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-rw-r--r--ChangeLog18
-rw-r--r--configure.ac2
-rw-r--r--gst-libs/gst/audio/gstaudiofilter.c8
-rw-r--r--gst/equalizer/Makefile.am6
-rw-r--r--gst/equalizer/gstiirequalizer.c440
5 files changed, 471 insertions, 3 deletions
diff --git a/ChangeLog b/ChangeLog
index 58ab3130..64f145b7 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,21 @@
+2004-10-28 Benjamin Otte <otte@gnome.org>
+
+ * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
+ fix link function to always query channels and query width for
+ floats
+ * configure.ac:
+ add equalizer dir
+ * gst/equalizer/Makefile.am:
+ * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
+ (gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
+ (gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
+ (arg_to_scale), (setup_filter),
+ (gst_iir_equalizer_compute_frequencies),
+ (gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
+ (gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
+ (plugin_init):
+ add an equalizer
+
2004-10-27 Thomas Vander Stichele <thomas at apestaart dot org>
Submitted by: Kjartan Maraas <kmaraas@broadpark.no>
diff --git a/configure.ac b/configure.ac
index f5bf7eb3..11b17ff0 100644
--- a/configure.ac
+++ b/configure.ac
@@ -342,6 +342,7 @@ GST_PLUGINS_ALL="\
debug \
deinterlace \
effectv \
+ equalizer \
festival \
ffmpegcolorspace \
filter \
@@ -1832,6 +1833,7 @@ gst/cutter/Makefile
gst/debug/Makefile
gst/deinterlace/Makefile
gst/effectv/Makefile
+gst/equalizer/Makefile
gst/festival/Makefile
gst/ffmpegcolorspace/Makefile
gst/filter/Makefile
diff --git a/gst-libs/gst/audio/gstaudiofilter.c b/gst-libs/gst/audio/gstaudiofilter.c
index 0666862c..da0064d3 100644
--- a/gst-libs/gst/audio/gstaudiofilter.c
+++ b/gst-libs/gst/audio/gstaudiofilter.c
@@ -141,15 +141,17 @@ gst_audiofilter_link (GstPad * pad, const GstCaps * caps)
if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
ret = gst_structure_get_int (structure, "depth", &audiofilter->depth);
ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
- ret &=
- gst_structure_get_int (structure, "channels", &audiofilter->channels);
} else if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float")
== 0) {
-
+ ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
} else {
g_assert_not_reached ();
}
ret &= gst_structure_get_int (structure, "rate", &audiofilter->rate);
+ ret &= gst_structure_get_int (structure, "channels", &audiofilter->channels);
+
+ if (!ret)
+ return GST_PAD_LINK_REFUSED;
audiofilter->bytes_per_sample = (audiofilter->width / 8) *
audiofilter->channels;
diff --git a/gst/equalizer/Makefile.am b/gst/equalizer/Makefile.am
new file mode 100644
index 00000000..d85bc5a4
--- /dev/null
+++ b/gst/equalizer/Makefile.am
@@ -0,0 +1,6 @@
+plugin_LTLIBRARIES = libgstequalizer.la
+
+libgstequalizer_la_SOURCES = gstiirequalizer.c
+libgstequalizer_la_CFLAGS = $(GST_CFLAGS)
+libgstequalizer_la_LIBADD =
+libgstequalizer_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -lm
diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c
new file mode 100644
index 00000000..c7819792
--- /dev/null
+++ b/gst/equalizer/gstiirequalizer.c
@@ -0,0 +1,440 @@
+/* GStreamer
+ * Copyright (C) <2004> Benjamin Otte <otte@gnome.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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <math.h>
+#include <string.h>
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+typedef struct _GstIirEqualizer GstIirEqualizer;
+typedef struct _GstIirEqualizerClass GstIirEqualizerClass;
+
+#define GST_TYPE_IIR_EQUALIZER \
+ (gst_iir_equalizer_get_type())
+#define GST_IIR_EQUALIZER(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_IIR_EQUALIZER,GstIirEqualizer))
