gstreamer/gst/audioscale/gstaudioscale.c
Thomas Vander Stichele e976ccb298 PadConnect -> PadLink
Original commit message from CVS:
PadConnect -> PadLink
2003-01-10 13:38:32 +00:00

345 lines
9.2 KiB
C

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 <string.h>
#include <math.h>
/*#define DEBUG_ENABLED */
#include <gstaudioscale.h>
#include <gst/audio/audio.h>
#include <gst/resample/resample.h>
/* elementfactory information */
static GstElementDetails audioscale_details = {
"Audio scaler",
"Filter/Audio",
"LGPL",
"Audio resampler",
VERSION,
"Wim Taymans <wim.taymans@chello.be>",
"(C) 2000",
};
/* Audioscale signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_FREQUENCY,
ARG_FILTERLEN,
ARG_METHOD,
/* FILL ME */
};
static GstPadTemplate *
sink_template (void)
{
static GstPadTemplate *template = NULL;
if (!template) {
template = gst_pad_template_new ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
gst_caps_new
("audioscale_sink",
"audio/raw", GST_AUDIO_INT_PAD_TEMPLATE_PROPS), NULL);
}
return template;
}
static GstPadTemplate *
src_template (void)
{
static GstPadTemplate *template = NULL;
if (!template) {
template = gst_pad_template_new ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
gst_caps_new
("audioscale_src",
"audio/raw", GST_AUDIO_INT_PAD_TEMPLATE_PROPS), NULL);
}
return template;
}
#define GST_TYPE_AUDIOSCALE_METHOD (gst_audioscale_method_get_type())
static GType
gst_audioscale_method_get_type (void)
{
static GType audioscale_method_type = 0;
static GEnumValue audioscale_methods[] = {
{ RESAMPLE_NEAREST, "0", "Nearest" },
{ RESAMPLE_BILINEAR, "1", "Bilinear" },
{ RESAMPLE_SINC, "2", "Sinc" },
{ 0, NULL, NULL },
};
if(!audioscale_method_type){
audioscale_method_type = g_enum_register_static("GstAudioscaleMethod",
audioscale_methods);
}
return audioscale_method_type;
}
static void gst_audioscale_class_init (AudioscaleClass *klass);
static void gst_audioscale_init (Audioscale *audioscale);
static void gst_audioscale_chain (GstPad *pad, GstBuffer *buf);
static void gst_audioscale_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audioscale_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_audioscale_signals[LAST_SIGNAL] = { 0 }; */
GType
audioscale_get_type (void)
{
static GType audioscale_type = 0;
if (!audioscale_type) {
static const GTypeInfo audioscale_info = {
sizeof(AudioscaleClass), NULL,
NULL,
(GClassInitFunc)gst_audioscale_class_init,
NULL,
NULL,
sizeof(Audioscale),
0,
(GInstanceInitFunc)gst_audioscale_init,
};
audioscale_type = g_type_register_static(GST_TYPE_ELEMENT, "Audioscale", &audioscale_info, 0);
}
return audioscale_type;
}
static void
gst_audioscale_class_init (AudioscaleClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_FREQUENCY,
g_param_spec_int ("frequency","frequency","frequency",
0,G_MAXINT,44100,G_PARAM_READWRITE|G_PARAM_CONSTRUCT));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE|G_PARAM_CONSTRUCT));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_METHOD,
g_param_spec_enum ("method", "method", "method", GST_TYPE_AUDIOSCALE_METHOD,
RESAMPLE_SINC, G_PARAM_READWRITE|G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
gobject_class->set_property = gst_audioscale_set_property;
gobject_class->get_property = gst_audioscale_get_property;
}
static GstPadLinkReturn
gst_audioscale_sinkconnect (GstPad * pad, GstCaps * caps)
{
Audioscale *audioscale;
resample_t *r;
GstCaps *newcaps;
gint rate;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
r = audioscale->resample;
gst_caps_get_int (caps, "rate", &rate);
gst_caps_get_int (caps, "channels", &r->channels);
r->i_rate = rate;
resample_reinit(r);
