*
* 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-audiocheblimit
- * @short_description: Chebyshev low pass and high pass filter
*
- * <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 ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
- * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
- * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit 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 "audiocheblimit.h"
+#include "gst/glib-compat-private.h"
+
#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
PROP_POLES
};
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
-
-GST_BOILERPLATE_FULL (GstAudioChebLimit,
- gst_audio_cheb_limit, GstAudioFXBaseIIRFilter,
- GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
+#define gst_audio_cheb_limit_parent_class parent_class
+G_DEFINE_TYPE (GstAudioChebLimit,
+ gst_audio_cheb_limit, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
static void gst_audio_cheb_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
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_cheb_limit_setup (GstAudioFilter * filter,
- GstRingBufferSpec * format);
+ const GstAudioInfo * info);
enum
{
/* GObject vmethod implementations */
static void
-gst_audio_cheb_limit_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_set_details_simple (element_class,
- "Low pass & high pass filter",
- "Filter/Effect/Audio",
- "Chebyshev low pass and high pass filter",
- "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
-static void
gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+ GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0,
+ "audiocheblimit element");
+
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",
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_cheb_limit_setup);
}
static void
-gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
- GstAudioChebLimitClass * 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;
+
+ g_mutex_init (&filter->lock);
}
static void
generate_biquad_coefficients (GstAudioChebLimit * filter,
- gint p, gdouble * a0, gdouble * a1, gdouble * a2,
- gdouble * b1, gdouble * b2)
+ 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;
}
static void
-generate_coefficients (GstAudioChebLimit * filter)
+generate_coefficients (GstAudioChebLimit * filter, const GstAudioInfo * info)
{
- if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+ gint rate;
+
+ if (info) {
+ rate = GST_AUDIO_INFO_RATE (info);
+ } else {
+ rate = GST_AUDIO_FILTER_RATE (filter);
+ }
+
+ GST_LOG_OBJECT (filter, "cutoff %f", filter->cutoff);
+
+ if (rate == 0) {
gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
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, NULL, 0);
+ (filter), a, 1, b, 1);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
- if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+ if (filter->cutoff >= rate / 2.0) {
gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
- a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+ 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, NULL, 0);
+ (filter), a, 1, b, 1);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
gdouble *a = g_new0 (gdouble, 1);
+ gdouble *b = g_new0 (gdouble, 1);
- a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+ 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, NULL, 0);
+ (filter), a, 1, b, 1);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
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
-1.0, 0.0);
for (i = 0; i <= np; i++) {
- a[i] /= gain;
+ b[i] /= gain;
}
}
#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",
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
- np + 1, -1.0, 0.0)),
- GST_AUDIO_FILTER (filter)->format.rate / 2);
+ np + 1, -1.0, 0.0)), rate);
}
}
static void
+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)
{
switch (prop_id) {
case PROP_MODE:
- GST_OBJECT_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->mode = g_value_get_enum (value);
- generate_coefficients (filter);
- GST_OBJECT_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_TYPE:
- GST_OBJECT_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->type = g_value_get_int (value);
- generate_coefficients (filter);
- GST_OBJECT_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_CUTOFF:
- GST_OBJECT_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->cutoff = g_value_get_float (value);
- generate_coefficients (filter);
- GST_OBJECT_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_RIPPLE:
- GST_OBJECT_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->ripple = g_value_get_float (value);
- generate_coefficients (filter);
- GST_OBJECT_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_POLES:
- GST_OBJECT_LOCK (filter);
+ g_mutex_lock (&filter->lock);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
- generate_coefficients (filter);
- GST_OBJECT_UNLOCK (filter);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
/* GstAudioFilter vmethod implementations */
static gboolean
-gst_audio_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+gst_audio_cheb_limit_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
- generate_coefficients (filter);
+ generate_coefficients (filter, info);
- return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
+ return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
}