mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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2081f9ad65
Original commit message from CVS: Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
650 lines
17 KiB
C
650 lines
17 KiB
C
/* GStreamer
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* Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.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 <stdlib.h>
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#ifdef HAVE_STDINT_H
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#include <stdint.h>
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#else
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/* (Ronald) hacky... can't include stdint.h because it's not available
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* everywhere. however, a52dec wants uint8_t/uint32_t... how? */
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#ifndef __uint8_t_defined
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#define __uint8_t_defined
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typedef guint8 uint8_t;
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#endif
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#ifndef __uint32_t_defined
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#define __uint32_t_defined
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typedef guint32 uint32_t;
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#endif
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/* grosj... but it works (tm) */
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#endif /* HAVE_STDINT_H */
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#include <gst/gst.h>
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#include <a52dec/a52.h>
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#include <a52dec/mm_accel.h>
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#include "gsta52dec.h"
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/* elementfactory information */
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static GstElementDetails gst_a52dec_details = {
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"ATSC A/52 audio decoder",
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"Codec/Audio/Decoder",
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"Decodes ATSC A/52 encoded audio streams",
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"David I. Lehn <dlehn@users.sourceforge.net>",
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};
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/* A52Dec signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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ARG_DRC,
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ARG_STREAMINFO
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};
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/*
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* "audio/a52" and "audio/ac3" are the same format. The name
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* "ac3" is now deprecated and should not be used in new code.
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*/
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GST_PAD_TEMPLATE_FACTORY (sink_factory,
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"a52dec_sink",
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"audio/x-ac3",
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NULL
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)
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);
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GST_PAD_TEMPLATE_FACTORY (src_factory,
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"src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_CAPS_NEW (
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"a52dec_src",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN (TRUE),
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"width", GST_PROPS_INT (16),
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"depth", GST_PROPS_INT (16),
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"rate", GST_PROPS_INT_RANGE (4000, 48000),
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"channels", GST_PROPS_INT_RANGE (1, 6)
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)
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);
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static void gst_a52dec_base_init (gpointer g_class);
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static void gst_a52dec_class_init (GstA52DecClass * klass);
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static void gst_a52dec_init (GstA52Dec * a52dec);
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static void gst_a52dec_loop (GstElement * element);
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static GstElementStateReturn
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gst_a52dec_change_state (GstElement * element);
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static void gst_a52dec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_a52dec_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstElementClass *parent_class = NULL;
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/* static guint gst_a52dec_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_a52dec_get_type (void)
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{
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static GType a52dec_type = 0;
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if (!a52dec_type) {
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static const GTypeInfo a52dec_info = {
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sizeof (GstA52DecClass),
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gst_a52dec_base_init,
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NULL, (GClassInitFunc) gst_a52dec_class_init,
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NULL,
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NULL,
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sizeof (GstA52Dec),
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0,
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(GInstanceInitFunc) gst_a52dec_init,
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};
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a52dec_type = g_type_register_static (GST_TYPE_ELEMENT, "GstA52Dec", &a52dec_info, 0);
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}
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return a52dec_type;
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}
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static void
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gst_a52dec_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_PAD_TEMPLATE_GET (sink_factory));
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gst_element_class_add_pad_template (element_class,
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GST_PAD_TEMPLATE_GET (src_factory));
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gst_element_class_set_details (element_class, &gst_a52dec_details);
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}
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static void
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gst_a52dec_class_init (GstA52DecClass * 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_ref (GST_TYPE_ELEMENT);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
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g_param_spec_boolean ("drc", "Dynamic Range Compression",
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"Use Dynamic Range Compression", FALSE,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_STREAMINFO,
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g_param_spec_boxed ("streaminfo", "Streaminfo", "Streaminfo",
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GST_TYPE_CAPS, G_PARAM_READABLE));
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gobject_class->set_property = gst_a52dec_set_property;
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gobject_class->get_property = gst_a52dec_get_property;
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gstelement_class->change_state = gst_a52dec_change_state;
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}
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static void
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gst_a52dec_init (GstA52Dec * a52dec)
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{
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/* create the sink and src pads */
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a52dec->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
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gst_element_set_loop_function ((GstElement *) a52dec, gst_a52dec_loop);
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a52dec->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
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gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
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a52dec->dynamic_range_compression = FALSE;
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a52dec->streaminfo = NULL;
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}
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/* BEGIN modified a52dec conversion code */
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static inline int16_t
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convert (int32_t i)
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{
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if (i > 0x43c07fff)
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return 32767;
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else if (i < 0x43bf8000)
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return -32768;
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else
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return i - 0x43c00000;
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}
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static inline void
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float_to_int (float *_f, int16_t * s16, int flags)
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{
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int i;
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int32_t *f = (int32_t *) _f;
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switch (flags) {
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case A52_MONO:
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for (i = 0; i < 256; i++) {
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s16[5 * i] = s16[5 * i + 1] = s16[5 * i + 2] = s16[5 * i + 3] = 0;
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s16[5 * i + 4] = convert (f[i]);
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}
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break;
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case A52_CHANNEL:
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case A52_STEREO:
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case A52_DOLBY:
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for (i = 0; i < 256; i++) {
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s16[2 * i] = convert (f[i]);
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s16[2 * i + 1] = convert (f[i + 256]);
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}
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break;
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case A52_3F:
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for (i = 0; i < 256; i++) {
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s16[5 * i] = convert (f[i]);
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s16[5 * i + 1] = convert (f[i + 512]);
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s16[5 * i + 2] = s16[5 * i + 3] = 0;
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s16[5 * i + 4] = convert (f[i + 256]);
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}
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break;
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case A52_2F2R:
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for (i = 0; i < 256; i++) {
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s16[4 * i] = convert (f[i]);
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s16[4 * i + 1] = convert (f[i + 256]);
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s16[4 * i + 2] = convert (f[i + 512]);
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s16[4 * i + 3] = convert (f[i + 768]);
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}
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break;
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case A52_3F2R:
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for (i = 0; i < 256; i++) {
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s16[5 * i] = convert (f[i]);
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s16[5 * i + 1] = convert (f[i + 512]);
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s16[5 * i + 2] = convert (f[i + 768]);
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s16[5 * i + 3] = convert (f[i + 1024]);
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s16[5 * i + 4] = convert (f[i + 256]);
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}
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break;
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case A52_MONO | A52_LFE:
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for (i = 0; i < 256; i++) {
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s16[6 * i] = s16[6 * i + 1] = s16[6 * i + 2] = s16[6 * i + 3] = 0;
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s16[6 * i + 4] = convert (f[i + 256]);
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s16[6 * i + 5] = convert (f[i]);
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}
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break;
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case A52_CHANNEL | A52_LFE:
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case A52_STEREO | A52_LFE:
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case A52_DOLBY | A52_LFE:
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for (i = 0; i < 256; i++) {
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s16[6 * i] = convert (f[i + 256]);
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s16[6 * i + 1] = convert (f[i + 512]);
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s16[6 * i + 2] = s16[6 * i + 3] = s16[6 * i + 4] = 0;
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s16[6 * i + 5] = convert (f[i]);
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}
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break;
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case A52_3F | A52_LFE:
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for (i = 0; i < 256; i++) {
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s16[6 * i] = convert (f[i + 256]);
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s16[6 * i + 1] = convert (f[i + 768]);
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s16[6 * i + 2] = s16[6 * i + 3] = 0;
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s16[6 * i + 4] = convert (f[i + 512]);
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s16[6 * i + 5] = convert (f[i]);
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}
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break;
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case A52_2F2R | A52_LFE:
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for (i = 0; i < 256; i++) {
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s16[6 * i] = convert (f[i + 256]);
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s16[6 * i + 1] = convert (f[i + 512]);
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s16[6 * i + 2] = convert (f[i + 768]);
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s16[6 * i + 3] = convert (f[i + 1024]);
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s16[6 * i + 4] = 0;
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s16[6 * i + 5] = convert (f[i]);
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}
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break;
