gst/audiofx/: Implement a base class for generic audio FIR filters.

Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.

11 files changed, 831 insertions, 1034 deletions

diff --git a/ChangeLog b/ChangeLog
index 120c3e3e..12343bdf 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,46 @@
+2009-01-11  Sebastian Dröge  <sebastian.droege@collabora.co.uk>
+
+	* gst/audiofx/Makefile.am:
+	* gst/audiofx/audiofxbasefirfilter.c:
+	(gst_audio_fx_base_fir_filter_dispose),
+	(gst_audio_fx_base_fir_filter_base_init),
+	(gst_audio_fx_base_fir_filter_class_init),
+	(gst_audio_fx_base_fir_filter_init),
+	(gst_audio_fx_base_fir_filter_push_residue),
+	(gst_audio_fx_base_fir_filter_setup),
+	(gst_audio_fx_base_fir_filter_transform),
+	(gst_audio_fx_base_fir_filter_start),
+	(gst_audio_fx_base_fir_filter_stop),
+	(gst_audio_fx_base_fir_filter_query),
+	(gst_audio_fx_base_fir_filter_query_type),
+	(gst_audio_fx_base_fir_filter_event),
+	(gst_audio_fx_base_fir_filter_set_kernel):
+	* gst/audiofx/audiofxbasefirfilter.h:
+	* gst/audiofx/audiofxbaseiirfilter.c:
+	Implement a base class for generic audio FIR filters.
+
+	* gst/audiofx/audiowsincband.c:
+	(gst_gst_audio_wsincband_mode_get_type),
+	(gst_gst_audio_wsincband_window_get_type),
+	(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
+	(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
+	(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
+	(gst_audio_wsincband_get_property):
+	* gst/audiofx/audiowsincband.h:
+	* gst/audiofx/audiowsinclimit.c:
+	(gst_audio_wsinclimit_mode_get_type),
+	(gst_audio_wsinclimit_window_get_type),
+	(gst_audio_wsinclimit_base_init),
+	(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
+	(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
+	(gst_audio_wsinclimit_set_property),
+	(gst_audio_wsinclimit_get_property):
+	* gst/audiofx/audiowsinclimit.h:
+	* tests/check/elements/audiowsincband.c: (GST_START_TEST):
+	* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
+	Use this new base class for audiowsincband and audiowsinclimit.
+	Also cleanup both elements.
+
 2009-01-08  Michael Smith <msmith@songbirdnest.com>
 
 	* gst/qtdemux/qtdemux.c:
diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am
index ac6439bb..d93d3e9a 100644
--- a/gst/audiofx/Makefile.am
+++ b/gst/audiofx/Makefile.am
@@ -12,6 +12,7 @@ libgstaudiofx_la_SOURCES = audiofx.c\
 	audiofxbaseiirfilter.c \
 	audiocheblimit.c \
 	audiochebband.c \
+	audiofxbasefirfilter.c \
 	audiowsincband.c \
 	audiowsinclimit.c
 
