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363 lines
11 KiB
C
363 lines
11 KiB
C
/*
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* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audiokaraoke
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*
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* Remove the voice from audio by filtering the center channel.
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* This plugin is useful for karaoke applications.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink
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* ]|
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <math.h>
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <gst/controller/gstcontroller.h>
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#include "audiokaraoke.h"
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#define GST_CAT_DEFAULT gst_audio_karaoke_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_LEVEL 1.0
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#define DEFAULT_MONO_LEVEL 1.0
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#define DEFAULT_FILTER_BAND 220.0
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#define DEFAULT_FILTER_WIDTH 100.0
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enum
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{
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PROP_0,
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PROP_LEVEL,
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PROP_MONO_LEVEL,
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PROP_FILTER_BAND,
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PROP_FILTER_WIDTH,
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PROP_LAST
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};
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#define ALLOWED_CAPS \
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"audio/x-raw-int," \
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" depth=(int)16," \
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" width=(int)16," \
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" endianness=(int)BYTE_ORDER," \
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" signed=(bool)TRUE," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]; " \
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"audio/x-raw-float," \
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" width=(int)32," \
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" endianness=(int)BYTE_ORDER," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]"
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_karaoke_debug, "audiokaraoke", 0, "audiokaraoke element");
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GST_BOILERPLATE_FULL (GstAudioKaraoke, gst_audio_karaoke, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static void gst_audio_karaoke_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_karaoke_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_karaoke_setup (GstAudioFilter * filter,
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GstRingBufferSpec * format);
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static GstFlowReturn gst_audio_karaoke_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static void gst_audio_karaoke_transform_int (GstAudioKaraoke * filter,
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gint16 * data, guint num_samples);
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static void gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,
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gfloat * data, guint num_samples);
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/* GObject vmethod implementations */
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static void
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gst_audio_karaoke_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstCaps *caps;
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gst_element_class_set_details_simple (element_class, "AudioKaraoke",
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"Filter/Effect/Audio",
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"Removes voice from sound", "Wim Taymans <wim.taymans@gmail.com>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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}
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static void
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gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_audio_karaoke_set_property;
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gobject_class->get_property = gst_audio_karaoke_get_property;
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g_object_class_install_property (gobject_class, PROP_LEVEL,
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g_param_spec_float ("level", "Level",
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"Level of the effect (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MONO_LEVEL,
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g_param_spec_float ("mono-level", "Mono Level",
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"Level of the mono channel (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_FILTER_BAND,
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g_param_spec_float ("filter-band", "Filter Band",
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"The Frequency band of the filter", 0.0, 441.0, DEFAULT_FILTER_BAND,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_FILTER_WIDTH,
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g_param_spec_float ("filter-width", "Filter Width",
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"The Frequency width of the filter", 0.0, 100.0, DEFAULT_FILTER_WIDTH,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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GST_AUDIO_FILTER_CLASS (klass)->setup =
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GST_DEBUG_FUNCPTR (gst_audio_karaoke_setup);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_karaoke_transform_ip);
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}
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static void
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gst_audio_karaoke_init (GstAudioKaraoke * filter, GstAudioKaraokeClass * klass)
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{
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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filter->level = DEFAULT_LEVEL;
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filter->mono_level = DEFAULT_MONO_LEVEL;
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filter->filter_band = DEFAULT_FILTER_BAND;
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filter->filter_width = DEFAULT_FILTER_WIDTH;
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}
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static void
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update_filter (GstAudioKaraoke * filter, gint rate)
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{
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gfloat A, B, C;
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if (rate == 0)
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return;
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C = exp (-2 * G_PI * filter->filter_width / rate);
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B = -4 * C / (1 + C) * cos (2 * G_PI * filter->filter_band / rate);
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A = sqrt (1 - B * B / (4 * C)) * (1 - C);
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filter->A = A;
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filter->B = B;
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filter->C = C;
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filter->y1 = 0.0;
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filter->y2 = 0.0;
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}
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static void
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gst_audio_karaoke_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioKaraoke *filter;
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filter = GST_AUDIO_KARAOKE (object);
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switch (prop_id) {
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case PROP_LEVEL:
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filter->level = g_value_get_float (value);
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break;
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case PROP_MONO_LEVEL:
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filter->mono_level = g_value_get_float (value);
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break;
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case PROP_FILTER_BAND:
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filter->filter_band = g_value_get_float (value);
