From 99753365c6fe868196f454402f94e42d4e0170f0 Mon Sep 17 00:00:00 2001 From: Sebastian Dröge Date: Thu, 22 Jan 2009 10:40:34 +0100 Subject: Rename audioreverb to audioecho. Fixes bug #568395. The element can add an echo and a simple reverb effect to an audio stream but for a real reverb filter it would need some additional filtering to prevent a metallic-sounding result. --- gst/audiofx/Makefile.am | 4 +- gst/audiofx/audioecho.c | 367 ++++++++++++++++++++++++++++++++++++++++++++++ gst/audiofx/audioecho.h | 68 +++++++++ gst/audiofx/audiofx.c | 6 +- gst/audiofx/audioreverb.c | 367 ---------------------------------------------- gst/audiofx/audioreverb.h | 68 --------- 6 files changed, 440 insertions(+), 440 deletions(-) create mode 100644 gst/audiofx/audioecho.c create mode 100644 gst/audiofx/audioecho.h delete mode 100644 gst/audiofx/audioreverb.c delete mode 100644 gst/audiofx/audioreverb.h (limited to 'gst/audiofx') diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index f4f02be2..22f5fd02 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -17,7 +17,7 @@ libgstaudiofx_la_SOURCES = audiofx.c\ audiowsincband.c \ audiowsinclimit.c \ audiofirfilter.c \ - audioreverb.c + audioecho.c # flags used to compile this plugin libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ @@ -47,6 +47,6 @@ noinst_HEADERS = audiopanorama.h \ audiowsincband.h \ audiowsinclimit.h \ audiofirfilter.h \ - audioreverb.h \ + audioecho.h \ math_compat.h diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c new file mode 100644 index 00000000..04d51240 --- /dev/null +++ b/gst/audiofx/audioecho.c @@ -0,0 +1,367 @@ +/* + * GStreamer + * Copyright (C) 2009 Sebastian Dröge + * + * 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-audioecho + * + * + * audioecho adds an echo or reverb effect to an audio stream. The echo + * delay, intensity and the percentage of feedback can be configured. + * + * + * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink + * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink + * + * + * + * + * Since: 0.10.12 + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include "audioecho.h" + +#define GST_CAT_DEFAULT gst_audio_echo_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +enum +{ + PROP_0, + PROP_DELAY, + PROP_INTENSITY, + PROP_FEEDBACK +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width=(int) { 32, 64 }, " \ + " endianness=(int)BYTE_ORDER," \ + " rate=(int)[1,MAX]," \ + " channels=(int)[1,MAX]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element"); + +GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter, + GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static void gst_audio_echo_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_audio_echo_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); +static void gst_audio_echo_finalize (GObject * object); + +static gboolean gst_audio_echo_setup (GstAudioFilter * self, + GstRingBufferSpec * format); +static gboolean gst_audio_echo_stop (GstBaseTransform * base); +static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base, + GstBuffer * buf); + +static void gst_audio_echo_transform_float (GstAudioEcho * self, + gfloat * data, guint num_samples); +static void gst_audio_echo_transform_double (GstAudioEcho * self, + gdouble * data, guint num_samples); + +/* GObject vmethod implementations */ + +static void +gst_audio_echo_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details_simple (element_class, "Audio echo", + "Filter/Effect/Audio", + "Adds an echo or reverb effect to an audio stream", + "Sebastian Dröge "); + + 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_echo_class_init (GstAudioEchoClass * klass) +{ + GObjectClass *gobject_class = (GObjectClass *) klass; + GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass; + GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_echo_set_property; + gobject_class->get_property = gst_audio_echo_get_property; + gobject_class->finalize = gst_audio_echo_finalize; + + g_object_class_install_property (gobject_class, PROP_DELAY, + g_param_spec_uint64 ("delay", "Delay", + "Delay of the echo in nanosecondsecho", 1, G_MAXUINT64, + 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS + | GST_PARAM_CONTROLLABLE)); + + g_object_class_install_property (gobject_class, PROP_INTENSITY, + g_param_spec_float ("intensity", "Intensity", + "Intensity of the echo", 0.0, 1.0, + 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS + | GST_PARAM_CONTROLLABLE)); + + g_object_class_install_property (gobject_class, PROP_FEEDBACK, + g_param_spec_float ("feedback", "Feedback", + "Amount of feedback", 0.0, 1.0, + 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS + | GST_PARAM_CONTROLLABLE)); + + audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup); + basetransform_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip); + basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop); +} + +static void +gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass) +{ + self->delay = 1; + self->intensity = 0.0; + self->feedback = 0.0; + + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE); +} + +static void +gst_audio_echo_finalize (GObject * object) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (object); + + g_free (self->buffer); + self->buffer = NULL; + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_audio_echo_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (object); + + switch (prop_id) { + case PROP_DELAY:{ + guint rate, width, channels; + + GST_BASE_TRANSFORM_LOCK (self); + self->delay = g_value_get_uint64 (value); + + rate = GST_AUDIO_FILTER (self)->format.rate; + width = GST_AUDIO_FILTER (self)->format.width / 8; + channels = GST_AUDIO_FILTER (self)->format.channels; + + if (self->buffer && rate > 0) { + guint new_echo = + MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); + guint new_size = new_echo * width * channels; + + if (new_size > self->buffer_size) { + guint i; + guint8 *old_buffer = self->buffer; + + self->buffer_size = new_size; + self->buffer = g_malloc0 (new_size); + + for (i = 0; i < self->buffer_size_frames; i++) { + memcpy (&self->buffer[i * width * channels], + &old_buffer[((i + + self->buffer_pos) % self->buffer_size_frames) * + width * channels], channels * width); + } + self->buffer_size_frames = self->delay_frames = new_echo; + self->buffer_pos = 0; + } + } else if (self->buffer) { + g_free (self->buffer); + self->buffer = NULL; + } + + GST_BASE_TRANSFORM_UNLOCK (self); + } + break; + case PROP_INTENSITY:{ + GST_BASE_TRANSFORM_LOCK (self); + self->intensity = g_value_get_float (value); + GST_BASE_TRANSFORM_UNLOCK (self); + } + break; + case PROP_FEEDBACK:{ + GST_BASE_TRANSFORM_LOCK (self); + self->feedback = g_value_get_float (value); + GST_BASE_TRANSFORM_UNLOCK (self); + } + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_echo_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (object); + + switch (prop_id) { + case PROP_DELAY: + GST_BASE_TRANSFORM_LOCK (self); + g_value_set_uint64 (value, self->delay); + GST_BASE_TRANSFORM_UNLOCK (self); + break; + case PROP_INTENSITY: + GST_BASE_TRANSFORM_LOCK (self); + g_value_set_float (value, self->intensity); + GST_BASE_TRANSFORM_UNLOCK (self); + break; + case PROP_FEEDBACK: + GST_BASE_TRANSFORM_LOCK (self); + g_value_set_float (value, self->feedback); + GST_BASE_TRANSFORM_UNLOCK (self); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (base); + gboolean ret = TRUE; + + if (format->type == GST_BUFTYPE_FLOAT && format->width == 32) + self->process = (GstAudioEchoProcessFunc) + gst_audio_echo_transform_float; + else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64) + self->process = (GstAudioEchoProcessFunc) + gst_audio_echo_transform_double; + else + ret = FALSE; + + g_free (self->buffer); + self->buffer = NULL; + self->buffer_pos = 0; + self->buffer_size = 0; + self->buffer_size_frames = 0; + + return ret; +} + +static gboolean +gst_audio_echo_stop (GstBaseTransform * base) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (base); + + g_free (self->buffer); + self->buffer = NULL; + self->buffer_pos = 0; + self->buffer_size = 0; + self->buffer_size_frames = 0; + + return TRUE; +} + +#define TRANSFORM_FUNC(name, type) \ +static void \ +gst_audio_echo_transform_##name (GstAudioEcho * self, \ + type * data, guint num_samples) \ +{ \ + type *buffer = (type *) self->buffer; \ + guint channels = GST_AUDIO_FILTER (self)->format.channels; \ + guint rate = GST_AUDIO_FILTER (self)->format.rate; \ + guint i, j; \ + guint echo_index = self->buffer_size_frames - self->delay_frames; \ + gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \ + \ + if (echo_off < 0.0) \ + echo_off = 0.0; \ + \ + num_samples /= channels; \ + \ + for (i = 0; i < num_samples; i++) { \ + guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \ + guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \ + guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \ + for (j = 0; j < channels; j++) { \ + gdouble in = data[i*channels + j]; \ + gdouble echo0 = buffer[echo0_index + j]; \ + gdouble echo1 = buffer[echo1_index + j]; \ + gdouble echo = echo0 + (echo1-echo0)*echo_off; \ + type out = in + self->intensity * echo; \ + \ + data[i*channels + j] = out; \ + \ + buffer[rbout_index + j] = in + self->feedback * echo; \ + } \ + self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \ + } \ +} + +TRANSFORM_FUNC (float, gfloat); +TRANSFORM_FUNC (double, gdouble); + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf) +{ + GstAudioEcho *self = GST_AUDIO_ECHO (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8); + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf)); + + if (self->buffer == NULL) { + guint width, rate, channels; + + width = GST_AUDIO_FILTER (self)->format.width / 8; + rate = GST_AUDIO_FILTER (self)->format.rate; + channels = GST_AUDIO_FILTER (self)->format.channels; + + self->delay_frames = + MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); + + self->buffer_size_frames = MAX (self->delay_frames, 1000); + self->buffer_size = self->buffer_size_frames * width * channels; + self->buffer = g_malloc0 (self->buffer_size); + self->buffer_pos = 0; + } + + self->process (self, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} diff --git a/gst/audiofx/audioecho.h b/gst/audiofx/audioecho.h new file mode 100644 index 00000000..9513bb8e --- /dev/null +++ b/gst/audiofx/audioecho.h @@ -0,0 +1,68 @@ +/* + * GStreamer + * Copyright (C) 2009 Sebastian Dröge + * + * 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_ECHO_H__ +#define __GST_AUDIO_ECHO_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_AUDIO_ECHO (gst_audio_echo_get_type()) +#define GST_AUDIO_ECHO(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_ECHO,GstAudioEcho)) +#define GST_IS_AUDIO_ECHO(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_ECHO)) +#define GST_AUDIO_ECHO_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_ECHO,GstAudioEchoClass)) +#define GST_IS_AUDIO_ECHO_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_ECHO)) +#define GST_AUDIO_ECHO_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_ECHO,GstAudioEchoClass)) +typedef struct _GstAudioEcho GstAudioEcho; +typedef struct _GstAudioEchoClass GstAudioEchoClass; + +typedef void (*GstAudioEchoProcessFunc) (GstAudioEcho *, guint8 *, guint); + +struct _GstAudioEcho +{ + GstAudioFilter audiofilter; + + guint64 delay; + gfloat intensity; + gfloat feedback; + + /* < private > */ + GstAudioEchoProcessFunc process; + guint delay_frames; + guint8 *buffer; + guint buffer_pos; + guint buffer_size; + guint buffer_size_frames; +}; + +struct _GstAudioEchoClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_echo_get_type (void); + +G_END_DECLS + +#endif /* __GST_AUDIO_ECHO_H__ */ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index e23a638f..a268724d 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -36,7 +36,7 @@ #include "audiowsincband.h" #include "audiowsinclimit.h" #include "audiofirfilter.h" -#include "audioreverb.h" +#include "audioecho.h" /* entry point to initialize the plug-in * initialize the plug-in itself @@ -71,8 +71,8 @@ plugin_init (GstPlugin * plugin) GST_TYPE_AUDIO_WSINC_BAND) && gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE, GST_TYPE_AUDIO_FIR_FILTER) && - gst_element_register (plugin, "audioreverb", GST_RANK_NONE, - GST_TYPE_AUDIO_REVERB)); + gst_element_register (plugin, "audioecho", GST_RANK_NONE, + GST_TYPE_AUDIO_ECHO)); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, diff --git a/gst/audiofx/audioreverb.c b/gst/audiofx/audioreverb.c deleted file mode 100644 index 2d661b16..00000000 --- a/gst/audiofx/audioreverb.c +++ /dev/null @@ -1,367 +0,0 @@ -/* - * GStreamer - * Copyright (C) 2009 Sebastian Dröge - * - * 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-audioreverb - * - * - * audioreverb adds an echo or revert effect to an audio stream. The echo - * reverb, intensity and the percentage of feedback can be configured. - * - * - * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioreverb reverb=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink - * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioreverb reverb=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink - * - * - * - * - * Since: 0.10.12 - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include -#include -#include -#include - -#include "audioreverb.h" - -#define GST_CAT_DEFAULT gst_audio_reverb_debug -GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); - -enum -{ - PROP_0, - PROP_DELAY, - PROP_INTENSITY, - PROP_FEEDBACK -}; - -#define ALLOWED_CAPS \ - "audio/x-raw-float," \ - " width=(int) { 32, 64 }, " \ - " endianness=(int)BYTE_ORDER," \ - " rate=(int)[1,MAX]," \ - " channels=(int)[1,MAX]" - -#define DEBUG_INIT(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_audio_reverb_debug, "audioreverb", 0, "audioreverb element"); - -GST_BOILERPLATE_FULL (GstAudioReverb, gst_audio_reverb, GstAudioFilter, - GST_TYPE_AUDIO_FILTER, DEBUG_INIT); - -static void gst_audio_reverb_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_audio_reverb_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_audio_reverb_finalize (GObject * object); - -static gboolean gst_audio_reverb_setup (GstAudioFilter * self, - GstRingBufferSpec * format); -static gboolean gst_audio_reverb_stop (GstBaseTransform * base); -static GstFlowReturn gst_audio_reverb_transform_ip (GstBaseTransform * base, - GstBuffer * buf); - -static void gst_audio_reverb_transform_float (GstAudioReverb * self, - gfloat * data, guint num_samples); -static void gst_audio_reverb_transform_double (GstAudioReverb * self, - gdouble * data, guint num_samples); - -/* GObject vmethod implementations */ - -static void -gst_audio_reverb_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - GstCaps *caps; - - gst_element_class_set_details_simple (element_class, "Audio reverb", - "Filter/Effect/Audio", - "Adds an echo or reverb effect to an audio stream", - "Sebastian Dröge "); - - 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_reverb_class_init (GstAudioReverbClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass; - GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass; - - gobject_class->set_property = gst_audio_reverb_set_property; - gobject_class->get_property = gst_audio_reverb_get_property; - gobject_class->finalize = gst_audio_reverb_finalize; - - g_object_class_install_property (gobject_class, PROP_DELAY, - g_param_spec_uint64 ("delay", "Delay", - "Delay of the echo in nanoseconds", 1, G_MAXUINT64, - 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS - | GST_PARAM_CONTROLLABLE)); - - g_object_class_install_property (gobject_class, PROP_INTENSITY, - g_param_spec_float ("intensity", "Intensity", - "Intensity of the echo", 0.0, 1.0, - 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS - | GST_PARAM_CONTROLLABLE)); - - g_object_class_install_property (gobject_class, PROP_FEEDBACK, - g_param_spec_float ("feedback", "Feedback", - "Amount of feedback", 0.0, 1.0, - 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS - | GST_PARAM_CONTROLLABLE)); - - audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_reverb_setup); - basetransform_class->transform_ip = - GST_DEBUG_FUNCPTR (gst_audio_reverb_transform_ip); - basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_reverb_stop); -} - -static void -gst_audio_reverb_init (GstAudioReverb * self, GstAudioReverbClass * klass) -{ - self->delay = 1; - self->intensity = 0.0; - self->feedback = 0.0; - - gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE); -} - -static void -gst_audio_reverb_finalize (GObject * object) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (object); - - g_free (self->buffer); - self->buffer = NULL; - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static void -gst_audio_reverb_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (object); - - switch (prop_id) { - case PROP_DELAY:{ - guint rate, width, channels; - - GST_BASE_TRANSFORM_LOCK (self); - self->delay = g_value_get_uint64 (value); - - rate = GST_AUDIO_FILTER (self)->format.rate; - width = GST_AUDIO_FILTER (self)->format.width / 8; - channels = GST_AUDIO_FILTER (self)->format.channels; - - if (self->buffer && rate > 0) { - guint new_reverb = - MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); - guint new_size = new_reverb * width * channels; - - if (new_size > self->buffer_size) { - guint i; - guint8 *old_buffer = self->buffer; - - self->buffer_size = new_size; - self->buffer = g_malloc0 (new_size); - - for (i = 0; i < self->buffer_size_frames; i++) { - memcpy (&self->buffer[i * width * channels], - &old_buffer[((i + - self->buffer_pos) % self->buffer_size_frames) * - width * channels], channels * width); - } - self->buffer_size_frames = self->delay_frames = new_reverb; - self->buffer_pos = 0; - } - } else if (self->buffer) { - g_free (self->buffer); - self->buffer = NULL; - } - - GST_BASE_TRANSFORM_UNLOCK (self); - } - break; - case PROP_INTENSITY:{ - GST_BASE_TRANSFORM_LOCK (self); - self->intensity = g_value_get_float (value); - GST_BASE_TRANSFORM_UNLOCK (self); - } - break; - case PROP_FEEDBACK:{ - GST_BASE_TRANSFORM_LOCK (self); - self->feedback = g_value_get_float (value); - GST_BASE_TRANSFORM_UNLOCK (self); - } - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_reverb_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (object); - - switch (prop_id) { - case PROP_DELAY: - GST_BASE_TRANSFORM_LOCK (self); - g_value_set_uint64 (value, self->delay); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_INTENSITY: - GST_BASE_TRANSFORM_LOCK (self); - g_value_set_float (value, self->intensity); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - case PROP_FEEDBACK: - GST_BASE_TRANSFORM_LOCK (self); - g_value_set_float (value, self->feedback); - GST_BASE_TRANSFORM_UNLOCK (self); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -/* GstAudioFilter vmethod implementations */ - -static gboolean -gst_audio_reverb_setup (GstAudioFilter * base, GstRingBufferSpec * format) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (base); - gboolean ret = TRUE; - - if (format->type == GST_BUFTYPE_FLOAT && format->width == 32) - self->process = (GstAudioReverbProcessFunc) - gst_audio_reverb_transform_float; - else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64) - self->process = (GstAudioReverbProcessFunc) - gst_audio_reverb_transform_double; - else - ret = FALSE; - - g_free (self->buffer); - self->buffer = NULL; - self->buffer_pos = 0; - self->buffer_size = 0; - self->buffer_size_frames = 0; - - return ret; -} - -static gboolean -gst_audio_reverb_stop (GstBaseTransform * base) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (base); - - g_free (self->buffer); - self->buffer = NULL; - self->buffer_pos = 0; - self->buffer_size = 0; - self->buffer_size_frames = 0; - - return