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Remove lpwsinc and bpwsinc elements - they've become audiowsinclimit and audiowsincband respectively, in the gst-plug...

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Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* gst/filter/Makefile.am:
* gst/filter/filter.vcproj:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstfilter.c:
* gst/filter/gstfilter.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
* tests/check/Makefile.am:
* tests/check/elements/bpwsinc.c:
* tests/check/elements/lpwsinc.c:
Remove lpwsinc and bpwsinc elements - they've become
audiowsinclimit and audiowsincband respectively, in the
gst-plugins-good audiofx plugin.
This commit is contained in:
Jan Schmidt 2008-02-07 21:53:39 +00:00
parent 37915fa611
commit 9749d146c6
17 changed files with 26 additions and 3679 deletions
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23
ChangeLog
View file

@ -1,3 +1,26 @@
2008-02-07 Jan Schmidt <jan.schmidt@sun.com>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* gst/filter/Makefile.am:
* gst/filter/filter.vcproj:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstfilter.c:
* gst/filter/gstfilter.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
* tests/check/Makefile.am:
* tests/check/elements/bpwsinc.c:
* tests/check/elements/lpwsinc.c:
Remove lpwsinc and bpwsinc elements - they've become
audiowsinclimit and audiowsincband respectively, in the
gst-plugins-good audiofx plugin.
2008-02-07 Sebastien Moutte <sebastien@moutte.net>
* ext\neon\gstneonhttpsrc.c:

2
docs/plugins/Makefile.am
View file

@ -122,8 +122,6 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/equalizer/gstiirequalizer10bands.h \
$(top_srcdir)/gst/equalizer/gstiirequalizernbands.h \
$(top_srcdir)/gst/festival/gstfestival.h \
$(top_srcdir)/gst/filter/gstlpwsinc.h \
$(top_srcdir)/gst/filter/gstbpwsinc.h \
$(top_srcdir)/gst/modplug/gstmodplug.h \
$(top_srcdir)/gst/multifile/gstmultifilesink.h \
$(top_srcdir)/gst/multifile/gstmultifilesrc.h \

2
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -16,7 +16,6 @@
<xi:include href="xml/element-amrwbenc.xml" />
<xi:include href="xml/element-amrwbparse.xml" />
<xi:include href="xml/element-audioparse.xml" />
<xi:include href="xml/element-bpwsinc.xml" />
<xi:include href="xml/element-dfb-example.xml" />
<xi:include href="xml/element-dfbvideosink.xml" />
<xi:include href="xml/element-dvbsrc.xml" />
@ -38,7 +37,6 @@
<xi:include href="xml/element-input-selector.xml" />
<xi:include href="xml/element-ivorbisdec.xml" />
<xi:include href="xml/element-jackaudiosink.xml" />
<xi:include href="xml/element-lpwsinc.xml" />
<xi:include href="xml/element-metadatademux.xml" />
<xi:include href="xml/element-metadatamux.xml" />
<xi:include href="xml/element-modplug.xml" />

30
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -280,36 +280,6 @@ GST_TYPE_SOUP_HTTP_SRC
gst_soup_http_src_get_type
</SECTION>
<SECTION>
<FILE>element-bpwsinc</FILE>
<TITLE>bpwsinc</TITLE>
GstBPWSinc
<SUBSECTION Standard>
GstBPWSincClass
GstBPWSincProcessFunc
GST_BPWSINC
GST_BPWSINC_CLASS
GST_IS_BPWSINC
GST_IS_BPWSINC_CLASS
GST_TYPE_BPWSINC
gst_bpwsinc_get_type
</SECTION>
<SECTION>
<FILE>element-lpwsinc</FILE>
<TITLE>lpwsinc</TITLE>
GstLPWSinc
<SUBSECTION Standard>
GstLPWSincClass
GstLPWSincProcessFunc
GST_LPWSINC
GST_LPWSINC_CLASS
GST_IS_LPWSINC
GST_IS_LPWSINC_CLASS
GST_TYPE_LPWSINC
gst_lpwsinc_get_type
</SECTION>
<SECTION>
<FILE>element-input-selector</FILE>
<TITLE>input-selector</TITLE>

100
docs/plugins/gst-plugins-bad-plugins.args
View file

@ -1518,106 +1518,6 @@
<DEFAULT>20</DEFAULT>
</ARG>
<ARG>
<NAME>GstBPWSinc::length</NAME>
<TYPE>gint</TYPE>
<RANGE>[3,50000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
<DEFAULT>101</DEFAULT>
</ARG>
<ARG>
<NAME>GstBPWSinc::lower-frequency</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Lower Frequency</NICK>
<BLURB>Cut-off lower frequency (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstBPWSinc::upper-frequency</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Upper Frequency</NICK>
<BLURB>Cut-off upper frequency (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstBPWSinc::mode</NAME>
<TYPE>GstBPWSincMode</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Band pass or band reject mode.</BLURB>
<DEFAULT>Band pass (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstBPWSinc::window</NAME>
<TYPE>GstBPWSincWindow</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Window</NICK>
<BLURB>Window function to use.</BLURB>
<DEFAULT>Hamming window (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstLPWSinc::frequency</NAME>
<TYPE>gdouble</TYPE>
<RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Frequency</NICK>
<BLURB>Cut-off Frequency (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstLPWSinc::length</NAME>
<TYPE>gint</TYPE>
<RANGE>[3,50000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
<DEFAULT>101</DEFAULT>
</ARG>
<ARG>
<NAME>GstLPWSinc::mode</NAME>
<TYPE>GstLPWSincMode</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Low pass or high pass mode.</BLURB>
<DEFAULT>Low pass (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstLPWSinc::window</NAME>
<TYPE>GstLPWSincWindow</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Window</NICK>
<BLURB>Window function to use.</BLURB>
<DEFAULT>Hamming window (default)</DEFAULT>
</ARG>
<ARG>
<NAME>GstLPWSinc::cutoff</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,100000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Cutoff</NICK>
<BLURB>Cut-off Frequency (Hz).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstIIR::A</NAME>
<TYPE>gdouble</TYPE>

2
docs/plugins/gst-plugins-bad-plugins.hierarchy
View file

@ -58,8 +58,6 @@ GObject
GstIirEqualizer3Bands
GstIirEqualizer10Bands
GstStereo
GstLPWSinc
GstBPWSinc
GstGLUpload
GstGLDownload
GstGLFilter

4
gst/filter/Makefile.am
View file

@ -1,7 +1,7 @@
plugin_LTLIBRARIES = libgstfilter.la
libgstfilter_la_SOURCES = gstfilter.c gstlpwsinc.c gstbpwsinc.c gstiir.c iir.c
libgstfilter_la_SOURCES = gstfilter.c gstiir.c iir.c
libgstfilter_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(GST_CONTROLLER_CFLAGS)
libgstfilter_la_LIBADD = \
$(GST_BASE_LIBS) \
@ -13,4 +13,4 @@ libgstfilter_la_LIBADD = \
libgstfilter_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstfilter.h gstlpwsinc.h gstbpwsinc.h gstiir.h iir.h
noinst_HEADERS = gstfilter.h gstiir.h iir.h

6
gst/filter/filter.vcproj
View file

@ -134,12 +134,6 @@
<File
RelativePath=".\iir.c">
</File>
<File
RelativePath=".\gstlpwsinc.c">
</File>
<File
RelativePath=".\gstbpwsinc.c">
</File>
</Filter>
<Filter
Name="Header Files"

