1208 lines
30 KiB
C
1208 lines
30 KiB
C
/*
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* Copyright (c) 1991, 1992, 1993
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* The Regents of the University of California. All rights reserved.
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*
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* This software was developed by the Computer Systems Engineering group
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* at Lawrence Berkeley Laboratory under DARPA contract BG 91-66 and
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* contributed to Berkeley.
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*
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* All advertising materials mentioning features or use of this software
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* must display the following acknowledgement:
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* This product includes software developed by the University of
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* California, Lawrence Berkeley Laboratory.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. All advertising materials mentioning features or use of this software
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* must display the following acknowledgement:
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* This product includes software developed by the University of
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* California, Berkeley and its contributors.
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* 4. Neither the name of the University nor the names of its contributors
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* may be used to endorse or promote products derived from this software
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* without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*
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* @(#)bsd_audio.c 8.1 (Berkeley) 6/11/93
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*
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* from: Header: bsd_audio.c,v 1.18 93/04/24 16:20:35 leres Exp (LBL)
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* $Id: bsd_audio.c,v 1.1 1993/10/02 10:22:33 deraadt Exp $
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*/
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#include "bsdaudio.h"
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#if NBSDAUDIO > 0
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#include <sys/param.h>
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#include <sys/systm.h>
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#if BSD < 199103
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#ifndef SUNOS
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#define SUNOS
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#endif
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#endif
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#include <sys/errno.h>
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#include <sys/file.h>
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#include <sys/proc.h>
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#include <sys/user.h>
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#include <sys/vnode.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#ifndef SUNOS
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#include <sys/tty.h>
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#endif
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#include <sys/uio.h>
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#ifdef SUNOS
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#include <sundev/mbvar.h>
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#include <sun4c/intreg.h>
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#else
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#include <sys/device.h>
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#include <machine/autoconf.h>
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#endif
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#include <machine/cpu.h>
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/*
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* Avoid name clashes with SunOS so we can config either the bsd or sun
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* streams driver in a SunOS kernel.
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*/
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#ifdef SUNOS
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#include <sbusdev/bsd_audioreg.h>
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#include <sbusdev/bsd_audiovar.h>
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#include <sbusdev/bsd_audioio.h>
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struct selinfo {
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struct proc *si_proc;
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int si_coll;
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};
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#else
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#include <sparc/dev/bsd_audioreg.h>
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#include <sparc/dev/bsd_audiovar.h>
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#include <machine/bsd_audioio.h>
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#endif
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#ifdef SUNOS
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#include "bsd_audiocompat.h"
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#endif
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/*
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* Initial/default block size is patchable.
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*/
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int audio_blocksize = DEFBLKSIZE;
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int audio_backlog = 400; /* 50ms in samples */
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/*
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* Software state, per AMD79C30 audio chip.
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*/
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struct audio_softc {
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#ifndef SUNOS
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struct device sc_dev; /* base device */
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struct intrhand sc_hwih; /* hardware interrupt vector */
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struct intrhand sc_swih; /* software interrupt vector */
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#endif
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int sc_interrupts; /* number of interrupts taken */
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int sc_open; /* single use device */
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u_long sc_wseek; /* timestamp of last frame written */
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u_long sc_rseek; /* timestamp of last frame read */
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struct mapreg sc_map; /* current contents of map registers */
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struct selinfo sc_wsel; /* write selector */
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struct selinfo sc_rsel; /* read selector */
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/*
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* keep track of levels so we don't have to convert back from
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* MAP gain constants
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*/
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int sc_rlevel; /* record level */
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int sc_plevel; /* play level */
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int sc_mlevel; /* monitor level */
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/* sc_au is special in that the hardware interrupt handler uses it */
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struct auio sc_au; /* recv and xmit buffers, etc */
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};
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/* interrupt interfaces */
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#ifndef AUDIO_C_HANDLER
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int audiohwintr __P((void *));
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#endif
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int audioswintr __P((void *));
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/* forward declarations */
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int audio_sleep __P((struct aucb *, int));
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void audio_setmap __P((volatile struct amd7930 *, struct mapreg *));
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static void init_amd();
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#if !defined(AUDIO_C_HANDLER) || defined(SUNOS)
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struct auio *audio_au;
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extern void audio_trap();
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#endif
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#ifdef SUNOS
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struct audio_softc audio_softc;
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#define SOFTC(dev) &audio_softc
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#define UIOMOVE(cp, len, code, uio) uiomove(cp, len, code, uio)
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#define AUDIOOPEN(d, f, i, p)\
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audioopen(d, f, i)\
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dev_t d; int f, i;
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#define AUDIOCLOSE(d, f, i, p)\
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audioclose(d, f, i)\
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dev_t d; int f, i;
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#define AUDIOREAD(d, u, f) \
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audioread(d, u) dev_t d; struct uio *u;
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#define AUDIOWRITE(d, u, f) \
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audiowrite(d, u) dev_t d; struct uio *u;
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#define AUDIOIOCTL(d, c, a, f, o)\
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audioioctl(d, c, a, f)\
