remove gst-libs from gst-plugins module as it is in gst-plugins-base now

Original commit message from CVS:
remove gst-libs from gst-plugins module as it is in gst-plugins-base now

1 files changed, 0 insertions, 280 deletions

diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
deleted file mode 100644
index b6a1edb5..00000000
--- a/gst-libs/gst/audio/audio.c
+++ /dev/null
@@ -1,280 +0,0 @@
-/* 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 "multichannel-enumtypes.h"
-
-#include <gst/gststructure.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;
-}
-
-/* _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);
-}
-
-void
-gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
-{
-  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);
-  if (flag & GST_AUDIO_FIELD_BUFFER_FRAMES)
-    gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 1,
-        G_MAXINT, NULL);
-}
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  gst_audio_channel_position_get_type ();
-
-  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);