863
gst/filter/gstbpwsinc.c
View file

@ -1,863 +0,0 @@
/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
* TODO: - Implement the convolution in place, probably only makes sense
* when using FFT convolution as currently the convolution itself
* is probably the bottleneck
* - Maybe allow cascading the filter to get a better stopband attenuation.
* Can be done by convolving a filter kernel with itself
* - Drop the first kernel_length/2 samples and append the same number of
* samples on EOS as the first few samples are essentialy zero.
*/
/**
* SECTION:element-bpwsinc
* @short_description: Windowed Sinc band pass and band reject filter
*
* <refsect2>
* <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
* </para>
* <para>
* This element has the advantage over the Chebyshev bandpass and bandreject filter that it has
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! bpwsinc mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! bpwsinc mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! bpwsinc mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "gstbpwsinc.h"
#define GST_CAT_DEFAULT gst_bpwsinc_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails bpwsinc_details =
GST_ELEMENT_DETAILS ("Band-pass and Band-reject Windowed sinc filter",
"Filter/Effect/Audio",
"Band-pass Windowed sinc filter",
"Thomas <thomas@apestaart.org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LENGTH,
PROP_LOWER_FREQUENCY,
PROP_UPPER_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_BAND_PASS = 0,
MODE_BAND_REJECT
};
#define GST_TYPE_BPWSINC_MODE (gst_bpwsinc_mode_get_type ())
static GType
gst_bpwsinc_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_BAND_PASS, "Band pass (default)",
"band-pass"},
{MODE_BAND_REJECT, "Band reject",
"band-reject"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstBPWSincMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN
};
#define GST_TYPE_BPWSINC_WINDOW (gst_bpwsinc_window_get_type ())
static GType
gst_bpwsinc_window_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{WINDOW_HAMMING, "Hamming window (default)",
"hamming"},
{WINDOW_BLACKMAN, "Blackman window",
"blackman"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstBPWSincWindow", values);
}
return gtype;
}
#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_bpwsinc_debug, "bpwsinc", 0, "Band-pass and Band-reject Windowed sinc filter plugin");
GST_BOILERPLATE_FULL (GstBPWSinc, gst_bpwsinc, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void bpwsinc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void bpwsinc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn bpwsinc_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean bpwsinc_start (GstBaseTransform * base);
static gboolean bpwsinc_event (GstBaseTransform * base, GstEvent * event);
static gboolean bpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
static gboolean bpwsinc_query (GstPad * pad, GstQuery * query);
static const GstQueryType *bpwsinc_query_type (GstPad * pad);
/* Element class */
static void
gst_bpwsinc_dispose (GObject * object)
{
GstBPWSinc *self = GST_BPWSINC (object);
if (self->residue) {
g_free (self->residue);
self->residue = NULL;
}
if (self->kernel) {
g_free (self->kernel);
self->kernel = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_bpwsinc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &bpwsinc_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_bpwsinc_class_init (GstBPWSincClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = bpwsinc_set_property;
gobject_class->get_property = bpwsinc_get_property;
gobject_class->dispose = gst_bpwsinc_dispose;
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider */
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_float ("lower-frequency", "Lower Frequency",
"Cut-off lower frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_float ("upper-frequency", "Upper Frequency",
"Cut-off upper frequency (Hz)", 0.0, 100000.0, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
3, 50000, 101, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Band pass or band reject mode", GST_TYPE_BPWSINC_MODE,
MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_BPWSINC_WINDOW,
WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
trans_class->transform = GST_DEBUG_FUNCPTR (bpwsinc_transform);
trans_class->start = GST_DEBUG_FUNCPTR (bpwsinc_start);
trans_class->event = GST_DEBUG_FUNCPTR (bpwsinc_event);
filter_class->setup = GST_DEBUG_FUNCPTR (bpwsinc_setup);
}
static void
gst_bpwsinc_init (GstBPWSinc * self, GstBPWSincClass * g_class)
{
self->kernel_length = 101;
self->latency = 50;
self->lower_frequency = 0.0;
self->upper_frequency = 0.0;
self->mode = MODE_BAND_PASS;
self->window = WINDOW_HAMMING;
self->kernel = NULL;
self->have_kernel = FALSE;
self->residue = NULL;
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, bpwsinc_query);
gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
bpwsinc_query_type);
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstBPWSinc * self, g##ctype * src, g##ctype * dst, guint input_samples) \
{ \
gint kernel_length = self->kernel_length; \
gint i, j, k, l; \
gint channels = GST_AUDIO_FILTER (self)->format.channels; \
gint res_start; \
\
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
k = i % channels; \
l = i / channels; \
for (j = 0; j < kernel_length; j++) \
if (l < j) \
dst[i] += \
self->residue[(kernel_length + l - j) * channels + \
k] * self->kernel[j]; \
else \
dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
} \
\
/* copy the tail of the current input buffer to the residue, while \
* keeping parts of the residue if the input buffer is smaller than \
* the kernel length */ \
if (input_samples < kernel_length * channels) \
res_start = kernel_length * channels - input_samples; \
else \
res_start = 0; \
\
for (i = 0; i < res_start; i++) \
self->residue[i] = self->residue[i + input_samples]; \
for (i = res_start; i < kernel_length * channels; i++) \
self->residue[i] = src[input_samples - kernel_length * channels + i]; \
\
self->residue_length += kernel_length * channels - res_start; \
if (self->residue_length > kernel_length * channels) \
self->residue_length = kernel_length * channels; \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
static void
bpwsinc_build_kernel (GstBPWSinc * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble *kernel_lp, *kernel_hp;
gdouble w;
len = self->kernel_length;
if (GST_AUDIO_FILTER (self)->format.rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (GST_AUDIO_FILTER (self)->format.channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp frequencies */
self->lower_frequency =
CLAMP (self->lower_frequency, 0.0,
GST_AUDIO_FILTER (self)->format.rate / 2);
self->upper_frequency =
CLAMP (self->upper_frequency, 0.0,
GST_AUDIO_FILTER (self)->format.rate / 2);
if (self->lower_frequency > self->upper_frequency) {
gint tmp = self->lower_frequency;
self->lower_frequency = self->upper_frequency;
self->upper_frequency = tmp;
}
GST_DEBUG ("bpwsinc: initializing filter kernel of length %d "
"with lower frequency %.2lf Hz "
", upper frequency %.2lf Hz for mode %s",
len, self->lower_frequency, self->upper_frequency,
(self->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject");
/* fill the lp kernel */
w = 2 * M_PI * (self->lower_frequency / GST_AUDIO_FILTER (self)->format.rate);
kernel_lp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
kernel_lp[i] = w;
else
kernel_lp[i] = sin (w * (i - len / 2))
/ (i - len / 2);
/* Windowing */
if (self->window == WINDOW_HAMMING)
kernel_lp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
kernel_lp[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_lp[i];
for (i = 0; i < len; ++i)
kernel_lp[i] /= sum;
/* fill the hp kernel */
w = 2 * M_PI * (self->upper_frequency / GST_AUDIO_FILTER (self)->format.rate);
kernel_hp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
kernel_hp[i] = w;
else
kernel_hp[i] = sin (w * (i - len / 2))
/ (i - len / 2);
/* Windowing */
if (self->window == WINDOW_HAMMING)
kernel_hp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
kernel_hp[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_hp[i];
for (i = 0; i < len; ++i)
kernel_hp[i] /= sum;
/* do spectral inversion to go from lowpass to highpass */
for (i = 0; i < len; ++i)
kernel_hp[i] = -kernel_hp[i];
kernel_hp[len / 2] += 1;
/* combine the two kernels */
if (self->kernel)
g_free (self->kernel);
self->kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i)
self->kernel[i] = kernel_lp[i] + kernel_hp[i];
/* free the helper kernels */
g_free (kernel_lp);
g_free (kernel_hp);
/* do spectral inversion to go from bandreject to bandpass
* if specified */
if (self->mode == MODE_BAND_PASS) {
for (i = 0; i < len; ++i)
self->kernel[i] = -self->kernel[i];
self->kernel[len / 2] += 1;
}
/* set up the residue memory space */
if (!self->residue) {
self->residue =
g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->residue_length = 0;
}
self->have_kernel = TRUE;
}
static void
bpwsinc_push_residue (GstBPWSinc * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint outsize, outsamples;
gint diffsize, diffsamples;
guint8 *in, *out;
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = MIN (self->latency, self->residue_length / channels);
outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (outsize == 0)
return;
/* Process the difference between latency and residue_length samples
* to start at the actual data instead of starting at the zeros before
* when we only got one buffer smaller than latency */
diffsamples = self->latency - self->residue_length / channels;
diffsize =
diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (diffsize > 0) {
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->process (self, in, out, diffsamples * channels);
g_free (in);
g_free (out);
}
res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
return;
}
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
g_free (in);
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (outsamples, GST_SECOND, rate);
self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
}
GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), outsamples);
res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
bpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstBPWSinc *self = GST_BPWSINC (base);
gboolean ret = TRUE;
if (format->width == 32)
self->process = (GstBPWSincProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstBPWSincProcessFunc) process_64;
else
ret = FALSE;
self->have_kernel = FALSE;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
bpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstBPWSinc *self = GST_BPWSINC (base);
GstClockTime timestamp;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
gint output_samples = input_samples;
gint diff;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
if (!self->have_kernel)
bpwsinc_build_kernel (self);
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf)) {
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
}
/* Calculate the number of samples we can push out now without outputting
* kernel_length/2 zeros in the beginning */
diff = (self->kernel_length / 2) * channels - self->residue_length;
if (diff > 0)
output_samples -= diff;
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
if (output_samples <= 0) {
/* Drop buffer and save original timestamp/offset for later use */
if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)
&& GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf);
if (self->next_off == GST_BUFFER_OFFSET_NONE
&& GST_BUFFER_OFFSET_IS_VALID (outbuf))
self->next_off = GST_BUFFER_OFFSET (outbuf);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
} else if (output_samples < input_samples) {
/* First (probably partial) buffer after starting from
* a clean residue. Use stored timestamp/offset here */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) =
self->next_off + output_samples / channels;
} else {
/* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) -= diff / channels;
}
if (GST_BUFFER_DURATION_IS_VALID (outbuf))
GST_BUFFER_DURATION (outbuf) -=
gst_util_uint64_scale (diff, GST_SECOND, channels * rate);
GST_BUFFER_DATA (outbuf) +=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
GST_BUFFER_SIZE (outbuf) -=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
} else {
GstClockTime ts_latency =
gst_util_uint64_scale (self->latency, GST_SECOND, rate);
/* Normal buffer, adjust timestamp/offset/etc by latency */
if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) {
GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency");
GST_BUFFER_TIMESTAMP (outbuf) = 0;
} else {
GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency;
}
if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET (outbuf) > self->latency) {
GST_BUFFER_OFFSET (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency");
GST_BUFFER_OFFSET (outbuf) = 0;
}
}
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) {
GST_BUFFER_OFFSET_END (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency");
GST_BUFFER_OFFSET_END (outbuf) = 0;
}
}
}
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
return GST_FLOW_OK;
}
static gboolean
bpwsinc_start (GstBaseTransform * base)
{
GstBPWSinc *self = GST_BPWSINC (base);
gint channels = GST_AUDIO_FILTER (self)->format.channels;
/* Reset the residue if already existing */
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
return TRUE;
}
static gboolean
bpwsinc_query (GstPad * pad, GstQuery * query)
{
GstBPWSinc *self = GST_BPWSINC (gst_pad_get_parent (pad));
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG_OBJECT (self, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
latency =
(rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND,
rate) : 0;
GST_DEBUG_OBJECT (self, "Our latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (self);
return res;
}
static const GstQueryType *
bpwsinc_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static gboolean
bpwsinc_event (GstBaseTransform * base, GstEvent * event)
{
GstBPWSinc *self = GST_BPWSINC (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
bpwsinc_push_residue (self);
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
}
static void
bpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstBPWSinc *self = GST_BPWSINC (object);
g_return_if_fail (GST_IS_BPWSINC (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
GST_BASE_TRANSFORM_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
if (self->residue) {
bpwsinc_push_residue (self);
g_free (self->residue);
self->residue = NULL;
}
self->kernel_length = val;
self->latency = val / 2;
bpwsinc_build_kernel (self);
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
case PROP_LOWER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->lower_frequency = g_value_get_float (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_UPPER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->upper_frequency = g_value_get_float (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (self);
self->mode = g_value_get_enum (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_WINDOW:
GST_BASE_TRANSFORM_LOCK (self);
self->window = g_value_get_enum (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
bpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBPWSinc *self = GST_BPWSINC (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_LOWER_FREQUENCY:
g_value_set_float (value, self->lower_frequency);
break;
case PROP_UPPER_FREQUENCY:
g_value_set_float (value, self->upper_frequency);
break;
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_WINDOW:
g_value_set_enum (value, self->window);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}

88
gst/filter/gstbpwsinc.h
View file

@ -1,88 +0,0 @@
/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
*
* 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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
*/
#ifndef __GST_BPWSINC_H__
#define __GST_BPWSINC_H__
#include "gstfilter.h"
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS
#define GST_TYPE_BPWSINC \
(gst_bpwsinc_get_type())
#define GST_BPWSINC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BPWSINC,GstBPWSinc))
#define GST_BPWSINC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BPWSINC,GstBPWSincClass))
#define GST_IS_BPWSINC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BPWSINC))
#define GST_IS_BPWSINC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BPWSINC))
typedef struct _GstBPWSinc GstBPWSinc;
typedef struct _GstBPWSincClass GstBPWSincClass;
typedef void (*GstBPWSincProcessFunc) (GstBPWSinc *, guint8 *, guint8 *, guint);
/**
* GstBPWSinc:
*
* Opaque data structure.
*/
struct _GstBPWSinc {
GstAudioFilter element;
/* < private > */
GstBPWSincProcessFunc process;
gint mode;
gint window;
gfloat lower_frequency, upper_frequency;
gint kernel_length; /* length of the filter kernel */
gdouble *residue; /* buffer for left-over samples from previous buffer */
gdouble *kernel;
gboolean have_kernel;
gint residue_length;
guint64 latency;
GstClockTime next_ts;
guint64 next_off;
};
struct _GstBPWSincClass {
GstAudioFilterClass parent_class;
};
GType gst_bpwsinc_get_type (void);
G_END_DECLS
#endif /* __GST_BPWSINC_H__ */

4
gst/filter/gstfilter.c
View file

@ -38,8 +38,6 @@ struct _elements_entry
static struct _elements_entry _elements[] = {
{"iir", gst_iir_get_type},
{"lpwsinc", gst_lpwsinc_get_type},
{"bpwsinc", gst_bpwsinc_get_type},
{NULL, 0},
};
@ -65,5 +63,5 @@ plugin_init (GstPlugin * plugin)
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"filter",
"IIR, lpwsinc and bpwsinc audio filter elements",
"IIR audio filter element",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);

2
gst/filter/gstfilter.h
View file

@ -29,7 +29,5 @@
#include <gst/gst.h>
GType gst_iir_get_type (void);
GType gst_lpwsinc_get_type (void);
GType gst_bpwsinc_get_type (void);
#endif /* __GST_FILTER_H__ */

