member name="u24be" value="19" c:identifier="GST_AUDIO_FORMAT_U24BE" glib:nick="u24be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, unsigned, big endian</doc> + </member> + <member name="s20le" value="20" c:identifier="GST_AUDIO_FORMAT_S20LE" glib:nick="s20le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, signed, little endian</doc> + </member> + <member name="s20be" value="21" c:identifier="GST_AUDIO_FORMAT_S20BE" glib:nick="s20be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, signed, big endian</doc> + </member> + <member name="u20le" value="22" c:identifier="GST_AUDIO_FORMAT_U20LE" glib:nick="u20le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, unsigned, little endian</doc> + </member> + <member name="u20be" value="23" c:identifier="GST_AUDIO_FORMAT_U20BE" glib:nick="u20be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, unsigned, big endian</doc> + </member> + <member name="s18le" value="24" c:identifier="GST_AUDIO_FORMAT_S18LE" glib:nick="s18le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, signed, little endian</doc> + </member> + <member name="s18be" value="25" c:identifier="GST_AUDIO_FORMAT_S18BE" glib:nick="s18be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, signed, big endian</doc> + </member> + <member name="u18le" value="26" c:identifier="GST_AUDIO_FORMAT_U18LE" glib:nick="u18le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, unsigned, little endian</doc> + </member> + <member name="u18be" value="27" c:identifier="GST_AUDIO_FORMAT_U18BE" glib:nick="u18be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, unsigned, big endian</doc> + </member> + <member name="f32le" value="28" c:identifier="GST_AUDIO_FORMAT_F32LE" glib:nick="f32le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32-bit floating point samples, little endian</doc> + </member> + <member name="f32be" value="29" c:identifier="GST_AUDIO_FORMAT_F32BE" glib:nick="f32be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32-bit floating point samples, big endian</doc> + </member> + <member name="f64le" value="30" c:identifier="GST_AUDIO_FORMAT_F64LE" glib:nick="f64le"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">64-bit floating point samples, little endian</doc> + </member> + <member name="f64be" value="31" c:identifier="GST_AUDIO_FORMAT_F64BE" glib:nick="f64be"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">64-bit floating point samples, big endian</doc> + </member> + <member name="s16" value="4" c:identifier="GST_AUDIO_FORMAT_S16" glib:nick="s16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, signed, native endianness</doc> + </member> + <member name="u16" value="6" c:identifier="GST_AUDIO_FORMAT_U16" glib:nick="u16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">16 bits in 16 bits, unsigned, native endianness</doc> + </member> + <member name="s24_32" value="8" c:identifier="GST_AUDIO_FORMAT_S24_32" glib:nick="s24-32"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, signed, native endianness</doc> + </member> + <member name="u24_32" value="10" c:identifier="GST_AUDIO_FORMAT_U24_32" glib:nick="u24-32"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 32 bits, unsigned, native endianness</doc> + </member> + <member name="s32" value="12" c:identifier="GST_AUDIO_FORMAT_S32" glib:nick="s32"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, signed, native endianness</doc> + </member> + <member name="u32" value="14" c:identifier="GST_AUDIO_FORMAT_U32" glib:nick="u32"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32 bits in 32 bits, unsigned, native endianness</doc> + </member> + <member name="s24" value="16" c:identifier="GST_AUDIO_FORMAT_S24" glib:nick="s24"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, signed, native endianness</doc> + </member> + <member name="u24" value="18" c:identifier="GST_AUDIO_FORMAT_U24" glib:nick="u24"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">24 bits in 24 bits, unsigned, native endianness</doc> + </member> + <member name="s20" value="20" c:identifier="GST_AUDIO_FORMAT_S20" glib:nick="s20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, signed, native endianness</doc> + </member> + <member name="u20" value="22" c:identifier="GST_AUDIO_FORMAT_U20" glib:nick="u20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">20 bits in 24 bits, unsigned, native endianness</doc> + </member> + <member name="s18" value="24" c:identifier="GST_AUDIO_FORMAT_S18" glib:nick="s18"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, signed, native endianness</doc> + </member> + <member name="u18" value="26" c:identifier="GST_AUDIO_FORMAT_U18" glib:nick="u18"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">18 bits in 24 bits, unsigned, native endianness</doc> + </member> + <member name="f32" value="28" c:identifier="GST_AUDIO_FORMAT_F32" glib:nick="f32"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">32-bit floating point samples, native endianness</doc> + </member> + <member name="f64" value="30" c:identifier="GST_AUDIO_FORMAT_F64" glib:nick="f64"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">64-bit floating point samples, native endianness</doc> + </member> + <function name="build_integer" c:identifier="gst_audio_format_build_integer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Construct a #GstAudioFormat with given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format +exists with the given parameters.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </return-value> + <parameters> + <parameter name="sign" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">signed or unsigned format</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + <parameter name="endianness" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">G_LITTLE_ENDIAN or G_BIG_ENDIAN</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="width" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">amount of bits used per sample</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="depth" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" deprecated="1" deprecated-version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Fill @length bytes in @dest with silence samples for @info.</doc> + <doc-deprecated xml:space="preserve">Use gst_audio_format_info_fill_silence() instead.</doc-deprecated> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormatInfo</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </parameter> + <parameter name="dest" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a destination + to fill</doc> + <array length="2" zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the length to fill</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </function> + <function name="from_string" c:identifier="gst_audio_format_from_string"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Convert the @format string to its #GstAudioFormat.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the +string is not a known format.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </return-value> + <parameters> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a format string</doc> + <type name="utf8" c:type="const gchar*"/> + </parameter> + </parameters> + </function> + <function name="get_info" c:identifier="gst_audio_format_get_info"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Get the #GstAudioFormatInfo for @format</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">The #GstAudioFormatInfo for @format.</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </return-value> + <parameters> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormat</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + </parameters> + </function> + <function name="to_string" c:identifier="gst_audio_format_to_string"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="utf8" c:type="const gchar*"/> + </return-value> + <parameters> + <parameter name="format" transfer-ownership="none"> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + </parameters> + </function> + </enumeration> + <bitfield name="AudioFormatFlags" glib:type-name="GstAudioFormatFlags" glib:get-type="gst_audio_format_flags_get_type" c:type="GstAudioFormatFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">The different audio flags that a format info can have.</doc> + <member name="integer" value="1" c:identifier="GST_AUDIO_FORMAT_FLAG_INTEGER" glib:nick="integer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">integer samples</doc> + </member> + <member name="float" value="2" c:identifier="GST_AUDIO_FORMAT_FLAG_FLOAT" glib:nick="float"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">float samples</doc> + </member> + <member name="signed" value="4" c:identifier="GST_AUDIO_FORMAT_FLAG_SIGNED" glib:nick="signed"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">signed samples</doc> + </member> + <member name="complex" value="16" c:identifier="GST_AUDIO_FORMAT_FLAG_COMPLEX" glib:nick="complex"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">complex layout</doc> + </member> + <member name="unpack" value="32" c:identifier="GST_AUDIO_FORMAT_FLAG_UNPACK" glib:nick="unpack"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">the format can be used in +#GstAudioFormatUnpack and #GstAudioFormatPack functions</doc> + </member> + </bitfield> + <record name="AudioFormatInfo" c:type="GstAudioFormatInfo"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Information for an audio format.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <field name="format" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">#GstAudioFormat</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </field> + <field name="name" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">string representation of the format</doc> + <type name="utf8" c:type="const gchar*"/> + </field> + <field name="description" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">user readable description of the format</doc> + <type name="utf8" c:type="const gchar*"/> + </field> + <field name="flags" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">#GstAudioFormatFlags</doc> + <type name="AudioFormatFlags" c:type="GstAudioFormatFlags"/> + </field> + <field name="endianness" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">the endianness</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="width" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">amount of bits used for one sample</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="depth" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">amount of valid bits in @width</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="silence" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">@width/8 bytes with 1 silent sample</doc> + <array zero-terminated="0" fixed-size="8"> + <type name="guint8" c:type="guint8"/> + </array> + </field> + <field name="unpack_format" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">the format of the unpacked samples</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </field> + <field name="unpack_func" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">function to unpack samples</doc> + <type name="AudioFormatUnpack" c:type="GstAudioFormatUnpack"/> + </field> + <field name="pack_func" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">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> + <method name="fill_silence" c:identifier="gst_audio_format_info_fill_silence" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Fill @length bytes in @dest with silence samples for @info.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormatInfo</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </instance-parameter> + <parameter name="dest" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a destination + to fill</doc> + <array length="1" zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the length to fill</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + </record> + <callback name="AudioFormatPack" c:type="GstAudioFormatPack"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Packs @length samples from @src to the data array in format @info. +The samples from source have each channel interleaved +and will be packed into @data.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">a #GstAudioFormatInfo</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">#GstAudioPackFlags</doc> + <type name="AudioPackFlags" c:type="GstAudioPackFlags"/> + </parameter> + <parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">a source array</doc> + <array zero-terminated="0" c:type="gconstpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">pointer to the destination + data</doc> + <array zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">the amount of samples to pack.</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </callback> + <callback name="AudioFormatUnpack" c:type="GstAudioFormatUnpack"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Unpacks @length samples from the given data of format @info. +The samples will be unpacked into @dest which each channel +interleaved. @dest should at least be big enough to hold @length * +channels * size(unpack_format) bytes.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">a #GstAudioFormatInfo</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">#GstAudioPackFlags</doc> + <type name="AudioPackFlags" c:type="GstAudioPackFlags"/> + </parameter> + <parameter name="dest" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">a destination array</doc> + <array zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">pointer to the audio data</doc> + <array zero-terminated="0" c:type="gconstpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">the amount of samples to unpack.</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </callback> + <record name="AudioInfo" c:type="GstAudioInfo" glib:type-name="GstAudioInfo" glib:get-type="gst_audio_info_get_type" c:symbol-prefix="audio_info"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">Information describing audio properties. This information can be filled +in from GstCaps with gst_audio_info_from_caps(). + +Use the provided macros to access the info in this structure.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <field name="finfo" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the format info of the audio</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </field> + <field name="flags" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">additional audio flags</doc> + <type name="AudioFlags" c:type="GstAudioFlags"/> + </field> + <field name="layout" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">audio layout</doc> + <type name="AudioLayout" c:type="GstAudioLayout"/> + </field> + <field name="rate" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the audio sample rate</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="channels" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the number of channels</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="bpf" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the number of bytes for one frame, this is the size of one + sample * @channels</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="position" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h">the positions for each channel</doc> + <array zero-terminated="0" fixed-size="64"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + <constructor name="new" c:identifier="gst_audio_info_new"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Allocate a new #GstAudioInfo that is also initialized with +gst_audio_info_init().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a new #GstAudioInfo. free with gst_audio_info_free().</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </return-value> + </constructor> + <constructor name="new_from_caps" c:identifier="gst_audio_info_new_from_caps" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Parse @caps to generate a #GstAudioInfo.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">A #GstAudioInfo, or %NULL if @caps couldn't be parsed</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </return-value> + <parameters> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstCaps</doc> + <type name="Gst.Caps" c:type="const GstCaps*"/> + </parameter> + </parameters> + </constructor> + <method name="convert" c:identifier="gst_audio_info_convert"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Converts among various #GstFormat types. This function handles +GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For +raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This +function can be used to handle pad queries of the type GST_QUERY_CONVERT.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">TRUE if the conversion was successful.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </instance-parameter> + <parameter name="src_fmt" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">#GstFormat of the @src_val</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </parameter> + <parameter name="src_val" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">value to convert</doc> + <type name="gint64" c:type="gint64"/> + </parameter> + <parameter name="dest_fmt" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">#GstFormat of the @dest_val</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </parameter> + <parameter name="dest_val" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">pointer to destination value</doc> + <type name="gint64" c:type="gint64*"/> + </parameter> + </parameters> + </method> + <method name="copy" c:identifier="gst_audio_info_copy"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Copy a GstAudioInfo structure.