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dev_t d; int c; caddr_t a; int f;
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#define AUDIOSELECT(d, r, p)\
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audio_select(d, r, p)\
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dev_t d; int r; struct proc *p;
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#define AUDIO_SET_SWINTR set_intreg(IR_SOFT_INT4, 1)
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int
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audioselect(dev, rw)
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register dev_t dev;
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int rw;
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{
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return (audio_select(dev, rw, u.u_procp));
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}
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static void
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selrecord(p, si)
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struct proc *p;
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struct selinfo *si;
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{
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if (si->si_proc != 0)
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si->si_coll = 1;
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else
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si->si_proc = p;
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}
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#define SELWAKEUP(si) \
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{\
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if ((si)->si_proc != 0) {\
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selwakeup((si)->si_proc, (si)->si_coll); \
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(si)->si_proc = 0;\
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(si)->si_coll = 0;\
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}\
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}
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static int audioattach();
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static int audioidentify();
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struct dev_ops bsdaudio_ops = {
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0,
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audioidentify,
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audioattach,
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};
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static int
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audioidentify(cp)
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char *cp;
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{
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return (strcmp(cp, "audio") == 0);
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}
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static int
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audioattach(dev)
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struct dev_info *dev;
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{
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register struct audio_softc *sc;
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register volatile struct amd7930 *amd;
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struct dev_reg *reg;
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sc = &audio_softc;
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if (dev->devi_nreg != 1 || dev->devi_nintr != 1) {
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printf("audio: bad config\n");
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return (-1);
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}
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reg = dev->devi_reg;
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amd = (struct amd7930 *)map_regs(reg->reg_addr, reg->reg_size,
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reg->reg_bustype);
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sc->sc_au.au_amd = amd;
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init_amd(amd);
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audio_au = &sc->sc_au;
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#ifndef AUDIO_C_HANDLER
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settrap(dev->devi_intr->int_pri, audio_trap);
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#else
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/* XXX */
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addintr(dev->devi_intr->int_pri, audiohwintr, dev->devi_name,
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dev->devi_unit);
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#endif
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addintr(4, audioswintr, dev->devi_name, dev->devi_unit);
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report_dev(dev);
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return (0);
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}
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#else
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#define AUDIOOPEN(d, f, i, p) audioopen(dev_t d, int f, int i, struct proc *p)
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#define AUDIOCLOSE(d, f, i, p) audioclose(dev_t d, int f, int i, \
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struct proc *p)
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#define AUDIOREAD(d, u, f) audioread(dev_t d, struct uio *u, int f)
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#define AUDIOWRITE(d, u, f) audiowrite(dev_t d, struct uio *u, int f)
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#define AUDIOIOCTL(d, c, a, f, o)\
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audioioctl(dev_t dev, int c, caddr_t a, int f, struct proc *p)
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#define AUDIOSELECT(d, r, p) audioselect(dev_t dev, int rw, struct proc *p)
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#define SELWAKEUP selwakeup
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#define AUDIO_SET_SWINTR ienab_bis(IE_L6)
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/* autoconfiguration driver */
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void audioattach(struct device *, struct device *, void *);
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struct cfdriver audiocd =
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{ NULL, "audio", matchbyname, audioattach,
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DV_DULL, sizeof(struct audio_softc) };
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#define SOFTC(dev) audiocd.cd_devs[minor(dev)]
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#define UIOMOVE(cp, len, code, uio) uiomove(cp, len, uio)
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/*
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* Audio chip found.
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*/
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void
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audioattach(parent, self, args)
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struct device *parent, *self;
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void *args;
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{
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register struct audio_softc *sc = (struct audio_softc *)self;
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register struct romaux *ra = args;
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register volatile struct amd7930 *amd;
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register int pri;
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if (ra->ra_nintr != 1) {
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printf(": expected 1 interrupt, got %d\n", ra->ra_nintr);
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return;
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}
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pri = ra->ra_intr[0].int_pri;
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printf(" pri %d, softpri %d\n", pri, PIL_AUSOFT);
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amd = (volatile struct amd7930 *)(ra->ra_vaddr ?
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ra->ra_vaddr : mapiodev(ra->ra_paddr, sizeof *amd));
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sc->sc_au.au_amd = amd;
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init_amd(amd);
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#ifndef AUDIO_C_HANDLER
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audio_au = &sc->sc_au;
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intr_fasttrap(pri, audio_trap);
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#else
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sc->sc_hwih.ih_fun = audiohwintr;
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sc->sc_hwih.ih_arg = &sc->sc_au;
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intr_establish(pri, &sc->sc_hwih);
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#endif
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sc->sc_swih.ih_fun = audioswintr;
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sc->sc_swih.ih_arg = sc;
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intr_establish(PIL_AUSOFT, &sc->sc_swih);
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}
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#endif
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static void
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init_amd(amd)
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register volatile struct amd7930 *amd;
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{
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/* disable interrupts */
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amd->cr = AMDR_INIT;
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amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE;
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/*
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* Initialize the mux unit. We use MCR3 to route audio (MAP)
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* through channel Bb. MCR1 and MCR2 are unused.
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* Setting the INT enable bit in MCR4 will generate an interrupt
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* on each converted audio sample.
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*/
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amd->cr = AMDR_MUX_1_4;
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amd->dr = 0;
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amd->dr = 0;
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amd->dr = (AMD_MCRCHAN_BB << 4) | AMD_MCRCHAN_BA;
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amd->dr = AMD_MCR4_INT_ENABLE;
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}
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static int audio_default_level = 150;
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static void ausetrgain __P((struct audio_softc *, int));
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static void ausetpgain __P((struct audio_softc *, int));
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static void ausetmgain __P((struct audio_softc *, int));
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static int audiosetinfo __P((struct audio_softc *, struct audio_info *));
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static int audiogetinfo __P((struct audio_softc *, struct audio_info *));
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struct sun_audio_info;
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static int sunaudiosetinfo __P((struct audio_softc *,
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struct sun_audio_info *));
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static int sunaudiogetinfo __P((struct audio_softc *,
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struct sun_audio_info *));
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static void audio_setmmr2 __P((volatile struct amd7930 *, int));
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/* ARGSUSED */
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int
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AUDIOOPEN(dev, flags, ifmt, p)
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{
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register struct audio_softc *sc;
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register volatile struct amd7930 *amd;
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int unit = minor(dev);
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#ifdef SUNOS
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if (unit > 0)
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return (ENXIO);
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sc = &audio_softc;
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#else
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if (unit >= audiocd.cd_ndevs || (sc = audiocd.cd_devs[unit]) == NULL)
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return (ENXIO);
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#endif
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if (sc->sc_open)
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return (EBUSY);
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sc->sc_open = 1;
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sc->sc_au.au_lowat = audio_blocksize;
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sc->sc_au.au_hiwat = AUCB_SIZE - sc->sc_au.au_lowat;
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sc->sc_au.au_blksize = audio_blocksize;
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sc->sc_au.au_backlog = audio_backlog;
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/* set up read and write blocks and `dead sound' zero value. */
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AUCB_INIT(&sc->sc_au.au_rb);
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sc->sc_au.au_rb.cb_thresh = AUCB_SIZE;
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AUCB_INIT(&sc->sc_au.au_wb);
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sc->sc_au.au_wb.cb_thresh = -1;
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/* nothing read or written yet */
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sc->sc_rseek = 0;
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sc->sc_wseek = 0;
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bzero((char *)&sc->sc_map, sizeof sc->sc_map);
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/* default to speaker */
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sc->sc_map.mr_mmr2 = AMD_MMR2_AINB | AMD_MMR2_LS;
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/* enable interrupts and set parameters established above */
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amd = sc->sc_au.au_amd;
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audio_setmmr2(amd, sc->sc_map.mr_mmr2);
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ausetrgain(sc, audio_default_level);
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ausetpgain(sc, audio_default_level);
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ausetmgain(sc, 0);
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amd->cr = AMDR_INIT;
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amd->dr = AMD_INIT_PMS_ACTIVE;
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return (0);
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}
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static int
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audio_drain(sc)
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register struct audio_softc *sc;
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{
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register int error;
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while (!AUCB_EMPTY(&sc->sc_au.au_wb))
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if ((error = audio_sleep(&sc->sc_au.au_wb, 0)) != 0)
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return (error);
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return (0);
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}
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/*
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* Close an audio chip.
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*/
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/* ARGSUSED */
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int
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AUDIOCLOSE(dev, flags, ifmt, p)
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{
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register struct audio_softc *sc = SOFTC(dev);
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register volatile struct amd7930 *amd;
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register struct aucb *cb;
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register int s;
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/*
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* Block until output drains, but allow ^C interrupt.
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*/
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sc->sc_au.au_lowat = 0; /* avoid excessive wakeups */
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s = splaudio();
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/*
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* If there is pending output, let it drain (unless
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* the output is paused).
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*/
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cb = &sc->sc_au.au_wb;
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if (!AUCB_EMPTY(cb) && !cb->cb_pause)
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(void)audio_drain(sc);
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/*
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* Disable interrupts, clear open flag, and done.
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*/
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amd = sc->sc_au.au_amd;
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amd->cr = AMDR_INIT;
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amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE;
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splx(s);
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sc->sc_open = 0;
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return (0);
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}
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int
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audio_sleep(cb, thresh)
|
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register struct aucb *cb;
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register int thresh;
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{
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register int error;
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register int s = splaudio();
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cb->cb_thresh = thresh;
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error = tsleep((caddr_t)cb, (PZERO + 1) | PCATCH, "audio", 0);
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splx(s);
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return (error);
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}
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|
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/* ARGSUSED */
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int
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AUDIOREAD(dev, uio, ioflag)
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{
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register struct audio_softc *sc = SOFTC(dev);
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register struct aucb *cb;
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register int n, head, taildata, error;
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register int blocksize = sc->sc_au.au_blksize;
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if (uio->uio_resid == 0)
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return (0);
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cb = &sc->sc_au.au_rb;
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error = 0;
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cb->cb_drops = 0;
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sc->sc_rseek = sc->sc_au.au_stamp - AUCB_LEN(cb);
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do {
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while (AUCB_LEN(cb) < blocksize) {
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#ifndef SUNOS
|
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if (ioflag & IO_NDELAY) {
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error = EWOULDBLOCK;
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return (error);
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}
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#endif
|
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if ((error = audio_sleep(cb, blocksize)) != 0)
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return (error);
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}
|
|
/*
|
|
* The space calculation can only err on the short
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* side if an interrupt occurs during processing:
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* only cb_tail is altered in the interrupt code.
|
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*/
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head = cb->cb_head;
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if ((n = AUCB_LEN(cb)) > uio->uio_resid)
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n = uio->uio_resid;
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taildata = AUCB_SIZE - head;
|
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if (n > taildata) {
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error = UIOMOVE((caddr_t)cb->cb_data + head,
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taildata, UIO_READ, uio);
|
|
if (error == 0)
|
|
error = UIOMOVE((caddr_t)cb->cb_data,
|
|
n - taildata, UIO_READ, uio);
|
|
} else
|
|
error = UIOMOVE((caddr_t)cb->cb_data + head, n,
|
|
UIO_READ, uio);
|
|
if (error)
|
|
break;
|
|
head = AUCB_MOD(head + n);
|
|
cb->cb_head = head;
|
|
} while (uio->uio_resid >= blocksize);
|
|
|
|
return (error);
|
|
}
|
|
|
|
/* ARGSUSED */
|
|
int
|
|
AUDIOWRITE(dev, uio, ioflag)
|
|
{
|
|
register struct audio_softc *sc = SOFTC(dev);
|
|
register struct aucb *cb = &sc->sc_au.au_wb;
|
|
register int n, tail, tailspace, error, first, watermark;
|
|
|
|
error = 0;
|
|
first = 1;
|
|
while (uio->uio_resid > 0) {
|
|
watermark = sc->sc_au.au_hiwat;
|
|
while (AUCB_LEN(cb) > watermark) {
|
|
#ifndef SUNOS
|
|
if (ioflag & IO_NDELAY) {
|
|
error = EWOULDBLOCK;
|
|
return (error);
|
|
}
|
|
#endif
|
|
if ((error = audio_sleep(cb, watermark)) != 0)
|
|
return (error);
|
|
watermark = sc->sc_au.au_lowat;
|
|
}
|
|
/*
|
|
* The only value that can change on an interrupt is
|
|
* cb->cb_head. We only pull that out once to decide
|
|
* how much to write into cb_data; if we lose a race
|
|
* and cb_head changes, we will merely be overly
|
|
* conservative. For a legitimate time stamp,
|
|
* however, we need to synchronize the accesses to
|
|
* au_stamp and cb_head at a high ipl below.
|
|
*/
|
|
tail = cb->cb_tail;
|
|
if ((n = (AUCB_SIZE - 1) - AUCB_LEN(cb)) > uio->uio_resid) {
|
|
n = uio->uio_resid;
|
|
if (cb->cb_head == tail &&
|
|
n <= sc->sc_au.au_blksize &&
|
|
sc->sc_au.au_stamp - sc->sc_wseek > 400) {
|
|
/*
|
|
* the write is 'small', the buffer is empty
|
|
* and we have been silent for at least 50ms
|
|
* so we might be dealing with an application
|
|
* that writes frames synchronously with
|
|
* reading them. If so, we need an output
|
|
* backlog to cover scheduling delays or
|
|
* there will be gaps in the sound output.
|
|
* Also take this opportunity to reset the
|
|
* buffer pointers in case we ended up on
|
|
* a bad boundary (odd byte, blksize bytes
|
|
* from end, etc.).
|
|
*/
|
|
register u_int* ip;
|
|
register int muzero = 0x7f7f7f7f;
|
|
register int i = splaudio();
|
|
cb->cb_head = cb->cb_tail = 0;
|
|
splx(i);
|
|
tail = sc->sc_au.au_backlog;
|
|
ip = (u_int*)cb->cb_data;
|
|
for (i = tail >> 2; --i >= 0; )
|
|
*ip++ = muzero;
|
|
}
|
|
}
|
|
tailspace = AUCB_SIZE - tail;
|
|
if (n > tailspace) {
|
|
/* write first part at tail and rest at head */
|
|
error = UIOMOVE((caddr_t)cb->cb_data + tail,
|
|
tailspace, UIO_WRITE, uio);
|
|
if (error == 0)
|
|
error = UIOMOVE((caddr_t)cb->cb_data,
|
|
n - tailspace, UIO_WRITE, uio);
|
|
} else
|
|
error = UIOMOVE((caddr_t)cb->cb_data + tail, n,
|
|
UIO_WRITE, uio);
|
|
if (error)
|
|
break;
|
|
|
|
tail = AUCB_MOD(tail + n);
|
|
if (first) {
|
|
register int s = splaudio();
|
|
sc->sc_wseek = AUCB_LEN(cb) + sc->sc_au.au_stamp + 1;
|
|
/*
|
|
* To guarantee that a write is contiguous in the
|
|
* sample space, we clear the drop count the first
|
|
* time through. If we later get drops, we will
|
|
* break out of the loop below, before writing
|
|
* a new frame.
|
|
*/
|
|
cb->cb_drops = 0;
|
|
cb->cb_tail = tail;
|
|
splx(s);
|
|
first = 0;
|
|
} else {
|
|
if (cb->cb_drops != 0)
|
|
break;
|
|
cb->cb_tail = tail;
|
|
}
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
/* Sun audio compatibility */
|
|
struct sun_audio_prinfo {
|
|
u_int sample_rate;
|
|
u_int channels;
|
|
u_int precision;
|
|
u_int encoding;
|
|
u_int gain;
|
|
u_int port;
|
|
u_int reserved0[4];
|
|
u_int samples;
|
|
u_int eof;
|
|
u_char pause;
|
|
u_char error;
|
|
u_char waiting;
|
|
u_char reserved1[3];
|
|
u_char open;
|
|
u_char active;
|
|
};
|
|
struct sun_audio_info {
|
|
struct sun_audio_prinfo play;
|
|
struct sun_audio_prinfo record;
|
|
u_int monitor_gain;
|
|
u_int reserved[4];
|
|
};
|
|
|
|
#ifndef SUNOS
|
|
#define SUNAUDIO_GETINFO _IOR('A', 1, struct sun_audio_info)
|
|
#define SUNAUDIO_SETINFO _IOWR('A', 2, struct sun_audio_info)
|
|
#else
|
|
#define SUNAUDIO_GETINFO _IOR(A, 1, struct sun_audio_info)
|
|
#define SUNAUDIO_SETINFO _IOWR(A, 2, struct sun_audio_info)
|
|
#endif
|
|
|
|
/* ARGSUSED */
|
|
int
|
|
AUDIOIOCTL(dev, cmd, addr, flag, p)
|
|
{
|
|
register struct audio_softc *sc = SOFTC(dev);
|
|
int error = 0, s;
|
|
|
|
switch (cmd) {
|
|
|
|
case AUDIO_GETMAP:
|
|
bcopy((caddr_t)&sc->sc_map, addr, sizeof(sc->sc_map));
|
|
break;
|
|
|
|
case AUDIO_SETMAP:
|
|
bcopy(addr, (caddr_t)&sc->sc_map, sizeof(sc->sc_map));
|
|
sc->sc_map.mr_mmr2 &= 0x7f;
|
|
audio_setmap(sc->sc_au.au_amd, &sc->sc_map);
|
|
break;
|
|
|
|
case AUDIO_FLUSH:
|
|
s = splaudio();
|
|
AUCB_INIT(&sc->sc_au.au_rb);
|
|
AUCB_INIT(&sc->sc_au.au_wb);
|
|
sc->sc_au.au_stamp = 0;
|
|
splx(s);
|
|
sc->sc_wseek = 0;
|
|
sc->sc_rseek = 0;
|
|
break;
|
|
|
|
/*
|
|
* Number of read samples dropped. We don't know where or
|
|
* when they were dropped.
