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Add an echo/reverb filter to the audiofx plugin, with configurable echo delay, intensity and feedback. Fixes bug #567...

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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:
* docs/plugins/inspect/plugin-spectrum.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
(gst_audio_reverb_class_init), (gst_audio_reverb_init),
(gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
(gst_audio_reverb_get_property), (gst_audio_reverb_setup),
(gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
* gst/audiofx/audioreverb.h:
* tests/check/Makefile.am:
* tests/check/elements/audioreverb.c: (setup_reverb),
(cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
Add an echo/reverb filter to the audiofx plugin, with configurable
echo delay, intensity and feedback. Fixes bug #567874.
This commit is contained in:
Sebastian Dröge 2009-01-19 11:19:08 +00:00
parent d912a42065
commit 344a9f4229
14 changed files with 798 additions and 3 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

23
ChangeLog
View file

@ -1,3 +1,26 @@
2009-01-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* 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:
* docs/plugins/inspect/plugin-spectrum.xml:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioreverb.c: (gst_audio_reverb_base_init),
(gst_audio_reverb_class_init), (gst_audio_reverb_init),
(gst_audio_reverb_finalize), (gst_audio_reverb_set_property),
(gst_audio_reverb_get_property), (gst_audio_reverb_setup),
(gst_audio_reverb_stop), (gst_audio_reverb_transform_ip):
* gst/audiofx/audioreverb.h:
* tests/check/Makefile.am:
* tests/check/elements/audioreverb.c: (setup_reverb),
(cleanup_reverb), (GST_START_TEST), (audioreverb_suite):
Add an echo/reverb filter to the audiofx plugin, with configurable
echo delay, intensity and feedback. Fixes bug #567874.
2009-01-19 Sebastian Dröge <sebastian.droege@collabora.co.uk> 2009-01-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state), * gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),

1
docs/plugins/Makefile.am
View file

@ -106,6 +106,7 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/alpha/gstalphacolor.h \ $(top_srcdir)/gst/alpha/gstalphacolor.h \
$(top_srcdir)/gst/apetag/gstapedemux.h \ $(top_srcdir)/gst/apetag/gstapedemux.h \
$(top_srcdir)/gst/audiofx/audioamplify.h \ $(top_srcdir)/gst/audiofx/audioamplify.h \
$(top_srcdir)/gst/audiofx/audioreverb.h \
$(top_srcdir)/gst/audiofx/audiodynamic.h \ $(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \ $(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audiokaraoke.h \ $(top_srcdir)/gst/audiofx/audiokaraoke.h \

1
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -23,6 +23,7 @@
<xi:include href="xml/element-audiowsincband.xml" /> <xi:include href="xml/element-audiowsincband.xml" />
<xi:include href="xml/element-audiowsinclimit.xml" /> <xi:include href="xml/element-audiowsinclimit.xml" />
<xi:include href="xml/element-audiofirfilter.xml" /> <xi:include href="xml/element-audiofirfilter.xml" />
<xi:include href="xml/element-audioreverb.xml" />
<xi:include href="xml/element-audiodynamic.xml" /> <xi:include href="xml/element-audiodynamic.xml" />
<xi:include href="xml/element-audioinvert.xml" /> <xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audiopanorama.xml" /> <xi:include href="xml/element-audiopanorama.xml" />

16
docs/plugins/gst-plugins-good-plugins-sections.txt
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@ -116,6 +116,22 @@ GST_TYPE_AUDIO_IIR_FILTER
gst_audio_iir_filter_get_type gst_audio_iir_filter_get_type
</SECTION> </SECTION>
<SECTION>
<FILE>element-audioreverb</FILE>
<TITLE>audioreverb</TITLE>
GstAudioReverb
<SUBSECTION Standard>
GstAudioReverbClass
GstAudioReverbProcessFunc
GST_AUDIO_REVERB
GST_AUDIO_REVERB_CLASS
GST_AUDIO_REVERB_GET_CLASS
GST_IS_AUDIO_REVERB
GST_IS_AUDIO_REVERB_CLASS
GST_TYPE_AUDIO_REVERB
gst_audio_reverb_get_type
</SECTION>
<SECTION> <SECTION>
<FILE>element-audiodynamic</FILE> <FILE>element-audiodynamic</FILE>
<TITLE>audiodynamic</TITLE> <TITLE>audiodynamic</TITLE>