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a new #GstAudioInfo. free with gst_audio_info_free.</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </instance-parameter> + </parameters> + </method> + <method name="free" c:identifier="gst_audio_info_free"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Free a GstAudioInfo structure previously allocated with gst_audio_info_new() +or gst_audio_info_copy().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </instance-parameter> + </parameters> + </method> + <method name="is_equal" c:identifier="gst_audio_info_is_equal" version="1.2"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Compares two #GstAudioInfo and returns whether they are equal or not</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">%TRUE if @info and @other are equal, else %FALSE.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </instance-parameter> + <parameter name="other" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + </parameters> + </method> + <method name="set_format" c:identifier="gst_audio_info_set_format"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Set the default info for the audio info of @format and @rate and @channels. + +Note: This initializes @info first, no values are preserved.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </instance-parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">the format</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">the samplerate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">the number of channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="position" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">the channel positions</doc> + <array zero-terminated="0" c:type="const GstAudioChannelPosition*" fixed-size="64"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </method> + <method name="to_caps" c:identifier="gst_audio_info_to_caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Convert the values of @info into a #GstCaps.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">the new #GstCaps containing the + info of @info.</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </return-value> + <parameters> + <instance-parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </instance-parameter> + </parameters> + </method> + <function name="from_caps" c:identifier="gst_audio_info_from_caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Parse @caps and update @info.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">TRUE if @caps could be parsed</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="info" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstCaps</doc> + <type name="Gst.Caps" c:type="const GstCaps*"/> + </parameter> + </parameters> + </function> + <function name="init" c:identifier="gst_audio_info_init"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Initialize @info with default values.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + </parameters> + </function> + </record> + <enumeration name="AudioLayout" glib:type-name="GstAudioLayout" glib:get-type="gst_audio_layout_get_type" c:type="GstAudioLayout"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">Layout of the audio samples for the different channels.</doc> + <member name="interleaved" value="0" c:identifier="GST_AUDIO_LAYOUT_INTERLEAVED" glib:nick="interleaved"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">interleaved audio</doc> + </member> + <member name="non_interleaved" value="1" c:identifier="GST_AUDIO_LAYOUT_NON_INTERLEAVED" glib:nick="non-interleaved"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">non-interleaved audio</doc> + </member> + </enumeration> + <record name="AudioLevelMeta" c:type="GstAudioLevelMeta" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <field name="meta" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">parent #GstMeta</doc> + <type name="Gst.Meta" c:type="GstMeta"/> + </field> + <field name="level" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the -dBov from 0-127 (127 is silence).</doc> + <type name="guint8" c:type="guint8"/> + </field> + <field name="voice_activity" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">whether the buffer contains voice activity</doc> + <type name="gboolean" c:type="gboolean"/> + </field> + <function name="get_info" c:identifier="gst_audio_level_meta_get_info" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Return the #GstMetaInfo associated with #GstAudioLevelMeta.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstMetaInfo</doc> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + </record> + <record name="AudioMeta" c:type="GstAudioMeta" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">#GstAudioDownmixMeta defines an audio downmix matrix to be send along with +audio buffers. These functions in this module help to create and attach the +meta as well as extracting it.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <field name="meta" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">parent #GstMeta</doc> + <type name="Gst.Meta" c:type="GstMeta"/> + </field> + <field name="info" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the audio properties of the buffer</doc> + <type name="AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="samples" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the number of valid samples in the buffer</doc> + <type name="gsize" c:type="gsize"/> + </field> + <field name="offsets" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h">the offsets (in bytes) where each channel plane starts in the + buffer or %NULL if the buffer has interleaved layout; if not %NULL, this + is guaranteed to be an array of @info.channels elements</doc> + <type name="gsize" c:type="gsize*"/> + </field> + <field name="priv_offsets_arr" readable="0" private="1"> + <array zero-terminated="0" fixed-size="8"> + <type name="gsize" c:type="gsize"/> + </array> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + <function name="get_info" c:identifier="gst_audio_meta_get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + </record> + <enumeration name="AudioNoiseShapingMethod" glib:type-name="GstAudioNoiseShapingMethod" glib:get-type="gst_audio_noise_shaping_method_get_type" c:type="GstAudioNoiseShapingMethod"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Set of available noise shaping methods</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_NOISE_SHAPING_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">No noise shaping (default)</doc> + </member> + <member name="error_feedback" value="1" c:identifier="GST_AUDIO_NOISE_SHAPING_ERROR_FEEDBACK" glib:nick="error-feedback"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Error feedback</doc> + </member> + <member name="simple" value="2" c:identifier="GST_AUDIO_NOISE_SHAPING_SIMPLE" glib:nick="simple"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Simple 2-pole noise shaping</doc> + </member> + <member name="medium" value="3" c:identifier="GST_AUDIO_NOISE_SHAPING_MEDIUM" glib:nick="medium"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Medium 5-pole noise shaping</doc> + </member> + <member name="high" value="4" c:identifier="GST_AUDIO_NOISE_SHAPING_HIGH" glib:nick="high"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">High 8-pole noise shaping</doc> + </member> + </enumeration> + <bitfield name="AudioPackFlags" glib:type-name="GstAudioPackFlags" glib:get-type="gst_audio_pack_flags_get_type" c:type="GstAudioPackFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">The different flags that can be used when packing and unpacking.</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_PACK_FLAG_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">No flag</doc> + </member> + <member name="truncate_range" value="1" c:identifier="GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE" glib:nick="truncate-range"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h">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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <method name="free" c:identifier="gst_audio_quantize_free"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">Free a #GstAudioQuantize.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="quant" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioQuantize</doc> + <type name="AudioQuantize" c:type="GstAudioQuantize*"/> + </instance-parameter> + </parameters> + </method> + <method name="reset" c:identifier="gst_audio_quantize_reset"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">Reset @quant to the state is was when created, clearing any +history it might have.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="quant" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioQuantize</doc> + <type name="AudioQuantize" c:type="GstAudioQuantize*"/> + </instance-parameter> + </parameters> + </method> + <method name="samples" c:identifier="gst_audio_quantize_samples"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">Perform quantization on @samples in @in and write the result to @out. + +In case the samples are interleaved, @in and @out must point to an +array with a single element pointing to a block of interleaved samples. + +If non-interleaved samples are used, @in and @out must point to an +array with pointers to memory blocks, one for each channel. + +@in and @out may point to the same memory location, in which case samples will be +modified in-place.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="quant" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioQuantize</doc> + <type name="AudioQuantize" c:type="GstAudioQuantize*"/> + </instance-parameter> + <parameter name="in" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">input samples</doc> + <type name="gpointer" c:type="const gpointer*"/> + </parameter> + <parameter name="out" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">output samples</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">number of samples</doc> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </method> + <function name="new" c:identifier="gst_audio_quantize_new" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">Create a new quantizer object with the given parameters. + +Output samples will be quantized to a multiple of @quantizer. Better +performance is achieved when @quantizer is a power of 2. + +Dithering and noise-shaping can be performed during quantization with +the @dither and @ns parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a new #GstAudioQuantize. Free with gst_audio_quantize_free().</doc> + <type name="AudioQuantize" c:type="GstAudioQuantize*"/> + </return-value> + <parameters> + <parameter name="dither" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioDitherMethod</doc> + <type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/> + </parameter> + <parameter name="ns" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioNoiseShapingMethod</doc> + <type name="AudioNoiseShapingMethod" c:type="GstAudioNoiseShapingMethod"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">#GstAudioQuantizeFlags</doc> + <type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the #GstAudioFormat of the samples</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the amount of channels in the samples</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="quantizer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the quantizer to use</doc> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </function> + </record> + <bitfield name="AudioQuantizeFlags" glib:type-name="GstAudioQuantizeFlags" glib:get-type="gst_audio_quantize_flags_get_type" c:type="GstAudioQuantizeFlags"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">Extra flags that can be passed to gst_audio_quantize_new()</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_QUANTIZE_FLAG_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">no flags</doc> + </member> + <member name="non_interleaved" value="1" c:identifier="GST_AUDIO_QUANTIZE_FLAG_NON_INTERLEAVED" glib:nick="non-interleaved"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h">samples are non-interleaved</doc> + </member> + </bitfield> + <record name="AudioResampler" c:type="GstAudioResampler" disguised="1" version="1.10"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">#GstAudioResampler is a structure which holds the information +required to perform various kinds of resampling filtering.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <method name="free" c:identifier="gst_audio_resampler_free"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Free a previously allocated #GstAudioResampler @resampler.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_in_frames" c:identifier="gst_audio_resampler_get_in_frames"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Get the number of input frames that would currently be needed +to produce @out_frames from @resampler.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">The number of input frames needed for producing +@out_frames of data from @resampler.</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + <parameter name="out_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="get_max_latency" c:identifier="gst_audio_resampler_get_max_latency"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Get the maximum number of input samples that the resampler would +need before producing output.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the latency of @resampler as expressed in the number of +frames.</doc> + <type name="gsize" c:type="gsize"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_out_frames" c:identifier="gst_audio_resampler_get_out_frames"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Get the number of output frames that would be currently available when +@in_frames are given to @resampler.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + <parameter name="in_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="resample" c:identifier="gst_audio_resampler_resample"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Perform resampling on @in_frames frames in @in and write @out_frames to @out. + +In case the samples are interleaved, @in and @out must point to an +array with a single element pointing to a block of interleaved samples. + +If non-interleaved samples are used, @in and @out must point to an +array with pointers to memory blocks, one for each channel. + +@in may be %NULL, in which case @in_frames of silence samples are pushed +into the resampler. + +This function always produces @out_frames of output and consumes @in_frames of +input. Use gst_audio_resampler_get_out_frames() and +gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames +are matching and @in and @out point to enough memory.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + <parameter name="in" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">input samples</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="in_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">number of input frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="out" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">output samples</doc> + <type name="gpointer" c:type="gpointer*"/> + </parameter> + <parameter name="out_frames" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">number of output frames</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </method> + <method name="reset" c:identifier="gst_audio_resampler_reset"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Reset @resampler to the state it was when it was first created, discarding +all sample history.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + </parameters> + </method> + <method name="update" c:identifier="gst_audio_resampler_update"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Update the resampler parameters for @resampler. This function should +not be called concurrently with any other function on @resampler. + +When @in_rate or @out_rate is 0, its value is unchanged. + +When @options is %NULL, the previously configured options are reused.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">%TRUE if the new parameters could be set</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="resampler" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResampler</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </instance-parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">new input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">new output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="options" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">new options or %NULL</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </method> + <function name="new" c:identifier="gst_audio_resampler_new"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Make a new resampler.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="full" skip="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">The new #GstAudioResampler.