|
|
*/
|
|
case AUDIO_RERROR:
|
|
*(int *)addr = sc->sc_au.au_rb.cb_drops != 0;
|
|
break;
|
|
|
|
/*
|
|
* How many samples will elapse until mike hears the first
|
|
* sample of what we last wrote?
|
|
*/
|
|
case AUDIO_WSEEK:
|
|
s = splaudio();
|
|
*(u_long *)addr = sc->sc_wseek - sc->sc_au.au_stamp
|
|
+ AUCB_LEN(&sc->sc_au.au_rb);
|
|
splx(s);
|
|
break;
|
|
|
|
case AUDIO_SETINFO:
|
|
error = audiosetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETINFO:
|
|
error = audiogetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case SUNAUDIO_GETINFO:
|
|
error = sunaudiogetinfo(sc, (struct sun_audio_info *)addr);
|
|
break;
|
|
|
|
case SUNAUDIO_SETINFO:
|
|
error = sunaudiosetinfo(sc, (struct sun_audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_DRAIN:
|
|
error = audio_drain(sc);
|
|
break;
|
|
|
|
default:
|
|
error = EINVAL;
|
|
break;
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
/* ARGSUSED */
|
|
int
|
|
AUDIOSELECT(dev, rw, p)
|
|
{
|
|
register struct audio_softc *sc = SOFTC(dev);
|
|
register struct aucb *cb;
|
|
register int s = splaudio();
|
|
|
|
switch (rw) {
|
|
|
|
case FREAD:
|
|
cb = &sc->sc_au.au_rb;
|
|
if (AUCB_LEN(cb) >= sc->sc_au.au_blksize) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
selrecord(p, &sc->sc_rsel);
|
|
cb->cb_thresh = sc->sc_au.au_blksize;
|
|
break;
|
|
|
|
case FWRITE:
|
|
cb = &sc->sc_au.au_wb;
|
|
if (AUCB_LEN(cb) <= sc->sc_au.au_lowat) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
selrecord(p, &sc->sc_wsel);
|
|
cb->cb_thresh = sc->sc_au.au_lowat;
|
|
break;
|
|
}
|
|
splx(s);
|
|
return (0);
|
|
}
|
|
|
|
#ifdef AUDIO_C_HANDLER
|
|
int
|
|
audiohwintr(au0)
|
|
void *au0;
|
|
{
|
|
#ifdef SUNOS
|
|
register struct auio *au = audio_au;
|
|
#else
|
|
register struct auio *au = au0;
|
|
#endif
|
|
register volatile struct amd7930 *amd = au->au_amd;
|
|
register struct aucb *cb;
|
|
register int h, t, k;
|
|
|
|
k = amd->ir; /* clear interrupt */
|
|
++au->au_stamp;
|
|
|
|
/* receive incoming data */
|
|
cb = &au->au_rb;
|
|
h = cb->cb_head;
|
|
t = cb->cb_tail;
|
|
k = AUCB_MOD(t + 1);
|
|
if (h == k)
|
|
cb->cb_drops++;
|
|
else if (cb->cb_pause != 0)
|
|
cb->cb_pdrops++;
|
|
else {
|
|
cb->cb_data[t] = amd->bbrb;
|
|
cb->cb_tail = t = k;
|
|
}
|
|
if (AUCB_MOD(t - h) >= cb->cb_thresh) {
|
|
cb->cb_thresh = AUCB_SIZE;
|
|
cb->cb_waking = 1;
|
|
AUDIO_SET_SWINTR;
|
|
}
|
|
/* send outgoing data */
|
|
cb = &au->au_wb;
|
|
h = cb->cb_head;
|
|
t = cb->cb_tail;
|
|
if (h == t)
|
|
cb->cb_drops++;
|
|
else if (cb->cb_pause != 0)
|
|
cb->cb_pdrops++;
|
|
else {
|
|
cb->cb_head = h = AUCB_MOD(h + 1);
|
|
amd->bbtb = cb->cb_data[h];
|
|
}
|
|
if (AUCB_MOD(t - h) <= cb->cb_thresh) {
|
|
cb->cb_thresh = -1;
|
|
cb->cb_waking = 1;
|
|
AUDIO_SET_SWINTR;
|
|
}
|
|
return (1);
|
|
}
|
|
#endif
|
|
|
|
/* ARGSUSED */
|
|
int
|
|
audioswintr(sc0)
|
|
void *sc0;
|
|
{
|
|
register struct audio_softc *sc;
|
|
register int s, ret = 0;
|
|
#ifdef SUNOS
|
|
sc = &audio_softc;
|
|
#else
|
|
sc = sc0;
|
|
#endif
|
|
s = splaudio();
|
|
if (sc->sc_au.au_rb.cb_waking != 0) {
|
|
sc->sc_au.au_rb.cb_waking = 0;
|
|
splx(s);
|
|
ret = 1;
|
|
wakeup((caddr_t)&sc->sc_au.au_rb);
|
|
SELWAKEUP(&sc->sc_rsel);
|
|
}
|
|
if (sc->sc_au.au_wb.cb_waking != 0) {
|
|
sc->sc_au.au_wb.cb_waking = 0;
|
|
splx(s);
|
|
ret = 1;
|
|
wakeup((caddr_t)&sc->sc_au.au_wb);
|
|
SELWAKEUP(&sc->sc_wsel);
|
|
} else
|
|
splx(s);
|
|
return (ret);
|
|
}
|
|
|
|
/* Write 16 bits of data from variable v to the data port of the audio chip */
|
|
|
|
#define WAMD16(amd, v) ((amd)->dr = (v), (amd)->dr = (v) >> 8)
|
|
|
|
void
|
|
audio_setmap(amd, map)
|
|
register volatile struct amd7930 *amd;
|
|
register struct mapreg *map;
|
|
{
|
|
register int i, s, v;
|
|
|
|
s = splaudio();
|
|
amd->cr = AMDR_MAP_1_10;
|
|
for (i = 0; i < 8; i++) {
|
|
v = map->mr_x[i];
|
|
WAMD16(amd, v);
|
|
}
|
|
for (i = 0; i < 8; ++i) {
|
|
v = map->mr_r[i];
|
|
WAMD16(amd, v);
|
|
}
|
|
v = map->mr_gx; WAMD16(amd, v);
|
|
v = map->mr_gr; WAMD16(amd, v);
|
|
v = map->mr_ger; WAMD16(amd, v);
|
|
v = map->mr_stgr; WAMD16(amd, v);
|
|
v = map->mr_ftgr; WAMD16(amd, v);
|
|
v = map->mr_atgr; WAMD16(amd, v);
|
|
amd->dr = map->mr_mmr1;
|
|
amd->dr = map->mr_mmr2;
|
|
splx(s);
|
|
}
|
|
|
|
/*
|
|
* Set the mmr1 register and one other 16 bit register in the audio chip.
|
|
* The other register is indicated by op and val.
|
|
*/
|
|
void
|
|
audio_setmmr1(amd, mmr1, op, val)
|
|
register volatile struct amd7930 *amd;
|
|
register int mmr1;
|
|
register int op;
|
|
register int val;
|
|
{
|
|
register int s = splaudio();
|
|
|
|
amd->cr = AMDR_MAP_MMR1;
|
|
amd->dr = mmr1;
|
|
amd->cr = op;
|
|
WAMD16(amd, val);
|
|
splx(s);
|
|
}
|
|
|
|
/*
|
|
* Set the mmr2 register.
|
|
*/
|
|
static void
|
|
audio_setmmr2(amd, mmr2)
|
|
register volatile struct amd7930 *amd;
|
|
register int mmr2;
|
|
{
|
|
register int s = splaudio();
|
|
|
|
amd->cr = AMDR_MAP_MMR2;
|
|
amd->dr = mmr2;
|
|
splx(s);
|
|
}
|
|
|
|
/*
|
|
* gx, gr & stg gains. this table must contain 256 elements with
|
|
* the 0th being "infinity" (the magic value 9008). The remaining
|
|
* elements match sun's gain curve (but with higher resolution):
|
|
* -18 to 0dB in .16dB steps then 0 to 12dB in .08dB steps.
|
|
*/
|
|
static const u_short gx_coeff[256] = {
|
|
0x9008, 0x8b7c, 0x8b51, 0x8b45, 0x8b42, 0x8b3b, 0x8b36, 0x8b33,
|
|
0x8b32, 0x8b2a, 0x8b2b, 0x8b2c, 0x8b25, 0x8b23, 0x8b22, 0x8b22,
|
|
0x9122, 0x8b1a, 0x8aa3, 0x8aa3, 0x8b1c, 0x8aa6, 0x912d, 0x912b,
|
|
0x8aab, 0x8b12, 0x8aaa, 0x8ab2, 0x9132, 0x8ab4, 0x913c, 0x8abb,
|
|
0x9142, 0x9144, 0x9151, 0x8ad5, 0x8aeb, 0x8a79, 0x8a5a, 0x8a4a,
|
|
0x8b03, 0x91c2, 0x91bb, 0x8a3f, 0x8a33, 0x91b2, 0x9212, 0x9213,
|
|
0x8a2c, 0x921d, 0x8a23, 0x921a, 0x9222, 0x9223, 0x922d, 0x9231,
|
|
0x9234, 0x9242, 0x925b, 0x92dd, 0x92c1, 0x92b3, 0x92ab, 0x92a4,
|
|
0x92a2, 0x932b, 0x9341, 0x93d3, 0x93b2, 0x93a2, 0x943c, 0x94b2,
|
|
0x953a, 0x9653, 0x9782, 0x9e21, 0x9d23, 0x9cd2, 0x9c23, 0x9baa,
|
|
0x9bde, 0x9b33, 0x9b22, 0x9b1d, 0x9ab2, 0xa142, 0xa1e5, 0x9a3b,
|
|
0xa213, 0xa1a2, 0xa231, 0xa2eb, 0xa313, 0xa334, 0xa421, 0xa54b,
|
|
0xada4, 0xac23, 0xab3b, 0xaaab, 0xaa5c, 0xb1a3, 0xb2ca, 0xb3bd,
|
|
0xbe24, 0xbb2b, 0xba33, 0xc32b, 0xcb5a, 0xd2a2, 0xe31d, 0x0808,
|
|
0x72ba, 0x62c2, 0x5c32, 0x52db, 0x513e, 0x4cce, 0x43b2, 0x4243,
|
|
0x41b4, 0x3b12, 0x3bc3, 0x3df2, 0x34bd, 0x3334, 0x32c2, 0x3224,
|
|
0x31aa, 0x2a7b, 0x2aaa, 0x2b23, 0x2bba, 0x2c42, 0x2e23, 0x25bb,
|
|
0x242b, 0x240f, 0x231a, 0x22bb, 0x2241, 0x2223, 0x221f, 0x1a33,
|
|
0x1a4a, 0x1acd, 0x2132, 0x1b1b, 0x1b2c, 0x1b62, 0x1c12, 0x1c32,
|
|
0x1d1b, 0x1e71, 0x16b1, 0x1522, 0x1434, 0x1412, 0x1352, 0x1323,
|
|
0x1315, 0x12bc, 0x127a, 0x1235, 0x1226, 0x11a2, 0x1216, 0x0a2a,
|
|
0x11bc, 0x11d1, 0x1163, 0x0ac2, 0x0ab2, 0x0aab, 0x0b1b, 0x0b23,
|
|
0x0b33, 0x0c0f, 0x0bb3, 0x0c1b, 0x0c3e, 0x0cb1, 0x0d4c, 0x0ec1,
|
|
0x079a, 0x0614, 0x0521, 0x047c, 0x0422, 0x03b1, 0x03e3, 0x0333,
|
|
0x0322, 0x031c, 0x02aa, 0x02ba, 0x02f2, 0x0242, 0x0232, 0x0227,
|
|
0x0222, 0x021b, 0x01ad, 0x0212, 0x01b2, 0x01bb, 0x01cb, 0x01f6,
|
|
0x0152, 0x013a, 0x0133, 0x0131, 0x012c, 0x0123, 0x0122, 0x00a2,
|
|
0x011b, 0x011e, 0x0114, 0x00b1, 0x00aa, 0x00b3, 0x00bd, 0x00ba,
|
|
0x00c5, 0x00d3, 0x00f3, 0x0062, 0x0051, 0x0042, 0x003b, 0x0033,
|
|
0x0032, 0x002a, 0x002c, 0x0025, 0x0023, 0x0022, 0x001a, 0x0021,
|
|
0x001b, 0x001b, 0x001d, 0x0015, 0x0013, 0x0013, 0x0012, 0x0012,
|
|
0x000a, 0x000a, 0x0011, 0x0011, 0x000b, 0x000b, 0x000c, 0x000e,
|
|
};
|
|
|
|
/*
|
|
* second stage play gain.
