gstreamer/gst/audiofx/audiowsinclimit.c
Sebastian Dröge 93e3ed5a86 Merge branch 'master' into 0.11
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
2012-01-10 14:32:32 +01:00

403 lines
12 KiB
C

/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
* For the window functions see
* http://en.wikipedia.org/wiki/Window_function
*/
/**
* SECTION:element-audiowsinclimit
*
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
*
* This element has the advantage over the Chebyshev lowpass and highpass filter that it has
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include "audiowsinclimit.h"
#include "gst/glib-compat-private.h"
#define GST_CAT_DEFAULT gst_audio_wsinclimit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
PROP_0,
PROP_LENGTH,
PROP_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (gst_audio_wsinclimit_mode_get_type ())
static GType
gst_audio_wsinclimit_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioWSincLimitMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN,
WINDOW_GAUSSIAN,
WINDOW_COSINE,
WINDOW_HANN
};
#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (gst_audio_wsinclimit_window_get_type ())
static GType
gst_audio_wsinclimit_window_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{WINDOW_HAMMING, "Hamming window (default)",
"hamming"},
{WINDOW_BLACKMAN, "Blackman window",
"blackman"},
{WINDOW_GAUSSIAN, "Gaussian window",
"gaussian"},
{WINDOW_COSINE, "Cosine window",
"cosine"},
{WINDOW_HANN, "Hann window",
"hann"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioWSincLimitWindow", values);
}
return gtype;
}
#define gst_audio_wsinclimit_parent_class parent_class
G_DEFINE_TYPE (GstAudioWSincLimit, gst_audio_wsinclimit,
GST_TYPE_AUDIO_FX_BASE_FIR_FILTER);
static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_audio_wsinclimit_finalize (GObject * object);
static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
const GstAudioInfo * info);
#define POW2(x) (x)*(x)
static void
gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0,
"Low-pass and High-pass Windowed sinc filter plugin");
gobject_class->set_property = gst_audio_wsinclimit_set_property;
gobject_class->get_property = gst_audio_wsinclimit_get_property;
gobject_class->finalize = gst_audio_wsinclimit_finalize;
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider */
g_object_class_install_property (gobject_class, PROP_FREQUENCY,
g_param_spec_float ("cutoff", "Cutoff",
"Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
3, 256000, 101,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE,
MODE_LOW_PASS,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW,
WINDOW_HAMMING,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
gst_element_class_set_details_simple (gstelement_class,
"Low pass & high pass filter", "Filter/Effect/Audio",
"Low pass and high pass windowed sinc filter",
"Thomas Vander Stichele <thomas at apestaart dot org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsinclimit_setup);
}
static void
gst_audio_wsinclimit_init (GstAudioWSincLimit * self)
{
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
self->cutoff = 0.0;
self->lock = g_mutex_new ();
}
static void
gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
gdouble *kernel = NULL;
gint rate, channels;
len = self->kernel_length;
rate = GST_AUDIO_FILTER_RATE (self);
channels = GST_AUDIO_FILTER_CHANNELS (self);
if (rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->cutoff = CLAMP (self->cutoff, 0.0, rate / 2);
GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->cutoff,
(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
/* fill the kernel */
w = 2 * G_PI * (self->cutoff / rate);
kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == (len - 1) / 2.0)
kernel[i] = w;
else
kernel[i] = sin (w * (i - (len - 1) / 2)) / (i - (len - 1) / 2.0);
/* windowing */
switch (self->window) {
case WINDOW_HAMMING:
kernel[i] *= (0.54 - 0.46 * cos (2 * G_PI * i / (len - 1)));
break;
case WINDOW_BLACKMAN:
kernel[i] *= (0.42 - 0.5 * cos (2 * G_PI * i / (len - 1)) +
0.08 * cos (4 * G_PI * i / (len - 1)));
break;
case WINDOW_GAUSSIAN:
kernel[i] *= exp (-0.5 * POW2 (3.0 / len * (2 * i - (len - 1))));
break;
case WINDOW_COSINE:
kernel[i] *= cos (G_PI * i / (len - 1) - G_PI / 2);
break;
case WINDOW_HANN:
kernel[i] *= 0.5 * (1 - cos (2 * G_PI * i / (len - 1)));
break;
}
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += kernel[i];
for (i = 0; i < len; ++i)
kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
kernel[i] = -kernel[i];
if (len % 2 == 1) {
kernel[(len - 1) / 2] += 1.0;
} else {
kernel[len / 2 - 1] += 0.5;
kernel[len / 2] += 0.5;
}
}
gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
kernel, self->kernel_length, (len - 1) / 2);
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_wsinclimit_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
gst_audio_wsinclimit_build_kernel (self);
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
}
static void
gst_audio_wsinclimit_finalize (GObject * object)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
g_mutex_free (self->lock);
self->lock = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
g_return_if_fail (GST_IS_AUDIO_WSINC_LIMIT (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
g_mutex_lock (self->lock);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
(self));
self->kernel_length = val;
gst_audio_wsinclimit_build_kernel (self);
}
g_mutex_unlock (self->lock);
break;
}
case PROP_FREQUENCY:
g_mutex_lock (self->lock);
self->cutoff = g_value_get_float (value);
gst_audio_wsinclimit_build_kernel (self);
g_mutex_unlock (self->lock);
break;
case PROP_MODE:
g_mutex_lock (self->lock);
self->mode = g_value_get_enum (value);
gst_audio_wsinclimit_build_kernel (self);
g_mutex_unlock (self->lock);
break;
case PROP_WINDOW:
g_mutex_lock (self->lock);
self->window = g_value_get_enum (value);
gst_audio_wsinclimit_build_kernel (self);
g_mutex_unlock (self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_FREQUENCY:
g_value_set_float (value, self->cutoff);
break;
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_WINDOW:
g_value_set_enum (value, self->window);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}