Commit 344a9f42 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Add an echo/reverb filter to the audiofx plugin, with configurable echo delay,...

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

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.
parent d912a420
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>
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
......
......@@ -106,6 +106,7 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/alpha/gstalphacolor.h \
$(top_srcdir)/gst/apetag/gstapedemux.h \
$(top_srcdir)/gst/audiofx/audioamplify.h \
$(top_srcdir)/gst/audiofx/audioreverb.h \
$(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audiokaraoke.h \
......
......@@ -23,6 +23,7 @@
<xi:include href="xml/element-audiowsincband.xml" />
<xi:include href="xml/element-audiowsinclimit.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-audioinvert.xml" />
<xi:include href="xml/element-audiopanorama.xml" />
......
......@@ -116,6 +116,22 @@ GST_TYPE_AUDIO_IIR_FILTER
gst_audio_iir_filter_get_type
</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>
<FILE>element-audiodynamic</FILE>
<TITLE>audiodynamic</TITLE>
......
......@@ -19708,3 +19708,63 @@
<DEFAULT></DEFAULT>
</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>
......@@ -64,6 +64,7 @@ GObject
GstAudioWSincLimit
GstAudioWSincBand
GstAudioFIRFilter
GstAudioReverb
GstIirEqualizer
GstIirEqualizerNBands
GstIirEqualizer3Bands
......@@ -221,6 +222,8 @@ GObject
GstRegistry
GstRingBuffer
GstSignalObject
GstMixerTrack
GstMixerOptions
GstCmmlTagStream
GstCmmlTagHead
GstCmmlTagClip
......
......@@ -198,6 +198,27 @@
</caps>
</pads>
</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>
<name>audiowsincband</name>
<longname>Band pass &amp; band reject filter</longname>
......
......@@ -14,7 +14,7 @@
<longname>Spectrum analyzer</longname>
<class>Filter/Analyzer/Audio</class>
<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>
<caps>
<name>sink</name>
......
......@@ -16,7 +16,8 @@ libgstaudiofx_la_SOURCES = audiofx.c\
audiofxbasefirfilter.c \
audiowsincband.c \
audiowsinclimit.c \
audiofirfilter.c
audiofirfilter.c \
audioreverb.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
......@@ -46,5 +47,6 @@ noinst_HEADERS = audiopanorama.h \
audiowsincband.h \
audiowsinclimit.h \
audiofirfilter.h \
audioreverb.h \
math_compat.h
......@@ -36,6 +36,7 @@
#include "audiowsincband.h"
#include "audiowsinclimit.h"
#include "audiofirfilter.h"
#include "audioreverb.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
......@@ -69,7 +70,9 @@ plugin_init (GstPlugin * plugin)
gst_element_register (plugin, "audiowsincband", GST_RANK_NONE,
GST_TYPE_AUDIO_WSINC_BAND) &&
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,
......
This diff is collapsed.
/*
* 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__ */
......@@ -74,6 +74,7 @@ check_PROGRAMS = \
elements/audiocheblimit \
elements/audioiirfilter \
elements/audioamplify \
elements/audioreverb \
elements/audiodynamic \
elements/audiowsincband \
elements/audiowsinclimit \
......
/* 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);
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment