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Added initial version of audioconvert, a generic converter of integer audio/raw formats.

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Original commit message from CVS:
Added initial version of audioconvert, a generic converter of integer audio/raw formats.
It currently supports conversion of
- channels (mono/stereo only, until someone tells me how to mix other channels)
- endianness (little/bi endian)
- signedness
- width (8, 1, 24 and 32 bits)
- depth (1 - width bits)
missing:
- enough testing (I intend to write a testsuite for this, but that's pending)
- samplerate conversion
- other goodies like format conversion etc
Expect bugs when using it.

problems this should solve:
- encoding wav files on big endian machines
- goom working with mono audio files in gst-player
- Iain's soundcard (that one is a problem in itself)
- complaints about missing conversion
- too many age old, nearly unmaintained plugins (stereo2mono etc.)
Have fun.
This commit is contained in:
Benjamin Otte 2003-04-14 01:20:30 +00:00
parent 40beb3e810
commit d41620bc75
3 changed files with 815 additions and 1 deletions
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3
configure.ac
View file

@ -232,7 +232,7 @@ AC_SUBST(GST_PLUGIN_LDFLAGS)
dnl these are all the gst plug-ins, compilable without additional libs
GST_PLUGINS_ALL="\
ac3parse adder audioscale auparse avi asfdemux cdxaparse chart\
ac3parse adder audioscale auparse avi asfdemux audioconvert cdxaparse chart\
cutter deinterlace effectv festival filter flx goom\
intfloat law level\
median mixmatrix mpeg1enc mpeg1sys mpeg1videoparse mpeg2enc mpeg2sub\
@ -1049,6 +1049,7 @@ gst-plugins.spec
gst/Makefile
gst/ac3parse/Makefile
gst/adder/Makefile
gst/audioconvert/Makefile
gst/audioscale/Makefile
gst/auparse/Makefile
gst/avi/Makefile

8
gst/audioconvert/Makefile.am Normal file
View file

@ -0,0 +1,8 @@
plugindir = $(libdir)/gstreamer-@GST_MAJORMINOR@
plugin_LTLIBRARIES = libgstaudioconvert.la
libgstaudioconvert_la_SOURCES = gstaudioconvert.c
libgstaudioconvert_la_CFLAGS = $(GST_CFLAGS)
libgstaudioconvert_la_LIBADD =
libgstaudioconvert_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)

805
gst/audioconvert/gstaudioconvert.c Normal file
View file

@ -0,0 +1,805 @@
/* 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 agressive;
/* 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_AGRESSIVE,
};
/*** 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_AGRESSIVE,
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_AGRESSIVE:
audio_convert->agressive = 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_AGRESSIVE:
g_value_set_boolean (value, audio_convert->agressive);
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;
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 == 8) {
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 == 8) {
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);
g_assert (this->caps_set[1 - nr]);
/* 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->agressive ? 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_readonly (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;
gint32 write;
guint8 *src, *dest;
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;
dest = ret->data;
for (i = 0; i < count; 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, &write, 4);
dest += 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 / 2;
src++;
*dest += (*src + 1) / 2;
src++;
dest++;
}
} else {
for (i = 0; i < count; i++) {
*dest = *src;
dest++;
*dest = *src;
dest++;
src++;
}
}
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
};