Commit 54b10ebf authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio...

gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...

Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
parent 1f51fd97
2007-01-24 Sebastian Dröge <slomo@circular-chaos.org>
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
2007-01-24 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
......
......@@ -77,6 +77,7 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/apetag/gstapedemux.h \
$(top_srcdir)/gst/audiofx/audiopanorama.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audioamplify.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \
......
......@@ -16,6 +16,7 @@
<xi:include href="xml/element-apev2mux.xml" />
<xi:include href="xml/element-audiopanorama.xml" />
<xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audioamplify.xml" />
<xi:include href="xml/element-autoaudiosink.xml" />
<xi:include href="xml/element-autovideosink.xml" />
<xi:include href="xml/element-avidemux.xml" />
......
......@@ -47,6 +47,16 @@ GST_AUDIO_INVERT
GST_AUDIO_INVERT_CLASS
</SECTION>
<SECTION>
<FILE>element-audioamplify</FILE>
GstAudioAmplify
<TITLE>audioamplify</TITLE>
<SUBSECTION Standard>
GstAudioAmplifyClass
GST_AUDIO_AMPLIFY
GST_AUDIO_AMPLIFY_CLASS
</SECTION>
<SECTION>
<FILE>element-autoaudiosink</FILE>
GstAutoAudioSink
......
......@@ -16648,3 +16648,23 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioAmplify::amplification</NAME>
<TYPE>gfloat</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Amplification</NICK>
<BLURB>Factor of amplification.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioAmplify::clipping-method</NAME>
<TYPE>GstAudioPanoramaClippingMethod</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Clipping method</NICK>
<BLURB>Selects how to handle values higher than the maximum.</BLURB>
<DEFAULT>Normal Clipping (default)</DEFAULT>
</ARG>
......@@ -9,6 +9,13 @@
<package>GStreamer</package>
<origin>http://gstreamer.net/</origin>
<elements>
<element>
<name>audioamplify</name>
<longname>AudioAmplify</longname>
<class>Filter/Effect/Audio</class>
<description>Amplifies an audio stream by a given factor</description>
<author>Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author>
</element>
<element>
<name>audioinvert</name>
<longname>AudioInvert</longname>
......
......@@ -5,7 +5,8 @@ plugin_LTLIBRARIES = libgstaudiofx.la
# sources used to compile this plug-in
libgstaudiofx_la_SOURCES = audiofx.c\
audiopanorama.c \
audioinvert.c
audioinvert.c \
audioamplify.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
......@@ -18,4 +19,5 @@ libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
# headers we need but don't want installed
noinst_HEADERS = audiopanorama.h \
audioinvert.h
audioinvert.h \
audioamplify.h
/*
* 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/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}
};
gtype = g_enum_register_static ("GstAudioPanoramaClippingMethod", values);
}
return gtype;
}
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32; "
"audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32; "
"audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
);
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element");
GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, 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_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
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);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details (element_class, &element_details);
}
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_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (gst_audio_amplify_set_caps);
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->width = 0;
filter->format_float = FALSE;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
}
static gboolean
gst_audio_amplify_set_process_function (GstAudioAmplify * filter)
{
gint format_index, method_index;
/* set processing function */
format_index = (filter->format_float) ? 1 : 0;
method_index = filter->clipping_method;
if (method_index >= NUM_METHODS || method_index < 0)
method_index = METHOD_CLIP;
filter->process = processing_functions[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;
}
}
/* GstBaseTransform vmethod implementations */
static gboolean
gst_audio_amplify_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
const GstStructure *structure;
gboolean ret;
gint width;
const gchar *fmt;
/*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
structure = gst_caps_get_structure (incaps, 0);
ret = gst_structure_get_int (structure, "width", &width);
if (!ret)
goto no_width;
filter->width = width / 8;
fmt = gst_structure_get_name (structure);
if (!strcmp (fmt, "audio/x-raw-int"))
filter->format_float = FALSE;
else
filter->format_float = TRUE;
GST_DEBUG ("try to process %s input", fmt);
ret = gst_audio_amplify_set_process_function (filter);
if (!ret)
GST_WARNING ("can't process input");
return TRUE;
no_width:
GST_DEBUG ("no width in caps");
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;
}
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
guint num_samples = GST_BUFFER_SIZE (buf) / filter->width;
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
/*
* 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.
*/
#ifndef __GST_AUDIO_AMPLIFY_H__
#define __GST_AUDIO_AMPLIFY_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_AMPLIFY (gst_audio_amplify_get_type())
#define GST_AUDIO_AMPLIFY(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplify))
#define GST_IS_AUDIO_AMPLIFY(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AMPLIFY))
#define GST_AUDIO_AMPLIFY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass))
#define GST_IS_AUDIO_AMPLIFY_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_AMPLIFY))
#define GST_AUDIO_AMPLIFY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_AMPLIFY,GstAudioAmplifyClass))
typedef struct _GstAudioAmplify GstAudioAmplify;
typedef struct _GstAudioAmplifyClass GstAudioAmplifyClass;
typedef void (*GstAudioAmplifyProcessFunc) (GstAudioAmplify *, guint8 *, guint);
struct _GstAudioAmplify
{
GstBaseTransform element;
gfloat amplification;
/* < private > */
GstAudioAmplifyProcessFunc process;
gint clipping_method;
gint width;
gboolean format_float;
};
struct _GstAudioAmplifyClass
{
GstBaseTransformClass parent;
};
GType gst_audio_amplify_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_AMPLIFY_H__ */
......@@ -27,6 +27,7 @@
#include "audiopanorama.h"
#include "audioinvert.h"