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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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638225aebb
Original commit message from CVS: * gst/auparse/gstauparse.c: (gst_au_parse_chain): Don't push buffers into the adapter that we are going to push downstream again without framing anyway. Also, the adaptor takes ownership of buffers put into it (fixes auparse pushing invalid buffers for .au files with ADPCM contents). Finally, set caps on all outgoing buffers.
428 lines
13 KiB
C
428 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* Element-Checklist-Version: 5 */
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/* 2001/04/03 - Updated parseau to use caps nego
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* Zaheer Abbas Merali <zaheerabbas at merali dot org>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include "gstauparse.h"
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#include <gst/audio/audio.h>
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/* elementfactory information */
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static GstElementDetails gst_au_parse_details =
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GST_ELEMENT_DETAILS (".au parser",
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"Codec/Demuxer/Audio",
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"Parse an .au file into raw audio",
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"Erik Walthinsen <omega@cse.ogi.edu>");
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static GstStaticPadTemplate gst_au_parse_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-au")
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);
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#define GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS \
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"audio/x-alaw, " \
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"rate = (int) [ 8000, 192000 ], " \
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"channels = (int) [ 1, 2 ]"
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#define GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS \
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"audio/x-mulaw, " \
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"rate = (int) [ 8000, 192000 ], " \
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"channels = (int) [ 1, 2 ]"
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/* Nothing to decode those ADPCM streams for now */
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#define GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS \
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"audio/x-adpcm, " \
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"layout = (string) { g721, g722, g723_3, g723_5 }"
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static GstStaticPadTemplate gst_au_parse_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS "; "
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GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS ";"
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GST_AU_PARSE_ALAW_PAD_TEMPLATE_CAPS ";"
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GST_AU_PARSE_MULAW_PAD_TEMPLATE_CAPS ";"
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GST_AU_PARSE_ADPCM_PAD_TEMPLATE_CAPS));
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static void gst_au_parse_dispose (GObject * object);
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static GstFlowReturn gst_au_parse_chain (GstPad * pad, GstBuffer * buf);
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static GstStateChangeReturn gst_au_parse_change_state (GstElement * element,
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GstStateChange transition);
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GST_BOILERPLATE (GstAuParse, gst_au_parse, GstElement, GST_TYPE_ELEMENT)
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static void gst_au_parse_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_au_parse_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_au_parse_src_template));
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gst_element_class_set_details (element_class, &gst_au_parse_details);
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}
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static void
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gst_au_parse_class_init (GstAuParseClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->dispose = gst_au_parse_dispose;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_au_parse_change_state);
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}
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static void
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gst_au_parse_init (GstAuParse * auparse, GstAuParseClass * klass)
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{
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auparse->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_au_parse_sink_template), "sink");
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gst_element_add_pad (GST_ELEMENT (auparse), auparse->sinkpad);
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gst_pad_set_chain_function (auparse->sinkpad, gst_au_parse_chain);
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auparse->srcpad = gst_pad_new_from_template (gst_static_pad_template_get
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(&gst_au_parse_src_template), "src");
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gst_pad_use_fixed_caps (auparse->srcpad);
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gst_element_add_pad (GST_ELEMENT (auparse), auparse->srcpad);
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auparse->offset = 0;
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auparse->buffer_offset = 0;
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auparse->adapter = gst_adapter_new ();
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auparse->size = 0;
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auparse->encoding = 0;
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auparse->frequency = 0;
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auparse->channels = 0;
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}
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static void
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gst_au_parse_dispose (GObject * object)
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{
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GstAuParse *au = GST_AU_PARSE (object);
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if (au->adapter != NULL) {
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g_object_unref (au->adapter);
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au->adapter = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static GstFlowReturn
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gst_au_parse_chain (GstPad * pad, GstBuffer * buf)
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{
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GstFlowReturn ret;
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GstAuParse *auparse;
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guchar *data;
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glong size;
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GstCaps *tempcaps;
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gint law = 0, depth = 0, ieee = 0;
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gchar layout[7];
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GstBuffer *databuf;
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GstEvent *event;
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layout[0] = 0;
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auparse = GST_AU_PARSE (gst_pad_get_parent (pad));
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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GST_DEBUG_OBJECT (auparse, "got buffer of size %ld", size);
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/* if we haven't seen any data yet... */
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if (auparse->size == 0) {
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guint32 *head = (guint32 *) data;
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/* normal format is big endian (au is a Sparc format) */
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if (GST_READ_UINT32_BE (head) == 0x2e736e64) { /* ".snd" */
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head++;