795
gst/filter/gstlpwsinc.c
View file

@ -1,795 +0,0 @@
/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
* TODO: - Implement the convolution in place, probably only makes sense
* when using FFT convolution as currently the convolution itself
* is probably the bottleneck
* - Maybe allow cascading the filter to get a better stopband attenuation.
* Can be done by convolving a filter kernel with itself
*/
/**
* SECTION:element-lpwsinc
* @short_description: Windowed Sinc low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
* </para>
* <para>
* This element has the advantage over the Chebyshev lowpass and highpass filter that it has
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! lpwsinc mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! lpwsinc mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "gstlpwsinc.h"
#define GST_CAT_DEFAULT gst_lpwsinc_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails lpwsinc_details = GST_ELEMENT_DETAILS ("LPWSinc",
"Filter/Effect/Audio",
"Low-pass and High-pass Windowed sinc filter",
"Thomas <thomas@apestaart.org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LENGTH,
PROP_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_LPWSINC_MODE (gst_lpwsinc_mode_get_type ())
static GType
gst_lpwsinc_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstLPWSincMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN
};
#define GST_TYPE_LPWSINC_WINDOW (gst_lpwsinc_window_get_type ())
static GType
gst_lpwsinc_window_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{WINDOW_HAMMING, "Hamming window (default)",
"hamming"},
{WINDOW_BLACKMAN, "Blackman window",
"blackman"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstLPWSincWindow", values);
}
return gtype;
}
#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_lpwsinc_debug, "lpwsinc", 0, "Low-pass and High-pass Windowed sinc filter plugin");
GST_BOILERPLATE_FULL (GstLPWSinc, gst_lpwsinc, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void lpwsinc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void lpwsinc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn lpwsinc_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean lpwsinc_start (GstBaseTransform * base);
static gboolean lpwsinc_event (GstBaseTransform * base, GstEvent * event);
static gboolean lpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
static gboolean lpwsinc_query (GstPad * pad, GstQuery * query);
static const GstQueryType *lpwsinc_query_type (GstPad * pad);
/* Element class */
static void
gst_lpwsinc_dispose (GObject * object)
{
GstLPWSinc *self = GST_LPWSINC (object);
if (self->residue) {
g_free (self->residue);
self->residue = NULL;
}
if (self->kernel) {
g_free (self->kernel);
self->kernel = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_lpwsinc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &lpwsinc_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_lpwsinc_class_init (GstLPWSincClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = lpwsinc_set_property;
gobject_class->get_property = lpwsinc_get_property;
gobject_class->dispose = gst_lpwsinc_dispose;
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider */
g_object_class_install_property (gobject_class, PROP_FREQUENCY,
g_param_spec_float ("cutoff", "Cutoff",
"Cut-off Frequency (Hz)", 0.0, 100000.0, 0.0,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
3, 50000, 101, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode", GST_TYPE_LPWSINC_MODE,
MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_LPWSINC_WINDOW,
WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
trans_class->transform = GST_DEBUG_FUNCPTR (lpwsinc_transform);
trans_class->start = GST_DEBUG_FUNCPTR (lpwsinc_start);
trans_class->event = GST_DEBUG_FUNCPTR (lpwsinc_event);
filter_class->setup = GST_DEBUG_FUNCPTR (lpwsinc_setup);
}
static void
gst_lpwsinc_init (GstLPWSinc * self, GstLPWSincClass * g_class)
{
self->mode = MODE_LOW_PASS;
self->window = WINDOW_HAMMING;
self->kernel_length = 101;
self->latency = 50;
self->cutoff = 0.0;
self->kernel = NULL;
self->residue = NULL;
self->have_kernel = FALSE;
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad, lpwsinc_query);
gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
lpwsinc_query_type);
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstLPWSinc * self, g##ctype * src, g##ctype * dst, guint input_samples) \
{ \
gint kernel_length = self->kernel_length; \
gint i, j, k, l; \
gint channels = GST_AUDIO_FILTER (self)->format.channels; \
gint res_start; \
\
/* convolution */ \
for (i = 0; i < input_samples; i++) { \
dst[i] = 0.0; \
k = i % channels; \
l = i / channels; \
for (j = 0; j < kernel_length; j++) \
if (l < j) \
dst[i] += \
self->residue[(kernel_length + l - j) * channels + \
k] * self->kernel[j]; \
else \
dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \
} \
\
/* copy the tail of the current input buffer to the residue, while \
* keeping parts of the residue if the input buffer is smaller than \
* the kernel length */ \
if (input_samples < kernel_length * channels) \
res_start = kernel_length * channels - input_samples; \
else \
res_start = 0; \
\
for (i = 0; i < res_start; i++) \
self->residue[i] = self->residue[i + input_samples]; \
for (i = res_start; i < kernel_length * channels; i++) \
self->residue[i] = src[input_samples - kernel_length * channels + i]; \
\
self->residue_length += kernel_length * channels - res_start; \
if (self->residue_length > kernel_length * channels) \
self->residue_length = kernel_length * channels; \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
static void
lpwsinc_build_kernel (GstLPWSinc * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble w;
len = self->kernel_length;
if (GST_AUDIO_FILTER (self)->format.rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (GST_AUDIO_FILTER (self)->format.channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp cutoff frequency between 0 and the nyquist frequency */
self->cutoff =
CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
GST_DEBUG ("lpwsinc: initializing filter kernel of length %d "
"with cutoff %.2lf Hz "
"for mode %s",
len, self->cutoff,
(self->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass");
/* fill the kernel */
w = 2 * M_PI * (self->cutoff / GST_AUDIO_FILTER (self)->format.rate);
if (self->kernel)
g_free (self->kernel);
self->kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
self->kernel[i] = w;
else
self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
/* windowing */
if (self->window == WINDOW_HAMMING)
self->kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
self->kernel[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
for (i = 0; i < len; ++i)
sum += self->kernel[i];
for (i = 0; i < len; ++i)
self->kernel[i] /= sum;
/* convert to highpass if specified */
if (self->mode == MODE_HIGH_PASS) {
for (i = 0; i < len; ++i)
self->kernel[i] = -self->kernel[i];
self->kernel[len / 2] += 1.0;
}
/* set up the residue memory space */
if (!self->residue) {
self->residue =
g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->residue_length = 0;
}
self->have_kernel = TRUE;
}
static void
lpwsinc_push_residue (GstLPWSinc * self)
{
GstBuffer *outbuf;
GstFlowReturn res;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint outsize, outsamples;
gint diffsize, diffsamples;
guint8 *in, *out;
/* Calculate the number of samples and their memory size that
* should be pushed from the residue */
outsamples = MIN (self->latency, self->residue_length / channels);
outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (outsize == 0)
return;
/* Process the difference between latency and residue_length samples
* to start at the actual data instead of starting at the zeros before
* when we only got one buffer smaller than latency */
