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GST-Tensordecoder-ov_ep/gst/audiofx/audiokaraoke.c
Wim Taymans 237ca1631f port some more elements to 0.11
2011-04-25 12:49:36 +02:00

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/*
* 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-audiokaraoke
*
* Remove the voice from audio by filtering the center channel.
* This plugin is useful for karaoke applications.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink
* ]|
* </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 "audiokaraoke.h"
#define GST_CAT_DEFAULT gst_audio_karaoke_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* 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]"
G_DEFINE_TYPE (GstAudioKaraoke, gst_audio_karaoke, GST_TYPE_AUDIO_FILTER);
static void gst_audio_karaoke_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_karaoke_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_audio_karaoke_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn gst_audio_karaoke_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static void gst_audio_karaoke_transform_int (GstAudioKaraoke * filter,
gint16 * data, guint num_samples);
static void gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,
gfloat * data, guint num_samples);
/* GObject vmethod implementations */
static void
gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_karaoke_debug, "audiokaraoke", 0,
"audiokaraoke element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audio_karaoke_set_property;
gobject_class->get_property = gst_audio_karaoke_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_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
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_PARAM_STATIC_STRINGS));
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 | G_PARAM_STATIC_STRINGS));
gst_element_class_set_details_simple (gstelement_class, "AudioKaraoke",
"Filter/Effect/Audio",
"Removes voice from sound", "Wim Taymans <wim.taymans@gmail.com>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_karaoke_setup);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_karaoke_transform_ip);
}
static void
gst_audio_karaoke_init (GstAudioKaraoke * filter)
{
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 (GstAudioKaraoke * filter, gint rate)
{
gfloat A, B, C;
if (rate == 0)
return;
C = exp (-2 * G_PI * filter->filter_width / rate);
B = -4 * C / (1 + C) * cos (2 * G_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_karaoke_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioKaraoke *filter;
filter = GST_AUDIO_KARAOKE (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_karaoke_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioKaraoke *filter;
filter = GST_AUDIO_KARAOKE (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_karaoke_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
gboolean ret = TRUE;
filter->channels = format->channels;
filter->rate = format->rate;
if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
filter->process = (GstAudioKaraokeProcessFunc)
gst_audio_karaoke_transform_float;
else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
filter->process = (GstAudioKaraokeProcessFunc)
gst_audio_karaoke_transform_int;
else
ret = FALSE;
update_filter (filter, format->rate);
return ret;
}
static void
gst_audio_karaoke_transform_int (GstAudioKaraoke * 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_karaoke_transform_float (GstAudioKaraoke * 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_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
guint num_samples;
GstClockTime timestamp, stream_time;
guint8 *data;
gsize size;
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);
if (gst_base_transform_is_passthrough (base) ||
G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
filter->process (filter, data, num_samples);
gst_buffer_unmap (buf, data, size);
return GST_FLOW_OK;
}
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