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23c774e5a3
Original commit message from CVS: reviewed by Benjamin Otte <otte@gnome.org> * ext/a52dec/gsta52dec.c: * ext/divx/gstdivxdec.c: * ext/divx/gstdivxenc.c: * ext/dts/gstdtsdec.c: (gst_dtsdec_base_init): * ext/faac/gstfaac.c: (gst_faac_base_init): * ext/faad/gstfaad.c: (gst_faad_base_init): * ext/ivorbis/vorbisfile.c: * ext/lame/gstlame.c: * ext/libfame/gstlibfame.c: * ext/mpeg2enc/gstmpeg2enc.cc: * ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init): * ext/sidplay/gstsiddec.cc: * ext/speex/gstspeexdec.c: * ext/speex/gstspeexenc.c: * ext/xvid/gstxviddec.c: * ext/xvid/gstxvidenc.c: correct klasses. Mostly s,Codec/(Audio|Video),\1/Codec, (fixes #142193)
450 lines
12 KiB
C
450 lines
12 KiB
C
/* GStreamer FAAD (Free AAC Decoder) plugin
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* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstfaad.h"
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GstStaticPadTemplate sink_template = 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/mpeg, "
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"systemstream = (bool) FALSE, " "mpegversion = { (int) 2, (int) 4 }")
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);
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GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"endianness = (int) BYTE_ORDER, "
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"signed = (bool) TRUE, "
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"width = (int) { 16, 24, 32 }, "
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"depth = (int) { 16, 24, 32 }, "
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"rate = (int) [ 8000, 96000 ], "
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"channels = (int) [ 1, 6 ]; "
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"audio/x-raw-float, "
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"endianness = (int) BYTE_ORDER, "
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"depth = (int) { 32, 64 }, "
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"rate = (int) [ 8000, 96000 ], " "channels = (int) [ 1, 6 ]")
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);
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static void gst_faad_base_init (GstFaadClass * klass);
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static void gst_faad_class_init (GstFaadClass * klass);
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static void gst_faad_init (GstFaad * faad);
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static GstPadLinkReturn
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gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps);
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static GstPadLinkReturn
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gst_faad_srcconnect (GstPad * pad, const GstCaps * caps);
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static GstCaps *gst_faad_srcgetcaps (GstPad * pad);
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static void gst_faad_chain (GstPad * pad, GstData * data);
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static GstElementStateReturn gst_faad_change_state (GstElement * element);
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static GstElementClass *parent_class = NULL;
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/* static guint gst_faad_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_faad_get_type (void)
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{
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static GType gst_faad_type = 0;
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if (!gst_faad_type) {
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static const GTypeInfo gst_faad_info = {
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sizeof (GstFaadClass),
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(GBaseInitFunc) gst_faad_base_init,
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NULL,
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(GClassInitFunc) gst_faad_class_init,
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NULL,
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NULL,
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sizeof (GstFaad),
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0,
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(GInstanceInitFunc) gst_faad_init,
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};
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gst_faad_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstFaad", &gst_faad_info, 0);
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}
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return gst_faad_type;
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}
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static void
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gst_faad_base_init (GstFaadClass * klass)
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{
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GstElementDetails gst_faad_details = {
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"Free AAC Decoder (FAAD)",
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"Codec/Decoder/Audio",
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"Free MPEG-2/4 AAC decoder",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>",
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};
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_details (element_class, &gst_faad_details);
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}
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static void
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gst_faad_class_init (GstFaadClass * klass)
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{
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
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gstelement_class->change_state = gst_faad_change_state;
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}
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static void
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gst_faad_init (GstFaad * faad)
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{
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faad->handle = NULL;
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faad->samplerate = -1;
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faad->channels = -1;
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GST_FLAG_SET (faad, GST_ELEMENT_EVENT_AWARE);
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faad->sinkpad =
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gst_pad_new_from_template (gst_static_pad_template_get (&sink_template),
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"sink");
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gst_element_add_pad (GST_ELEMENT (faad), faad->sinkpad);
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gst_pad_set_chain_function (faad->sinkpad, gst_faad_chain);
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gst_pad_set_link_function (faad->sinkpad, gst_faad_sinkconnect);
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faad->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get (&src_template),
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"src");
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gst_element_add_pad (GST_ELEMENT (faad), faad->srcpad);
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gst_pad_set_link_function (faad->srcpad, gst_faad_srcconnect);
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gst_pad_set_getcaps_function (faad->srcpad, gst_faad_srcgetcaps);
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}
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static GstPadLinkReturn
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gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
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{
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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GstStructure *str = gst_caps_get_structure (caps, 0);
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const GValue *value;
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GstBuffer *buf;
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if ((value = gst_structure_get_value (str, "codec_data"))) {
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GstPadLinkReturn ret;
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gulong samplerate;
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guchar channels;
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buf = g_value_get_boxed (value);
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/* someone forgot that char can be unsigned when writing the API */
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if ((gint8) faacDecInit2 (faad->handle, GST_BUFFER_DATA (buf),
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GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
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return GST_PAD_LINK_REFUSED;
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faad->samplerate = samplerate;