diffsamples = self->latency - self->residue_length / channels;
diffsize =
diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);
if (diffsize > 0) {
in = g_new0 (guint8, diffsize);
out = g_new0 (guint8, diffsize);
self->process (self, in, out, diffsamples * channels);
g_free (in);
g_free (out);
}
res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,
GST_BUFFER_OFFSET_NONE, outsize,
GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);
return;
}
/* Convolve the residue with zeros to get the actual remaining data */
in = g_new0 (guint8, outsize);
self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);
g_free (in);
/* Set timestamp, offset, etc from the values we
* saved when processing the regular buffers */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
else
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (outsamples, GST_SECOND, rate);
self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;
}
GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), outsamples);
res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);
if (G_UNLIKELY (res != GST_FLOW_OK)) {
GST_WARNING_OBJECT (self, "failed to push residue");
}
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
lpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstLPWSinc *self = GST_LPWSINC (base);
gboolean ret = TRUE;
if (format->width == 32)
self->process = (GstLPWSincProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstLPWSincProcessFunc) process_64;
else
ret = FALSE;
self->have_kernel = FALSE;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
lpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstLPWSinc *self = GST_LPWSINC (base);
GstClockTime timestamp;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
gint output_samples = input_samples;
gint diff;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
if (!self->have_kernel)
lpwsinc_build_kernel (self);
/* Reset the residue if already existing on discont buffers */
if (GST_BUFFER_IS_DISCONT (inbuf)) {
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
}
/* Calculate the number of samples we can push out now without outputting
* kernel_length/2 zeros in the beginning */
diff = (self->kernel_length / 2) * channels - self->residue_length;
if (diff > 0)
output_samples -= diff;
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
if (output_samples <= 0) {
/* Drop buffer and save original timestamp/offset for later use */
if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)
&& GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf);
if (self->next_off == GST_BUFFER_OFFSET_NONE
&& GST_BUFFER_OFFSET_IS_VALID (outbuf))
self->next_off = GST_BUFFER_OFFSET (outbuf);
return GST_BASE_TRANSFORM_FLOW_DROPPED;
} else if (output_samples < input_samples) {
/* First (probably partial) buffer after starting from
* a clean residue. Use stored timestamp/offset here */
if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
if (self->next_off != GST_BUFFER_OFFSET_NONE) {
GST_BUFFER_OFFSET (outbuf) = self->next_off;
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) =
self->next_off + output_samples / channels;
} else {
/* We dropped no buffer, offset is valid, offset_end must be adjusted by diff */
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf))
GST_BUFFER_OFFSET_END (outbuf) -= diff / channels;
}
if (GST_BUFFER_DURATION_IS_VALID (outbuf))
GST_BUFFER_DURATION (outbuf) -=
gst_util_uint64_scale (diff, GST_SECOND, channels * rate);
GST_BUFFER_DATA (outbuf) +=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
GST_BUFFER_SIZE (outbuf) -=
diff * (GST_AUDIO_FILTER (self)->format.width / 8);
} else {
GstClockTime ts_latency =
gst_util_uint64_scale (self->latency, GST_SECOND, rate);
/* Normal buffer, adjust timestamp/offset/etc by latency */
if (GST_BUFFER_TIMESTAMP (outbuf) < ts_latency) {
GST_WARNING_OBJECT (self, "GST_BUFFER_TIMESTAMP (outbuf) < latency");
GST_BUFFER_TIMESTAMP (outbuf) = 0;
} else {
GST_BUFFER_TIMESTAMP (outbuf) -= ts_latency;
}
if (GST_BUFFER_OFFSET_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET (outbuf) > self->latency) {
GST_BUFFER_OFFSET (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET (outbuf) < latency");
GST_BUFFER_OFFSET (outbuf) = 0;
}
}
if (GST_BUFFER_OFFSET_END_IS_VALID (outbuf)) {
if (GST_BUFFER_OFFSET_END (outbuf) > self->latency) {
GST_BUFFER_OFFSET_END (outbuf) -= self->latency;
} else {
GST_WARNING_OBJECT (self, "GST_BUFFER_OFFSET_END (outbuf) < latency");
GST_BUFFER_OFFSET_END (outbuf) = 0;
}
}
}
GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"
" offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);
self->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
self->next_off = GST_BUFFER_OFFSET_END (outbuf);
return GST_FLOW_OK;
}
static gboolean
lpwsinc_start (GstBaseTransform * base)
{
GstLPWSinc *self = GST_LPWSINC (base);
gint channels = GST_AUDIO_FILTER (self)->format.channels;
/* Reset the residue if already existing */
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
self->residue_length = 0;
self->next_ts = GST_CLOCK_TIME_NONE;
self->next_off = GST_BUFFER_OFFSET_NONE;
return TRUE;
}
static gboolean
lpwsinc_query (GstPad * pad, GstQuery * query)
{
GstLPWSinc *self = GST_LPWSINC (gst_pad_get_parent (pad));
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime min, max;
gboolean live;
guint64 latency;
GstPad *peer;
gint rate = GST_AUDIO_FILTER (self)->format.rate;
if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {
if ((res = gst_pad_query (peer, query))) {
gst_query_parse_latency (query, &live, &min, &max);
GST_DEBUG_OBJECT (self, "Peer latency: min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
/* add our own latency */
latency =
(rate != 0) ? gst_util_uint64_scale (self->latency, GST_SECOND,
rate) : 0;
GST_DEBUG_OBJECT (self, "Our latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (latency));
min += latency;
if (max != GST_CLOCK_TIME_NONE)
max += latency;
GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
gst_object_unref (peer);
}
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
gst_object_unref (self);
return res;
}
static const GstQueryType *
lpwsinc_query_type (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_LATENCY,
0
};
return types;
}
static gboolean
lpwsinc_event (GstBaseTransform * base, GstEvent * event)
{
GstLPWSinc *self = GST_LPWSINC (base);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
lpwsinc_push_residue (self);
break;
default:
break;
}
return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
}
static void
lpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
g_return_if_fail (GST_IS_LPWSINC (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
GST_BASE_TRANSFORM_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
if (val != self->kernel_length) {
if (self->residue) {
lpwsinc_push_residue (self);
g_free (self->residue);
self->residue = NULL;
}
self->kernel_length = val;
self->latency = val / 2;
lpwsinc_build_kernel (self);
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_latency (GST_OBJECT (self)));
}
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
case PROP_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->cutoff = g_value_get_float (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (self);
self->mode = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_WINDOW:
GST_BASE_TRANSFORM_LOCK (self);
self->window = g_value_get_enum (value);
lpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
lpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstLPWSinc *self = GST_LPWSINC (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_FREQUENCY:
g_value_set_float (value, self->cutoff);
break;
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_WINDOW:
g_value_set_enum (value, self->window);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}