|
|
*/
|
|
static const u_short ger_coeff[] = {
|
|
0x431f, /* 5. dB */
|
|
0x331f, /* 5.5 dB */
|
|
0x40dd, /* 6. dB */
|
|
0x11dd, /* 6.5 dB */
|
|
0x440f, /* 7. dB */
|
|
0x411f, /* 7.5 dB */
|
|
0x311f, /* 8. dB */
|
|
0x5520, /* 8.5 dB */
|
|
0x10dd, /* 9. dB */
|
|
0x4211, /* 9.5 dB */
|
|
0x410f, /* 10. dB */
|
|
0x111f, /* 10.5 dB */
|
|
0x600b, /* 11. dB */
|
|
0x00dd, /* 11.5 dB */
|
|
0x4210, /* 12. dB */
|
|
0x110f, /* 13. dB */
|
|
0x7200, /* 14. dB */
|
|
0x2110, /* 15. dB */
|
|
0x2200, /* 15.9 dB */
|
|
0x000b, /* 16.9 dB */
|
|
0x000f /* 18. dB */
|
|
#define NGER (sizeof(ger_coeff) / sizeof(ger_coeff[0]))
|
|
};
|
|
|
|
static void
|
|
ausetrgain(sc, level)
|
|
register struct audio_softc *sc;
|
|
register int level;
|
|
{
|
|
level &= 0xff;
|
|
sc->sc_rlevel = level;
|
|
sc->sc_map.mr_mmr1 |= AMD_MMR1_GX;
|
|
sc->sc_map.mr_gx = gx_coeff[level];
|
|
audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1,
|
|
AMDR_MAP_GX, sc->sc_map.mr_gx);
|
|
}
|
|
|
|
static void
|
|
ausetpgain(sc, level)
|
|
register struct audio_softc *sc;
|
|
register int level;
|
|
{
|
|
register int gi, s;
|
|
register volatile struct amd7930 *amd;
|
|
|
|
level &= 0xff;
|
|
sc->sc_plevel = level;
|
|
sc->sc_map.mr_mmr1 |= AMD_MMR1_GER|AMD_MMR1_GR;
|
|
level *= 256 + NGER;
|
|
level >>= 8;
|
|
if (level >= 256) {
|
|
gi = level - 256;
|
|
level = 255;
|
|
} else
|
|
gi = 0;
|
|
sc->sc_map.mr_ger = ger_coeff[gi];
|
|
sc->sc_map.mr_gr = gx_coeff[level];
|
|
|
|
amd = sc->sc_au.au_amd;
|
|
s = splaudio();
|
|
amd->cr = AMDR_MAP_MMR1;
|
|
amd->dr = sc->sc_map.mr_mmr1;
|
|
amd->cr = AMDR_MAP_GR;
|
|
gi = sc->sc_map.mr_gr;
|
|
WAMD16(amd, gi);
|
|
amd->cr = AMDR_MAP_GER;
|
|
gi = sc->sc_map.mr_ger;
|
|
WAMD16(amd, gi);
|
|
splx(s);
|
|
}
|
|
|
|
static void
|
|
ausetmgain(sc, level)
|
|
register struct audio_softc *sc;
|
|
register int level;
|
|
{
|
|
level &= 0xff;
|
|
sc->sc_mlevel = level;
|
|
sc->sc_map.mr_mmr1 |= AMD_MMR1_STG;
|
|
sc->sc_map.mr_stgr = gx_coeff[level];
|
|
audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1,
|
|
AMDR_MAP_STG, sc->sc_map.mr_stgr);
|
|
}
|
|
|
|
static int
|
|
audiosetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
register int s, bsize;
|
|
|
|
if (p->gain != ~0)
|
|
ausetpgain(sc, p->gain);
|
|
if (r->gain != ~0)
|
|
ausetrgain(sc, r->gain);
|
|
if (ai->monitor_gain != ~0)
|
|
ausetmgain(sc, ai->monitor_gain);
|
|
if (p->port == AUDIO_SPEAKER) {
|
|
sc->sc_map.mr_mmr2 |= AMD_MMR2_LS;
|
|
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
|
|
} else if (p->port == AUDIO_HEADPHONE) {
|
|
sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS;
|
|
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
|
|
}
|
|
if (p->pause != (u_char)~0)
|
|
sc->sc_au.au_wb.cb_pause = p->pause;
|
|
if (r->pause != (u_char)~0)
|
|
sc->sc_au.au_rb.cb_pause = r->pause;
|
|
|
|
if (ai->blocksize != ~0) {
|
|
if (ai->blocksize == 0)
|
|
bsize = ai->blocksize = DEFBLKSIZE;
|
|
else if (ai->blocksize > MAXBLKSIZE)
|
|
bsize = ai->blocksize = MAXBLKSIZE;
|
|
else
|
|
bsize = ai->blocksize;
|
|
|
|
s = splaudio();
|
|
sc->sc_au.au_blksize = bsize;
|
|
/* AUDIO_FLUSH */
|
|
AUCB_INIT(&sc->sc_au.au_rb);
|
|
AUCB_INIT(&sc->sc_au.au_wb);
|
|
splx(s);
|
|
|
|
}
|
|
if (ai->hiwat != ~0 && (unsigned)ai->hiwat < AUCB_SIZE)
|
|
sc->sc_au.au_hiwat = ai->hiwat;
|
|
if (ai->lowat != ~0 && ai->lowat < AUCB_SIZE)
|
|
sc->sc_au.au_lowat = ai->lowat;
|
|
if (ai->backlog != ~0 && ai->backlog < (AUCB_SIZE/2))
|
|
sc->sc_au.au_backlog = ai->backlog;
|
|
|
|
return (0);
|
|
}
|
|
|
|
static int
|
|
sunaudiosetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct sun_audio_info *ai;
|
|
{
|
|
struct sun_audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
|
|
if (p->gain != ~0)
|
|
ausetpgain(sc, p->gain);
|
|
if (r->gain != ~0)
|
|
ausetrgain(sc, r->gain);
|
|
if (ai->monitor_gain != ~0)
|
|
ausetmgain(sc, ai->monitor_gain);
|
|
if (p->port == AUDIO_SPEAKER) {
|
|
sc->sc_map.mr_mmr2 |= AMD_MMR2_LS;
|
|
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
|
|
} else if (p->port == AUDIO_HEADPHONE) {
|
|
sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS;
|
|
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
|
|
}
|
|
/*
|
|
* The bsd driver does not distinguish between paused and active.
|
|
* (In the sun driver, not active means samples are not ouput
|
|
* at all, but paused means the last streams buffer is drained
|
|
* and then output stops.) If either are 0, then when stop output.
|
|
* Otherwise, if either are non-zero, we resume.
|
|
*/
|
|
if (p->pause == 0 || p->active == 0)
|
|
sc->sc_au.au_wb.cb_pause = 0;
|
|
else if (p->pause != (u_char)~0 || p->active != (u_char)~0)
|
|
sc->sc_au.au_wb.cb_pause = 1;
|
|
if (r->pause == 0 || r->active == 0)
|
|
sc->sc_au.au_rb.cb_pause = 0;
|
|
else if (r->pause != (u_char)~0 || r->active != (u_char)~0)
|
|
sc->sc_au.au_rb.cb_pause = 1;
|
|
|
|
return (0);
|
|
}
|
|
|
|
static int
|
|
audiogetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
|
|
p->sample_rate = r->sample_rate = 8000;
|
|
p->channels = r->channels = 1;
|
|
p->precision = r->precision = 8;
|
|
p->encoding = r->encoding = AUDIO_ENCODING_ULAW;
|
|
|
|
ai->monitor_gain = sc->sc_mlevel;
|
|
r->gain = sc->sc_rlevel;
|
|
p->gain = sc->sc_plevel;
|
|
r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ?
|
|
AUDIO_SPEAKER : AUDIO_HEADPHONE;
|
|
|
|
p->pause = sc->sc_au.au_wb.cb_pause;
|
|
r->pause = sc->sc_au.au_rb.cb_pause;
|
|
p->error = sc->sc_au.au_wb.cb_drops != 0;
|
|
r->error = sc->sc_au.au_rb.cb_drops != 0;
|
|
|
|
p->open = sc->sc_open;
|
|
r->open = sc->sc_open;
|
|
|
|
p->samples = sc->sc_au.au_stamp - sc->sc_au.au_wb.cb_pdrops;
|
|
r->samples = sc->sc_au.au_stamp - sc->sc_au.au_rb.cb_pdrops;
|
|
|
|
p->seek = sc->sc_wseek;
|
|
r->seek = sc->sc_rseek;
|
|
|
|
ai->blocksize = sc->sc_au.au_blksize;
|
|
ai->hiwat = sc->sc_au.au_hiwat;
|
|
ai->lowat = sc->sc_au.au_lowat;
|
|
ai->backlog = sc->sc_au.au_backlog;
|
|
|
|
return (0);
|
|
}
|
|
|
|
static int
|
|
sunaudiogetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct sun_audio_info *ai;
|
|
{
|
|
struct sun_audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
|
|
p->sample_rate = r->sample_rate = 8000;
|
|
p->channels = r->channels = 1;
|
|
p->precision = r->precision = 8;
|
|
p->encoding = r->encoding = AUDIO_ENCODING_ULAW;
|
|
|
|
ai->monitor_gain = sc->sc_mlevel;
|
|
r->gain = sc->sc_rlevel;
|
|
p->gain = sc->sc_plevel;
|
|
r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ?
|
|
AUDIO_SPEAKER : AUDIO_HEADPHONE;
|
|
|
|
p->active = p->pause = sc->sc_au.au_wb.cb_pause;
|
|
r->active = r->pause = sc->sc_au.au_rb.cb_pause;
|
|
p->error = sc->sc_au.au_wb.cb_drops != 0;
|
|
r->error = sc->sc_au.au_rb.cb_drops != 0;
|
|
|
|
p->waiting = 0;
|
|
r->waiting = 0;
|
|
p->eof = 0;
|
|
r->eof = 0;
|
|
|
|
p->open = sc->sc_open;
|
|
r->open = sc->sc_open;
|
|
|
|
p->samples = sc->sc_au.au_stamp - sc->sc_au.au_wb.cb_pdrops;
|
|
r->samples = sc->sc_au.au_stamp - sc->sc_au.au_rb.cb_pdrops;
|
|
|
|
return (0);
|
|
}
|
|
#endif
|