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/* 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, 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, " \
"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__ */
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