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update_filter (filter, filter->rate);
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break;
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case PROP_FILTER_WIDTH:
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filter->filter_width = g_value_get_float (value);
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update_filter (filter, filter->rate);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_karaoke_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioKaraoke *filter;
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filter = GST_AUDIO_KARAOKE (object);
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switch (prop_id) {
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case PROP_LEVEL:
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g_value_set_float (value, filter->level);
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break;
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case PROP_MONO_LEVEL:
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g_value_set_float (value, filter->mono_level);
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break;
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case PROP_FILTER_BAND:
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g_value_set_float (value, filter->filter_band);
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break;
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case PROP_FILTER_WIDTH:
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g_value_set_float (value, filter->filter_width);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* GstAudioFilter vmethod implementations */
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static gboolean
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gst_audio_karaoke_setup (GstAudioFilter * base, GstRingBufferSpec * format)
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{
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GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
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gboolean ret = TRUE;
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filter->channels = format->channels;
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filter->rate = format->rate;
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if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
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filter->process = (GstAudioKaraokeProcessFunc)
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gst_audio_karaoke_transform_float;
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else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
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filter->process = (GstAudioKaraokeProcessFunc)
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gst_audio_karaoke_transform_int;
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else
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ret = FALSE;
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update_filter (filter, format->rate);
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return ret;
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}
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static void
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gst_audio_karaoke_transform_int (GstAudioKaraoke * filter,
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gint16 * data, guint num_samples)
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{
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gint i, l, r, o, x;
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gint channels;
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gdouble y;
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gint level;
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channels = filter->channels;
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level = filter->level * 256;
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for (i = 0; i < num_samples; i += channels) {
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/* get left and right inputs */
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l = data[i];
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r = data[i + 1];
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/* do filtering */
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x = (l + r) / 2;
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y = (filter->A * x - filter->B * filter->y1) - filter->C * filter->y2;
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filter->y2 = filter->y1;
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filter->y1 = y;
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/* filter mono signal */
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o = (int) (y * filter->mono_level);
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o = CLAMP (o, G_MININT16, G_MAXINT16);
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o = (o * level) >> 8;
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/* now cut the center */
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x = l - ((r * level) >> 8) + o;
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r = r - ((l * level) >> 8) + o;
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data[i] = CLAMP (x, G_MININT16, G_MAXINT16);
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data[i + 1] = CLAMP (r, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,
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gfloat * data, guint num_samples)
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{
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gint i;
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gint channels;
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gdouble l, r, o;
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gdouble y;
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channels = filter->channels;
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for (i = 0; i < num_samples; i += channels) {
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/* get left and right inputs */
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l = data[i];
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r = data[i + 1];
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/* do filtering */
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y = (filter->A * ((l + r) / 2.0) - filter->B * filter->y1) -
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filter->C * filter->y2;
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filter->y2 = filter->y1;
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filter->y1 = y;
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/* filter mono signal */
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o = y * filter->mono_level * filter->level;
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/* now cut the center */
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data[i] = l - (r * filter->level) + o;
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data[i + 1] = r - (l * filter->level) + o;
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}
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}
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/* GstBaseTransform vmethod implementations */
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static GstFlowReturn
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gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
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{
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GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
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guint num_samples;
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GstClockTime timestamp, stream_time;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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stream_time =
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gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
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GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (G_OBJECT (filter), stream_time);
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num_samples =
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GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
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if (gst_base_transform_is_passthrough (base) ||
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G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
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return GST_FLOW_OK;
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filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
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return GST_FLOW_OK;
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}
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