/* 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:gstaudiofilter

 * @title: 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 class_init function to easily configure the set of caps/formats that

 * the element is able to handle.

 *

 * Derived classes should override the #GstAudioFilterClass.setup() and

 * #GstBaseTransformClass.transform_ip() and/or

 * #GstBaseTransformClass.transform()

 * 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 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, gsize * size);

static GstFlowReturn gst_audio_filter_submit_input_buffer (GstBaseTransform *

    btrans, gboolean is_discont, GstBuffer * input);

#define do_init G_STMT_START { \

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

} G_STMT_END

G_DEFINE_ABSTRACT_TYPE_WITH_CODE (GstAudioFilter, gst_audio_filter,

    GST_TYPE_BASE_TRANSFORM, do_init);

static gboolean

gst_audio_filter_transform_meta (GstBaseTransform * trans, GstBuffer * inbuf,

    GstMeta * meta, GstBuffer * outbuf)

{

  const GstMetaInfo *info = meta->info;

  const gchar *const *tags;

  tags = gst_meta_api_type_get_tags (info->api);

  if (!tags || (g_strv_length ((gchar **) tags) == 1

          && gst_meta_api_type_has_tag (info->api,

              g_quark_from_string (GST_META_TAG_AUDIO_STR))))

    return TRUE;

  return

      GST_BASE_TRANSFORM_CLASS (gst_audio_filter_parent_class)->transform_meta

      (trans, inbuf, meta, outbuf);

}

static void

gst_audio_filter_class_init (GstAudioFilterClass * klass)

{

  GstBaseTransformClass *basetrans_class = (GstBaseTransformClass *) klass;

  GstElementClass *gstelement_class = (GstElementClass *) 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);

  basetrans_class->transform_meta = gst_audio_filter_transform_meta;

  basetrans_class->submit_input_buffer = gst_audio_filter_submit_input_buffer;

}

static void

gst_audio_filter_init (GstAudioFilter * self)

{

  gst_audio_info_init (&self->info);

}

/* 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 = GST_AUDIO_FILTER (element);

  ret =

      GST_ELEMENT_CLASS (gst_audio_filter_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_audio_info_init (&filter->info);

      break;

    default:

      break;

  }

  return ret;

}

static gboolean

gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,

    GstCaps * outcaps)

{

  GstAudioFilterClass *klass;

  GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);

  GstAudioInfo info;

  gboolean ret = TRUE;

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

  GST_LOG_OBJECT (filter, "info: %d", GST_AUDIO_FILTER_RATE (filter));

  if (!gst_audio_info_from_caps (&info, incaps))

    goto invalid_format;

  klass = GST_AUDIO_FILTER_GET_CLASS (filter);

  if (klass->setup)

    ret = klass->setup (filter, &info;;

  if (ret) {

    filter->info = info;

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

  }

  return ret;

  /* ERROR */

invalid_format:

  {

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

    return FALSE;

  }

}

static GstFlowReturn

gst_audio_filter_submit_input_buffer (GstBaseTransform * btrans,

    gboolean is_discont, GstBuffer * input)

{

  GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);

  if (btrans->segment.format == GST_FORMAT_TIME) {

    input =

        gst_audio_buffer_clip (input, &btrans->segment, filter->info.rate,

        filter->info.bpf);

    if (!input)

      return GST_FLOW_OK;

  }

  return

      GST_BASE_TRANSFORM_CLASS

      (gst_audio_filter_parent_class)->submit_input_buffer (btrans, is_discont,

      input);

}

static gboolean

gst_audio_filter_get_unit_size (GstBaseTransform * btrans, GstCaps * caps,

    gsize * size)

{

  GstAudioInfo info;

  if (!gst_audio_info_from_caps (&info, caps))

    return FALSE;

  *size = GST_AUDIO_INFO_BPF (&info;;

  return TRUE;

}

/**

 * 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 class_init function.

 */

void

gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,

    GstCaps * allowed_caps)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  GstPadTemplate *pad_template;

  g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));

  g_return_if_fail (GST_IS_CAPS (allowed_caps));

  pad_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,

      allowed_caps);

  gst_element_class_add_pad_template (element_class, pad_template);

  pad_template = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,

      allowed_caps);

  gst_element_class_add_pad_template (element_class, pad_template);

}