/*
* GStreamer
- * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * Copyright (C) 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
*
* 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.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/*
*/
/**
- * SECTION:element-audiochebyshevfreqlimit
- * @short_description: Chebyshev low pass and high pass filter
+ * SECTION:element-audiocheblimit
*
- * <refsect2>
- * <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
- * </para>
- * <para>
+ *
* This element has the advantage over the windowed sinc lowpass and highpass filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
- * </para>
- * <para>
+ *
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
- * </para>
- * <para>
+ *
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
- * </para>
- * <para>
+ *
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
- * <para><note>
+ * <note><para>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
- * </note></para>
- * <title>Example launch line</title>
+ * </para></note>
* <para>
- * <programlisting>
- * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
- * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
- * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
- * </programlisting>
- * </para>
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
+ * ]|
* </refsect2>
*/
#include "config.h"
#endif
+#include <string.h>
+
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
-#include <gst/controller/gstcontroller.h>
#include <math.h>
-#include "audiochebyshevfreqlimit.h"
+#include "math_compat.h"
-#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
-GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+#include "audiocheblimit.h"
-static const GstElementDetails element_details =
-GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
- "Filter/Effect/Audio",
- "Chebyshev low pass and high pass filter",
- "Sebastian Dröge <slomo@circular-chaos.org>");
+#include "gst/glib-compat-private.h"
-/* Filter signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
+#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
PROP_POLES
};
-#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_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
+#define gst_audio_cheb_limit_parent_class parent_class
+G_DEFINE_TYPE (GstAudioChebLimit,
+ gst_audio_cheb_limit, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
-GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
- gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
- DEBUG_INIT);
-
-static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
+static void gst_audio_cheb_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
+static void gst_audio_cheb_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_audio_cheb_limit_finalize (GObject * object);
-static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
- GstRingBufferSpec * format);
-static GstFlowReturn
-gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
- GstBuffer * buf);
-static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
-
-static void process_64 (GstAudioChebyshevFreqLimit * filter,
- gdouble * data, guint num_samples);
-static void process_32 (GstAudioChebyshevFreqLimit * filter,
- gfloat * data, guint num_samples);
+static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
+ const GstAudioInfo * info);
enum
{
MODE_HIGH_PASS
};
-#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
static GType
-gst_audio_chebyshev_freq_limit_mode_get_type (void)
+gst_audio_cheb_limit_mode_get_type (void)
{
static GType gtype = 0;
{0, NULL, NULL}
};
- gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
+ gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
-gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
- GstCaps *caps;
-
- gst_element_class_set_details (element_class, &element_details);
-
- caps = gst_caps_from_string (ALLOWED_CAPS);
- gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
- caps);
- gst_caps_unref (caps);
-}
-
-static void
-gst_audio_chebyshev_freq_limit_dispose (GObject * object)
-{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
-
- if (filter->a) {
- g_free (filter->a);
- filter->a = NULL;
- }
-
- if (filter->b) {
- g_free (filter->b);
- filter->b = NULL;
- }
-
- if (filter->channels) {
- GstAudioChebyshevFreqLimitChannelCtx *ctx;
- gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
-
- for (i = 0; i < channels; i++) {
- ctx = &filter->channels[i];
- g_free (ctx->x);
- g_free (ctx->y);
- }
-
- g_free (filter->channels);
- filter->channels = NULL;
- }
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
- klass)
+gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
{
- GObjectClass *gobject_class;
- GstBaseTransformClass *trans_class;
- GstAudioFilterClass *filter_class;
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *gstelement_class = (GstElementClass *) klass;
+ GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
- gobject_class = (GObjectClass *) klass;
