mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-03-30 04:45:36 +00:00
Code Issues1 Projects Releases Wiki Activity

New DTS decoder.

Browse source 
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/dts/Makefile.am:
* ext/dts/gstdtsdec.c: (gst_dtsdec_get_type),
(gst_dtsdec_base_init), (gst_dtsdec_class_init), (gst_dtsdec_init),
(gst_dtsdec_channels), (gst_dtsdec_renegotiate),
(gst_dtsdec_handle_event), (gst_dtsdec_update_streaminfo),
(gst_dtsdec_loop), (gst_dtsdec_change_state),
(gst_dtsdec_set_property), (gst_dtsdec_get_property),
(plugin_init):
* ext/dts/gstdtsdec.h:
New DTS decoder.
* ext/faad/gstfaad.c: (gst_faad_sinkconnect),
(gst_faad_srcconnect):
Add ESDS atom handling (.m4a).
This commit is contained in:
Ronald S. Bultje 2004-04-29 00:00:25 +00:00
parent 48892c24ed
commit 8b8776f69c
7 changed files with 662 additions and 4 deletions

18
ChangeLog
View file

@ -1,3 +1,21 @@
2004-04-28 Ronald Bultje <rbultje@ronald.bitfreak.net>
* configure.ac:
* ext/Makefile.am:
* ext/dts/Makefile.am:
* ext/dts/gstdtsdec.c: (gst_dtsdec_get_type),
(gst_dtsdec_base_init), (gst_dtsdec_class_init), (gst_dtsdec_init),
(gst_dtsdec_channels), (gst_dtsdec_renegotiate),
(gst_dtsdec_handle_event), (gst_dtsdec_update_streaminfo),
(gst_dtsdec_loop), (gst_dtsdec_change_state),
(gst_dtsdec_set_property), (gst_dtsdec_get_property),
(plugin_init):
* ext/dts/gstdtsdec.h:
New DTS decoder.
* ext/faad/gstfaad.c: (gst_faad_sinkconnect),
(gst_faad_srcconnect):
Add ESDS atom handling (.m4a).
2004-04-27 Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/divx/gstdivxdec.c: (plugin_init):

8
configure.ac
View file

@ -808,6 +808,13 @@ return 0;
fi
])
dnl *** DTS ***
translit(dnm, m, l) AM_CONDITIONAL(USE_DTS, true)
GST_CHECK_FEATURE(DTS, [dts library], dtsdec, [
GST_CHECK_LIBHEADER(DTS, dts_pic, dts_init, -lm, dts.h, DTS_LIBS="-ldts_pic -lm")
AC_SUBST(DTS_LIBS)
])
dnl *** dvdread ***
translit(dnm, m, l) AM_CONDITIONAL(USE_DVDREAD, true)
GST_CHECK_FEATURE(DVDREAD, [dvdread library], dvdreadsrc, [
@ -1778,6 +1785,7 @@ ext/artsd/Makefile
ext/audiofile/Makefile
ext/cdparanoia/Makefile
ext/divx/Makefile
ext/dts/Makefile
ext/dv/Makefile
ext/dvdread/Makefile
ext/dvdnav/Makefile

8
ext/Makefile.am
View file

@ -46,6 +46,12 @@ else
DIVX_DIR=
endif
if USE_DTS
DTS_DIR=dvdread
else
DTS_DIR=
endif
if USE_DVDREAD
DVDREAD_DIR=dvdread
else
@ -331,6 +337,7 @@ SUBDIRS=\
$(AUDIOFILE_DIR) \
$(CDPARANOIA_DIR) \
$(DIVX_DIR) \
$(DTS_DIR) \
$(DVDREAD_DIR) \
$(DVDNAV_DIR) \
$(ESD_DIR) \
@ -386,6 +393,7 @@ DIST_SUBDIRS=\
audiofile \
cdparanoia \
divx \
dts \
dv \
dvdread \
dvdnav \

8
ext/dts/Makefile.am Normal file
View file

@ -0,0 +1,8 @@
plugin_LTLIBRARIES = libgstdtsdec.la
libgstdtsdec_la_SOURCES = gstdtsdec.c
libgstdtsdec_la_CFLAGS = $(GST_CFLAGS)
libgstdtsdec_la_LIBADD = $(DTS_LIBS)
libgstdtsdec_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstdtsdec.h

