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Diffstat (limited to 'gst-libs/gst/audio/audio.h')
| -rw-r--r-- | gst-libs/gst/audio/audio.h | 131 |
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__ */ |