TRUE; -} - -#define TRANSFORM_FUNC(name, type) \ -static void \ -gst_audio_reverb_transform_##name (GstAudioReverb * self, \ - type * data, guint num_samples) \ -{ \ - type *buffer = (type *) self->buffer; \ - guint channels = GST_AUDIO_FILTER (self)->format.channels; \ - guint rate = GST_AUDIO_FILTER (self)->format.rate; \ - guint i, j; \ - guint reverb_index = self->buffer_size_frames - self->delay_frames; \ - gdouble reverb_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \ - \ - if (reverb_off < 0.0) \ - reverb_off = 0.0; \ - \ - num_samples /= channels; \ - \ - for (i = 0; i < num_samples; i++) { \ - guint echo0_index = ((reverb_index + self->buffer_pos) % self->buffer_size_frames) * channels; \ - guint echo1_index = ((reverb_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \ - guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \ - for (j = 0; j < channels; j++) { \ - gdouble in = data[i*channels + j]; \ - gdouble echo0 = buffer[echo0_index + j]; \ - gdouble echo1 = buffer[echo1_index + j]; \ - gdouble echo = echo0 + (echo1-echo0)*reverb_off; \ - type out = in + self->intensity * echo; \ - \ - data[i*channels + j] = out; \ - \ - buffer[rbout_index + j] = in + self->feedback * echo; \ - } \ - self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \ - } \ -} - -TRANSFORM_FUNC (float, gfloat); -TRANSFORM_FUNC (double, gdouble); - -/* GstBaseTransform vmethod implementations */ -static GstFlowReturn -gst_audio_reverb_transform_ip (GstBaseTransform * base, GstBuffer * buf) -{ - GstAudioReverb *self = GST_AUDIO_REVERB (base); - guint num_samples = - GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8); - - if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) - gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf)); - - if (self->buffer == NULL) { - guint width, rate, channels; - - width = GST_AUDIO_FILTER (self)->format.width / 8; - rate = GST_AUDIO_FILTER (self)->format.rate; - channels = GST_AUDIO_FILTER (self)->format.channels; - - self->delay_frames = - MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); - - self->buffer_size_frames = MAX (self->delay_frames, 1000); - self->buffer_size = self->buffer_size_frames * width * channels; - self->buffer = g_malloc0 (self->buffer_size); - self->buffer_pos = 0; - } - - self->process (self, GST_BUFFER_DATA (buf), num_samples); - - return GST_FLOW_OK; -} diff --git a/gst/audiofx/audioreverb.h b/gst/audiofx/audioreverb.h deleted file mode 100644 index 3ef5682e..00000000 --- a/gst/audiofx/audioreverb.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * GStreamer - * Copyright (C) 2009 Sebastian Dröge - * - * 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_REVERB_H__ -#define __GST_AUDIO_REVERB_H__ - -#include -#include -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AUDIO_REVERB (gst_audio_reverb_get_type()) -#define GST_AUDIO_REVERB(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_REVERB,GstAudioReverb)) -#define GST_IS_AUDIO_REVERB(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_REVERB)) -#define GST_AUDIO_REVERB_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass)) -#define GST_IS_AUDIO_REVERB_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_REVERB)) -#define GST_AUDIO_REVERB_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_REVERB,GstAudioReverbClass)) -typedef struct _GstAudioReverb GstAudioReverb; -typedef struct _GstAudioReverbClass GstAudioReverbClass; - -typedef void (*GstAudioReverbProcessFunc) (GstAudioReverb *, guint8 *, guint); - -struct _GstAudioReverb -{ - GstAudioFilter audiofilter; - - guint64 delay; - gfloat intensity; - gfloat feedback; - - /* < private > */ - GstAudioReverbProcessFunc process; - guint delay_frames; - guint8 *buffer; - guint buffer_pos; - guint buffer_size; - guint buffer_size_frames; -}; - -struct _GstAudioReverbClass -{ - GstAudioFilterClass parent; -}; - -GType gst_audio_reverb_get_type (void); - -G_END_DECLS - -#endif /* __GST_AUDIO_REVERB_H__ */ -- cgit