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gstreamer/gst/audiofx/audiodynamic.c

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/*
* GStreamer
* Copyright (C) 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.
*/
/**
* SECTION:element-audiodynamic
*
* This element can act as a compressor or expander. A compressor changes the
* amplitude of all samples above a specific threshold with a specific ratio,
* a expander does the same for all samples below a specific threshold. If
* soft-knee mode is selected the ratio is applied smoothly.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
* ]|
* </refsect2>
*/
/* TODO: Implement attack and release parameters */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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 "audiodynamic.h"
#define GST_CAT_DEFAULT gst_audio_dynamic_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_CHARACTERISTICS,
PROP_MODE,
PROP_THRESHOLD,
PROP_RATIO
};
#define ALLOWED_CAPS \
"audio/x-raw-int," \
" depth=(int)16," \
" width=(int)16," \
" endianness=(int)BYTE_ORDER," \
" signed=(bool)TRUE," \
" rate=(int)[1,MAX]," \
" channels=(int)[1,MAX]; " \
"audio/x-raw-float," \
" width=(int)32," \
" endianness=(int)BYTE_ORDER," \
" rate=(int)[1,MAX]," \
" channels=(int)[1,MAX]"
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0, "audiodynamic element");
GST_BOILERPLATE_FULL (GstAudioDynamic, gst_audio_dynamic, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void
gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
filter, gfloat * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
filter, gfloat * data, guint num_samples);
static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
* filter, gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
gfloat * data, guint num_samples);
static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
* filter, gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
gfloat * data, guint num_samples);
static GstAudioDynamicProcessFunc process_functions[] = {
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_hard_knee_compressor_int,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_hard_knee_compressor_float,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_soft_knee_compressor_int,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_soft_knee_compressor_float,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_hard_knee_expander_int,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_hard_knee_expander_float,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_soft_knee_expander_int,
(GstAudioDynamicProcessFunc)
gst_audio_dynamic_transform_soft_knee_expander_float
};
enum
{
CHARACTERISTICS_HARD_KNEE = 0,
CHARACTERISTICS_SOFT_KNEE
};
#define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
static GType
gst_audio_dynamic_characteristics_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
"hard-knee"},
{CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
"soft-knee"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
}
return gtype;
}
enum
{
MODE_COMPRESSOR = 0,
MODE_EXPANDER
};
#define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
static GType
gst_audio_dynamic_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_COMPRESSOR, "Compressor (default)",
"compressor"},
{MODE_EXPANDER, "Expander", "expander"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioDynamicMode", values);
}
return gtype;
}
static gboolean
gst_audio_dynamic_set_process_function (GstAudioDynamic * filter)
{
gint func_index;
func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
func_index +=
(GST_AUDIO_FILTER (filter)->format.type == GST_BUFTYPE_FLOAT) ? 1 : 0;
if (func_index >= 0 && func_index < 8) {
filter->process = process_functions[func_index];
return TRUE;
}
return FALSE;
}
/* GObject vmethod implementations */
static void
gst_audio_dynamic_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details_simple (element_class,
"Dynamic range controller", "Filter/Effect/Audio",
"Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
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_dynamic_class_init (GstAudioDynamicClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audio_dynamic_set_property;
gobject_class->get_property = gst_audio_dynamic_get_property;
g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
g_param_spec_enum ("characteristics", "Characteristics",
"Selects whether the ratio should be applied smooth (soft-knee) "
"or hard (hard-knee).",
GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Selects whether the filter should work on loud samples (compressor) or"
"quiet samples (expander).",
GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_THRESHOLD,
g_param_spec_float ("threshold", "Threshold",
"Threshold until the filter is activated", 0.0, 1.0,
0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RATIO,
g_param_spec_float ("ratio", "Ratio",
"Ratio that should be applied", 0.0, G_MAXFLOAT,
1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
}
static void
gst_audio_dynamic_init (GstAudioDynamic * filter, GstAudioDynamicClass * klass)
{
filter->ratio = 1.0;
filter->threshold = 0.0;
filter->characteristics = CHARACTERISTICS_HARD_KNEE;
filter->mode = MODE_COMPRESSOR;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
}
static void
gst_audio_dynamic_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
switch (prop_id) {
case PROP_CHARACTERISTICS:
filter->characteristics = g_value_get_enum (value);
gst_audio_dynamic_set_process_function (filter);
break;
case PROP_MODE:
filter->mode = g_value_get_enum (value);
gst_audio_dynamic_set_process_function (filter);
break;
case PROP_THRESHOLD:
filter->threshold = g_value_get_float (value);
break;
case PROP_RATIO:
filter->ratio = g_value_get_float (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_dynamic_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
switch (prop_id) {
case PROP_CHARACTERISTICS:
g_value_set_enum (value, filter->characteristics);
break;
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_THRESHOLD:
g_value_set_float (value, filter->threshold);
break;
case PROP_RATIO:
g_value_set_float (value, filter->ratio);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_dynamic_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
gboolean ret = TRUE;
ret = gst_audio_dynamic_set_process_function (filter);
return ret;
}
static void
gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples)
{
glong val;
glong thr_p = filter->threshold * G_MAXINT16;
glong thr_n = filter->threshold * G_MININT16;
/* Nothing to do for us if ratio is 1.0 or if the threshold
* equals 1.0. */
if (filter->threshold == 1.0 || filter->ratio == 1.0)
return;
for (; num_samples; num_samples--) {
val = *data;
if (val > thr_p) {
val = thr_p + (val - thr_p) * filter->ratio;
} else if (val < thr_n) {
val = thr_n + (val - thr_n) * filter->ratio;
}
*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
filter, gfloat * data, guint num_samples)
{
gdouble val, threshold = filter->threshold;
/* Nothing to do for us if ratio == 1.0.
* As float values can be above 1.0 we have to do something
* if threshold is greater than 1.0. */
if (filter->ratio == 1.0)
return;
for (; num_samples; num_samples--) {
val = *data;
if (val > threshold) {
val = threshold + (val - threshold) * filter->ratio;
} else if (val < -threshold) {
val = -threshold + (val + threshold) * filter->ratio;
}
*data++ = (gfloat) val;
}
}
static void
gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples)
{
glong val;
glong thr_p = filter->threshold * G_MAXINT16;
glong thr_n = filter->threshold * G_MININT16;
gdouble a_p, b_p, c_p;
gdouble a_n, b_n, c_n;
/* Nothing to do for us if ratio is 1.0 or if the threshold
* equals 1.0. */
if (filter->threshold == 1.0 || filter->ratio == 1.0)
return;
/* We build a 2nd degree polynomial here for
* values greater than threshold or small than
* -threshold with:
* f(t) = t, f'(t) = 1, f'(m) = r
* =>
* a = (1-r)/(2*(t-m))
* b = (r*t - m)/(t-m)
* c = t * (1 - b - a*t)
* f(x) = ax^2 + bx + c
*/
/* shouldn't happen because this would only be the case
* for threshold == 1.0 which we catch above */
g_assert (thr_p - G_MAXINT16 != 0);
g_assert (thr_n - G_MININT != 0);
a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
c_p = thr_p * (1 - b_p - a_p * thr_p);
a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
c_n = thr_n * (1 - b_n - a_n * thr_n);
for (; num_samples; num_samples--) {
val = *data;
if (val > thr_p) {
val = a_p * val * val + b_p * val + c_p;
} else if (val < thr_n) {
val = a_n * val * val + b_n * val + c_n;
}
*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
filter, gfloat * data, guint num_samples)
{
gdouble val;
gdouble threshold = filter->threshold;
gdouble a_p, b_p, c_p;
gdouble a_n, b_n, c_n;
/* Nothing to do for us if ratio == 1.0.
* As float values can be above 1.0 we have to do something
* if threshold is greater than 1.0. */
if (filter->ratio == 1.0)
return;
/* We build a 2nd degree polynomial here for
* values greater than threshold or small than
* -threshold with:
* f(t) = t, f'(t) = 1, f'(m) = r
* =>
* a = (1-r)/(2*(t-m))
* b = (r*t - m)/(t-m)
* c = t * (1 - b - a*t)
* f(x) = ax^2 + bx + c
*/
/* FIXME: If treshold is the same as the maximum
* we need to raise it a bit to prevent
* division by zero. */
if (threshold == 1.0)
threshold = 1.0 + 0.00001;
a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);
c_p = threshold * (1.0 - b_p - a_p * threshold);
a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));
b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);
c_n = -threshold * (1.0 - b_n + a_n * threshold);
for (; num_samples; num_samples--) {
val = *data;
if (val > 1.0) {
val = 1.0 + (val - 1.0) * filter->ratio;
} else if (val > threshold) {
val = a_p * val * val + b_p * val + c_p;
} else if (val < -1.0) {
val = -1.0 + (val + 1.0) * filter->ratio;
} else if (val < -threshold) {
val = a_n * val * val + b_n * val + c_n;
}
*data++ = (gfloat) val;
}
}
static void
gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples)
{
glong val;
glong thr_p = filter->threshold * G_MAXINT16;
glong thr_n = filter->threshold * G_MININT16;
gdouble zero_p, zero_n;
/* Nothing to do for us here if threshold equals 0.0
* or ratio equals 1.0 */
if (filter->threshold == 0.0 || filter->ratio == 1.0)
return;
/* zero crossing of our function */
if (filter->ratio != 0.0) {
zero_p = thr_p - thr_p / filter->ratio;
zero_n = thr_n - thr_n / filter->ratio;
} else {
zero_p = zero_n = 0.0;
}
if (zero_p < 0.0)
zero_p = 0.0;
if (zero_n > 0.0)
zero_n = 0.0;
for (; num_samples; num_samples--) {
val = *data;
if (val < thr_p && val > zero_p) {
val = filter->ratio * val + thr_p * (1 - filter->ratio);
} else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
val = 0;
} else if (val > thr_n && val < zero_n) {
val = filter->ratio * val + thr_n * (1 - filter->ratio);
}
*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
gfloat * data, guint num_samples)
{
gdouble val, threshold = filter->threshold, zero;
/* Nothing to do for us here if threshold equals 0.0
* or ratio equals 1.0 */
