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gstreamer/gst/audioconvert/gstaudioconvert.c
David Schleef b036fe4fda fix: width is in bytes, not bits. Remove incorrect assertion. 
Original commit message from CVS:
fix: width is in bytes, not bits.  Remove incorrect assertion.
2003-06-10 06:32:59 +00:00

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/* GStreamer
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
*
* gstaudioconvert.c: Convert audio to different audio formats automatically
*
* 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.
*/
#include <gst/gst.h>
#include <string.h>
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#if 0
static void
print_caps (GstCaps *caps)
{
GValue v = { 0, };
GValue s = { 0, };
g_value_init (&v, GST_TYPE_CAPS);
g_value_init (&s, G_TYPE_STRING);
g_value_set_boxed (&v, caps);
g_value_transform (&v, &s);
g_print ("%s\n", g_value_get_string (&s));
g_value_unset (&v);
g_value_unset (&s);
}
#endif
/*** DEFINITIONS **************************************************************/
#define GST_TYPE_AUDIO_CONVERT (gst_audio_convert_get_type())
#define GST_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
#define GST_AUDIO_CONVERT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CONVERT,GstAudioConvert))
#define GST_IS_AUDIO_CONVERT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CONVERT))
#define GST_IS_AUDIO_CONVERT_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CONVERT))
typedef struct _GstAudioConvert GstAudioConvert;
typedef struct _GstAudioConvertClass GstAudioConvertClass;
struct _GstAudioConvert {
GstElement element;
/* pads */
GstPad * sink;
GstPad * src;
/* properties */
gboolean aggressive;
/* caps: 0 = sink, 1 = src, so always convert from 0 to 1 */
gboolean caps_set[2];
gint law[2];
gint endian[2];
gint sign[2];
gint depth[2]; /* in BITS */
gint width[2]; /* in BYTES */
gint rate[2];
gint channels[2];
};
struct _GstAudioConvertClass {
GstElementClass parent_class;
};
/* type functions */
static GType gst_audio_convert_get_type (void);
static void gst_audio_convert_class_init (GstAudioConvertClass *klass);
static void gst_audio_convert_init (GstAudioConvert *audio_convert);
static void gst_audio_convert_set_property (GObject *object,
guint prop_id,
const GValue *value,
GParamSpec *pspec);
static void gst_audio_convert_get_property (GObject *object,
guint prop_id,
GValue *value,
GParamSpec *pspec);
/* gstreamer functions */
static void gst_audio_convert_chain (GstPad *pad,
GstBuffer *buf);
static GstPadLinkReturn gst_audio_convert_link (GstPad *pad,
GstCaps *caps);
static GstElementStateReturn gst_audio_convert_change_state (GstElement *element);
/* actual work */
static gboolean gst_audio_convert_set_caps (GstPad *pad);
static GstBuffer * gst_audio_convert_buffer_to_default_format (GstAudioConvert *this,
GstBuffer *buf);
static GstBuffer * gst_audio_convert_buffer_from_default_format (GstAudioConvert *this,
GstBuffer *buf);
static GstBuffer * gst_audio_convert_channels (GstAudioConvert *this,
GstBuffer *buf);
static GstElementClass *parent_class = NULL;
/*static guint gst_audio_convert_signals[LAST_SIGNAL] = { 0 }; */
/* AudioConvert signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_AGGRESSIVE,
};
/*** GSTREAMER PROTOTYPES *****************************************************/
GST_PAD_TEMPLATE_FACTORY (audio_convert_src_template_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audio_convert_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_LIST (
GST_PROPS_INT (G_LITTLE_ENDIAN),
GST_PROPS_INT (G_BIG_ENDIAN)
),
"signed", GST_PROPS_LIST (
GST_PROPS_BOOLEAN (FALSE),
GST_PROPS_BOOLEAN (TRUE)
),
"depth", GST_PROPS_INT_RANGE (1, 32),
"width", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16),
GST_PROPS_INT (24),
GST_PROPS_INT (32)
),
"rate", GST_PROPS_INT_RANGE (8000, 192000),
"channels", GST_PROPS_INT_RANGE (1, 2)
)
)
GST_PAD_TEMPLATE_FACTORY (audio_convert_sink_template_factory,
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audio_convert_sink",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_LIST (
GST_PROPS_INT (G_LITTLE_ENDIAN),
GST_PROPS_INT (G_BIG_ENDIAN)
),
"signed", GST_PROPS_LIST (
GST_PROPS_BOOLEAN (FALSE),
GST_PROPS_BOOLEAN (TRUE)
),
"depth", GST_PROPS_INT_RANGE (1, 32),
"width", GST_PROPS_LIST (
GST_PROPS_INT (8),
GST_PROPS_INT (16),
GST_PROPS_INT (24),
GST_PROPS_INT (32)
),
"rate", GST_PROPS_INT_RANGE (8000, 192000),
"channels", GST_PROPS_INT_RANGE (1, 2)
)
)
/*** TYPE FUNCTIONS ***********************************************************/
GType
gst_audio_convert_get_type(void) {
static GType audio_convert_type = 0;
if (!audio_convert_type) {
static const GTypeInfo audio_convert_info = {
sizeof(GstAudioConvertClass), NULL,
NULL,
(GClassInitFunc)gst_audio_convert_class_init,
NULL,
NULL,
sizeof(GstAudioConvert),
0,
(GInstanceInitFunc)gst_audio_convert_init,
};
audio_convert_type = g_type_register_static(GST_TYPE_ELEMENT, "GstAudioConvert", &audio_convert_info, 0);
}
return audio_convert_type;
}
static void
gst_audio_convert_class_init (GstAudioConvertClass *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_AGGRESSIVE,
g_param_spec_boolean("aggressive","aggressive mode","if true, tries any possible format before giving up",
FALSE, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
gstelement_class->change_state = gst_audio_convert_change_state;
}
static void
gst_audio_convert_init (GstAudioConvert *this)
{
/* sinkpad */
this->sink = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (
audio_convert_sink_template_factory), "sink");
gst_pad_set_link_function (this->sink, gst_audio_convert_link);
gst_element_add_pad (GST_ELEMENT(this), this->sink);
/* srcpad */
this->src = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (
audio_convert_src_template_factory), "src");
gst_pad_set_link_function (this->src, gst_audio_convert_link);
gst_element_add_pad (GST_ELEMENT(this), this->src);
gst_pad_set_chain_function(this->sink, gst_audio_convert_chain);
gst_pad_set_chain_function(this->sink, gst_audio_convert_chain);
/* clear important variables */
this->caps_set[0] = this->caps_set[1] = FALSE;
}
static void
gst_audio_convert_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstAudioConvert *audio_convert;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_AUDIO_CONVERT(object));
audio_convert = GST_AUDIO_CONVERT(object);
switch (prop_id) {
case ARG_AGGRESSIVE:
audio_convert->aggressive = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_convert_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstAudioConvert *audio_convert;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_AUDIO_CONVERT(object));
audio_convert = GST_AUDIO_CONVERT(object);
switch (prop_id) {
case ARG_AGGRESSIVE:
g_value_set_boolean (value, audio_convert->aggressive);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/*** GSTREAMER FUNCTIONS ******************************************************/
static void
gst_audio_convert_chain (GstPad *pad, GstBuffer *buf)
{
GstAudioConvert *this;
g_return_if_fail(GST_IS_PAD(pad));
g_return_if_fail(buf != NULL);
g_return_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)));
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad));
/* FIXME */
if (GST_IS_EVENT (buf)) {
gst_pad_event_default (pad, GST_EVENT (buf));
return;
}
if (!this->caps_set[1]) {
if (!gst_audio_convert_set_caps (this->src)) {
gst_element_error (GST_ELEMENT (this), "AudioConvert: could not set caps on pad %s",
GST_PAD_NAME(this->src));
return;
}
}
/**
* Theory of operation:
* - convert the format (endianness, signedness, width, depth) to
* (G_BYTE_ORDER, TRUE, 32, 32)
* - convert rate and channels
* - convert back to output format
*/
buf = gst_audio_convert_buffer_to_default_format (this, buf);
buf = gst_audio_convert_channels (this, buf);
buf = gst_audio_convert_buffer_from_default_format (this, buf);
gst_pad_push (this->src, buf);
}
static GstPadLinkReturn
gst_audio_convert_link (GstPad *pad, GstCaps *caps)
{
GstAudioConvert *this;
gint nr = 0;
gint rate, endianness, depth, width, channels;
gboolean sign;
g_return_val_if_fail(GST_IS_PAD(pad), GST_PAD_LINK_REFUSED);
g_return_val_if_fail(GST_IS_AUDIO_CONVERT(GST_OBJECT_PARENT (pad)), GST_PAD_LINK_REFUSED);
this = GST_AUDIO_CONVERT(GST_OBJECT_PARENT (pad));
/* could we do better? */
if (!GST_CAPS_IS_FIXED (caps))
return GST_PAD_LINK_DELAYED;
if (pad == this->sink) {
nr = 0;
} else if (pad == this->src) {
nr = 1;
} else {
g_assert_not_reached ();
}
if (!gst_caps_get (caps, "rate", &rate,
"channels", &channels,
"signed", &sign,
"depth", &depth,
"width", &width,
NULL
))
return GST_PAD_LINK_DELAYED;
if (!gst_caps_get_int (caps, "endianness", &endianness)) {
if (width == 1) {
endianness = G_BYTE_ORDER;
} else {
return GST_PAD_LINK_DELAYED;
}
}
/* we cannot yet convert this, so check */
if (this->caps_set[1 - nr]) {
if (rate != this->rate[1 - nr])
return GST_PAD_LINK_REFUSED;
}
this->caps_set[nr] = TRUE;
this->rate[nr] = rate;
this->channels[nr] = channels;
this->sign[nr] = sign;
this->endian[nr] = endianness;
this->depth[nr] = depth;
this->width[nr] = width / 8;
return GST_PAD_LINK_OK;
}
static GstElementStateReturn
gst_audio_convert_change_state (GstElement *element)
{
GstAudioConvert *this = GST_AUDIO_CONVERT (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY:
this->caps_set[0] = this->caps_set[1] = FALSE;
break;
default:
break;
}
if (parent_class->change_state) {
return parent_class->change_state (element);
} else {
return GST_STATE_SUCCESS;
}
}
/*** ACTUAL WORK **************************************************************/
static GstCaps*
make_caps (gint endianness, gboolean sign, gint depth, gint width, gint rate, gint channels)
{
if (width == 1) {
return GST_CAPS_NEW (
"audio_convert_caps",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"signed", GST_PROPS_BOOLEAN (sign),
"depth", GST_PROPS_INT (depth),
"width", GST_PROPS_INT (width * 8),
"rate", GST_PROPS_INT (rate),
"channels", GST_PROPS_INT (channels)
);
} else {
return GST_CAPS_NEW (
"audio_convert_caps",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (endianness),
"signed", GST_PROPS_BOOLEAN (sign),
"depth", GST_PROPS_INT (depth),
"width", GST_PROPS_INT (width * 8),
"rate", GST_PROPS_INT (rate),
"channels", GST_PROPS_INT (channels)
);
}
}
static gboolean
gst_audio_convert_set_caps (GstPad *pad)
{
GstCaps *caps;
gint nr;
GstPadLinkReturn ret;
GstAudioConvert *this;
gint channels, endianness, depth, width; /*, rate; */
gboolean sign;
this = GST_AUDIO_CONVERT (GST_PAD_PARENT (pad));
nr = this->src == pad ? 1 : this->sink == pad ? 0 : -1;
g_assert (nr > -1);
/* try 1:1 first */
caps = make_caps (this->endian[1 - nr], this->sign[1 - nr], this->depth[1 - nr],
this->width[1 - nr], this->rate[1 - nr], this->channels[1 - nr]);
ret = gst_pad_try_set_caps (pad, caps);
if (ret == GST_PAD_LINK_DONE || ret == GST_PAD_LINK_OK)
goto success;
/* now do some iterating, this is gonna be fun */
/* stereo is most important */
channels = 2;
while (channels > 0) {
/* endianness comes second */
endianness = 0;