</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </return-value> + <parameters> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResamplerMethod</doc> + <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">#GstAudioResamplerFlags</doc> + <type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the #GstAudioFormat</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the number of channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="options" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">extra options</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </function> + <function name="options_set_quality" c:identifier="gst_audio_resampler_options_set_quality"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Set the parameters for resampling from @in_rate to @out_rate using @method +for @quality in @options.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResamplerMethod</doc> + <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/> + </parameter> + <parameter name="quality" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the quality</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="options" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">The different filter interpolation methods.</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">no interpolation</doc> + </member> + <member name="linear" value="1" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR" glib:nick="linear"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Select for the filter tables should be set up.</doc> + <member name="interpolated" value="0" c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED" glib:nick="interpolated"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Use interpolated filter tables. This + uses less memory but more CPU and is slightly less accurate but it allows for more + efficient variable rate resampling with gst_audio_resampler_update().</doc> + </member> + <member name="full" value="1" c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_FULL" glib:nick="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Use full filter table. This uses more memory + but less CPU.</doc> + </member> + <member name="auto" value="2" c:identifier="GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO" glib:nick="auto"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Different resampler flags.</doc> + <member name="none" value="0" c:identifier="GST_AUDIO_RESAMPLER_FLAG_NONE" glib:nick="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">no flags</doc> + </member> + <member name="non_interleaved_in" value="1" c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN" glib:nick="non-interleaved-in"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">input samples are non-interleaved. + an array of blocks of samples, one for each channel, should be passed to the + resample function.</doc> + </member> + <member name="non_interleaved_out" value="2" c:identifier="GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT" glib:nick="non-interleaved-out"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">output samples are non-interleaved. + an array of blocks of samples, one for each channel, should be passed to the + resample function.</doc> + </member> + <member name="variable_rate" value="4" c:identifier="GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE" glib:nick="variable-rate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Different subsampling and upsampling methods</doc> + <member name="nearest" value="0" c:identifier="GST_AUDIO_RESAMPLER_METHOD_NEAREST" glib:nick="nearest"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Duplicates the samples when + upsampling and drops when downsampling</doc> + </member> + <member name="linear" value="1" c:identifier="GST_AUDIO_RESAMPLER_METHOD_LINEAR" glib:nick="linear"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Uses linear interpolation to reconstruct + missing samples and averaging to downsample</doc> + </member> + <member name="cubic" value="2" c:identifier="GST_AUDIO_RESAMPLER_METHOD_CUBIC" glib:nick="cubic"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Uses cubic interpolation</doc> + </member> + <member name="blackman_nuttall" value="3" c:identifier="GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL" glib:nick="blackman-nuttall"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Uses Blackman-Nuttall windowed sinc interpolation</doc> + </member> + <member name="kaiser" value="4" c:identifier="GST_AUDIO_RESAMPLER_METHOD_KAISER" glib:nick="kaiser"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h">Uses Kaiser windowed sinc interpolation</doc> + </member> + </enumeration> + <class name="AudioRingBuffer" c:symbol-prefix="audio_ring_buffer" c:type="GstAudioRingBuffer" parent="Gst.Object" abstract="1" glib:type-name="GstAudioRingBuffer" glib:get-type="gst_audio_ring_buffer_get_type" glib:type-struct="AudioRingBufferClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">This object is the base class for audio ringbuffers used by the base +audio source and sink classes. + +The ringbuffer abstracts a circular buffer of data. One reader and +one writer can operate on the data from different threads in a lockfree +manner. The base class is sufficiently flexible to be used as an +abstraction for DMA based ringbuffers as well as a pure software +implementations.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <function name="debug_spec_buff" c:identifier="gst_audio_ring_buffer_debug_spec_buff"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Print debug info about the buffer sized in @spec to the debug log.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the spec to debug</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </function> + <function name="debug_spec_caps" c:identifier="gst_audio_ring_buffer_debug_spec_caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Print debug info about the parsed caps in @spec to the debug log.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the spec to debug</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </function> + <function name="parse_caps" c:identifier="gst_audio_ring_buffer_parse_caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Parse @caps into @spec.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the caps could be parsed.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">a spec</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">a #GstCaps</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </parameter> + </parameters> + </function> + <virtual-method name="acquire" invoker="acquire"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Allocate the resources for the ringbuffer. This function fills +in the data pointer of the ring buffer with a valid #GstBuffer +to which samples can be written.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be acquired, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to acquire</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the specs of the buffer</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="activate" invoker="activate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Activate @buf to start or stop pulling data. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be activated in the requested mode, +FALSE on error.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to activate</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="active" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the new mode</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="clear_all" invoker="clear_all"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Clear all samples from the ringbuffer. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to clear</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="close_device" invoker="close_device"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Close the audio device associated with the ring buffer. The ring buffer +should already have been released via gst_audio_ring_buffer_release().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be closed, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="commit" invoker="commit"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Commit @in_samples samples pointed to by @data to the ringbuffer @buf. + +@in_samples and @out_samples define the rate conversion to perform on the +samples in @data. For negative rates, @out_samples must be negative and +@in_samples positive. + +When @out_samples is positive, the first sample will be written at position @sample +in the ringbuffer. When @out_samples is negative, the last sample will be written to +@sample in reverse order. + +@out_samples does not need to be a multiple of the segment size of the ringbuffer +although it is recommended for optimal performance. + +@accum will hold a temporary accumulator used in rate conversion and should be +set to 0 when this function is first called. In case the commit operation is +interrupted, one can resume the processing by passing the previously returned +@accum value back to this function. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples written to the ringbuffer or -1 on error. The +number of samples written can be less than @out_samples when @buf was interrupted +with a flush or stop.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to commit</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="sample" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the sample position of the data</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the data to commit</doc> + <array length="2" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="in_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples in the data to commit</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples to write to the ringbuffer</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="accum" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">accumulator for rate conversion.</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="delay" invoker="delay"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">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> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples queued in the audio device. + +MT safe.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to query</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open_device" invoker="open_device"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Open the audio device associated with the ring buffer. Does not perform any +setup on the device. You must open the device before acquiring the ring +buffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be opened, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="pause" invoker="pause"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Pause processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be paused, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to pause</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="release" invoker="release"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Free the resources of the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be released, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to release</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="resume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="start" invoker="start"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Start processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be started, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to start</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="stop" invoker="stop"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Stop processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be stopped, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to stop</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </virtual-method> + <method name="acquire" c:identifier="gst_audio_ring_buffer_acquire"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Allocate the resources for the ringbuffer. This function fills +in the data pointer of the ring buffer with a valid #GstBuffer +to which samples can be written.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be acquired, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to acquire</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the specs of the buffer</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </method> + <method name="activate" c:identifier="gst_audio_ring_buffer_activate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Activate @buf to start or stop pulling data. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be activated in the requested mode, +FALSE on error.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to activate</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="active" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the new mode</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="advance" c:identifier="gst_audio_ring_buffer_advance"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Subclasses should call this function to notify the fact that +@advance segments are now processed by the device. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to advance</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="advance" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of segments written</doc> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </method> + <method name="clear" c:identifier="gst_audio_ring_buffer_clear"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Clear the given segment of the buffer with silence samples. +This function is used by subclasses. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to clear</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="segment" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the segment to clear</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + <method name="clear_all" c:identifier="gst_audio_ring_buffer_clear_all"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Clear all samples from the ringbuffer. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to clear</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="close_device" c:identifier="gst_audio_ring_buffer_close_device"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Close the audio device associated with the ring buffer. The ring buffer +should already have been released via gst_audio_ring_buffer_release().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be closed, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="commit" c:identifier="gst_audio_ring_buffer_commit"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Commit @in_samples samples pointed to by @data to the ringbuffer @buf. + +@in_samples and @out_samples define the rate conversion to perform on the +samples in @data. For negative rates, @out_samples must be negative and +@in_samples positive. + +When @out_samples is positive, the first sample will be written at position @sample +in the ringbuffer. When @out_samples is negative, the last sample will be written to +@sample in reverse order. + +@out_samples does not need to be a multiple of the segment size of the ringbuffer +although it is recommended for optimal performance. + +@accum will hold a temporary accumulator used in rate conversion and should be +set to 0 when this function is first called. In case the commit operation is +interrupted, one can resume the processing by passing the previously returned +@accum value back to this function. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples written to the ringbuffer or -1 on error. The +number of samples written can be less than @out_samples when @buf was interrupted +with a flush or stop.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to commit</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="sample" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the sample position of the data</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the data to commit</doc> + <array length="2" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="in_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples in the data to commit</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples to write to the ringbuffer</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="accum" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">accumulator for rate conversion.</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </method> + <method name="convert" c:identifier="gst_audio_ring_buffer_convert"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result +will be put in @dest_val.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the conversion succeeded.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="src_fmt" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the source format</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </parameter> + <parameter name="src_val" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the source value</doc> + <type name="gint64" c:type="gint64"/> + </parameter> + <parameter name="dest_fmt" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the destination format</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </parameter> + <parameter name="dest_val" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">a location to store the converted value</doc> + <type name="gint64" c:type="gint64*"/> + </parameter> + </parameters> + </method> + <method name="delay" c:identifier="gst_audio_ring_buffer_delay"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">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> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples queued in the audio device. + +MT safe.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to query</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="device_is_open" c:identifier="gst_audio_ring_buffer_device_is_open"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Checks the status of the device associated with the ring buffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device was open, FALSE if it was closed. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="is_acquired" c:identifier="gst_audio_ring_buffer_is_acquired"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Check if the ringbuffer is acquired and ready to use.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the ringbuffer is acquired, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to check</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="is_active" c:identifier="gst_audio_ring_buffer_is_active"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Check if @buf is activated. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device is active.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="is_flushing" c:identifier="gst_audio_ring_buffer_is_flushing"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Check if @buf is flushing. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device is flushing.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="may_start" c:identifier="gst_audio_ring_buffer_may_start"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Tell the ringbuffer that it is allowed to start playback when +the ringbuffer is filled with samples. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="allowed" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the new value</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="open_device" c:identifier="gst_audio_ring_buffer_open_device"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Open the audio device associated with the ring buffer. Does not perform any +setup on the device. You must open the device before acquiring the ring +buffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be opened, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="pause" c:identifier="gst_audio_ring_buffer_pause"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Pause processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be paused, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to pause</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="prepare_read" c:identifier="gst_audio_ring_buffer_prepare_read"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Returns a pointer to memory where the data from segment @segment +can be found. This function is mostly used by subclasses.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">FALSE if the buffer is not started. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to read from</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="segment" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the segment to read</doc> + <type name="gint" c:type="gint*"/> + </parameter> + <parameter name="readptr" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c"> + the pointer to the memory where samples can be read</doc> + <array length="2" zero-terminated="0" c:type="guint8**"> + <type name="guint8" c:type="guint8*"/> + </array> + </parameter> + <parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of bytes to read</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </method> + <method name="read" c:identifier="gst_audio_ring_buffer_read"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Read @len samples from the ringbuffer into the memory pointed +to by @data. +The first sample should be read from position @sample in +the ringbuffer. + +@len should not be a multiple of the segment size of the ringbuffer +although it is recommended. + +@timestamp will return the timestamp associated with the data returned.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples read from the ringbuffer or -1 on +error. + +MT safe.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to read from</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="sample" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the sample position of the data</doc> + <type name="guint64" c:type="guint64"/> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">where the data should be read</doc> + <array length="2" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="len" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples in data to read</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="timestamp" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">where the timestamp is returned</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </method> + <method name="release" c:identifier="gst_audio_ring_buffer_release"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Free the resources of the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be released, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to release</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="samples_done" c:identifier="gst_audio_ring_buffer_samples_done"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Get the number of samples that were processed by the ringbuffer +since it was last started. This does not include the number of samples not +yet processed (see gst_audio_ring_buffer_delay()).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples processed by the ringbuffer. + +MT safe.</doc> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to query</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="set_callback" c:identifier="gst_audio_ring_buffer_set_callback" shadowed-by="set_callback_full" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Sets the given callback function on the buffer. This function +will be called every time a segment has been written to a device. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to set the callback on</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="cb" transfer-ownership="none" nullable="1" allow-none="1" closure="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the callback to set</doc> + <type name="AudioRingBufferCallback" c:type="GstAudioRingBufferCallback"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">user data passed to the callback</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + </parameters> + </method> + <method name="set_callback_full" c:identifier="gst_audio_ring_buffer_set_callback_full" shadows="set_callback" version="1.12"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Sets the given callback function on the buffer. This function +will be called every time a segment has been written to a device. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to set the callback on</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="cb" transfer-ownership="none" nullable="1" allow-none="1" scope="notified" closure="1" destroy="2"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the callback to set</doc> + <type name="AudioRingBufferCallback" c:type="GstAudioRingBufferCallback"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">user data passed to the callback</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + <parameter name="notify" transfer-ownership="none" scope="async"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">function to be called when @user_data is no longer needed</doc> + <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/> + </parameter> + </parameters> + </method> + <method name="set_channel_positions" c:identifier="gst_audio_ring_buffer_set_channel_positions"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Tell the ringbuffer about the device's channel positions. This must +be called in when the ringbuffer is acquired.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the device channel positions</doc> + <array zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </method> + <method name="set_flushing" c:identifier="gst_audio_ring_buffer_set_flushing"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Set the ringbuffer to flushing mode or normal mode. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to flush</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="flushing" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the new mode</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_sample" c:identifier="gst_audio_ring_buffer_set_sample"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Make sure that the next sample written to the device is +accounted for as being the @sample sample written to the +device. This value will be used in reporting the current +sample position of the ringbuffer. + +This function will also clear the buffer with silence. + +MT safe.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to use</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="sample" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the sample number to set</doc> + <type name="guint64" c:type="guint64"/> + </parameter> + </parameters> + </method> + <method name="set_timestamp" c:identifier="gst_audio_ring_buffer_set_timestamp"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + <parameter name="readseg" transfer-ownership="none"> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="timestamp" transfer-ownership="none"> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="start" c:identifier="gst_audio_ring_buffer_start"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Start processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be started, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to start</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <method name="stop" c:identifier="gst_audio_ring_buffer_stop"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">Stop processing samples from the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be stopped, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to stop</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </instance-parameter> + </parameters> + </method> + <field name="object"> + <type name="Gst.Object" c:type="GstObject"/> + </field> + <field name="cond"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">used to signal start/stop/pause/resume actions</doc> + <type name="GLib.Cond" c:type="GCond"/> + </field> + <field name="open"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">boolean indicating that the ringbuffer is open</doc> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="acquired"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">boolean indicating that the ringbuffer is acquired</doc> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="memory"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">data in the ringbuffer</doc> + <type name="guint8" c:type="guint8*"/> + </field> + <field name="size"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">size of data in the ringbuffer</doc> + <type name="gsize" c:type="gsize"/> + </field> + <field name="timestamps" readable="0" private="1"> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </field> + <field name="spec"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">format and layout of the ringbuffer data</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec"/> + </field> + <field name="samples_per_seg"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">number of samples in one segment</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="empty_seg"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">pointer to memory holding one segment of silence samples</doc> + <type name="guint8" c:type="guint8*"/> + </field> + <field name="state"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">state of the buffer</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="segdone"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">readpointer in the ringbuffer</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="segbase"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">segment corresponding to segment 0 (unused)</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="waiting"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">is a reader or writer waiting for a free segment</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="callback" readable="0" private="1"> + <type name="AudioRingBufferCallback" c:type="GstAudioRingBufferCallback"/> + </field> + <field name="cb_data" readable="0" private="1"> + <type name="gpointer" c:type="gpointer"/> + </field> + <field name="need_reorder" readable="0" private="1"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="channel_reorder_map" readable="0" private="1"> + <array zero-terminated="0" fixed-size="64"> + <type name="gint" c:type="gint"/> + </array> + </field> + <field name="flushing" readable="0" private="1"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="may_start" readable="0" private="1"> + <type name="gint" c:type="gint"/> + </field> + <field name="active" readable="0" private="1"> + <type name="gboolean" c:type="gboolean"/> + </field> + <field name="cb_data_notify" readable="0" private="1"> + <type name="GLib.DestroyNotify" c:type="GDestroyNotify"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="3"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <callback name="AudioRingBufferCallback" c:type="GstAudioRingBufferCallback"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">This function is set with gst_audio_ring_buffer_set_callback() and is +called to fill the memory at @data with @len bytes of samples.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="rbuf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">a #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">target to fill</doc> + <array length="2" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="len" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">amount to fill</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="3"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">user data</doc> + <type name="gpointer" c:type="gpointer"/> + </parameter> + </parameters> + </callback> + <record name="AudioRingBufferClass" c:type="GstAudioRingBufferClass" glib:is-gtype-struct-for="AudioRingBuffer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The vmethods that subclasses can override to implement the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <field name="parent_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">parent class</doc> + <type name="Gst.ObjectClass" c:type="GstObjectClass"/> + </field> + <field name="open_device"> + <callback name="open_device"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be opened, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="acquire"> + <callback name="acquire"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be acquired, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to acquire</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the specs of the buffer</doc> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="release"> + <callback name="release"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be released, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to release</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close_device"> + <callback name="close_device"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be closed, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="start"> + <callback name="start"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be started, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to start</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="pause"> + <callback name="pause"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be paused, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to pause</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="resume"> + <callback name="resume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="stop"> + <callback name="stop"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be stopped, FALSE on error. + +MT safe.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to stop</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="delay"> + <callback name="delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples queued in the audio device. + +MT safe.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to query</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="activate"> + <callback name="activate"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">TRUE if the device could be activated in the requested mode, +FALSE on error.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to activate</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + <parameter name="active" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the new mode</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </callback> + </field> + <field name="commit"> + <callback name="commit"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">The number of samples written to the ringbuffer or -1 on error. The +number of samples written can be less than @out_samples when @buf was interrupted +with a flush or stop.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to commit</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + <parameter name="sample" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the sample position of the data</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the data to commit</doc> + <array length="3" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="in_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples in the data to commit</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the number of samples to write to the ringbuffer</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="accum" direction="inout" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">accumulator for rate conversion.</doc> + <type name="gint" c:type="gint*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="clear_all"> + <callback name="clear_all"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="buf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.c">the #GstAudioRingBuffer to clear</doc> + <type name="AudioRingBuffer" c:type="GstAudioRingBuffer*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <enumeration name="AudioRingBufferFormatType" glib:type-name="GstAudioRingBufferFormatType" glib:get-type="gst_audio_ring_buffer_format_type_get_type" c:type="GstAudioRingBufferFormatType"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The format of the samples in the ringbuffer.</doc> + <member name="raw" value="0" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW" glib:nick="raw"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in linear or float</doc> + </member> + <member name="mu_law" value="1" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW" glib:nick="mu-law"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in mulaw</doc> + </member> + <member name="a_law" value="2" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW" glib:nick="a-law"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in alaw</doc> + </member> + <member name="ima_adpcm" value="3" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM" glib:nick="ima-adpcm"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in ima adpcm</doc> + </member> + <member name="mpeg" value="4" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG" glib:nick="mpeg"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in mpeg audio (but not AAC) format</doc> + </member> + <member name="gsm" value="5" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM" glib:nick="gsm"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in gsm format</doc> + </member> + <member name="iec958" value="6" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958" glib:nick="iec958"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in IEC958 frames (e.g. AC3)</doc> + </member> + <member name="ac3" value="7" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3" glib:nick="ac3"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in AC3 format</doc> + </member> + <member name="eac3" value="8" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3" glib:nick="eac3"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in EAC3 format</doc> + </member> + <member name="dts" value="9" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS" glib:nick="dts"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in DTS format</doc> + </member> + <member name="mpeg2_aac" value="10" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC" glib:nick="mpeg2-aac"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in MPEG-2 AAC ADTS format</doc> + </member> + <member name="mpeg4_aac" value="11" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC" glib:nick="mpeg4-aac"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in MPEG-4 AAC ADTS format</doc> + </member> + <member name="mpeg2_aac_raw" value="12" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC_RAW" glib:nick="mpeg2-aac-raw"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in MPEG-2 AAC raw format (Since: 1.12)</doc> + </member> + <member name="mpeg4_aac_raw" value="13" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC_RAW" glib:nick="mpeg4-aac-raw"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in MPEG-4 AAC raw format (Since: 1.12)</doc> + </member> + <member name="flac" value="14" c:identifier="GST_AUDIO_RING_BUFFER_FORMAT_TYPE_FLAC" glib:nick="flac"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">samples in FLAC format (Since: 1.12)</doc> + </member> + </enumeration> + <record name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The structure containing the format specification of the ringbuffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <field name="caps" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The caps that generated the Spec.</doc> + <type name="Gst.Caps" c:type="GstCaps*"/> + </field> + <field name="type" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the sample type</doc> + <type name="AudioRingBufferFormatType" c:type="GstAudioRingBufferFormatType"/> + </field> + <field name="info" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo"/> + </field> + <field name="latency_time" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the latency in microseconds</doc> + <type name="guint64" c:type="guint64"/> + </field> + <field name="buffer_time" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the total buffer size in microseconds</doc> + <type name="guint64" c:type="guint64"/> + </field> + <field name="segsize" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the size of one segment in bytes</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="segtotal" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">the total number of segments</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="seglatency" writable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">number of segments queued in the lower level device, + defaults to segtotal</doc> + <type name="gint" c:type="gint"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <enumeration name="AudioRingBufferState" glib:type-name="GstAudioRingBufferState" glib:get-type="gst_audio_ring_buffer_state_get_type" c:type="GstAudioRingBufferState"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The state of the ringbuffer.</doc> + <member name="stopped" value="0" c:identifier="GST_AUDIO_RING_BUFFER_STATE_STOPPED" glib:nick="stopped"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The ringbuffer is stopped</doc> + </member> + <member name="paused" value="1" c:identifier="GST_AUDIO_RING_BUFFER_STATE_PAUSED" glib:nick="paused"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The ringbuffer is paused</doc> + </member> + <member name="started" value="2" c:identifier="GST_AUDIO_RING_BUFFER_STATE_STARTED" glib:nick="started"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The ringbuffer is started</doc> + </member> + <member name="error" value="3" c:identifier="GST_AUDIO_RING_BUFFER_STATE_ERROR" glib:nick="error"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h">The ringbuffer has encountered an + error after it has been started, e.g. because the device was + disconnected (Since: 1.2)</doc> + </member> + </enumeration> + <class name="AudioSink" c:symbol-prefix="audio_sink" c:type="GstAudioSink" parent="AudioBaseSink" glib:type-name="GstAudioSink" glib:get-type="gst_audio_sink_get_type" glib:type-struct="AudioSinkClass"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.c">This is the most simple base class for audio sinks that only requires +subclasses to implement a set of simple functions: + +* `open()` :Open the device. + +* `prepare()` :Configure the device with the specified format. + +* `write()` :Write samples to the device. + +* `reset()` :Unblock writes and flush the device. + +* `delay()` :Get the number of samples written but not yet played +by the device. + +* `unprepare()` :Undo operations done by prepare. + +* `close()` :Close the device. + +All scheduling of samples and timestamps is done in this base class +together with #GstAudioBaseSink using a default implementation of a +#GstAudioRingBuffer that uses threads.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <virtual-method name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </instance-parameter> + <parameter name="spec" transfer-ownership="none"> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="reset"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="unprepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h">Write samples to the device.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h">the sample data</doc> + <array length="1" zero-terminated="0" c:type="gpointer"> + <type name="guint8" c:type="gpointer"/> + </array> + </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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <field name="parent_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h">the parent class structure.</doc> + <type name="AudioBaseSinkClass" c:type="GstAudioBaseSinkClass"/> + </field> + <field name="open"> + <callback name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="prepare"> + <callback name="prepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </parameter> + <parameter name="spec" transfer-ownership="none"> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="unprepare"> + <callback name="unprepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close"> + <callback name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="write"> + <callback name="write"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h">the sample data</doc> + <array length="2" zero-terminated="0" c:type="gpointer"> + <type name="guint8" c:type="gpointer"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </callback> + </field> + <field name="delay"> + <callback name="delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="sink" transfer-ownership="none"> + <type name="AudioSink" c:type="GstAudioSink*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="reset"> + <callback name="reset"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h">class extension structure. Since: 1.18</doc> + <type name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension*"/> + </field> + </record> + <record name="AudioSinkClassExtension" c:type="GstAudioSinkClassExtension"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <field name="clear_all"> + <callback name="clear_all"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.c">This is the most simple base class for audio sources that only requires +subclasses to implement a set of simple functions: + +* `open()` :Open the device. +* `prepare()` :Configure the device with the specified format. +* `read()` :Read samples from the device. +* `reset()` :Unblock reads and flush the device. +* `delay()` :Get the number of samples in the device but not yet read. +* `unprepare()` :Undo operations done by prepare. +* `close()` :Close the device. + +All scheduling of samples and timestamps is done in this base class +together with #GstAudioBaseSrc using a default implementation of a +#GstAudioRingBuffer that uses threads.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <virtual-method name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="prepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">Read samples from the device.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">the sample data</doc> + <array length="1" zero-terminated="0" c:type="gpointer"> + <type name="guint8" c:type="gpointer"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="timestamp" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">a #GstClockTime</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </virtual-method> + <virtual-method name="reset"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <virtual-method name="unprepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </instance-parameter> + </parameters> + </virtual-method> + <field name="element"> + <type name="AudioBaseSrc" c:type="GstAudioBaseSrc"/> + </field> + <field name="thread" readable="0" private="1"> + <type name="GLib.Thread" c:type="GThread*"/> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </class> + <record name="AudioSrcClass" c:type="GstAudioSrcClass" glib:is-gtype-struct-for="AudioSrc"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">#GstAudioSrc class. Override the vmethod to implement +functionality.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <field name="parent_class"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">the parent class.</doc> + <type name="AudioBaseSrcClass" c:type="GstAudioBaseSrcClass"/> + </field> + <field name="open"> + <callback name="open"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="prepare"> + <callback name="prepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + <parameter name="spec" transfer-ownership="none"> + <type name="AudioRingBufferSpec" c:type="GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="unprepare"> + <callback name="unprepare"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="close"> + <callback name="close"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="read"> + <callback name="read"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <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"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">the sample data</doc> + <array length="2" zero-terminated="0" c:type="gpointer"> + <type name="guint8" c:type="gpointer"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="timestamp" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h">a #GstClockTime</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="delay"> + <callback name="delay"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="reset"> + <callback name="reset"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <type name="AudioSrc" c:type="GstAudioSrc*"/> + </parameter> + </parameters> + </callback> + </field> + <field name="_gst_reserved" readable="0" private="1"> + <array zero-terminated="0" fixed-size="4"> + <type name="gpointer" c:type="gpointer"/> + </array> + </field> + </record> + <record name="AudioStreamAlign" c:type="GstAudioStreamAlign" version="1.14" glib:type-name="GstAudioStreamAlign" glib:get-type="gst_audio_stream_align_get_type" c:symbol-prefix="audio_stream_align"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">#GstAudioStreamAlign provides a helper object that helps tracking audio +stream alignment and discontinuities, and detects discontinuities if +possible. + +See gst_audio_stream_align_new() for a description of its parameters and +gst_audio_stream_align_process() for the details of the processing.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <constructor name="new" c:identifier="gst_audio_stream_align_new" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Allocate a new #GstAudioStreamAlign with the given configuration. All +processing happens according to sample rate @rate, until +gst_audio_stream_align_set_rate() is called with a new @rate. +A negative rate can be used for reverse playback. + +@alignment_threshold gives the tolerance in nanoseconds after which a +timestamp difference is considered a discontinuity. Once detected, +@discont_wait nanoseconds have to pass without going below the threshold +again until the output buffer is marked as a discontinuity. These can later +be re-configured with gst_audio_stream_align_set_alignment_threshold() and +gst_audio_stream_align_set_discont_wait().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a new #GstAudioStreamAlign. free with gst_audio_stream_align_free().</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </return-value> + <parameters> + <parameter name="rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a sample rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="alignment_threshold" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a alignment threshold in nanoseconds</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="discont_wait" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">discont wait in nanoseconds</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </constructor> + <method name="copy" c:identifier="gst_audio_stream_align_copy" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Copy a GstAudioStreamAlign structure.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a new #GstAudioStreamAlign. free with gst_audio_stream_align_free.</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="free" c:identifier="gst_audio_stream_align_free" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new() +or gst_audio_stream_align_copy().</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_alignment_threshold" c:identifier="gst_audio_stream_align_get_alignment_threshold" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Gets the currently configured alignment threshold.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">The currently configured alignment threshold</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_discont_wait" c:identifier="gst_audio_stream_align_get_discont_wait" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Gets the currently configured discont wait.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">The currently configured discont wait</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_rate" c:identifier="gst_audio_stream_align_get_rate" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Gets the currently configured sample rate.