+#define GST_IIR_EQUALIZER_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_IIR_EQUALIZER,GstIirEqualizerClass))
+#define GST_IS_IIR_EQUALIZER(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_IIR_EQUALIZER))
+#define GST_IS_IIR_EQUALIZER_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_IIR_EQUALIZER))
+
+#define LOWEST_FREQ (20.0)
+#define HIGHEST_FREQ (20000.0)
+
+typedef void (*ProcessFunc) (GstIirEqualizer * equ, guint8 * data, guint size,
+ guint channels);
+
+typedef struct
+{
+ gdouble alpha; /* IIR coefficients for outputs */
+ gdouble beta; /* IIR coefficients for inputs */
+ gdouble gamma; /* IIR coefficients for inputs */
+} SecondOrderFilter;
+
+struct _GstIirEqualizer
+{
+ GstAudiofilter audiofilter;
+
+ /* properties */
+ guint freq_count;
+ gdouble bandwidth;
+ gdouble *freqs;
+ gdouble *values;
+
+ /* data */
+ SecondOrderFilter *filter;
+ gpointer history;
+ ProcessFunc process;
+ guint history_size;
+};
+
+struct _GstIirEqualizerClass
+{
+ GstAudiofilterClass audiofilter_class;
+};
+
+enum
+{
+ ARG_0,
+ ARG_BANDS,
+ ARG_BANDWIDTH,
+ ARG_VALUES
+ /* FILL ME */
+};
+
+static void gst_iir_equalizer_base_init (gpointer g_class);
+static void gst_iir_equalizer_class_init (gpointer g_class,
+ gpointer class_data);
+static void gst_iir_equalizer_init (GTypeInstance * instance, gpointer g_class);
+static void gst_iir_equalizer_finalize (GObject * object);
+
+static void gst_iir_equalizer_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_iir_equalizer_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static void gst_iir_equalizer_setup (GstAudiofilter * iir_equalizer);
+static void gst_iir_equalizer_filter_inplace (GstAudiofilter *
+ iir_equalizer, GstBuffer * buf);
+
+static GstAudiofilterClass *parent_class;
+
+GType
+gst_iir_equalizer_get_type (void)
+{
+ static GType iir_equalizer_type = 0;
+
+ if (!iir_equalizer_type) {
+ static const GTypeInfo iir_equalizer_info = {
+ sizeof (GstIirEqualizerClass),
+ gst_iir_equalizer_base_init,
+ NULL,
+ gst_iir_equalizer_class_init,
+ NULL,
+ gst_iir_equalizer_init,
+ sizeof (GstIirEqualizer),
+ 0,
+ NULL,
+ };
+
+ iir_equalizer_type = g_type_register_static (GST_TYPE_AUDIOFILTER,
+ "GstIirEqualizer", &iir_equalizer_info, 0);
+ }
+ return iir_equalizer_type;
+}
+
+static void
+gst_iir_equalizer_base_init (gpointer g_class)
+{
+ static GstElementDetails iir_equalizer_details = {
+ "Equalizer",
+ "Filter/Effect/Audio",
+ "Direct Form IIR equalizer",
+ "Benjamin Otte <otte@gnome.org>"
+ };
+ GstIirEqualizerClass *klass = (GstIirEqualizerClass *) g_class;
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &iir_equalizer_details);
+
+ caps = gst_caps_from_string ("audio/x-raw-int, depth=(int)16, width=(int)16, "
+ "endianness=(int)BYTE_ORDER, signed=(bool)TRUE, "
+ "rate=(int)[1000,MAX], channels=(int)[1,6];"
+ "audio/x-raw-float, width=(int)32, endianness=(int)BYTE_ORDER,"
+ "rate=(int)[1000,MAX], channels=(int)[1,6]");
+ gst_audiofilter_class_add_pad_templates (GST_AUDIOFILTER_CLASS (g_class),
+ caps);
+ gst_caps_free (caps);
+}
+
+static void
+gst_iir_equalizer_class_init (gpointer g_class, gpointer class_data)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstIirEqualizerClass *klass;
+ GstAudiofilterClass *audiofilter_class;
+
+ klass = (GstIirEqualizerClass *) g_class;
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ audiofilter_class = (GstAudiofilterClass *) g_class;
+
+ gobject_class->set_property = gst_iir_equalizer_set_property;
+ gobject_class->get_property = gst_iir_equalizer_get_property;
+ gobject_class->finalize = gst_iir_equalizer_finalize;
+
+ parent_class = g_type_class_peek_parent (g_class);
+
+ g_object_class_install_property (gobject_class, ARG_BANDS,
+ g_param_spec_uint ("bands", "bands", "number of different bands to use",
+ 2, 64, 15, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+ g_object_class_install_property (gobject_class, ARG_BANDWIDTH,
+ g_param_spec_double ("bandwidth", "bandwidth",
+ "bandwidth calculated as distance between bands * this value", 0.1,
+ 5.0, 1.0, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+ /* FIXME FIXME FIXME */
+ g_object_class_install_property (gobject_class, ARG_VALUES,
+ g_param_spec_pointer ("values", "values",
+ "expects a gdouble* of values to use for the bands",
+ G_PARAM_WRITABLE));
+
+ audiofilter_class->setup = gst_iir_equalizer_setup;
+ audiofilter_class->filter_inplace = gst_iir_equalizer_filter_inplace;
+}
+
+static void
+gst_iir_equalizer_init (GTypeInstance * instance, gpointer g_class)
+{
+}
+
+static void
+gst_iir_equalizer_finalize (GObject * object)
+{
+ GstIirEqualizer *equ = GST_IIR_EQUALIZER (object);
+
+ g_free (equ->freqs);
+ g_free (equ->values);
+ g_free (equ->filter);
+ g_free (equ->history);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* args are in the range [-1 ... 1] with 0 meaning "no action"
+ * convert to [-0.2 ... 1] with 0 meaning no action via the function
+ * f(x) = 0.25 * 5 ^ x - 0.25
+ */
+static gdouble
+arg_to_scale (gdouble arg)
+{
+ return 0.25 * exp (log (5) * arg) - 0.25;
+}
+
+static void
+setup_filter (GstIirEqualizer * equ, SecondOrderFilter * filter, gdouble gain,
+ gdouble frequency)
+{
+ gdouble q =
+ pow (HIGHEST_FREQ / LOWEST_FREQ,
+ 1.0 / (equ->freq_count - 1)) * equ->bandwidth;
+ gdouble theta = frequency * 2 * M_PI;
+
+ filter->beta = (q - theta / 2) / (2 * q + theta);
+ filter->gamma = (0.5 + filter->beta) * cos (theta);
+ filter->alpha = (0.5 - filter->beta) / 2;
+
+ filter->beta *= 2.0;
+ filter->alpha *= 2.0 * gain;
+ filter->gamma *= 2.0;
+ GST_INFO ("gain = %g, frequency = %g, alpha = %g, beta = %g, gamma=%g\n",
+ gain, frequency, filter->alpha, filter->beta, filter->gamma);
+}
+
+static void
+gst_iir_equalizer_compute_frequencies (GstIirEqualizer * equ, guint band_count)
+{
+ gdouble *old_values;
+ guint old_count, i;
+ gdouble step = pow (HIGHEST_FREQ / LOWEST_FREQ, 1.0 / (band_count - 1));
+ GstAudiofilter *audio = GST_AUDIOFILTER (equ);
+
+ old_count = equ->freq_count;
+ equ->freq_count = band_count;
+ old_values = equ->values;
+ if (old_count < band_count) {
+ equ->freqs = g_realloc (equ->freqs, sizeof (gdouble) * band_count);
+ memset (equ->freqs + sizeof (gdouble) * old_count, 0,
+ sizeof (gdouble) * (band_count - old_count));
+ equ->values = g_realloc (equ->values, sizeof (gdouble) * band_count);
+ memset (equ->values + sizeof (gdouble) * old_count, 0,
+ sizeof (gdouble) * (band_count - old_count));
+ equ->filter =
+ g_realloc (equ->filter, sizeof (SecondOrderFilter) * band_count);
+ memset (equ->filter + sizeof (SecondOrderFilter) * old_count, 0,
+ sizeof (SecondOrderFilter) * (band_count - old_count));
+ }
+ equ->history =
+ g_realloc (equ->history,
+ equ->history_size * audio->channels * band_count);
+ memset (equ->history, 0, equ->history_size * audio->channels * band_count);
+ equ->freqs[0] = LOWEST_FREQ;
+ for (i = 1; i < band_count; i++) {
+ equ->freqs[i] = equ->freqs[i - 1] * step;
+ }
+
+ if (audio->rate) {
+ guint i;
+
+ for (i = 0; i < band_count; i++) {
+ setup_filter (equ, &equ->filter[i], arg_to_scale (equ->values[i]),
+ equ->freqs[i] / audio->rate);
+ }
+ }
+}
+
+static void
+gst_iir_equalizer_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstIirEqualizer *equ = GST_IIR_EQUALIZER (object);
+
+ switch (prop_id) {
+ case ARG_BANDS:
+ gst_iir_equalizer_compute_frequencies (equ, g_value_get_uint (value));
+ break;
+ case ARG_BANDWIDTH:
+ if (g_value_get_double (value) != equ->bandwidth) {
+ equ->bandwidth = g_value_get_double (value);
