6292c7eda0
(but not necessarily MIDI) is defined. Fixes PR port-i386/6615 (Krister Walfridsson <cato@ulysses.df.lth.se>). (XXX this should probably go into a separate file to get rid of the #ifdef mess)
2839 lines
67 KiB
C
2839 lines
67 KiB
C
/* $NetBSD: audio.c,v 1.108 1998/12/20 14:26:44 drochner 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 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, poll, signals,
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* and silence fill.
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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/poll.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/kernel.h>
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#include <sys/signalvar.h>
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#include <sys/conf.h>
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#include <sys/audioio.h>
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#include <sys/device.h>
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#include <dev/audio_if.h>
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#include <dev/audiovar.h>
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#include <vm/vm.h>
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#include <vm/vm_prot.h>
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#include <machine/endian.h>
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#ifdef AUDIO_DEBUG
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#define DPRINTF(x) if (audiodebug) printf x
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#define DPRINTFN(n,x) if (audiodebug>(n)) printf x
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int audiodebug = 0;
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#else
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#define DPRINTF(x)
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#define DPRINTFN(n,x)
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#endif
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#define ROUNDSIZE(x) x &= -16 /* round to nice boundary */
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int audio_blk_ms = AUDIO_BLK_MS;
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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_poll __P((dev_t, int, struct proc *));
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int audio_mmap __P((dev_t, int, int));
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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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static void mixer_remove __P((struct audio_softc *, struct proc *p));
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static void mixer_signal __P((struct audio_softc *));
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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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int audiostartr __P((struct audio_softc *));
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int audiostartp __P((struct audio_softc *));
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void audio_rint __P((void *));
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void audio_pint __P((void *));
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int audio_check_params __P((struct audio_params *));
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void audio_calc_blksize __P((struct audio_softc *, int));
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void audio_fill_silence __P((struct audio_params *, u_char *, int));
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int audio_silence_copyout __P((struct audio_softc *, int, struct uio *));
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void audio_init_ringbuffer __P((struct audio_ringbuffer *));
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int audio_initbufs __P((struct audio_softc *));
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void audio_calcwater __P((struct audio_softc *));
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static __inline int audio_sleep_timo __P((int *, char *, int));
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static __inline int audio_sleep __P((int *, char *));
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static __inline void audio_wakeup __P((int *));
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int audio_drain __P((struct audio_softc *));
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void audio_clear __P((struct audio_softc *));
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static __inline void audio_pint_silence __P((struct audio_softc *, struct audio_ringbuffer *, u_char *, int));
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int audio_alloc_ring __P((struct audio_softc *, struct audio_ringbuffer *, int));
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void audio_free_ring __P((struct audio_softc *, struct audio_ringbuffer *));
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int audioprobe __P((struct device *, struct cfdata *, void *));
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void audioattach __P((struct device *, struct device *, void *));
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struct portname {
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char *name;
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int mask;
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};
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static struct portname itable[] = {
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{ AudioNmicrophone, AUDIO_MICROPHONE },
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{ AudioNline, AUDIO_LINE_IN },
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{ AudioNcd, AUDIO_CD },
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{ 0 }
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};
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static struct portname otable[] = {
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{ AudioNspeaker, AUDIO_SPEAKER },
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{ AudioNheadphone, AUDIO_HEADPHONE },
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{ AudioNline, AUDIO_LINE_OUT },
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{ 0 }
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};
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void au_check_ports __P((struct audio_softc *, struct au_mixer_ports *,
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mixer_devinfo_t *, int, char *, char *,
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struct portname *));
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int au_set_gain __P((struct audio_softc *, struct au_mixer_ports *,
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int, int));
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void au_get_gain __P((struct audio_softc *, struct au_mixer_ports *,
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u_int *, u_char *));
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int au_set_port __P((struct audio_softc *, struct au_mixer_ports *,
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u_int));
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int au_get_port __P((struct audio_softc *, struct au_mixer_ports *));
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int au_get_lr_value __P((struct audio_softc *, mixer_ctrl_t *,
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int *, int *r));
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int au_set_lr_value __P((struct audio_softc *, mixer_ctrl_t *,
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int, int));
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int au_portof __P((struct audio_softc *, char *));
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/* The default audio mode: 8 kHz mono ulaw */
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struct audio_params audio_default =
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{ 8000, AUDIO_ENCODING_ULAW, 8, 1, 0, 1 };
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struct cfattach audio_ca = {
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sizeof(struct audio_softc), audioprobe, audioattach
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};
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extern struct cfdriver audio_cd;
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int
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audioprobe(parent, match, aux)
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struct device *parent;
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struct cfdata *match;
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void *aux;
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{
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struct audio_attach_args *sa = aux;
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DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n",
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sa->type, sa, sa->hwif));
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return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0;
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}
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void
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audioattach(parent, self, aux)
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struct device *parent, *self;
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void *aux;
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{
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struct audio_softc *sc = (void *)self;
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struct audio_attach_args *sa = aux;
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struct audio_hw_if *hwp = sa->hwif;
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void *hdlp = sa->hdl;
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int error;
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mixer_devinfo_t mi;
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int iclass, oclass;
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printf("\n");
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#ifdef DIAGNOSTIC
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if (hwp == 0 ||
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hwp->open == 0 ||
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hwp->close == 0 ||
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hwp->query_encoding == 0 ||
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hwp->set_params == 0 ||
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(hwp->start_output == 0 && hwp->trigger_output == 0) ||
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(hwp->start_input == 0 && hwp->trigger_input == 0) ||
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hwp->halt_output == 0 ||
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hwp->halt_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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hwp->get_props == 0) {
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printf("audio: missing method\n");
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sc->hw_if = 0;
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return;
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}
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#endif
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sc->hw_if = hwp;
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sc->hw_hdl = hdlp;
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sc->sc_dev = parent;
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error = audio_alloc_ring(sc, &sc->sc_pr, AU_RING_SIZE);
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if (error) {
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sc->hw_if = 0;
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printf("audio: could not allocate play buffer\n");
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return;
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}
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error = audio_alloc_ring(sc, &sc->sc_rr, AU_RING_SIZE);
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if (error) {
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audio_free_ring(sc, &sc->sc_pr);
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sc->hw_if = 0;
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printf("audio: could not allocate record buffer\n");
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return;
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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_pparams = audio_default;
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sc->sc_rparams = audio_default;
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/* Set up some default values */
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sc->sc_blkset = 0;
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audio_calc_blksize(sc, AUMODE_RECORD);
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audio_calc_blksize(sc, AUMODE_PLAY);
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audio_init_ringbuffer(&sc->sc_rr);
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audio_init_ringbuffer(&sc->sc_pr);
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audio_calcwater(sc);
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iclass = oclass = -1;
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sc->sc_inports.index = -1;
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sc->sc_inports.nports = 0;
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sc->sc_inports.isenum = 0;
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sc->sc_inports.allports = 0;
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sc->sc_outports.index = -1;
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sc->sc_outports.nports = 0;
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sc->sc_outports.isenum = 0;
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sc->sc_outports.allports = 0;
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sc->sc_monitor_port = -1;
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for(mi.index = 0; ; mi.index++) {
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if (hwp->query_devinfo(hdlp, &mi) != 0)
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break;
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if (mi.type == AUDIO_MIXER_CLASS &&
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strcmp(mi.label.name, AudioCrecord) == 0)
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iclass = mi.index;
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if (mi.type == AUDIO_MIXER_CLASS &&
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strcmp(mi.label.name, AudioCmonitor) == 0)
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oclass = mi.index;
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}
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for(mi.index = 0; ; mi.index++) {
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if (hwp->query_devinfo(hdlp, &mi) != 0)
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break;
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if (mi.type == AUDIO_MIXER_CLASS)
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continue;
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au_check_ports(sc, &sc->sc_inports, &mi, iclass,
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AudioNsource, AudioNrecord, itable);
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au_check_ports(sc, &sc->sc_outports, &mi, oclass,
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AudioNoutput, AudioNmaster, otable);
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if (mi.mixer_class == oclass &&
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strcmp(mi.label.name, AudioNmonitor))
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sc->sc_monitor_port = mi.index;
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}
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DPRINTF(("audio_attach: inputs ports=0x%x, output ports=0x%x\n",
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sc->sc_inports.allports, sc->sc_outports.allports));
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}
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int
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au_portof(sc, name)
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struct audio_softc *sc;
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char *name;
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{
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mixer_devinfo_t mi;
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for(mi.index = 0;
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sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0;
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mi.index++)
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if (strcmp(mi.label.name, name) == 0)
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return mi.index;
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return -1;
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}
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void
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au_check_ports(sc, ports, mi, cls, name, mname, tbl)
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struct audio_softc *sc;
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struct au_mixer_ports *ports;
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mixer_devinfo_t *mi;
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int cls;
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char *name;
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char *mname;
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struct portname *tbl;
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{
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int i, j;
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if (mi->mixer_class != cls)
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return;
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if (strcmp(mi->label.name, mname) == 0) {
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ports->master = mi->index;
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return;
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}
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if (strcmp(mi->label.name, name) != 0)
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return;
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if (mi->type == AUDIO_MIXER_ENUM) {
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ports->index = mi->index;
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for(i = 0; tbl[i].name; i++) {
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for(j = 0; j < mi->un.e.num_mem; j++) {
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if (strcmp(mi->un.e.member[j].label.name,
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tbl[i].name) == 0) {
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ports->aumask[ports->nports] = tbl[i].mask;
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ports->misel [ports->nports] = mi->un.e.member[j].ord;
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ports->miport[ports->nports++] =
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au_portof(sc, mi->un.e.member[j].label.name);
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ports->allports |= tbl[i].mask;
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}
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}
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}
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ports->isenum = 1;
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} else if (mi->type == AUDIO_MIXER_SET) {
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ports->index = mi->index;
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for(i = 0; tbl[i].name; i++) {
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for(j = 0; j < mi->un.s.num_mem; j++) {
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if (strcmp(mi->un.s.member[j].label.name,
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tbl[i].name) == 0) {
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ports->aumask[ports->nports] = tbl[i].mask;
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ports->misel [ports->nports] = mi->un.s.member[j].mask;
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ports->miport[ports->nports++] =
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au_portof(sc, mi->un.s.member[j].label.name);
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ports->allports |= tbl[i].mask;
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}
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}
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}
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}
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}
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|
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/*
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* Called from hardware driver. This is where the MI audio driver gets
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* probed/attached to the hardware driver.