do {
if (endianness == G_BIG_ENDIAN)
break;
endianness = endianness == 0 ? G_LITTLE_ENDIAN : G_BIG_ENDIAN;
/* signedness */
sign = TRUE;
do {
sign = !sign;
/* depth */
for (width = 4; width >= 1; width--) {
/* width */
for (depth = width * 8; depth >= 1; depth -= this->aggressive ? 1 : 8) {
/* rate - not supported yet*/
caps = make_caps (endianness, sign, depth,
width, this->rate[1 - nr], channels);
ret = gst_pad_try_set_caps (pad, caps);
if (ret == GST_PAD_LINK_DONE || ret == GST_PAD_LINK_OK) {
goto success;
}
}
}
} while (sign != TRUE);
} while TRUE;
channels--;
}
goto fail;
fail:
return FALSE;
success:
g_assert (gst_audio_convert_link (pad, caps) == GST_PAD_LINK_OK);
return TRUE;
}
/* return a writable buffer of size which ideally is the same as before
- You must unref the new buffer
- The size of the old buffer is undefined after this operation */
static GstBuffer*
gst_audio_convert_get_buffer (GstBuffer *buf, guint size)
{
GstBuffer *ret;
if (buf->maxsize >= size && gst_buffer_is_writable (buf)) {
gst_buffer_ref (buf);
buf->size = size;
return buf;
} else if (buf->maxsize >= size) {
buf = gst_buffer_copy (buf);
buf->size = size;
return buf;
} else {
g_assert ((ret = gst_buffer_new_and_alloc (size)));
ret->timestamp = buf->timestamp;
return ret;
}
}
static inline guint8 GUINT8_IDENTITY (guint8 x) { return x; }
static inline guint8 GINT8_IDENTITY (gint8 x) { return x; }
#define CONVERT_TO(to, from, type, sign, endianness, LE_FUNC, BE_FUNC) G_STMT_START{\
type value; \
memcpy (&value, from, sizeof (type)); \
from -= sizeof (type); \
value = (endianness == G_LITTLE_ENDIAN) ? LE_FUNC (value) : BE_FUNC (value); \
if (sign) { \
to = value; \
} else { \
to = (gint64) value - (1 << (sizeof (type) * 8 - 1)); \
} \
}G_STMT_END;
static GstBuffer*
gst_audio_convert_buffer_to_default_format (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
gint i, count;
gint64 cur = 0;
gint32 write;
gint32 *dest;
guint8 *src;
if (this->width[0] == 4 && this->depth[0] == 32 &&
this->endian[0] == G_BYTE_ORDER && this->sign[0] == TRUE)
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * 4 / this->width[0]);
count = ret->size / 4;
src = buf->data + (count - 1) * this->width[0];
dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) {
switch (this->width[0]) {
case 1:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint8, this->sign[0], this->endian[0], GINT8_IDENTITY, GINT8_IDENTITY);
} else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 2:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint16, this->sign[0], this->endian[0], GINT16_FROM_LE, GINT16_FROM_BE);
} else {
CONVERT_TO (cur, src, guint16, this->sign[0], this->endian[0], GUINT16_FROM_LE, GUINT16_FROM_BE);
}
break;
case 3:
if (this->sign[0]) {
gint32 value;
if (this->endian[0] == G_BIG_ENDIAN) {
gpointer p = &value;
p++;
memcpy (p, src, 3);
value = GINT32_FROM_BE (value);
} else if (this->endian[0] == G_LITTLE_ENDIAN) {
memcpy (&value, src, 3);
value = GINT32_FROM_LE (value);
} else {
g_assert_not_reached();
}
cur = value;
} else {
guint32 value;
if (this->endian[0] == G_BIG_ENDIAN) {
gpointer p = &value;
p++;
memcpy (p, src, 3);
value = GUINT32_FROM_BE (value);
} else if (this->endian[0] == G_LITTLE_ENDIAN) {
memcpy (&value, src, 3);
value = GUINT32_FROM_LE (value);
} else {
g_assert_not_reached();
}
cur = (gint64) value - (1 << 23);
}
src -= 3;
break;
case 4:
if (this->sign[0]) {
CONVERT_TO (cur, src, gint32, this->sign[0], this->endian[0], GINT32_FROM_LE, GINT32_FROM_BE);
} else {
CONVERT_TO (cur, src, guint32, this->sign[0], this->endian[0], GUINT32_FROM_LE, GUINT32_FROM_BE);
}
break;
default:
g_assert_not_reached ();
}
cur = cur * ((gint64) 1 << (32 - this->depth[0]));