- trans_class = (GstBaseTransformClass *) klass;
- filter_class = (GstAudioFilterClass *) klass;
+ GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0,
+ "audiocheblimit element");
- gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
- gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
- gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
+ gobject_class->set_property = gst_audio_cheb_limit_set_property;
+ gobject_class->get_property = gst_audio_cheb_limit_get_property;
+ gobject_class->finalize = gst_audio_cheb_limit_finalize;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode",
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
- G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
- G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
- 100000.0, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 100000.0, 0.0,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
- 200.0, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 200.0, 0.25,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
/* FIXME: What to do about this upper boundary? With a cutoff frequency of
* rate/4 32 poles are completely possible, with a cutoff frequency very low
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next even number",
- 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 2, 32, 4,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "Low pass & high pass filter",
+ "Filter/Effect/Audio",
+ "Chebyshev low pass and high pass filter",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
- filter_class->setup =
- GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
- trans_class->transform_ip =
- GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
- trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
}
static void
-gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
- GstAudioChebyshevFreqLimitClass * klass)
+gst_audio_cheb_limit_init (GstAudioChebLimit * filter)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
- gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
- filter->have_coeffs = FALSE;
- filter->num_a = 0;
- filter->num_b = 0;
- filter->channels = NULL;
+ g_mutex_init (&filter->lock);
}
static void
-generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
- gint p, gdouble * a0, gdouble * a1, gdouble * a2,
- gdouble * b1, gdouble * b2)
+generate_biquad_coefficients (GstAudioChebLimit * filter,
+ gint p, gint rate, gdouble * b0, gdouble * b1, gdouble * b2,
+ gdouble * a1, gdouble * a2)
{
gint np = filter->poles;
gdouble ripple = filter->ripple;
gdouble rp, ip;
/* zero location in s-plane */
- gdouble rz = 0.0, iz = 0.0;
+ gdouble iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
/* Calculate pole location for lowpass at frequency 1 */
{
- gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
+ gdouble angle = (G_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
- gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
+ gdouble angle = G_PI / (np * 2.0) + ((p - 1) * G_PI) / (np);
gdouble mag2;
- rz = 0.0;
iz = cos (angle);
- mag2 = rz * rz + iz * iz;
- rz /= mag2;
+ mag2 = iz * iz;
iz /= mag2;
}
*/
{
gdouble k, d;
- gdouble omega =
- 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble omega = 2.0 * G_PI * (filter->cutoff / rate);
if (filter->mode == MODE_LOW_PASS)
k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
d = 1.0 + y1 * k - y2 * k * k;
- *a0 = (x0 + k * (-x1 + k * x2)) / d;
- *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
- *a2 = (x0 * k * k - x1 * k + x2) / d;
- *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
- *b2 = (-k * k - y1 * k + y2) / d;
+ *b0 = (x0 + k * (-x1 + k * x2)) / d;
+ *b1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
+ *b2 = (x0 * k * k - x1 * k + x2) / d;
+ *a1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
+ *a2 = (-k * k - y1 * k + y2) / d;
if (filter->mode == MODE_HIGH_PASS) {
*a1 = -*a1;
}
}
-/* Evaluate the transfer function that corresponds to the IIR
- * coefficients at zr + zi*I and return the magnitude */
-static gdouble
-calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
- gdouble zi)
-{
- gdouble sum_ar, sum_ai;
- gdouble sum_br, sum_bi;
- gdouble gain_r, gain_i;
-
- gdouble sum_r_old;
- gdouble sum_i_old;
-
- gint i;
-
- sum_ar = 0.0;
- sum_ai = 0.0;
- for (i = num_a; i >= 0; i--) {
- sum_r_old = sum_ar;
- sum_i_old = sum_ai;
-
- sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
- sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
- }
-
- sum_br = 0.0;
- sum_bi = 0.0;
- for (i = num_b; i >= 0; i--) {
- sum_r_old = sum_br;
- sum_i_old = sum_bi;
-
- sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
- sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
- }
- sum_br += 1.0;
- sum_bi += 0.0;
-
- gain_r =
- (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
- gain_i =
- (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
-
- return (sqrt (gain_r * gain_r + gain_i * gain_i));
-}
-
static void
-generate_coefficients (GstAudioChebyshevFreqLimit * filter)
+generate_coefficients (GstAudioChebLimit * filter, const GstAudioInfo * info)
{
- gint channels = GST_AUDIO_FILTER (filter)->format.channels;
-
- if (filter->a) {
- g_free (filter->a);
- filter->a = NULL;
- }
+ gint rate;
- if (filter->b) {
- g_free (filter->b);
- filter->b = NULL;
+ if (info) {
+ rate = GST_AUDIO_INFO_RATE (info);
+ } else {
+ rate = GST_AUDIO_FILTER_RATE (filter);
}
- if (filter->channels) {
- GstAudioChebyshevFreqLimitChannelCtx *ctx;
- gint i;
+ GST_LOG_OBJECT (filter, "cutoff %f", filter->cutoff);
- for (i = 0; i < channels; i++) {
- ctx = &filter->channels[i];
- g_free (ctx->x);
- g_free (ctx->y);
- }
+ if (rate == 0) {
+ gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
- g_free (filter->channels);
- filter->channels = NULL;
- }
+ a[0] = 1.0;
+ b[0] = 1.0;
+ gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+ (filter), a, 1, b, 1);
- if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
- filter->num_a = 1;
- filter->a = g_new0 (gdouble, 1);
- filter->a[0] = 1.0;
- filter->num_b = 0;
- filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
- filter->have_coeffs = TRUE;
+ if (filter->cutoff >= rate / 2.0) {
+ gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
- if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
- filter->num_a = 1;
- filter->a = g_new0 (gdouble, 1);
- filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
- filter->num_b = 0;
- filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ a[0] = 1.0;
+ b[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+ gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+ (filter), a, 1, b, 1);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
- filter->num_a = 1;
- filter->a = g_new0 (gdouble, 1);
- filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
- filter->num_b = 0;
- filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+ gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
+
+ a[0] = 1.0;
+ b[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+ gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+ (filter), a, 1, b, 1);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
gdouble *a, *b;
gint i, p;
- filter->num_a = np + 1;
- filter->a = a = g_new0 (gdouble, np + 3);
- filter->num_b = np + 1;
- filter->b = b = g_new0 (gdouble, np + 3);
-
- filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
- for (i = 0; i < channels; i++) {
- GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
-
- ctx->x = g_new0 (gdouble, np + 1);
- ctx->y = g_new0 (gdouble, np + 1);
- }
+ a = g_new0 (gdouble, np + 3);
+ b = g_new0 (gdouble, np + 3);
/* Calculate transfer function coefficients */
a[2] = 1.0;
b[2] = 1.0;
for (p = 1; p <= np / 2; p++) {
- gdouble a0, a1, a2, b1, b2;
+ gdouble b0, b1, b2, a1, a2;
gdouble *ta = g_new0 (gdouble, np + 3);
gdouble *tb = g_new0 (gdouble, np + 3);
- generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
+ generate_biquad_coefficients (filter, p, rate, &b0, &b1, &b2, &a1, &a2);
memcpy (ta, a, sizeof (gdouble) * (np + 3));
memcpy (tb, b, sizeof (gdouble) * (np + 3));
* to the cascade by multiplication of the transfer
* functions */
for (i = 2; i < np + 3; i++) {
- a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
- b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
+ b[i] = b0 * tb[i] + b1 * tb[i - 1] + b2 * tb[i - 2];
+ a[i] = ta[i] - a1 * ta[i - 1] - a2 * ta[i - 2];
}
g_free (ta);
g_free (tb);
}
- /* Move coefficients to the beginning of the array
- * and multiply the b coefficients with -1 to move from
+ /* Move coefficients to the beginning of the array to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
- b[2] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 2];
- b[i] = -b[i + 2];
+ b[i] = b[i + 2];
}
/* Normalize to unity gain at frequency 0 for lowpass
gdouble gain;
if (filter->mode == MODE_LOW_PASS)
- gain = calculate_gain (a, b, np, np, 1.0, 0.0);
+ gain =
+ gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+ 1.0, 0.0);
else
- gain = calculate_gain (a, b, np, np, -1.0, 0.0);
+ gain =
+ gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+ -1.0, 0.0);
for (i = 0; i <= np; i++) {
- a[i] /= gain;
+ b[i] /= gain;
}
}
+ gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+ (filter), a, np + 1, b, np + 1);
+
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
(filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
filter->type, filter->poles, filter->cutoff, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
- 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+ 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
+ np + 1, 1.0, 0.0)));
+
+#ifndef GST_DISABLE_GST_DEBUG
{
- gdouble wc =
- 2.0 * M_PI * (filter->cutoff /
- GST_AUDIO_FILTER (filter)->format.rate);
+ gdouble wc = 2.0 * G_PI * (filter->cutoff / rate);
gdouble zr = cos (wc), zi = sin (wc);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
- 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
- (int) filter->cutoff);
+ 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