if (filter->threshold == 0.0 || filter->ratio == 1.0)
return;
/* zero crossing of our function */
if (filter->ratio != 0.0)
zero = threshold - threshold / filter->ratio;
else
zero = 0.0;
if (zero < 0.0)
zero = 0.0;
for (; num_samples; num_samples--) {
val = *data;
if (val < threshold && val > zero) {
val = filter->ratio * val + threshold * (1.0 - filter->ratio);
} else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
val = 0.0;
} else if (val > -threshold && val < -zero) {
val = filter->ratio * val - threshold * (1.0 - filter->ratio);
}
*data++ = (gfloat) val;
}
}
static void
gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,
gint16 * data, guint num_samples)
{
glong val;
glong thr_p = filter->threshold * G_MAXINT16;
glong thr_n = filter->threshold * G_MININT16;
gdouble zero_p, zero_n;
gdouble a_p, b_p, c_p;
gdouble a_n, b_n, c_n;
/* Nothing to do for us here if threshold equals 0.0
* or ratio equals 1.0 */
if (filter->threshold == 0.0 || filter->ratio == 1.0)
return;
/* zero crossing of our function */
zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
if (zero_p < 0.0)
zero_p = 0.0;
if (zero_n > 0.0)
zero_n = 0.0;
/* shouldn't happen as this would only happen
* with threshold == 0.0 */
g_assert (thr_p != 0);
g_assert (thr_n != 0);
/* We build a 2n degree polynomial here for values between
* 0 and threshold or 0 and -threshold with:
* f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
* z between 0 and t
* =>
* a = (1 - r^2) / (4 * t)
* b = (1 + r^2) / 2
* c = t * (1.0 - b - a*t)
* f(x) = ax^2 + bx + c */
a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_p);
b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
c_p = thr_p * (1.0 - b_p - a_p * thr_p);
a_n = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_n);
b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;
c_n = thr_n * (1.0 - b_n - a_n * thr_n);
for (; num_samples; num_samples--) {
val = *data;
if (val < thr_p && val > zero_p) {
val = a_p * val * val + b_p * val + c_p;
} else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
val = 0;
} else if (val > thr_n && val < zero_n) {
val = a_n * val * val + b_n * val + c_n;
}
*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
gfloat * data, guint num_samples)
{
gdouble val;
gdouble threshold = filter->threshold;
gdouble zero;
gdouble a_p, b_p, c_p;
gdouble a_n, b_n, c_n;
/* Nothing to do for us here if threshold equals 0.0
* or ratio equals 1.0 */
if (filter->threshold == 0.0 || filter->ratio == 1.0)
return;
/* zero crossing of our function */
zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
if (zero < 0.0)
zero = 0.0;
/* shouldn't happen as this only happens with
* threshold == 0.0 */
g_assert (threshold != 0.0);
/* We build a 2n degree polynomial here for values between
* 0 and threshold or 0 and -threshold with:
* f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
* z between 0 and t
* =>
* a = (1 - r^2) / (4 * t)
* b = (1 + r^2) / 2
* c = t * (1.0 - b - a*t)
* f(x) = ax^2 + bx + c */
a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * threshold);
b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
c_p = threshold * (1.0 - b_p - a_p * threshold);
a_n = (1.0 - filter->ratio * filter->ratio) / (-4.0 * threshold);
b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;
c_n = -threshold * (1.0 - b_n + a_n * threshold);
for (; num_samples; num_samples--) {
val = *data;
if (val < threshold && val > zero) {
val = a_p * val * val + b_p * val + c_p;
} else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
val = 0.0;
} else if (val > -threshold && val < -zero) {
val = a_n * val * val + b_n * val + c_n;
}
*data++ = (gfloat) val;
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
guint num_samples;
GstClockTime timestamp, stream_time;
timestamp = GST_BUFFER_TIMESTAMP (buf);
stream_time =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (G_OBJECT (filter), stream_time);
num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
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