From 9c17a6006f1d1dbe921fddbe4b7315b5a3714bcc Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Wed, 28 May 2008 14:07:21 +0000 Subject: Rename audiovoice to audiokaraoke and add it to the docs. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/gst-plugins-good-plugins.hierarchy: * docs/plugins/inspect/plugin-audiofx.xml: * gst/audiofx/Makefile.am: * gst/audiofx/audiofx.c: * gst/audiofx/audiokaraoke.c: * gst/audiofx/audiokaraoke.h: * gst/audiofx/audiovoice.c: * gst/audiofx/audiovoice.h: Rename audiovoice to audiokaraoke and add it to the docs. --- gst/audiofx/Makefile.am | 4 +- gst/audiofx/audiofx.c | 6 +- gst/audiofx/audiokaraoke.c | 359 +++++++++++++++++++++++++++++++++++++++++++++ gst/audiofx/audiokaraoke.h | 70 +++++++++ gst/audiofx/audiovoice.c | 359 --------------------------------------------- gst/audiofx/audiovoice.h | 70 --------- 6 files changed, 434 insertions(+), 434 deletions(-) create mode 100644 gst/audiofx/audiokaraoke.c create mode 100644 gst/audiofx/audiokaraoke.h delete mode 100644 gst/audiofx/audiovoice.c delete mode 100644 gst/audiofx/audiovoice.h (limited to 'gst/audiofx') diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index 28de5e2a..5754eba9 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -8,7 +8,7 @@ libgstaudiofx_la_SOURCES = audiofx.c\ audioinvert.c \ audioamplify.c \ audiodynamic.c \ - audiovoice.c \ + audiokaraoke.c \ audiocheblimit.c \ audiochebband.c \ audiowsincband.c \ @@ -32,7 +32,7 @@ noinst_HEADERS = audiopanorama.h \ audioinvert.h \ audioamplify.h \ audiodynamic.h \ - audiovoice.h \ + audiokaraoke.h \ audiocheblimit.h \ audiochebband.h \ audiowsincband.h \ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index f9b62ca9..43d1b0cc 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -27,7 +27,7 @@ #include "audiopanorama.h" #include "audioinvert.h" -#include "audiovoice.h" +#include "audiokaraoke.h" #include "audioamplify.h" #include "audiodynamic.h" #include "audiocheblimit.h" @@ -50,8 +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, "audiokaraoke", GST_RANK_NONE, + GST_TYPE_AUDIO_KARAOKE) && 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/audiokaraoke.c b/gst/audiofx/audiokaraoke.c new file mode 100644 index 00000000..ec505681 --- /dev/null +++ b/gst/audiofx/audiokaraoke.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-audiokaraoke + * @short_description: Voice removal element + * + * + * Remove the voice from audio by filtering the center channel. + * This plugin is useful for karaoke applications. + * Example launch line + * + * + * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include +#include +#include +#include +#include + +#include "audiokaraoke.h" + +#define GST_CAT_DEFAULT gst_audio_karaoke_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioKaraoke", + "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_karaoke_debug, "audiokaraoke", 0, "audiokaraoke element"); + +GST_BOILERPLATE_FULL (GstAudioKaraoke, gst_audio_karaoke, GstAudioFilter, + GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +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_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_karaoke_class_init (GstAudioKaraokeClass * klass) +{ + GObjectClass *gobject_class; + + gobject_class = (GObjectClass *) 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_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_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, GstAudioKaraokeClass * 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 (GstAudioKaraoke * 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_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 = + 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/audiokaraoke.h b/gst/audiofx/audiokaraoke.h new file mode 100644 index 00000000..727fafae --- /dev/null +++ b/gst/audiofx/audiokaraoke.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_KARAOKE_H__ +#define __GST_AUDIO_KARAOKE_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_KARAOKE (gst_audio_karaoke_get_type()) +#define GST_AUDIO_KARAOKE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_KARAOKE,GstAudioKaraoke)) +#define GST_IS_AUDIO_KARAOKE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_KARAOKE)) +#define GST_AUDIO_KARAOKE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_KARAOKE,GstAudioKaraokeClass)) +#define GST_IS_AUDIO_KARAOKE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_KARAOKE)) +#define GST_AUDIO_KARAOKE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_KARAOKE,GstAudioKaraokeClass)) +typedef struct _GstAudioKaraoke GstAudioKaraoke; +typedef struct _GstAudioKaraokeClass GstAudioKaraokeClass; + +typedef void (*GstAudioKaraokeProcessFunc) (GstAudioKaraoke *, guint8 *, guint); + +struct _GstAudioKaraoke +{ + 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 > */ + GstAudioKaraokeProcessFunc process; +}; + +struct _GstAudioKaraokeClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_karaoke_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_KARAOKE_H__ */ diff --git a/gst/audiofx/audiovoice.c b/gst/audiofx/audiovoice.c deleted file mode 100644 index 08916c11..00000000 --- a/gst/audiofx/audiovoice.c +++ /dev/null @@ -1,359 +0,0 @@ -/* - * 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 deleted file mode 100644 index cf3ff4f6..00000000 --- a/gst/audiofx/audiovoice.h +++ /dev/null @@ -1,70 +0,0 @@ -/* - * 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__ */ -- cgit