/* GStreamer

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

 *

 * 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.

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include "audio.h"

#include <gst/gststructure.h>

/**

 * gst_audio_frame_byte_size:

 * @pad: the #GstPad to get the caps from

 *

 * Calculate byte size of an audio frame.

 *

 * Returns: the byte size, or 0 if there was an error

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

int

gst_audio_frame_byte_size (GstPad * pad)

{

  /* FIXME: this should be moved closer to the gstreamer core

   * and be implemented for every mime type IMO

   */

  int width = 0;

  int channels = 0;

  const GstCaps *caps = NULL;

  GstStructure *structure;

  /* get caps of pad */

  caps = GST_PAD_CAPS (pad);

  if (caps == NULL) {

    /* ERROR: could not get caps of pad */

    g_warning ("gstaudio: could not get caps of pad %s:%s\n",

        GST_DEBUG_PAD_NAME (pad));

    return 0;

  }

  structure = gst_caps_get_structure (caps, 0);

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

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

  return (width / 8) * channels;

}

/**

 * gst_audio_frame_length:

 * @pad: the #GstPad to get the caps from

 * @buf: the #GstBuffer

 *

 * Calculate length of buffer in frames.

 *

 * Returns: 0 if there's an error, or the number of frames if everything's ok

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

long

gst_audio_frame_length (GstPad * pad, GstBuffer * buf)

{

  /* FIXME: this should be moved closer to the gstreamer core

   * and be implemented for every mime type IMO

   */

  int frame_byte_size = 0;

  frame_byte_size = gst_audio_frame_byte_size (pad);

  if (frame_byte_size == 0)

    /* error */

    return 0;

  /* FIXME: this function assumes the buffer size to be a whole multiple

   *        of the frame byte size

   */

  return GST_BUFFER_SIZE (buf) / frame_byte_size;

}

/**

 * gst_audio_duration_from_pad_buffer:

 * @pad: the #GstPad to get the caps from

 * @buf: the #GstBuffer

 *

 * Calculate length in nanoseconds of audio buffer @buf based on capabilities of

 * @pad.

 *

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

GstClockTime

gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf)

{

  long bytes = 0;

  int width = 0;

  int channels = 0;

  int rate = 0;

  GstClockTime length;

  const GstCaps *caps = NULL;

  GstStructure *structure;

  g_assert (GST_IS_BUFFER (buf));

  /* get caps of pad */

  caps = GST_PAD_CAPS (pad);

  if (caps == NULL) {

    /* ERROR: could not get caps of pad */

    g_warning ("gstaudio: could not get caps of pad %s:%s\n",

        GST_DEBUG_PAD_NAME (pad));

    length = GST_CLOCK_TIME_NONE;

  } else {

    structure = gst_caps_get_structure (caps, 0);

    bytes = GST_BUFFER_SIZE (buf);

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

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

    gst_structure_get_int (structure, "rate", &rate);

    g_assert (bytes != 0);

    g_assert (width != 0);

    g_assert (channels != 0);

    g_assert (rate != 0);

    length = (bytes * 8 * GST_SECOND) / (rate * channels * width);

  }

  return length;

}

/**

 * gst_audio_is_buffer_framed:

 * @pad: the #GstPad to get the caps from

 * @buf: the #GstBuffer

 *

 * Check if the buffer size is a whole multiple of the frame size.

 *

 * Returns: %TRUE if buffer size is multiple.

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

gboolean

gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)

{

  if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)

    return TRUE;

  else

    return FALSE;

}

/* _getcaps helper functions

 * sets structure fields to default for audio type

 * flag determines which structure fields to set to default

 * keep these functions in sync with the templates in audio.h

 */

/* private helper function

 * sets a list on the structure

 * pass in structure, fieldname for the list, type of the list values,

 * number of list values, and each of the values, terminating with NULL

 */

static void

_gst_audio_structure_set_list (GstStructure * structure,

    const gchar * fieldname, GType type, int number, ...)

{

  va_list varargs;

  GValue value = { 0 };

  GArray *array;

  int j;

  g_return_if_fail (structure != NULL);

  g_value_init (&value, GST_TYPE_LIST);

  array = g_value_peek_pointer (&value);

  va_start (varargs, number);

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

    int i;

    gboolean b;

    GValue list_value = { 0 };

    switch (type) {

      case G_TYPE_INT:

        i = va_arg (varargs, int);

        g_value_init (&list_value, G_TYPE_INT);

        g_value_set_int (&list_value, i);

        break;

      case G_TYPE_BOOLEAN:

        b = va_arg (varargs, gboolean);

        g_value_init (&list_value, G_TYPE_BOOLEAN);

        g_value_set_boolean (&list_value, b);

        break;

      default:

        g_warning

            ("_gst_audio_structure_set_list: LIST of given type not implemented.");

    }

    g_array_append_val (array, list_value);

  }

  gst_structure_set_value (structure, fieldname, &value);

  va_end (varargs);

}

/**

 * gst_audio_structure_set_int:

 * @structure: a #GstStructure

 * @flag: a set of #GstAudioFieldFlag

 *

 * Do not use anymore.

 *

 * Deprecated: use gst_structure_set()

 */

#ifndef GST_REMOVE_DEPRECATED

#ifdef GST_DISABLE_DEPRECATED

typedef enum

{

  GST_AUDIO_FIELD_RATE = (1 << 0),

  GST_AUDIO_FIELD_CHANNELS = (1 << 1),

  GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),

  GST_AUDIO_FIELD_WIDTH = (1 << 3),

  GST_AUDIO_FIELD_DEPTH = (1 << 4),

  GST_AUDIO_FIELD_SIGNED = (1 << 5),

} GstAudioFieldFlag;

#endif /* GST_DISABLE_DEPRECATED */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)

{

  /* was added here:

   * http://webcvs.freedesktop.org/gstreamer/gst-plugins-base/gst-libs/gst/audio/audio.c?r1=1.16&r2=1.17

   * but it is not used

   */

  if (flag & GST_AUDIO_FIELD_RATE)

    gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,

        NULL);

  if (flag & GST_AUDIO_FIELD_CHANNELS)

    gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,

        NULL);

  if (flag & GST_AUDIO_FIELD_ENDIANNESS)

    _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,

        G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);

  if (flag & GST_AUDIO_FIELD_WIDTH)

    _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,

        NULL);

  if (flag & GST_AUDIO_FIELD_DEPTH)

    gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);

  if (flag & GST_AUDIO_FIELD_SIGNED)

    _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,

        FALSE, NULL);

}

#endif /* GST_REMOVE_DEPRECATED */

/**

 * gst_audio_buffer_clip:

 * @buffer: The buffer to clip.

 * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which the buffer should be clipped.

 * @rate: sample rate.

 * @frame_size: size of one audio frame in bytes.

 *

 * Clip the the buffer to the given %GstSegment.

 *

 * After calling this function the caller does not own a reference to 

 * @buffer anymore.

 *

 * Returns: %NULL if the buffer is completely outside the configured segment,

 * otherwise the clipped buffer is returned.

 *

 * If the buffer has no timestamp, it is assumed to be inside the segment and

 * is not clipped 

 *

 * Since: 0.10.14

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

GstBuffer *

gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,

    gint frame_size)

{

  GstBuffer *ret;

  GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;

  guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;

  guint8 *data;

  guint size;

  gboolean change_duration = TRUE, change_offset = TRUE, change_offset_end =

      TRUE;

  g_return_val_if_fail (segment->format == GST_FORMAT_TIME ||

      segment->format == GST_FORMAT_DEFAULT, buffer);

  g_return_val_if_fail (GST_IS_BUFFER (buffer), NULL);

  if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer))

    /* No timestamp - assume the buffer is completely in the segment */

    return buffer;

  /* Get copies of the buffer metadata to change later. 

   * Calculate the missing values for the calculations,

   * they won't be changed later though. */

  data = GST_BUFFER_DATA (buffer);

  size = GST_BUFFER_SIZE (buffer);

  timestamp = GST_BUFFER_TIMESTAMP (buffer);

  if (GST_BUFFER_DURATION_IS_VALID (buffer)) {

    duration = GST_BUFFER_DURATION (buffer);

  } else {

    change_duration = FALSE;

    duration = gst_util_uint64_scale (size / frame_size, GST_SECOND, rate);

  }

  if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {

    offset = GST_BUFFER_OFFSET (buffer);

  } else {

    change_offset = FALSE;

    offset = 0;

  }

  if (GST_BUFFER_OFFSET_END_IS_VALID (buffer)) {

    offset_end = GST_BUFFER_OFFSET_END (buffer);

  } else {

    change_offset_end = FALSE;

    offset_end = offset + size / frame_size;

  }

  if (segment->format == GST_FORMAT_TIME) {

    /* Handle clipping for GST_FORMAT_TIME */

    gint64 start, stop, cstart, cstop, diff;

    start = timestamp;

    stop = timestamp + duration;

    if (gst_segment_clip (segment, GST_FORMAT_TIME,

            start, stop, &cstart, &cstop)) {

      diff = cstart - start;

      if (diff > 0) {

        timestamp = cstart;

        if (change_duration)

          duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);

        if (change_offset)

          offset += diff;

        data += diff * frame_size;

        size -= diff * frame_size;

      }

      diff = stop - cstop;

      if (diff > 0) {

        /* duration is always valid if stop is valid */

        duration -= diff;

        diff = gst_util_uint64_scale (diff, rate, GST_SECOND);

        if (change_offset_end)

          offset_end -= diff;

        size -= diff * frame_size;

      }

    } else {

      gst_buffer_unref (buffer);

      return NULL;

    }

  } else {

    /* Handle clipping for GST_FORMAT_DEFAULT */

    gint64 start, stop, cstart, cstop, diff;

    g_return_val_if_fail (GST_BUFFER_OFFSET_IS_VALID (buffer), buffer);

    start = offset;

    stop = offset_end;

    if (gst_segment_clip (segment, GST_FORMAT_DEFAULT,

            start, stop, &cstart, &cstop)) {

      diff = cstart - start;

      if (diff > 0) {

        offset = cstart;

        timestamp = gst_util_uint64_scale (cstart, GST_SECOND, rate);

        if (change_duration)

          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        data += diff * frame_size;

        size -= diff * frame_size;

      }

      diff = stop - cstop;

      if (diff > 0) {

        offset_end = cstop;

        if (change_duration)

          duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);

        size -= diff * frame_size;

      }

    } else {

      gst_buffer_unref (buffer);

      return NULL;

    }

  }

  /* Get a metadata writable buffer and apply all changes */

  ret = gst_buffer_make_metadata_writable (buffer);

  GST_BUFFER_TIMESTAMP (ret) = timestamp;

  GST_BUFFER_SIZE (ret) = size;

  GST_BUFFER_DATA (ret) = data;

  if (change_duration)

    GST_BUFFER_DURATION (ret) = duration;

  if (change_offset)

    GST_BUFFER_OFFSET (ret) = offset;

  if (change_offset_end)

    GST_BUFFER_OFFSET_END (ret) = offset_end;

  return ret;

}