88
gst/filter/gstlpwsinc.h
View file

@ -1,88 +0,0 @@
/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
*
* 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.
*
*
* this windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
*/
#ifndef __GST_LPWSINC_H__
#define __GST_LPWSINC_H__
#include "gstfilter.h"
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
G_BEGIN_DECLS
#define GST_TYPE_LPWSINC \
(gst_lpwsinc_get_type())
#define GST_LPWSINC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LPWSINC,GstLPWSinc))
#define GST_LPWSINC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LPWSINC,GstLPWSincClass))
#define GST_IS_LPWSINC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LPWSINC))
#define GST_IS_LPWSINC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LPWSINC))
typedef struct _GstLPWSinc GstLPWSinc;
typedef struct _GstLPWSincClass GstLPWSincClass;
typedef void (*GstLPWSincProcessFunc) (GstLPWSinc *, guint8 *, guint8 *, guint);
/**
* GstLPWSinc:
*
* Opaque data structure.
*/
struct _GstLPWSinc {
GstAudioFilter element;
/* < private > */
GstLPWSincProcessFunc process;
gint mode;
gint window;
gfloat cutoff;
gint kernel_length; /* length of the filter kernel */
gdouble *residue; /* buffer for left-over samples from previous buffer */
gdouble *kernel; /* filter kernel */
gboolean have_kernel;
gint residue_length;
guint64 latency;
GstClockTime next_ts;
guint64 next_off;
};
struct _GstLPWSincClass {
GstAudioFilterClass parent_class;
};
GType gst_lpwsinc_get_type (void);
G_END_DECLS
#endif /* __GST_LPWSINC_H__ */

2
tests/check/Makefile.am
View file

@ -74,10 +74,8 @@ check_PROGRAMS = \
$(check_neon) \
$(check_soup) \
$(check_timidity) \
elements/bpwsinc \
elements/equalizer \
elements/interleave \
elements/lpwsinc \
elements/multifile \
elements/rganalysis \
elements/rglimiter \

998
tests/check/elements/bpwsinc.c
View file

@ -1,998 +0,0 @@
/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* bpwsinc.c: Unit test for the bpwsinc element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* 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 BPWSINC_CAPS_STRING_32 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32" \
#define BPWSINC_CAPS_STRING_64 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"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, "
"rate = (int) 44100, "
"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, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 } ")
);
GstElement *
setup_bpwsinc ()
{
GstElement *bpwsinc;
GST_DEBUG ("setup_bpwsinc");
bpwsinc = gst_check_setup_element ("bpwsinc");
mysrcpad = gst_check_setup_src_pad (bpwsinc, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (bpwsinc, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return bpwsinc;
}
void
cleanup_bpwsinc (GstElement * bpwsinc)
{
GST_DEBUG ("cleanup_bpwsinc");
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 (bpwsinc);
gst_check_teardown_sink_pad (bpwsinc);
gst_check_teardown_element (bpwsinc);
}
/* Test if data containing only one frequency component
* at rate/2 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_bp_0hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at the band center is preserved with bandreject mode
* and a 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_bp_11025hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.4);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_bp_22050hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.3);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_br_0hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at the band center is erased with bandreject mode
* and a 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_br_11025hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.35);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_32_br_22050hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_32_small_buffer)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in;
gfloat *res;
gint i;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 101, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency",
44100 / 4.0 - 44100 / 16.0, NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency",
44100 / 4.0 + 44100 / 16.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with bandpass mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_bp_0hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at the band center is preserved with bandreject mode
* and a 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_bp_11025hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.4);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_bp_22050hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.3);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_br_0hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at the band center is erased with bandreject mode
* and a 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_br_11025hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 4) {
in[i] = 0.0;
in[i + 1] = 1.0;
in[i + 2] = 0.0;
in[i + 3] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.35);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with bandreject mode and a
* 2000Hz frequency band around rate/4 */
GST_START_TEST (test_64_br_22050hz)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
bpwsinc = setup_bpwsinc ();
/* Set to bandreject */
g_object_set (G_OBJECT (bpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 31, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency", 44100 / 4.0 - 1000,
NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency", 44100 / 4.0 + 1000,
NULL);
inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 1024; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_64_small_buffer)
{
GstElement *bpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in;
gdouble *res;
gint i;
bpwsinc = setup_bpwsinc ();
/* Set to bandpass */
g_object_set (G_OBJECT (bpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (bpwsinc), "length", 101, NULL);
fail_unless (gst_element_set_state (bpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (bpwsinc), "lower-frequency",
44100 / 4.0 - 44100 / 16.0, NULL);
g_object_set (G_OBJECT (bpwsinc), "upper-frequency",
44100 / 4.0 + 44100 / 16.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (BPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
/* cleanup */
cleanup_bpwsinc (bpwsinc);
}
GST_END_TEST;
Suite *
bpwsinc_suite (void)
{
Suite *s = suite_create ("bpwsinc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_32_bp_0hz);
tcase_add_test (tc_chain, test_32_bp_11025hz);
tcase_add_test (tc_chain, test_32_bp_22050hz);
tcase_add_test (tc_chain, test_32_br_0hz);
tcase_add_test (tc_chain, test_32_br_11025hz);
tcase_add_test (tc_chain, test_32_br_22050hz);
tcase_add_test (tc_chain, test_32_small_buffer);
tcase_add_test (tc_chain, test_64_bp_0hz);
tcase_add_test (tc_chain, test_64_bp_11025hz);
tcase_add_test (tc_chain, test_64_bp_22050hz);
tcase_add_test (tc_chain, test_64_br_0hz);
tcase_add_test (tc_chain, test_64_br_11025hz);
tcase_add_test (tc_chain, test_64_br_22050hz);
tcase_add_test (tc_chain, test_64_small_buffer);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = bpwsinc_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

696
tests/check/elements/lpwsinc.c
View file

@ -1,696 +0,0 @@
/* GStreamer
*
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* lpwsinc.c: Unit test for the lpwsinc element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/check/gstcheck.h>
#include <math.h>
/* 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 LPWSINC_CAPS_STRING_32 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32" \
#define LPWSINC_CAPS_STRING_64 \
"audio/x-raw-float, " \
"channels = (int) 1, " \
"rate = (int) 44100, " \
"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, "
"rate = (int) 44100, "
"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, "
"rate = (int) 44100, "
"endianness = (int) BYTE_ORDER, " "width = (int) { 32, 64 } ")
);
GstElement *
setup_lpwsinc ()
{
GstElement *lpwsinc;
GST_DEBUG ("setup_lpwsinc");
lpwsinc = gst_check_setup_element ("lpwsinc");
mysrcpad = gst_check_setup_src_pad (lpwsinc, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (lpwsinc, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return lpwsinc;
}
void
cleanup_lpwsinc (GstElement * lpwsinc)
{
GST_DEBUG ("cleanup_lpwsinc");
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 (lpwsinc);
gst_check_teardown_sink_pad (lpwsinc);
gst_check_teardown_element (lpwsinc);
}
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_lp_0hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_lp_22050hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_hp_0hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to highpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_32_hp_22050hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to highpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gfloat *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gfloat);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_32_small_buffer)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gfloat *in;
gfloat *res;
gint i;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 101, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
in = (gfloat *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_32);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is preserved with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_lp_0hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
/* cutoff = sampling rate / 4, data = 0 */
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is erased with lowpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_lp_22050hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at 0 is erased with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_hp_0hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to highpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms <= 0.1);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if data containing only one frequency component
* at rate/2 is preserved with highpass mode and a cutoff
* at rate/4 */
GST_START_TEST (test_64_hp_22050hz)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in, *res, rms;
gint i;
GList *node;
lpwsinc = setup_lpwsinc ();
/* Set to highpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 1, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 21, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 128; i += 2) {
in[i] = 1.0;
in[i + 1] = -1.0;
}
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
for (node = buffers; node; node = node->next) {
gint buffer_length;
fail_if ((outbuffer = (GstBuffer *) node->data) == NULL);
res = (gdouble *) GST_BUFFER_DATA (outbuffer);
buffer_length = GST_BUFFER_SIZE (outbuffer) / sizeof (gdouble);
rms = 0.0;
for (i = 0; i < buffer_length; i++)
rms += res[i] * res[i];
rms = sqrt (rms / buffer_length);
fail_unless (rms >= 0.9);
}
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
/* Test if buffers smaller than the kernel size are handled
* correctly without accessing wrong memory areas */
GST_START_TEST (test_64_small_buffer)
{
GstElement *lpwsinc;
GstBuffer *inbuffer, *outbuffer;
GstCaps *caps;
gdouble *in;
gdouble *res;
gint i;
lpwsinc = setup_lpwsinc ();
/* Set to lowpass */
g_object_set (G_OBJECT (lpwsinc), "mode", 0, NULL);
g_object_set (G_OBJECT (lpwsinc), "length", 101, NULL);
fail_unless (gst_element_set_state (lpwsinc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_object_set (G_OBJECT (lpwsinc), "cutoff", 44100 / 4.0, NULL);
inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
in = (gdouble *) GST_BUFFER_DATA (inbuffer);
for (i = 0; i < 20; i++)
in[i] = 1.0;
caps = gst_caps_from_string (LPWSINC_CAPS_STRING_64);
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);
fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()));
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) >= 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
/* cleanup */
cleanup_lpwsinc (lpwsinc);
}
GST_END_TEST;
Suite *
lpwsinc_suite (void)
{
Suite *s = suite_create ("lpwsinc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_32_lp_0hz);
tcase_add_test (tc_chain, test_32_lp_22050hz);
tcase_add_test (tc_chain, test_32_hp_0hz);
tcase_add_test (tc_chain, test_32_hp_22050hz);
tcase_add_test (tc_chain, test_32_small_buffer);
tcase_add_test (tc_chain, test_64_lp_0hz);
tcase_add_test (tc_chain, test_64_lp_22050hz);
tcase_add_test (tc_chain, test_64_hp_0hz);
tcase_add_test (tc_chain, test_64_hp_22050hz);
tcase_add_test (tc_chain, test_64_small_buffer);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = lpwsinc_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}