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, 131 deletions

diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
deleted file mode 100644
index 8556ce1f..00000000
--- a/gst-libs/gst/audio/audio.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/* GStreamer
- * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
- * Library       <2001> Thomas Vander Stichele <thomas@apestaart.org>
- *
- * 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.
- */
-
-#include <gst/gst.h>
-
-#include <gst/audio/audioclock.h>
-
-#ifndef __GST_AUDIO_AUDIO_H__
-#define __GST_AUDIO_AUDIO_H__
-
-G_BEGIN_DECLS
-
-/* For people that are looking at this source: the purpose of these defines is
- * to make GstCaps a bit easier, in that you don't have to know all of the
- * properties that need to be defined. you can just use these macros. currently
- * (8/01) the only plugins that use these are the passthrough, speed, volume,
- * adder, and [de]interleave plugins. These are for convenience only, and do not
- * specify the 'limits' of GStreamer. you might also use these definitions as a
- * base for making your own caps, if need be.
- *
- * For example, to make a source pad that can output streams of either mono
- * float or any channel int:
- *
- *  template = gst_pad_template_new
- *    ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
- *    gst_caps_append(gst_caps_new ("sink_int",  "audio/x-raw-int",
- *                                  GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
- *                    gst_caps_new ("sink_float", "audio/x-raw-float",
- *                                  GST_AUDIO_FLOAT_PAD_TEMPLATE_PROPS)),
- *    NULL);
- *
- *  sinkpad = gst_pad_new_from_template(template, "sink");
- *
- * Andy Wingo, 18 August 2001
- * Thomas, 6 September 2002 */
-
-#define GST_AUDIO_DEF_RATE 44100
-
-#define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
-  "audio/x-raw-int, " \
-  "rate = (int) [ 1, MAX ], " \
-  "channels = (int) [ 1, MAX ], " \
-  "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
-  "width = (int) { 8, 16, 24, 32 }, " \
-  "depth = (int) [ 1, 32 ], " \
-  "signed = (boolean) { true, false }" 
-
-
-/* "standard" int audio is native order, 16 bit stereo. */
-#define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
-  "audio/x-raw-int, " \
-  "rate = (int) [ 1, MAX ], " \
-  "channels = (int) 2, " \
-  "endianness = (int) BYTE_ORDER, " \
-  "width = (int) 16, " \
-  "depth = (int) 16, " \
-  "signed = (boolean) true" 
-
-#define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
-  "audio/x-raw-float, " \
-  "rate = (int) [ 1, MAX ], " \
-  "channels = (int) [ 1, MAX ], " \
-  "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
-  "width = (int) { 32, 64 }, " \
-  "buffer-frames = (int) [ 1, MAX]"
-
-/* "standard" float audio is native order, 32 bit mono. */
-#define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
-  "audio/x-raw-float, " \
-  "width = (int) 32, " \
-  "rate = (int) [ 1, MAX ], " \
-  "channels = (int) 1, " \
-  "endianness = (int) BYTE_ORDER, " \
-  "buffer-frames = (int) [ 1, MAX]"
-
-/*
- * this library defines and implements some helper functions for audio
- * handling
- */
-
-/* get byte size of audio frame (based on caps of pad */
-int      gst_audio_frame_byte_size      (GstPad* pad);
-
-/* get length in frames of buffer */
-long     gst_audio_frame_length         (GstPad* pad, GstBuffer* buf);
-
-/* get frame rate based on caps */
-long     gst_audio_frame_rate           (GstPad *pad);
-
-/* calculate length in seconds of audio buffer buf based on caps of pad */
-double   gst_audio_length               (GstPad* pad, GstBuffer* buf);
-
-/* calculate highest possible sample value based on capabilities of pad */
-long     gst_audio_highest_sample_value (GstPad* pad);
-
-/* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed     (GstPad* pad, GstBuffer* buf);
-
-/* functions useful for _getcaps functions */
-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),
-  GST_AUDIO_FIELD_BUFFER_FRAMES = (1 << 6)
-} GstAudioFieldFlag;
-
-void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
-
-G_END_DECLS
-
-#endif /* __GST_AUDIO_AUDIO_H__ */