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faad->channels = channels;
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ret = gst_pad_renegotiate (faad->srcpad);
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if (ret == GST_PAD_LINK_DELAYED)
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ret = GST_PAD_LINK_OK;
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return ret;
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}
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/* if there's no decoderspecificdata, it's all fine. We cannot know
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* much more at this point... */
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return GST_PAD_LINK_OK;
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}
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static GstCaps *
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gst_faad_srcgetcaps (GstPad * pad)
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{
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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if (faad->handle != NULL && faad->channels != -1 && faad->samplerate != -1) {
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GstCaps *caps = gst_caps_new_empty ();
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GstStructure *str;
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gint fmt[] = {
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FAAD_FMT_16BIT,
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FAAD_FMT_24BIT,
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FAAD_FMT_32BIT,
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FAAD_FMT_FLOAT,
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FAAD_FMT_DOUBLE,
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-1
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}
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, n;
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for (n = 0; fmt[n] != -1; n++) {
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switch (n) {
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case FAAD_FMT_16BIT:
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str = gst_structure_new ("audio/x-raw-int",
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
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break;
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case FAAD_FMT_24BIT:
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str = gst_structure_new ("audio/x-raw-int",
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 24, "depth", G_TYPE_INT, 24, NULL);
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break;
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case FAAD_FMT_32BIT:
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str = gst_structure_new ("audio/x-raw-int",
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"signed", G_TYPE_BOOLEAN, TRUE,
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"width", G_TYPE_INT, 32, "depth", G_TYPE_INT, 32, NULL);
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break;
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case FAAD_FMT_FLOAT:
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str = gst_structure_new ("audio/x-raw-float",
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"depth", G_TYPE_INT, 32, NULL);
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break;
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case FAAD_FMT_DOUBLE:
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str = gst_structure_new ("audio/x-raw-float",
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"depth", G_TYPE_INT, 64, NULL);
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break;
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default:
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str = NULL;
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break;
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}
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if (!str)
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continue;
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if (faad->samplerate != -1) {
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gst_structure_set (str, "rate", G_TYPE_INT, faad->samplerate, NULL);
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} else {
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gst_structure_set (str, "rate", GST_TYPE_INT_RANGE, 8000, 96000, NULL);
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}
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if (faad->channels != -1) {
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gst_structure_set (str, "channels", G_TYPE_INT, faad->channels, NULL);
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} else {
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gst_structure_set (str, "channels", GST_TYPE_INT_RANGE, 1, 6, NULL);
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}
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gst_structure_set (str, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
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gst_caps_append_structure (caps, str);
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}
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return caps;
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}
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return gst_caps_copy (GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad)));
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}
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static GstPadLinkReturn
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gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
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{
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GstStructure *structure;
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const gchar *mimetype;
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gint fmt = -1;
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gint depth, rate, channels;
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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if (!faad->handle || (faad->samplerate == -1 || faad->channels == -1)) {
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return GST_PAD_LINK_DELAYED;
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}
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structure = gst_caps_get_structure (caps, 0);
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mimetype = gst_structure_get_name (structure);
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/* Samplerate and channels are normally provided through
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* the getcaps function */
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "rate", &rate) ||
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rate != faad->samplerate || channels != faad->channels) {
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return GST_PAD_LINK_REFUSED;
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}
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if (!strcmp (mimetype, "audio/x-raw-int")) {
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gint width;
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if (!gst_structure_get_int (structure, "depth", &depth) ||
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!gst_structure_get_int (structure, "width", &width))
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return GST_PAD_LINK_REFUSED;
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if (depth != width)
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return GST_PAD_LINK_REFUSED;
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switch (depth) {
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case 16:
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fmt = FAAD_FMT_16BIT;
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break;
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case 24:
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fmt = FAAD_FMT_24BIT;
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break;
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case 32:
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fmt = FAAD_FMT_32BIT;
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break;
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}
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} else {
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if (!gst_structure_get_int (structure, "depth", &depth))
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return GST_PAD_LINK_REFUSED;
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switch (depth) {
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case 32:
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fmt = FAAD_FMT_FLOAT;
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break;
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case 64:
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fmt = FAAD_FMT_DOUBLE;
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break;
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}
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}
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if (fmt != -1) {
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faacDecConfiguration *conf;
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conf = faacDecGetCurrentConfiguration (faad->handle);
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conf->outputFormat = fmt;
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faacDecSetConfiguration (faad->handle, conf);
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/* FIXME: handle return value, how? */
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faad->bps = depth / 8;
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return GST_PAD_LINK_OK;
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}
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return GST_PAD_LINK_REFUSED;
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}
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static void
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gst_faad_chain (GstPad * pad, GstData * data)
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{
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GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
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GstBuffer *buf, *outbuf;
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faacDecFrameInfo info;
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void *out;
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if (GST_IS_EVENT (data)) {
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GstEvent *event = GST_EVENT (data);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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gst_element_set_eos (GST_ELEMENT (faad));
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gst_pad_push (faad->srcpad, data);
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return;
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default:
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gst_pad_event_default (pad, event);
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return;
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}
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}
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buf = GST_BUFFER (data);
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if (faad->samplerate == -1 || faad->channels == -1) {
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GstPadLinkReturn ret;
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gulong samplerate;
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guchar channels;
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faacDecInit (faad->handle,
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GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), &samplerate, &channels);
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faad->samplerate = samplerate;
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faad->channels = channels;
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ret = gst_pad_renegotiate (faad->srcpad);
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if (GST_PAD_LINK_FAILED (ret)) {
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GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
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gst_buffer_unref (buf);
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return;
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}
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}
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out = faacDecDecode (faad->handle, &info,
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GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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if (info.error) {
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GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
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("Failed to decode buffer: %s", faacDecGetErrorMessage (info.error)));
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gst_buffer_unref (buf);
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return;
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}
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if (info.samplerate != faad->samplerate || info.channels != faad->channels) {
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GstPadLinkReturn ret;
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faad->samplerate = info.samplerate;
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faad->channels = info.channels;
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ret = gst_pad_renegotiate (faad->srcpad);
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if (GST_PAD_LINK_FAILED (ret)) {
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GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL), (NULL));
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gst_buffer_unref (buf);
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return;
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}
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}
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if (info.samples == 0) {
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gst_buffer_unref (buf);
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return;
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}
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/* FIXME: did it handle the whole buffer? */
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outbuf = gst_buffer_new_and_alloc (info.samples * faad->bps);
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/* ugh */
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memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
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GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
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GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (buf);
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gst_buffer_unref (buf);
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gst_pad_push (faad->srcpad, GST_DATA (outbuf));
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}
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static GstElementStateReturn
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gst_faad_change_state (GstElement * element)
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{
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GstFaad *faad = GST_FAAD (element);
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switch (GST_STATE_TRANSITION (element)) {
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case GST_STATE_NULL_TO_READY:
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if (!(faad->handle = faacDecOpen ()))
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return GST_STATE_FAILURE;
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else {
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faacDecConfiguration *conf;
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conf = faacDecGetCurrentConfiguration (faad->handle);
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conf->defObjectType = LC;
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faacDecSetConfiguration (faad->handle, conf);
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}
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break;
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case GST_STATE_PAUSED_TO_READY:
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faad->samplerate = -1;
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faad->channels = -1;
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break;
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case GST_STATE_READY_TO_NULL:
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faacDecClose (faad->handle);
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faad->handle = NULL;
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break;
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default:
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break;
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}
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if (GST_ELEMENT_CLASS (parent_class)->change_state)
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return GST_ELEMENT_CLASS (parent_class)->change_state (element);
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return GST_STATE_SUCCESS;
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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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return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
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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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"faad",
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"Free AAC Decoder (FAAD)",
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plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN)
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