514
ext/dts/gstdtsdec.c Normal file
View file

@ -0,0 +1,514 @@
/* GStreamer DTS decoder plugin based on libdtsdec
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "_stdint.h"
#include <stdlib.h>
#include <gst/gst.h>
#include <dts.h>
#include "gstdtsdec.h"
GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
#define GST_CAT_DEFAULT (dtsdec_debug)
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DRC
/* FILL ME */
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-dts")
);
#if defined(LIBDTS_FIXED)
#define DTS_CAPS "audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16"
#define SAMPLE_WIDTH 16
#elif defined(LIBDTS_DOUBLE)
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64"
#define SAMPLE_WIDTH 64
#else
#define DTS_CAPS "audio/x-raw-float, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32"
#define SAMPLE_WIDTH 32
#endif
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (DTS_CAPS ", "
"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
static void gst_dtsdec_base_init (GstDtsDecClass * klass);
static void gst_dtsdec_class_init (GstDtsDecClass * klass);
static void gst_dtsdec_init (GstDtsDec * dtsdec);
static void gst_dtsdec_loop (GstElement * element);
static GstElementStateReturn gst_dtsdec_change_state (GstElement * element);
static void gst_dtsdec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtsdec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/* static guint gst_dtsdec_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_dtsdec_get_type (void)
{
static GType dtsdec_type = 0;
if (!dtsdec_type) {
static const GTypeInfo dtsdec_info = {
sizeof (GstDtsDecClass),
(GBaseInitFunc) gst_dtsdec_base_init,
NULL, (GClassInitFunc) gst_dtsdec_class_init,
NULL,
NULL,
sizeof (GstDtsDec),
0,
(GInstanceInitFunc) gst_dtsdec_init,
};
dtsdec_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstDtsDec", &dtsdec_info, 0);
GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
}
return dtsdec_type;
}
static void
gst_dtsdec_base_init (GstDtsDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_dtsdec_details = {
"DTS audio decoder",
"Codec/Audio/Decoder",
"Decodes DTS audio streams",
"Ronald Bultje <rbultje@ronald.bitfreak.net>"
};
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_dtsdec_details);
}
static void
gst_dtsdec_class_init (GstDtsDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
gobject_class->set_property = gst_dtsdec_set_property;
gobject_class->get_property = gst_dtsdec_get_property;
gstelement_class->change_state = gst_dtsdec_change_state;
}
static void
gst_dtsdec_init (GstDtsDec * dtsdec)
{
GstElement *element = GST_ELEMENT (dtsdec);
/* create the sink and src pads */
dtsdec->sinkpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(dtsdec), "sink"), "sink");
gst_element_add_pad (element, dtsdec->sinkpad);
gst_element_set_loop_function (element, gst_dtsdec_loop);
dtsdec->srcpad =
gst_pad_new_from_template (gst_element_get_pad_template (element,
"src"), "src");
gst_pad_use_explicit_caps (dtsdec->srcpad);
gst_element_add_pad (element, dtsdec->srcpad);
GST_FLAG_SET (element, GST_ELEMENT_EVENT_AWARE);
dtsdec->dynamic_range_compression = FALSE;
}
static gint
gst_dtsdec_channels (uint32_t flags)
{
gint chans = 0;
switch (flags & DTS_CHANNEL_MASK) {
case DTS_MONO:
chans = 1;
break;
case DTS_CHANNEL:
case DTS_STEREO:
case DTS_STEREO_SUMDIFF:
case DTS_STEREO_TOTAL:
case DTS_DOLBY:
chans = 2;
break;
case DTS_3F:
case DTS_2F1R:
chans = 3;
break;
case DTS_3F1R:
case DTS_2F2R:
chans = 4;
break;
case DTS_3F2R:
chans = 5;
break;
case DTS_4F2R:
chans = 6;
break;
default:
/* error */
g_warning ("dtsdec: invalid flags 0x%x", flags);
return 0;
}
if (flags & DTS_LFE)
chans += 1;
return chans;
}
static gboolean
gst_dtsdec_renegotiate (GstDtsDec * dts)
{
GstCaps *caps = gst_caps_from_string (DTS_CAPS);
gint channels = gst_dtsdec_channels (dts->using_channels);
GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
channels, dts->sample_rate);
gst_caps_set_simple (caps,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
return gst_pad_set_explicit_caps (dts->srcpad, caps);
}
static void
gst_dtsdec_handle_event (GstDtsDec * dts)
{
guint32 remaining;
GstEvent *event;
gst_bytestream_get_status (dts->bs, &remaining, &event);
if (!event) {
GST_ELEMENT_ERROR (dts, RESOURCE, READ, (NULL), (NULL));
return;
}
GST_LOG ("Handling event of type %d timestamp %llu", GST_EVENT_TYPE (event),
GST_EVENT_TIMESTAMP (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_DISCONTINUOUS:
case GST_EVENT_FLUSH:
gst_bytestream_flush_fast (dts->bs, remaining);
break;
default:
break;
}
gst_pad_event_default (dts->sinkpad, event);
}
static void
gst_dtsdec_update_streaminfo (GstDtsDec * dts)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (dts),
