soundcard.h

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/*
 * soundcard.h
 *
 * Copyright by Hannu Savolainen 1993
 * Modified for the new FreeBSD sound driver by Luigi Rizzo, 1997
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2. Redistributions in binary form must reproduce the above
 *    copyright notice, this list of conditions and the following
 *    disclaimer in the documentation and/or other materials provided
 *    with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS''
 * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
 * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A
 * PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR
 * OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
 * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
 * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
 * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
 * ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
 * POSSIBILITY OF SUCH DAMAGE.
 */

#ifndef SOUNDCARD_H
#define SOUNDCARD_H
 /* 
  * If you make modifications to this file, please contact me before
  * distributing the modified version. There is already enough 
  * diversity in the world.
  *
  * Regards,
  * Hannu Savolainen
  * [e-mail]
  *
  **********************************************************************
  * PS.	The Hacker's Guide to VoxWare available from 
  *     nic.funet.fi:pub/OS/Linux/ALPHA/sound. The file is
  *	snd-sdk-doc-0.1.ps.gz (gzipped postscript). It contains
  *	some useful information about programming with VoxWare.
  *	(NOTE! The pub/OS/Linux/ALPHA/ directories are hidden. You have
  *	to cd inside them before the files are accessible.)
  **********************************************************************
  */

/*
 * SOUND_VERSION is only used by the voxware driver. Hopefully apps
 * should not depend on it, but rather look at the capabilities
 * of the driver in the kernel!
 */
#define SOUND_VERSION  301
#define VOXWARE		/* does this have any use ? */

/*
 * Supported card ID numbers (Should be somewhere else? We keep
 * them here just for compativility with the old driver, but these
 * constants are of little or no use).
 */

#define SNDCARD_ADLIB          1
#define SNDCARD_SB             2
#define SNDCARD_PAS            3
#define SNDCARD_GUS            4
#define SNDCARD_MPU401         5
#define SNDCARD_SB16           6
#define SNDCARD_SB16MIDI       7
#define SNDCARD_UART6850       8
#define SNDCARD_GUS16          9
#define SNDCARD_MSS            10
#define SNDCARD_PSS            11
#define SNDCARD_SSCAPE         12
#define SNDCARD_PSS_MPU        13
#define SNDCARD_PSS_MSS        14
#define SNDCARD_SSCAPE_MSS     15
#define SNDCARD_TRXPRO         16
#define SNDCARD_TRXPRO_SB      17
#define SNDCARD_TRXPRO_MPU     18
#define SNDCARD_MAD16          19
#define SNDCARD_MAD16_MPU      20
#define SNDCARD_CS4232         21
#define SNDCARD_CS4232_MPU     22
#define SNDCARD_MAUI           23
#define SNDCARD_PSEUDO_MSS     24
#define SNDCARD_AWE32           25
#define SNDCARD_NSS            26

#include <sys/types.h>
#ifndef _IOWR
#include <sys/ioccom.h>
#endif  /* !_IOWR */

/*
 * The first part of this file contains the new FreeBSD sound ioctl
 * interface. Tries to minimize the number of different ioctls, and
 * to be reasonably general.
 *
 * 970821: some of the new calls have not been implemented yet.
 */

/*
 * the following three calls extend the generic file descriptor
 * interface. AIONWRITE is the dual of FIONREAD, i.e. returns the max
 * number of bytes for a write operation to be non-blocking.
 *
 * AIOGSIZE/AIOSSIZE are used to change the behaviour of the device,
 * from a character device (default) to a block device. In block mode,
 * (not to be confused with blocking mode) the main difference for the
 * application is that select() will return only when a complete
 * block can be read/written to the device, whereas in character mode
 * select will return true when one byte can be exchanged. For audio
 * devices, character mode makes select almost useless since one byte
 * will always be ready by the next sample time (which is often only a
 * handful of microseconds away). 
 * Use a size of 0 or 1 to return to character mode.
 */
#define	AIONWRITE   _IOR('A', 10, int)   /* get # bytes to write */
struct snd_size {
    int play_size;
    int rec_size;
};
#define	AIOGSIZE    _IOR('A', 11, struct snd_size)/* read current blocksize */
#define	AIOSSIZE    _IOWR('A', 11, struct snd_size)  /* sets blocksize */

/*
 * The following constants define supported audio formats. The
 * encoding follows voxware conventions, i.e. 1 bit for each supported
 * format. We extend it by using bit 31 (RO) to indicate full-duplex
 * capability, and bit 29 (RO) to indicate that the card supports/
 * needs different formats on capture & playback channels.
 * Bit 29 (RW) is used to indicate/ask stereo.
 */

#   define AFMT_QUERY		0x00000000	/* Return current fmt */
#   define AFMT_MU_LAW		0x00000001
#   define AFMT_A_LAW		0x00000002
#   define AFMT_IMA_ADPCM	0x00000004
#   define AFMT_U8		0x00000008
#   define AFMT_S16_LE		0x00000010	/* Little endian signed 16*/
#   define AFMT_S16_BE		0x00000020	/* Big endian signed 16 */
#   define AFMT_S8		0x00000040
#   define AFMT_U16_LE		0x00000080	/* Little endian U16 */
#   define AFMT_U16_BE		0x00000100	/* Big endian U16 */
#   define AFMT_MPEG		0x00000200	/* MPEG (2) audio */

#   define AFMT_STEREO		0x10000000	/* can do/want stereo	*/

/*
 * the following are really capabilities
 */
#   define AFMT_WEIRD		0x20000000	/* weird hardware...	*/
    /*
     * AFMT_WEIRD reports that the hardware might need to operate
     * with different formats in the playback and capture
     * channels when operating in full duplex.
     * As an example, SoundBlaster16 cards only support U8 in one
     * direction and S16 in the other one, and applications should
     * be aware of this limitation.
     */
#   define AFMT_FULLDUPLEX	0x80000000	/* can do full duplex	*/

/*
 * The following structure is used to get/set format and sampling rate.
 * While it would be better to have things such as stereo, bits per
 * sample, endiannes, etc split in different variables, it turns out
 * that formats are not that many, and not all combinations are possible.
 * So we followed the Voxware approach of associating one bit to each
 * format.
 */

typedef struct _snd_chan_param {
    u_long	play_rate;	/* sampling rate			*/
    u_long	rec_rate;	/* sampling rate			*/
    u_long	play_format;	/* everything describing the format	*/
    u_long	rec_format;	/* everything describing the format	*/
} snd_chan_param;
#define	AIOGFMT    _IOR('f', 12, snd_chan_param)   /* get format */
#define	AIOSFMT    _IOWR('f', 12, snd_chan_param)  /* sets format */

/*
 * The following structure is used to get/set the mixer setting.
 * Up to 32 mixers are supported, each one with up to 32 channels.
 */
typedef struct _snd_mix_param {
    u_char	subdev;	/* which output				*/
    u_char	line;	/* which input				*/
    u_char	left,right; /* volumes, 0..255, 0 = mute	*/
} snd_mix_param ;

/* XXX AIOGMIX, AIOSMIX not implemented yet */
#define AIOGMIX	_IOWR('A', 13, snd_mix_param)	/* return mixer status */
#define AIOSMIX	_IOWR('A', 14, snd_mix_param)	/* sets mixer status   */

/*
 * channel specifiers used in AIOSTOP and AIOSYNC
 */
#define	AIOSYNC_PLAY	0x1	/* play chan */
#define	AIOSYNC_CAPTURE	0x2	/* capture chan */
/* AIOSTOP stop & flush a channel, returns the residual count */
#define	AIOSTOP	_IOWR ('A', 15, int)

/* alternate method used to notify the sync condition */
#define	AIOSYNC_SIGNAL	0x100
#define	AIOSYNC_SELECT	0x200

/* what the 'pos' field refers to */
#define AIOSYNC_READY	0x400
#define AIOSYNC_FREE	0x800

typedef struct _snd_sync_parm {
    long chan ; /* play or capture channel, plus modifier */
    long pos;
} snd_sync_parm;
#define	AIOSYNC	_IOWR ('A', 15, snd_sync_parm)	/* misc. synchronization */

/*
 * The following is used to return device capabilities. If the structure
 * passed to the ioctl is zeroed, default values are returned for rate
 * and formats, a bitmap of available mixers is returned, and values
 * (inputs, different levels) for the first one are returned.
 *
 * If  formats, mixers, inputs are instantiated, then detailed info
 * are returned depending on the call.
 */
typedef struct _snd_capabilities {
    u_long	rate_min, rate_max;	/* min-max sampling rate */
    u_long	formats;
    u_long	bufsize; /* DMA buffer size */
    u_long	mixers; /* bitmap of available mixers */
    u_long	inputs; /* bitmap of available inputs (per mixer) */
    u_short	left, right;	/* how many levels are supported */
} snd_capabilities;
#define AIOGCAP	_IOWR('A', 15, snd_capabilities)	/* get capabilities */

