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

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.

1 files changed, 0 insertions, 795 deletions

diff --git a/gst/filter/gstlpwsinc.c b/gst/filter/gstlpwsinc.c
deleted file mode 100644
index 7189aaa78..000000000
--- 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;
-  }
-}