gstreamer/gst/audiofx/audiowsinclimit.c
Sebastian Dröge a1c029bab5 gst/filter/: Reset the residue in BaseTransform::start to get a clean residue on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
2007-08-12 12:46:20 +00:00

533 lines
15 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 Sebastian Dröge <slomo@circular-chaos.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.
*
*
* 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/
*
* TODO: - Implement the convolution in place, probably only makes sense
* when using FFT convolution as currently the convolution itself
* is probably the bottleneck
* - Maybe allow cascading the filter to get a better stopband attenuation.
* Can be done by convolving a filter kernel with itself.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "gstlpwsinc.h"
#define GST_CAT_DEFAULT gst_lpwsinc_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails lpwsinc_details = GST_ELEMENT_DETAILS ("LPWSinc",
"Filter/Effect/Audio",
"Low-pass and High-pass Windowed sinc filter",
"Thomas <thomas@apestaart.org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LENGTH,
PROP_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_LPWSINC_MODE (gst_lpwsinc_mode_get_type ())
static GType
gst_lpwsinc_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 ("GstLPWSincMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN
};
#define GST_TYPE_LPWSINC_WINDOW (gst_lpwsinc_window_get_type ())
static GType
gst_lpwsinc_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"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstLPWSincWindow", values);
}
return gtype;
}
#define ALLOWED_CAPS \
"audio/x-raw-float, " \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_lpwsinc_debug, "lpwsinc", 0, "Low-pass and High-pass Windowed sinc filter plugin");
GST_BOILERPLATE_FULL (GstLPWSinc, gst_lpwsinc, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void lpwsinc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void lpwsinc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn lpwsinc_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean lpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size);
static gboolean lpwsinc_start (GstBaseTransform * base);
static gboolean lpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
/* Element class */
static void
gst_lpwsinc_dispose (GObject * object)
{
GstLPWSinc *self = GST_LPWSINC (object);
if (self->residue) {
g_free (self->residue);
self->residue = NULL;
}
if (self->kernel) {
g_free (self->kernel);
self->kernel = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_lpwsinc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &lpwsinc_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_lpwsinc_class_init (GstLPWSincClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = lpwsinc_set_property;
gobject_class->get_property = lpwsinc_get_property;
gobject_class->dispose = gst_lpwsinc_dispose;
g_object_class_install_property (gobject_class, PROP_FREQUENCY,
g_param_spec_double ("frequency", "Frequency",
"Cut-off Frequency (Hz)", 0.0, G_MAXDOUBLE, 0.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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, G_MAXINT, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_LPWSINC_MODE,
MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_LPWSINC_WINDOW,
WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
trans_class->transform = GST_DEBUG_FUNCPTR (lpwsinc_transform);
trans_class->get_unit_size = GST_DEBUG_FUNCPTR (lpwsinc_get_unit_size);
trans_class->start = GST_DEBUG_FUNCPTR (lpwsinc_start);
filter_class->setup = GST_DEBUG_FUNCPTR (lpwsinc_setup);
}
static void
gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
{
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
self->frequency = 0.0;
self->kernel = NULL;
self->residue = NULL;
self->have_kernel = FALSE;
}
static void
process_32 (GstLPWSinc * self, gfloat * src, gfloat * dst, guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
}
static void
process_64 (GstLPWSinc * self, gdouble * src, gdouble * dst,
guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
}
static void
lpwsinc_build_kernel (GstLPWSinc * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
/* fill the kernel */
len = self->kernel_length;
GST_DEBUG ("lpwsinc: initializing filter kernel of length %d", len);
if (GST_AUDIO_FILTER (self)->format.rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (GST_AUDIO_FILTER (self)->format.channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->frequency =
CLAMP (self->frequency, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
w = 2 * M_PI * (self->frequency / GST_AUDIO_FILTER (self)->format.rate);
if (self->kernel)
g_free (self->kernel);
self->kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
self->kernel[i] = w;
else
self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
/* windowing */
if (self->window == WINDOW_HAMMING)
self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
self->kernel[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += self->kernel[i];
for (i = 0; i < len; ++i)
self->kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
self->kernel[i] = -self->kernel[i];
self->kernel[len / 2] += 1.0;
}
/* set up the residue memory space */
if (self->residue)
g_free (self->residue);
self->residue =
g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->have_kernel = TRUE;
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
lpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstLPWSinc *self = GST_LPWSINC (base);
gboolean ret = TRUE;
if (format->width == 32)
self->process = (GstLPWSincProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstLPWSincProcessFunc) process_64;
else
ret = FALSE;
self->have_kernel = FALSE;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static gboolean
lpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
g_assert (size);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
*size = width * channels / 8;
return ret;
}
static GstFlowReturn
lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstLPWSinc *self = GST_LPWSINC (base);
GstClockTime timestamp;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
/* don't process data in passthrough-mode */
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
if (!self->have_kernel)
lpwsinc_build_kernel (self);
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
return GST_FLOW_OK;
}
static gboolean
lpwsinc_start (GstBaseTransform * base)
{
GstLPWSinc *self = GST_LPWSINC (base);
gint channels = GST_AUDIO_FILTER (self)->format.channels;
/* Reset the residue if already existing */
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
return TRUE;
}
static void
lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
g_return_if_fail (GST_IS_LPWSINC (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
GST_BASE_TRANSFORM_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
self->kernel_length = val;
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
case PROP_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->frequency = g_value_get_double (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (self);
self->mode = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_WINDOW:
GST_BASE_TRANSFORM_LOCK (self);
self->window = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
lpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_FREQUENCY:
g_value_set_double (value, self->frequency);
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;
}
}