@@ -38,6 +39,7 @@ noinst_HEADERS = audiopanorama.h \
 	audiofxbaseiirfilter.h \
 	audiocheblimit.h \
 	audiochebband.h \
+	audiofxbasefirfilter.h \
 	audiowsincband.h \
         audiowsinclimit.h \
 	math_compat.h
diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c
new file mode 100644
index 00000000..059c2aa3
--- /dev/null
+++ b/gst/audiofx/audiofxbasefirfilter.c
@@ -0,0 +1,527 @@
+/* -*- c-basic-offset: 2 -*-
+ * 
+ * GStreamer
+ * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
+ *               2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ *               2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ * 
+ * 
+ * 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
+ */
+
+#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 "audiofxbasefirfilter.h"
+
+#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float, "                                             \
+    " width = (int) { 32, 64 }, "                                     \
+    " endianness = (int) BYTE_ORDER, "                                \
+    " rate = (int) [ 1, MAX ], "                                      \
+    " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \
+      "FIR filter base class");
+
+GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
+    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
+    base, GstBuffer * inbuf, GstBuffer * outbuf);
+static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);
+static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
+static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,
+    GstEvent * event);
+static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format);
+
+static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,
+    GstQuery * query);
+static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *
+    pad);
+
+/* Element class */
+
+static void
+gst_audio_fx_base_fir_filter_dispose (GObject * object)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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_audio_fx_base_fir_filter_base_init (gpointer g_class)
+{
+  GstCaps *caps;
+
+  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_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
+  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;
+
+  trans_class->transform =
+      GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
+  trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);
+  trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);
+}
+
+static void
+gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
+    GstAudioFXBaseFIRFilterClass * g_class)
+{
+  self->kernel = NULL;
+  self->residue = NULL;
+
+  self->next_ts = GST_CLOCK_TIME_NONE;
+  self->next_off = GST_BUFFER_OFFSET_NONE;
+
+  gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,
+      gst_audio_fx_base_fir_filter_query);
+  gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
+      gst_audio_fx_base_fir_filter_query_type);
+}
+
+#define DEFINE_PROCESS_FUNC(width,ctype) \
+static void \
+process_##width (GstAudioFXBaseFIRFilter * 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
+
+void
+gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * 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;
+
+  if (channels == 0 || rate == 0) {
+    self->residue_length = 0;
+    return;
+  }
+
+  /* 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) {
+    self->residue_length = 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);
+    self->residue_length = 0;
+    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;
+    self->next_off = GST_BUFFER_OFFSET_END (outbuf);
+  }
+
+  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");
+  }
+
+  self->residue_length = 0;
+}
+
+/* GstAudioFilter vmethod implementations */
+
+/* get notified of caps and plug in the correct process function */
+static gboolean
+gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+  gboolean ret = TRUE;
+
+  if (self->residue) {
+    gst_audio_fx_base_fir_filter_push_residue (self);
+    g_free (self->residue);
+    self->residue = NULL;
+    self->residue_length = 0;
+    self->next_ts = GST_CLOCK_TIME_NONE;
+    self->next_off = GST_BUFFER_OFFSET_NONE;
+  }
+
+  if (format->width == 32)
+    self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;
+  else if (format->width == 64)
+    self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;
+  else
+    ret = FALSE;
+
+  return TRUE;
+}
+
+/* GstBaseTransform vmethod implementations */
+
+static GstFlowReturn
+gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
+    GstBuffer * inbuf, GstBuffer * outbuf)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (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 = 0;
+
+  timestamp = GST_BUFFER_TIMESTAMP (outbuf);
+  if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
+    GST_ERROR_OBJECT (self, "Invalid timestamp");
+    return GST_FLOW_ERROR;
+  }
+
+  gst_object_sync_values (G_OBJECT (self), timestamp);
+
+  g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);
+  g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);
+
+  if (!self->residue)
+    self->residue = g_new0 (gdouble, self->kernel_length * channels);
+
+  /* Reset the residue if already existing on discont buffers */
+  if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)
+          && timestamp - gst_util_uint64_scale (MIN (self->latency,
+                  self->residue_length / channels), GST_SECOND,
+              rate) - self->next_ts > 5 * GST_MSECOND)) {
+    GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");
+    if (GST_CLOCK_TIME_IS_VALID (self->next_ts))
+      gst_audio_fx_base_fir_filter_push_residue (self);
+    self->residue_length = 0;
+    self->next_ts = timestamp;
+    self->next_off = GST_BUFFER_OFFSET (inbuf);
+  } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {
+    self->next_ts = timestamp;
+    self->next_off = GST_BUFFER_OFFSET (inbuf);
+  }
+
+  /* Calculate the number of samples we can push out now without outputting
+   * latency zeros in the beginning */
+  diff = self->latency * 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) {
+    return GST_BASE_TRANSFORM_FLOW_DROPPED;
+  }
+
+  GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;
+  GST_BUFFER_DURATION (outbuf) =
+      gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);
+  GST_BUFFER_OFFSET (outbuf) = self->next_off;
+  if (GST_BUFFER_OFFSET_IS_VALID (outbuf))
+    GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;
+  else
+    GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;
+
+  if (output_samples < input_samples) {
+    GST_BUFFER_DATA (outbuf) +=
+        diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+    GST_BUFFER_SIZE (outbuf) -=
+        diff * (GST_AUDIO_FILTER (self)->format.width / 8);
+  }
+
+  self->next_ts += GST_BUFFER_DURATION (outbuf);
+  self->next_off = GST_BUFFER_OFFSET_END (outbuf);
+
+  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);
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+
+  self->residue_length = 0;
+  self->next_ts = GST_CLOCK_TIME_NONE;
+  self->next_off = GST_BUFFER_OFFSET_NONE;
+
+  return TRUE;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+
+  g_free (self->residue);
+  self->residue = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)
+{
+  GstAudioFXBaseFIRFilter *self =
+      GST_AUDIO_FX_BASE_FIR_FILTER (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 (rate == 0) {
+        res = FALSE;
+      } else 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 = gst_util_uint64_scale (self->latency, GST_SECOND, rate);
+
+          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 *
+gst_audio_fx_base_fir_filter_query_type (GstPad * pad)
+{
+  static const GstQueryType types[] = {
+    GST_QUERY_LATENCY,
+    0
+  };
+
+  return types;
+}
+
+static gboolean
+gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)
+{
+  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      gst_audio_fx_base_fir_filter_push_residue (self);
+      self->next_ts = GST_CLOCK_TIME_NONE;
+      self->next_off = GST_BUFFER_OFFSET_NONE;
+      break;
+    default:
+      break;
+  }
+
+  return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+}
+
+void
+gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,
+    gdouble * kernel, guint kernel_length, guint64 latency)
+{
+  g_return_if_fail (kernel != NULL);
+  g_return_if_fail (self != NULL);
+
+  GST_BASE_TRANSFORM_LOCK (self);
+  if (self->residue) {
+    gst_audio_fx_base_fir_filter_push_residue (self);
+    self->next_ts = GST_CLOCK_TIME_NONE;
+    self->next_off = GST_BUFFER_OFFSET_NONE;
+    self->residue_length = 0;
+  }
+
+  g_free (self->kernel);
+  g_free (self->residue);
+
+  self->kernel = kernel;
+  self->kernel_length = kernel_length;
+
+  if (GST_AUDIO_FILTER (self)->format.channels) {
+    self->residue =
+        g_new0 (gdouble,
+        kernel_length * GST_AUDIO_FILTER (self)->format.channels);
+    self->residue_length = 0;
+  }
+
+  if (self->latency != latency) {
+    self->latency = latency;
+    gst_element_post_message (GST_ELEMENT (self),
+        gst_message_new_latency (GST_OBJECT (self)));
+  }
+
+  GST_BASE_TRANSFORM_UNLOCK (self);
+}
diff --git a/gst/audiofx/audiofxbasefirfilter.h b/gst/audiofx/audiofxbasefirfilter.h
new file mode 100644
index 00000000..52f42482
--- /dev/null
+++ b/gst/audiofx/audiofxbasefirfilter.h
@@ -0,0 +1,81 @@
+/* -*- c-basic-offset: 2 -*-
+ * 
+ * GStreamer
+ * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
+ *               2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ *               2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ * 
+ */
+
+#ifndef __GST_AUDIO_FX_BASE_FIR_FILTER_H__
+#define __GST_AUDIO_FX_BASE_FIR_FILTER_H__
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_FX_BASE_FIR_FILTER \
+  (gst_audio_fx_base_fir_filter_get_type())
+#define GST_AUDIO_FX_BASE_FIR_FILTER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilter))
+#define GST_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER,GstAudioFXBaseFIRFilterClass))
+#define GST_IS_AUDIO_FX_BASE_FIR_FILTER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER))
+#define GST_IS_AUDIO_FX_BASE_FIR_FILTER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FX_BASE_FIR_FILTER))
+
+typedef struct _GstAudioFXBaseFIRFilter GstAudioFXBaseFIRFilter;
+typedef struct _GstAudioFXBaseFIRFilterClass GstAudioFXBaseFIRFilterClass;
+
+typedef void (*GstAudioFXBaseFIRFilterProcessFunc) (GstAudioFXBaseFIRFilter *, guint8 *, guint8 *, guint);
+
+/**
+ * GstAudioFXBaseFIRFilter:
+ *
+ * Opaque data structure.
+ */
+struct _GstAudioFXBaseFIRFilter {
+  GstAudioFilter element;
+
+  /* < private > */
+  GstAudioFXBaseFIRFilterProcessFunc process;
+
+  gdouble *kernel;              /* filter kernel */
+  guint kernel_length;           /* length of the filter kernel */
+  gdouble *residue;             /* buffer for left-over samples from previous buffer */
+  guint residue_length;
+
+  guint64 latency;
+
+  GstClockTime next_ts;
+  guint64 next_off;
+};
+
+struct _GstAudioFXBaseFIRFilterClass {
+  GstAudioFilterClass parent_class;
+};
+
+GType gst_audio_fx_base_fir_filter_get_type (void);
+void gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter *filter, gdouble *kernel, guint kernel_length, guint64 latency);
+void gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter *filter);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_FX_BASE_FIR_FILTER_H__ */
diff --git a/gst/audiofx/audiofxbaseiirfilter.c b/gst/audiofx/audiofxbaseiirfilter.c
index 29cb2440..4571a2da 100644
--- a/gst/audiofx/audiofxbaseiirfilter.c
+++ b/gst/audiofx/audiofxbaseiirfilter.c
@@ -43,7 +43,7 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
     " channels = (int) [ 1, MAX ]"
 