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*/
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void
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audio_attach_mi(ahwp, hdlp, dev)
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struct audio_hw_if *ahwp;
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void *hdlp;
|
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struct device *dev;
|
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{
|
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struct audio_attach_args arg;
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|
|
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#ifdef DIAGNOSTIC
|
|
if (ahwp == NULL) {
|
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printf("audio_attach_mi: NULL\n");
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return;
|
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}
|
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#endif
|
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arg.type = AUDIODEV_TYPE_AUDIO;
|
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arg.hwif = ahwp;
|
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arg.hdl = hdlp;
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(void)config_found(dev, &arg, audioprint);
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}
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|
|
#ifdef AUDIO_DEBUG
|
|
void audio_printsc __P((struct audio_softc *));
|
|
void audio_print_params __P((char *, struct audio_params *));
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|
|
|
void
|
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audio_printsc(sc)
|
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struct audio_softc *sc;
|
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{
|
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printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
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printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode);
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printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan);
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printf("rring used 0x%x pring used=%d\n", sc->sc_rr.used, sc->sc_pr.used);
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printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus);
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printf("blksize %d", sc->sc_pr.blksize);
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printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow);
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}
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|
|
void
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audio_print_params(s, p)
|
|
char *s;
|
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struct audio_params *p;
|
|
{
|
|
printf("audio: %s sr=%ld, enc=%d, chan=%d, prec=%d\n", s,
|
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p->sample_rate, p->encoding, p->channels, p->precision);
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}
|
|
#endif
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|
|
|
int
|
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audio_alloc_ring(sc, r, bufsize)
|
|
struct audio_softc *sc;
|
|
struct audio_ringbuffer *r;
|
|
int bufsize;
|
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{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
void *hdl = sc->hw_hdl;
|
|
/*
|
|
* Alloc DMA play and record buffers
|
|
*/
|
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ROUNDSIZE(bufsize);
|
|
if (bufsize < AUMINBUF)
|
|
bufsize = AUMINBUF;
|
|
if (hw->round_buffersize)
|
|
bufsize = hw->round_buffersize(hdl, bufsize);
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r->bufsize = bufsize;
|
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if (hw->allocm)
|
|
r->start = hw->allocm(hdl, r->bufsize, M_DEVBUF, M_WAITOK);
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else
|
|
r->start = malloc(bufsize, M_DEVBUF, M_WAITOK);
|
|
if (r->start == 0)
|
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return ENOMEM;
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
audio_free_ring(sc, r)
|
|
struct audio_softc *sc;
|
|
struct audio_ringbuffer *r;
|
|
{
|
|
if (sc->hw_if->freem) {
|
|
sc->hw_if->freem(sc->hw_hdl, r->start, M_DEVBUF);
|
|
} else {
|
|
free(r->start, M_DEVBUF);
|
|
}
|
|
}
|
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|
|
int
|
|
audioopen(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
case AUDIOCTL_DEVICE:
|
|
return (audio_open(dev, flags, ifmt, p));
|
|
case MIXER_DEVICE:
|
|
return (mixer_open(dev, flags, ifmt, p));
|
|
default:
|
|
return (ENXIO);
|
|
}
|
|
}
|
|
|
|
int
|
|
audioclose(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
return (audio_close(dev, flags, ifmt, p));
|
|
case MIXER_DEVICE:
|
|
return (mixer_close(dev, flags, ifmt, p));
|
|
case AUDIOCTL_DEVICE:
|
|
return 0;
|
|
default:
|
|
return (ENXIO);
|
|
}
|
|
}
|
|
|
|
int
|
|
audioread(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
return (audio_read(dev, uio, ioflag));
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
return (ENODEV);
|
|
default:
|
|
return (ENXIO);
|
|
}
|
|
}
|
|
|
|
int
|
|
audiowrite(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
return (audio_write(dev, uio, ioflag));
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
return (ENODEV);
|
|
default:
|
|
return (ENXIO);
|
|
}
|
|
}
|
|
|
|
int
|
|
audioioctl(dev, cmd, addr, flag, p)
|
|
dev_t dev;
|
|
u_long cmd;
|
|
caddr_t addr;
|
|
int flag;
|
|
struct proc *p;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
case AUDIOCTL_DEVICE:
|
|
return (audio_ioctl(dev, cmd, addr, flag, p));
|
|
case MIXER_DEVICE:
|
|
return (mixer_ioctl(dev, cmd, addr, flag, p));
|
|
default:
|
|
return (ENXIO);
|
|
}
|
|
}
|
|
|
|
int
|
|
audiopoll(dev, events, p)
|
|
dev_t dev;
|
|
int events;
|
|
struct proc *p;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
return (audio_poll(dev, events, p));
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
return (0);
|
|
default:
|
|
return (0);
|
|
}
|
|
}
|
|
|
|
int
|
|
audiommap(dev, off, prot)
|
|
dev_t dev;
|
|
int off, prot;
|
|
{
|
|
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
return (audio_mmap(dev, off, prot));
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
return -1;
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Audio driver
|
|
*/
|
|
void
|
|
audio_init_ringbuffer(rp)
|
|
struct audio_ringbuffer *rp;
|
|
{
|
|
int nblks;
|
|
int blksize = rp->blksize;
|
|
|
|
if (blksize < AUMINBLK)
|
|
blksize = AUMINBLK;
|
|
nblks = rp->bufsize / blksize;
|
|
if (nblks < AUMINNOBLK) {
|
|
nblks = AUMINNOBLK;
|
|
blksize = rp->bufsize / nblks;
|
|
ROUNDSIZE(blksize);
|
|
}
|
|
DPRINTF(("audio_init_ringbuffer: blksize=%d\n", blksize));
|
|
rp->blksize = blksize;
|
|
rp->maxblks = nblks;
|
|
rp->used = 0;
|
|
rp->end = rp->start + nblks * blksize;
|
|
rp->inp = rp->outp = rp->start;
|
|
rp->stamp = 0;
|
|
rp->drops = 0;
|
|
rp->pause = 0;
|
|
rp->copying = 0;
|
|
rp->needfill = 0;
|
|
rp->mmapped = 0;
|
|
}
|
|
|
|
int
|
|
audio_initbufs(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int error;
|
|
|
|
DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
|
|
audio_init_ringbuffer(&sc->sc_rr);
|
|
if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) {
|
|
error = hw->init_input(sc->hw_hdl, sc->sc_rr.start,
|
|
sc->sc_rr.end - sc->sc_rr.start);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
audio_init_ringbuffer(&sc->sc_pr);
|
|
sc->sc_sil_count = 0;
|
|
if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) {
|
|
error = hw->init_output(sc->hw_hdl, sc->sc_pr.start,
|
|
sc->sc_pr.end - sc->sc_pr.start);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
#define double u_long
|
|
sc->sc_pnintr = 0;
|
|
sc->sc_pblktime = (u_long)(
|
|
(double)sc->sc_pr.blksize * 100000 /
|
|
(double)(sc->sc_pparams.precision / NBBY *
|
|
sc->sc_pparams.channels *
|
|
sc->sc_pparams.sample_rate)) * 10;
|
|
DPRINTF(("audio: play blktime = %lu for %d\n",
|
|
sc->sc_pblktime, sc->sc_pr.blksize));
|
|
sc->sc_rnintr = 0;
|
|
sc->sc_rblktime = (u_long)(
|
|
(double)sc->sc_rr.blksize * 100000 /
|
|
(double)(sc->sc_rparams.precision / NBBY *
|
|
sc->sc_rparams.channels *
|
|
sc->sc_rparams.sample_rate)) * 10;
|
|
DPRINTF(("audio: record blktime = %lu for %d\n",
|
|
sc->sc_rblktime, sc->sc_rr.blksize));
|
|
#undef double
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
audio_calcwater(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
sc->sc_pr.usedhigh = sc->sc_pr.end - sc->sc_pr.start;
|
|
sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4; /* set lowater at 75% */
|
|
if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh)
|
|
sc->sc_pr.usedlow -= sc->sc_pr.blksize;
|
|
sc->sc_rr.usedhigh = sc->sc_pr.end - sc->sc_pr.start - sc->sc_pr.blksize;
|
|
sc->sc_rr.usedlow = 0;
|
|
}
|
|
|
|
static __inline int
|
|
audio_sleep_timo(chan, label, timo)
|
|
int *chan;
|
|
char *label;
|
|
int timo;
|
|
{
|
|
int st;
|
|
|
|
if (!label)
|
|
label = "audio";
|
|
|
|
DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n",
|
|
chan, label, timo));
|
|
*chan = 1;
|
|
st = tsleep(chan, PWAIT | PCATCH, label, timo);
|
|
*chan = 0;
|
|
#ifdef AUDIO_DEBUG
|
|
if (st != 0)
|
|
printf("audio_sleep: woke up st=%d\n", st);
|
|
#endif
|
|
return (st);
|
|
}
|
|
|
|
static __inline int
|
|
audio_sleep(chan, label)
|
|
int *chan;
|
|
char *label;
|
|
{
|
|
return audio_sleep_timo(chan, label, 0);
|
|
}
|
|
|
|
/* call at splaudio() */
|
|
static __inline void
|
|
audio_wakeup(chan)
|
|
int *chan;
|
|
{
|
|
DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan));
|
|
if (*chan) {
|
|
wakeup(chan);
|
|
*chan = 0;
|
|
}
|
|
}
|
|
|
|
int
|
|
audio_open(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc;
|
|
int error;
|
|
int mode;
|
|
struct audio_hw_if *hw;
|
|
struct audio_info ai;
|
|
|
|
if (unit >= audio_cd.cd_ndevs ||
|
|
(sc = audio_cd.cd_devs[unit]) == NULL)
|
|
return ENXIO;
|
|
|
|
hw = sc->hw_if;
|
|
if (!hw)
|
|
return ENXIO;
|
|
|
|
DPRINTF(("audio_open: dev=0x%x flags=0x%x sc=%p hdl=%p\n", dev, flags, sc, sc->hw_hdl));
|
|
|
|
if (ISDEVAUDIOCTL(dev))
|
|
return 0;
|
|
|
|
if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0)
|
|
return (EBUSY);
|
|
|
|
error = hw->open(sc->hw_hdl, flags);
|
|
if (error)
|
|
return (error);
|
|
|
|
sc->sc_async_audio = 0;
|
|
sc->sc_rchan = 0;
|
|
sc->sc_wchan = 0;
|
|
sc->sc_blkset = 0; /* Block sizes not set yet */
|
|
sc->sc_sil_count = 0;
|
|
sc->sc_rbus = 0;
|
|
sc->sc_pbus = 0;
|
|
sc->sc_eof = 0;
|
|
sc->sc_playdrop = 0;
|
|
|
|
sc->sc_full_duplex = 0;
|
|
/* doesn't always work right on SB.
|
|
(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
|
|
(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX);
|
|
*/
|
|
|
|
mode = 0;
|
|
if (flags & FREAD) {
|
|
sc->sc_open |= AUOPEN_READ;
|
|
mode |= AUMODE_RECORD;
|
|
}
|
|
if (flags & FWRITE) {
|
|
sc->sc_open |= AUOPEN_WRITE;
|
|
mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
|
|
}
|
|
|
|
/*
|
|
* Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
|
|
* The /dev/audio is always (re)set to 8-bit MU-Law mono
|
|
* For the other devices, you get what they were last set to.
|
|
*/
|
|
if (ISDEVAUDIO(dev)) {
|
|
/* /dev/audio */
|
|
sc->sc_rparams = audio_default;
|
|
sc->sc_pparams = audio_default;
|
|
}
|
|
#ifdef DIAGNOSTIC
|
|
/*
|
|
* Sample rate and precision are supposed to be set to proper
|
|
* default values by the hardware driver, so that it may give
|
|
* us these values.
|
|
*/
|
|
if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) {
|
|
printf("audio_open: 0 precision\n");
|
|
return EINVAL;
|
|
}
|
|
#endif
|
|
|
|
AUDIO_INITINFO(&ai);
|
|
ai.record.sample_rate = sc->sc_rparams.sample_rate;
|
|
ai.record.encoding = sc->sc_rparams.encoding;
|
|
ai.record.channels = sc->sc_rparams.channels;
|
|
ai.record.precision = sc->sc_rparams.precision;
|
|
ai.play.sample_rate = sc->sc_pparams.sample_rate;
|
|
ai.play.encoding = sc->sc_pparams.encoding;
|
|
ai.play.channels = sc->sc_pparams.channels;
|
|
ai.play.precision = sc->sc_pparams.precision;
|
|
ai.mode = mode;
|
|
sc->sc_pr.blksize = sc->sc_rr.blksize = 0; /* force recalculation */
|
|
error = audiosetinfo(sc, &ai);
|
|
if (error)
|
|
goto bad;
|
|
|
|
DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode));
|
|
|
|
return 0;
|
|
|
|
bad:
|
|
hw->close(sc->hw_hdl);
|
|
sc->sc_open = 0;
|
|
sc->sc_mode = 0;
|
|
sc->sc_full_duplex = 0;
|
|
return error;
|
|
}
|
|
|
|
/*
|
|
* Must be called from task context.
|
|
*/
|
|
void
|
|
audio_init_record(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int s = splaudio();
|
|
|
|
if (sc->hw_if->speaker_ctl &&
|
|
(!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0))
|
|
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_wstamp = sc->sc_pr.stamp;
|
|
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, drops;
|
|
struct audio_ringbuffer *cb = &sc->sc_pr;
|
|
int s;
|
|
|
|
DPRINTF(("audio_drain: enter busy=%d used=%d\n",
|
|
sc->sc_pbus, sc->sc_pr.used));
|
|
if (sc->sc_pr.mmapped || sc->sc_pr.used <= 0)
|
|
return 0;
|
|
if (!sc->sc_pbus) {
|
|
/* We've never started playing, probably because the
|
|
* block was too short. Pad it and start now.
|
|
*/
|
|
int cc;
|
|
u_char *inp = cb->inp;
|
|
|
|
cc = cb->blksize - (inp - cb->start) % cb->blksize;
|
|
audio_fill_silence(&sc->sc_pparams, inp, cc);
|
|
inp += cc;
|
|
if (inp >= cb->end)
|
|
inp = cb->start;
|
|
s = splaudio();
|
|
cb->used += cc;
|
|
cb->inp = inp;
|
|
error = audiostartp(sc);
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
/*
|
|
* Play until a silence block has been played, then we
|
|
* know all has been drained.
|
|
* XXX This should be done some other way to avoid
|
|
* playing silence.
|
|
*/
|
|
#ifdef DIAGNOSTIC
|
|
if (cb->copying) {
|
|
printf("audio_drain: copying in progress!?!\n");
|
|
cb->copying = 0;
|
|
}
|
|
#endif
|
|
drops = cb->drops;
|
|
error = 0;
|
|
s = splaudio();
|
|
while (cb->drops == drops && !error) {
|
|
DPRINTF(("audio_drain: used=%d, drops=%ld\n", sc->sc_pr.used, cb->drops));
|
|
/*
|
|
* When the process is exiting, it ignores all signals and
|
|
* we can't interrupt this sleep, so we set a timeout just in case.
|
|
*/
|
|
error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz);
|
|
}
|
|
splx(s);
|
|
return error;
|
|
}
|
|
|
|
/*
|
|
* 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_cd.cd_devs[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
int s;
|
|
|
|
DPRINTF(("audio_close: unit=%d\n", unit));
|
|
|
|
s = splaudio();
|
|
/* Stop recording. */
|
|
if ((flags & FREAD) && sc->sc_rbus) {
|
|
/*
|
|
* XXX Some drivers (e.g. SB) use the same routine
|
|
* to halt input and output so don't halt input if
|
|
* in full duplex mode. These drivers should be fixed.
|
|
*/
|
|
if (!sc->sc_full_duplex || sc->hw_if->halt_input != sc->hw_if->halt_output)
|
|
sc->hw_if->halt_input(sc->hw_hdl);
|
|
sc->sc_rbus = 0;
|
|
}
|
|
/*
|
|
* Block until output drains, but allow ^C interrupt.
|
|
*/
|
|
sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */
|
|
/*
|
|
* If there is pending output, let it drain (unless
|
|
* the output is paused).
|
|
*/
|
|
if ((flags & FWRITE) && sc->sc_pbus) {
|
|
if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain)
|
|
(void)hw->drain(sc->hw_hdl);
|
|
sc->hw_if->halt_output(sc->hw_hdl);
|
|
sc->sc_pbus = 0;
|
|
}
|
|
|
|
hw->close(sc->hw_hdl);
|
|
|
|
if (flags & FREAD) {
|
|
sc->sc_open &= ~AUOPEN_READ;
|
|
sc->sc_mode &= ~AUMODE_RECORD;
|
|
}
|
|
if (flags & FWRITE) {
|
|
sc->sc_open &= ~AUOPEN_WRITE;
|
|
sc->sc_mode &= ~(AUMODE_PLAY|AUMODE_PLAY_ALL);
|
|
}
|
|
|
|
sc->sc_async_audio = 0;
|
|
sc->sc_full_duplex = 0;
|
|
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_cd.cd_devs[unit];
|
|
struct audio_ringbuffer *cb = &sc->sc_rr;
|
|
u_char *outp;
|
|
int error, s, used, cc, n;
|
|
|
|
if (cb->mmapped)
|
|
return EINVAL;
|
|
|
|
DPRINTFN(1,("audio_read: cc=%d mode=%d\n",
|
|
uio->uio_resid, sc->sc_mode));
|
|
|
|
error = 0;
|
|
/*
|
|
* If hardware is half-duplex and currently playing, return
|
|
* silence blocks based on the number of blocks we have output.
|
|
*/
|
|
if (!sc->sc_full_duplex &&
|
|
(sc->sc_mode & AUMODE_PLAY)) {
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
for(;;) {
|
|
cc = sc->sc_pr.stamp - sc->sc_wstamp;
|
|
if (cc > 0)
|
|
break;
|
|
DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n",
|
|
sc->sc_pr.stamp, sc->sc_wstamp));
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return EWOULDBLOCK;
|
|
}
|
|
error = audio_sleep(&sc->sc_rchan, "aud_hr");
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
splx(s);
|
|
|
|
if (uio->uio_resid < cc)
|
|
cc = uio->uio_resid;
|
|
DPRINTFN(1, ("audio_read: reading in write mode, cc=%d\n", cc));
|
|
error = audio_silence_copyout(sc, cc, uio);
|
|
sc->sc_wstamp += cc;
|
|
}
|
|
return (error);
|
|
}
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
while (cb->used <= 0) {
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return EWOULDBLOCK;
|
|
}
|
|
if (!sc->sc_rbus) {
|
|
error = audiostartr(sc);
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
DPRINTFN(2, ("audio_read: sleep used=%d\n", cb->used));
|
|
error = audio_sleep(&sc->sc_rchan, "aud_rd");
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
used = cb->used;
|
|
outp = cb->outp;
|
|
cb->copying = 1;
|
|
splx(s);
|
|
cc = used - cb->usedlow; /* maximum to read */
|
|
n = cb->end - outp;
|
|
if (n < cc)
|
|
cc = n; /* don't read beyond end of buffer */
|
|
|
|
if (uio->uio_resid < cc)
|
|
cc = uio->uio_resid; /* and no more than we want */
|
|
|
|
if (sc->sc_rparams.sw_code)
|
|
sc->sc_rparams.sw_code(sc->hw_hdl, outp, cc);
|
|
DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc));
|
|
error = uiomove(outp, cc, uio);
|
|
used -= cc;
|
|
outp += cc;
|
|
if (outp >= cb->end)
|
|
outp = cb->start;
|
|
s = splaudio();
|
|
cb->outp = outp;
|
|
cb->used = used;
|
|
cb->copying = 0;
|
|
splx(s);
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
void
|
|
audio_clear(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int s = splaudio();
|
|
|
|
if (sc->sc_rbus) {
|
|
audio_wakeup(&sc->sc_rchan);
|
|
sc->hw_if->halt_input(sc->hw_hdl);
|
|
sc->sc_rbus = 0;
|
|
}
|
|
if (sc->sc_pbus) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
sc->hw_if->halt_output(sc->hw_hdl);
|
|
sc->sc_pbus = 0;
|
|
}
|
|
splx(s);
|
|
}
|
|
|
|
void
|
|
audio_calc_blksize(sc, mode)
|
|
struct audio_softc *sc;
|
|
int mode;
|
|
{
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_params *parm;
|
|
struct audio_ringbuffer *rb;
|
|
int bs;
|
|
|
|
if (sc->sc_blkset)
|
|
return;
|
|
|
|
if (mode == AUMODE_PLAY) {
|
|
parm = &sc->sc_pparams;
|
|
rb = &sc->sc_pr;
|
|
} else {
|
|
parm = &sc->sc_rparams;
|
|
rb = &sc->sc_rr;
|
|
}
|
|
|
|
bs = parm->sample_rate * audio_blk_ms / 1000 *
|
|
parm->channels * parm->precision / NBBY *
|
|
parm->factor;
|
|
ROUNDSIZE(bs);
|
|
if (hw->round_blocksize)
|
|
bs = hw->round_blocksize(sc->hw_hdl, bs);
|
|
rb->blksize = bs;
|
|
|
|
DPRINTF(("audio_calc_blksize: %s blksize=%d\n",
|
|
mode == AUMODE_PLAY ? "play" : "record", bs));
|
|
}
|
|
|
|
void
|
|
audio_fill_silence(params, p, n)
|
|
struct audio_params *params;
|
|
u_char *p;
|
|
int n;
|
|
{
|
|
u_char auzero0, auzero1 = 0; /* initialize to please gcc */
|
|
int nfill = 1;
|
|
|
|
switch (params->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
auzero0 = 0x7f;
|
|
break;
|
|
case AUDIO_ENCODING_ALAW:
|
|
auzero0 = 0x55;
|
|
break;
|
|
case AUDIO_ENCODING_MPEG_L1_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L1_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
|
|
case AUDIO_ENCODING_MPEG_L2_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L2_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
|
|
case AUDIO_ENCODING_ADPCM: /* is this right XXX */
|
|
case AUDIO_ENCODING_SLINEAR_LE:
|
|
case AUDIO_ENCODING_SLINEAR_BE:
|
|
auzero0 = 0; /* fortunately this works for both 8 and 16 bits */
|
|
break;
|
|
case AUDIO_ENCODING_ULINEAR_LE:
|
|
case AUDIO_ENCODING_ULINEAR_BE:
|
|
if (params->precision == 16) {
|
|
nfill = 2;
|
|
if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) {
|
|
auzero0 = 0;
|
|
auzero1 = 0x80;
|
|
} else {
|
|
auzero0 = 0x80;
|
|
auzero1 = 0;
|
|
}
|
|
} else
|
|
auzero0 = 0x80;
|
|
break;
|
|
default:
|
|
DPRINTF(("audio: bad encoding %d\n", params->encoding));
|
|
auzero0 = 0;
|
|
break;
|
|
}
|
|
if (nfill == 1) {
|
|
while (--n >= 0)
|
|
*p++ = auzero0; /* XXX memset */
|
|
} else /* nfill must be 2 */ {
|
|
while (n > 1) {
|
|
*p++ = auzero0;
|
|
*p++ = auzero1;
|
|
n -= 2;
|
|
}
|
|
}
|
|
}
|
|
|
|
int
|
|
audio_silence_copyout(sc, n, uio)
|
|
struct audio_softc *sc;
|
|
int n;
|
|
struct uio *uio;
|
|
{
|
|
int error;
|
|
int k;
|
|
u_char zerobuf[128];
|
|
|
|
audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf);
|
|
|
|
error = 0;
|
|
while (n > 0 && uio->uio_resid > 0 && !error) {
|
|
k = min(n, min(uio->uio_resid, sizeof zerobuf));
|
|
error = uiomove(zerobuf, k, uio);
|
|
n -= k;
|
|
}
|
|
return (error);
|
|
}
|
|
|
|
int
|
|
audio_write(dev, uio, ioflag)
|
|
dev_t dev;
|
|
struct uio *uio;
|
|
int ioflag;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_cd.cd_devs[unit];
|
|
struct audio_ringbuffer *cb = &sc->sc_pr;
|
|
u_char *inp, *einp;
|
|
int error, s, n, cc, used;
|
|
|
|
DPRINTFN(2, ("audio_write: sc=%p(unit=%d) count=%d used=%d(hi=%d)\n", sc, unit,
|
|
uio->uio_resid, sc->sc_pr.used, sc->sc_pr.usedhigh));
|
|
|
|
if (cb->mmapped)
|
|
return EINVAL;
|
|
|
|
if (uio->uio_resid == 0) {
|
|
sc->sc_eof++;
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* If half-duplex and currently recording, throw away data.
|
|
*/
|
|
if (!sc->sc_full_duplex &&
|
|
(sc->sc_mode & AUMODE_RECORD)) {
|
|
uio->uio_offset += uio->uio_resid;
|
|
uio->uio_resid = 0;
|
|
DPRINTF(("audio_write: half-dpx read busy\n"));
|
|
return (0);
|
|
}
|
|
|
|
if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) {
|
|
n = min(sc->sc_playdrop, uio->uio_resid);
|
|
DPRINTF(("audio_write: playdrop %d\n", n));
|
|
uio->uio_offset += n;
|
|
uio->uio_resid -= n;
|
|
sc->sc_playdrop -= n;
|
|
if (uio->uio_resid == 0)
|
|
return 0;
|
|
}
|
|
|
|
DPRINTFN(1, ("audio_write: sr=%ld, enc=%d, prec=%d, chan=%d, sw=%p, fact=%d\n",
|
|
sc->sc_pparams.sample_rate, sc->sc_pparams.encoding,
|
|
sc->sc_pparams.precision, sc->sc_pparams.channels,
|
|
sc->sc_pparams.sw_code, sc->sc_pparams.factor));
|
|
|
|
error = 0;
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
while (cb->used >= cb->usedhigh) {
|
|
DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d hiwat=%d\n",
|
|
cb->used, cb->usedlow, cb->usedhigh));
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return (EWOULDBLOCK);
|
|
}
|
|
error = audio_sleep(&sc->sc_wchan, "aud_wr");
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
used = cb->used;
|
|
inp = cb->inp;
|
|
cb->copying = 1;
|
|
splx(s);
|
|
cc = cb->usedhigh - used; /* maximum to write */
|
|
n = cb->end - inp;
|
|
if (sc->sc_pparams.factor != 1) {
|
|
/* Compensate for software coding expansion factor. */
|
|
n /= sc->sc_pparams.factor;
|
|
cc /= sc->sc_pparams.factor;
|
|
}
|
|
if (n < cc)
|
|
cc = n; /* don't write beyond end of buffer */
|
|
if (uio->uio_resid < cc)
|
|
cc = uio->uio_resid; /* and no more than we have */
|
|
|
|
#ifdef DIAGNOSTIC
|
|
/*
|
|
* This should never happen since the block size and and
|
|
* block pointers are always nicely aligned.
|
|
*/
|
|
if (cc == 0) {
|
|
printf("audio_write: cc == 0, swcode=%p, factor=%d\n",
|
|
sc->sc_pparams.sw_code, sc->sc_pparams.factor);
|
|
cb->copying = 0;
|
|
return EINVAL;
|
|
}
|
|
#endif
|
|
DPRINTFN(1, ("audio_write: uiomove cc=%d inp=%p, left=%d\n",
|
|
cc, inp, uio->uio_resid));
|
|
n = uio->uio_resid;
|
|
error = uiomove(inp, cc, uio);
|
|
cc = n - uio->uio_resid; /* number of bytes actually moved */
|
|
#ifdef AUDIO_DEBUG
|
|
if (error)
|
|
printf("audio_write:(1) uiomove failed %d; cc=%d inp=%p\n",
|
|
error, cc, inp);
|
|
#endif
|
|
/*
|
|
* Continue even if uiomove() failed because we may have
|
|
* gotten a partial block.
|
|
*/
|
|
|
|
if (sc->sc_pparams.sw_code) {
|
|
sc->sc_pparams.sw_code(sc->hw_hdl, inp, cc);
|
|
/* Adjust count after the expansion. */
|
|
cc *= sc->sc_pparams.factor;
|
|
DPRINTFN(1, ("audio_write: expanded cc=%d\n", cc));
|
|
}
|
|
|
|
einp = cb->inp + cc;
|
|
if (einp >= cb->end)
|
|
einp = cb->start;
|
|
|
|
s = splaudio();
|
|
/*
|
|
* This is a very suboptimal way of keeping track of
|
|
* silence in the buffer, but it is simple.
|
|
*/
|
|
sc->sc_sil_count = 0;
|
|
|
|
cb->inp = einp;
|
|
cb->used += cc;
|
|
/* If the interrupt routine wants the last block filled AND
|
|
* the copy did not fill the last block completely it needs to
|
|
* be padded.
|
|
*/
|
|
if (cb->needfill &&
|
|
(inp - cb->start) / cb->blksize ==
|
|
(einp - cb->start) / cb->blksize) {
|
|
/* Figure out how many bytes there is to a block boundary. */
|
|
cc = cb->blksize - (einp - cb->start) % cb->blksize;
|
|
DPRINTF(("audio_write: partial fill %d\n", cc));
|
|
} else
|
|
cc = 0;
|
|
cb->needfill = 0;
|
|
cb->copying = 0;
|
|
if (!sc->sc_pbus && !cb->pause)
|
|
error = audiostartp(sc); /* XXX should not clobber error */
|
|
splx(s);
|
|
if (cc) {
|
|
DPRINTFN(1, ("audio_write: fill %d\n", cc));
|
|
audio_fill_silence(&sc->sc_pparams, einp, cc);
|
|
}
|
|
}
|
|
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_cd.cd_devs[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_offset *ao;
|
|
int error = 0, s, offs, fd;
|
|
int rbus, pbus;
|
|
|
|
DPRINTF(("audio_ioctl(%d,'%c',%d)\n",
|
|
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
|
|
switch (cmd) {
|
|
case FIONBIO:
|
|
/* All handled in the upper FS layer. */
|
|
break;
|
|
|
|
case FIOASYNC:
|
|
if (*(int *)addr) {
|
|
if (sc->sc_async_audio)
|
|
return (EBUSY);
|
|
sc->sc_async_audio = p;
|
|
DPRINTF(("audio_ioctl: FIOASYNC %p\n", p));
|
|
} else
|
|
sc->sc_async_audio = 0;
|
|
break;
|
|
|
|
case AUDIO_FLUSH:
|
|
DPRINTF(("AUDIO_FLUSH\n"));
|
|
rbus = sc->sc_rbus;
|
|
pbus = sc->sc_pbus;
|
|
audio_clear(sc);
|
|
s = splaudio();
|
|
error = audio_initbufs(sc);
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus)
|
|
error = audiostartp(sc);
|
|
if (!error &&
|
|
(sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus)
|
|
error = audiostartr(sc);
|
|
splx(s);
|
|
break;
|
|
|
|
/*
|
|
* Number of read (write) samples dropped. We don't know where or
|
|
* when they were dropped.
|
|
*/
|
|
case AUDIO_RERROR:
|
|
*(int *)addr = sc->sc_rr.drops;
|
|
break;
|
|
|
|
case AUDIO_PERROR:
|
|
*(int *)addr = sc->sc_pr.drops;
|
|
break;
|
|
|
|
/*
|
|
* Offsets into buffer.
|
|
*/
|
|
case AUDIO_GETIOFFS:
|
|
s = splaudio();
|
|
/* figure out where next DMA will start */
|
|
ao = (struct audio_offset *)addr;
|
|
ao->samples = sc->sc_rr.stamp;
|
|
ao->deltablks = (sc->sc_rr.stamp - sc->sc_rr.stamp_last) / sc->sc_rr.blksize;
|
|
sc->sc_rr.stamp_last = sc->sc_rr.stamp;
|
|
ao->offset = sc->sc_rr.inp - sc->sc_rr.start;
|
|
splx(s);
|
|
break;
|
|
|
|
case AUDIO_GETOOFFS:
|
|
s = splaudio();
|
|
/* figure out where next DMA will start */
|
|
ao = (struct audio_offset *)addr;
|
|
offs = sc->sc_pr.outp - sc->sc_pr.start + sc->sc_pr.blksize;
|
|
if (sc->sc_pr.start + offs >= sc->sc_pr.end)
|
|
offs = 0;
|
|
ao->samples = sc->sc_pr.stamp;
|
|
ao->deltablks = (sc->sc_pr.stamp - sc->sc_pr.stamp_last) / sc->sc_pr.blksize;
|
|
sc->sc_pr.stamp_last = sc->sc_pr.stamp;
|
|
ao->offset = offs;
|
|
splx(s);
|
|
break;
|
|
|
|
/*
|
|
* How many bytes will elapse until mike hears the first
|
|
* sample of what we write next?
