*
* 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.
*/
/*
* <refsect2>
* <title>Example launch line</title>
* |[
- * 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
+ * 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>
*/
2, 32, 4,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_set_static_metadata (gstelement_class,
"Low pass & high pass filter",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
filter->poles = 4;
filter->ripple = 0.25;
- filter->lock = g_mutex_new ();
+ 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 k, d;
- gdouble omega =
- 2.0 * G_PI * (filter->cutoff / GST_AUDIO_FILTER_RATE (filter));
+ 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_RATE (filter) == 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_RATE (filter) / 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 * G_PI * (filter->cutoff / GST_AUDIO_FILTER_RATE (filter));
+ 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_RATE (filter) / 2);
+ np + 1, -1.0, 0.0)), rate);
}
}
{
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
- g_mutex_free (filter->lock);
- filter->lock = NULL;
+ g_mutex_clear (&filter->lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
switch (prop_id) {
case PROP_MODE:
- g_mutex_lock (filter->lock);
+ g_mutex_lock (&filter->lock);
filter->mode = g_value_get_enum (value);
- generate_coefficients (filter);
- g_mutex_unlock (filter->lock);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_TYPE:
- g_mutex_lock (filter->lock);
+ g_mutex_lock (&filter->lock);
filter->type = g_value_get_int (value);
- generate_coefficients (filter);
- g_mutex_unlock (filter->lock);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_CUTOFF:
- g_mutex_lock (filter->lock);
+ g_mutex_lock (&filter->lock);
filter->cutoff = g_value_get_float (value);
- generate_coefficients (filter);
- g_mutex_unlock (filter->lock);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_RIPPLE:
- g_mutex_lock (filter->lock);
+ g_mutex_lock (&filter->lock);
filter->ripple = g_value_get_float (value);
- generate_coefficients (filter);
- g_mutex_unlock (filter->lock);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
case PROP_POLES:
- g_mutex_lock (filter->lock);
+ g_mutex_lock (&filter->lock);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
- generate_coefficients (filter);
- g_mutex_unlock (filter->lock);
+ generate_coefficients (filter, NULL);
+ g_mutex_unlock (&filter->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
{
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
- generate_coefficients (filter);
+ generate_coefficients (filter, info);
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
}