60
docs/plugins/gst-plugins-good-plugins.args
View file

@ -19708,3 +19708,63 @@
<DEFAULT></DEFAULT> <DEFAULT></DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstAudioDelay::delay</NAME>
<TYPE>guint64</TYPE>
<RANGE>>= 1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Delay</NICK>
<BLURB>Delay in nanoseconds.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioDelay::feedback</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Feedback</NICK>
<BLURB>Amount of feedback.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioDelay::intensity</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Intensity</NICK>
<BLURB>Intensity of the echo.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioReverb::delay</NAME>
<TYPE>guint64</TYPE>
<RANGE>>= 1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Delay</NICK>
<BLURB>Delay of the echo in nanoseconds.</BLURB>
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioReverb::feedback</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Feedback</NICK>
<BLURB>Amount of feedback.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstAudioReverb::intensity</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,1]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Intensity</NICK>
<BLURB>Intensity of the echo.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>

3
docs/plugins/gst-plugins-good-plugins.hierarchy
View file

@ -64,6 +64,7 @@ GObject
GstAudioWSincLimit GstAudioWSincLimit
GstAudioWSincBand GstAudioWSincBand
GstAudioFIRFilter GstAudioFIRFilter
GstAudioReverb
GstIirEqualizer GstIirEqualizer
GstIirEqualizerNBands GstIirEqualizerNBands
GstIirEqualizer3Bands GstIirEqualizer3Bands
@ -221,6 +222,8 @@ GObject
GstRegistry GstRegistry
GstRingBuffer GstRingBuffer
GstSignalObject GstSignalObject
GstMixerTrack
GstMixerOptions
GstCmmlTagStream GstCmmlTagStream
GstCmmlTagHead GstCmmlTagHead
GstCmmlTagClip GstCmmlTagClip

21
docs/plugins/inspect/plugin-audiofx.xml
View file

@ -198,6 +198,27 @@
</caps> </caps>
</pads> </pads>
</element> </element>
<element>
<name>audioreverb</name>
<longname>Audio reverb</longname>
<class>Filter/Effect/Audio</class>
<description>Adds an echo or reverb effect to an audio stream</description>
<author>Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-float, width=(int){ 32, 64 }, endianness=(int)1234, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element> <element>
<name>audiowsincband</name> <name>audiowsincband</name>
<longname>Band pass &amp; band reject filter</longname> <longname>Band pass &amp; band reject filter</longname>

2
docs/plugins/inspect/plugin-spectrum.xml
View file

@ -14,7 +14,7 @@
<longname>Spectrum analyzer</longname> <longname>Spectrum analyzer</longname>
<class>Filter/Analyzer/Audio</class> <class>Filter/Analyzer/Audio</class>
<description>Run an FFT on the audio signal, output spectrum data</description> <description>Run an FFT on the audio signal, output spectrum data</description>
<author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;, Stefan Kost &lt;ensonic@users.sf.net&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</author> <author>Erik Walthinsen &lt;omega@cse.ogi.edu&gt;, Stefan Kost &lt;ensonic@users.sf.net&gt;, Sebastian Dröge &lt;sebastian.droege@collabora.co.uk&gt;</author>
<pads> <pads>
<caps> <caps>
<name>sink</name> <name>sink</name>

4
gst/audiofx/Makefile.am
View file

@ -16,7 +16,8 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audiofxbasefirfilter.c \ audiofxbasefirfilter.c \
audiowsincband.c \ audiowsincband.c \
audiowsinclimit.c \ audiowsinclimit.c \
audiofirfilter.c audiofirfilter.c \
audioreverb.c
# flags used to compile this plugin # flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@ -46,5 +47,6 @@ noinst_HEADERS = audiopanorama.h \
audiowsincband.h \ audiowsincband.h \
audiowsinclimit.h \ audiowsinclimit.h \
audiofirfilter.h \ audiofirfilter.h \
audioreverb.h \
math_compat.h math_compat.h

5
gst/audiofx/audiofx.c
View file

@ -36,6 +36,7 @@
#include "audiowsincband.h" #include "audiowsincband.h"
#include "audiowsinclimit.h" #include "audiowsinclimit.h"
#include "audiofirfilter.h" #include "audiofirfilter.h"
#include "audioreverb.h"
/* entry point to initialize the plug-in /* entry point to initialize the plug-in
* initialize the plug-in itself * initialize the plug-in itself
@ -69,7 +70,9 @@ plugin_init (GstPlugin * plugin)
gst_element_register (plugin, "audiowsincband", GST_RANK_NONE, gst_element_register (plugin, "audiowsincband", GST_RANK_NONE,
GST_TYPE_AUDIO_WSINC_BAND) && GST_TYPE_AUDIO_WSINC_BAND) &&
gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE, gst_element_register (plugin, "audiofirfilter", GST_RANK_NONE,
GST_TYPE_AUDIO_FIR_FILTER)); GST_TYPE_AUDIO_FIR_FILTER) &&
gst_element_register (plugin, "audioreverb", GST_RANK_NONE,
GST_TYPE_AUDIO_REVERB));
} }
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

367
gst/audiofx/audioreverb.c Normal file
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@ -0,0 +1,367 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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
*
* <refsect2>
* audioreverb adds an echo or revert effect to an audio stream. The echo
* reverb, intensity and the percentage of feedback can be configured.
* <para>
* <programlisting>
* 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
* </programlisting>
* </para>
* </refsect2>
*
* Since: 0.10.12
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#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 "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 <sebastian.droege@collabora.co.uk>");
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 = 0;
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;
}

68
gst/audiofx/audioreverb.h Normal file
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@ -0,0 +1,68 @@
/*
* GStreamer
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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 <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_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__ */

1
tests/check/Makefile.am
View file

@ -74,6 +74,7 @@ check_PROGRAMS = \
elements/audiocheblimit \ elements/audiocheblimit \
elements/audioiirfilter \ elements/audioiirfilter \
elements/audioamplify \ elements/audioamplify \
elements/audioreverb \
elements/audiodynamic \ elements/audiodynamic \
elements/audiowsincband \ elements/audiowsincband \
elements/audiowsinclimit \ elements/audiowsinclimit \

229
tests/check/elements/audioreverb.c Normal file
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@ -0,0 +1,229 @@
/* GStreamer
*
* Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* 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.
*/
#include <gst/check/gstcheck.h>
gboolean have_eos = FALSE;
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
GstPad *mysrcpad, *mysinkpad;
#define REVERB_CAPS_STRING \
"audio/x-raw-float, " \
"channels = (int) 2, " \
"rate = (int) 100000, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 64"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"channels = (int) [ 1, 2 ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 }")
);
GstElement *
setup_reverb ()
{
GstElement *reverb;
GST_DEBUG ("setup_reverb");
reverb = gst_check_setup_element ("audioreverb");
mysrcpad = gst_check_setup_src_pad (reverb, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (reverb, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return reverb;
}
void
cleanup_reverb (GstElement * reverb)
{
GST_DEBUG ("cleanup_reverb");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (reverb);
gst_check_teardown_sink_pad (reverb);
gst_check_teardown_element (reverb);
}
GST_START_TEST (test_passthrough)
{
GstElement *reverb;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble in[] = { 1.0, -1.0, 0.0, 0.5, -0.5, 0.0 };
gdouble *res;
reverb = setup_reverb ();
g_object_set (G_OBJECT (reverb), "delay", 1, "intensity", 0.0, "feedback",
0.0, NULL);
fail_unless (gst_element_set_state (reverb,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
caps = gst_caps_from_string (REVERB_CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
GST_INFO
("expected %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf",
in[0], in[1], in[2], in[3], in[4], in[5], res[0], res[1], res[2], res[3],
res[4], res[5]);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), in, sizeof (in)) == 0);
/* cleanup */
cleanup_reverb (reverb);
}
GST_END_TEST;
GST_START_TEST (test_reverb)
{
GstElement *reverb;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 0.0, 0.0 };
gdouble *res;
reverb = setup_reverb ();
g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
0.0, NULL);
fail_unless (gst_element_set_state (reverb,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
caps = gst_caps_from_string (REVERB_CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
GST_INFO
("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
res[8], res[9]);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
/* cleanup */
cleanup_reverb (reverb);
}
GST_END_TEST;
GST_START_TEST (test_feedback)
{
GstElement *reverb;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble in[] = { 1.0, -1.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, };
gdouble out[] = { 1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 0.0, 1.0, -1.0 };
gdouble *res;
reverb = setup_reverb ();
g_object_set (G_OBJECT (reverb), "delay", 20000, "intensity", 1.0, "feedback",
1.0, NULL);
fail_unless (gst_element_set_state (reverb,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
inbuffer = gst_buffer_new_and_alloc (sizeof (in));
memcpy (GST_BUFFER_DATA (inbuffer), in, sizeof (in));
fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, sizeof (in)) == 0);
caps = gst_caps_from_string (REVERB_CAPS_STRING);
gst_buffer_set_caps (inbuffer, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... but it ends up being collected on the global buffer list */
fail_unless_equals_int (g_list_length (buffers), 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
GST_INFO
("expected %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf real %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf %+lf",
out[0], out[1], out[2], out[3], out[4], out[5], out[6], out[7], out[8],
out[9], res[0], res[1], res[2], res[3], res[4], res[5], res[6], res[7],
res[8], res[9]);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, sizeof (out)) == 0);
/* cleanup */
cleanup_reverb (reverb);
}
GST_END_TEST;
static Suite *
audioreverb_suite (void)
{
Suite *s = suite_create ("audioreverb");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_passthrough);
tcase_add_test (tc_chain, test_reverb);
tcase_add_test (tc_chain, test_feedback);
return s;
}
GST_CHECK_MAIN (audioreverb);