|
|
*/
|
|
case AUDIO_WSEEK:
|
|
*(u_long *)addr = sc->sc_rr.used;
|
|
break;
|
|
|
|
case AUDIO_SETINFO:
|
|
DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode));
|
|
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 = sc->sc_full_duplex;
|
|
break;
|
|
|
|
case AUDIO_SETFD:
|
|
DPRINTF(("AUDIO_SETFD\n"));
|
|
fd = *(int *)addr;
|
|
if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) {
|
|
if (hw->setfd)
|
|
error = hw->setfd(sc->hw_hdl, fd);
|
|
else
|
|
error = 0;
|
|
if (!error)
|
|
sc->sc_full_duplex = fd;
|
|
} else {
|
|
if (fd)
|
|
error = ENOTTY;
|
|
else
|
|
error = 0;
|
|
}
|
|
break;
|
|
|
|
case AUDIO_GETPROPS:
|
|
DPRINTF(("AUDIO_GETPROPS\n"));
|
|
*(int *)addr = hw->get_props(sc->hw_hdl);
|
|
break;
|
|
|
|
default:
|
|
DPRINTF(("audio_ioctl: 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_poll(dev, events, p)
|
|
dev_t dev;
|
|
int events;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_cd.cd_devs[unit];
|
|
int revents = 0;
|
|
int s = splaudio();
|
|
|
|
DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode));
|
|
|
|
if (events & (POLLIN | POLLRDNORM))
|
|
if ((sc->sc_mode & AUMODE_PLAY) ?
|
|
sc->sc_pr.stamp > sc->sc_wstamp :
|
|
sc->sc_rr.used > sc->sc_rr.usedlow)
|
|
revents |= events & (POLLIN | POLLRDNORM);
|
|
|
|
if (events & (POLLOUT | POLLWRNORM))
|
|
if (sc->sc_mode & AUMODE_RECORD ||
|
|
sc->sc_pr.used <= sc->sc_pr.usedlow)
|
|
revents |= events & (POLLOUT | POLLWRNORM);
|
|
|
|
if (revents == 0) {
|
|
if (events & (POLLIN | POLLRDNORM))
|
|
selrecord(p, &sc->sc_rsel);
|
|
|
|
if (events & (POLLOUT | POLLWRNORM))
|
|
selrecord(p, &sc->sc_wsel);
|
|
}
|
|
|
|
splx(s);
|
|
return (revents);
|
|
}
|
|
|
|
int
|
|
audio_mmap(dev, off, prot)
|
|
dev_t dev;
|
|
int off, prot;
|
|
{
|
|
int s;
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_cd.cd_devs[unit];
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_ringbuffer *cb;
|
|
|
|
DPRINTF(("audio_mmap: off=%d, prot=%d\n", off, prot));
|
|
|
|
if (!(hw->get_props(sc->hw_hdl) & AUDIO_PROP_MMAP) || !hw->mappage)
|
|
return -1;
|
|
#if 0
|
|
/* XXX
|
|
* The idea here was to use the protection to determine if
|
|
* we are mapping the read or write buffer, but it fails.
|
|
* The VM system is broken in (at least) two ways.
|
|
* 1) If you map memory VM_PROT_WRITE you SIGSEGV
|
|
* when writing to it, so VM_PROT_READ|VM_PROT_WRITE
|
|
* has to be used for mmapping the play buffer.
|
|
* 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE
|
|
* audio_mmap will get called at some point with VM_PROT_READ
|
|
* only.
|
|
* So, alas, we always map the play buffer for now.
|
|
*/
|
|
if (prot == (VM_PROT_READ|VM_PROT_WRITE) ||
|
|
prot == VM_PROT_WRITE)
|
|
cb = &sc->sc_pr;
|
|
else if (prot == VM_PROT_READ)
|
|
cb = &sc->sc_rr;
|
|
else
|
|
return -1;
|
|
#else
|
|
cb = &sc->sc_pr;
|
|
#endif
|
|
|
|
if ((u_int)off >= cb->bufsize)
|
|
return -1;
|
|
if (!cb->mmapped) {
|
|
cb->mmapped = 1;
|
|
if (cb == &sc->sc_pr) {
|
|
audio_fill_silence(&sc->sc_pparams, cb->start, cb->bufsize);
|
|
s = splaudio();
|
|
if (!sc->sc_pbus)
|
|
(void)audiostartp(sc);
|
|
splx(s);
|
|
} else {
|
|
s = splaudio();
|
|
if (!sc->sc_rbus)
|
|
(void)audiostartr(sc);
|
|
splx(s);
|
|
}
|
|
}
|
|
|
|
return hw->mappage(sc->hw_hdl, cb->start, off, prot);
|
|
}
|
|
|
|
int
|
|
audiostartr(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int error;
|
|
|
|
DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n",
|
|
sc->sc_rr.start, sc->sc_rr.used, sc->sc_rr.usedhigh,
|
|
sc->sc_rr.mmapped));
|
|
|
|
if (sc->hw_if->trigger_input)
|
|
error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.start,
|
|
sc->sc_rr.end, sc->sc_rr.blksize,
|
|
audio_rint, (void *)sc, &sc->sc_rparams);
|
|
else
|
|
error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.start,
|
|
sc->sc_rr.blksize, audio_rint, (void *)sc);
|
|
if (error) {
|
|
DPRINTF(("audiostartr failed: %d\n", error));
|
|
return error;
|
|
}
|
|
sc->sc_rbus = 1;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audiostartp(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
int error;
|
|
|
|
DPRINTF(("audiostartp: start=%p used=%d(hi=%d) mmapped=%d\n",
|
|
sc->sc_pr.start, sc->sc_pr.used, sc->sc_pr.usedhigh,
|
|
sc->sc_pr.mmapped));
|
|
|
|
if (!sc->sc_pr.mmapped && sc->sc_pr.used < sc->sc_pr.blksize)
|
|
return 0;
|
|
|
|
if (sc->hw_if->trigger_output)
|
|
error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.start,
|
|
sc->sc_pr.end, sc->sc_pr.blksize,
|
|
audio_pint, (void *)sc, &sc->sc_pparams);
|
|
else
|
|
error = sc->hw_if->start_output(sc->hw_hdl, sc->sc_pr.outp,
|
|
sc->sc_pr.blksize, audio_pint, (void *)sc);
|
|
if (error) {
|
|
DPRINTF(("audiostartp failed: %d\n", error));
|
|
return error;
|
|
}
|
|
sc->sc_pbus = 1;
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* When the play interrupt routine finds that the write isn't keeping
|
|
* the buffer filled it will insert silence in the buffer to make up
|
|
* for this. The part of the buffer that is filled with silence
|
|
* is kept track of in a very approximate way: it starts at sc_sil_start
|
|
* and extends sc_sil_count bytes. If there is already silence in
|
|
* the requested area nothing is done; so when the whole buffer is
|
|
* silent nothing happens. When the writer starts again sc_sil_count
|
|
* is set to 0.
|
|
*/
|
|
/* XXX
|
|
* Putting silence into the output buffer should not really be done
|
|
* at splaudio, but there is no softaudio level to do it at yet.