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">The currently configured sample rate</doc> + <type name="gint" c:type="gint"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_samples_since_discont" c:identifier="gst_audio_stream_align_get_samples_since_discont" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Returns the number of samples that were processed since the last +discontinuity was detected.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">The number of samples processed since the last discontinuity.</doc> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_timestamp_at_discont" c:identifier="gst_audio_stream_align_get_timestamp_at_discont" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Timestamp that was passed when a discontinuity was detected, i.e. the first +timestamp after the discontinuity.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">The last timestamp at when a discontinuity was detected</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="const GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="mark_discont" c:identifier="gst_audio_stream_align_mark_discont" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Marks the next buffer as discontinuous and resets timestamp tracking.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + </parameters> + </method> + <method name="process" c:identifier="gst_audio_stream_align_process" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Processes data with @timestamp and @n_samples, and returns the output +timestamp, duration and sample position together with a boolean to signal +whether a discontinuity was detected or not. All non-discontinuous data +will have perfect timestamps and durations. + +A discontinuity is detected once the difference between the actual +timestamp and the timestamp calculated from the sample count since the last +discontinuity differs by more than the alignment threshold for a duration +longer than discont wait. + +Note: In reverse playback, every buffer is considered discontinuous in the +context of buffer flags because the last sample of the previous buffer is +discontinuous with the first sample of the current one. However for this +function they are only considered discontinuous in reverse playback if the +first sample of the previous buffer is discontinuous with the last sample +of the current one.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">%TRUE if a discontinuity was detected, %FALSE otherwise.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + <parameter name="discont" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">if this data is considered to be discontinuous</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + <parameter name="timestamp" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstClockTime of the start of the data</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + <parameter name="n_samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">number of samples to process</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="out_timestamp" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">output timestamp of the data</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="out_duration" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">output duration of the data</doc> + <type name="Gst.ClockTime" c:type="GstClockTime*"/> + </parameter> + <parameter name="out_sample_position" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">output sample position of the start of the data</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + </parameters> + </method> + <method name="set_alignment_threshold" c:identifier="gst_audio_stream_align_set_alignment_threshold" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Sets @alignment_treshold as new alignment threshold for the following processing.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + <parameter name="alignment_threshold" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a new alignment threshold</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_discont_wait" c:identifier="gst_audio_stream_align_set_discont_wait" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Sets @alignment_treshold as new discont wait for the following processing.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + <parameter name="discont_wait" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a new discont wait</doc> + <type name="Gst.ClockTime" c:type="GstClockTime"/> + </parameter> + </parameters> + </method> + <method name="set_rate" c:identifier="gst_audio_stream_align_set_rate" version="1.14"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">Sets @rate as new sample rate for the following processing. If the sample +rate differs this implicitly marks the next data as discontinuous.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="align" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a #GstAudioStreamAlign</doc> + <type name="AudioStreamAlign" c:type="GstAudioStreamAlign*"/> + </instance-parameter> + <parameter name="rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiostreamalign.c">a new sample rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </method> + </record> + <function-macro name="CLOCK_TIME_TO_FRAMES" c:identifier="GST_CLOCK_TIME_TO_FRAMES" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">Calculate frames from @clocktime and sample @rate.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <parameters> + <parameter name="clocktime"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">clock time</doc> + </parameter> + <parameter name="rate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">sampling rate</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="FRAMES_TO_CLOCK_TIME" c:identifier="GST_FRAMES_TO_CLOCK_TIME" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">Calculate clocktime from sample @frames and @rate.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <parameters> + <parameter name="frames"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">sample frames</doc> + </parameter> + <parameter name="rate"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">sampling rate</doc> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR" c:identifier="GST_IS_AUDIO_AGGREGATOR" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR_CLASS" c:identifier="GST_IS_AUDIO_AGGREGATOR_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR_CONVERT_PAD" c:identifier="GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS" c:identifier="GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR_PAD" c:identifier="GST_IS_AUDIO_AGGREGATOR_PAD" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_AGGREGATOR_PAD_CLASS" c:identifier="GST_IS_AUDIO_AGGREGATOR_PAD_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioaggregator.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_BASE_SINK" c:identifier="GST_IS_AUDIO_BASE_SINK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_BASE_SINK_CLASS" c:identifier="GST_IS_AUDIO_BASE_SINK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesink.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_BASE_SRC" c:identifier="GST_IS_AUDIO_BASE_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_BASE_SRC_CLASS" c:identifier="GST_IS_AUDIO_BASE_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiobasesrc.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_CD_SRC" c:identifier="GST_IS_AUDIO_CD_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_CD_SRC_CLASS" c:identifier="GST_IS_AUDIO_CD_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiocdsrc.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_CLOCK" c:identifier="GST_IS_AUDIO_CLOCK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_CLOCK_CLASS" c:identifier="GST_IS_AUDIO_CLOCK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioclock.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_DECODER" c:identifier="GST_IS_AUDIO_DECODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_DECODER_CLASS" c:identifier="GST_IS_AUDIO_DECODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_ENCODER" c:identifier="GST_IS_AUDIO_ENCODER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_ENCODER_CLASS" c:identifier="GST_IS_AUDIO_ENCODER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_FILTER" c:identifier="GST_IS_AUDIO_FILTER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_FILTER_CLASS" c:identifier="GST_IS_AUDIO_FILTER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiofilter.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_RING_BUFFER" c:identifier="GST_IS_AUDIO_RING_BUFFER" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_RING_BUFFER_CLASS" c:identifier="GST_IS_AUDIO_RING_BUFFER_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioringbuffer.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_SINK" c:identifier="GST_IS_AUDIO_SINK" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_SINK_CLASS" c:identifier="GST_IS_AUDIO_SINK_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosink.h"/> + <parameters> + <parameter name="klass"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_SRC" c:identifier="GST_IS_AUDIO_SRC" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <parameters> + <parameter name="obj"> + </parameter> + </parameters> + </function-macro> + <function-macro name="IS_AUDIO_SRC_CLASS" c:identifier="GST_IS_AUDIO_SRC_CLASS" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiosrc.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">This metadata stays relevant as long as channels are unchanged.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="META_TAG_AUDIO_RATE_STR" value="rate" c:type="GST_META_TAG_AUDIO_RATE_STR" version="1.8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">This metadata stays relevant as long as sample rate is unchanged.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <constant name="META_TAG_AUDIO_STR" value="audio" c:type="GST_META_TAG_AUDIO_STR" version="1.2"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h">This metadata is relevant for audio streams.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <type name="utf8" c:type="gchar*"/> + </constant> + <function-macro name="STREAM_VOLUME_GET_INTERFACE" c:identifier="GST_STREAM_VOLUME_GET_INTERFACE" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">This interface is implemented by elements that provide a stream volume. Examples for +such elements are #volume and #playbin. + +Applications can use this interface to get or set the current stream volume. For this +the "volume" #GObject property can be used or the helper functions gst_stream_volume_set_volume() +and gst_stream_volume_get_volume(). This volume is always a linear factor, i.e. 0.0 is muted +1.0 is 100%. For showing the volume in a GUI it might make sense to convert it to +a different format by using gst_stream_volume_convert_volume(). Volume sliders should usually +use a cubic volume. + +Separate from the volume the stream can also be muted by the "mute" #GObject property or +gst_stream_volume_set_mute() and gst_stream_volume_get_mute(). + +Elements that provide some kind of stream volume should implement the "volume" and +"mute" #GObject properties and handle setting and getting of them properly. +The volume property is defined to be a linear volume factor.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <function name="convert_volume" c:identifier="gst_stream_volume_convert_volume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">the converted volume</doc> + <type name="gdouble" c:type="gdouble"/> + </return-value> + <parameters> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat to convert from</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat to convert to</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + <parameter name="val" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">Volume in @from format that should be converted</doc> + <type name="gdouble" c:type="gdouble"/> + </parameter> + </parameters> + </function> + <method name="get_mute" c:identifier="gst_stream_volume_get_mute"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">Returns %TRUE if the stream is muted</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <instance-parameter name="volume" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolume that should be used</doc> + <type name="StreamVolume" c:type="GstStreamVolume*"/> + </instance-parameter> + </parameters> + </method> + <method name="get_volume" c:identifier="gst_stream_volume_get_volume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">The current stream volume as linear factor</doc> + <type name="gdouble" c:type="gdouble"/> + </return-value> + <parameters> + <instance-parameter name="volume" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolume that should be used</doc> + <type name="StreamVolume" c:type="GstStreamVolume*"/> + </instance-parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat which should be returned</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + </parameters> + </method> + <method name="set_mute" c:identifier="gst_stream_volume_set_mute"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="volume" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolume that should be used</doc> + <type name="StreamVolume" c:type="GstStreamVolume*"/> + </instance-parameter> + <parameter name="mute" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">Mute state that should be set</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </method> + <method name="set_volume" c:identifier="gst_stream_volume_set_volume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <instance-parameter name="volume" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolume that should be used</doc> + <type name="StreamVolume" c:type="GstStreamVolume*"/> + </instance-parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat of @val</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + <parameter name="val" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">Linear volume factor that should be set</doc> + <type name="gdouble" c:type="gdouble"/> + </parameter> + </parameters> + </method> + <property name="mute" writable="1" transfer-ownership="none"> + <type name="gboolean" c:type="gboolean"/> + </property> + <property name="volume" writable="1" transfer-ownership="none"> + <type name="gdouble" c:type="gdouble"/> + </property> + </interface> + <enumeration name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h">Different representations of a stream volume. gst_stream_volume_convert_volume() +allows to convert between the different representations. + +Formulas to convert from a linear to a cubic or dB volume are +cbrt(val) and 20 * log10 (val).</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <member name="linear" value="0" c:identifier="GST_STREAM_VOLUME_FORMAT_LINEAR"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h">Linear scale factor, 1.0 = 100%</doc> + </member> + <member name="cubic" value="1" c:identifier="GST_STREAM_VOLUME_FORMAT_CUBIC"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h">Cubic volume scale</doc> + </member> + <member name="db" value="2" c:identifier="GST_STREAM_VOLUME_FORMAT_DB"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h">Logarithmic volume scale (dB, amplitude not power)</doc> + </member> + </enumeration> + <record name="StreamVolumeInterface" c:type="GstStreamVolumeInterface" glib:is-gtype-struct-for="StreamVolume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <field name="iface"> + <type name="GObject.TypeInterface" c:type="GTypeInterface"/> + </field> + </record> + <function name="audio_buffer_clip" c:identifier="gst_audio_buffer_clip" moved-to="AudioBuffer.clip"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">Clip the buffer to the given %GstSegment. + +After calling this function the caller does not own a reference to +@buffer anymore.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <return-value transfer-ownership="full" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">%NULL if the buffer is completely outside the configured segment, +otherwise the clipped buffer is returned. + +If the buffer has no timestamp, it is assumed to be inside the segment and +is not clipped</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">The buffer to clip.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="segment" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which + the buffer should be clipped.</doc> + <type name="Gst.Segment" c:type="const GstSegment*"/> + </parameter> + <parameter name="rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">sample rate.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="bpf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">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_map" c:identifier="gst_audio_buffer_map" moved-to="AudioBuffer.map" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">Maps an audio @gstbuffer so that it can be read or written and stores the +result of the map operation in @buffer. + +This is especially useful when the @gstbuffer is in non-interleaved (planar) +layout, in which case this function will use the information in the +@gstbuffer's attached #GstAudioMeta in order to map each channel in a +separate "plane" in #GstAudioBuffer. If a #GstAudioMeta is not attached +on the @gstbuffer, then it must be in interleaved layout. + +If a #GstAudioMeta is attached, then the #GstAudioInfo on the meta is checked +against @info. Normally, they should be equal, but in case they are not, +a g_critical will be printed and the #GstAudioInfo from the meta will be +used. + +In non-interleaved buffers, it is possible to have each channel on a separate +#GstMemory. In this case, each memory will be mapped separately to avoid +copying their contents in a larger memory area. Do note though that it is +not supported to have a single channel spanning over two or more different +#GstMemory objects. Although the map operation will likely succeed in this +case, it will be highly sub-optimal and it is recommended to merge all the +memories in the buffer before calling this function. + +Note: The actual #GstBuffer is not ref'ed, but it is required to stay valid +as long as it's mapped.