1424 lines
31 KiB
C
1424 lines
31 KiB
C
/* $NetBSD: bsd_audio.c,v 1.1.1.1 1996/05/05 12:17:03 oki Exp $ */
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/*
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* Copyright (c) 1991-1993 Regents of the University of California.
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* All rights reserved.
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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 Computer Systems
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* Engineering Group at Lawrence Berkeley Laboratory.
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* 4. Neither the name of the University nor of the Laboratory may be used
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* to endorse or promote products derived from this software without
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* 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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/*
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* This is a (partially) SunOS-compatible /dev/audio driver for NetBSD.
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*
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* This code assumes SoundBlaster type hardware, supported by the
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* code in isa/sb.c. A major problem with this hardware is that it
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* is half-duplex. E.g., you cannot simultaneously record and play
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* samples. Thus, it doesn't really make sense to allow O_RDWR access.
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* However, opening and closing the device to "turn around the line"
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* is relatively expensive and costs a card reset (which can take
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* some time). Instead, we allow O_RDWR access, and provide an
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* ioctl to set the "mode", e.g., playing or recording. If you
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* write to the device in record mode, the data is tossed. If you
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* read from the device in play mode, you get zero filled buffers
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* at the rate at which samples are naturally generated.
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*/
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#include "audio.h"
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#if NAUDIO > 0
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#include <sys/param.h>
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#include <sys/ioctl.h>
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#include <sys/systm.h>
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#include <sys/vnode.h>
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#include <sys/select.h>
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#include <sys/malloc.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/device.h>
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#include <machine/cpu.h>
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#include <machine/pte.h>
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extern u_int kvtop __P((register caddr_t addr));
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int uiomove __P((caddr_t cp, int n, struct uio *uio));
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#include <x68k/dev/bsd_audiovar.h>
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#include <x68k/dev/bsd_audioreg.h>
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#include <machine/bsd_audioio.h>
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#include <x68k/x68k/iodevice.h>
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#define AUDIODEBUG if (audiodebug) printf
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int audiodebug = 0;
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#define dma3 (IODEVbase->io_dma[3])
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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 MSM6258V audio chip.
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*/
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struct audio_softc {
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struct adpcm_softc sc_adpcm;
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u_char sc_open; /* single use device */
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u_char sc_mode; /* */
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u_char sc_rbus; /* input dma in progress */
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u_char sc_pbus; /* output dma in progress */
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u_char sc_rencoding;
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u_char sc_pencoding;
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u_char sc_pad[2];
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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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u_long sc_orate; /* input sampling rate */
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u_long sc_irate; /* output sampling rate */
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struct selinfo sc_wsel; /* write selector */
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struct selinfo sc_rsel; /* read selector */
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int sc_rlevel; /* record level */
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int sc_plevel; /* play level */
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/*
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* Sleep channels for reading and writing.
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*/
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int sc_rchan;
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int sc_wchan;
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int sc_ochan;
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/*
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* Buffer management.
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*/
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u_char *sc_hp; /* head */
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u_char *sc_tp; /* tail */
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u_char *sc_bp; /* start of buffer */
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u_char *sc_ep; /* end of buffer */
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u_char *sc_zp; /* block of silence */
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int sc_nblk;
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int sc_maxblk;
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int sc_lowat; /* xmit low water mark (for wakeup) */
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int sc_hiwat; /* xmit high water mark (for wakeup) */
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int sc_blksize; /* recv block (chunk) size */
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int sc_backlog; /* # blks of xmit backlog to gen. */
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int sc_finish;
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/* sc_au is special in that the hardware interrupt handler uses it */
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int sc_rblks; /* number of phantom record blocks */
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} audio_softc[NAUDIO];
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/* forward declarations */
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inline static int audio_sleep __P((int *));
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/* autoconfiguration driver */
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int audioattach();
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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 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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void audio_init_record __P((struct audio_softc *));
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void audio_init_play __P((struct audio_softc *));
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void audiostartr __P((struct audio_softc *));
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void audiostartp __P((struct audio_softc *));
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inline void audio_rint __P((struct audio_softc *));
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inline void audio_pint __P((struct audio_softc *));
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void audio_tomulaw __P((short *, register int));
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void audio_frommulaw __P((u_char *, register int));
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void audio_tolinear __P((struct audio_softc *, register u_char *, register int));
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void audio_fromlinear __P((struct audio_softc *, register u_char *, register int));
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void
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audio_initbuf(sc)
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struct audio_softc *sc;
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{
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register int nblk = NBPG / sc->sc_blksize;
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sc->sc_ep = sc->sc_bp + nblk * sc->sc_blksize;
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sc->sc_hp = sc->sc_tp = sc->sc_bp;
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sc->sc_maxblk = nblk;
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sc->sc_nblk = 0;
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sc->sc_lowat = 3;
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sc->sc_hiwat = nblk - 1;
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}
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static inline int
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audio_sleep(int *chan)
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{
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int st;
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*chan = 1;
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st = (tsleep((caddr_t)chan, PWAIT | PCATCH, "audio", 0));
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*chan = 0;
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return (st);
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}
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static inline void
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audio_wakeup(chan)
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int *chan;
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{
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if (*chan) {
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wakeup((caddr_t)chan);
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*chan = 0;
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}
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}
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/*XXX*/
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int auzero[NBPG/4];
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short transbuf[NBPG*2];
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u_char transbuf2[NBPG];
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int audiomatch __P((struct device *, void *, void *));
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void audioattach __P((struct device *, struct device *, void *));
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struct cfattach audio_ca = {
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sizeof(struct audio_softc), audiomatch, audioattach
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};
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struct cfdriver audio_cd = {
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NULL, "audio", DV_DULL
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};
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int
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audiomatch(parent, match, aux)
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struct device *parent;
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void *match, *aux;
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{
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struct cfdata *cf = match;
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if (strcmp(aux, "adpcm") || cf->cf_unit > 0)
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return 0;
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return 1;
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}
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void
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audioattach(parent, self, aux)
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struct device *parent;
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struct device *self;
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void *aux;
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{
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struct dmac *dmac = &IODEVbase->io_dma[3];
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register struct audio_softc *sc = &audio_softc[0];
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dmac->csr = 0xff;
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dmac->dcr = 0x80;
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dmac->ocr = 0x32;
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dmac->scr = 0x04;
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dmac->mfc = 0x05;
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dmac->dfc = 0x05;
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dmac->bfc = 0x05;
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dmac->dar = (unsigned long)kvtop(&(adpcm.data));
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dmac->niv = 0x6a;
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dmac->eiv = 0x6b;
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sc->sc_rbus = 0;
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sc->sc_pbus = 0;
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/*printf("audio0: MSM6258V ADPCM chip.\n");*/
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}
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int
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audioopen(dev, flags, ifmt, p)
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dev_t dev;
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int flags;
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int ifmt;
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struct proc *p;
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{
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register struct audio_softc *sc = &audio_softc[0];
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int s, i;
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int unit = minor(dev);
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int error;
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AUDIODEBUG("audio: open\n");
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if((unit & 0x0000003f) >= NAUDIO)
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return (ENXIO);
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if (sc->sc_open != 0)
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return (EBUSY);
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/*
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* Allocate a single page so it won't cross a page boundary.
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* This way the dma carried out in the sb module will be efficient
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* (i.e., at_dma() won't have to make a copy)
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*/
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sc->sc_zp = malloc(NBPG, M_DEVBUF, M_WAITOK);
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if (sc->sc_zp == 0)
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goto nobufs;
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sc->sc_bp = malloc(NBPG, M_DEVBUF, M_WAITOK);
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if (sc->sc_bp == 0) {
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free(sc->sc_zp, M_DEVBUF);
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goto nobufs;
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}
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sc->sc_blksize = audio_blocksize;
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sc->sc_backlog = audio_backlog;
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audio_initbuf(sc);
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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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sc->sc_rchan = 0;
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sc->sc_wchan = 0;
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sc->sc_adpcm.sc_amp = 0;
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sc->sc_adpcm.sc_estim = 0;
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/*
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* Here's a hack: do ulaw conversion if 6-7 bit of
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* minor device is set. That way, we can have /dev/audio
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* (minor 0x80) do ulaw conversion, and /dev/sound or
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* whatever, do adpcm.
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*/
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switch (minor(dev) >> 6) {
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case 0x00:
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/* /dev/adpcm */
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AUDIODEBUG("audioopen: adpcm mode\n");
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sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_ADPCM;
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sc->sc_orate = 15625;
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sc->sc_irate = 15625;
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for (i = NBPG / 4; --i >= 0; ) {
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((u_long *)sc->sc_zp)[i] = 0x88008800;
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auzero[i] = 0x88008800;
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}
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break;
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case 0x01:
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/* /dev/sound */
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AUDIODEBUG("audioopen: linear mode\n");
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sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_LINEAR;
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sc->sc_orate = 15625;
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sc->sc_irate = 15625;
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for (i = NBPG / 4; --i >= 0; ) {
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((u_long *)sc->sc_zp)[i] = 0x88008800;
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auzero[i] = 0x88008800;
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}
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break;
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case 0x02:
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/* /dev/audio */
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AUDIODEBUG("audioopen: ulaw mode\n");
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sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_ULAW;
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sc->sc_orate = 7813;
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sc->sc_irate = 7813;
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for (i = NBPG / 4; --i >= 0; ) {
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((u_long *)sc->sc_zp)[i] = 0x7f7f7f7f;
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auzero[i] = 0x80808080;
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}
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break;
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case 0x03:
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/* /dev/audio */
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AUDIODEBUG("audioopen: ulaw mode\n");
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sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_ULAW;
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sc->sc_orate = 7813;
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sc->sc_irate = 7813;
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for (i = NBPG / 4; --i >= 0; ) {
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((u_long *)sc->sc_zp)[i] = 0x7f7f7f7f;
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auzero[i] = 0x80808080;
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}
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break;
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}
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#if 0
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ausetrgain(sc, audio_default_level);
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ausetpgain(sc, audio_default_level);
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#endif
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/* XXX */
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s = splaudio();
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if ((sc->sc_pbus == 1) || (sc->sc_rbus == 1))
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error = audio_sleep(&sc->sc_ochan);
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sc->sc_open = 1;
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if ((flags & FREAD) != 0) {
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audio_init_record(sc);
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} else {
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audio_init_play(sc);
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}
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splx(s);
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return (0);
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nobufs:
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sc->sc_open = 0;
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return (ENOBUFS);
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}
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/*
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* Must be called from task context.
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*/
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void
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audio_init_record(sc)
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struct audio_softc *sc;
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{
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register int s;
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s = splaudio();
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dma3.csr = 0xff;
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dma3.ocr = 0xb2;
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sc->sc_mode = AUMODE_RECORD;
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sc->sc_rblks = 0;
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adpcm.stat = ADPCM_CMD_STOP;
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adpcm_set_sr(sc->sc_irate);
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splx(s);
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}
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/*
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* Must be called from task context.
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*/
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void
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audio_init_play(sc)
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struct audio_softc *sc;
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{
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register int s;
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s = splaudio();
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dma3.csr = 0xff;
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dma3.ocr = 0x32;
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sc->sc_mode = AUMODE_PLAY;
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sc->sc_rblks = 0;
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adpcm.stat = ADPCM_CMD_STOP;
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adpcm_set_sr(sc->sc_orate);
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splx(s);
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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 (sc->sc_nblk > 0) {
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error = audio_sleep(&sc->sc_wchan);
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if (error != 0) {
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AUDIODEBUG("audio: Interrupted?\n");
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return (error);
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}
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}
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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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dev_t dev;
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int flags;
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int ifmt;
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struct proc *p;
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{
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register struct audio_softc *sc =&audio_softc[0];
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register int s;
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AUDIODEBUG("audio: close\n");
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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_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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if (sc->sc_pbus && sc->sc_nblk > 0)
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(void)audio_drain(sc);
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splx(s);
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free(sc->sc_bp, M_DEVBUF);
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free(sc->sc_zp, M_DEVBUF);
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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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audioread(dev, uio, ioflag)
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dev_t dev;
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struct uio *uio;
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int ioflag;
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{
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register struct audio_softc *sc = &audio_softc[0];
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register u_char *hp;
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register int blocksize = sc->sc_blksize;
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register int error, error2, s;
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AUDIODEBUG("audio: read\n");
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if (uio->uio_resid == 0)
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return (0);
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if (uio->uio_resid < blocksize)
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return (EINVAL);
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if (sc->sc_mode == AUMODE_PLAY) {
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/*
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* If we're in play mode, return silence blocks
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* based on the number of blocks we have output.
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*/
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do {
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s = splaudio();
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while (sc->sc_rblks <= 0) {
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if (ioflag & IO_NDELAY) {
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splx(s);
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return (EWOULDBLOCK);
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}
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error = audio_sleep(&sc->sc_rchan);
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if (error != 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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splx(s);
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/*XXX handle ulaw 0 */
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error = uiomove(sc->sc_zp, blocksize, uio);
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if (error)
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break;
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--sc->sc_rblks;
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} while (uio->uio_resid >= blocksize);
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return (error);
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}
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error = error2 = 0;
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do {
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while ((sc->sc_nblk <= 0) && (error2 == 0)){
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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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s = splaudio();
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if (!sc->sc_rbus)
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audiostartr(sc);
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error2 = error = audio_sleep(&sc->sc_rchan);
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splx(s);
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if (error != 0) {
|
|
AUDIODEBUG("audio: read Interrupted?\n");
|
|
if (sc->sc_rbus == 1){
|
|
s = splaudio();
|
|
dma3.csr = 0xff;
|
|
dma3.ccr = 0x08;
|
|
sc->sc_finish = 1;
|
|
splx(s);
|
|
audio_sleep(&sc->sc_rchan);
|
|
} else {
|
|
return (error);
|
|
}
|
|
}
|
|
}
|
|
hp = sc->sc_hp;
|
|
switch (sc->sc_rencoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
audio_tolinear(sc, hp, blocksize);
|
|
audio_tomulaw(transbuf, blocksize*2);
|
|
error = uiomove((u_char *)transbuf2, blocksize*2, uio);
|
|
break;
|
|
case AUDIO_ENCODING_LINEAR:
|
|
audio_tolinear(sc, hp, blocksize);
|
|
error = uiomove((u_char *)transbuf, blocksize*4, uio);
|
|
break;
|
|
case AUDIO_ENCODING_ADPCM:
|
|
error = uiomove(hp, blocksize, uio);
|
|
break;
|
|
}
|
|
if (error) {
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
hp += blocksize;
|
|
if (hp >= sc->sc_ep)
|
|
hp = sc->sc_bp;
|
|
sc->sc_hp = hp;
|
|
--sc->sc_nblk;
|
|
} while (uio->uio_resid >= blocksize);
|
|
|
|
return (error ? error : error2);
|
|
}
|
|
|
|
void
|
|
audio_clear(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
register int s = splaudio();
|
|
|
|
if (sc->sc_rbus || sc->sc_pbus) {
|
|
dma3.ccr = 0x10;
|
|
sc->sc_rbus = 0;
|
|
sc->sc_pbus = 0;
|
|
}
|
|
sc->sc_nblk = 0;
|
|
sc->sc_hp = sc->sc_tp = sc->sc_bp;
|
|
splx(s);
|
|
}
|
|
|
|
int
|
|
audiowrite(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
register struct audio_softc *sc;
|
|
register u_char *tp;
|
|
register int error, s, cc;
|
|
register int blocksize;
|
|
|
|
AUDIODEBUG("audio: write\n");
|
|
sc = &audio_softc[0];
|
|
blocksize = sc->sc_blksize;
|
|
/*
|
|
* If currently recording, throw away data.
|
|
*/
|
|
if (sc->sc_mode != AUMODE_PLAY) {
|
|
uio->uio_offset += uio->uio_resid;
|
|
uio->uio_resid = 0;
|
|
return (0);
|
|
}
|
|
error = 0;
|
|
|
|
while (uio->uio_resid > 0) {
|
|
register int watermark = sc->sc_hiwat;
|
|
s = splaudio();
|
|
while (sc->sc_nblk > watermark) {
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
error = EWOULDBLOCK;
|
|
printf("audiowrite: ioflag=%x\n", ioflag);
|
|
return (error);
|
|
}
|
|
error = audio_sleep(&sc->sc_wchan);
|
|
if (error != 0) {
|
|
splx(s);
|
|
AUDIODEBUG("audiowrite: Interrupted?\n");
|
|
return (error);
|
|
}
|
|
watermark = sc->sc_lowat;
|
|
}
|
|
splx(s);
|
|
tp = sc->sc_tp;
|
|
cc = uio->uio_resid;
|
|
switch (sc->sc_pencoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
if (cc < blocksize*2) {
|
|
error = uiomove((u_char *)transbuf2, cc, uio);
|
|
if (error){
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
AUDIODEBUG("audiowrite: zero suppress(%x,%x,%d)\n",transbuf, cc, blocksize*4 - cc);
|
|
bcopy((char *)auzero, (u_char *)transbuf2 + cc, blocksize*2 - cc);
|
|
} else {
|
|
error = uiomove((u_char *)transbuf2, blocksize*2, uio);
|
|
if (error) {
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
case AUDIO_ENCODING_ADPCM:
|
|
if (cc < blocksize) {
|
|
error = uiomove(tp, cc, uio);
|
|
if (error){
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
bcopy((char *)auzero, tp + cc, blocksize - cc);
|
|
} else {
|
|
error = uiomove(tp, blocksize, uio);
|
|
if (error) {
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
case AUDIO_ENCODING_LINEAR:
|
|
if (cc < blocksize*4) {
|
|
error = uiomove((u_char *)transbuf, cc, uio);
|
|
if (error){
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
AUDIODEBUG("audiowrite: zero suppress(%x,%x,%d)\n",transbuf, cc, blocksize*4 - cc);
|
|
bzero((char *)transbuf + cc, blocksize*4 - cc);
|
|
} else {
|
|
error = uiomove((u_char *)transbuf, blocksize*4, uio);
|
|
if (error) {
|
|
printf("audio: uiomove failed\n");
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
switch (sc->sc_pencoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
audio_frommulaw(transbuf2, blocksize*2);
|
|
audio_fromlinear(sc, tp, blocksize*2);
|
|
break;
|
|
case AUDIO_ENCODING_LINEAR:
|
|
audio_fromlinear(sc, tp, blocksize*2);
|
|
break;
|
|
case AUDIO_ENCODING_ADPCM:
|
|
break;
|
|
}
|
|
tp += blocksize;
|
|
if (tp >= sc->sc_ep)
|
|
tp = sc->sc_bp;
|
|
sc->sc_tp = tp;
|
|
++sc->sc_nblk;
|
|
sc->sc_finish = 0;
|
|
/*
|
|
* If output isn't active, start it up.
|
|
*/
|
|
s = splaudio();
|
|
if (sc->sc_pbus == 0)
|
|
audiostartp(sc);
|
|
splx(s);
|
|
}
|
|
AUDIODEBUG("audiowrite: exit\n");
|
|
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 balance;
|
|
u_short minordev;
|
|
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];
|
|
};
|
|
|
|
int
|
|
audioioctl(dev, cmd, addr, flag, p)
|
|
dev_t dev;
|
|
u_long cmd;
|
|
caddr_t addr;
|
|
int flag;
|
|
struct proc *p;
|
|
{
|
|
register struct audio_softc *sc = &audio_softc[0];
|
|
int error = 0, s;
|
|
|
|
AUDIODEBUG("audio: ioctl(0x%x)\n", cmd);
|
|
switch (cmd) {
|
|
|
|
#if 0
|
|
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_msm, &sc->sc_map);
|
|
break;
|
|
|
|
case AUDIO_FLUSH:
|
|
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;
|
|
#endif
|
|
case AUDIO_SETINFO:
|
|
error = audiosetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETINFO:
|
|
error = audiogetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_DRAIN:
|
|
error = audio_drain(sc);
|
|
break;
|
|
|
|
default:
|
|
error = EINVAL;
|
|
break;
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
/* ARGSUSED */
|
|
int
|
|
audioselect(dev, rw, p)
|
|
dev_t dev;
|
|
int rw;
|
|
struct proc *p;
|
|
{
|
|
register struct audio_softc *sc = &audio_softc[0];
|
|
register int s = splaudio();
|
|
|
|
switch (rw) {
|
|
|
|
case FREAD:
|
|
if (sc->sc_mode == AUMODE_PLAY) {
|
|
if (sc->sc_rblks > 0) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
} else if (sc->sc_nblk > 0) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
selrecord(p, &sc->sc_rsel);
|
|
break;
|
|
|
|
case FWRITE:
|
|
/*
|
|
* Can write if we're recording because it gets preempted.
|
|
* Otherwise, can write when below low water.
|
|
* XXX this won't work right if we're in
|
|
* record mode -- we need to note that a write
|
|
* select has happed and flip the speaker.
|
|
*/
|
|
if (sc->sc_mode != AUMODE_PLAY ||
|
|
sc->sc_nblk < sc->sc_lowat) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
selrecord(p, &sc->sc_wsel);
|
|
break;
|
|
}
|
|
splx(s);
|
|
return (0);
|
|
}
|
|
|
|
static inline void
|
|
audio_dmastart(read, addr, count)
|
|
int read;
|
|
u_char *addr;
|
|
int count;
|
|
{
|
|
dma3.csr = 0xff;
|
|
dma3.mtc = (u_short)count;
|
|
asm("nop");
|
|
asm("nop");
|
|
dma3.mar = (u_long)kvtop(addr);
|
|
#if defined(M68040)
|
|
/*
|
|
* Push back dirty cache lines
|
|
*/
|
|
if (mmutype == MMU_68040)
|
|
DCFP(kvtop(addr));
|
|
#endif
|
|
adpcm.stat = read ? ADPCM_CMD_PLAY : ADPCM_CMD_REC;
|
|
dma3.ccr = 0x88;
|
|
}
|
|
|
|
void
|
|
audiostartr(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
register u_char *tp = sc->sc_tp;
|
|
register int cc = sc->sc_blksize;
|
|
|
|
AUDIODEBUG("audio: startr\n");
|
|
audio_dmastart(0, tp, cc);
|
|
sc->sc_rbus = 1;
|
|
}
|
|
|
|
void
|
|
audiostartp(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
audio_dmastart(1, sc->sc_hp, sc->sc_blksize);
|
|
sc->sc_pbus = 1;
|
|
}
|
|
|
|
void
|
|
audiointr()
|
|
{
|
|
register struct audio_softc *sc = &audio_softc[0];
|
|
|
|
dma3.csr = 0xff;
|
|
PCIA(); /* XXX? by oki */
|
|
if (sc->sc_pbus == 1){
|
|
audio_pint(sc);
|
|
} else if (sc->sc_rbus == 1){
|
|
audio_rint(sc);
|
|
} else {
|
|
printf("audiointr: sc_pbus == sc_rbus == 0. Why interrupt?\n");
|
|
}
|
|
if (sc->sc_open == 0) {
|
|
audio_wakeup(&sc->sc_ochan);
|
|
}
|
|
}
|
|
|
|
void
|
|
audioerrintr()
|
|
{
|
|
register struct audio_softc *sc = &audio_softc[0];
|
|
|
|
printf("audioerrintr: software abort?\ncsr=%x, cer=%x\n pbus = %d, rbus = %d\n", dma3.csr, dma3.cer, sc->sc_pbus, sc->sc_rbus);
|
|
|
|
dma3.csr = 0xff;
|
|
if (sc->sc_pbus == 1){
|
|
audio_pint(sc);
|
|
} else if (sc->sc_rbus == 1){
|
|
audio_rint(sc);
|
|
} else {
|
|
printf("audioerrintr: sc_pbus == sc_rbus == 0. Why interrupt?\n");
|
|
}
|
|
if (sc->sc_open == 0) {
|
|
audio_wakeup(&sc->sc_ochan);
|
|
}
|
|
}
|
|
|
|
inline void
|
|
audio_pint(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
register u_char *hp = sc->sc_hp;
|
|
register int cc = sc->sc_blksize;
|
|
register int s;
|
|
|
|
AUDIODEBUG("audio: pint sc_nblk %d\n", sc->sc_nblk);
|
|
s = splaudio();
|
|
if (sc->sc_finish == 1) {
|
|
adpcm.stat = ADPCM_CMD_STOP;
|
|
sc->sc_pbus = 0;
|
|
} else {
|
|
--sc->sc_nblk;
|
|
hp = sc->sc_hp;
|
|
hp += cc;
|
|
if (hp >= sc->sc_ep)
|
|
hp = sc->sc_bp;
|
|
sc->sc_hp = hp;
|
|
if (sc->sc_nblk > 0) {
|
|
audio_dmastart(1, hp, cc);
|
|
sc->sc_finish = 0;
|
|
} else {
|
|
audio_dmastart(1, (char *)&auzero, cc);
|
|
sc->sc_finish = 1;
|
|
}
|
|
}
|
|
splx(s);
|
|
++sc->sc_rblks;
|
|
|
|
if (sc->sc_mode == AUMODE_PLAY) {
|
|
if (sc->sc_nblk <= sc->sc_lowat) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
selwakeup(&sc->sc_wsel);
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from sb module on completion of dma input.
|
|
* Copy the input frame into the ring buffer at the
|
|
* current position. Do a wakeup if necessary.
|
|
*/
|
|
void
|
|
audio_rint(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
register u_char *tp = sc->sc_tp;
|
|
register int cc = sc->sc_blksize;
|
|
|
|
tp = sc->sc_tp;
|
|
tp += cc;
|
|
if (tp >= sc->sc_ep)
|
|
tp = sc->sc_bp;
|
|
if (++sc->sc_nblk < sc->sc_maxblk)
|
|
audio_dmastart(0, tp, cc);
|
|
else {
|
|
adpcm.stat = ADPCM_CMD_STOP;
|
|
sc->sc_rbus = 0;
|
|
}
|
|
sc->sc_tp = tp;
|
|
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selwakeup(&sc->sc_rsel);
|
|
}
|
|
|
|
static void
|
|
ausetrgain(sc, level)
|
|
register struct audio_softc *sc;
|
|
register int level;
|
|
{
|
|
#ifdef x68k
|
|
/* XXX */
|
|
#endif
|
|
}
|
|
|
|
/*
|
|
* XXX Looks like we need a pro to do volume control...
|
|
*/
|
|
static void
|
|
ausetpgain(sc, level)
|
|
register struct audio_softc *sc;
|
|
register int level;
|
|
{
|
|
#ifdef x68k
|
|
/* XXX */
|
|
#endif
|
|
}
|
|
|
|
static int
|
|
audiosetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
register int cleared = 0;
|
|
register int s, bsize;
|
|
|
|
if (p->gain != ~0)
|
|
ausetpgain(sc, p->gain);
|
|
if (r->gain != ~0)
|
|
ausetrgain(sc, r->gain);
|
|
|
|
if (p->sample_rate != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
p->sample_rate = adpcm_round_sr(p->sample_rate);
|
|
printf("audiosetinfo: rate=%d\n", p->sample_rate);
|
|
sc->sc_orate = p->sample_rate;
|
|
if (sc->sc_mode == AUMODE_PLAY)
|
|
(void)adpcm_set_sr(sc->sc_orate);
|
|
}
|
|
if (r->sample_rate != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
r->sample_rate = adpcm_round_sr(r->sample_rate);
|
|
sc->sc_irate = r->sample_rate;
|
|
if (sc->sc_mode != AUMODE_PLAY)
|
|
(void)adpcm_set_sr(sc->sc_irate);
|
|
}
|
|
if (p->encoding != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
switch (p->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
sc->sc_pencoding = AUDIO_ENCODING_ULAW;
|
|
break;
|
|
case AUDIO_ENCODING_LINEAR:
|
|
sc->sc_pencoding = AUDIO_ENCODING_LINEAR;
|
|
break;
|
|
default:
|
|
sc->sc_pencoding = AUDIO_ENCODING_ADPCM;
|
|
p->encoding = AUDIO_ENCODING_ADPCM;
|
|
}
|
|
}
|
|
if (r->encoding != ~0) {
|
|
switch (r->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
sc->sc_rencoding = AUDIO_ENCODING_ULAW;
|
|
break;
|
|
case AUDIO_ENCODING_LINEAR:
|
|
sc->sc_rencoding = AUDIO_ENCODING_LINEAR;
|
|
break;
|
|
default:
|
|
sc->sc_rencoding = AUDIO_ENCODING_ADPCM;
|
|
r->encoding = AUDIO_ENCODING_ADPCM;
|
|
}
|
|
}
|
|
#ifdef notdef
|
|
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;
|
|
#endif
|
|
if (ai->blocksize != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
if (ai->blocksize == 0)
|
|
bsize = audio_blocksize;
|
|
else if (ai->blocksize > NBPG/2)
|
|
bsize = NBPG/2;
|
|
else
|
|
bsize = ai->blocksize;
|
|
ai->blocksize = sc->sc_blksize = bsize;
|
|
audio_initbuf(sc);
|
|
}
|
|
|
|
if (ai->hiwat != ~0) {
|
|
if ((unsigned)ai->hiwat > sc->sc_maxblk)
|
|
ai->hiwat = sc->sc_maxblk;
|
|
sc->sc_hiwat = ai->hiwat;
|
|
}
|
|
if (ai->lowat != ~0) {
|
|
if ((unsigned)ai->lowat > sc->sc_maxblk)
|
|
ai->lowat = sc->sc_maxblk;
|
|
sc->sc_lowat = ai->lowat;
|
|
}
|
|
if (ai->backlog != ~0) {
|
|
if ((unsigned)ai->backlog > (sc->sc_maxblk/2))
|
|
ai->backlog = sc->sc_maxblk/2;
|
|
sc->sc_backlog = ai->backlog;
|
|
}
|
|
if (ai->mode != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
sc->sc_mode = ai->mode;
|
|
if (sc->sc_mode == AUMODE_PLAY)
|
|
audio_init_play(sc);
|
|
else
|
|
audio_init_record(sc);
|
|
}
|
|
#if 0
|
|
if (cleared) {
|
|
if (sc->sc_mode != AUMODE_PLAY)
|
|
audiostartr(sc);
|
|
else
|
|
audiostartp(sc);
|
|
}
|
|
#endif
|
|
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 = sc->sc_orate;
|
|
r->sample_rate = sc->sc_irate;
|
|
p->channels = r->channels = 1;
|
|
p->precision = r->precision = 8;
|
|
p->encoding = sc->sc_pencoding;
|
|
r->encoding = sc->sc_rencoding;
|
|
|
|
ai->monitor_gain = 0;
|
|
r->gain = sc->sc_rlevel;
|
|
p->gain = sc->sc_plevel;
|
|
r->port = 1; p->port = AUDIO_SPEAKER;
|
|
|
|
#ifdef notdef
|
|
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;
|
|
#endif
|
|
p->open = sc->sc_open;
|
|
r->open = sc->sc_open;
|
|
|
|
#ifdef notdef
|
|
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;
|
|
#endif
|
|
|
|
p->seek = sc->sc_wseek;
|
|
r->seek = sc->sc_rseek;
|
|
|
|
ai->blocksize = sc->sc_blksize;
|
|
ai->hiwat = sc->sc_hiwat;
|
|
ai->lowat = sc->sc_lowat;
|
|
ai->backlog = sc->sc_backlog;
|
|
ai->mode = sc->sc_mode;
|
|
|
|
return (0);
|
|
}
|
|
u_char mulawtolin[256] = {
|
|
128, 4, 8, 12, 16, 20, 24, 28,
|
|
32, 36, 40, 44, 48, 52, 56, 60,
|
|
64, 66, 68, 70, 72, 74, 76, 78,
|
|
80, 82, 84, 86, 88, 90, 92, 94,
|
|
96, 97, 98, 99, 100, 101, 102, 103,
|
|
104, 105, 106, 107, 108, 109, 110, 111,
|
|
112, 112, 113, 113, 114, 114, 115, 115,
|
|
116, 116, 117, 117, 118, 118, 119, 119,
|
|
120, 120, 120, 121, 121, 121, 121, 122,
|
|
122, 122, 122, 123, 123, 123, 123, 124,
|
|
124, 124, 124, 124, 125, 125, 125, 125,
|
|
125, 125, 125, 125, 126, 126, 126, 126,
|
|
126, 126, 126, 126, 126, 126, 126, 126,
|
|
127, 127, 127, 127, 127, 127, 127, 127,
|
|
127, 127, 127, 127, 127, 127, 127, 127,
|
|
127, 127, 127, 127, 127, 127, 127, 127,
|
|
255, 251, 247, 243, 239, 235, 231, 227,
|
|
223, 219, 215, 211, 207, 203, 199, 195,
|
|
191, 189, 187, 185, 183, 181, 179, 177,
|
|
175, 173, 171, 169, 167, 165, 163, 161,
|
|
159, 158, 157, 156, 155, 154, 153, 152,
|
|
151, 150, 149, 148, 147, 146, 145, 144,
|
|
143, 143, 142, 142, 141, 141, 140, 140,
|
|
139, 139, 138, 138, 137, 137, 136, 136,
|
|
135, 135, 135, 134, 134, 134, 134, 133,
|
|
133, 133, 133, 132, 132, 132, 132, 131,
|
|
131, 131, 131, 131, 130, 130, 130, 130,
|
|
130, 130, 130, 130, 129, 129, 129, 129,
|
|
129, 129, 129, 129, 129, 129, 129, 129,
|
|
128, 128, 128, 128, 128, 128, 128, 128,
|
|
128, 128, 128, 128, 128, 128, 128, 128,
|
|
128, 128, 128, 128, 128, 128, 128, 128,
|
|
};
|
|
u_char lintomulaw[256] = {
|
|
0, 0, 0, 0, 0, 1, 1, 1,
|
|
1, 2, 2, 2, 2, 3, 3, 3,
|
|
3, 4, 4, 4, 4, 5, 5, 5,
|
|
5, 6, 6, 6, 6, 7, 7, 7,
|
|
7, 8, 8, 8, 8, 9, 9, 9,
|
|
9, 10, 10, 10, 10, 11, 11, 11,
|
|
11, 12, 12, 12, 12, 13, 13, 13,
|
|
13, 14, 14, 14, 14, 15, 15, 15,
|
|
15, 16, 16, 17, 17, 18, 18, 19,
|
|
19, 20, 20, 21, 21, 22, 22, 23,
|
|
23, 24, 24, 25, 25, 26, 26, 27,
|
|
27, 28, 28, 29, 29, 30, 30, 31,
|
|
31, 32, 33, 34, 35, 36, 37, 38,
|
|
39, 40, 41, 42, 43, 44, 45, 46,
|
|
47, 48, 50, 52, 54, 56, 58, 60,
|
|
62, 65, 69, 73, 77, 83, 91, 103,
|
|
255, 231, 219, 211, 205, 201, 197, 193,
|
|
190, 188, 186, 184, 182, 180, 178, 176,
|
|
175, 174, 173, 172, 171, 170, 169, 168,
|
|
167, 166, 165, 164, 163, 162, 161, 160,
|
|
159, 159, 158, 158, 157, 157, 156, 156,
|
|
155, 155, 154, 154, 153, 153, 152, 152,
|
|
151, 151, 150, 150, 149, 149, 148, 148,
|
|
147, 147, 146, 146, 145, 145, 144, 144,
|
|
143, 143, 143, 143, 142, 142, 142, 142,
|
|
141, 141, 141, 141, 140, 140, 140, 140,
|
|
139, 139, 139, 139, 138, 138, 138, 138,
|
|
137, 137, 137, 137, 136, 136, 136, 136,
|
|
135, 135, 135, 135, 134, 134, 134, 134,
|
|
133, 133, 133, 133, 132, 132, 132, 132,
|
|
131, 131, 131, 131, 130, 130, 130, 130,
|
|
129, 129, 129, 129, 128, 128, 128, 128,
|
|
};
|
|
|
|
#if 0
|
|
void
|
|
audio_tomulaw(p, cc)
|
|
register u_char *p;
|
|
register int cc;
|
|
{
|
|
register u_char *utab = lintomulaw;
|
|
while (--cc >= 0) {
|
|
*p = utab[*p];
|
|
++p;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
static inline char
|
|
short2char(x)
|
|
short x;
|
|
{
|
|
if (x < 0){
|
|
x /= -256;
|
|
return (-1 * (char)x);
|
|
} else {
|
|
return ((char)x/256);
|
|
}
|
|
}
|
|
|
|
void
|
|
audio_tomulaw(p, cc)
|
|
short *p;
|
|
register int cc;
|
|
{
|
|
register u_char *utab = lintomulaw;
|
|
register int i;
|
|
for (i = 0; i < cc; i++) {
|
|
transbuf2[i] = utab[(u_char)(short2char(p[i]))];
|
|
}
|
|
}
|
|
|
|
void
|
|
audio_frommulaw(p, cc)
|
|
u_char *p;
|
|
register int cc;
|
|
{
|
|
register u_char *utab = mulawtolin;
|
|
register int i;
|
|
for (i = 0; i < cc; i++) {
|
|
transbuf[i] = (short)utab[p[i]];
|
|
}
|
|
}
|
|
|
|
double adpcm_estimindex[16] = {
|
|
1.0/8, 3.0/8, 5.0/8, 7.0/8, 9.0/8, 11.0/8, 13.0/8, 15.0/8,
|
|
-1.0/8, -3.0/8, -5.0/8, -7.0/8, -9.0/8, -11.0/8, -13.0/8, -15.0/8
|
|
};
|
|
|
|
double adpcm_estim[49] = {
|
|
16, 17, 19, 21, 23, 25, 28, 31, 34, 37,
|
|
41, 45, 50, 55, 60, 66, 73, 80, 88, 97,
|
|
107, 118, 130, 143, 157, 173, 190, 209, 230, 253,
|
|
279, 307, 337, 371, 408, 449, 494, 544, 598, 658,
|
|
724, 796, 875, 963, 1060, 1166, 1282, 1411, 1552 };
|
|
|
|
u_char adpcm_estimindex_correct[16] = {
|
|
-1, -1, -1, -1, 2, 4, 6, 8,
|
|
-1, -1, -1, -1, 2, 4, 6, 8
|
|
};
|
|
|
|
static inline void
|
|
adpcm2pcm_step(b, y, x)
|
|
u_char b;
|
|
short *y;
|
|
signed char *x;
|
|
{
|
|
*y += (short)(adpcm_estimindex[b] * adpcm_estim[*x]);
|
|
*x += adpcm_estimindex_correct[b];
|
|
if (*x < 0)
|
|
*x = 0;
|
|
else if (*x > 48)
|
|
*x = 48;
|
|
}
|
|
|
|
void
|
|
audio_tolinear(sc, p, cc)
|
|
struct audio_softc *sc;
|
|
register u_char *p;
|
|
register int cc;
|
|
{
|
|
signed char *x = &(sc->sc_adpcm.sc_estim);
|
|
short *y = &(sc->sc_adpcm.sc_amp);
|
|
u_char a, b;
|
|
int i;
|
|
|
|
AUDIODEBUG("audio_tolinear:\n");
|
|
for (i = 0; i < cc*2;) {
|
|
a = *p;
|
|
p++;
|
|
b = a & 0x0f;
|
|
adpcm2pcm_step(b, y, x);
|
|
transbuf[i++] = *y;
|
|
b = a >> 4;
|
|
adpcm2pcm_step(b, y, x);
|
|
transbuf[i++] = *y;
|
|
}
|
|
}
|
|
#if 0
|
|
void
|
|
audio_tolinear(sc, p, cc)
|
|
struct audio_softc *sc;
|
|
register u_char *p;
|
|
register int cc;
|
|
{
|
|
signed char x = 0;
|
|
short y = audio_softc[0].sc_amp;
|
|
u_char a, b;
|
|
int i;
|
|
|
|
for (i = 0; i < cc; i++) {
|
|
a = *p;
|
|
p++;
|
|
b = a & 0x0f;
|
|
y += (short)floor(adpcm_estimindex[b] * adpcm_estim[x]);
|
|
transbuf[i] = y;
|
|
x += adpcm_estimindex_correct[b];
|
|
if (x < 0)
|
|
x = 0;
|
|
else if (x > 48)
|
|
x = 48;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
inline u_char
|
|
pcm2adpcm_step(a, y, x)
|
|
short a;
|
|
short *y;
|
|
signed char *x;
|
|
{
|
|
register u_char b;
|
|
double c, d;
|
|
|
|
a -= *y;
|
|
c = (double)a*4.0 / (d = adpcm_estim[*x]);
|
|
if (c < 0.0) {
|
|
b = (u_char)-c;
|
|
if (b >= 8)
|
|
b = 7;
|
|
b |= 0x08;
|
|
} else {
|
|
b = (u_char)c;
|
|
if (b >= 8)
|
|
b = 7;
|
|
}
|
|
|
|
*y += (short)(adpcm_estimindex[b] * d);
|
|
*x += adpcm_estimindex_correct[b];
|
|
if (*x < 0)
|
|
*x = 0;
|
|
else if (*x > 48)
|
|
*x = 48;
|
|
return b;
|
|
}
|
|
|
|
void
|
|
audio_fromlinear(sc, p, cc)
|
|
struct audio_softc *sc;
|
|
register u_char *p;
|
|
register int cc;
|
|
{
|
|
signed char *x = &(sc->sc_adpcm.sc_estim);
|
|
short *y = &(sc->sc_adpcm.sc_amp);
|
|
u_char f;
|
|
register int i;
|
|
|
|
for (i = 0; i < cc;) {
|
|
f = pcm2adpcm_step(transbuf[i++], y, x);
|
|
*p++ = f + (pcm2adpcm_step(transbuf[i++], y, x) << 4);
|
|
}
|
|
}
|
|
|
|
#endif
|