|
|
*/
|
|
static __inline void
|
|
audio_pint_silence(sc, cb, inp, cc)
|
|
struct audio_softc *sc;
|
|
struct audio_ringbuffer *cb;
|
|
u_char *inp;
|
|
int cc;
|
|
{
|
|
u_char *s, *e, *p, *q;
|
|
|
|
if (sc->sc_sil_count > 0) {
|
|
s = sc->sc_sil_start; /* start of silence */
|
|
e = s + sc->sc_sil_count; /* end of silence, may be beyond end */
|
|
p = inp; /* adjusted pointer to area to fill */
|
|
if (p < s)
|
|
p += cb->end - cb->start;
|
|
q = p+cc;
|
|
/* Check if there is already silence. */
|
|
if (!(s <= p && p < e &&
|
|
s <= q && q <= e)) {
|
|
if (s <= p)
|
|
sc->sc_sil_count = max(sc->sc_sil_count, q-s);
|
|
DPRINTFN(5, ("audio_pint_silence: fill cc=%d inp=%p, count=%d size=%d\n",
|
|
cc, inp, sc->sc_sil_count, (int)(cb->end - cb->start)));
|
|
audio_fill_silence(&sc->sc_pparams, inp, cc);
|
|
} else {
|
|
DPRINTFN(5, ("audio_pint_silence: already silent cc=%d inp=%p\n", cc, inp));
|
|
|
|
}
|
|
} else {
|
|
sc->sc_sil_start = inp;
|
|
sc->sc_sil_count = cc;
|
|
DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n",
|
|
inp, cc));
|
|
audio_fill_silence(&sc->sc_pparams, inp, cc);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* 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(v)
|
|
void *v;
|
|
{
|
|
struct audio_softc *sc = v;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_ringbuffer *cb = &sc->sc_pr;
|
|
u_char *inp;
|
|
int cc, ccr;
|
|
int blksize;
|
|
int error;
|
|
|
|
if (!sc->sc_open)
|
|
return; /* ignore interrupt if not open */
|
|
|
|
blksize = cb->blksize;
|
|
|
|
cb->outp += blksize;
|
|
if (cb->outp >= cb->end)
|
|
cb->outp = cb->start;
|
|
cb->stamp += blksize / sc->sc_pparams.factor;
|
|
if (cb->mmapped) {
|
|
DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n",
|
|
cb->outp, blksize, cb->inp));
|
|
if (!hw->trigger_output)
|
|
(void)hw->start_output(sc->hw_hdl, cb->outp,
|
|
blksize, audio_pint, (void *)sc);
|
|
return;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
{
|
|
struct timeval tv;
|
|
u_long t;
|
|
microtime(&tv);
|
|
t = tv.tv_usec + 1000000 * tv.tv_sec;
|
|
if (sc->sc_pnintr) {
|
|
long lastdelta, totdelta;
|
|
lastdelta = t - sc->sc_plastintr - sc->sc_pblktime;
|
|
if (lastdelta > sc->sc_pblktime / 3) {
|
|
printf("audio: play interrupt(%d) off relative by %ld us (%lu)\n",
|
|
sc->sc_pnintr, lastdelta, sc->sc_pblktime);
|
|
}
|
|
totdelta = t - sc->sc_pfirstintr - sc->sc_pblktime * sc->sc_pnintr;
|
|
if (totdelta > sc->sc_pblktime) {
|
|
printf("audio: play interrupt(%d) off absolute by %ld us (%lu) (LOST)\n",
|
|
sc->sc_pnintr, totdelta, sc->sc_pblktime);
|
|
sc->sc_pnintr++; /* avoid repeated messages */
|
|
}
|
|
} else
|
|
sc->sc_pfirstintr = t;
|
|
sc->sc_plastintr = t;
|
|
sc->sc_pnintr++;
|
|
}
|
|
#endif
|
|
|
|
cb->used -= blksize;
|
|
if (cb->used < blksize) {
|
|
/* we don't have a full block to use */
|
|
if (cb->copying) {
|
|
/* writer is in progress, don't disturb */
|
|
cb->needfill = 1;
|
|
DPRINTFN(1, ("audio_pint: copying in progress\n"));
|
|
} else {
|
|
inp = cb->inp;
|
|
cc = blksize - (inp - cb->start) % blksize;
|
|
ccr = cc / sc->sc_pparams.factor;
|
|
if (cb->pause)
|
|
cb->pdrops += ccr;
|
|
else {
|
|
cb->drops += ccr;
|
|
sc->sc_playdrop += ccr;
|
|
}
|
|
audio_pint_silence(sc, cb, inp, cc);
|
|
inp += cc;
|
|
if (inp >= cb->end)
|
|
inp = cb->start;
|
|
cb->inp = inp;
|
|
cb->used += cc;
|
|
|
|
/* Clear next block so we keep ahead of the DMA. */
|
|
if (cb->used + cc < cb->usedhigh)
|
|
audio_pint_silence(sc, cb, inp, blksize);
|
|
}
|
|
}
|
|
|
|
DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->outp, blksize));
|
|
if (!hw->trigger_output) {
|
|
error = hw->start_output(sc->hw_hdl, cb->outp, blksize,
|
|
audio_pint, (void *)sc);
|
|
if (error) {
|
|
/* XXX does this really help? */
|
|
DPRINTF(("audio_pint restart failed: %d\n", error));
|
|
audio_clear(sc);
|
|
}
|
|
}
|
|
|
|
DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n",
|
|
sc->sc_mode, cb->pause, cb->used, cb->usedlow));
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
|
|
if (cb->used <= cb->usedlow) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
selwakeup(&sc->sc_wsel);
|
|
if (sc->sc_async_audio) {
|
|
DPRINTFN(3, ("audio_pint: sending SIGIO %p\n",
|
|
sc->sc_async_audio));
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Possible to return one or more "phantom blocks" now. */
|
|
if (!sc->sc_full_duplex && sc->sc_rchan) {
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selwakeup(&sc->sc_rsel);
|
|
if (sc->sc_async_audio)
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* 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(v)
|
|
void *v;
|
|
{
|
|
struct audio_softc *sc = v;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_ringbuffer *cb = &sc->sc_rr;
|
|
int blksize;
|
|
int error;
|
|
|
|
if (!sc->sc_open)
|
|
return; /* ignore interrupt if not open */
|
|
|
|
blksize = cb->blksize;
|
|
|
|
cb->inp += blksize;
|
|
if (cb->inp >= cb->end)
|
|
cb->inp = cb->start;
|
|
cb->stamp += blksize;
|
|
if (cb->mmapped) {
|
|
DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n",
|
|
cb->inp, blksize));
|
|
if (!hw->trigger_input)
|
|
(void)hw->start_input(sc->hw_hdl, cb->inp, blksize,
|
|
audio_rint, (void *)sc);
|
|
return;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
{
|
|
struct timeval tv;
|
|
u_long t;
|
|
microtime(&tv);
|
|
t = tv.tv_usec + 1000000 * tv.tv_sec;
|
|
if (sc->sc_rnintr) {
|
|
long lastdelta, totdelta;
|
|
lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime;
|
|
if (lastdelta > sc->sc_rblktime / 5) {
|
|
printf("audio: record interrupt(%d) off relative by %ld us (%lu)\n",
|
|
sc->sc_rnintr, lastdelta, sc->sc_rblktime);
|
|
}
|
|
totdelta = t - sc->sc_rfirstintr - sc->sc_rblktime * sc->sc_rnintr;
|
|
if (totdelta > sc->sc_rblktime / 2) {
|
|
sc->sc_rnintr++;
|
|
printf("audio: record interrupt(%d) off absolute by %ld us (%lu)\n",
|
|
sc->sc_rnintr, totdelta, sc->sc_rblktime);
|
|
sc->sc_rnintr++; /* avoid repeated messages */
|
|
}
|
|
} else
|
|
sc->sc_rfirstintr = t;
|
|
sc->sc_rlastintr = t;
|
|
sc->sc_rnintr++;
|
|
}
|
|
#endif
|
|
|
|
cb->used += blksize;
|
|
if (cb->pause) {
|
|
DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops));
|
|
cb->pdrops += blksize;
|
|
cb->outp += blksize;
|
|
cb->used -= blksize;
|
|
} else if (cb->used + blksize >= cb->usedhigh && !cb->copying) {
|
|
DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops));
|
|
cb->drops += blksize;
|
|
cb->outp += blksize;
|
|
cb->used -= blksize;
|
|
}
|
|
|
|
DPRINTFN(2, ("audio_rint: inp=%p cc=%d used=%d\n",
|
|
cb->inp, blksize, cb->used));
|
|
if (!hw->trigger_input) {
|
|
error = hw->start_input(sc->hw_hdl, cb->inp, blksize,
|
|
audio_rint, (void *)sc);
|
|
if (error) {
|
|
/* XXX does this really help? */
|
|
DPRINTF(("audio_rint: restart failed: %d\n", error));
|
|
audio_clear(sc);
|
|
}
|
|
}
|
|
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selwakeup(&sc->sc_rsel);
|
|
if (sc->sc_async_audio)
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
|
|
int
|
|
audio_check_params(p)
|
|
struct audio_params *p;
|
|
{
|
|
if (p->encoding == AUDIO_ENCODING_PCM16) {
|
|
if (p->precision == 8)
|
|
p->encoding = AUDIO_ENCODING_ULINEAR;
|
|
else
|
|
p->encoding = AUDIO_ENCODING_SLINEAR;
|
|
} else if (p->encoding == AUDIO_ENCODING_PCM8) {
|
|
if (p->precision == 8)
|
|
p->encoding = AUDIO_ENCODING_ULINEAR;
|
|
else
|
|
return EINVAL;
|
|
}
|
|
|
|
if (p->encoding == AUDIO_ENCODING_SLINEAR)
|
|
#if BYTE_ORDER == LITTLE_ENDIAN
|
|
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
|
|
#else
|
|
p->encoding = AUDIO_ENCODING_SLINEAR_BE;
|
|
#endif
|
|
if (p->encoding == AUDIO_ENCODING_ULINEAR)
|
|
#if BYTE_ORDER == LITTLE_ENDIAN
|
|
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
|
|
#else
|
|
p->encoding = AUDIO_ENCODING_ULINEAR_BE;
|
|
#endif
|
|
|
|
switch (p->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
case AUDIO_ENCODING_ALAW:
|
|
case AUDIO_ENCODING_ADPCM:
|
|
if (p->precision != 8)
|
|
return (EINVAL);
|
|
break;
|
|
case AUDIO_ENCODING_SLINEAR_LE:
|
|
case AUDIO_ENCODING_SLINEAR_BE:
|
|
case AUDIO_ENCODING_ULINEAR_LE:
|
|
case AUDIO_ENCODING_ULINEAR_BE:
|
|
if (p->precision != 8 && p->precision != 16)
|
|
return (EINVAL);
|
|
break;
|
|
