--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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__ */
projects / platform / upstream / gstreamer.git / commitdiff