cur = CLAMP (cur, -((gint64)1 << 32), (gint64) 0x7FFFFFFF);
write = cur;
memcpy (&dest[i], &write, 4);
}
gst_buffer_unref (buf);
return ret;
}
#define POPULATE(format, be_func, le_func) G_STMT_START{ \
format val; \
format* p = (format *) dest; \
int_value >>= (32 - this->depth[1]); \
val = (format) int_value; \
switch (this->endian[1]) { \
case G_LITTLE_ENDIAN: \
val = le_func (val); \
break; \
case G_BIG_ENDIAN: \
val = be_func (val); \
break; \
default: \
g_assert_not_reached (); \
}; \
*p = val; \
p ++; \
dest = (guint8 *) p; \
}G_STMT_END
static GstBuffer *
gst_audio_convert_buffer_from_default_format (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
guint8 *dest;
guint count, i;
gint32 *src;
if (this->width[1] == 4 && this->depth[1] == 32 &&
this->endian[1] == G_BYTE_ORDER && this->sign[1] == TRUE)
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size * this->width[1] / 4);
dest = ret->data;
src = (gint32 *) buf->data;
count = ret->size / this->width[1];
for (i = 0; i < count; i++) {
gint32 int_value = *src;
src++;
switch (this->width[1]) {
case 1:
if (this->sign[1]) {
POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 2:
if (this->sign[1]) {
POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
} else {
POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
}
break;
case 3:
if (this->sign[1]) {
gpointer p;
gint32 val = (gint32) int_value;
switch (this->endian[1]) {
case G_LITTLE_ENDIAN:
val = GINT32_TO_LE (val);
break;
case G_BIG_ENDIAN:
val = GINT32_TO_BE (val);
break;
default:
g_assert_not_reached ();
};
p = &val;
if (this->endian[1] == G_BIG_ENDIAN)
p++;
memcpy (dest, p, 3);
dest += 3;
} else {
gpointer p;
guint32 val = (guint32) int_value;
switch (this->endian[1]) {
case G_LITTLE_ENDIAN:
val = GUINT32_TO_LE (val);
break;
case G_BIG_ENDIAN:
val = GUINT32_TO_BE (val);
break;
default:
g_assert_not_reached ();
};
p = &val;
if (this->endian[1] == G_BIG_ENDIAN)
p++;
memcpy (dest, p, 3);
dest += 3;
}
break;
case 4:
if (this->sign[1]) {
POPULATE (gint32, GINT32_TO_BE, GINT32_TO_LE);
} else {
POPULATE (guint32, GUINT32_TO_BE, GUINT32_TO_LE);
}
break;
default:
g_assert_not_reached ();
}
}
gst_buffer_unref(buf);
return ret;
}
static GstBuffer *
gst_audio_convert_channels (GstAudioConvert *this, GstBuffer *buf)
{
GstBuffer *ret;
guint i, count;
guint32 *src, *dest;
if (this->channels[0] == this->channels[1])
return buf;
ret = gst_audio_convert_get_buffer (buf, buf->size / this->channels[0] * this->channels[1]);
count = ret->size / 4 / this->channels[1];
src = (guint32 *) buf->data;
dest = (guint32 *) ret->data;
if (this->channels[0] > this->channels[1]) {
for (i = 0; i < count; i++) {
*dest = *src >> 1;
src++;
*dest += (*src + 1) >> 1;
src++;
dest++;
}
} else {
for (i = count - 1; i >= 0; i--) {
dest[2 * i] = dest[2 * i + 1] = src[i];
}
}
gst_buffer_unref(buf);
return ret;
}
/*** PLUGIN DETAILS ***********************************************************/
static GstElementDetails audio_convert_details = {
"Audio Conversion",
"Filter/Convert",
"LGPL",
"Convert audio to different formats",
VERSION,
"Benjamin Otte <in7y118@public.uni-hamburg.de",
"(C) 2003",
};
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_element_factory_new("audioconvert", GST_TYPE_AUDIO_CONVERT,
&audio_convert_details);
g_return_val_if_fail(factory != NULL, FALSE);
gst_element_factory_add_pad_template (factory,
GST_PAD_TEMPLATE_GET (audio_convert_src_template_factory));
gst_element_factory_add_pad_template (factory,
GST_PAD_TEMPLATE_GET (audio_convert_sink_template_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstaudioconvert",
plugin_init
};
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