dts->srcpad, dts->current_ts, taglist);
}
static void
gst_dtsdec_loop (GstElement * element)
{
GstDtsDec *dts = GST_DTSDEC (element);
guint8 *data;
GstBuffer *buf, *out;
sample_t *samples;
gint i, length, flags, sample_rate, bit_rate, frame_length, s, c, num_c;
gint channels, skipped = 0, num_blocks;
guint32 got_bytes;
gboolean need_renegotiation = FALSE;
GstClockTime timestamp = 0;
/* find sync. Don't know what 3840 is based on... */
#define MAX_SKIP 3840
while (skipped < MAX_SKIP) {
got_bytes = gst_bytestream_peek_bytes (dts->bs, &data, 7);
if (got_bytes < 7) {
gst_dtsdec_handle_event (dts);
return;
}
length = dts_syncinfo (dts->state, data, &flags,
&sample_rate, &bit_rate, &frame_length);
if (length == 0) {
/* shift window to re-find sync */
gst_bytestream_flush_fast (dts->bs, 1);
skipped++;
GST_LOG ("Skipped");
} else
break;
}
if (skipped >= MAX_SKIP) {
GST_ELEMENT_ERROR (dts, RESOURCE, SYNC, (NULL), (NULL));
return;
}
/* go over stream properties, update caps/streaminfo if needed */
if (dts->sample_rate != sample_rate) {
need_renegotiation = TRUE;
dts->sample_rate = sample_rate;
}
dts->stream_channels = flags;
if (bit_rate != dts->bit_rate) {
dts->bit_rate = bit_rate;
gst_dtsdec_update_streaminfo (dts);
}
/* read the header + rest of frame */
got_bytes = gst_bytestream_read (dts->bs, &buf, length);
if (got_bytes < length) {
gst_dtsdec_handle_event (dts);
return;
}
data = GST_BUFFER_DATA (buf);
timestamp = gst_bytestream_get_timestamp (dts->bs);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (timestamp == dts->last_ts) {
timestamp = dts->current_ts;
} else {
dts->last_ts = timestamp;
}
}
/* process */
flags = dts->request_channels | DTS_ADJUST_LEVEL;
dts->level = 1;
if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
GST_WARNING ("dts_frame error");
goto end;
}
channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
if (dts->using_channels != channels) {
need_renegotiation = TRUE;
dts->using_channels = channels;
}
if (need_renegotiation == TRUE) {
GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
dts->sample_rate, dts->stream_channels, dts->using_channels);
if (!gst_dtsdec_renegotiate (dts))
goto end;
}
if (dts->dynamic_range_compression == FALSE) {
dts_dynrng (dts->state, NULL, NULL);
}
/* handle decoded data, one block is 256 samples */
num_blocks = dts_blocks_num (dts->state);
for (i = 0; i < num_blocks; i++) {
if (dts_block (dts->state)) {
GST_WARNING ("dts_block error %d", i);
continue;
}
samples = dts_samples (dts->state);
num_c = gst_dtsdec_channels (dts->using_channels);
out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
GST_BUFFER_TIMESTAMP (out) = timestamp;
GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
/* libdts returns buffers in 256-sample-blocks per channel,
* we want interleaved. And we need to copy anyway... */
data = GST_BUFFER_DATA (out);
for (s = 0; s < 256; s++) {
for (c = 0; c < num_c; c++) {
*(sample_t *) data = samples[s + c * 256];
data += (SAMPLE_WIDTH / 8);
}
}
/* push on */
gst_pad_push (dts->srcpad, GST_DATA (out));
timestamp += GST_SECOND * 256 / dts->sample_rate;
}
dts->current_ts = timestamp;
end:
gst_buffer_unref (buf);
}
static GstElementStateReturn
gst_dtsdec_change_state (GstElement * element)
{
GstDtsDec *dts = GST_DTSDEC (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:{
GstCPUFlags cpuflags;
uint32_t mm_accel = 0;
dts->bs = gst_bytestream_new (dts->sinkpad);
cpuflags = gst_cpu_get_flags ();
if (cpuflags & GST_CPU_FLAG_MMX)
mm_accel |= MM_ACCEL_X86_MMX;
if (cpuflags & GST_CPU_FLAG_3DNOW)
mm_accel |= MM_ACCEL_X86_3DNOW;
if (cpuflags & GST_CPU_FLAG_MMXEXT)
mm_accel |= MM_ACCEL_X86_MMXEXT;
dts->state = dts_init (mm_accel);
break;
}
case GST_STATE_READY_TO_PAUSED:
dts->samples = dts_samples (dts->state);
dts->bit_rate = -1;
dts->sample_rate = -1;
dts->stream_channels = 0;
/* FIXME force stereo for now */
dts->request_channels = DTS_STEREO;
dts->using_channels = 0;
dts->level = 1;
dts->bias = 0;
dts->last_ts = 0;
dts->current_ts = 0;
break;
case GST_STATE_PAUSED_TO_READY:
dts->samples = NULL;
break;
case GST_STATE_READY_TO_NULL:
gst_bytestream_destroy (dts->bs);
dts->bs = NULL;
dts_free (dts->state);
dts->state = NULL;
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static void
gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
dts->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDtsDec *dts = GST_DTSDEC (object);
switch (prop_id) {
case ARG_DRC:
g_value_set_boolean (value, dts->dynamic_range_compression);
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, "dtsdec", GST_RANK_PRIMARY,
GST_TYPE_DTSDEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"dtsdec",
"Decodes DTS audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);

77
ext/dts/gstdtsdec.h Normal file
View file

@ -0,0 +1,77 @@
/* GStreamer DTS decoder plugin based on libdtsdec
* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_DTSDEC_H__
#define __GST_DTSDEC_H__
#include <gst/gst.h>
#include <gst/bytestream/bytestream.h>
G_BEGIN_DECLS
#define GST_TYPE_DTSDEC \
(gst_dtsdec_get_type())
#define GST_DTSDEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DTSDEC,GstDtsDec))
#define GST_DTSDEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DTSDEC,GstDtsDecClass))
#define GST_IS_DTSDEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DTSDEC))
#define GST_IS_DTSDEC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DTSDEC))
typedef struct _GstDtsDec GstDtsDec;
typedef struct _GstDtsDecClass GstDtsDecClass;
struct _GstDtsDec {
GstElement element;
/* pads */
GstPad *sinkpad,
*srcpad;
/* stream properties */
gint bit_rate;
gint sample_rate;
gint stream_channels;
gint request_channels;
gint using_channels;
/* decoding properties */
sample_t level;
sample_t bias;
gboolean dynamic_range_compression;
sample_t *samples;
dts_state_t *state;
GstByteStream *bs;
/* keep track of time */
GstClockTime last_ts;
GstClockTime current_ts;
};
struct _GstDtsDecClass {
GstElementClass parent_class;
};
G_END_DECLS
#endif /* __GST_DTSDEC_H__ */

33
ext/faad/gstfaad.c
View file

@ -146,8 +146,33 @@ gst_faad_init (GstFaad * faad)
static GstPadLinkReturn
gst_faad_sinkconnect (GstPad * pad, const GstCaps * caps)
{
/* oh, we really don't care what's in here. We'll
* get AAC audio (MPEG-2/4) anyway, so why bother? */
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
GstStructure *str = gst_caps_get_structure (caps, 0);
const GValue *value;
GstBuffer *buf;
if ((value = gst_structure_get_value (str, "codec_data"))) {
GstPadLinkReturn ret;
gulong samplerate;
guchar channels;
buf = g_value_get_boxed (value);
if (faacDecInit2 (faad->handle, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf), &samplerate, &channels) < 0)
return GST_PAD_LINK_REFUSED;
faad->samplerate = samplerate;
faad->channels = channels;
ret = gst_pad_renegotiate (faad->srcpad);
if (ret == GST_PAD_LINK_DELAYED)
ret = GST_PAD_LINK_OK;
return ret;
}
/* if there's no decoderspecificdata, it's all fine. We cannot know
* much more at this point... */
return GST_PAD_LINK_OK;
}
@ -229,7 +254,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
{
GstStructure *structure;
const gchar *mimetype;
gint fmt = 0;
gint fmt = -1;
gint depth, rate, channels;
GstFaad *faad = GST_FAAD (gst_pad_get_parent (pad));
@ -282,7 +307,7 @@ gst_faad_srcconnect (GstPad * pad, const GstCaps * caps)
}
}
if (fmt) {
if (fmt != -1) {
faacDecConfiguration *conf;
conf = faacDecGetCurrentConfiguration (faad->handle);