mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-19 06:46:38 +00:00
Code Issues 1 Projects Releases Wiki Activity

audioresample: remove old audioresample plugin that was never even ported to 0.10

Browse source
This commit is contained in:
Tim-Philipp Müller 2012-06-12 19:02:57 +01:00
parent 2ca936ef57
commit c1facd0c36
4 changed files with 0 additions and 523 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

7
ext/Makefile.am
View file

@ -34,12 +34,6 @@ ARTSC_DIR=
AUDIOFILE_DIR=
# endif
# if USE_AUDIORESAMPLE
# AUDIORESAMPLE_DIR=audioresample
# else
AUDIORESAMPLE_DIR=
# endif
if USE_BZ2
BZ2_DIR=bz2
else
@ -405,7 +399,6 @@ SUBDIRS=\
$(ARTS_DIR) \
$(ARTSC_DIR) \
$(AUDIOFILE_DIR) \
$(AUDIORESAMPLE_DIR) \
$(BZ2_DIR) \
$(CDAUDIO_DIR) \
$(CELT_DIR) \

11
ext/audioresample/Makefile.am
View file

@ -1,11 +0,0 @@
plugin_LTLIBRARIES = libgstaudioresample.la
libgstaudioresample_la_SOURCES = gstaudioresample.c
libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(AUDIORESAMPLE_CFLAGS)
libgstaudioresample_la_LIBADD = $(AUDIORESAMPLE_LIBS)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudioresample_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstaudioresample.h

