1725 lines
39 KiB
C
1725 lines
39 KiB
C
/* $NetBSD: audio.c,v 1.9 1995/06/04 15:33:50 brezak 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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/*
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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 tries to do something half-way sensible with
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* half-duplex hardware, such as with the SoundBlaster hardware. With
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* half-duplex hardware allowing O_RDWR access doesn't really make
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* sense. However, closing and opening the device to "turn around the
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* line" is relatively expensive and costs a card reset (which can
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* take some time, at least for the SoundBlaster hardware). Instead
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* we allow O_RDWR access, and provide an ioctl to set the "mode",
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* i.e. playing or recording.
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*
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* If you write to a half-duplex device in record mode, the data is
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* tossed. If you read from the device in play mode, you get silence
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* filled buffers at the rate at which samples are naturally
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* generated.
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*
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* If you try to set both play and record mode on a half-duplex
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* device, playing takes precedence.
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*/
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/*
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* Todo:
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* - Add softaudio() isr processing for wakeup, select and signals.
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* - Allow opens for READ and WRITE (one open each)
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* - Setup for single isr for full-duplex
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* - Add SIGIO generation for changes in the mixer device
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* - Fixup SunOS compat for mixer device changes in ioctl.
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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/fcntl.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/proc.h>
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#include <sys/systm.h>
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#include <sys/syslog.h>
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#include <sys/audioio.h>
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#include <dev/audiovar.h>
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#include <dev/audio_if.h>
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#ifdef DEBUG
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#include <machine/stdarg.h>
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#ifndef TOLOG
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#define TOLOG 0x04
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#endif
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void kprintf __P((const char *fmt, int flags, struct tty *tp, va_list ap));
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void
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#ifdef __STDC__
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Dprintf(const char *fmt, ...)
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#else
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Dprintf(fmt, va_alist)
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char *fmt;
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#endif
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{
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va_list ap;
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va_start(ap, fmt);
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kprintf(fmt, TOLOG, NULL, ap);
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va_end(ap);
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}
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#define DPRINTF(x) if (audiodebug) Dprintf x
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int audiodebug = 0;
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#else
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#define DPRINTF(x)
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#endif
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int naudio; /* Count of attached hardware drivers */
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int audio_blk_ms = AUDIO_BLK_MS;
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int audio_backlog = AUDIO_BACKLOG;
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int audio_blocksize;
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/* XXX */
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#define splaudio splclock
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/* A block of silence */
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char *auzero_block;
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struct audio_softc *audio_softc[NAUDIO];
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int audiosetinfo __P((struct audio_softc *, struct audio_info *));
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int audiogetinfo __P((struct audio_softc *, struct audio_info *));
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int audio_open __P((dev_t, int, int, struct proc *));
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int audio_close __P((dev_t, int, int, struct proc *));
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int audio_read __P((dev_t, struct uio *, int));
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int audio_write __P((dev_t, struct uio *, int));
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int audio_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
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int audio_select __P((dev_t, int, struct proc *));
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int mixer_open __P((dev_t, int, int, struct proc *));
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int mixer_close __P((dev_t, int, int, struct proc *));
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int mixer_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
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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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void audio_rint __P((struct audio_softc *));
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void audio_pint __P((struct audio_softc *));
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int audio_calc_blksize __P((struct audio_softc *));
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void audio_silence_fill __P((struct audio_softc *, u_char *, int));
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int audio_silence_copyout __P((struct audio_softc *, int, struct uio *));
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void audio_alloc_auzero __P((struct audio_softc *, int));
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#ifdef DEBUG
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void
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audio_printsc(struct audio_softc *sc)
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{
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printf("hwhandle %x hw_if %x ", sc->hw_hdl, sc->hw_if);
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printf("open %x mode %x\n", sc->sc_open, sc->sc_mode);
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printf("rchan %x wchan %x ", sc->sc_rchan, sc->sc_wchan);
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printf("rring blk %x pring nblk %x\n", sc->rr.nblk, sc->pr.nblk);
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printf("rbus %x pbus %x ", sc->sc_rbus, sc->sc_pbus);
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printf("blksz %d sib %d ", sc->sc_blksize, sc->sc_smpl_in_blk);
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printf("sp50ms %d backlog %d\n", sc->sc_50ms, sc->sc_backlog);
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printf("hiwat %d lowat %d rblks %d\n", sc->sc_hiwat, sc->sc_lowat,
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sc->sc_rblks);
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}
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#endif
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void
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audioattach(num)
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int num;
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{
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}
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/*
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* Called from hardware driver.
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*/
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int
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audio_hardware_attach(hwp, hdlp)
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struct audio_hw_if *hwp;
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void *hdlp;
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{
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int *zp, i;
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struct audio_softc *sc;
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if (naudio > NAUDIO) {
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DPRINTF(("audio_hardware_attach: not enough audio devices: %d > %d\n",
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naudio, NAUDIO));
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return(EINVAL);
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}
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/*
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* Malloc a softc for the device
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*/
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/* XXX Find the first free slot */
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audio_softc[naudio] = malloc(sizeof(struct audio_softc), M_DEVBUF, M_WAITOK);
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sc = audio_softc[naudio];
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bzero(sc, sizeof(struct audio_softc));
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/* XXX too paranoid? */
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if (hwp->open == 0 ||
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hwp->close == 0 ||
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hwp->set_in_sr == 0 ||
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hwp->get_in_sr == 0 ||
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hwp->set_out_sr == 0 ||
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hwp->get_out_sr == 0 ||
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hwp->set_encoding == 0 ||
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hwp->get_encoding == 0 ||
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hwp->set_precision == 0 ||
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hwp->get_precision == 0 ||
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hwp->set_channels == 0 ||
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hwp->get_channels == 0 ||
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hwp->round_blocksize == 0 ||
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hwp->set_out_port == 0 ||
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hwp->get_out_port == 0 ||
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hwp->set_in_port == 0 ||
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hwp->get_in_port == 0 ||
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hwp->commit_settings == 0 ||
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hwp->get_silence == 0 ||
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hwp->start_output == 0 ||
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hwp->start_input == 0 ||
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hwp->halt_output == 0 ||
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hwp->halt_input == 0 ||
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hwp->cont_output == 0 ||
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hwp->cont_input == 0 ||
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hwp->getdev == 0 ||
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hwp->set_port == 0 ||
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hwp->get_port == 0 ||
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hwp->query_devinfo == 0)
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return(EINVAL);
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sc->hw_if = hwp;
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sc->hw_hdl = hdlp;
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/*
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* Alloc DMA play and record buffers
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*/
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sc->rr.bp = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK);
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if (sc->rr.bp == 0) {
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return (ENOMEM);
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}
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sc->pr.bp = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK);
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if (sc->pr.bp == 0) {
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free(sc->rr.bp, M_DEVBUF);
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return (ENOMEM);
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}
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/*
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* Set default softc params
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*/
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sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_LINEAR;
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/*
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* Return the audio unit number
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*/
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hwp->audio_unit = naudio++;
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#ifdef DEBUG
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printf("audio: unit %d attached\n", hwp->audio_unit);
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#endif
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return(0);
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}
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int
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audio_hardware_detach(hwp)
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struct audio_hw_if *hwp;
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{
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struct audio_softc *sc;
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#ifdef DIAGNOSTIC
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if (!hwp)
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panic("audio_hardware_detach: null hwp");
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if (hwp->audio_unit > naudio)
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panic("audio_hardware_detach: invalid audio unit");
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#endif
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sc = audio_softc[hwp->audio_unit];
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if (hwp != sc->hw_if)
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return(EINVAL);
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if (sc->sc_open != 0)
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return(EBUSY);
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sc->hw_if = 0;
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/* Free audio buffers */
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free(sc->rr.bp, M_DEVBUF);
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free(sc->pr.bp, M_DEVBUF);
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free(sc, M_DEVBUF);
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audio_softc[hwp->audio_unit] = NULL;
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return(0);
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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, ifmt;
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struct proc *p;
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{
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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_open(dev, flags, ifmt, p);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return mixer_open(dev, flags, ifmt, p);
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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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, ifmt;
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struct proc *p;
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{
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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_close(dev, flags, ifmt, p);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return mixer_close(dev, flags, ifmt, p);
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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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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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_read(dev, uio, ioflag);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return ENODEV;
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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}
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int
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audiowrite(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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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_write(dev, uio, ioflag);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return ENODEV;
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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}
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int
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audioioctl(dev, cmd, addr, flag, p)
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dev_t dev;
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int cmd;
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caddr_t addr;
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int flag;
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struct proc *p;
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{
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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_ioctl(dev, cmd, addr, flag, p);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return mixer_ioctl(dev, cmd, addr, flag, p);
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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}
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int
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audioselect(dev, rw, p)
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dev_t dev;
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int rw;
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struct proc *p;
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{
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switch(AUDIODEV(dev)) {
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case SOUND_DEVICE:
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case AUDIO_DEVICE:
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return audio_select(dev, rw, p);
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/*NOTREACHED*/
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case MIXER_DEVICE:
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return 1;
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/*NOTREACHED*/
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default:
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break;
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}
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return EIO;
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}
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/*
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* Audio driver
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*/
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void
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audio_init_ring(rp, blksize)
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struct audio_buffer *rp;
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int blksize;
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{
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int nblk = AU_RING_SIZE / blksize;
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rp->ep = rp->bp + nblk * blksize;
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rp->hp = rp->tp = rp->bp;
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rp->maxblk = nblk;
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rp->nblk = 0;
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rp->cb_drops = 0;
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rp->cb_pdrops = 0;
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}
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void
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audio_initbufs(sc)
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struct audio_softc *sc;
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{
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int nblk = AU_RING_SIZE / sc->sc_blksize;
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audio_init_ring(&sc->rr, sc->sc_blksize);
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audio_init_ring(&sc->pr, sc->sc_blksize);
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sc->sc_lowat = 1;
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sc->sc_hiwat = nblk - sc->sc_lowat;
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}
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static inline int
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audio_sleep(chan, label)
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int *chan;
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char *label;
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{
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int st;
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if (!label)
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label = "audio";
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*chan = 1;
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st = (tsleep(chan, PWAIT | PCATCH, label, 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(chan);
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*chan = 0;
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}
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}
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int
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audio_open(dev, flags, ifmt, p)
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dev_t dev;
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int flags, ifmt;
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struct proc *p;
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{
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int unit = AUDIOUNIT(dev);
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struct audio_softc *sc;
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int s, error;
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struct audio_hw_if *hw;
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if (unit >= NAUDIO || !audio_softc[unit]) {
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DPRINTF(("audioopen: invalid device unit - %d\n", unit));
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return (ENODEV);
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}
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sc = audio_softc[unit];
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hw = sc->hw_if;
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DPRINTF(("audio: open(%x): flags=0x%x sc=0x%x hdl=0x%x\n", dev, flags, sc, sc->hw_hdl));
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if (hw == 0) /* Hardware has not attached to us... */
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return (ENXIO);
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if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0) /* XXX use flags */
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return (EBUSY);
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if ((error = hw->open(dev, flags)) != 0)
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return (error);
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if (flags&FREAD)
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sc->sc_open |= AUOPEN_READ;
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if (flags&FWRITE)
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sc->sc_open |= AUOPEN_WRITE;
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/*
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* Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
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* The /dev/audio is always (re)set to 8-bit MU-Law mono
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* For the other devices, you get what they were last set to.
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*/
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if (ISDEVAUDIO(dev)) {
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/* /dev/audio */
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hw->set_precision(sc->hw_hdl, 8);
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hw->set_encoding(sc->hw_hdl, AUDIO_ENCODING_ULAW);
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hw->set_in_sr(sc->hw_hdl, 8000);
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hw->set_out_sr(sc->hw_hdl, 8000);
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hw->set_channels(sc->hw_hdl, 1);
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sc->sc_pencoding = AUDIO_ENCODING_ULAW;
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sc->sc_rencoding = AUDIO_ENCODING_ULAW;
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}
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/*
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* Sample rate and precision are supposed to be set to proper
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* default values by the hardware driver, so that it may give
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* us these values.
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*/
|
|
#ifdef DIAGNOSTIC
|
|
if (hw->get_precision(sc->hw_hdl) == 0)
|
|
panic("audioopen: hardware driver returned 0 for get_precision");
|
|
#endif
|
|
sc->sc_blksize = audio_blocksize = audio_calc_blksize(sc);
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
|
|
sc->sc_smpl_in_blk = audio_blocksize /
|
|
(hw->get_precision(sc->hw_hdl) / NBBY);
|
|
sc->sc_50ms = 50 * hw->get_out_sr(sc->hw_hdl) / 1000;
|
|
sc->sc_backlog = audio_backlog;
|
|
|
|
audio_initbufs(sc);
|
|
DPRINTF(("audio: rr.bp=%x-%x pr.bp=%x-%x\n",
|
|
sc->rr.bp, sc->rr.ep, sc->pr.bp, sc->pr.ep));
|
|
|
|
hw->commit_settings(sc->hw_hdl);
|
|
|
|
s = splaudio();
|
|
|
|
/* nothing read or written yet */
|
|
sc->sc_rseek = 0;
|
|
sc->sc_wseek = 0;
|
|
|
|
sc->sc_rchan = 0;
|
|
sc->sc_wchan = 0;
|
|
|
|
sc->sc_rbus = 0;
|
|
sc->sc_pbus = 0;
|
|
|
|
if ((flags & FWRITE) != 0) {
|
|
audio_init_play(sc);
|
|
/* audio_pint(sc); ??? */
|
|
}
|
|
if ((flags & FREAD) != 0) {
|
|
/* Play takes precedence if HW is half-duplex */
|
|
if (hw->full_duplex || ((flags & FWRITE) == 0)) {
|
|
audio_init_record(sc);
|
|
/* audiostartr(sc); don't start recording until read */
|
|
}
|
|
}
|
|
if (ISDEVAUDIO(dev)) {
|
|
/* if open only for read or only for write, then set specific mode */
|
|
if ((flags & (FWRITE|FREAD)) == FWRITE) {
|
|
sc->sc_mode = 1 << AUMODE_PLAY;
|
|
sc->pr.cb_pause = 0;
|
|
sc->rr.cb_pause = 1;
|
|
audiostartp(sc);
|
|
} else if ((flags & (FWRITE|FREAD)) == FREAD) {
|
|
sc->sc_mode = 1 << AUMODE_RECORD;
|
|
sc->rr.cb_pause = 0;
|
|
sc->pr.cb_pause = 1;
|
|
audiostartr(sc);
|
|
}
|
|
}
|
|
splx(s);
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Must be called from task context.
|
|
*/
|
|
void
|
|
audio_init_record(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int s = splaudio();
|
|
|
|
sc->sc_mode |= 1<<AUMODE_RECORD;
|
|
if (sc->hw_if->speaker_ctl)
|
|
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF);
|
|
splx(s);
|
|
}
|
|
|
|
/*
|
|
* Must be called from task context.
|
|
*/
|
|
void
|
|
audio_init_play(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int s = splaudio();
|
|
|
|
sc->sc_mode |= 1<<AUMODE_PLAY;
|
|
sc->sc_rblks = 0;
|
|
if (sc->hw_if->speaker_ctl)
|
|
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
|
|
splx(s);
|
|
}
|
|
|
|
int
|
|
audio_drain(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int error;
|
|
|
|
while (sc->pr.nblk > 0) {
|
|
DPRINTF(("audio_drain: nblk=%d\n", sc->pr.nblk));
|
|
error = audio_sleep(&sc->sc_wchan, "aud dr");
|
|
if (error != 0)
|
|
return (error);
|
|
}
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Close an audio chip.
|
|
*/
|
|
/* ARGSUSED */
|
|
int
|
|
audio_close(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int s;
|
|
|
|
DPRINTF(("audio: close; unit %d\n", unit));
|
|
|
|
/*
|
|
* Block until output drains, but allow ^C interrupt.
|
|
*/
|
|
sc->sc_lowat = 0; /* avoid excessive wakeups */
|
|
s = splaudio();
|
|
/*
|
|
* If there is pending output, let it drain (unless
|
|
* the output is paused).
|
|
*/
|
|
if (sc->sc_pbus && sc->pr.nblk > 0 && !sc->pr.cb_pause) {
|
|
if (!audio_drain(sc) && hw->drain)
|
|
(void)hw->drain(sc->hw_hdl);
|
|
}
|
|
|
|
hw->close(sc->hw_hdl);
|
|
|
|
if (flags&FREAD)
|
|
sc->sc_open &= ~AUOPEN_READ;
|
|
if (flags&FWRITE)
|
|
sc->sc_open &= ~AUOPEN_WRITE;
|
|
|
|
splx(s);
|
|
DPRINTF(("audio: close done\n"));
|
|
|
|
return (0);
|
|
}
|
|
|
|
int
|
|
audio_read(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_buffer *cb = &sc->rr;
|
|
u_char *hp;
|
|
int blocksize = sc->sc_blksize;
|
|
int error, s;
|
|
|
|
DPRINTF(("audio: read cc=%d mode=%d rblks=%d\n",
|
|
uio->uio_resid, sc->sc_mode, sc->sc_rblks));
|
|
|
|
if (uio->uio_resid == 0)
|
|
return (0);
|
|
|
|
if (uio->uio_resid < blocksize)
|
|
return (EINVAL);
|
|
|
|
/* First sample we'll read in sample space */
|
|
sc->sc_rseek = cb->au_stamp - AU_RING_LEN(cb);
|
|
|
|
/*
|
|
* If hardware is half-duplex and currently playing, return
|
|
* silence blocks based on the number of blocks we have output.
|
|
*/
|
|
if ((!hw->full_duplex) &&
|
|
(sc->sc_mode & 1<<AUMODE_PLAY)) {
|
|
do {
|
|
s = splaudio();
|
|
while (sc->sc_rblks <= 0) {
|
|
DPRINTF(("audioread: sc_rblks=%d\n", sc->sc_rblks));
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return (EWOULDBLOCK);
|
|
}
|
|
error = audio_sleep(&sc->sc_rchan, "aud hr");
|
|
if (error != 0) {
|
|
splx(s);
|
|
return (error);
|
|
}
|
|
}
|
|
splx(s);
|
|
error = audio_silence_copyout(sc, blocksize, uio);
|
|
if (error)
|
|
break;
|
|
--sc->sc_rblks;
|
|
} while (uio->uio_resid >= blocksize);
|
|
return (error);
|
|
}
|
|
error = 0;
|
|
do {
|
|
while (cb->nblk <= 0) {
|
|
if (ioflag & IO_NDELAY) {
|
|
error = EWOULDBLOCK;
|
|
return (error);
|
|
}
|
|
s = splaudio();
|
|
if (!sc->sc_rbus)
|
|
audiostartr(sc);
|
|
error = audio_sleep(&sc->sc_rchan, "aud rd");
|
|
splx(s);
|
|
if (error != 0)
|
|
return (error);
|
|
}
|
|
hp = cb->hp;
|
|
if (hw->sw_decode)
|
|
hw->sw_decode(sc->sc_rencoding, hp, blocksize);
|
|
error = uiomove(hp, blocksize, uio);
|
|
if (error)
|
|
break;
|
|
hp += blocksize;
|
|
if (hp >= cb->ep)
|
|
hp = cb->bp;
|
|
cb->hp = hp;
|
|
--cb->nblk;
|
|
} while (uio->uio_resid >= blocksize);
|
|
|
|
return (error);
|
|
}
|
|
|
|
void
|
|
audio_clear(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int s = splaudio();
|
|
|
|
if (sc->sc_rbus || sc->sc_pbus) {
|
|
sc->hw_if->halt_output(sc->hw_hdl);
|
|
sc->hw_if->halt_input(sc->hw_hdl);
|
|
sc->sc_rbus = 0;
|
|
sc->sc_pbus = 0;
|
|
}
|
|
AU_RING_INIT(&sc->rr);
|
|
AU_RING_INIT(&sc->pr);
|
|
|
|
splx(s);
|
|
}
|
|
|
|
int
|
|
audio_calc_blksize(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int bs;
|
|
|
|
bs = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
bs = hw->get_out_sr(sc->hw_hdl) * audio_blk_ms * bs / 1000;
|
|
bs *= hw->get_channels(sc->hw_hdl);
|
|
bs = hw->round_blocksize(sc->hw_hdl, bs);
|
|
if (bs > AU_RING_SIZE)
|
|
bs = AU_RING_SIZE;
|
|
|
|
bs &= ~1; /* make it even, in case of stereo */
|
|
return(bs);
|
|
}
|
|
|
|
void
|
|
audio_silence_fill(sc, p, n)
|
|
struct audio_softc *sc;
|
|
u_char *p;
|
|
int n;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int i;
|
|
u_int auzero;
|
|
|
|
auzero = hw->get_silence(sc->sc_pencoding);
|
|
|
|
while (--n >= 0)
|
|
*p++ = auzero;
|
|
}
|
|
|
|
audio_silence_copyout(sc, n, uio)
|
|
struct audio_softc *sc;
|
|
int n;
|
|
struct uio *uio;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct iovec *iov;
|
|
int error = 0;
|
|
u_int auzero;
|
|
|
|
auzero = hw->get_silence(sc->sc_rencoding);
|
|
|
|
while (n > 0 && uio->uio_resid) {
|
|
iov = uio->uio_iov;
|
|
if (iov->iov_len == 0) {
|
|
uio->uio_iov++;
|
|
uio->uio_iovcnt--;
|
|
continue;
|
|
}
|
|
switch (uio->uio_segflg) {
|
|
case UIO_USERSPACE:
|
|
error = copyout(&auzero, iov->iov_base, 1);
|
|
if (error)
|
|
return (error);
|
|
break;
|
|
|
|
case UIO_SYSSPACE:
|
|
bcopy(&auzero, iov->iov_base, 1);
|
|
break;
|
|
}
|
|
iov->iov_base++;
|
|
iov->iov_len--;
|
|
uio->uio_resid--;
|
|
uio->uio_offset++;
|
|
n--;
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
void
|
|
audio_alloc_auzero(sc, bs)
|
|
struct audio_softc *sc;
|
|
int bs;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
u_char *p, silence;
|
|
int i;
|
|
|
|
if (auzero_block)
|
|
free(auzero_block, M_DEVBUF);
|
|
|
|
auzero_block = malloc(bs, M_DEVBUF, M_WAITOK);
|
|
#ifdef DIAGNOSTIC
|
|
if (auzero_block == 0) {
|
|
panic("audio_alloc_auzero: malloc auzero_block failed");
|
|
}
|
|
#endif
|
|
silence = hw->get_silence(sc->sc_pencoding);
|
|
|
|
p = auzero_block;
|
|
for (i = 0; i < bs; i++)
|
|
*p++ = silence;
|
|
|
|
if (hw->sw_encode)
|
|
hw->sw_encode(sc->sc_pencoding, auzero_block, bs);
|
|
}
|
|
|
|
|
|
int
|
|
audio_write(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_buffer *cb = &sc->pr;
|
|
u_char *tp;
|
|
int error, s, cc;
|
|
int blocksize = sc->sc_blksize;
|
|
|
|
DPRINTF(("audio: write cc=%d hiwat=%d\n", uio->uio_resid, sc->sc_hiwat));
|
|
/*
|
|
* If half-duplex and currently recording, throw away data.
|
|
*/
|
|
if (!hw->full_duplex &&
|
|
(sc->sc_mode & 1<<AUMODE_RECORD)) {
|
|
uio->uio_offset += uio->uio_resid;
|
|
uio->uio_resid = 0;
|
|
DPRINTF(("audiowrite: half-dpx read busy\n"));
|
|
return (0);
|
|
}
|
|
error = 0;
|
|
|
|
while (uio->uio_resid > 0) {
|
|
int watermark = sc->sc_hiwat;
|
|
while (cb->nblk > watermark) {
|
|
DPRINTF(("audiowrite: nblk=%d watermark=%d\n", cb->nblk, watermark));
|
|
if (ioflag & IO_NDELAY) {
|
|
error = EWOULDBLOCK;
|
|
return (error);
|
|
}
|
|
error = audio_sleep(&sc->sc_wchan, "aud wr");
|
|
if (error != 0) {
|
|
return (error);
|
|
}
|
|
watermark = sc->sc_lowat;
|
|
}
|
|
#if 0
|
|
if (cb->nblk == 0 &&
|
|
cb->maxblk > sc->sc_backlog &&
|
|
uio->uio_resid <= blocksize &&
|
|
(cb->au_stamp - sc->sc_wseek) > sc->sc_50ms) {
|
|
/*
|
|
* 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.).
|
|
*/
|
|
DPRINTF(("audiowrite: set backlog %d\n", sc->sc_backlog));
|
|
s = splaudio();
|
|
cb->hp = cb->bp;
|
|
cb->nblk = sc->sc_backlog;
|
|
cb->tp = cb->hp + sc->sc_backlog * blocksize;
|
|
splx(s);
|
|
audio_silence_fill(sc, cb->hp, sc->sc_backlog * blocksize);
|
|
}
|
|
#endif
|
|
/* Calculate sample number of first sample in block we write */
|
|
s = splaudio();
|
|
sc->sc_wseek = AU_RING_LEN(cb) + cb->au_stamp;
|
|
splx(s);
|
|
|
|
tp = cb->tp;
|
|
cc = uio->uio_resid;
|
|
|
|
#ifdef DEBUG
|
|
if (audiodebug > 1) {
|
|
int left = cb->ep - tp;
|
|
Dprintf("audiowrite: cc=%d tp=0x%x bs=%d nblk=%d left=%d\n", cc, tp, blocksize, cb->nblk, left);
|
|
}
|
|
#endif
|
|
#ifdef DIAGNOSTIC
|
|
{
|
|
int towrite = (cc < blocksize)?cc:blocksize;
|
|
|
|
/* check for an overwrite. Should never happen */
|
|
if ((tp + towrite) > cb->ep) {
|
|
DPRINTF(("audiowrite: overwrite tp=0x%x towrite=%d ep=0x%x bs=%d\n",
|
|
tp, towrite, cb->ep, blocksize));
|
|
printf("audiowrite: overwrite tp=0x%x towrite=%d ep=0x%x\n",
|
|
tp, towrite, cb->ep);
|
|
tp = cb->bp;
|
|
}
|
|
}
|
|
#endif
|
|
if (cc < blocksize) {
|
|
error = uiomove(tp, cc, uio);
|
|
if (error == 0) {
|
|
/* fill with audio silence */
|
|
tp += cc;
|
|
cc = blocksize - cc;
|
|
audio_silence_fill(sc, tp, cc);
|
|
DPRINTF(("audiowrite: auzero 0x%x %d 0x%x\n",
|
|
tp, cc, *(int *)tp));
|
|
tp += cc;
|
|
}
|
|
} else {
|
|
error = uiomove(tp, blocksize, uio);
|
|
if (error == 0) {
|
|
tp += blocksize;
|
|
}
|
|
}
|
|
if (error) {
|
|
#ifdef DEBUG
|
|
printf("audiowrite1: uiomove failed %d; cc=%d tp=0x%x bs=%d\n", error, cc, tp, blocksize);
|
|
#endif
|
|
break;
|
|
}
|
|
|
|
if (hw->sw_encode) {
|
|
hw->sw_encode(sc->sc_pencoding, cb->tp, blocksize);
|
|
}
|
|
|
|
/* wrap the ring buffer if at end */
|
|
s = splaudio();
|
|
if (tp >= cb->ep)
|
|
tp = cb->bp;
|
|
cb->tp = tp;
|
|
++cb->nblk;
|
|
|
|
/*
|
|
* If output isn't active, start it up.
|
|
*/
|
|
if (sc->sc_pbus == 0)
|
|
audiostartp(sc);
|
|
splx(s);
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
int
|
|
audio_ioctl(dev, cmd, addr, flag, p)
|
|
dev_t dev;
|
|
int cmd;
|
|
caddr_t addr;
|
|
int flag;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int error = 0, s;
|
|
|
|
DPRINTF(("audio: ioctl(%d,'%c',%d)\n",
|
|
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
|
|
switch (cmd) {
|
|
|
|
case AUDIO_FLUSH:
|
|
DPRINTF(("AUDIO_FLUSH\n"));
|
|
audio_clear(sc);
|
|
s = splaudio();
|
|
if ((sc->sc_mode & 1<<AUMODE_PLAY) && (sc->sc_pbus == 0))
|
|
audiostartp(sc);
|
|
/* Again, play takes precedence on half-duplex hardware */
|
|
if ((sc->sc_mode & 1<<AUMODE_RECORD) &&
|
|
(hw->full_duplex ||
|
|
((sc->sc_mode & 1<<AUMODE_PLAY) == 0)))
|
|
audiostartr(sc);
|
|
splx(s);
|
|
break;
|
|
|
|
/*
|
|
* Number of read samples dropped. We don't know where or
|
|
* when they were dropped.
|
|
*/
|
|
case AUDIO_RERROR:
|
|
*(int *)addr = sc->rr.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->pr.au_stamp
|
|
+ AU_RING_LEN(&sc->rr);
|
|
splx(s);
|
|
break;
|
|
case AUDIO_SETINFO:
|
|
DPRINTF(("AUDIO_SETINFO\n"));
|
|
error = audiosetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETINFO:
|
|
DPRINTF(("AUDIO_GETINFO\n"));
|
|
error = audiogetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_DRAIN:
|
|
DPRINTF(("AUDIO_DRAIN\n"));
|
|
error = audio_drain(sc);
|
|
if (!error && hw->drain)
|
|
error = hw->drain(sc->hw_hdl);
|
|
break;
|
|
|
|
case AUDIO_GETDEV:
|
|
DPRINTF(("AUDIO_GETDEV\n"));
|
|
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETENC:
|
|
DPRINTF(("AUDIO_GETENC\n"));
|
|
error = hw->query_encoding(sc->hw_hdl, (struct audio_encoding *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETFD:
|
|
DPRINTF(("AUDIO_GETFD\n"));
|
|
*(int *)addr = hw->full_duplex;
|
|
break;
|
|
|
|
case AUDIO_SETFD:
|
|
DPRINTF(("AUDIO_SETFD\n"));
|
|
error = hw->setfd(sc->hw_hdl, *(int *)addr);
|
|
break;
|
|
|
|
default:
|
|
DPRINTF(("audio: unknown ioctl\n"));
|
|
error = EINVAL;
|
|
break;
|
|
}
|
|
DPRINTF(("audio: ioctl(%d,'%c',%d) result %d\n",
|
|
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
|
|
return (error);
|
|
}
|
|
|
|
int
|
|
audio_select(dev, rw, p)
|
|
dev_t dev;
|
|
int rw;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
int s = splaudio();
|
|
|
|
#if 0
|
|
DPRINTF(("audio: select rw=%d mode=%d rblks=%d rr.nblk=%d\n",
|
|
rw, sc->sc_mode, sc->sc_rblks, sc->rr.nblk));
|
|
#endif
|
|
switch (rw) {
|
|
|
|
case FREAD:
|
|
if (sc->sc_mode & 1<<AUMODE_PLAY) {
|
|
if (sc->sc_rblks > 0) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
} else if (sc->rr.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.
|
|
*
|
|
* XXX The above XXX-comment is SoundBlaster-dependent,
|
|
* right? Or maybe specific to half-duplex devices?
|
|
*/
|
|
if (sc->sc_mode & 1<<AUMODE_RECORD ||
|
|
sc->pr.nblk < sc->sc_lowat) {
|
|
splx(s);
|
|
return (1);
|
|
}
|
|
selrecord(p, &sc->sc_wsel);
|
|
break;
|
|
}
|
|
splx(s);
|
|
return (0);
|
|
}
|
|
|
|
void
|
|
audiostartr(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int err;
|
|
|
|
DPRINTF(("audiostartr: tp=0x%x\n", sc->rr.tp));
|
|
|
|
if (err = sc->hw_if->start_input(sc->hw_hdl, sc->rr.tp, sc->sc_blksize,
|
|
audio_rint, (void *)sc)) {
|
|
DPRINTF(("audiostartr failed: %d\n", err));
|
|
audio_clear(sc);
|
|
}
|
|
else
|
|
sc->sc_rbus = 1;
|
|
}
|
|
|
|
void
|
|
audiostartp(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int rval;
|
|
|
|
DPRINTF(("audiostartp: hp=0x%x nblk=%d\n", sc->pr.hp, sc->pr.nblk));
|
|
|
|
if (sc->pr.nblk > 0) {
|
|
u_char *hp = sc->pr.hp;
|
|
if (rval = sc->hw_if->start_output(sc->hw_hdl, hp, sc->sc_blksize,
|
|
audio_pint, (void *)sc)) {
|
|
DPRINTF(("audiostartp failed: %d\n", rval));
|
|
}
|
|
else {
|
|
--sc->pr.nblk;
|
|
sc->sc_pbus = 1;
|
|
hp += sc->sc_blksize;
|
|
if (hp >= sc->pr.ep)
|
|
hp = sc->pr.bp;
|
|
sc->pr.hp = hp;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from HW driver module on completion of dma output.
|
|
* Start output of new block, wrap in ring buffer if needed.
|
|
* If no more buffers to play, output zero instead.
|
|
* Do a wakeup if necessary.
|
|
*/
|
|
void
|
|
audio_pint(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
u_char *hp;
|
|
int cc = sc->sc_blksize;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_buffer *cb = &sc->pr;
|
|
int err;
|
|
|
|
if (cb->nblk > 0) {
|
|
hp = cb->hp;
|
|
if (cb->cb_pause) {
|
|
cb->cb_pdrops++;
|
|
#ifdef DEBUG
|
|
if (audiodebug > 1)
|
|
Dprintf("audio_pint: paused %d\n", cb->cb_pdrops);
|
|
#endif
|
|
goto psilence;
|
|
}
|
|
else {
|
|
#ifdef DEBUG
|
|
if (audiodebug > 1)
|
|
Dprintf("audio_pint: hp=0x%x cc=%d\n", hp, cc);
|
|
#endif
|
|
if (err = hw->start_output(sc->hw_hdl, hp, cc,
|
|
audio_pint, (void *)sc)) {
|
|
DPRINTF(("audio_pint restart failed: %d\n", err));
|
|
audio_clear(sc);
|
|
}
|
|
else {
|
|
--cb->nblk;
|
|
hp += cc;
|
|
if (hp >= cb->ep)
|
|
hp = cb->bp;
|
|
cb->hp = hp;
|
|
cb->au_stamp += sc->sc_smpl_in_blk;
|
|
|
|
++sc->sc_rblks;
|
|
}
|
|
}
|
|
}
|
|
else {
|
|
cb->cb_drops++;
|
|
#ifdef DEBUG
|
|
if (audiodebug > 1)
|
|
Dprintf("audio_pint: drops=%d auzero %d 0x%x\n", cb->cb_drops, cc, *(int *)auzero_block);
|
|
#endif
|
|
psilence:
|
|
if (err = hw->start_output(sc->hw_hdl,
|
|
auzero_block, cc,
|
|
audio_pint, (void *)sc)) {
|
|
DPRINTF(("audio_pint zero failed: %d\n", err));
|
|
audio_clear(sc);
|
|
}
|
|
}
|
|
|
|
#ifdef DEBUG
|
|
DPRINTF(("audio_pint: mode=%d pause=%d nblk=%d lowat=%d\n",
|
|
sc->sc_mode, cb->cb_pause, cb->nblk, sc->sc_lowat));
|
|
#endif
|
|
if (sc->sc_mode & 1<<AUMODE_PLAY && !cb->cb_pause) {
|
|
if (cb->nblk <= sc->sc_lowat) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
selwakeup(&sc->sc_wsel);
|
|
}
|
|
}
|
|
|
|
/* XXX possible to return one or more "phantom blocks" now. */
|
|
/* Only in half duplex? */
|
|
if (hw->full_duplex) {
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selwakeup(&sc->sc_rsel);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from HW driver module on completion of dma input.
|
|
* Mark it as input in the ring buffer (fiddle pointers).
|
|
* Do a wakeup if necessary.
|
|
*/
|
|
void
|
|
audio_rint(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
u_char *tp;
|
|
int cc = sc->sc_blksize;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_buffer *cb = &sc->rr;
|
|
int err;
|
|
|
|
tp = cb->tp;
|
|
if (cb->cb_pause) {
|
|
cb->cb_pdrops++;
|
|
DPRINTF(("audio_rint: pdrops %d\n", cb->cb_pdrops));
|
|
}
|
|
else {
|
|
tp += cc;
|
|
if (tp >= cb->ep)
|
|
tp = cb->bp;
|
|
if (++cb->nblk < cb->maxblk) {
|
|
#ifdef DEBUG
|
|
if (audiodebug > 1)
|
|
Dprintf("audio_rint: tp=0x%x cc=%d\n", tp, cc);
|
|
#endif
|
|
if (err = hw->start_input(sc->hw_hdl, tp, cc,
|
|
audio_rint, (void *)sc)) {
|
|
DPRINTF(("audio_rint start failed: %d\n", err));
|
|
audio_clear(sc);
|
|
}
|
|
cb->au_stamp += sc->sc_smpl_in_blk;
|
|
} else {
|
|
/* XXX */
|
|
/* How do we count dropped input samples due to overrun? */
|
|
/* Start a "dummy DMA transfer" when the input ring buffer */
|
|
/* is full and count # of these? Seems pretty lame to */
|
|
/* me, but how else are we going to do this? */
|
|
cb->cb_drops++;
|
|
sc->sc_rbus = 0;
|
|
DPRINTF(("audio_rint: drops %d\n", cb->cb_drops));
|
|
}
|
|
cb->tp = tp;
|
|
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selwakeup(&sc->sc_rsel);
|
|
}
|
|
}
|
|
|
|
int
|
|
audiosetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
int cleared = 0;
|
|
int bsize, bps, error = 0;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
mixer_ctrl_t ct;
|
|
|
|
if (hw == 0) /* HW has not attached */
|
|
return(ENXIO);
|
|
|
|
if (p->sample_rate != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_out_sr(sc->hw_hdl, p->sample_rate);
|
|
if (error != 0)
|
|
return(error);
|
|
sc->sc_50ms = 50 * hw->get_out_sr(sc->hw_hdl) / 1000;
|
|
}
|
|
if (r->sample_rate != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_in_sr(sc->hw_hdl, r->sample_rate);
|
|
if (error != 0)
|
|
return(error);
|
|
sc->sc_50ms = 50 * hw->get_in_sr(sc->hw_hdl) / 1000;
|
|
}
|
|
if (p->encoding != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_encoding(sc->hw_hdl, p->encoding);
|
|
if (error != 0)
|
|
return(error);
|
|
sc->sc_pencoding = p->encoding;
|
|
}
|
|
if (r->encoding != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_encoding(sc->hw_hdl, r->encoding);
|
|
if (error != 0)
|
|
return(error);
|
|
sc->sc_rencoding = r->encoding;
|
|
}
|
|
if (p->precision != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_precision(sc->hw_hdl, p->precision);
|
|
if (error != 0)
|
|
return(error);
|
|
|
|
sc->sc_blksize = audio_blocksize = audio_calc_blksize(sc);
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
bps = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
sc->sc_smpl_in_blk = sc->sc_blksize / bps;
|
|
audio_initbufs(sc);
|
|
}
|
|
if (r->precision != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_precision(sc->hw_hdl, r->precision);
|
|
if (error != 0)
|
|
return(error);
|
|
|
|
sc->sc_blksize = audio_blocksize = audio_calc_blksize(sc);
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
bps = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
sc->sc_smpl_in_blk = sc->sc_blksize / bps;
|
|
audio_initbufs(sc);
|
|
}
|
|
if (p->channels != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_channels(sc->hw_hdl, p->channels);
|
|
if (error != 0)
|
|
return(error);
|
|
|
|
sc->sc_blksize = audio_blocksize = audio_calc_blksize(sc);
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
bps = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
sc->sc_smpl_in_blk = sc->sc_blksize / bps;
|
|
audio_initbufs(sc);
|
|
}
|
|
if (r->channels != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_channels(sc->hw_hdl, r->channels);
|
|
if (error != 0)
|
|
return(error);
|
|
|
|
sc->sc_blksize = audio_blocksize = audio_calc_blksize(sc);
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
bps = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
sc->sc_smpl_in_blk = sc->sc_blksize / bps;
|
|
audio_initbufs(sc);
|
|
}
|
|
if (p->port != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_out_port(sc->hw_hdl, p->port);
|
|
if (error != 0)
|
|
return(error);
|
|
}
|
|
if (r->port != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
error = hw->set_in_port(sc->hw_hdl, r->port);
|
|
if (error != 0)
|
|
return(error);
|
|
}
|
|
if (p->gain != ~0) {
|
|
ct.dev = hw->get_out_port(sc->hw_hdl);
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = p->gain;
|
|
hw->set_port(sc->hw_hdl, &ct);
|
|
}
|
|
if (r->gain != ~0) {
|
|
ct.dev = hw->get_in_port(sc->hw_hdl);
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = r->gain;
|
|
hw->set_port(sc->hw_hdl, &ct);
|
|
}
|
|
|
|
if (p->pause != (u_char)~0) {
|
|
sc->pr.cb_pause = p->pause;
|
|
if (!p->pause) {
|
|
int s = splaudio();
|
|
audiostartp(sc);
|
|
splx(s);
|
|
}
|
|
}
|
|
if (r->pause != (u_char)~0) {
|
|
sc->rr.cb_pause = r->pause;
|
|
if (!r->pause) {
|
|
int s = splaudio();
|
|
audiostartr(sc);
|
|
splx(s);
|
|
}
|
|
}
|
|
|
|
if (ai->blocksize != ~0) { /* Explicitly specified, possibly */
|
|
/* overriding changes done above. */
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
if (ai->blocksize == 0)
|
|
bsize = audio_blocksize;
|
|
else if (ai->blocksize > AU_RING_SIZE)
|
|
bsize = AU_RING_SIZE;
|
|
else
|
|
bsize = ai->blocksize;
|
|
bsize = hw->round_blocksize(sc->hw_hdl, bsize);
|
|
if (bsize > AU_RING_SIZE)
|
|
bsize = AU_RING_SIZE;
|
|
ai->blocksize = sc->sc_blksize = bsize;
|
|
audio_alloc_auzero(sc, sc->sc_blksize);
|
|
bps = hw->get_precision(sc->hw_hdl) / NBBY;
|
|
sc->sc_smpl_in_blk = bsize / bps;
|
|
audio_initbufs(sc);
|
|
}
|
|
if (ai->hiwat != ~0) {
|
|
if ((unsigned)ai->hiwat > sc->pr.maxblk)
|
|
ai->hiwat = sc->pr.maxblk;
|
|
if (sc->sc_hiwat != 0)
|
|
sc->sc_hiwat = ai->hiwat;
|
|
}
|
|
if (ai->lowat != ~0) {
|
|
if ((unsigned)ai->lowat > sc->pr.maxblk)
|
|
ai->lowat = sc->pr.maxblk;
|
|
sc->sc_lowat = ai->lowat;
|
|
}
|
|
if (ai->backlog != ~0) {
|
|
if ((unsigned)ai->backlog > (sc->pr.maxblk/2))
|
|
ai->backlog = sc->pr.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 & 1<<AUMODE_PLAY) {
|
|
audio_init_play(sc);
|
|
if (!hw->full_duplex) /* Play takes precedence */
|
|
sc->sc_mode &= ~(1<<AUMODE_RECORD);
|
|
}
|
|
if (sc->sc_mode & 1<<AUMODE_RECORD)
|
|
audio_init_record(sc);
|
|
}
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
if (error != 0)
|
|
return (error);
|
|
|
|
if (cleared) {
|
|
int s = splaudio();
|
|
if (sc->sc_mode & 1<<AUMODE_PLAY)
|
|
audiostartp(sc);
|
|
if (sc->sc_mode & 1<<AUMODE_RECORD)
|
|
audiostartr(sc);
|
|
splx(s);
|
|
}
|
|
|
|
return (0);
|
|
}
|
|
|
|
int
|
|
audiogetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
mixer_ctrl_t ct;
|
|
|
|
if (hw == 0) /* HW has not attached */
|
|
return(ENXIO);
|
|
|
|
p->sample_rate = hw->get_out_sr(sc->hw_hdl);
|
|
r->sample_rate = hw->get_in_sr(sc->hw_hdl);
|
|
p->channels = r->channels = hw->get_channels(sc->hw_hdl);
|
|
p->precision = r->precision = hw->get_precision(sc->hw_hdl);
|
|
p->encoding = hw->get_encoding(sc->hw_hdl);
|
|
r->encoding = hw->get_encoding(sc->hw_hdl);
|
|
|
|
r->port = hw->get_in_port(sc->hw_hdl);
|
|
p->port = hw->get_out_port(sc->hw_hdl);
|
|
|
|
ct.dev = r->port;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
if (hw->get_port(sc->hw_hdl, &ct) == 0)
|
|
r->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
else
|
|
r->gain = AUDIO_MAX_GAIN/2;
|
|
|
|
ct.dev = p->port;
|
|
ct.un.value.num_channels = 1;
|
|
if (hw->get_port(sc->hw_hdl, &ct) == 0)
|
|
p->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
else
|
|
p->gain = AUDIO_MAX_GAIN/2;
|
|
|
|
p->pause = sc->pr.cb_pause;
|
|
r->pause = sc->rr.cb_pause;
|
|
p->error = sc->pr.cb_drops != 0;
|
|
r->error = sc->rr.cb_drops != 0;
|
|
|
|
p->open = ((sc->sc_open & AUOPEN_WRITE) != 0);
|
|
r->open = ((sc->sc_open & AUOPEN_READ) != 0);
|
|
|
|
p->samples = sc->pr.au_stamp - sc->pr.cb_pdrops;
|
|
r->samples = sc->rr.au_stamp - sc->rr.cb_pdrops;
|
|
|
|
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 & (1<<AUMODE_PLAY))?AUMODE_PLAY:AUMODE_RECORD;
|
|
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Mixer driver
|
|
*/
|
|
int
|
|
mixer_open(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc;
|
|
struct audio_hw_if *hw;
|
|
|
|
if (unit >= NAUDIO || !audio_softc[unit]) {
|
|
DPRINTF(("mixeropen: invalid device unit - %d\n", unit));
|
|
return (ENODEV);
|
|
}
|
|
|
|
sc = audio_softc[unit];
|
|
hw = sc->hw_if;
|
|
|
|
DPRINTF(("mixer: open(%x): flags=0x%x sc=0x%x\n", dev, flags, sc));
|
|
if (hw == 0) /* Hardware has not attached to us... */
|
|
return (ENXIO);
|
|
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Close a mixer device
|
|
*/
|
|
/* ARGSUSED */
|
|
int
|
|
mixer_close(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
DPRINTF(("mixer: close; unit %d\n", AUDIOUNIT(dev)));
|
|
|
|
return (0);
|
|
}
|
|
|
|
int
|
|
mixer_ioctl(dev, cmd, addr, flag, p)
|
|
dev_t dev;
|
|
int cmd;
|
|
caddr_t addr;
|
|
int flag;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_softc[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int error = EINVAL;
|
|
|
|
DPRINTF(("mixer: ioctl(%d,'%c',%d)\n",
|
|
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
|
|
|
|
switch (cmd) {
|
|
case AUDIO_GETDEV:
|
|
DPRINTF(("AUDIO_GETDEV\n"));
|
|
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_DEVINFO:
|
|
DPRINTF(("AUDIO_MIXER_DEVINFO\n"));
|
|
error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_READ:
|
|
DPRINTF(("AUDIO_MIXER_READ\n"));
|
|
error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_WRITE:
|
|
DPRINTF(("AUDIO_MIXER_WRITE\n"));
|
|
error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
|
|
if (error == 0)
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
break;
|
|
|
|
default:
|
|
error = EINVAL;
|
|
break;
|
|
}
|
|
DPRINTF(("mixer: ioctl(%d,'%c',%d) result %d\n",
|
|
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
|
|
return (error);
|
|
}
|
|
#endif
|