mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
configure.ac: add audioresample and cairo plugins. Remove
Original commit message from CVS: * configure.ac: add audioresample and cairo plugins. Remove HAVE_MMX stuff, because it's not used. * ext/Makefile.am: same * ext/audioresample/Makefile.am: You are not ready for an audio resampling element based on audioresample. * ext/audioresample/gstaudioresample.c: * ext/audioresample/gstaudioresample.h: * ext/cairo/Makefile.am: You are not ready for overlay elements based on cairo. Don't look too closely, these elements kinda suck right now. * ext/cairo/gstcairo.c: new * ext/cairo/gsttextoverlay.c: new * ext/cairo/gsttextoverlay.h: new * ext/cairo/gsttimeoverlay.c: new * ext/cairo/gsttimeoverlay.h: new * gst-libs/gst/media-info/media-info-priv.h: fix compile problem with compilers that don't support variadic macros.
This commit is contained in:
David Schleef
parent
7de6c44f44
commit
e803c9f86d
7 changed files with 574 additions and 15 deletions
20
ChangeLog
20
ChangeLog
|
|
@ -1,3 +1,23 @@
|
|||
2004-12-15 David Schleef <ds@schleef.org>
|
||||
|
||||
* configure.ac: add audioresample and cairo plugins. Remove
|
||||
HAVE_MMX stuff, because it's not used.
|
||||
* ext/Makefile.am: same
|
||||
* ext/audioresample/Makefile.am: You are not ready for an
|
||||
audio resampling element based on audioresample.
|
||||
* ext/audioresample/gstaudioresample.c:
|
||||
* ext/audioresample/gstaudioresample.h:
|
||||
* ext/cairo/Makefile.am: You are not ready for overlay elements
|
||||
based on cairo. Don't look too closely, these elements kinda
|
||||
suck right now.
|
||||
* ext/cairo/gstcairo.c: new
|
||||
* ext/cairo/gsttextoverlay.c: new
|
||||
* ext/cairo/gsttextoverlay.h: new
|
||||
* ext/cairo/gsttimeoverlay.c: new
|
||||
* ext/cairo/gsttimeoverlay.h: new
|
||||
* gst-libs/gst/media-info/media-info-priv.h: fix compile
|
||||
problem with compilers that don't support variadic macros.
|
||||
|
||||
2004-12-15 Balamurali Viswanathan <balamurali.viswanathan@wipro.com>
|
||||
|
||||
Reviewed by: David Schleef <ds@schleef.org>
|
||||
|
|
|
|||
33
configure.ac
33
configure.ac
|
|
@ -806,6 +806,22 @@ GST_CHECK_FEATURE(AUDIOFILE, [audiofile], afsink afsrc, [
|
|||
AC_CHECK_LIB(audiofile, af_virtual_file_new, , HAVE_AUDIOFILE="no")
|
||||
fi])
|
||||
|
||||
dnl *** audioresample ***
|
||||
translit(dnm, m, l) AM_CONDITIONAL(USE_AUDIORESAMPLE, true)
|
||||
GST_CHECK_FEATURE(AUDIORESAMPLE, [audioresample plug-in], audioresample, [
|
||||
PKG_CHECK_MODULES(AUDIORESAMPLE, audioresample-0.1, HAVE_AUDIORESAMPLE=yes, HAVE_AUDIORESAMPLE=no)
|
||||
AC_SUBST(AUDIORESAMPLE_CFLAGS)
|
||||
AC_SUBST(AUDIORESAMPLE_LIBS)
|
||||
])
|
||||
|
||||
dnl *** cairo ***
|
||||
translit(dnm, m, l) AM_CONDITIONAL(USE_CAIRO, true)
|
||||
GST_CHECK_FEATURE(CAIRO, [cairo plug-in], cairo, [
|
||||
PKG_CHECK_MODULES(CAIRO, cairo, HAVE_CAIRO=yes, HAVE_CAIRO=no)
|
||||
AC_SUBST(CAIRO_CFLAGS)
|
||||
AC_SUBST(CAIRO_LIBS)
|
||||
])
|
||||
|
||||
dnl *** cdaudio ***
|
||||
translit(dnm, m, l) AM_CONDITIONAL(USE_CDAUDIO, true)
|
||||
GST_CHECK_FEATURE(CDAUDIO, [cdaudio], cdaudio, [
|
||||
|
|
@ -1722,15 +1738,6 @@ dnl ######################################################################
|
|||
dnl # Check command line parameters, and set shell variables accordingly #
|
||||
dnl ######################################################################
|
||||
|
||||
AC_ARG_ENABLE(libmmx,
|
||||
AC_HELP_STRING([--enable-libmmx],[use libmmx, if available]),
|
||||
[case "${enableval}" in
|
||||
yes) USE_LIBMMX=$HAVE_LIBMMX ;;
|
||||
no) USE_LIBMMX=no ;;
|
||||
*) AC_MSG_ERROR(bad value ${enableval} for --enable-libmmx) ;;
|
||||
esac],
|
||||
[USE_LIBMMX=$HAVE_LIBMMX]) dnl Default value
|
||||
|
||||
AC_ARG_ENABLE(atomic,
|
||||
AC_HELP_STRING([--enable-atomic],[use atomic reference counting header]),
|
||||
[case "${enableval}" in
|
||||
|
|
@ -1783,10 +1790,6 @@ dnl # Set defines according to variables set above #
