/* 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>

#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 */

/* conversion macros */

/**

 * GST_FRAMES_TO_CLOCK_TIME:

 * @frames: sample frames

 * @rate: sampling rate

 * 

 * Calculate clocktime from sample @frames and @rate.

 */

#define GST_FRAMES_TO_CLOCK_TIME(frames, rate) \

  ((GstClockTime) gst_util_uint64_scale (frames, GST_SECOND, rate))

/**

 * GST_CLOCK_TIME_TO_FRAMES:

 * @clocktime: clock time

 * @rate: sampling rate

 * 

 * Calculate frames from @clocktime and sample @rate.

 */

#define GST_CLOCK_TIME_TO_FRAMES(clocktime, rate) \

  gst_util_uint64_scale (clocktime, rate, GST_SECOND)

/**

 * GST_AUDIO_DEF_RATE:

 * 

 * Standard sampling rate used in consumer audio.

 */

#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 }"

#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 }"

#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"

/*

 * this library defines and implements some helper functions for audio

 * handling

 */

/* get byte size of audio frame (based on caps of pad */

#ifdef __SYMBIAN32__

IMPORT_C

#endif

int      gst_audio_frame_byte_size      (GstPad* pad);

/* get length in frames of buffer */

#ifdef __SYMBIAN32__

IMPORT_C

#endif

long     gst_audio_frame_length         (GstPad* pad, GstBuffer* buf);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstClockTime gst_audio_duration_from_pad_buffer (GstPad * pad, GstBuffer * buf);

/* check if the buffer size is a whole multiple of the frame size */

#ifdef __SYMBIAN32__

IMPORT_C

#endif

gboolean gst_audio_is_buffer_framed     (GstPad* pad, GstBuffer* buf);

/* functions useful for _getcaps functions */

/**

 * GstAudioFieldFlag:

 *

 * Do not use anymore.

 *

 * Deprecated: use gst_structure_set() directly

 */

#ifndef GST_DISABLE_DEPRECATED

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),

} GstAudioFieldFlag;

#endif

#ifndef GST_DISABLE_DEPRECATED

void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);

#endif /* GST_DISABLE_DEPRECATED */

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstBuffer *gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint frame_size);

G_END_DECLS

#endif /* __GST_AUDIO_AUDIO_H__ */