gstreamer/gst-libs/gst/audio/audio.c

/* 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"

int
gst_audio_frame_byte_size (GstPad* pad)
{
/* calculate byte size of an audio frame
 * this should be moved closer to the gstreamer core
 * and be implemented for every mime type IMO
 * returns -1 if there's an error (to avoid division by zero), 
 * or the byte size if everything's ok
 */

  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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_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; 
}

long
gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
/* calculate length of buffer in frames
 * this should be moved closer to the gstreamer core
 * and be implemented for every mime type IMO
 * returns 0 if there's an error, or the number of frames if everything's ok
 */
{
  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;
}

long
gst_audio_frame_rate (GstPad *pad)
/*
 * calculate frame rate (based on caps of pad)
 * returns 0 if failed, rate if success
 */
{
  const GstCaps *caps = NULL;
  gint rate;
  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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
    return 0;
  }
  else {
    structure = gst_caps_get_structure (caps, 0);
    gst_structure_get_int  (structure, "rate", &rate);
    return rate;
  }
}

double 
gst_audio_length (GstPad* pad, GstBuffer* buf)
{
/* calculate length in seconds
 * of audio buffer buf
 * based on capabilities of pad
 */

  long bytes = 0;
  int width = 0;
  int channels = 0;
  int rate = 0;

  double 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_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
    length = 0.0;
  }
  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 * 8.0) / (double) (rate * channels * width);
  }
  /* g_print ("DEBUG: audio: returning length of %f\n", length); */
  return length;
}

long 
gst_audio_highest_sample_value (GstPad* pad)
/* calculate highest possible sample value
 * based on capabilities of pad
 */
{
  gboolean is_signed = FALSE;
  gint width = 0;
  const GstCaps *caps = NULL;
  GstStructure *structure;
  
  caps = GST_PAD_CAPS (pad);
  if (caps == NULL)
  {
    g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
  }
  
  structure = gst_caps_get_structure (caps, 0);
  gst_structure_get_int  (structure, "width", &width);
  gst_structure_get_boolean  (structure, "signed", &is_signed);
  
  if (is_signed) --width;
  /* example : 16 bit, signed : samples between -32768 and 32767 */
  return ((long) (1 << width));
}

gboolean 
gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
/* check if the buffer size is a whole multiple of the frame size */
{
  if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
    return TRUE;
  else
    return FALSE;
}

static gboolean
plugin_init (GstPlugin *plugin)
{
  return TRUE;
}

GST_PLUGIN_DEFINE (
  GST_VERSION_MAJOR,
  GST_VERSION_MINOR,
  "gstaudio",
  "Support services for audio plugins",
  plugin_init,
  VERSION,
  GST_LICENSE,
  GST_PACKAGE,
  GST_ORIGIN
);