 #define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiobaseiirfilter", 0, "Audio IIR Filter Base Class");
+  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class");
 
 GST_BOILERPLATE_FULL (GstAudioFXBaseIIRFilter,
     gst_audio_fx_base_iir_filter, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 60848b1b..62034308 100644
--- a/gst/audiofx/audiowsincband.c
+++ b/gst/audiofx/audiowsincband.c
@@ -3,7 +3,7 @@
  * 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>
+ *               2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -74,25 +74,9 @@
 
 #include "audiowsincband.h"
 
-#define GST_CAT_DEFAULT gst_audio_wsincband_debug
+#define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
-static const GstElementDetails audio_wsincband_details =
-GST_ELEMENT_DETAILS ("Band pass & band reject filter",
-    "Filter/Effect/Audio",
-    "Band pass and band reject windowed sinc filter",
-    "Thomas Vander Stichele <thomas at apestaart dot 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,
@@ -109,9 +93,9 @@ enum
   MODE_BAND_REJECT
 };
 
-#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_audio_wsincband_mode_get_type ())
+#define GST_TYPE_AUDIO_WSINC_BAND_MODE (gst_gst_audio_wsincband_mode_get_type ())
 static GType
-gst_audio_wsincband_mode_get_type (void)
+gst_gst_audio_wsincband_mode_get_type (void)
 {
   static GType gtype = 0;
 
@@ -135,9 +119,9 @@ enum
   WINDOW_BLACKMAN
 };
 
-#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_audio_wsincband_window_get_type ())
+#define GST_TYPE_AUDIO_WSINC_BAND_WINDOW (gst_gst_audio_wsincband_window_get_type ())
 static GType
-gst_audio_wsincband_window_get_type (void)
+gst_gst_audio_wsincband_window_get_type (void)
 {
   static GType gtype = 0;
 
@@ -155,193 +139,96 @@ gst_audio_wsincband_window_get_type (void)
   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_audio_wsincband_debug, "audiowsincband", 0, \
+  GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0, \
       "Band-pass and Band-reject Windowed sinc filter plugin");
 
-GST_BOILERPLATE_FULL (GstAudioWSincBand, audio_wsincband, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+GST_BOILERPLATE_FULL (GstAudioWSincBand, gst_audio_wsincband, GstAudioFilter,
+    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
 
-static void audio_wsincband_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
-static void audio_wsincband_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsincband_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
-static GstFlowReturn audio_wsincband_transform (GstBaseTransform * base,
-    GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audio_wsincband_start (GstBaseTransform * base);
-static gboolean audio_wsincband_event (GstBaseTransform * base,
-    GstEvent * event);
-
-static gboolean audio_wsincband_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsincband_setup (GstAudioFilter * base,
     GstRingBufferSpec * format);
 
-static gboolean audio_wsincband_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsincband_query_type (GstPad * pad);
-
 /* Element class */
-
 static void
-audio_wsincband_dispose (GObject * object)
-{
-  GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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
-audio_wsincband_base_init (gpointer g_class)
+gst_audio_wsincband_base_init (gpointer g_class)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-  GstCaps *caps;
 
-  gst_element_class_set_details (element_class, &audio_wsincband_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);
+  gst_element_class_set_details_simple (element_class,
+      "Band pass & band reject filter", "Filter/Effect/Audio",
+      "Band pass and band reject windowed sinc filter",
+      "Thomas Vander Stichele <thomas at apestaart dot org>, "
+      "Steven W. Smith, "
+      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
 }
 
 static void
-audio_wsincband_class_init (GstAudioWSincBandClass * klass)
+gst_audio_wsincband_class_init (GstAudioWSincBandClass * klass)
 {
-  GObjectClass *gobject_class;
-  GstBaseTransformClass *trans_class;
-  GstAudioFilterClass *filter_class;
-
-  gobject_class = (GObjectClass *) klass;
-  trans_class = (GstBaseTransformClass *) klass;
-  filter_class = (GstAudioFilterClass *) klass;
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
-  gobject_class->set_property = audio_wsincband_set_property;
-  gobject_class->get_property = audio_wsincband_get_property;
-  gobject_class->dispose = audio_wsincband_dispose;
+  gobject_class->set_property = gst_audio_wsincband_set_property;
+  gobject_class->get_property = gst_audio_wsincband_get_property;
 
   /* 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));
+          "Cut-off lower frequency (Hz)", 0.0, 100000.0, 0,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
   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));
+          "Cut-off upper frequency (Hz)", 0.0, 100000.0, 0,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
   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));
+          "Filter kernel length, will be rounded to the next odd number", 3,
+          50000, 101,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
   g_object_class_install_property (gobject_class, PROP_MODE,
       g_param_spec_enum ("mode", "Mode",
           "Band pass or band reject mode", GST_TYPE_AUDIO_WSINC_BAND_MODE,
-          MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+          MODE_BAND_PASS,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
   g_object_class_install_property (gobject_class, PROP_WINDOW,
       g_param_spec_enum ("window", "Window",
           "Window function to use", GST_TYPE_AUDIO_WSINC_BAND_WINDOW,
-          WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+          WINDOW_HAMMING,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
-  trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsincband_transform);
-  trans_class->start = GST_DEBUG_FUNCPTR (audio_wsincband_start);
-  trans_class->event = GST_DEBUG_FUNCPTR (audio_wsincband_event);
-  filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsincband_setup);
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsincband_setup);
 }
 
 static void
-audio_wsincband_init (GstAudioWSincBand * self,
+gst_audio_wsincband_init (GstAudioWSincBand * self,
     GstAudioWSincBandClass * 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,
-      audio_wsincband_query);
-  gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
-      audio_wsincband_query_type);
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioWSincBand * 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
-audio_wsincband_build_kernel (GstAudioWSincBand * self)
+gst_audio_wsincband_build_kernel (GstAudioWSincBand * self)
 {
   gint i = 0;
   gdouble sum = 0.0;
   gint len = 0;
   gdouble *kernel_lp, *kernel_hp;
   gdouble w;
+  gdouble *kernel;
 
   len = self->kernel_length;
 
@@ -369,7 +256,7 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self)
     self->upper_frequency = tmp;
   }
 
-  GST_DEBUG ("audio_wsincband: initializing filter kernel of length %d "
+  GST_DEBUG ("gst_audio_wsincband: 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,
@@ -431,12 +318,10 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self)
   kernel_hp[len / 2] += 1;
 
   /* combine the two kernels */
-  if (self->kernel)
-    g_free (self->kernel);
-  self->kernel = g_new (gdouble, len);
+  kernel = g_new (gdouble, len);
 
   for (i = 0; i < len; ++i)
-    self->kernel[i] = kernel_lp[i] + kernel_hp[i];
+    kernel[i] = kernel_lp[i] + kernel_hp[i];
 
   /* free the helper kernels */
   g_free (kernel_lp);
@@ -446,338 +331,29 @@ audio_wsincband_build_kernel (GstAudioWSincBand * self)
    * 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
-audio_wsincband_push_residue (GstAudioWSincBand * 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");
+      kernel[i] = -kernel[i];
+    kernel[len / 2] += 1;
   }
 
+  gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
+      kernel, self->kernel_length, (len - 1) / 2);
 }
 
 /* GstAudioFilter vmethod implementations */
 
 /* get notified of caps and plug in the correct process function */
 static gboolean
-audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format)
-{
-  GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base);
-
-  gboolean ret = TRUE;
-
-  if (format->width == 32)
-    self->process = (GstAudioWSincBandProcessFunc) process_32;
-  else if (format->width == 64)
-    self->process = (GstAudioWSincBandProcessFunc) process_64;
-  else
-    ret = FALSE;
-
-  self->have_kernel = FALSE;
-
-  return TRUE;
-}
-
-/* GstBaseTransform vmethod implementations */
-
-static GstFlowReturn
-audio_wsincband_transform (GstBaseTransform * base, GstBuffer * inbuf,
-    GstBuffer * outbuf)
-{
-  GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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)
-    audio_wsincband_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
-audio_wsincband_start (GstBaseTransform * base)
-{
-  GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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
-audio_wsincband_query (GstPad * pad, GstQuery * query)
-{
-  GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (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 *
-audio_wsincband_query_type (GstPad * pad)
-{
-  static const GstQueryType types[] = {
-    GST_QUERY_LATENCY,
-    0
-  };
-
-  return types;
-}
-
-static gboolean
-audio_wsincband_event (GstBaseTransform * base, GstEvent * event)
+gst_audio_wsincband_setup (GstAudioFilter * base, GstRingBufferSpec * format)
 {
   GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (base);
 
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_EOS:
-      audio_wsincband_push_residue (self);
-      break;
-    default:
-      break;
-  }
+  gst_audio_wsincband_build_kernel (self);
 
-  return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+  return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
 }
 
 static void
-audio_wsincband_set_property (GObject * object, guint prop_id,
+gst_audio_wsincband_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
   GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
@@ -788,49 +364,43 @@ audio_wsincband_set_property (GObject * object, guint prop_id,
     case PROP_LENGTH:{
       gint val;
 
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       val = g_value_get_int (value);
       if (val % 2 == 0)
         val++;
 
       if (val != self->kernel_length) {
-        if (self->residue) {
-          audio_wsincband_push_residue (self);
-          g_free (self->residue);
-          self->residue = NULL;
-        }
+        gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
+            (self));
         self->kernel_length = val;
-        self->latency = val / 2;
-        audio_wsincband_build_kernel (self);
-        gst_element_post_message (GST_ELEMENT (self),
-            gst_message_new_latency (GST_OBJECT (self)));
+        gst_audio_wsincband_build_kernel (self);
       }
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     }
     case PROP_LOWER_FREQUENCY:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->lower_frequency = g_value_get_float (value);
-      audio_wsincband_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsincband_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     case PROP_UPPER_FREQUENCY:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->upper_frequency = g_value_get_float (value);
-      audio_wsincband_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsincband_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     case PROP_MODE:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->mode = g_value_get_enum (value);
-      audio_wsincband_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsincband_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     case PROP_WINDOW:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->window = g_value_get_enum (value);
-      audio_wsincband_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsincband_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -839,8 +409,8 @@ audio_wsincband_set_property (GObject * object, guint prop_id,
 }
 
 static void
-audio_wsincband_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
+gst_audio_wsincband_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
 {
   GstAudioWSincBand *self = GST_AUDIO_WSINC_BAND (object);
 
diff --git a/gst/audiofx/audiowsincband.h b/gst/audiofx/audiowsincband.h
index d99c788d..977a41f0 100644
--- a/gst/audiofx/audiowsincband.h
+++ b/gst/audiofx/audiowsincband.h
@@ -3,6 +3,7 @@
  * GStreamer
  * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
  *               2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ *               2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -33,10 +34,12 @@
 #include <gst/gst.h>
 #include <gst/audio/gstaudiofilter.h>
 
+#include "audiofxbasefirfilter.h"
+
 G_BEGIN_DECLS
 
 #define GST_TYPE_AUDIO_WSINC_BAND \
-  (audio_wsincband_get_type())
+  (gst_audio_wsincband_get_type())
 #define GST_AUDIO_WSINC_BAND(obj) \
   (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_BAND,GstAudioWSincBand))
 #define GST_AUDIO_WSINC_BAND_CLASS(klass) \
@@ -49,38 +52,26 @@ G_BEGIN_DECLS
 typedef struct _GstAudioWSincBand GstAudioWSincBand;
 typedef struct _GstAudioWSincBandClass GstAudioWSincBandClass;
 
-typedef void (*GstAudioWSincBandProcessFunc) (GstAudioWSincBand *, guint8 *, guint8 *, guint);
-
 /**
  * GstAudioWSincBand:
  *
  * Opaque data structure.
  */
 struct _GstAudioWSincBand {
-  GstAudioFilter element;
+  GstAudioFXBaseFIRFilter parent;
 
   /* < private > */
-  GstAudioWSincBandProcessFunc 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 _GstAudioWSincBandClass {
-  GstAudioFilterClass parent_class;
+  GstAudioFilterClass parent;
 };
 
-GType audio_wsincband_get_type (void);
+GType gst_audio_wsincband_get_type (void);
 
 G_END_DECLS
 
diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c
index 109b89bd..68e8522c 100644
--- a/gst/audiofx/audiowsinclimit.c
+++ b/gst/audiofx/audiowsinclimit.c
@@ -3,7 +3,7 @@
  * 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>
+ *               2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -72,25 +72,9 @@
 
 #include "audiowsinclimit.h"
 
-#define GST_CAT_DEFAULT audio_wsinclimit_debug
+#define GST_CAT_DEFAULT gst_audio_wsinclimit_debug
 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
 
-static const GstElementDetails audio_wsinclimit_details =
-GST_ELEMENT_DETAILS ("Low pass & high pass filter",
-    "Filter/Effect/Audio",
-    "Low pass and high pass windowed sinc filter",
-    "Thomas Vander Stichele <thomas at apestaart dot 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,
@@ -106,9 +90,9 @@ enum
   MODE_HIGH_PASS
 };
 
-#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (audio_wsinclimit_mode_get_type ())
+#define GST_TYPE_AUDIO_WSINC_LIMIT_MODE (gst_audio_wsinclimit_mode_get_type ())
 static GType
-audio_wsinclimit_mode_get_type (void)
+gst_audio_wsinclimit_mode_get_type (void)
 {
   static GType gtype = 0;
 
@@ -132,9 +116,9 @@ enum
   WINDOW_BLACKMAN
 };
 
-#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (audio_wsinclimit_window_get_type ())
+#define GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW (gst_audio_wsinclimit_window_get_type ())
 static GType
-audio_wsinclimit_window_get_type (void)
+gst_audio_wsinclimit_window_get_type (void)
 {
   static GType gtype = 0;
 
@@ -152,189 +136,91 @@ audio_wsinclimit_window_get_type (void)
   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 (audio_wsinclimit_debug, "audiowsinclimit", 0, \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0, \
       "Low-pass and High-pass Windowed sinc filter plugin");
 
-GST_BOILERPLATE_FULL (GstAudioWSincLimit, audio_wsinclimit, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+GST_BOILERPLATE_FULL (GstAudioWSincLimit, gst_audio_wsinclimit, GstAudioFilter,
+    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
 
-static void audio_wsinclimit_set_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
-static void audio_wsinclimit_get_property (GObject * object, guint prop_id,
+static void gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
-static GstFlowReturn audio_wsinclimit_transform (GstBaseTransform * base,
-    GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audio_wsinclimit_start (GstBaseTransform * base);
-static gboolean audio_wsinclimit_event (GstBaseTransform * base,
-    GstEvent * event);
-static gboolean audio_wsinclimit_setup (GstAudioFilter * base,
+static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
     GstRingBufferSpec * format);
 
-static gboolean audio_wsinclimit_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audio_wsinclimit_query_type (GstPad * pad);
-
 /* Element class */
 
 static void
-audio_wsinclimit_dispose (GObject * object)
-{
-  GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (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
-audio_wsinclimit_base_init (gpointer g_class)
+gst_audio_wsinclimit_base_init (gpointer g_class)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-  GstCaps *caps;
-
-  gst_element_class_set_details (element_class, &audio_wsinclimit_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);
+  gst_element_class_set_details_simple (element_class,
+      "Low pass & high pass filter", "Filter/Effect/Audio",
+      "Low pass and high pass windowed sinc filter",
+      "Thomas Vander Stichele <thomas at apestaart dot org>, "
+      "Steven W. Smith, "
+      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
 }
 
 static void
-audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
+gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * 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 = audio_wsinclimit_set_property;
-  gobject_class->get_property = audio_wsinclimit_get_property;
-  gobject_class->dispose = audio_wsinclimit_dispose;
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  gobject_class->set_property = gst_audio_wsinclimit_set_property;
+  gobject_class->get_property = gst_audio_wsinclimit_get_property;
 
   /* 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_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
   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));
+          3, 50000, 101,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
   g_object_class_install_property (gobject_class, PROP_MODE,
       g_param_spec_enum ("mode", "Mode",
           "Low pass or high pass mode", GST_TYPE_AUDIO_WSINC_LIMIT_MODE,
-          MODE_LOW_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+          MODE_LOW_PASS,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
   g_object_class_install_property (gobject_class, PROP_WINDOW,
       g_param_spec_enum ("window", "Window",
           "Window function to use", GST_TYPE_AUDIO_WSINC_LIMIT_WINDOW,
-          WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+          WINDOW_HAMMING,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
-  trans_class->transform = GST_DEBUG_FUNCPTR (audio_wsinclimit_transform);
-  trans_class->start = GST_DEBUG_FUNCPTR (audio_wsinclimit_start);
-  trans_class->event = GST_DEBUG_FUNCPTR (audio_wsinclimit_event);
-  filter_class->setup = GST_DEBUG_FUNCPTR (audio_wsinclimit_setup);
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsinclimit_setup);
 }
 
 static void
-audio_wsinclimit_init (GstAudioWSincLimit * self,
+gst_audio_wsinclimit_init (GstAudioWSincLimit * self,
     GstAudioWSincLimitClass * 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,
-      audio_wsinclimit_query);
-  gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,
-      audio_wsinclimit_query_type);
-}
-
-#define DEFINE_PROCESS_FUNC(width,ctype) \
-static void \
-process_##width (GstAudioWSincLimit * 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
-audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
+gst_audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
 {
   gint i = 0;
   gdouble sum = 0.0;
   gint len = 0;
   gdouble w;
+  gdouble *kernel = NULL;
 
   len = self->kernel_length;
 
@@ -352,7 +238,7 @@ audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
   self->cutoff =
       CLAMP (self->cutoff, 0.0, GST_AUDIO_FILTER (self)->format.rate / 2);
 
-  GST_DEBUG ("audio_wsinclimit_: initializing filter kernel of length %d "
+  GST_DEBUG ("gst_audio_wsinclimit_: initializing filter kernel of length %d "
       "with cutoff %.2lf Hz "
       "for mode %s",
       len, self->cutoff,
@@ -361,365 +247,53 @@ audio_wsinclimit_build_kernel (GstAudioWSincLimit * self)
   /* 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);
+  kernel = g_new (gdouble, len);
 
   for (i = 0; i < len; ++i) {
     if (i == len / 2)
-      self->kernel[i] = w;
+      kernel[i] = w;
     else
-      self->kernel[i] = sin (w * (i - len / 2)) / (i - len / 2);
+      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));
+      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) +
+      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];
+    sum += kernel[i];
   for (i = 0; i < len; ++i)
-    self->kernel[i] /= sum;
+    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
-audio_wsinclimit_push_residue (GstAudioWSincLimit * 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");
+      kernel[i] = -kernel[i];
+    kernel[len / 2] += 1.0;
   }
 
+  gst_audio_fx_base_fir_filter_set_kernel (GST_AUDIO_FX_BASE_FIR_FILTER (self),
+      kernel, self->kernel_length, (len - 1) / 2);
 }
 
 /* GstAudioFilter vmethod implementations */
 
 /* get notified of caps and plug in the correct process function */
 static gboolean
-audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+gst_audio_wsinclimit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
 {
   GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
 
-  gboolean ret = TRUE;
-
-  if (format->width == 32)
-    self->process = (GstAudioWSincLimitProcessFunc) process_32;
-  else if (format->width == 64)
-    self->process = (GstAudioWSincLimitProcessFunc) process_64;
-  else
-    ret = FALSE;
-
-  self->have_kernel = FALSE;
-
-  return TRUE;
-}
-
-/* GstBaseTransform vmethod implementations */
-
-static GstFlowReturn
-audio_wsinclimit_transform (GstBaseTransform * base, GstBuffer * inbuf,
-    GstBuffer * outbuf)
-{
-  GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (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)
-    audio_wsinclimit_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
-audio_wsinclimit_start (GstBaseTransform * base)
-{
-  GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (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
-audio_wsinclimit_query (GstPad * pad, GstQuery * query)
-{
-  GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (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 *
-audio_wsinclimit_query_type (GstPad * pad)
-{
-  static const GstQueryType types[] = {
-    GST_QUERY_LATENCY,
-    0
-  };
-
-  return types;
-}
-
-static gboolean
-audio_wsinclimit_event (GstBaseTransform * base, GstEvent * event)
-{
-  GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (base);
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_EOS:
-      audio_wsinclimit_push_residue (self);
-      break;
-    default:
-      break;
-  }
+  gst_audio_wsinclimit_build_kernel (self);
 
-  return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
+  return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
 }
 
 static void
-audio_wsinclimit_set_property (GObject * object, guint prop_id,
+gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
   GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
@@ -730,43 +304,37 @@ audio_wsinclimit_set_property (GObject * object, guint prop_id,
     case PROP_LENGTH:{
       gint val;
 
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       val = g_value_get_int (value);
       if (val % 2 == 0)
         val++;
 
       if (val != self->kernel_length) {
-        if (self->residue) {
-          audio_wsinclimit_push_residue (self);
-          g_free (self->residue);
-          self->residue = NULL;
-        }
+        gst_audio_fx_base_fir_filter_push_residue (GST_AUDIO_FX_BASE_FIR_FILTER
+            (self));
         self->kernel_length = val;
-        self->latency = val / 2;
-        audio_wsinclimit_build_kernel (self);
-        gst_element_post_message (GST_ELEMENT (self),
-            gst_message_new_latency (GST_OBJECT (self)));
+        gst_audio_wsinclimit_build_kernel (self);
       }
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     }
     case PROP_FREQUENCY:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->cutoff = g_value_get_float (value);
-      audio_wsinclimit_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsinclimit_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     case PROP_MODE:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->mode = g_value_get_enum (value);
-      audio_wsinclimit_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsinclimit_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     case PROP_WINDOW:
-      GST_BASE_TRANSFORM_LOCK (self);
+      GST_OBJECT_LOCK (self);
       self->window = g_value_get_enum (value);
-      audio_wsinclimit_build_kernel (self);
-      GST_BASE_TRANSFORM_UNLOCK (self);
+      gst_audio_wsinclimit_build_kernel (self);
+      GST_OBJECT_UNLOCK (self);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
@@ -775,8 +343,8 @@ audio_wsinclimit_set_property (GObject * object, guint prop_id,
 }
 
 static void
-audio_wsinclimit_get_property (GObject * object, guint prop_id, GValue * value,
-    GParamSpec * pspec)
+gst_audio_wsinclimit_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
 {
   GstAudioWSincLimit *self = GST_AUDIO_WSINC_LIMIT (object);
 
diff --git a/gst/audiofx/audiowsinclimit.h b/gst/audiofx/audiowsinclimit.h
index b781a422..d30b3938 100644
--- a/gst/audiofx/audiowsinclimit.h
+++ b/gst/audiofx/audiowsinclimit.h
@@ -3,6 +3,7 @@
  * GStreamer
  * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
  *               2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
+ *               2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -33,10 +34,12 @@
 #include <gst/gst.h>
 #include <gst/audio/gstaudiofilter.h>
 
+#include "audiofxbasefirfilter.h"
+
 G_BEGIN_DECLS
 
 #define GST_TYPE_AUDIO_WSINC_LIMIT \
-  (audio_wsinclimit_get_type())
+  (gst_audio_wsinclimit_get_type())
 #define GST_AUDIO_WSINC_LIMIT(obj) \
   (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_WSINC_LIMIT,GstAudioWSincLimit))
 #define GST_AUDIO_WSINC_LIMIT_CLASS(klass) \
@@ -49,38 +52,26 @@ G_BEGIN_DECLS
 typedef struct _GstAudioWSincLimit GstAudioWSincLimit;
 typedef struct _GstAudioWSincLimitClass GstAudioWSincLimitClass;
 
-typedef void (*GstAudioWSincLimitProcessFunc) (GstAudioWSincLimit *, guint8 *, guint8 *, guint);
-
 /**
  * GstAudioWSincLimit:
  *
  * Opaque data structure.
  */
 struct _GstAudioWSincLimit {
-  GstAudioFilter element;
+  GstAudioFXBaseFIRFilter parent;
 
   /* < private > */
-  GstAudioWSincLimitProcessFunc 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;
+  gint kernel_length;
 };
 
 struct _GstAudioWSincLimitClass {
-  GstAudioFilterClass parent_class;
+  GstAudioFXBaseFIRFilterClass parent;
 };
 
-GType audio_wsinclimit_get_type (void);
+GType gst_audio_wsinclimit_get_type (void);
 
 G_END_DECLS
 
diff --git a/tests/check/elements/audiowsincband.c b/tests/check/elements/audiowsincband.c
index c3ea37f0..fc4805df 100644
--- a/tests/check/elements/audiowsincband.c
+++ b/tests/check/elements/audiowsincband.c
@@ -119,6 +119,7 @@ GST_START_TEST (test_32_bp_0hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i++)
     in[i] = 1.0;
@@ -180,6 +181,7 @@ GST_START_TEST (test_32_bp_11025hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 4) {
     in[i] = 0.0;
@@ -246,6 +248,7 @@ GST_START_TEST (test_32_bp_22050hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 2) {
     in[i] = 1.0;
@@ -309,6 +312,7 @@ GST_START_TEST (test_32_br_0hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i++)
     in[i] = 1.0;
@@ -370,6 +374,7 @@ GST_START_TEST (test_32_br_11025hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
 
   for (i = 0; i < 1024; i += 4) {
@@ -437,6 +442,7 @@ GST_START_TEST (test_32_br_22050hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 2) {
     in[i] = 1.0;
@@ -498,6 +504,7 @@ GST_START_TEST (test_32_small_buffer)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 44100 / 16.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 20; i++)
     in[i] = 1.0;
@@ -553,6 +560,7 @@ GST_START_TEST (test_64_bp_0hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i++)
     in[i] = 1.0;
@@ -614,6 +622,7 @@ GST_START_TEST (test_64_bp_11025hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 4) {
     in[i] = 0.0;
@@ -680,6 +689,7 @@ GST_START_TEST (test_64_bp_22050hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 2) {
     in[i] = 1.0;
@@ -743,6 +753,7 @@ GST_START_TEST (test_64_br_0hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i++)
     in[i] = 1.0;
@@ -804,6 +815,7 @@ GST_START_TEST (test_64_br_11025hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
 
   for (i = 0; i < 1024; i += 4) {
@@ -871,6 +883,7 @@ GST_START_TEST (test_64_br_22050hz)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 1000, NULL);
   inbuffer = gst_buffer_new_and_alloc (1024 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 1024; i += 2) {
     in[i] = 1.0;
@@ -932,6 +945,7 @@ GST_START_TEST (test_64_small_buffer)
   g_object_set (G_OBJECT (audiowsincband), "upper-frequency",
       44100 / 4.0 + 44100 / 16.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 20; i++)
     in[i] = 1.0;
diff --git a/tests/check/elements/audiowsinclimit.c b/tests/check/elements/audiowsinclimit.c
index da176049..3a6280d3 100644
--- a/tests/check/elements/audiowsinclimit.c
+++ b/tests/check/elements/audiowsinclimit.c
@@ -117,6 +117,7 @@ GST_START_TEST (test_32_lp_0hz)
   /* cutoff = sampling rate / 4, data = 0 */
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
     in[i] = 1.0;
@@ -175,6 +176,7 @@ GST_START_TEST (test_32_lp_22050hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
     in[i] = 1.0;
@@ -235,6 +237,7 @@ GST_START_TEST (test_32_hp_0hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
     in[i] = 1.0;
@@ -293,6 +296,7 @@ GST_START_TEST (test_32_hp_22050hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
     in[i] = 1.0;
@@ -352,6 +356,7 @@ GST_START_TEST (test_32_small_buffer)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gfloat));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gfloat *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 20; i++)
     in[i] = 1.0;
@@ -398,6 +403,7 @@ GST_START_TEST (test_64_lp_0hz)
   /* cutoff = sampling rate / 4, data = 0 */
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
     in[i] = 1.0;
@@ -456,6 +462,7 @@ GST_START_TEST (test_64_lp_22050hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
     in[i] = 1.0;
@@ -516,6 +523,7 @@ GST_START_TEST (test_64_hp_0hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i++)
     in[i] = 1.0;
@@ -574,6 +582,7 @@ GST_START_TEST (test_64_hp_22050hz)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (128 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 128; i += 2) {
     in[i] = 1.0;
@@ -633,6 +642,7 @@ GST_START_TEST (test_64_small_buffer)
 
   g_object_set (G_OBJECT (audiowsinclimit), "cutoff", 44100 / 4.0, NULL);
   inbuffer = gst_buffer_new_and_alloc (20 * sizeof (gdouble));
+  GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   in = (gdouble *) GST_BUFFER_DATA (inbuffer);
   for (i = 0; i < 20; i++)
     in[i] = 1.0;