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-rw-r--r--gst/filter/Makefile.am4
-rw-r--r--gst/filter/filter.vcproj6
-rw-r--r--gst/filter/gstbpwsinc.c863
-rw-r--r--gst/filter/gstbpwsinc.h88
-rw-r--r--gst/filter/gstfilter.c4
-rw-r--r--gst/filter/gstfilter.h2
-rw-r--r--gst/filter/gstlpwsinc.c795
-rw-r--r--gst/filter/gstlpwsinc.h88
8 files changed, 3 insertions, 1847 deletions
diff --git a/gst/filter/Makefile.am b/gst/filter/Makefile.am
index e4f27784..35132517 100644
--- a/gst/filter/Makefile.am
+++ b/gst/filter/Makefile.am
@@ -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
diff --git a/gst/filter/filter.vcproj b/gst/filter/filter.vcproj
index 59434e85..5b8f3124 100644
--- a/gst/filter/filter.vcproj
+++ b/gst/filter/filter.vcproj
@@ -134,12 +134,6 @@
<File
RelativePath=".\iir.c">
</File>
- <File
- RelativePath=".\gstlpwsinc.c">
- </File>
- <File
- RelativePath=".\gstbpwsinc.c">
- </File>
</Filter>
<Filter
Name="Header Files"
diff --git a/gst/filter/gstbpwsinc.c b/gst/filter/gstbpwsinc.c
deleted file mode 100644
index 929daec6..00000000
--- a/gst/filter/gstbpwsinc.c
+++ /dev/null
@@ -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;
- }
-}
diff --git a/gst/filter/gstbpwsinc.h b/gst/filter/gstbpwsinc.h
deleted file mode 100644
index 24a4d2da..00000000
--- a/gst/filter/gstbpwsinc.h
+++ /dev/null
@@ -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__ */
diff --git a/gst/filter/gstfilter.c b/gst/filter/gstfilter.c
index 36c37ec4..11c43cc5 100644
--- a/gst/filter/gstfilter.c
+++ b/gst/filter/gstfilter.c
@@ -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);
diff --git a/gst/filter/gstfilter.h b/gst/filter/gstfilter.h
index d8e43cb0..d676fa3b 100644
--- a/gst/filter/gstfilter.h
+++ b/gst/filter/gstfilter.h
@@ -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__ */
diff --git a/gst/filter/gstlpwsinc.c b/gst/filter/gstlpwsinc.c
deleted file mode 100644
index 7189aaa7..00000000
--- a/gst/filter/gstlpwsinc.c
+++ /dev/null
@@ -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;
- }
-}
diff --git a/gst/filter/gstlpwsinc.h b/gst/filter/gstlpwsinc.h
deleted file mode 100644
index f56f5a4d..00000000
--- a/gst/filter/gstlpwsinc.h
+++ /dev/null
@@ -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__ */