gstreamer/gst/audioscale/gstaudioscale.c

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/* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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 gst_audioscale_details = GST_ELEMENT_DETAILS (
"Audio scaler",
"Filter/Audio",
"Audio resampler",
"David Schleef <ds@schleef.org>"
);
/* Audioscale signals and args */
enum {
/* FILL ME */
LAST_SIGNAL
};
enum {
ARG_0,
ARG_FILTERLEN,
ARG_METHOD,
/* FILL ME */
};
GST_PAD_TEMPLATE_FACTORY (sink_factory,
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
gst_caps_new ("audioscale_sink",
"audio/x-raw-int",
GST_AUDIO_INT_PAD_TEMPLATE_PROPS)
);
GST_PAD_TEMPLATE_FACTORY (src_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
gst_caps_new ("audioscale_src",
"audio/x-raw-int",
GST_AUDIO_INT_PAD_TEMPLATE_PROPS)
);
#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_base_init (gpointer g_class);
static void gst_audioscale_class_init (AudioscaleClass *klass);
static void gst_audioscale_init (Audioscale *audioscale);
static void gst_audioscale_chain (GstPad *pad, GstData *_data);
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),
gst_audioscale_base_init,
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_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
GST_PAD_TEMPLATE_GET (src_factory));
gst_element_class_add_pad_template (gstelement_class,
GST_PAD_TEMPLATE_GET (sink_factory));
gst_element_class_set_details (gstelement_class, &gst_audioscale_details);
}
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_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 GstCaps *
gst_audioscale_getcaps (GstPad *pad, GstCaps *caps)
{
Audioscale *audioscale;
GstCaps *peercaps;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
if (pad == audioscale->srcpad){
peercaps = gst_pad_get_allowed_caps (audioscale->sinkpad);
}else{
peercaps = gst_pad_get_allowed_caps (audioscale->srcpad);
}
if(peercaps == GST_CAPS_NONE){
return GST_CAPS_NONE;
}
caps = gst_caps_copy (peercaps);
#if 1
/* we do this hack, because the audioscale lib doesn't handle
* rate conversions larger than a factor of 2 */
if(gst_caps_has_property_typed(caps, "rate", GST_PROPS_INT_RANGE_TYPE)){
int rate_min, rate_max;
gst_props_entry_get_int_range (gst_props_get_entry(caps->properties, "rate"),
&rate_min, &rate_max);
gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE((rate_min+1)/2,
rate_max*2));
}else{
int rate;
gst_caps_get_int (caps, "rate", &rate);
gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE((rate+1)/2,rate*2));
}
#else
gst_caps_set (caps, "rate", GST_PROPS_INT_RANGE(4000,96000));
#endif
return caps;
}
static GstPadLinkReturn
gst_audioscale_sink_link (GstPad * pad, GstCaps * caps)
{
Audioscale *audioscale;
resample_t *r;
GstCaps *caps1;
GstCaps *caps2;
GstCaps *peercaps;
gint rate;
int ret;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
r = audioscale->resample;
if (!GST_CAPS_IS_FIXED (caps)){
return GST_PAD_LINK_DELAYED;
}
ret = gst_pad_try_set_caps (audioscale->srcpad, caps);
if(ret == GST_PAD_LINK_OK || ret == GST_PAD_LINK_DONE){
audioscale->passthru = TRUE;
return ret;
}
audioscale->passthru = FALSE;
gst_caps_get_int (caps, "rate", &rate);
gst_caps_get_int (caps, "channels", &r->channels);
r->i_rate = rate;
resample_reinit(r);
peercaps = gst_pad_get_allowed_caps (audioscale->srcpad);
caps1 = gst_caps_copy (caps);
#if 1
/* we do this hack, because the audioscale lib doesn't handle
* rate conversions larger than a factor of 2 */
if(gst_caps_has_property_typed(caps1, "rate", GST_PROPS_INT_RANGE_TYPE)){
int rate_min, rate_max;
gst_props_entry_get_int_range (gst_props_get_entry(caps1->properties, "rate"),
&rate_min, &rate_max);
gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE((rate_min+1)/2,
rate_max*2));
}else{
gst_caps_get_int (caps1, "rate", &rate);
gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE((rate+1)/2,rate*2));
}
#else
gst_caps_set (caps1, "rate", GST_PROPS_INT_RANGE(4000,96000));
#endif
caps2 = gst_caps_intersect(caps1, peercaps);
gst_caps_unref(caps1);
if(caps2 == GST_CAPS_NONE){
return GST_PAD_LINK_REFUSED;
}
if (GST_CAPS_IS_FIXED (caps2)) {
ret = gst_pad_try_set_caps (audioscale->srcpad, caps2);
gst_caps_get_int (caps, "rate", &rate);
r->o_rate = rate;
audioscale->targetfrequency = rate;
resample_reinit(r);
return ret;
}
gst_caps_unref (caps2);
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_factory), "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_sink_link);
gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps);
audioscale->srcpad = gst_pad_new_from_template (
GST_PAD_TEMPLATE_GET (src_factory), "src");
gst_element_add_pad(GST_ELEMENT(audioscale),audioscale->srcpad);
gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
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, GstData *_data)
{
GstBuffer *buf = GST_BUFFER (_data);
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));
if (audioscale->passthru){
gst_pad_push (audioscale->srcpad, GST_DATA (buf));
return;
}
data = GST_BUFFER_DATA(buf);
size = GST_BUFFER_SIZE(buf);
GST_DEBUG (
"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, GST_DATA (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_FILTERLEN:
r->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (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_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 (GstPlugin *plugin)
{
/* load support library */
if (!gst_library_load ("gstresample"))
return FALSE;
if (!gst_element_register (plugin, "audioscale", GST_RANK_NONE,
GST_TYPE_AUDIOSCALE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioscale",
"Resamples audio",
plugin_init,
VERSION,
"LGPL",
GST_COPYRIGHT,
GST_PACKAGE,
GST_ORIGIN
)