/* GStreamer

 * Copyright (C) 2011 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>.

 * Copyright (C) 2011 Nokia Corporation. All rights reserved.

 *   Contact: Stefan Kost <stefan.kost@nokia.com>

 *

 * 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.

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <gst/audio/audio.h>

#include "gstaudioutilsprivate.h"

/*

 * Takes caps and copies its audio fields to tmpl_caps

 */

static GstCaps *

__gst_audio_element_proxy_caps (GstElement * element, GstCaps * templ_caps,

    GstCaps * caps)

{

  GstCaps *result = gst_caps_new_empty ();

  gint i, j;

  gint templ_caps_size = gst_caps_get_size (templ_caps);

  gint caps_size = gst_caps_get_size (caps);

  for (i = 0; i < templ_caps_size; i++) {

    GQuark q_name =

        gst_structure_get_name_id (gst_caps_get_structure (templ_caps, i));

    GstCapsFeatures *features = gst_caps_get_features (templ_caps, i);

    for (j = 0; j < caps_size; j++) {

      const GstStructure *caps_s = gst_caps_get_structure (caps, j);

      const GValue *val;

      GstStructure *s;

      GstCaps *tmp = gst_caps_new_empty ();

      s = gst_structure_new_id_empty (q_name);

      if ((val = gst_structure_get_value (caps_s, "rate")))

        gst_structure_set_value (s, "rate", val);

      if ((val = gst_structure_get_value (caps_s, "channels")))

        gst_structure_set_value (s, "channels", val);

      if ((val = gst_structure_get_value (caps_s, "channels-mask")))

        gst_structure_set_value (s, "channels-mask", val);

      gst_caps_append_structure_full (tmp, s,

          gst_caps_features_copy (features));

      result = gst_caps_merge (result, tmp);

    }

  }

  return result;

}

/**

 * __gst_audio_element_proxy_getcaps:

 * @element: a #GstElement

 * @sinkpad: the element's sink #GstPad

 * @srcpad: the element's source #GstPad

 * @initial_caps: initial caps

 * @filter: filter caps

 *

 * Returns caps that express @initial_caps (or sink template caps if

 * @initial_caps == NULL) restricted to rate/channels/...

 * combinations supported by downstream elements (e.g. muxers).

 *

 * Returns: a #GstCaps owned by caller

 */

GstCaps *

__gst_audio_element_proxy_getcaps (GstElement * element, GstPad * sinkpad,

    GstPad * srcpad, GstCaps * initial_caps, GstCaps * filter)

{

  GstCaps *templ_caps, *src_templ_caps;

  GstCaps *peer_caps;

  GstCaps *allowed;

  GstCaps *fcaps, *filter_caps;

  /* Allow downstream to specify rate/channels constraints

   * and forward them upstream for audio converters to handle

   */

  templ_caps = initial_caps ? gst_caps_ref (initial_caps) :

      gst_pad_get_pad_template_caps (sinkpad);

  src_templ_caps = gst_pad_get_pad_template_caps (srcpad);

  if (filter && !gst_caps_is_any (filter)) {

    GstCaps *proxy_filter =

        __gst_audio_element_proxy_caps (element, src_templ_caps, filter);

    peer_caps = gst_pad_peer_query_caps (srcpad, proxy_filter);

    gst_caps_unref (proxy_filter);

  } else {

    peer_caps = gst_pad_peer_query_caps (srcpad, NULL);

  }

  allowed = gst_caps_intersect_full (peer_caps, src_templ_caps,

      GST_CAPS_INTERSECT_FIRST);

  gst_caps_unref (src_templ_caps);

  gst_caps_unref (peer_caps);

  if (!allowed || gst_caps_is_any (allowed)) {

    fcaps = templ_caps;

    goto done;

  } else if (gst_caps_is_empty (allowed)) {

    fcaps = gst_caps_ref (allowed);

    goto done;

  }

  GST_LOG_OBJECT (element, "template caps %" GST_PTR_FORMAT, templ_caps);

  GST_LOG_OBJECT (element, "allowed caps %" GST_PTR_FORMAT, allowed);

  filter_caps = __gst_audio_element_proxy_caps (element, templ_caps, allowed);

  fcaps = gst_caps_intersect (filter_caps, templ_caps);

  gst_caps_unref (filter_caps);

  gst_caps_unref (templ_caps);

  if (filter) {

    GST_LOG_OBJECT (element, "intersecting with %" GST_PTR_FORMAT, filter);

    filter_caps = gst_caps_intersect (fcaps, filter);

    gst_caps_unref (fcaps);

    fcaps = filter_caps;

  }

done:

  gst_caps_replace (&allowed, NULL);

  GST_LOG_OBJECT (element, "proxy caps %" GST_PTR_FORMAT, fcaps);

  return fcaps;

}

/**

 * __gst_audio_encoded_audio_convert:

 * @fmt: audio format of the encoded audio

 * @bytes: number of encoded bytes

 * @samples: number of encoded samples

 * @src_format: source format

 * @src_value: source value

 * @dest_format: destination format

 * @dest_value: destination format

 *

 * Helper function to convert @src_value in @src_format to @dest_value in

 * @dest_format for encoded audio data.  Conversion is possible between

 * BYTE and TIME format by using estimated bitrate based on

 * @samples and @bytes (and @fmt).

 */

gboolean

__gst_audio_encoded_audio_convert (GstAudioInfo * fmt,

    gint64 bytes, gint64 samples, GstFormat src_format,

    gint64 src_value, GstFormat * dest_format, gint64 * dest_value)

{

  gboolean res = FALSE;

  g_return_val_if_fail (dest_format != NULL, FALSE);

  g_return_val_if_fail (dest_value != NULL, FALSE);

  if (G_UNLIKELY (src_format == *dest_format || src_value == 0 ||

          src_value == -1)) {

    if (dest_value)

      *dest_value = src_value;

    return TRUE;

  }

  if (samples == 0 || bytes == 0 || fmt->rate == 0) {

    GST_DEBUG ("not enough metadata yet to convert");

    goto exit;

  }

  bytes *= fmt->rate;

  switch (src_format) {

    case GST_FORMAT_BYTES:

      switch (*dest_format) {

        case GST_FORMAT_TIME:

          *dest_value = gst_util_uint64_scale (src_value,

              GST_SECOND * samples, bytes);

          res = TRUE;

          break;

        default:

          res = FALSE;

      }

      break;

    case GST_FORMAT_TIME:

      switch (*dest_format) {

        case GST_FORMAT_BYTES:

          *dest_value = gst_util_uint64_scale (src_value, bytes,

              samples * GST_SECOND);

          res = TRUE;

          break;

        default:

          res = FALSE;

      }

      break;

    default:

      res = FALSE;

  }

exit:

  return res;

}