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">%TRUE if the map operation succeeded or %FALSE on failure</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buffer" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">pointer to a #GstAudioBuffer</doc> + <type name="AudioBuffer" c:type="GstAudioBuffer*"/> + </parameter> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the audio properties of the buffer</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + <parameter name="gstbuffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the #GstBuffer to be mapped</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-buffer.c">the access mode for the memory</doc> + <type name="Gst.MapFlags" c:type="GstMapFlags"/> + </parameter> + </parameters> + </function> + <function name="audio_buffer_reorder_channels" c:identifier="gst_audio_buffer_reorder_channels" moved-to="AudioBuffer.reorder_channels"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Reorders @buffer from the channel positions @from to the channel +positions @to. @from and @to must contain the same number of +positions and the same positions, only in a different order. +@buffer must be writable.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the reordering was possible.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The buffer to reorder.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioFormat of the buffer.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions in the buffer.</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to convert to.</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </function> + <function name="audio_buffer_truncate" c:identifier="gst_audio_buffer_truncate" moved-to="AudioBuffer.truncate" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">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. + +This function does not know the audio rate, therefore the caller is +responsible for re-setting the correct timestamp and duration to the +buffer. However, timestamp will be preserved if trim == 0, and duration +will also be preserved if there is no trimming to be done. Offset and +offset end will be preserved / updated. + +After calling this function the caller does not own a reference to +@buffer anymore.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the truncated buffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">The buffer to truncate.</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="bpf" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">size of one audio frame in bytes. This is the size of one sample * +number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="trim" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the number of samples to remove from the beginning of the buffer</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio.c">the final number of samples that should exist in this buffer or -1 +to use all the remaining samples if you are only removing samples from the +beginning.</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_get_fallback_mask" c:identifier="gst_audio_channel_get_fallback_mask" version="1.8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Get the fallback channel-mask for the given number of channels. + +This function returns a reasonable fallback channel-mask and should be +called as a last resort when the specific channel map is unknown.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">a fallback channel-mask for @channels or 0 when there is no +mask and mono.</doc> + <type name="guint64" c:type="guint64"/> + </return-value> + <parameters> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">the number of channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_mixer_new" c:identifier="gst_audio_channel_mixer_new" moved-to="AudioChannelMixer.new" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Create a new channel mixer object for the given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a new #GstAudioChannelMixer object. + Free with gst_audio_channel_mixer_free() after usage.</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </return-value> + <parameters> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">#GstAudioChannelMixerFlags</doc> + <type name="AudioChannelMixerFlags" c:type="GstAudioChannelMixerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="in_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of input channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="in_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">positions of input channels</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </parameter> + <parameter name="out_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of output channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">positions of output channels</doc> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition*"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_mixer_new_with_matrix" c:identifier="gst_audio_channel_mixer_new_with_matrix" moved-to="AudioChannelMixer.new_with_matrix" version="1.14" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">Create a new channel mixer object for the given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">a new #GstAudioChannelMixer object. + Free with gst_audio_channel_mixer_free() after usage.</doc> + <type name="AudioChannelMixer" c:type="GstAudioChannelMixer*"/> + </return-value> + <parameters> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">#GstAudioChannelMixerFlags</doc> + <type name="AudioChannelMixerFlags" c:type="GstAudioChannelMixerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="in_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of input channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">number of output channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="matrix" transfer-ownership="full" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channel-mixer.c">channel conversion matrix, m[@in_channels][@out_channels]. + If identity matrix, passthrough applies. If %NULL, a (potentially truncated) + identity matrix is generated.</doc> + <type name="gfloat" c:type="gfloat**"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_positions_from_mask" c:identifier="gst_audio_channel_positions_from_mask"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Convert the @channels present in @channel_mask to a @position array +(which should have at least @channels entries ensured by caller). +If @channel_mask is set to 0, it is considered as 'not present' for purpose +of conversion. +A partially valid @channel_mask with less bits set than the number +of channels is considered valid.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if channel and channel mask are valid and could be converted</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="channel_mask" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The input channel_mask</doc> + <type name="guint64" c:type="guint64"/> + </parameter> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The + %GstAudioChannelPosition<!-- -->s</doc> + <array length="0" zero-terminated="0" c:type="GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </function> + <function name="audio_channel_positions_to_mask" c:identifier="gst_audio_channel_positions_to_mask"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Convert the @position array of @channels channels to a bitmask. + +If @force_order is %TRUE it additionally checks if the channels are +in the order required by GStreamer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the channel positions are valid and could be converted.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioChannelPositions</doc> + <array length="1" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="force_order" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Only consider the GStreamer channel order.</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + <parameter name="channel_mask" direction="out" caller-allocates="0" transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">the output channel mask</doc> + <type name="guint64" c:type="guint64*"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_positions_to_string" c:identifier="gst_audio_channel_positions_to_string" version="1.10"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Converts @position to a human-readable string representation for +debugging purposes.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">a newly allocated string representing +@position</doc> + <type name="utf8" c:type="gchar*"/> + </return-value> + <parameters> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioChannelPositions + to convert.</doc> + <array length="1" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </function> + <function name="audio_channel_positions_to_valid_order" c:identifier="gst_audio_channel_positions_to_valid_order"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Reorders the channel positions in @position from any order to +the GStreamer channel order.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the channel positions are valid and reordering +was successful.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to + reorder to.</doc> + <array length="1" zero-terminated="0" c:type="GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </function> + <function name="audio_check_valid_channel_positions" c:identifier="gst_audio_check_valid_channel_positions"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Checks if @position contains valid channel positions for +@channels channels. If @force_order is %TRUE it additionally +checks if the channels are in the order required by GStreamer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the channel positions are valid.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioChannelPositions + to check.</doc> + <array length="1" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="force_order" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Only consider the GStreamer channel order.</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </function> + <function name="audio_clipping_meta_api_get_type" c:identifier="gst_audio_clipping_meta_api_get_type"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="GType" c:type="GType"/> + </return-value> + </function> + <function name="audio_clipping_meta_get_info" c:identifier="gst_audio_clipping_meta_get_info" moved-to="AudioClippingMeta.get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + <function name="audio_downmix_meta_api_get_type" c:identifier="gst_audio_downmix_meta_api_get_type"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="GType" c:type="GType"/> + </return-value> + </function> + <function name="audio_downmix_meta_get_info" c:identifier="gst_audio_downmix_meta_get_info" moved-to="AudioDownmixMeta.get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + <function name="audio_format_build_integer" c:identifier="gst_audio_format_build_integer" moved-to="AudioFormat.build_integer"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Construct a #GstAudioFormat with given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format +exists with the given parameters.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </return-value> + <parameters> + <parameter name="sign" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">signed or unsigned format</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + <parameter name="endianness" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">G_LITTLE_ENDIAN or G_BIG_ENDIAN</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="width" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">amount of bits used per sample</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="depth" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" deprecated="1" deprecated-version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Fill @length bytes in @dest with silence samples for @info.</doc> + <doc-deprecated xml:space="preserve">Use gst_audio_format_info_fill_silence() instead.</doc-deprecated> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormatInfo</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </parameter> + <parameter name="dest" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a destination + to fill</doc> + <array length="2" zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="length" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the length to fill</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + </parameters> + </function> + <function name="audio_format_from_string" c:identifier="gst_audio_format_from_string" moved-to="AudioFormat.from_string"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Convert the @format string to its #GstAudioFormat.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the +string is not a known format.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </return-value> + <parameters> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a format string</doc> + <type name="utf8" c:type="const gchar*"/> + </parameter> + </parameters> + </function> + <function name="audio_format_get_info" c:identifier="gst_audio_format_get_info" moved-to="AudioFormat.get_info"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Get the #GstAudioFormatInfo for @format</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">The #GstAudioFormatInfo for @format.</doc> + <type name="AudioFormatInfo" c:type="const GstAudioFormatInfo*"/> + </return-value> + <parameters> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">a #GstAudioFormat</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + </parameters> + </function> + <function name="audio_format_info_get_type" c:identifier="gst_audio_format_info_get_type"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <type name="GType" c:type="GType"/> + </return-value> + </function> + <function name="audio_format_to_string" c:identifier="gst_audio_format_to_string" moved-to="AudioFormat.to_string"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">Return all the raw audio formats supported by GStreamer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Returns a reorder map for @from to @to that can be used in +custom channel reordering code, e.g. to convert from or to the +GStreamer channel order. @from and @to must contain the same +number of positions and the same positions, only in a +different order. + +The resulting @reorder_map can be used for reordering by assigning +channel i of the input to channel reorder_map[i] of the output.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the channel positions are valid and reordering +is possible.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to reorder from.</doc> + <array length="0" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to reorder to.</doc> + <array length="0" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="reorder_map" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Pointer to the reorder map.</doc> + <array length="0" zero-terminated="0" c:type="gint*"> + <type name="gint" c:type="gint"/> + </array> + </parameter> + </parameters> + </function> + <function name="audio_iec61937_frame_size" c:identifier="gst_audio_iec61937_frame_size"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">Calculated the size of the buffer expected by gst_audio_iec61937_payload() for +payloading type from @spec.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">the size or 0 if the given @type is not supported or cannot be +payloaded.</doc> + <type name="guint" c:type="guint"/> + </return-value> + <parameters> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">the ringbufer spec</doc> + <type name="AudioRingBufferSpec" c:type="const GstAudioRingBufferSpec*"/> + </parameter> + </parameters> + </function> + <function name="audio_iec61937_payload" c:identifier="gst_audio_iec61937_payload"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">Payloads @src in the form specified by IEC 61937 for the type from @spec and +stores the result in @dst. @src must contain exactly one frame of data and +the frame is not checked for errors.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">transfer-full: %TRUE if the payloading was successful, %FALSE +otherwise.