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gstreamer/gst/audioconvert/gstaudioconvert.c
Thomas Vander Stichele 2a415f1abd fix up audioconvert caps nego, remove float stuff, remove rate stuff gst-launch-0.7 -v sinesrc ! audioconvert ! audi... 
Original commit message from CVS:

fix up audioconvert caps nego, remove float stuff, remove rate stuff
gst-launch-0.7  -v sinesrc ! audioconvert ! audio/x-raw-int,rate=23000 ! wavenc ! filesink location=test.wav now writes a completely useless 23000 Hz wave file
2004-01-12 19:46:45 +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.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_convert_debug);
#define GST_CAT_DEFAULT (audio_convert_debug)
#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 _GstAudioConvertCaps GstAudioConvertCaps;
typedef struct _GstAudioConvertClass GstAudioConvertClass;
/* this struct is a handy way of passing around all the caps info ... */
struct _GstAudioConvertCaps {
/* general caps */
gint endianness;
gint width;
gint rate;
gint channels;
/* int audio caps */
gint depth;
gboolean is_signed;
};
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];
GstAudioConvertCaps caps[2];
gint law[2];
gint endian[2];
gint sign[2];
gint depth[2]; /* in BITS */
gint width[2]; /* in BITS */
gint rate[2];
gint channels[2];
/* conversion functions */
GstBuffer * (* convert_internal) (GstAudioConvert *this, GstBuffer *buf);
};
struct _GstAudioConvertClass {
GstElementClass parent_class;
};
static GstElementDetails audio_convert_details = {
"Audio Conversion",
"Filter/Converter/Audio",
"Convert audio to different formats",
"Benjamin Otte <in7y118@public.uni-hamburg.de",
};
/* type functions */
static GType gst_audio_convert_get_type (void);
static void gst_audio_convert_base_init (gpointer g_class);
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, GstData *_data);
static GstPadLinkReturn gst_audio_convert_link (GstPad *pad, const GstCaps *caps);
static GstElementStateReturn gst_audio_convert_change_state (GstElement *element);
/* actual work */
#if 0
static gboolean gst_audio_convert_set_caps (GstPad *pad);
#endif
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);
/* AudioConvert signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_AGGRESSIVE,
};
static GstElementClass *parent_class = NULL;
/*static guint gst_audio_convert_signals[LAST_SIGNAL] = { 0 }; */
/*** GSTREAMER PROTOTYPES *****************************************************/
static GstStaticPadTemplate gst_audio_convert_src_template =
GST_STATIC_PAD_TEMPLATE (
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS
)
);
static GstStaticPadTemplate gst_audio_convert_sink_template =
GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (
GST_AUDIO_INT_PAD_TEMPLATE_CAPS
)
);
/*** 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),
gst_audio_convert_base_init,
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);
}
GST_DEBUG_CATEGORY_INIT (audio_convert_debug, "audioconvert", 0, "audio conversion element");
return audio_convert_type;
}
static void
gst_audio_convert_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_convert_sink_template));
gst_element_class_set_details (element_class, &audio_convert_details);
}
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);
gobject_class->set_property = gst_audio_convert_set_property;
gobject_class->get_property = gst_audio_convert_get_property;
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));
gstelement_class->change_state = gst_audio_convert_change_state;
}
static GstCaps *
gst_audioconvert_getcaps (GstPad *pad)
{
GstAudioConvert *this;
GstCaps *othercaps;
GstCaps *caps;
GstPad *otherpad;
int i;
GST_DEBUG ("gst_audioconvert_getcaps");
this = GST_AUDIO_CONVERT (gst_pad_get_parent (pad));
otherpad = (pad == this->src) ? this->sink : this->src;
othercaps = gst_pad_get_allowed_caps (otherpad);
GST_DEBUG_CAPS ("othercaps are", othercaps);
caps = gst_caps_copy (othercaps);
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
gst_audio_structure_set_int (structure, GST_AUDIO_FIELD_CHANNELS |
GST_AUDIO_FIELD_ENDIANNESS |
GST_AUDIO_FIELD_WIDTH |
GST_AUDIO_FIELD_DEPTH |
GST_AUDIO_FIELD_SIGNED);
/* FIXME:
* since gst_structure_set doesn't handle lists, we need to do lists
* manually; would be nice if this could be done more easily */
}
return caps;
}
static void
gst_audio_convert_init (GstAudioConvert *this)
{
/* sinkpad */
this->sink = gst_pad_new_from_template (
gst_static_pad_template_get (&gst_audio_convert_sink_template), "sink");
gst_pad_set_link_function (this->sink, gst_audio_convert_link);
