diff --git a/ChangeLog b/ChangeLog index 31c7fe7f97..a342b60348 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,16 @@ +2008-05-26 Wim Taymans + + * 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 Patch by: William M. Brack diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index 3c0df213f9..28de5e2a90 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -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 \ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index ea6a8c2e63..f9b62ca943 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -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, diff --git a/gst/audiofx/audiovoice.c b/gst/audiofx/audiovoice.c new file mode 100644 index 0000000000..08916c113c --- /dev/null +++ b/gst/audiofx/audiovoice.c @@ -0,0 +1,359 @@ +/* + * GStreamer + * Copyright (C) 2008 Wim Taymans + * + * 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 + * + * + * Remove the voice from audio by removing the center channel. + * This plugin is useful for karaoke applications. + * Example launch line + * + * + * gst-launch filesrc location="song.ogg" ! oggdemux ! vorbisdec ! audiovoice ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include +#include +#include +#include +#include + +#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 "); + +/* 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; +} diff --git a/gst/audiofx/audiovoice.h b/gst/audiofx/audiovoice.h new file mode 100644 index 0000000000..cf3ff4f6f6 --- /dev/null +++ b/gst/audiofx/audiovoice.h @@ -0,0 +1,70 @@ +/* + * GStreamer + * Copyright (C) 2008 Wim Taymans + * + * 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 +#include +#include +#include + +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__ */