431
ext/audioresample/gstaudioresample.c
View file

@ -1,431 +0,0 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
*
* 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 "gstaudioresample.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
#define GST_CAT_DEFAULT audioresample_debug
/* Audioresample signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FILTERLEN
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS (\
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true"
#if 0
/* disabled because it segfaults */
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32"
#endif
)
static GstStaticPadTemplate gst_audioresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_base_init (gpointer g_class);
static void gst_audioresample_class_init (AudioresampleClass * klass);
static void gst_audioresample_init (Audioresample * audioresample);
static void gst_audioresample_dispose (GObject * object);
static void gst_audioresample_chain (GstPad * pad, GstData * _data);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
GType audioresample_get_type (void)
{
static GType audioresample_type = 0;
if (!audioresample_type)
{
static const GTypeInfo audioresample_info = {
sizeof (AudioresampleClass),
gst_audioresample_base_init,
NULL,
(GClassInitFunc) gst_audioresample_class_init,
NULL,
NULL,
sizeof (Audioresample), 0,
(GInstanceInitFunc) gst_audioresample_init,};
audioresample_type =
g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
&audioresample_info, 0);
}
return audioresample_type;
}
static void gst_audioresample_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_audioresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_sink_template));
gst_element_class_set_details_simple (gstelement_class, "Audio scaler",
"Filter/Converter/Audio",
"Resample audio", "David Schleef <ds@schleef.org>");
}
static void gst_audioresample_class_init (AudioresampleClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
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_PARAM_STATIC_STRINGS));
parent_class = g_type_class_peek_parent (klass);
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
"audioresample element");
}
static void gst_audioresample_expand_caps (GstCaps * caps)
{
gint i;
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return;
}
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
}
}
static GstCaps *gst_audioresample_getcaps (GstPad * pad)
{
Audioresample *audioresample;
GstCaps *caps;
GstPad *otherpad;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
caps = gst_pad_get_allowed_caps (otherpad);
gst_audioresample_expand_caps (caps);
return caps;
}
static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
{
Audioresample *audioresample;
GstPad *otherpad;
int rate;
GstCaps *copy;
GstStructure *structure;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (pad == audioresample->srcpad) {
otherpad = audioresample->sinkpad;
rate = audioresample->i_rate;
} else
{
otherpad = audioresample->srcpad;
rate = audioresample->o_rate;
}
if (!GST_PAD_IS_NEGOTIATING (otherpad))
return NULL;
if (gst_caps_get_size (caps) > 1)
return NULL;
copy = gst_caps_copy (caps);
structure = gst_caps_get_structure (copy, 0);
if (rate) {
if (gst_structure_fixate_field_nearest_int (structure, "rate", rate)) {
return copy;
}
}
gst_caps_free (copy);
return NULL;
}
static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
const GstCaps * caps)
{
Audioresample *audioresample;
GstStructure *structure;
int rate;
int channels;
gboolean ret;
GstPad *otherpad;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rate);
ret &= gst_structure_get_int (structure, "channels", &channels);
if (!ret)
{
return GST_PAD_LINK_REFUSED;
}
if (gst_pad_is_negotiated (otherpad))
{
GstCaps *othercaps = gst_caps_copy (caps);
int otherrate;
GstPadLinkReturn linkret;
if (pad == audioresample->srcpad) {
otherrate = audioresample->i_rate;
} else {
otherrate = audioresample->o_rate;
}
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
linkret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (linkret)) {
return GST_PAD_LINK_REFUSED;
}
}
audioresample->channels = channels;
resample_set_n_channels (audioresample->resample, audioresample->channels);
if (pad == audioresample->srcpad) {
audioresample->o_rate = rate;
resample_set_output_rate (audioresample->resample, audioresample->o_rate);
GST_DEBUG ("set o_rate to %d", rate);
} else {
audioresample->i_rate = rate;
resample_set_input_rate (audioresample->resample, audioresample->i_rate);
GST_DEBUG ("set i_rate to %d", rate);
}
return GST_PAD_LINK_OK;
}
static void gst_audioresample_init (Audioresample * audioresample)
{
ResampleState *r;
audioresample->sinkpad =
gst_pad_new_from_static_template (&gst_audioresample_sink_template,
"sink");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->sinkpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->sinkpad,
gst_audioresample_fixate);
audioresample->srcpad =
gst_pad_new_from_static_template (&gst_audioresample_src_template, "src");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->srcpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
r = resample_new ();
audioresample->resample = r;
resample_set_filter_length (r, 64);
resample_set_format (r, RESAMPLE_FORMAT_S16);
}
static void gst_audioresample_dispose (GObject * object)
{
Audioresample *audioresample = GST_AUDIORESAMPLE (object);
if (audioresample->resample) {
resample_free (audioresample->resample);
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void gst_audioresample_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
Audioresample *audioresample;
ResampleState *r;
guchar *data;
gulong size;
int outsize;
GstBuffer *outbuf;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (!GST_IS_BUFFER (_data)) {
gst_pad_push (audioresample->srcpad, _data);
return;
}
if (audioresample->passthru) {
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
return;
}
r = audioresample->resample;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG ("got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
buf);
outsize = resample_get_output_size (r);
/* FIXME this is audioresample being dumb. dunno why */
if (outsize == 0) {
GST_ERROR ("overriding outbuf size");
outsize = size;
}
outbuf = gst_buffer_new_and_alloc (outsize);
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
Audioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
break;
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
Audioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
g_value_set_int (value, audioresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean plugin_init (GstPlugin * plugin)
{
resample_init ();
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIORESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audioresample,
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN)

74
ext/audioresample/gstaudioresample.h
View file

@ -1,74 +0,0 @@
/* 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.
*/
#ifndef __AUDIORESAMPLE_H__
#define __AUDIORESAMPLE_H__
#include <gst/gst.h>
#include <audioresample/resample.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIORESAMPLE \
(audioresample_get_type())
#define GST_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,Audioresample))
#define GST_AUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,AudioresampleClass))
#define GST_IS_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
typedef struct _Audioresample Audioresample;
typedef struct _AudioresampleClass AudioresampleClass;
struct _Audioresample {
GstElement element;
GstPad *sinkpad,*srcpad;
gboolean passthru;
gint64 offset;
int channels;
int i_rate;
int o_rate;
int filter_length;
ResampleState * resample;
};
struct _AudioresampleClass {
GstElementClass parent_class;
};
GType gst_audioresample_get_type(void);
G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */