+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),
$(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 \
<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" />
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>
<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>
+
GstAudioWSincLimit
GstAudioWSincBand
GstAudioFIRFilter
+ GstAudioReverb
GstIirEqualizer
GstIirEqualizerNBands
GstIirEqualizer3Bands
GstRegistry
GstRingBuffer
GstSignalObject
+ GstMixerTrack
+ GstMixerOptions
GstCmmlTagStream
GstCmmlTagHead
GstCmmlTagClip
</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 <sebastian.droege@collabora.co.uk></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 & band reject filter</longname>
<longname>Spectrum analyzer</longname>
<class>Filter/Analyzer/Audio</class>
<description>Run an FFT on the audio signal, output spectrum data</description>
- <author>Erik Walthinsen <omega@cse.ogi.edu>, Stefan Kost <ensonic@users.sf.net>, Sebastian Dröge <slomo@circular-chaos.org></author>
+ <author>Erik Walthinsen <omega@cse.ogi.edu>, Stefan Kost <ensonic@users.sf.net>, Sebastian Dröge <sebastian.droege@collabora.co.uk></author>
<pads>
<caps>
<name>sink</name>
audiofxbasefirfilter.c \
audiowsincband.c \
audiowsinclimit.c \
- audiofirfilter.c
+ audiofirfilter.c \
+ audioreverb.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
audiowsincband.h \
audiowsinclimit.h \
audiofirfilter.h \
+ audioreverb.h \
math_compat.h
#include "audiowsincband.h"
#include "audiowsinclimit.h"
#include "audiofirfilter.h"
+#include "audioreverb.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
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,
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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__ */
elements/audiocheblimit \
elements/audioiirfilter \
elements/audioamplify \
+ elements/audioreverb \
elements/audiodynamic \
elements/audiowsincband \
elements/audiowsinclimit \
--- /dev/null
+/* 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);
projects / platform / upstream / gst-plugins-good.git / commitdiff