/*      $NetBSD: audiovar.h,v 1.13 2023/04/23 08:06:05 mlelstv Exp $    */

/*-
* Copyright (c) 2002 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by TAMURA Kent
*
* 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 NETBSD FOUNDATION, INC. 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 FOUNDATION 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.
*/

/*
* Copyright (c) 1991-1993 Regents of the University of California.
* All rights reserved.
*
* 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.
* 3. All advertising materials mentioning features or use of this software
*    must display the following acknowledgement:
*      This product includes software developed by the Computer Systems
*      Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
*    to endorse or promote products derived from this software without
*    specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS 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 REGENTS 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.
*
*      From: Header: audiovar.h,v 1.3 93/07/18 14:07:25 mccanne Exp  (LBL)
*/

#ifndef _SYS_DEV_AUDIO_AUDIOVAR_H_
#define _SYS_DEV_AUDIO_AUDIOVAR_H_

#include <sys/condvar.h>
#include <sys/proc.h>
#include <sys/pserialize.h>
#include <sys/psref.h>
#include <sys/queue.h>

#include <dev/audio/audio_if.h>
#include <dev/audio/audiofil.h>

/*
* Whether supports [US]LINEAR24/24 as userland format.
*/
#define AUDIO_SUPPORT_LINEAR24

/*
* Frequency range.
* For lower limit, there are some antique machines who supports under
* 4000Hz, so that we accept 1000Hz as lower limit, regardless of
* practicality(?).
* For upper limit, there are some devices who supports 384000Hz, but
* I don't have them. :-)
*/
#define AUDIO_MIN_FREQUENCY (1000)
#define AUDIO_MAX_FREQUENCY (192000)

typedef struct audio_file audio_file_t;
typedef struct audio_trackmixer audio_trackmixer_t;

/* ring buffer */
typedef struct {
       audio_format2_t fmt;    /* format */
       int  capacity;          /* capacity by frame */
       int  head;              /* head position in frame */
       int  used;              /* used frame count */
       void *mem;              /* sample ptr */
} audio_ring_t;

#define AUDIO_N_PORTS 4

struct au_mixer_ports {
       int     index;          /* index of port-selector mixerctl */
       int     master;         /* index of master mixerctl */
       int     nports;         /* number of selectable ports */
       bool    isenum;         /* selector is enum type */
       u_int   allports;       /* all aumasks or'd */
       u_int   aumask[AUDIO_N_PORTS];  /* exposed value of "ports" */
       int     misel [AUDIO_N_PORTS];  /* ord of port, for selector */
       int     miport[AUDIO_N_PORTS];  /* index of port's mixerctl */
       bool    isdual;         /* has working mixerout */
       int     mixerout;       /* ord of mixerout, for dual case */
       int     cur_port;       /* the port that gain actually controls when
                                  mixerout is selected, for dual case */
};

struct audio_softc {
       /* Myself (e.g.; audio0, audio1, ...) */
       device_t        sc_dev;

       /* Hardware device struct (e.g.; sb0, hdafg0, ...) */
       device_t        hw_dev;

       /*
        * Hardware interface and driver handle.
        * hw_if == NULL means that the device is (attached but) disabled.
        */
       const struct audio_hw_if *hw_if;
       void            *hw_hdl;

       /*
        * Properties obtained by get_props().
        * No need any locks to read this variable.
        */
       int sc_props;

       /*
        * List of opened descriptors.
        * Must be protected by sc_lock || sc_intr_lock for traversal(FOREACH).
        * Must be protected by sc_lock && sc_intr_lock for insertion/removal.
        */
       SLIST_HEAD(, audio_file) sc_files;

       /*
        * Blocksize in msec.
        * Must be protected by sc_exlock.
        */
       int sc_blk_ms;

       /*
        * Track mixer for playback and recording.
        * If null, the mixer is disabled.
        * Must be protected by sc_exlock.
        */
       audio_trackmixer_t *sc_pmixer;
       audio_trackmixer_t *sc_rmixer;

       /*
        * Opening track counter.
        * Must be protected by sc_lock && sc_exlock for modifying.
        * Must be protected by sc_lock || sc_exlock for reference.
        */
       int sc_popens;
       int sc_ropens;

       /*
        * true if the track mixer is running.
        * Must be protected by sc_exlock && sc_intr_lock for modifying.
        * Must be protected by sc_exlock || sc_intr_lock for reference.
        */
       bool sc_pbusy;
       bool sc_rbusy;

       /*
        * These four are the parameters sustained with /dev/sound.
        * Must be protected by sc_exlock.
        */
       audio_format2_t sc_sound_pparams;
       audio_format2_t sc_sound_rparams;
       bool            sc_sound_ppause;
       bool            sc_sound_rpause;

       /* recent info for /dev/sound */
       /* XXX TODO */
       struct audio_info sc_ai;

       /*
        * Playback(write)/Recording(read) selector.
        * Must be protected by sc_lock.
        */
       struct selinfo sc_wsel;
       struct selinfo sc_rsel;

       /*
        * Processes who want mixer SIGIO.
        * sc_am is an array of pids, or NULL if empty.
        * sc_am_capacity is the number of allocated elements.
        * sc_am_used is the number of elements actually used.
        * Must be protected by sc_exlock.
        */
       pid_t *sc_am;
       int sc_am_capacity;
       int sc_am_used;

       /*
        * Thread lock and interrupt lock obtained by get_locks().
        */
       kmutex_t *sc_lock;
       kmutex_t *sc_intr_lock;

       /*
        * Critical section.
        * Must be protected by sc_lock.
        */
       int sc_exlock;
       kcondvar_t sc_exlockcv;

       /*
        * Passive reference to prevent a race between detach and fileops.
        * pserialize_perform(sc_psz) must be protected by sc_lock.
        */
       pserialize_t sc_psz;
       struct psref_target sc_psref;

       /*
        * Must be protected by sc_lock (?)
        */
       bool            sc_dying;

       /*
        * Indicates that about to suspend.
        * Must be protected by sc_lock.
        */
       bool            sc_suspending;

       /*
        * If multiuser is false, other users who have different euid
        * than the first user cannot open this device.
        * Must be protected by sc_exlock.
        */
       bool sc_multiuser;
       kauth_cred_t sc_cred;

       struct sysctllog *sc_log;

       mixer_ctrl_t    *sc_mixer_state;
       int             sc_nmixer_states;
       struct au_mixer_ports sc_inports;
       struct au_mixer_ports sc_outports;
       int             sc_monitor_port;
       u_int   sc_lastgain;
};

#ifdef DIAGNOSTIC
#define DIAGNOSTIC_filter_arg(arg) audio_diagnostic_filter_arg(__func__, (arg))
#define DIAGNOSTIC_format2(fmt) audio_diagnostic_format2(__func__, (fmt))
extern void audio_diagnostic_filter_arg(const char *,
       const audio_filter_arg_t *);
extern void audio_diagnostic_format2(const char *, const audio_format2_t *);
#else
#define DIAGNOSTIC_filter_arg(arg)
#define DIAGNOSTIC_format2(fmt)
#endif

/*
* Return true if 'fmt' is the internal format.
* It does not check for frequency and number of channels.
*/
static __inline bool
audio_format2_is_internal(const audio_format2_t *fmt)
{

       if (fmt->encoding != AUDIO_ENCODING_SLINEAR_NE)
               return false;
       if (fmt->precision != AUDIO_INTERNAL_BITS)
               return false;
       if (fmt->stride != AUDIO_INTERNAL_BITS)
               return false;
       return true;
}

/*
* Return true if fmt's encoding is one of LINEAR.
*/
static __inline bool
audio_format2_is_linear(const audio_format2_t *fmt)
{
       return (fmt->encoding == AUDIO_ENCODING_SLINEAR_LE)
           || (fmt->encoding == AUDIO_ENCODING_SLINEAR_BE)
           || (fmt->encoding == AUDIO_ENCODING_ULINEAR_LE)
           || (fmt->encoding == AUDIO_ENCODING_ULINEAR_BE);
}

/*
* Return true if fmt's encoding is one of SLINEAR.
*/
static __inline bool
audio_format2_is_signed(const audio_format2_t *fmt)
{
       return (fmt->encoding == AUDIO_ENCODING_SLINEAR_LE)
           || (fmt->encoding == AUDIO_ENCODING_SLINEAR_BE);
}

/*
* Return fmt's endian as LITTLE_ENDIAN or BIG_ENDIAN.
*/
static __inline int
audio_format2_endian(const audio_format2_t *fmt)
{
       if (fmt->stride == 8) {
               /* HOST ENDIAN */
               return BYTE_ORDER;
       }

       if (fmt->encoding == AUDIO_ENCODING_SLINEAR_LE ||
           fmt->encoding == AUDIO_ENCODING_ULINEAR_LE) {
               return LITTLE_ENDIAN;
       }
       if (fmt->encoding == AUDIO_ENCODING_SLINEAR_BE ||
           fmt->encoding == AUDIO_ENCODING_ULINEAR_BE) {
               return BIG_ENDIAN;
       }
       return BYTE_ORDER;
}

/* Interfaces for audiobell. */
int audiobellopen(dev_t, audio_file_t **);
int audiobellsetrate(audio_file_t *, u_int);
int audiobellclose(audio_file_t *);
int audiobellwrite(audio_file_t *, struct uio *);

#endif /* !_SYS_DEV_AUDIO_AUDIOVAR_H_ */