/*
 * here is the old (Voxware) ioctl interface
 */

/*
 * IOCTL Commands for /dev/sequencer
 */

#define SNDCTL_SEQ_RESET	_IO  ('Q', 0)
#define SNDCTL_SEQ_SYNC		_IO  ('Q', 1)
#define SNDCTL_SYNTH_INFO	_IOWR('Q', 2, struct synth_info)
#define SNDCTL_SEQ_CTRLRATE	_IOWR('Q', 3, int) /* Set/get timer res.(hz) */
#define SNDCTL_SEQ_GETOUTCOUNT	_IOR ('Q', 4, int)
#define SNDCTL_SEQ_GETINCOUNT	_IOR ('Q', 5, int)
#define SNDCTL_SEQ_PERCMODE	_IOW ('Q', 6, int)
#define SNDCTL_FM_LOAD_INSTR	_IOW ('Q', 7, struct sbi_instrument)	/* Valid for FM only */
#define SNDCTL_SEQ_TESTMIDI	_IOW ('Q', 8, int)
#define SNDCTL_SEQ_RESETSAMPLES	_IOW ('Q', 9, int)
#define SNDCTL_SEQ_NRSYNTHS	_IOR ('Q',10, int)
#define SNDCTL_SEQ_NRMIDIS	_IOR ('Q',11, int)
#define SNDCTL_MIDI_INFO	_IOWR('Q',12, struct midi_info)
#define SNDCTL_SEQ_THRESHOLD	_IOW ('Q',13, int)
#define SNDCTL_SEQ_TRESHOLD	SNDCTL_SEQ_THRESHOLD	/* there was once a typo */
#define SNDCTL_SYNTH_MEMAVL	_IOWR('Q',14, int) /* in=dev#, out=memsize */
#define SNDCTL_FM_4OP_ENABLE	_IOW ('Q',15, int) /* in=dev# */
#define SNDCTL_PMGR_ACCESS	_IOWR('Q',16, struct patmgr_info)
#define SNDCTL_SEQ_PANIC	_IO  ('Q',17)
#define SNDCTL_SEQ_OUTOFBAND	_IOW ('Q',18, struct seq_event_rec)

struct seq_event_rec {
	u_char arr[8];
};

#define SNDCTL_TMR_TIMEBASE	_IOWR('T', 1, int)
#define SNDCTL_TMR_START	_IO  ('T', 2)
#define SNDCTL_TMR_STOP		_IO  ('T', 3)
#define SNDCTL_TMR_CONTINUE	_IO  ('T', 4)
#define SNDCTL_TMR_TEMPO	_IOWR('T', 5, int)
#define SNDCTL_TMR_SOURCE	_IOWR('T', 6, int)
#   define TMR_INTERNAL		0x00000001
#   define TMR_EXTERNAL		0x00000002
#	define TMR_MODE_MIDI	0x00000010
#	define TMR_MODE_FSK	0x00000020
#	define TMR_MODE_CLS	0x00000040
#	define TMR_MODE_SMPTE	0x00000080
#define SNDCTL_TMR_METRONOME	_IOW ('T', 7, int)
#define SNDCTL_TMR_SELECT	_IOW ('T', 8, int)

/*
 *	Endian aware patch key generation algorithm.
 */

#if defined(_AIX) || defined(AIX)
#  define _PATCHKEY(id) (0xfd00|id)
#else
#  define _PATCHKEY(id) ((id<<8)|0xfd)
#endif

/*
 *	Sample loading mechanism for internal synthesizers (/dev/sequencer)
 *	The following patch_info structure has been designed to support
 *	Gravis UltraSound. It tries to be universal format for uploading
 *	sample based patches but is probably too limited.
 */

struct patch_info {
/*		u_short key;		 Use GUS_PATCH here */
	short key;		 /* Use GUS_PATCH here */
#define GUS_PATCH	_PATCHKEY(0x04)
#define OBSOLETE_GUS_PATCH	_PATCHKEY(0x02)

	short device_no;	/* Synthesizer number */
	short instr_no;		/* Midi pgm# */

	u_long mode;
/*
 * The least significant byte has the same format than the GUS .PAT
 * files
 */
#define WAVE_16_BITS	0x01	/* bit 0 = 8 or 16 bit wave data. */
#define WAVE_UNSIGNED	0x02	/* bit 1 = Signed - Unsigned data. */
#define WAVE_LOOPING	0x04	/* bit 2 = looping enabled-1. */
#define WAVE_BIDIR_LOOP	0x08	/* bit 3 = Set is bidirectional looping. */
#define WAVE_LOOP_BACK	0x10	/* bit 4 = Set is looping backward. */
#define WAVE_SUSTAIN_ON	0x20	/* bit 5 = Turn sustaining on. (Env. pts. 3)*/
#define WAVE_ENVELOPES	0x40	/* bit 6 = Enable envelopes - 1 */
				/* 	(use the env_rate/env_offs fields). */
/* Linux specific bits */
#define WAVE_VIBRATO	0x00010000	/* The vibrato info is valid */
#define WAVE_TREMOLO	0x00020000	/* The tremolo info is valid */
#define WAVE_SCALE	0x00040000	/* The scaling info is valid */
/* Other bits must be zeroed */

	long len;	/* Size of the wave data in bytes */
	long loop_start, loop_end; /* Byte offsets from the beginning */

/* 
 * The base_freq and base_note fields are used when computing the
 * playback speed for a note. The base_note defines the tone frequency
 * which is heard if the sample is played using the base_freq as the
 * playback speed.
 *
 * The low_note and high_note fields define the minimum and maximum note