gst_pad_set_getcaps_function (this->sink, gst_audioconvert_getcaps);
gst_element_add_pad (GST_ELEMENT(this), this->sink);
/* srcpad */
this->src = gst_pad_new_from_template (
gst_static_pad_template_get (&gst_audio_convert_src_template), "src");
gst_pad_set_link_function (this->src, gst_audio_convert_link);
gst_pad_set_getcaps_function (this->src, gst_audioconvert_getcaps);
gst_element_add_pad (GST_ELEMENT(this), this->src);
gst_pad_set_chain_function(this->sink, gst_audio_convert_chain);
/* clear important variables */
this->caps_set[0] = this->caps_set[1] = FALSE;
this->convert_internal = NULL;
}
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, GstData *_data)
{
GstBuffer *buf = GST_BUFFER (_data);
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;
}
g_assert(this->caps_set[0] && this->caps_set[1]);
/**
* 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, GST_DATA (buf));
}
static GstPadLinkReturn
gst_audio_convert_link (GstPad *pad, const GstCaps *caps)
{
GstAudioConvert *this;
gint nr = 0;
gint rate, endianness, depth, width, channels;
gboolean sign;
GstStructure *structure;
gboolean ret;
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));
/* nr is 0 for sink pad, 1 for src pad */
nr = (pad == this->sink) ? 0 : (pad == this->src) ? 1 : -1;
g_assert (nr > -1);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "channels", &channels);
ret &= gst_structure_get_boolean (structure, "signed", &sign);
ret &= gst_structure_get_int (structure, "depth", &depth);
ret &= gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "rate", &rate);
endianness = G_BYTE_ORDER;
if (width != 8) {
ret &= gst_structure_get_int (structure, "endianness", &endianness);
}
if (!ret) {
GST_DEBUG ("could not get some values from structure");
return GST_PAD_LINK_REFUSED;
}
/* we don't convert rate changes, this is done by audioscale */
/* 1 - nr is "the other caps" */
if ((this->caps_set[1 - nr]) &&
(rate != this->rate[1 - nr])) {
GstPad *otherpad;
GstPadLinkReturn ret;
otherpad = (nr) ? this->src : this->sink;
if (gst_pad_is_negotiated (otherpad))
{
GstCaps *othercaps = gst_caps_copy (othercaps);
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, rate, NULL);
ret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (ret))
{
GST_DEBUG ("could not gst_pad_try_set_caps on otherpad using othercaps");
return GST_PAD_LINK_REFUSED;
}
}
this->rate[1 - nr] = rate;
}
GST_DEBUG ("setting caps_set[%d] to TRUE", nr);
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;
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;
this->convert_internal = NULL;
break;
default:
break;
}
if (parent_class->change_state) {
return parent_class->change_state (element);
} else {
return GST_STATE_SUCCESS;
}
}
/* 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] == 32 && 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 * 32 / this->width[0]);
count = ret->size / 4;
src = buf->data + (count - 1) * (this->width[0] / 8);
dest = (gint32 *) ret->data;
for (i = count - 1; i >= 0; i--) {
switch (this->width[0]) {
case 8:
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 16:
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 32:
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] == 32 && 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] / 32);
dest = ret->data;
src = (gint32 *) buf->data;
count = ret->size / (this->width[1] / 8);
for (i = 0; i < count; i++) {
gint32 int_value = *src;
src++;
switch (this->width[1]) {
case 8:
if (this->sign[1]) {
POPULATE (gint8, GINT8_IDENTITY, GINT8_IDENTITY);
} else {
POPULATE (guint8, GUINT8_IDENTITY, GUINT8_IDENTITY);
}
break;
case 16:
if (this->sign[1]) {
POPULATE (gint16, GINT16_TO_BE, GINT16_TO_LE);
} else {
POPULATE (guint16, GUINT16_TO_BE, GUINT16_TO_LE);
}
break;
case 32:
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;
gint i, count;
guint32 *src, *dest;
if (this->channels[0] == this->channels[1])
return buf;
count = GST_BUFFER_SIZE (buf) / 4 / this->channels[0];
ret = gst_audio_convert_get_buffer (buf, count * 4 * this->channels[1]);
src = (guint32 *) GST_BUFFER_DATA (buf);
dest = (guint32 *) GST_BUFFER_DATA (ret);
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 gboolean
plugin_init (GstPlugin *plugin)
{
if (!gst_element_register (plugin, "audioconvert", GST_RANK_NONE, GST_TYPE_AUDIO_CONVERT))
return FALSE;
if (!gst_plugin_load ("gstaudio")) return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstaudioconvert",
"Convert audio to different formats",
plugin_init,
VERSION,
"LGPL",
GST_PACKAGE,
GST_ORIGIN)
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