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="src" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">a buffer containing the data to payload</doc> + <array length="1" zero-terminated="0" c:type="const guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="src_n" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">size of @src in bytes</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="dst" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">the destination buffer to store the + payloaded contents in. Should not overlap with @src</doc> + <array length="3" zero-terminated="0" c:type="guint8*"> + <type name="guint8" c:type="guint8"/> + </array> + </parameter> + <parameter name="dst_n" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">size of @dst in bytes</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="spec" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">the ringbufer spec for @src</doc> + <type name="AudioRingBufferSpec" c:type="const GstAudioRingBufferSpec*"/> + </parameter> + <parameter name="endianness" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioiec61937.c">the expected byte order of the payloaded data</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </function> + <function name="audio_info_from_caps" c:identifier="gst_audio_info_from_caps" moved-to="AudioInfo.from_caps"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Parse @caps and update @info.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">TRUE if @caps could be parsed</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="info" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + <parameter name="caps" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstCaps</doc> + <type name="Gst.Caps" c:type="const GstCaps*"/> + </parameter> + </parameters> + </function> + <function name="audio_info_init" c:identifier="gst_audio_info_init" moved-to="AudioInfo.init"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">Initialize @info with default values.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="info" direction="out" caller-allocates="1" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-info.c">a #GstAudioInfo</doc> + <type name="AudioInfo" c:type="GstAudioInfo*"/> + </parameter> + </parameters> + </function> + <function name="audio_level_meta_api_get_type" c:identifier="gst_audio_level_meta_api_get_type" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Return the #GType associated with #GstAudioLevelMeta.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GType</doc> + <type name="GType" c:type="GType"/> + </return-value> + </function> + <function name="audio_level_meta_get_info" c:identifier="gst_audio_level_meta_get_info" moved-to="AudioLevelMeta.get_info" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Return the #GstMetaInfo associated with #GstAudioLevelMeta.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstMetaInfo</doc> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + <function name="audio_make_raw_caps" c:identifier="gst_audio_make_raw_caps" version="1.18"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.h"/> + <return-value transfer-ownership="full"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">the size of @formats</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="layout" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-format.c">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"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="GType" c:type="GType"/> + </return-value> + </function> + <function name="audio_meta_get_info" c:identifier="gst_audio_meta_get_info" moved-to="AudioMeta.get_info"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <type name="Gst.MetaInfo" c:type="const GstMetaInfo*"/> + </return-value> + </function> + <function name="audio_quantize_new" c:identifier="gst_audio_quantize_new" moved-to="AudioQuantize.new" introspectable="0"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">Create a new quantizer object with the given parameters. + +Output samples will be quantized to a multiple of @quantizer. Better +performance is achieved when @quantizer is a power of 2. + +Dithering and noise-shaping can be performed during quantization with +the @dither and @ns parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.h"/> + <return-value> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a new #GstAudioQuantize. Free with gst_audio_quantize_free().</doc> + <type name="AudioQuantize" c:type="GstAudioQuantize*"/> + </return-value> + <parameters> + <parameter name="dither" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioDitherMethod</doc> + <type name="AudioDitherMethod" c:type="GstAudioDitherMethod"/> + </parameter> + <parameter name="ns" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">a #GstAudioNoiseShapingMethod</doc> + <type name="AudioNoiseShapingMethod" c:type="GstAudioNoiseShapingMethod"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">#GstAudioQuantizeFlags</doc> + <type name="AudioQuantizeFlags" c:type="GstAudioQuantizeFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the #GstAudioFormat of the samples</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the amount of channels in the samples</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="quantizer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-quantize.c">the quantizer to use</doc> + <type name="guint" c:type="guint"/> + </parameter> + </parameters> + </function> + <function name="audio_reorder_channels" c:identifier="gst_audio_reorder_channels"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">Reorders @data from the channel positions @from to the channel +positions @to. @from and @to must contain the same number of +positions and the same positions, only in a different order. + +Note: this function assumes the audio data is in interleaved layout</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">%TRUE if the reordering was possible.</doc> + <type name="gboolean" c:type="gboolean"/> + </return-value> + <parameters> + <parameter name="data" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The pointer to + the memory.</doc> + <array length="1" zero-terminated="0" c:type="gpointer"> + <type name="guint8"/> + </array> + </parameter> + <parameter name="size" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The size of the memory.</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The %GstAudioFormat of the buffer.</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The number of channels.</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions in the buffer.</doc> + <array length="3" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-channels.c">The channel positions to convert to.</doc> + <array length="3" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + </parameters> + </function> + <function name="audio_resampler_new" c:identifier="gst_audio_resampler_new" moved-to="AudioResampler.new"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Make a new resampler.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="full" skip="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">The new #GstAudioResampler.</doc> + <type name="AudioResampler" c:type="GstAudioResampler*"/> + </return-value> + <parameters> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResamplerMethod</doc> + <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/> + </parameter> + <parameter name="flags" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">#GstAudioResamplerFlags</doc> + <type name="AudioResamplerFlags" c:type="GstAudioResamplerFlags"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the #GstAudioFormat</doc> + <type name="AudioFormat" c:type="GstAudioFormat"/> + </parameter> + <parameter name="channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the number of channels</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="options" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">extra options</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </function> + <function name="audio_resampler_options_set_quality" c:identifier="gst_audio_resampler_options_set_quality" moved-to="AudioResampler.options_set_quality"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">Set the parameters for resampling from @in_rate to @out_rate using @method +for @quality in @options.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.h"/> + <return-value transfer-ownership="none"> + <type name="none" c:type="void"/> + </return-value> + <parameters> + <parameter name="method" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstAudioResamplerMethod</doc> + <type name="AudioResamplerMethod" c:type="GstAudioResamplerMethod"/> + </parameter> + <parameter name="quality" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the quality</doc> + <type name="guint" c:type="guint"/> + </parameter> + <parameter name="in_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the input rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="out_rate" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">the output rate</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="options" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/audio-resampler.c">a #GstStructure</doc> + <type name="Gst.Structure" c:type="GstStructure*"/> + </parameter> + </parameters> + </function> + <function name="buffer_add_audio_clipping_meta" c:identifier="gst_buffer_add_audio_clipping_meta" version="1.8"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Attaches #GstAudioClippingMeta metadata to @buffer with the given parameters.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioClippingMeta on @buffer.</doc> + <type name="AudioClippingMeta" c:type="GstAudioClippingMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="format" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples</doc> + <type name="Gst.Format" c:type="GstFormat"/> + </parameter> + <parameter name="start" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Amount of audio to clip from start of buffer</doc> + <type name="guint64" c:type="guint64"/> + </parameter> + <parameter name="end" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Amount of to clip from end of buffer</doc> + <type name="guint64" c:type="guint64"/> + </parameter> + </parameters> + </function> + <function name="buffer_add_audio_downmix_meta" c:identifier="gst_buffer_add_audio_downmix_meta"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Attaches #GstAudioDownmixMeta metadata to @buffer with the given parameters. + +@matrix is an two-dimensional array of @to_channels times @from_channels +coefficients, i.e. the i-th output channels is constructed by multiplicating +the input channels with the coefficients in @matrix[i] and taking the sum +of the results.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioDownmixMeta on @buffer.</doc> + <type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="from_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the channel positions + of the source</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="from_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">The number of channels of the source</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="to_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the channel positions of + the destination</doc> + <array length="4" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">The number of channels of the destination</doc> + <type name="gint" c:type="gint"/> + </parameter> + <parameter name="matrix" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">The matrix coefficients.</doc> + <type name="gfloat" c:type="const gfloat**"/> + </parameter> + </parameters> + </function> + <function name="buffer_add_audio_level_meta" c:identifier="gst_buffer_add_audio_level_meta" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Attaches audio level information to @buffer. (RFC 6464)</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioLevelMeta on @buffer.</doc> + <type name="AudioLevelMeta" c:type="GstAudioLevelMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="level" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the -dBov from 0-127 (127 is silence).</doc> + <type name="guint8" c:type="guint8"/> + </parameter> + <parameter name="voice_activity" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">whether the buffer contains voice activity.</doc> + <type name="gboolean" c:type="gboolean"/> + </parameter> + </parameters> + </function> + <function name="buffer_add_audio_meta" c:identifier="gst_buffer_add_audio_meta" version="1.16"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Allocates and attaches a #GstAudioMeta on @buffer, which must be writable +for that purpose. The fields of the #GstAudioMeta are directly populated +from the arguments of this function. + +When @info->layout is %GST_AUDIO_LAYOUT_NON_INTERLEAVED and @offsets is +%NULL, the offsets are calculated with a formula that assumes the planes are +tightly packed and in sequence: +offsets[channel] = channel * @samples * sample_stride + +It is not allowed for channels to overlap in memory, +i.e. for each i in [0, channels), the range +[@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap +with any other such range. This function will assert if the parameters +specified cause this restriction to be violated. + +It is, obviously, also not allowed to specify parameters that would cause +out-of-bounds memory access on @buffer. This is also checked, which means +that you must add enough memory on the @buffer before adding this meta.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioMeta that was attached on the @buffer</doc> + <type name="AudioMeta" c:type="GstAudioMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="info" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the audio properties of the buffer</doc> + <type name="AudioInfo" c:type="const GstAudioInfo*"/> + </parameter> + <parameter name="samples" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the number of valid samples in the buffer</doc> + <type name="gsize" c:type="gsize"/> + </parameter> + <parameter name="offsets" transfer-ownership="none" nullable="1" allow-none="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the offsets (in bytes) where each channel plane starts + in the buffer or %NULL to calculate it (see below); must be %NULL also + when @info->layout is %GST_AUDIO_LAYOUT_INTERLEAVED</doc> + <type name="gsize" c:type="gsize*"/> + </parameter> + </parameters> + </function> + <function-macro name="buffer_get_audio_clipping_meta" c:identifier="gst_buffer_get_audio_clipping_meta" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function-macro name="buffer_get_audio_downmix_meta" c:identifier="gst_buffer_get_audio_downmix_meta" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function name="buffer_get_audio_downmix_meta_for_channels" c:identifier="gst_buffer_get_audio_downmix_meta_for_channels"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Find the #GstAudioDownmixMeta on @buffer for the given destination +channel positions.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioDownmixMeta on @buffer.</doc> + <type name="AudioDownmixMeta" c:type="GstAudioDownmixMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + <parameter name="to_position" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the channel positions of + the destination</doc> + <array length="2" zero-terminated="0" c:type="const GstAudioChannelPosition*"> + <type name="AudioChannelPosition" c:type="GstAudioChannelPosition"/> + </array> + </parameter> + <parameter name="to_channels" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">The number of channels of the destination</doc> + <type name="gint" c:type="gint"/> + </parameter> + </parameters> + </function> + <function name="buffer_get_audio_level_meta" c:identifier="gst_buffer_get_audio_level_meta" version="1.20"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">Find the #GstAudioLevelMeta on @buffer.</doc> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <return-value transfer-ownership="none" nullable="1"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">the #GstAudioLevelMeta or %NULL when +there is no such metadata on @buffer.</doc> + <type name="AudioLevelMeta" c:type="GstAudioLevelMeta*"/> + </return-value> + <parameters> + <parameter name="buffer" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.c">a #GstBuffer</doc> + <type name="Gst.Buffer" c:type="GstBuffer*"/> + </parameter> + </parameters> + </function> + <function-macro name="buffer_get_audio_meta" c:identifier="gst_buffer_get_audio_meta" introspectable="0"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiometa.h"/> + <parameters> + <parameter name="b"> + </parameter> + </parameters> + </function-macro> + <function name="stream_volume_convert_volume" c:identifier="gst_stream_volume_convert_volume" moved-to="StreamVolume.convert_volume"> + <source-position filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.h"/> + <return-value transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">the converted volume</doc> + <type name="gdouble" c:type="gdouble"/> + </return-value> + <parameters> + <parameter name="from" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat to convert from</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + <parameter name="to" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">#GstStreamVolumeFormat to convert to</doc> + <type name="StreamVolumeFormat" c:type="GstStreamVolumeFormat"/> + </parameter> + <parameter name="val" transfer-ownership="none"> + <doc xml:space="preserve" filename="../subprojects/gst-plugins-base/gst-libs/gst/audio/streamvolume.c">Volume in @from format that should be converted</doc> + <type name="gdouble" c:type="gdouble"/> + </parameter> + </parameters> + </function> + </namespace> +</repository> |