gstreamer/gst/audiofx/audioamplify.c
Sebastian Dröge b76819bbd2 Make elements GST_BUFFER_FLAG_GAP aware and call gst_base_transform_set_gap_aware for this.
Original commit message from CVS:
* configure.ac:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_init), (gst_audio_amplify_transform_ip):
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_init),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
(gst_audio_invert_transform_ip):
* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
(gst_audio_panorama_transform):
* gst/level/gstlevel.c: (gst_level_init):
Make elements GST_BUFFER_FLAG_GAP aware and call
gst_base_transform_set_gap_aware for this.
Bump core requirement to CVS.
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_transform_ip):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_transform_ip):
Also sync GObject properties to the controller if operating
in passthrough mode.
2008-01-08 14:58:18 +00:00

429 lines
13 KiB
C

/*
* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
* Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
*
* 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-audioamplify
* @short_description: Amplifies an audio stream with selectable clipping mode
*
* <refsect2>
* Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
* The difference between the clipping modes is best evaluated by testing.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink
* gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#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 "audioamplify.h"
#define GST_CAT_DEFAULT gst_audio_amplify_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioAmplify",
"Filter/Effect/Audio",
"Amplifies an audio stream by a given factor",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_AMPLIFICATION,
PROP_CLIPPING_METHOD
};
enum
{
METHOD_CLIP = 0,
METHOD_WRAP_NEGATIVE,
METHOD_WRAP_POSITIVE,
NUM_METHODS
};
#define GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD (gst_audio_amplify_clipping_method_get_type ())
static GType
gst_audio_amplify_clipping_method_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{METHOD_CLIP, "Normal Clipping (default)", "clip"},
{METHOD_WRAP_NEGATIVE,
"Push overdriven values back from the opposite side",
"wrap-negative"},
{METHOD_WRAP_POSITIVE, "Push overdriven values back from the same side",
"wrap-positive"},
{0, NULL, NULL}
};
/* FIXME 0.11: rename to GstAudioAmplifyClippingMethod */
gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values);
}
return gtype;
}
#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_amplify_debug, "audioamplify", 0, "audioamplify element");
GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_amplify_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_amplify_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_amplify_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn gst_audio_amplify_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter,
gint16 * data, guint num_samples);
static void gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify *
filter, gint16 * data, guint num_samples);
static void gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify *
filter, gint16 * data, guint num_samples);
static void gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter,
gfloat * data, guint num_samples);
static void gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify *
filter, gfloat * data, guint num_samples);
static void gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify *
filter, gfloat * data, guint num_samples);
/* table of processing functions: [format][clipping_method] */
static GstAudioAmplifyProcessFunc processing_functions[2][3] = {
{
(GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_int_clip,
(GstAudioAmplifyProcessFunc)
gst_audio_amplify_transform_int_wrap_negative,
(GstAudioAmplifyProcessFunc)
gst_audio_amplify_transform_int_wrap_positive},
{
(GstAudioAmplifyProcessFunc) gst_audio_amplify_transform_float_clip,
(GstAudioAmplifyProcessFunc)
gst_audio_amplify_transform_float_wrap_negative,
(GstAudioAmplifyProcessFunc)
gst_audio_amplify_transform_float_wrap_positive}
};
/* GObject vmethod implementations */
static void
gst_audio_amplify_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_amplify_class_init (GstAudioAmplifyClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audio_amplify_set_property;
gobject_class->get_property = gst_audio_amplify_get_property;
g_object_class_install_property (gobject_class, PROP_AMPLIFICATION,
g_param_spec_float ("amplification", "Amplification",
"Factor of amplification", 0.0, G_MAXFLOAT,
1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/**
* GstAudioAmplify:clipping-method
*
* Clipping method: clip mode set values higher than the maximum to the
* maximum. The wrap-negative mode pushes those values back from the
* opposite side, wrap-positive pushes them back from the same side.
*
**/
g_object_class_install_property (gobject_class, PROP_CLIPPING_METHOD,
g_param_spec_enum ("clipping-method", "Clipping method",
"Selects how to handle values higher than the maximum",
GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP,
G_PARAM_READWRITE));
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_amplify_setup);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_amplify_transform_ip);
}
static void
gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass)
{
filter->amplification = 1.0;
filter->clipping_method = METHOD_CLIP;
filter->format_index = 0;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
}
static gboolean
gst_audio_amplify_set_process_function (GstAudioAmplify * filter)
{
gint method_index;
/* set processing function */
method_index = filter->clipping_method;
if (method_index >= NUM_METHODS || method_index < 0)
method_index = METHOD_CLIP;
filter->process = processing_functions[filter->format_index][method_index];
return TRUE;
}
static void
gst_audio_amplify_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
switch (prop_id) {
case PROP_AMPLIFICATION:
filter->amplification = g_value_get_float (value);
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
filter->amplification == 1.0);
break;
case PROP_CLIPPING_METHOD:
filter->clipping_method = g_value_get_enum (value);
gst_audio_amplify_set_process_function (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_amplify_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (object);
switch (prop_id) {
case PROP_AMPLIFICATION:
g_value_set_float (value, filter->amplification);
break;
case PROP_CLIPPING_METHOD:
g_value_set_enum (value, filter->clipping_method);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_amplify_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
gboolean ret;
if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
filter->format_index = 0;
else if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
filter->format_index = 1;
else
goto wrong_format;
ret = gst_audio_amplify_set_process_function (filter);
if (!ret)
GST_WARNING ("can't process input");
return ret;
wrong_format:
GST_DEBUG ("wrong format");
return FALSE;
}
static void
gst_audio_amplify_transform_int_clip (GstAudioAmplify * filter,
gint16 * data, guint num_samples)
{
gint i;
glong val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_amplify_transform_int_wrap_negative (GstAudioAmplify * filter,
gint16 * data, guint num_samples)
{
gint i;
glong val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
if (val > G_MAXINT16)
val = ((val - G_MININT16) & 0xffff) + G_MININT16;
else if (val < G_MININT16)
val = ((val - G_MAXINT16) & 0xffff) + G_MAXINT16;
*data++ = val;
}
}
static void
gst_audio_amplify_transform_int_wrap_positive (GstAudioAmplify * filter,
gint16 * data, guint num_samples)
{
gint i;
glong val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
while (val > G_MAXINT16 || val < G_MININT16) {
if (val > G_MAXINT16)
val = G_MAXINT16 - (val - G_MAXINT16);
else if (val < G_MININT16)
val = G_MININT16 - (val - G_MININT16);
}
*data++ = val;
}
}
static void
gst_audio_amplify_transform_float_clip (GstAudioAmplify * filter,
gfloat * data, guint num_samples)
{
gint i;
gfloat val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
if (val > 1.0)
val = 1.0;
else if (val < -1.0)
val = -1.0;
*data++ = val;
}
}
static void
gst_audio_amplify_transform_float_wrap_negative (GstAudioAmplify * filter,
gfloat * data, guint num_samples)
{
gint i;
gfloat val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
while (val > 1.0 || val < -1.0) {
if (val > 1.0)
val = -1.0 + (val - 1.0);
else if (val < -1.0)
val = 1.0 + (val + 1.0);
}
*data++ = val;
}
}
static void
gst_audio_amplify_transform_float_wrap_positive (GstAudioAmplify * filter,
gfloat * data, guint num_samples)
{
gint i;
gfloat val;
for (i = 0; i < num_samples; i++) {
val = (*data) * filter->amplification;
while (val > 1.0 || val < -1.0) {
if (val > 1.0)
val = 1.0 - (val - 1.0);
else if (val < -1.0)
val = -1.0 - (val + 1.0);
}
*data++ = val;
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (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) ||
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;
}