|
|||
dnl ################################################
|
||||
|
||||
|
||||
if test "x$USE_LIBMMX" = xyes; then
|
||||
AC_DEFINE(HAVE_LIBMMX, 1, [Define if libmmx is available])
|
||||
fi
|
||||
|
||||
if test "x$USE_ATOMIC_H" = xyes; then
|
||||
AC_DEFINE(HAVE_ATOMIC_H, 1, [Define if atomic.h header file is available])
|
||||
fi
|
||||
|
|
@ -1804,8 +1807,6 @@ dnl #############################
|
|||
|
||||
dnl These should be "USE_*" instead of "HAVE_*", but some packages expect
|
||||
dnl HAVE_ and it is likely to be easier to stick with the old name
|
||||
AM_CONDITIONAL(HAVE_LIBMMX, test "x$USE_LIBMMX" = "xyes")
|
||||
|
||||
AM_CONDITIONAL(HAVE_ATOMIC_H, test "x$USE_ATOMIC_H" = "xyes")
|
||||
|
||||
AM_CONDITIONAL(EXPERIMENTAL, test "$EXPERIMENTAL" = "$xyes")
|
||||
|
|
@ -1980,6 +1981,8 @@ ext/alsa/Makefile
|
|||
ext/arts/Makefile
|
||||
ext/artsd/Makefile
|
||||
ext/audiofile/Makefile
|
||||
ext/audioresample/Makefile
|
||||
ext/cairo/Makefile
|
||||
ext/cdaudio/Makefile
|
||||
ext/cdparanoia/Makefile
|
||||
ext/dirac/Makefile
|
||||
|
|
|
|||
|
|
@ -34,6 +34,18 @@ else
|
|||
AUDIOFILE_DIR=
|
||||
endif
|
||||
|
||||
if USE_AUDIORESAMPLE
|
||||
AUDIORESAMPLE_DIR=audioresample
|
||||
else
|
||||
AUDIORESAMPLE_DIR=
|
||||
endif
|
||||
|
||||
if USE_CAIRO
|
||||
CAIRO_DIR=cairo
|
||||
else
|
||||
CAIRO_DIR=
|
||||
endif
|
||||
|
||||
if USE_CDAUDIO
|
||||
CDAUDIO_DIR=cdaudio
|
||||
else
|
||||
|
|
@ -383,6 +395,8 @@ SUBDIRS=\
|
|||
$(ARTS_DIR) \
|
||||
$(ARTSC_DIR) \
|
||||
$(AUDIOFILE_DIR) \
|
||||
$(AUDIORESAMPLE_DIR) \
|
||||
$(CAIRO_DIR) \
|
||||
$(CDAUDIO_DIR) \
|
||||
$(CDPARANOIA_DIR) \
|
||||
$(DIRAC_DIR) \
|
||||
|
|
@ -446,6 +460,8 @@ DIST_SUBDIRS=\
|
|||
arts \
|
||||
artsd \
|
||||
audiofile \
|
||||
audioresample \
|
||||
cairo \
|
||||
cdaudio \
|
||||
cdparanoia \
|
||||
dirac \
|
||||
|
|
|
|||
10
ext/audioresample/Makefile.am
Normal file
10
ext/audioresample/Makefile.am
Normal file
|
|
@ -0,0 +1,10 @@
|
|||
|
||||
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)
|
||||
|
||||
noinst_HEADERS = gstaudioresample.h
|
||||
|
||||
435
ext/audioresample/gstaudioresample.c
Normal file
435
ext/audioresample/gstaudioresample.c
Normal file
|
|
@ -0,0 +1,435 @@
|
|||
/* 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
|
||||
|
||||
/* elementfactory information */
|
||||
static GstElementDetails gst_audioresample_details =
|
||||
GST_ELEMENT_DETAILS ("Audio scaler",
|
||||
"Filter/Converter/Audio",
|
||||
"Resample audio",
|
||||
"David Schleef <ds@schleef.org>");
|
||||
|
||||
/* 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 (gstelement_class, &gst_audioresample_details);
|
||||
}
|
||||
|
||||
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));
|
||||
|
||||
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
|
||||
|
||||
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_caps_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_template (gst_static_pad_template_get
|
||||
(&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_template (gst_static_pad_template_get
|
||||
(&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;
|
||||
|
||||
/* it's not null if we got it, but it might not be ours */
|
||||
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, GST_ORIGIN)
|
||||
74
ext/audioresample/gstaudioresample.h
Normal file
74
ext/audioresample/gstaudioresample.h
Normal file
|
|
@ -0,0 +1,74 @@
|
|||
/* 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,Audioresample))
|
||||
#define GST_IS_AUDIORESAMPLE(obj) \
|
||||
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
|
||||
#define GST_IS_AUDIORESAMPLE_CLASS(obj) \
|
||||
(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__ */
|
||||
|
|
@ -23,6 +23,7 @@
|
|||
#define __GST_MEDIA_INFO_PRIV_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <glib/gprintf.h>
|
||||
|
||||
/* debug */
|
||||
GST_DEBUG_CATEGORY_EXTERN (gst_media_info_debug);
|
||||
|
|
|
|||
Loading…
Reference in a new issue