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case A52_3F2R | A52_LFE:
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for (i = 0; i < 256; i++) {
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s16[6 * i] = convert (f[i + 256]);
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s16[6 * i + 1] = convert (f[i + 768]);
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s16[6 * i + 2] = convert (f[i + 1024]);
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s16[6 * i + 3] = convert (f[i + 1280]);
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s16[6 * i + 4] = convert (f[i + 512]);
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s16[6 * i + 5] = convert (f[i]);
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}
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break;
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}
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}
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static int
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gst_a52dec_channels (int flags)
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{
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int chans = 0;
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if (flags & A52_LFE) {
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chans += 1;
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}
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flags &= A52_CHANNEL_MASK;
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switch (flags) {
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case A52_3F2R:
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chans += 5;
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break;
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case A52_2F2R:
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case A52_3F1R:
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chans += 4;
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break;
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case A52_2F1R:
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case A52_3F:
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chans += 3;
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case A52_CHANNEL:
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case A52_STEREO:
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case A52_DOLBY:
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chans += 2;
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break;
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default:
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/* error */
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g_warning ("a52dec invalid flags %d", flags);
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return 0;
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}
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return chans;
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}
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static int
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gst_a52dec_push (GstPad * srcpad, int flags, sample_t * _samples, gint64 timestamp)
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{
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GstBuffer *buf;
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int chans;
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#ifdef LIBA52_DOUBLE
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float samples[256 * 6];
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int i;
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for (i = 0; i < 256 * 6; i++)
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samples[i] = _samples[i];
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#else
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float *samples = _samples;
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#endif
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flags &= A52_CHANNEL_MASK | A52_LFE;
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chans = gst_a52dec_channels (flags);
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if (!chans) {
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return 1;
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}
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buf = gst_buffer_new ();
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GST_BUFFER_SIZE (buf) = sizeof (int16_t) * 256 * chans;
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GST_BUFFER_DATA (buf) = g_malloc (GST_BUFFER_SIZE (buf));
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GST_BUFFER_TIMESTAMP (buf) = timestamp;
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float_to_int (samples, (int16_t *) GST_BUFFER_DATA (buf), flags);
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gst_pad_push (srcpad, GST_DATA (buf));
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return 0;
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}
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/* END modified a52dec conversion code */
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static void
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gst_a52dec_reneg (GstPad * pad, int channels, int rate)
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{
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GST_INFO ( "a52dec: reneg channels:%d rate:%d\n", channels, rate);
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if (gst_pad_try_set_caps (pad,
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GST_CAPS_NEW ("a52dec_src_caps",
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"audio/x-raw-int",
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"endianness", GST_PROPS_INT (G_BYTE_ORDER),
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"signed", GST_PROPS_BOOLEAN (TRUE),
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"width", GST_PROPS_INT (16),
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"depth", GST_PROPS_INT (16),
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"channels", GST_PROPS_INT (channels),
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"rate", GST_PROPS_INT (rate))
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) <= 0) {
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gst_element_error (GST_PAD_PARENT (pad), "could not set caps on source pad, aborting...");
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}
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}
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static void
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gst_a52dec_handle_event (GstA52Dec *a52dec)
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{
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guint32 remaining;
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GstEvent *event;
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gst_bytestream_get_status (a52dec->bs, &remaining, &event);
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if (!event) {
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g_warning ("a52dec: no bytestream event");
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return;
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}
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_DISCONTINUOUS:
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gst_bytestream_flush_fast (a52dec->bs, remaining);
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default:
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gst_pad_event_default (a52dec->sinkpad, event);
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break;
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}
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}
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static void
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gst_a52dec_update_streaminfo (GstA52Dec *a52dec)
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{
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GstProps *props;
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GstPropsEntry *entry;
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props = gst_props_empty_new ();
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entry = gst_props_entry_new ("bitrate", GST_PROPS_INT (a52dec->bit_rate));
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gst_props_add_entry (props, (GstPropsEntry *) entry);
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gst_caps_unref (a52dec->streaminfo);
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a52dec->streaminfo = gst_caps_new ("a52dec_streaminfo",
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"application/x-gst-streaminfo",
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props);
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g_object_notify (G_OBJECT (a52dec), "streaminfo");
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}
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static void
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gst_a52dec_loop (GstElement *element)
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{
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GstA52Dec *a52dec;
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guint8 *data;
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int i, length, flags, sample_rate, bit_rate;
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int channels;
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GstBuffer *buf;
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guint32 got_bytes;
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gboolean need_reneg;
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GstClockTime timestamp;
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a52dec = GST_A52DEC (element);
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/* find and read header */
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while (1) {
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got_bytes = gst_bytestream_peek_bytes (a52dec->bs, &data, 7);
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if (got_bytes < 7) {
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gst_a52dec_handle_event (a52dec);
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return;
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}
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length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
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if (length == 0) {
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/* slide window to next 7 bytesa */
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gst_bytestream_flush_fast (a52dec->bs, 1);
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}
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else
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break;
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/* FIXME this can potentially be an infinite loop, we might
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* have to insert a yield operation here */
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}
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need_reneg = FALSE;
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if (a52dec->sample_rate != sample_rate) {
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need_reneg = TRUE;
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a52dec->sample_rate = sample_rate;
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}
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a52dec->stream_channels = flags & A52_CHANNEL_MASK;
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if (bit_rate != a52dec->bit_rate) {
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a52dec->bit_rate = bit_rate;
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gst_a52dec_update_streaminfo (a52dec);
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}
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/* read the header + rest of frame */
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got_bytes = gst_bytestream_read (a52dec->bs, &buf, length);
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if (got_bytes < length) {
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gst_a52dec_handle_event (a52dec);
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return;
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}
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data = GST_BUFFER_DATA (buf);
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timestamp = gst_bytestream_get_timestamp (a52dec->bs);
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if (timestamp == a52dec->last_ts) {
|
|
timestamp = a52dec->current_ts;
|
|
}
|
|
else {
|
|
a52dec->last_ts = timestamp;
|
|
}
|
|
|
|
/* process */
|
|
flags = a52dec->request_channels | A52_ADJUST_LEVEL;
|
|
a52dec->level = 1;
|
|
|
|
if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
|
|
g_warning ("a52dec: a52_frame error\n");
|
|
goto end;
|
|
}
|
|
|
|
channels = flags & A52_CHANNEL_MASK;
|
|
|
|
if (a52dec->using_channels != channels) {
|
|
need_reneg = TRUE;
|
|
a52dec->using_channels = channels;
|
|
}
|
|
|
|
if (need_reneg == TRUE) {
|
|
GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d\n",
|
|
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
|
|
gst_a52dec_reneg (a52dec->srcpad,
|
|
gst_a52dec_channels (a52dec->using_channels), a52dec->sample_rate);
|
|
}
|
|
|
|
if (a52dec->dynamic_range_compression == FALSE) {
|
|
a52_dynrng (a52dec->state, NULL, NULL);
|
|
}
|
|
|
|
for (i = 0; i < 6; i++) {
|
|
if (a52_block (a52dec->state)) {
|
|
g_warning ("a52dec a52_block error %d\n", i);
|
|
continue;
|
|
}
|
|
/* push on */
|
|
if (gst_a52dec_push (a52dec->srcpad, a52dec->using_channels, a52dec->samples, timestamp)) {
|
|
g_warning ("a52dec push error\n");
|
|
}
|
|
else {
|
|
timestamp += sizeof (int16_t) * 256 * GST_SECOND / a52dec->sample_rate;
|
|
}
|
|
}
|
|
a52dec->current_ts = timestamp;
|
|
|
|
end:
|
|
gst_buffer_unref (buf);
|
|
}
|
|
|
|
static GstElementStateReturn
|
|
gst_a52dec_change_state (GstElement * element)
|
|
{
|
|
GstA52Dec *a52dec = GST_A52DEC (element);
|
|
|
|
switch (GST_STATE_TRANSITION (element)) {
|
|
case GST_STATE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_READY_TO_PAUSED:
|
|
a52dec->bs = gst_bytestream_new (a52dec->sinkpad);
|
|
a52dec->state = a52_init (0); /* mm_accel()); */
|
|
a52dec->samples = a52_samples (a52dec->state);
|
|
a52dec->bit_rate = -1;
|
|
a52dec->sample_rate = -1;
|
|
a52dec->stream_channels = A52_CHANNEL;
|
|
/* FIXME force stereo for now */
|
|
a52dec->request_channels = A52_STEREO;
|
|
a52dec->using_channels = A52_CHANNEL;
|
|
a52dec->level = 1;
|
|
a52dec->bias = 384;
|
|
a52dec->last_ts = -1;
|
|
a52dec->current_ts = 0;
|
|
break;
|
|
case GST_STATE_PAUSED_TO_PLAYING:
|
|
break;
|
|
case GST_STATE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_PAUSED_TO_READY:
|
|
gst_bytestream_destroy (a52dec->bs);
|
|
a52dec->bs = NULL;
|
|
a52dec->samples = NULL;
|
|
a52_free (a52dec->state);
|
|
a52dec->state = NULL;
|
|
gst_caps_unref (a52dec->streaminfo);
|
|
break;
|
|
case GST_STATE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
|
|
}
|
|
|
|
GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
|
|
return GST_STATE_SUCCESS;
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_A52DEC (object));
|
|
src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
src->dynamic_range_compression = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstA52Dec *src;
|
|
|
|
/* it's not null if we got it, but it might not be ours */
|
|
g_return_if_fail (GST_IS_A52DEC (object));
|
|
src = GST_A52DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_DRC:
|
|
g_value_set_boolean (value, src->dynamic_range_compression);
|
|
break;
|
|
case ARG_STREAMINFO:
|
|
g_value_set_boxed (value, src->streaminfo);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
plugin_init (GstPlugin * plugin)
|
|
{
|
|
|
|
if (!gst_library_load ("gstbytestream"))
|
|
return FALSE;
|
|
|
|
if (!gst_element_register (plugin, "a52dec", GST_RANK_PRIMARY, GST_TYPE_A52DEC))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
GST_PLUGIN_DEFINE (
|
|
GST_VERSION_MAJOR,
|
|
GST_VERSION_MINOR,
|
|
"a52dec",
|
|
"Decodes ATSC A/52 encoded audio streams",
|
|
plugin_init,
|
|
VERSION,
|
|
"GPL",
|
|
GST_COPYRIGHT,
|
|
GST_PACKAGE,
|
|
GST_ORIGIN
|
|
);
|