gstreamer/gst/audioscale/gstaudioscale.c
David Schleef 4f3158a64c gst/audioscale/gstaudioscale.c: Fix getcaps to expand and union lists. (bug )
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_expand_value),
(gst_audioscale_getcaps): Fix getcaps to expand and union lists.
(bug )
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c: (gst_break_my_data_plugin_init):
* gst/debug/gstdebug.c: (plugin_init):  Merge elements into one
plugin.
* gst/debug/negotiation.c: (gst_gst_negotiation_get_type),
(gst_negotiation_base_init), (gst_negotiation_class_init),
(gst_negotiation_init), (gst_negotiation_getcaps),
(gst_negotiation_pad_link), (gst_negotiation_chain),
(gst_negotiation_set_property), (gst_negotiation_get_property),
(gst_negotiation_plugin_init):  New element to talk about random
negotiation things happening in a pipeline.
2004-03-31 22:36:36 +00:00

488 lines
13 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.
*/
/* Element-Checklist-Version: 5 */
#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/Converter/Audio",
"Resample audio",
"David Schleef <ds@schleef.org>");
/* Audioscale signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FILTERLEN,
ARG_METHOD,
/* FILL ME */
};
static GstStaticPadTemplate gst_audioscale_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
);
static GstStaticPadTemplate gst_audioscale_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
);
#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[] = {
{GST_RESAMPLE_NEAREST, "0", "Nearest"},
{GST_RESAMPLE_BILINEAR, "1", "Bilinear"},
{GST_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_static_pad_template_get (&gst_audioscale_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioscale_sink_template));
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;
gobject_class->set_property = gst_audioscale_set_property;
gobject_class->get_property = gst_audioscale_get_property;
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, GST_RESAMPLE_SINC,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
}
static void
gst_audioscale_expand_value (GValue * dest, const GValue * src)
{
int rate_min, rate_max;
if (G_VALUE_TYPE (src) == G_TYPE_INT ||
G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE) {
if (G_VALUE_TYPE (src) == G_TYPE_INT) {
rate_min = g_value_get_int (src);
rate_max = rate_min;
} else {
rate_min = gst_value_get_int_range_min (src);
rate_max = gst_value_get_int_range_max (src);
}
rate_min /= 2;
if (rate_min < 1)
rate_min = 1;
if (rate_max < G_MAXINT / 2) {
rate_max *= 2;
} else {
rate_max = G_MAXINT;
}
g_value_init (dest, GST_TYPE_INT_RANGE);
gst_value_set_int_range (dest, rate_min, rate_max);
return;
}
if (G_VALUE_TYPE (src) == GST_TYPE_LIST) {
int i;
g_value_init (dest, GST_TYPE_LIST);
for (i = 0; i < gst_value_list_get_size (src); i++) {
const GValue *s = gst_value_list_get_value (src, i);
GValue d = { 0 };
int j;
gst_audioscale_expand_value (&d, s);
for (j = 0; j < gst_value_list_get_size (dest); j++) {
const GValue *s2 = gst_value_list_get_value (dest, j);
GValue d2 = { 0 };
gst_value_union (&d2, &d, s2);
if (G_VALUE_TYPE (&d2) == GST_TYPE_INT_RANGE) {
g_value_unset ((GValue *) s2);
gst_value_init_and_copy ((GValue *) s2, &d2);
break;
}
g_value_unset (&d2);
}
if (j == gst_value_list_get_size (dest)) {
gst_value_list_append_value (dest, &d);
}
g_value_unset (&d);
}
if (gst_value_list_get_size (dest) == 1) {
const GValue *s = gst_value_list_get_value (dest, 0);
GValue d = { 0 };
gst_value_init_and_copy (&d, s);
g_value_unset (dest);
gst_value_init_and_copy (dest, &d);
g_value_unset (&d);
}
return;
}
GST_ERROR ("unexpected value type");
}
static GstCaps *
gst_audioscale_getcaps (GstPad * pad)
{
Audioscale *audioscale;
GstCaps *caps;
GstPad *otherpad;
int i;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
otherpad = (pad == audioscale->srcpad) ? audioscale->sinkpad :
audioscale->srcpad;
caps = gst_pad_get_allowed_caps (otherpad);
/* we do this hack, because the audioscale lib doesn't handle
* rate conversions larger than a factor of 2 */
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
GValue dest = { 0 };
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return NULL;
}
gst_audioscale_expand_value (&dest, value);
gst_structure_set_value (structure, "rate", &dest);
}
return caps;
}
static GstPadLinkReturn
gst_audioscale_link (GstPad * pad, const GstCaps * caps)
{
Audioscale *audioscale;
gst_resample_t *r;
GstStructure *structure;
int rate;
int channels;
int ret;
GstPadLinkReturn link_ret;
GstPad *otherpad;
audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
r = audioscale->gst_resample;
otherpad = (pad == audioscale->srcpad) ? audioscale->sinkpad
: audioscale->srcpad;
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rate);
ret &= gst_structure_get_int (structure, "channels", &channels);
link_ret = gst_pad_try_set_caps (otherpad, gst_caps_copy (caps));
if (GST_PAD_LINK_SUCCESSFUL (link_ret)) {
audioscale->passthru = TRUE;
r->channels = channels;
r->i_rate = rate;
r->o_rate = rate;
return link_ret;
}
audioscale->passthru = FALSE;
if (gst_pad_is_negotiated (otherpad)) {
GstCaps *trycaps = gst_caps_copy (caps);
gst_caps_set_simple (trycaps,
"rate", G_TYPE_INT,
(int) ((pad == audioscale->srcpad) ? r->i_rate : r->o_rate), NULL);
link_ret = gst_pad_try_set_caps (otherpad, trycaps);
if (GST_PAD_LINK_FAILED (link_ret)) {
return link_ret;
}
}
r->channels = channels;
if (pad == audioscale->srcpad) {
r->o_rate = rate;
} else {
r->i_rate = rate;
}
gst_resample_reinit (r);
return GST_PAD_LINK_OK;
}
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->gst_resample->o_rate;
audioscale->offset +=
size / sizeof (gint16) / audioscale->gst_resample->channels;
return GST_BUFFER_DATA (audioscale->outbuf);
}
static void
gst_audioscale_init (Audioscale * audioscale)
{
gst_resample_t *r;
audioscale->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioscale_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_link);
gst_pad_set_getcaps_function (audioscale->sinkpad, gst_audioscale_getcaps);
audioscale->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioscale_src_template), "src");
gst_element_add_pad (GST_ELEMENT (audioscale), audioscale->srcpad);
gst_pad_set_link_function (audioscale->srcpad, gst_audioscale_link);
gst_pad_set_getcaps_function (audioscale->srcpad, gst_audioscale_getcaps);
r = g_new0 (gst_resample_t, 1);
audioscale->gst_resample = r;
r->priv = audioscale;
r->get_buffer = gst_audioscale_get_buffer;
r->method = GST_RESAMPLE_SINC;
r->channels = 0;
r->filter_length = 16;
r->i_rate = -1;
r->o_rate = -1;
r->format = GST_RESAMPLE_S16;
/*r->verbose = 1; */
gst_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)));
gst_resample_scale (audioscale->gst_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;
gst_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->gst_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;
}
gst_resample_reinit (r);
}
static void
gst_audioscale_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
Audioscale *src;
gst_resample_t *r;
src = GST_AUDIOSCALE (object);
r = src->gst_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_PACKAGE, GST_ORIGIN)