newcaps = gst_caps_copy (caps);
gst_caps_set (newcaps, "rate", GST_PROPS_INT_TYPE, audioscale->targetfrequency, NULL);
if (GST_CAPS_IS_FIXED (caps))
return gst_pad_try_set_caps (audioscale->srcpad, newcaps);
else
return GST_PAD_LINK_DELAYED;
}
static void *
gst_audioscale_get_buffer (void *priv, unsigned int size)
{
Audioscale * audioscale = priv;
audioscale->outbuf = gst_buffer_new();
GST_BUFFER_SIZE(audioscale->outbuf) = size;
GST_BUFFER_DATA(audioscale->outbuf) = g_malloc(size);
GST_BUFFER_TIMESTAMP(audioscale->outbuf) = audioscale->offset * GST_SECOND / audioscale->targetfrequency;
audioscale->offset += size / sizeof(gint16) / audioscale->resample->channels;
return GST_BUFFER_DATA(audioscale->outbuf);
}
static void
gst_audioscale_init (Audioscale *audioscale)
{
resample_t *r;
audioscale->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_template), "sink");
gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->sinkpad);
gst_pad_set_chain_function(audioscale->sinkpad,gst_audioscale_chain);
gst_pad_set_link_function (audioscale->sinkpad, gst_audioscale_sinkconnect);
audioscale->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_template), "src");
gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->srcpad);
r = g_new0(resample_t,1);
audioscale->resample = r;
r->priv = audioscale;
r->get_buffer = gst_audioscale_get_buffer;
r->method = RESAMPLE_SINC;
r->channels = 0;
r->filter_length = 16;
r->i_rate = -1;
r->o_rate = -1;
r->format = RESAMPLE_S16;
/*r->verbose = 1; */
resample_init(r);
/* we will be reinitialized when the G_PARAM_CONSTRUCTs hit */
}
static void
gst_audioscale_chain (GstPad *pad, GstBuffer *buf)
{
Audioscale *audioscale;
guchar *data;
gulong size;
g_return_if_fail(pad != NULL);
g_return_if_fail(GST_IS_PAD(pad));
g_return_if_fail(buf != NULL);
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
data = GST_BUFFER_DATA(buf);
size = GST_BUFFER_SIZE(buf);
GST_DEBUG (0,
"gst_audioscale_chain: got buffer of %ld bytes in '%s'\n",
size, gst_element_get_name (GST_ELEMENT (audioscale)));
resample_scale (audioscale->resample, data, size);
gst_pad_push (audioscale->srcpad, audioscale->outbuf);
gst_buffer_unref (buf);
}
static void
gst_audioscale_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
Audioscale *src;
resample_t *r;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail(GST_IS_AUDIOSCALE(object));
src = GST_AUDIOSCALE(object);
r = src->resample;
switch (prop_id) {
case ARG_FREQUENCY:
src->targetfrequency = g_value_get_int (value);
r->o_rate = src->targetfrequency;
break;
case ARG_FILTERLEN:
r->filter_length = g_value_get_int (value);
GST_DEBUG_ELEMENT (0, GST_ELEMENT(src), "new filter length %d\n", r->filter_length);
break;
case ARG_METHOD:
r->method = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
resample_reinit (r);
}
static void
gst_audioscale_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
Audioscale *src;
resample_t *r;
src = GST_AUDIOSCALE (object);
r = src->resample;
switch (prop_id) {
case ARG_FREQUENCY:
g_value_set_int (value, src->targetfrequency);
break;
case ARG_FILTERLEN:
g_value_set_int (value, r->filter_length);
break;
case ARG_METHOD:
g_value_set_enum (value, r->method);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
/* create an elementfactory for the audioscale element */
factory = gst_element_factory_new ("audioscale", GST_TYPE_AUDIOSCALE, &audioscale_details);
g_return_val_if_fail(factory != NULL, FALSE);
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
/* load support library */
if (!gst_library_load ("gstresample"))
return FALSE;
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioscale",
plugin_init
};