+ if (GST_AUDIOFILTER (equ)->rate) {
+ guint i;
+
+ for (i = 0; i < equ->freq_count; i++) {
+ setup_filter (equ, &equ->filter[i], arg_to_scale (equ->values[i]),
+ equ->freqs[i] / GST_AUDIOFILTER (equ)->rate);
+ }
+ }
+ }
+ break;
+ case ARG_VALUES:
+ {
+ gdouble *new = g_value_get_pointer (value);
+ guint i;
+
+ for (i = 0; i < equ->freq_count; i++) {
+ if (new[i] != equ->values[i]) {
+ equ->values[i] = new[i];
+ setup_filter (equ, &equ->filter[i], arg_to_scale (new[i]),
+ equ->freqs[i] / GST_AUDIOFILTER (equ)->rate);
+ }
+ }
+ }
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_iir_equalizer_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstIirEqualizer *equ = GST_IIR_EQUALIZER (object);
+
+ switch (prop_id) {
+ case ARG_BANDS:
+ g_value_set_uint (value, equ->freq_count);
+ break;
+ case ARG_BANDWIDTH:
+ g_value_set_double (value, equ->bandwidth);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* start of code that is type specific */
+
+#define CREATE_OPTIMIZED_FUNCTIONS(TYPE,BIG_TYPE,MIN_VAL,MAX_VAL) \
+typedef struct { \
+ TYPE x1, x2; /* history of input values for a filter */ \
+ TYPE y1, y2; /* history of output values for a filter */ \
+} SecondOrderHistory ## TYPE; \
+ \
+static inline TYPE \
+one_step_ ## TYPE (SecondOrderFilter *filter, \
+ SecondOrderHistory ## TYPE *history, TYPE input) \
+{ \
+ /* calculate output */ \
+ TYPE output = filter->alpha * (input - history->x2) + \
+ filter->gamma * history->y1 - filter->beta * history->y2; \
+ /* update history */ \
+ history->y2 = history->y1; \
+ history->y1 = output; \
+ history->x2 = history->x1; \
+ history->x1 = input; \
+ \
+ return output; \
+} \
+ \
+static const guint history_size_ ## TYPE = sizeof (SecondOrderHistory ## TYPE); \
+ \
+static void \
+gst_iir_equ_process_ ## TYPE (GstIirEqualizer *equ, guint8 *data, guint size, \
+ guint channels) \
+{ \
+ guint frames = size / channels / sizeof (TYPE); \
+ guint i, c, f; \
+ BIG_TYPE cur; \
+ TYPE val; \
+ \
+ for (i = 0; i < frames; i++) { \
+ for (c = 0; c < channels; c++) { \
+ SecondOrderHistory ## TYPE *history = equ->history; \
+ val = *((TYPE *) data); \
+ cur = 0; \
+ for (f = 0; f < equ->freq_count; f++) { \
+ SecondOrderFilter *filter = &equ->filter[f]; \
+ \
+ cur += one_step_ ## TYPE (filter, history, val); \
+ history++; \
+ } \
+ cur += val * 0.25; \
+ cur = CLAMP (cur, MIN_VAL, MAX_VAL); \
+ *((TYPE *) data) = (TYPE) cur; \
+ data += sizeof (TYPE); \
+ } \
+ } \
+}
+
+CREATE_OPTIMIZED_FUNCTIONS (gint16, gint, -32768, 32767)
+ CREATE_OPTIMIZED_FUNCTIONS (gfloat, gfloat, -1.0, 1.0)
+
+ static void
+ gst_iir_equalizer_filter_inplace (GstAudiofilter * filter,
+ GstBuffer * buf)
+{
+ GstIirEqualizer *equ = GST_IIR_EQUALIZER (filter);
+
+ equ->process (equ, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+ filter->channels);
+}
+
+static void
+gst_iir_equalizer_setup (GstAudiofilter * audio)
+{
+ GstIirEqualizer *equ = GST_IIR_EQUALIZER (audio);
+
+ if (audio->width == 16) {
+ equ->history_size = history_size_gint16;
+ equ->process = gst_iir_equ_process_gint16;
+ } else if (audio->width == 32) {
+ equ->history_size = history_size_gfloat;
+ equ->process = gst_iir_equ_process_gfloat;
+ } else {
+ g_assert_not_reached ();
+ }
+ gst_iir_equalizer_compute_frequencies (equ, equ->freq_count);
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ if (!gst_library_load ("gstaudiofilter"))
+ return FALSE;
+
+ return gst_element_register (plugin, "equalizer", GST_RANK_NONE,
+ GST_TYPE_IIR_EQUALIZER);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "gstequalizer",
+ "GStreamer equalizers",
+ plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
generated by cgit v1.2.1 (git 2.18.0) at 2024-11-01 09:00:14 +0000