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auparse->le = 0;
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auparse->offset = GST_READ_UINT32_BE (head);
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head++;
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/* Do not trust size, could be set to -1 : unknown */
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auparse->size = GST_READ_UINT32_BE (head);
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head++;
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auparse->encoding = GST_READ_UINT32_BE (head);
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head++;
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auparse->frequency = GST_READ_UINT32_BE (head);
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head++;
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auparse->channels = GST_READ_UINT32_BE (head);
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head++;
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/* and of course, someone had to invent a little endian
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* version. Used by DEC systems. */
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} else if (GST_READ_UINT32_LE (head) == 0x0064732E) { /* other source say it is "dns." */
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head++;
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auparse->le = 1;
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auparse->offset = GST_READ_UINT32_LE (head);
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head++;
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/* Do not trust size, could be set to -1 : unknown */
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auparse->size = GST_READ_UINT32_LE (head);
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head++;
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auparse->encoding = GST_READ_UINT32_LE (head);
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head++;
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auparse->frequency = GST_READ_UINT32_LE (head);
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head++;
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auparse->channels = GST_READ_UINT32_LE (head);
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head++;
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} else {
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GST_ELEMENT_ERROR (auparse, STREAM, WRONG_TYPE, (NULL), (NULL));
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gst_buffer_unref (buf);
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gst_object_unref (auparse);
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return GST_FLOW_ERROR;
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}
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GST_DEBUG
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("offset %ld, size %ld, encoding %ld, frequency %ld, channels %ld\n",
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auparse->offset, auparse->size, auparse->encoding, auparse->frequency,
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auparse->channels);
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/*
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Docs :
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http://www.opengroup.org/public/pubs/external/auformat.html
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http://astronomy.swin.edu.au/~pbourke/dataformats/au/
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Solaris headers : /usr/include/audio/au.h
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libsndfile : src/au.c
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Samples :
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http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/AU/Samples.html
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*/
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switch (auparse->encoding) {
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case 1: /* 8-bit ISDN mu-law G.711 */
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law = 1;
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depth = 8;
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break;
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case 27: /* 8-bit ISDN A-law G.711 */
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law = 2;
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depth = 8;
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break;
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case 2: /* 8-bit linear PCM */
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depth = 8;
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break;
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case 3: /* 16-bit linear PCM */
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depth = 16;
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break;
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case 4: /* 24-bit linear PCM */
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depth = 24;
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break;
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case 5: /* 32-bit linear PCM */
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depth = 32;
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break;
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case 6: /* 32-bit IEEE floating point */
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ieee = 1;
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depth = 32;
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break;
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case 7: /* 64-bit IEEE floating point */
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ieee = 1;
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depth = 64;
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break;
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case 23: /* 4-bit CCITT G.721 ADPCM 32kbps -> modplug/libsndfile (compressed 8-bit mu-law) */
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strcpy (layout, "g721");
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break;
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case 24: /* 8-bit CCITT G.722 ADPCM -> rtp */
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strcpy (layout, "g722");
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break;
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case 25: /* 3-bit CCITT G.723.3 ADPCM 24kbps -> rtp/xine/modplug/libsndfile */
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strcpy (layout, "g723_3");
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break;
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case 26: /* 5-bit CCITT G.723.5 ADPCM 40kbps -> rtp/xine/modplug/libsndfile */
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strcpy (layout, "g723_5");
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break;
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case 8: /* Fragmented sample data */
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case 9: /* AU_ENCODING_NESTED */
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case 10: /* DSP program */
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case 11: /* DSP 8-bit fixed point */
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case 12: /* DSP 16-bit fixed point */
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case 13: /* DSP 24-bit fixed point */
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case 14: /* DSP 32-bit fixed point */
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case 16: /* AU_ENCODING_DISPLAY : non-audio display data */
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case 17: /* AU_ENCODING_MULAW_SQUELCH */
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case 18: /* 16-bit linear with emphasis */
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case 19: /* 16-bit linear compressed (NeXT) */
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case 20: /* 16-bit linear with emphasis and compression */
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case 21: /* Music kit DSP commands */
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case 22: /* Music kit DSP commands samples */
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default:
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GST_ELEMENT_ERROR (auparse, STREAM, FORMAT, (NULL), (NULL));
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gst_buffer_unref (buf);
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gst_object_unref (auparse);
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return GST_FLOW_ERROR;
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}
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if (law) {
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tempcaps =
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gst_caps_new_simple ((law == 1) ? "audio/x-mulaw" : "audio/x-alaw",
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"rate", G_TYPE_INT, auparse->frequency,
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"channels", G_TYPE_INT, auparse->channels, NULL);
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auparse->sample_size = auparse->channels;
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} else if (ieee) {
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tempcaps = gst_caps_new_simple ("audio/x-raw-float",
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"rate", G_TYPE_INT, auparse->frequency,
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"channels", G_TYPE_INT, auparse->channels,
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"endianness", G_TYPE_INT,
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auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN,
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"width", G_TYPE_INT, depth, NULL);
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auparse->sample_size = auparse->channels * depth / 8;
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} else if (layout[0]) {
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tempcaps = gst_caps_new_simple ("audio/x-adpcm",
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"layout", G_TYPE_STRING, layout, NULL);
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auparse->sample_size = 0;
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} else {
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tempcaps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, auparse->frequency,
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"channels", G_TYPE_INT, auparse->channels,
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"endianness", G_TYPE_INT,
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auparse->le ? G_LITTLE_ENDIAN : G_BIG_ENDIAN, "depth", G_TYPE_INT,
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depth, "width", G_TYPE_INT, depth, "signed", G_TYPE_BOOLEAN, TRUE,
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NULL);
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auparse->sample_size = auparse->channels * depth / 8;
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}
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gst_pad_use_fixed_caps (auparse->srcpad);
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gst_pad_set_caps (auparse->srcpad, tempcaps);
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gst_pad_set_active (auparse->srcpad, TRUE);
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event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_DEFAULT,
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0, GST_CLOCK_TIME_NONE, 0);
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gst_pad_push_event (auparse->srcpad, event);
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databuf = gst_buffer_create_sub (buf, auparse->offset,
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size - auparse->offset);
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gst_buffer_unref (buf);
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buf = NULL;
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} else {
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databuf = buf;
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buf = NULL;
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}
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if (auparse->sample_size) {
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int avail;
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gst_adapter_push (auparse->adapter, databuf);
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/* Ensure we push a buffer that's a multiple of the frame size downstream */
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avail = gst_adapter_available (auparse->adapter);
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avail -= avail % auparse->sample_size;
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if (avail > 0) {
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const guint8 *data = gst_adapter_peek (auparse->adapter, avail);
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GstBuffer *newbuf;
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if ((ret =
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gst_pad_alloc_buffer_and_set_caps (auparse->srcpad,
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auparse->buffer_offset, avail, GST_PAD_CAPS (auparse->srcpad),
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&newbuf)) == GST_FLOW_OK) {
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memcpy (GST_BUFFER_DATA (newbuf), data, avail);
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gst_adapter_flush (auparse->adapter, avail);
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auparse->buffer_offset += avail;
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ret = gst_pad_push (auparse->srcpad, newbuf);
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}
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} else {
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ret = GST_FLOW_OK;
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}
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} else {
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/* It's something non-trivial (such as ADPCM), we don't understand it, so
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* just push downstream and assume this will know what to do with it */
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gst_buffer_set_caps (databuf, GST_PAD_CAPS (auparse->srcpad));
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ret = gst_pad_push (auparse->srcpad, databuf);
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}
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gst_object_unref (auparse);
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return ret;
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}
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static GstStateChangeReturn
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gst_au_parse_change_state (GstElement * element, GstStateChange transition)
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{
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GstAuParse *auparse = GST_AU_PARSE (element);
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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ret = parent_class->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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return ret;
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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gst_adapter_clear (auparse->adapter);
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auparse->buffer_offset = 0;
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auparse->offset = 0;
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auparse->size = 0;
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auparse->encoding = 0;
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auparse->frequency = 0;
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auparse->channels = 0;
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default:
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break;
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}
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return ret;
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}
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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if (!gst_element_register (plugin, "auparse", GST_RANK_SECONDARY,
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GST_TYPE_AU_PARSE)) {
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return FALSE;
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}
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return TRUE;
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"auparse",
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"parses au streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
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GST_PACKAGE_ORIGIN)
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