+ b, np + 1, zr, zi)), (int) filter->cutoff);
}
+#endif
+
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
- 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
- GST_AUDIO_FILTER (filter)->format.rate / 2);
+ 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
+ np + 1, -1.0, 0.0)), rate);
}
}
static void
-gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
+gst_audio_cheb_limit_finalize (GObject * object)
+{
+ GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
+
+ g_mutex_clear (&filter->lock);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_audio_cheb_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+ GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
- GST_BASE_TRANSFORM_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->mode = g_value_get_enum (value);
- generate_coefficients (filter);
- GST_BASE_TRANSFORM_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_TYPE:
- GST_BASE_TRANSFORM_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->type = g_value_get_int (value);
- generate_coefficients (filter);
- GST_BASE_TRANSFORM_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_CUTOFF:
- GST_BASE_TRANSFORM_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->cutoff = g_value_get_float (value);
- generate_coefficients (filter);
- GST_BASE_TRANSFORM_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_RIPPLE:
- GST_BASE_TRANSFORM_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->ripple = g_value_get_float (value);
- generate_coefficients (filter);
- GST_BASE_TRANSFORM_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_POLES:
- GST_BASE_TRANSFORM_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
- generate_coefficients (filter);
- GST_BASE_TRANSFORM_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
+gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+ GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
/* GstAudioFilter vmethod implementations */
static gboolean
-gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
- GstRingBufferSpec * format)
-{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
- gboolean ret = TRUE;
-
- if (format->width == 32)
- filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
- process_32;
- else if (format->width == 64)
- filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
- process_64;
- else
- ret = FALSE;
-
- filter->have_coeffs = FALSE;
-
- return ret;
-}
-
-static inline gdouble
-process (GstAudioChebyshevFreqLimit * filter,
- GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
-{
- gdouble val = filter->a[0] * x0;
- gint i, j;
-
- for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
- val += filter->a[i] * ctx->x[j];
- j--;
- if (j < 0)
- j = filter->num_a - 1;
- }
-
- for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
- val += filter->b[i] * ctx->y[j];
- j--;
- if (j < 0)
- j = filter->num_b - 1;
- }
-
- if (ctx->x) {
- ctx->x_pos++;
- if (ctx->x_pos > filter->num_a - 1)
- ctx->x_pos = 0;
- ctx->x[ctx->x_pos] = x0;
- }
-
- if (ctx->y) {
- ctx->y_pos++;
- if (ctx->y_pos > filter->num_b - 1)
- ctx->y_pos = 0;
-
- ctx->y[ctx->y_pos] = val;
- }
-
- return val;
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioChebyshevFreqLimit * filter, \
- g##ctype * data, guint num_samples) \
-{ \
- gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
- gdouble val; \
- \
- for (i = 0; i < num_samples / channels; i++) { \
- for (j = 0; j < channels; j++) { \
- val = process (filter, &filter->channels[j], *data); \
- *data++ = val; \
- } \
- } \
-}
-
-DEFINE_PROCESS_FUNC (32, float);
-DEFINE_PROCESS_FUNC (64, double);
-
-#undef DEFINE_PROCESS_FUNC
-
-/* GstBaseTransform vmethod implementations */
-static GstFlowReturn
-gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
- GstBuffer * buf)
+gst_audio_cheb_limit_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
- guint num_samples =
- GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
- if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
- gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
-
- if (gst_base_transform_is_passthrough (base))
- return GST_FLOW_OK;
-
- if (!filter->have_coeffs)
- generate_coefficients (filter);
+ GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
- filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+ generate_coefficients (filter, info);
- return GST_FLOW_OK;
-}
-
-
-static gboolean
-gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
-{
- GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
- gint channels = GST_AUDIO_FILTER (filter)->format.channels;
- GstAudioChebyshevFreqLimitChannelCtx *ctx;
- gint i;
-
- /* Reset the history of input and output values if
- * already existing */
- if (channels && filter->channels) {
- for (i = 0; i < channels; i++) {
- ctx = &filter->channels[i];
- if (ctx->x)
- memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
- if (ctx->y)
- memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
- }
- }
- return TRUE;
+ return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff