Commit 0007831a authored by Wim Taymans's avatar Wim Taymans
Browse files

gst/audiofx/: Add simple voice removal element. Yay karaoke.

Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audiovoice.c: (gst_audio_voice_base_init),
(gst_audio_voice_class_init), (gst_audio_voice_init),
(update_filter), (gst_audio_voice_set_property),
(gst_audio_voice_get_property), (gst_audio_voice_setup),
(gst_audio_voice_transform_int), (gst_audio_voice_transform_float),
(gst_audio_voice_transform_ip):
* gst/audiofx/audiovoice.h:
Add simple voice removal element. Yay karaoke.
parent 61597d99
2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audiovoice.c: (gst_audio_voice_base_init),
(gst_audio_voice_class_init), (gst_audio_voice_init),
(update_filter), (gst_audio_voice_set_property),
(gst_audio_voice_get_property), (gst_audio_voice_setup),
(gst_audio_voice_transform_int), (gst_audio_voice_transform_float),
(gst_audio_voice_transform_ip):
* gst/audiofx/audiovoice.h:
Add simple voice removal element. Yay karaoke.
2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
 
Patch by: William M. Brack <wbrack at mmm dot com dot hk>
......
......@@ -8,6 +8,7 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audioinvert.c \
audioamplify.c \
audiodynamic.c \
audiovoice.c \
audiocheblimit.c \
audiochebband.c \
audiowsincband.c \
......@@ -31,6 +32,7 @@ noinst_HEADERS = audiopanorama.h \
audioinvert.h \
audioamplify.h \
audiodynamic.h \
audiovoice.h \
audiocheblimit.h \
audiochebband.h \
audiowsincband.h \
......
......@@ -27,6 +27,7 @@
#include "audiopanorama.h"
#include "audioinvert.h"
#include "audiovoice.h"
#include "audioamplify.h"
#include "audiodynamic.h"
#include "audiocheblimit.h"
......@@ -49,6 +50,8 @@ plugin_init (GstPlugin * plugin)
GST_TYPE_AUDIO_PANORAMA) &&
gst_element_register (plugin, "audioinvert", GST_RANK_NONE,
GST_TYPE_AUDIO_INVERT) &&
gst_element_register (plugin, "audiovoice", GST_RANK_NONE,
GST_TYPE_AUDIO_VOICE) &&
gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
GST_TYPE_AUDIO_AMPLIFY) &&
gst_element_register (plugin, "audiodynamic", GST_RANK_NONE,
......
/*
* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>
*
* 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-audiovoice
* @short_description: Voice removal element
*
* <refsect2>
* Remove the voice from audio by removing the center channel.
* This plugin is useful for karaoke applications.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch filesrc location="song.ogg" ! oggdemux ! vorbisdec ! audiovoice ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#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 "audiovoice.h"
#define GST_CAT_DEFAULT gst_audio_voice_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioVoice",
"Filter/Effect/Audio",
"Removes voice from sound",
"Wim Taymans <wim.taymans@gmail.com>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_LEVEL 1.0
#define DEFAULT_MONO_LEVEL 1.0
#define DEFAULT_FILTER_BAND 220.0
#define DEFAULT_FILTER_WIDTH 100.0
enum
{
PROP_0,
PROP_LEVEL,
PROP_MONO_LEVEL,
PROP_FILTER_BAND,
PROP_FILTER_WIDTH,
PROP_LAST
};
#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_voice_debug, "audiovoice", 0, "audiovoice element");
GST_BOILERPLATE_FULL (GstAudioVoice, gst_audio_voice, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void gst_audio_voice_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_voice_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_voice_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn gst_audio_voice_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void gst_audio_voice_transform_int (GstAudioVoice * filter,
gint16 * data, guint num_samples);
static void gst_audio_voice_transform_float (GstAudioVoice * filter,
gfloat * data, guint num_samples);
/* GObject vmethod implementations */
static void
gst_audio_voice_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_voice_class_init (GstAudioVoiceClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audio_voice_set_property;
gobject_class->get_property = gst_audio_voice_get_property;
g_object_class_install_property (gobject_class, PROP_LEVEL,
g_param_spec_float ("level", "Level",
"Level of the effect (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_MONO_LEVEL,
g_param_spec_float ("mono-level", "Mono Level",
"Level of the mono channel (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_FILTER_BAND,
g_param_spec_float ("filter-band", "Filter Band",
"The Frequency band of the filter", 0.0, 441.0, DEFAULT_FILTER_BAND,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_FILTER_WIDTH,
g_param_spec_float ("filter-width", "Filter Width",
"The Frequency width of the filter", 0.0, 100.0, DEFAULT_FILTER_WIDTH,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_voice_setup);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_voice_transform_ip);
}
static void
gst_audio_voice_init (GstAudioVoice * filter, GstAudioVoiceClass * klass)
{
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
filter->level = DEFAULT_LEVEL;
filter->mono_level = DEFAULT_MONO_LEVEL;
filter->filter_band = DEFAULT_FILTER_BAND;
filter->filter_width = DEFAULT_FILTER_WIDTH;
}
static void
update_filter (GstAudioVoice * filter, gint rate)
{
gfloat A, B, C;
if (rate == 0)
return;
C = exp (-2 * M_PI * filter->filter_width / rate);
B = -4 * C / (1 + C) * cos (2 * M_PI * filter->filter_band / rate);
A = sqrt (1 - B * B / (4 * C)) * (1 - C);
filter->A = A;
filter->B = B;
filter->C = C;
filter->y1 = 0.0;
filter->y2 = 0.0;
}
static void
gst_audio_voice_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioVoice *filter;
filter = GST_AUDIO_VOICE (object);
switch (prop_id) {
case PROP_LEVEL:
filter->level = g_value_get_float (value);
break;
case PROP_MONO_LEVEL:
filter->mono_level = g_value_get_float (value);
break;
case PROP_FILTER_BAND:
filter->filter_band = g_value_get_float (value);
update_filter (filter, filter->rate);
break;
case PROP_FILTER_WIDTH:
filter->filter_width = g_value_get_float (value);
update_filter (filter, filter->rate);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_voice_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioVoice *filter;
filter = GST_AUDIO_VOICE (object);
switch (prop_id) {
case PROP_LEVEL:
g_value_set_float (value, filter->level);
break;
case PROP_MONO_LEVEL:
g_value_set_float (value, filter->mono_level);
break;
case PROP_FILTER_BAND:
g_value_set_float (value, filter->filter_band);
break;
case PROP_FILTER_WIDTH:
g_value_set_float (value, filter->filter_width);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_voice_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioVoice *filter = GST_AUDIO_VOICE (base);
gboolean ret = TRUE;
filter->channels = format->channels;
filter->rate = format->rate;
if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
filter->process = (GstAudioVoiceProcessFunc)
gst_audio_voice_transform_float;
else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
filter->process = (GstAudioVoiceProcessFunc)
gst_audio_voice_transform_int;
else
ret = FALSE;
update_filter (filter, format->rate);
return ret;
}
static void
gst_audio_voice_transform_int (GstAudioVoice * filter,
gint16 * data, guint num_samples)
{
gint i, l, r, o, x;
gint channels;
gdouble y;
gint level;
channels = filter->channels;
level = filter->level * 256;
for (i = 0; i < num_samples; i += channels) {
/* get left and right inputs */
l = data[i];
r = data[i + 1];
/* do filtering */
x = (l + r) / 2;
y = (filter->A * x - filter->B * filter->y1) - filter->C * filter->y2;
filter->y2 = filter->y1;
filter->y1 = y;
/* filter mono signal */
o = (int) (y * filter->mono_level);
o = CLAMP (o, G_MININT16, G_MAXINT16);
o = (o * level) >> 8;
/* now cut the center */
x = l - ((r * level) >> 8) + o;
r = r - ((l * level) >> 8) + o;
data[i] = CLAMP (x, G_MININT16, G_MAXINT16);
data[i + 1] = CLAMP (r, G_MININT16, G_MAXINT16);
}
}
static void
gst_audio_voice_transform_float (GstAudioVoice * filter,
gfloat * data, guint num_samples)
{
gint i;
gint channels;
gdouble l, r, o;
gdouble y;
channels = filter->channels;
for (i = 0; i < num_samples; i += channels) {
/* get left and right inputs */
l = data[i];
r = data[i + 1];
/* do filtering */
y = (filter->A * ((l + r) / 2.0) - filter->B * filter->y1) -
filter->C * filter->y2;
filter->y2 = filter->y1;
filter->y1 = y;
/* filter mono signal */
o = y * filter->mono_level * filter->level;
/* now cut the center */
data[i] = l - (r * filter->level) + o;
data[i + 1] = r - (l * filter->level) + o;
}
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_voice_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioVoice *filter = GST_AUDIO_VOICE (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;
}
/*
* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>
*
* 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_VOICE_H__
#define __GST_AUDIO_VOICE_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_VOICE (gst_audio_voice_get_type())
#define GST_AUDIO_VOICE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_VOICE,GstAudioVoice))
#define GST_IS_AUDIO_VOICE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_VOICE))
#define GST_AUDIO_VOICE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_VOICE,GstAudioVoiceClass))
#define GST_IS_AUDIO_VOICE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_VOICE))
#define GST_AUDIO_VOICE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_VOICE,GstAudioVoiceClass))
typedef struct _GstAudioVoice GstAudioVoice;
typedef struct _GstAudioVoiceClass GstAudioVoiceClass;
typedef void (*GstAudioVoiceProcessFunc) (GstAudioVoice *, guint8 *, guint);
struct _GstAudioVoice
{
GstAudioFilter audiofilter;
gint channels;
gint rate;
/* properties */
gfloat level;
gfloat mono_level;
gfloat filter_band;
gfloat filter_width;
/* filter coef */
gfloat A, B, C;
gfloat y1, y2;
/* < private > */
GstAudioVoiceProcessFunc process;
};
struct _GstAudioVoiceClass
{
GstAudioFilterClass parent;
};
GType gst_audio_voice_get_type (void);
G_END_DECLS
#endif /* __GST_AUDIO_VOICE_H__ */
Supports Markdown
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment