1 files changed, 68 insertions, 500 deletions

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);