/* GStreamer audio filter base class

 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>

 * Copyright (C) <2003> David Schleef <ds@schleef.org>

 * Copyright (C) <2007> Tim-Philipp Müller <tim centricular net>

 *

 * This library is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Library General Public

 * License as published by the Free Software Foundation; either

 * version 2 of the License, or (at your option) any later version.

 *

 * This library is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Library General Public License for more details.

 *

 * You should have received a copy of the GNU Library General Public

 * License along with this library; if not, write to the

 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

 * Boston, MA 02111-1307, USA.

 */

/**

 * SECTION:gstaudiofilter

 * @short_description: Base class for simple audio filters

 *

 * #GstAudioFilter is a #GstBaseTransform-derived base class for simple audio

 * filters, ie. those that output the same format that they get as input.

 *

 * #GstAudioFilter will parse the input format for you (with error checking)

 * before calling your setup function. Also, elements deriving from

 * #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from

 * their base_init function to easily configure the set of caps/formats that

 * the element is able to handle.

 *

 * virtual functions in their class_init function.

 *

 *

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include "gstaudiofilter.h"

#include <string.h>

GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);

#define GST_CAT_DEFAULT audiofilter_dbg

static void gst_audio_filter_class_init (gpointer g_class, gpointer class_data);

static void gst_audio_filter_init (GTypeInstance * instance, gpointer g_class);

static GstStateChangeReturn gst_audio_filter_change_state (GstElement * element,

    GstStateChange transition);

static gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,

    GstCaps * incaps, GstCaps * outcaps);

static gboolean gst_audio_filter_get_unit_size (GstBaseTransform * btrans,

    GstCaps * caps, guint * size);

static GstElementClass *parent_class = NULL;

#ifdef __SYMBIAN32__

EXPORT_C

#endif

GType

gst_audio_filter_get_type (void)

{

  static GType audio_filter_type = 0;

  if (!audio_filter_type) {

    const GTypeInfo audio_filter_info = {

      sizeof (GstAudioFilterClass),

      NULL,

      NULL,

      gst_audio_filter_class_init,

      NULL,

      NULL,

      sizeof (GstAudioFilter),

      0,

      gst_audio_filter_init,

    };

    GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter");

    audio_filter_type = g_type_register_static (GST_TYPE_BASE_TRANSFORM,

        "GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);

  }

  return audio_filter_type;

}

static void

gst_audio_filter_class_init (gpointer klass, gpointer class_data)

{

  GstBaseTransformClass *basetrans_class;

  GstElementClass *gstelement_class;

  parent_class = g_type_class_peek_parent (klass);

  gstelement_class = (GstElementClass *) klass;

  basetrans_class = (GstBaseTransformClass *) klass;

  gstelement_class->change_state =

      GST_DEBUG_FUNCPTR (gst_audio_filter_change_state);

  basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);

  basetrans_class->get_unit_size =

      GST_DEBUG_FUNCPTR (gst_audio_filter_get_unit_size);

  /* FIXME: Ref the GstRingerBuffer class to get it's debug category

   * initialized. gst_ring_buffer_parse_caps () which we use later

   * uses this debug category.

   */

  g_type_class_ref (GST_TYPE_RING_BUFFER);

}

static void

gst_audio_filter_init (GTypeInstance * instance, gpointer g_class)

{

  /* nothing to do here */

}

/* we override the state change vfunc here instead of GstBaseTransform's stop

 * vfunc, so GstAudioFilter-derived elements can override ::stop() for their

 * own purposes without having to worry about chaining up */

static GstStateChangeReturn

gst_audio_filter_change_state (GstElement * element, GstStateChange transition)

{

  GstStateChangeReturn ret;

  GstAudioFilter *filter;

  filter = GST_AUDIO_FILTER (element);

  switch (transition) {

    case GST_STATE_CHANGE_NULL_TO_READY:

      memset (&filter->format, 0, sizeof (GstRingBufferSpec));

      /* to make gst_buffer_spec_parse_caps() happy */

      filter->format.latency_time = GST_SECOND;

      break;

    default:

      break;

  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  if (ret == GST_STATE_CHANGE_FAILURE)

    return ret;

  switch (transition) {

    case GST_STATE_CHANGE_PAUSED_TO_READY:

    case GST_STATE_CHANGE_READY_TO_NULL:

      gst_caps_replace (&filter->format.caps, NULL);

      break;

    default:

      break;

  }

  return ret;

}

static gboolean

gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,

    GstCaps * outcaps)

{

  GstAudioFilterClass *klass;

  GstAudioFilter *filter;

  gboolean ret = TRUE;

  filter = GST_AUDIO_FILTER (btrans);

  GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);

  if (!gst_ring_buffer_parse_caps (&filter->format, incaps)) {

    GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);

    return FALSE;

  }

  klass = GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (filter));

  if (klass->setup)

    ret = klass->setup (filter, &filter->format);

  return ret;

}

static gboolean

gst_audio_filter_get_unit_size (GstBaseTransform * btrans, GstCaps * caps,

    guint * size)

{

  GstStructure *structure;

  gboolean ret = TRUE;

  gint width, channels;

  structure = gst_caps_get_structure (caps, 0);

  ret &= gst_structure_get_int (structure, "width", &width);

  ret &= gst_structure_get_int (structure, "channels", &channels);

  if (ret)

    *size = (width / 8) * channels;

  return ret;

}

/**

 * gst_audio_filter_class_add_pad_templates:

 * @klass: an #GstAudioFilterClass

 * @allowed_caps: what formats the filter can handle, as #GstCaps

 *

 * Convenience function to add pad templates to this element class, with

 * @allowed_caps as the caps that can be handled.

 *

 * This function is usually used from within a GObject base_init function.

 *

 * Since: 0.10.12

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,

    const GstCaps * allowed_caps)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));

  g_return_if_fail (allowed_caps != NULL);

  g_return_if_fail (GST_IS_CAPS (allowed_caps));

}