case AUDIO_ENCODING_MPEG_L1_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L1_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
|
|
case AUDIO_ENCODING_MPEG_L2_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L2_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
|
|
break;
|
|
default:
|
|
return (EINVAL);
|
|
}
|
|
|
|
if (p->channels < 1 || p->channels > 8) /* sanity check # of channels */
|
|
return (EINVAL);
|
|
|
|
return (0);
|
|
}
|
|
|
|
int
|
|
au_set_lr_value(sc, ct, l, r)
|
|
struct audio_softc *sc;
|
|
mixer_ctrl_t *ct;
|
|
int l, r;
|
|
{
|
|
ct->type = AUDIO_MIXER_VALUE;
|
|
ct->un.value.num_channels = 2;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
|
|
if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0)
|
|
return 0;
|
|
ct->un.value.num_channels = 1;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
|
|
return sc->hw_if->set_port(sc->hw_hdl, ct);
|
|
}
|
|
|
|
int
|
|
au_set_gain(sc, ports, gain, balance)
|
|
struct audio_softc *sc;
|
|
struct au_mixer_ports *ports;
|
|
int gain;
|
|
int balance;
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, error;
|
|
int l, r;
|
|
u_int mask;
|
|
int nset;
|
|
|
|
if (balance == AUDIO_MID_BALANCE) {
|
|
l = r = gain;
|
|
} else if (balance < AUDIO_MID_BALANCE) {
|
|
r = gain;
|
|
l = (balance * gain) / AUDIO_MID_BALANCE;
|
|
} else {
|
|
l = gain;
|
|
r = ((AUDIO_RIGHT_BALANCE - balance) * gain)
|
|
/ AUDIO_MID_BALANCE;
|
|
}
|
|
DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n",
|
|
gain, balance, l, r));
|
|
|
|
if (ports->index == -1) {
|
|
usemaster:
|
|
if (ports->master == -1)
|
|
return 0; /* just ignore it silently */
|
|
ct.dev = ports->master;
|
|
error = au_set_lr_value(sc, &ct, l, r);
|
|
} else {
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return error;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] == ct.un.ord) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_set_lr_value(sc, &ct, l, r))
|
|
goto usemaster;
|
|
else
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return error;
|
|
mask = ct.un.mask;
|
|
nset = 0;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] & mask) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev != -1 &&
|
|
au_set_lr_value(sc, &ct, l, r) == 0)
|
|
nset++;
|
|
}
|
|
}
|
|
if (nset == 0)
|
|
goto usemaster;
|
|
}
|
|
}
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
au_get_lr_value(sc, ct, l, r)
|
|
struct audio_softc *sc;
|
|
mixer_ctrl_t *ct;
|
|
int *l, *r;
|
|
{
|
|
int error;
|
|
|
|
ct->un.value.num_channels = 2;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) {
|
|
*l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
|
|
*r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
|
|
} else {
|
|
ct->un.value.num_channels = 1;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, ct);
|
|
if (error)
|
|
return error;
|
|
*r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
au_get_gain(sc, ports, pgain, pbalance)
|
|
struct audio_softc *sc;
|
|
struct au_mixer_ports *ports;
|
|
u_int *pgain;
|
|
u_char *pbalance;
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, l, r, n;
|
|
int lgain = AUDIO_MAX_GAIN/2, rgain = AUDIO_MAX_GAIN/2;
|
|
|
|
if (ports->index == -1) {
|
|
usemaster:
|
|
if (ports->master == -1)
|
|
goto bad;
|
|
ct.dev = ports->master;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
if (au_get_lr_value(sc, &ct, &lgain, &rgain))
|
|
goto bad;
|
|
} else {
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
goto bad;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] == ct.un.ord) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_get_lr_value(sc, &ct,
|
|
&lgain, &rgain))
|
|
goto usemaster;
|
|
else
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
goto bad;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
lgain = rgain = n = 0;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] & ct.un.mask) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_get_lr_value(sc, &ct, &l, &r))
|
|
goto usemaster;
|
|
else {
|
|
lgain += l;
|
|
rgain += r;
|
|
n++;
|
|
}
|
|
}
|
|
}
|
|
if (n != 0) {
|
|
lgain /= n;
|
|
rgain /= n;
|
|
}
|
|
}
|
|
}
|
|
bad:
|
|
if (lgain == rgain) { /* handles lgain==rgain==0 */
|
|
*pgain = lgain;
|
|
*pbalance = AUDIO_MID_BALANCE;
|
|
} else if (lgain < rgain) {
|
|
*pgain = rgain;
|
|
*pbalance = (AUDIO_MID_BALANCE * lgain) / rgain;
|
|
} else /* lgain > rgain */ {
|
|
*pgain = lgain;
|
|
*pbalance = AUDIO_RIGHT_BALANCE -
|
|
(AUDIO_MID_BALANCE * rgain) / lgain;
|
|
}
|
|
}
|
|
|
|
int
|
|
au_set_port(sc, ports, port)
|
|
struct audio_softc *sc;
|
|
struct au_mixer_ports *ports;
|
|
u_int port;
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, error;
|
|
|
|
if (port == 0 && ports->allports == 0)
|
|
return 0; /* allow this special case */
|
|
|
|
if (ports->index == -1)
|
|
return EINVAL;
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
if (port & (port-1))
|
|
return EINVAL; /* Only one port allowed */
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
error = EINVAL;
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->aumask[i] == port) {
|
|
ct.un.ord = ports->misel[i];
|
|
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
|
|
break;
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
ct.un.mask = 0;
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->aumask[i] & port)
|
|
ct.un.mask |= ports->misel[i];
|
|
if (port != 0 && ct.un.mask == 0)
|
|
error = EINVAL;
|
|
else
|
|
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
|
|
}
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
au_get_port(sc, ports)
|
|
struct audio_softc *sc;
|
|
struct au_mixer_ports *ports;
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, aumask;
|
|
|
|
if (ports->index == -1)
|
|
return 0;
|
|
ct.dev = ports->index;
|
|
ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
return 0;
|
|
aumask = 0;
|
|
if (ports->isenum) {
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ct.un.ord == ports->misel[i])
|
|
aumask = ports->aumask[i];
|
|
} else {
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ct.un.mask & ports->misel[i])
|
|
aumask |= ports->aumask[i];
|
|
}
|
|
return aumask;
|
|
}
|
|
|
|
int
|
|
audiosetinfo(sc, ai)
|
|
struct audio_softc *sc;
|
|
struct audio_info *ai;
|
|
{
|
|
struct audio_prinfo *r = &ai->record, *p = &ai->play;
|
|
int cleared;
|
|
int s, setmode, modechange = 0;
|
|
int error;
|
|
struct audio_hw_if *hw = sc->hw_if;
|
|
struct audio_params pp, rp;
|
|
int np, nr;
|
|
unsigned int blks;
|
|
int oldpblksize, oldrblksize;
|
|
int rbus, pbus;
|
|
u_int gain;
|
|
u_char balance;
|
|
|
|
if (hw == 0) /* HW has not attached */
|
|
return(ENXIO);
|
|
|
|
rbus = sc->sc_rbus;
|
|
pbus = sc->sc_pbus;
|
|
error = 0;
|
|
cleared = 0;
|
|
|
|
pp = sc->sc_pparams; /* Temporary encoding storage in */
|
|
rp = sc->sc_rparams; /* case setting the modes fails. */
|
|
nr = np = 0;
|
|
|
|
if (p->sample_rate != ~0) {
|
|
pp.sample_rate = p->sample_rate;
|
|
np++;
|
|
}
|
|
if (r->sample_rate != ~0) {
|
|
rp.sample_rate = r->sample_rate;
|
|
nr++;
|
|
}
|
|
if (p->encoding != ~0) {
|
|
pp.encoding = p->encoding;
|
|
np++;
|
|
}
|
|
if (r->encoding != ~0) {
|
|
rp.encoding = r->encoding;
|
|
nr++;
|
|
}
|
|
if (p->precision != ~0) {
|
|
pp.precision = p->precision;
|
|
np++;
|
|
}
|
|
if (r->precision != ~0) {
|
|
rp.precision = r->precision;
|
|
nr++;
|
|
}
|
|
if (p->channels != ~0) {
|
|
pp.channels = p->channels;
|
|
np++;
|
|
}
|
|
if (r->channels != ~0) {
|
|
rp.channels = r->channels;
|
|
nr++;
|
|
}
|
|
#ifdef AUDIO_DEBUG
|
|
if (audiodebug && nr)
|
|
audio_print_params("Setting record params", &rp);
|
|
if (audiodebug && np)
|
|
audio_print_params("Setting play params", &pp);
|
|
#endif
|
|
if (nr && (error = audio_check_params(&rp)))
|
|
return error;
|
|
if (np && (error = audio_check_params(&pp)))
|
|
return error;
|
|
setmode = 0;
|
|
if (nr) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
modechange = cleared = 1;
|
|
rp.sw_code = 0;
|
|
rp.factor = 1;
|
|
setmode |= AUMODE_RECORD;
|
|
}
|
|
if (np) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
modechange = cleared = 1;
|
|
pp.sw_code = 0;
|
|
pp.factor = 1;
|
|
setmode |= AUMODE_PLAY;
|
|
}
|
|
|
|
if (ai->mode != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
modechange = cleared = 1;
|
|
sc->sc_mode = ai->mode;
|
|
if (sc->sc_mode & AUMODE_PLAY_ALL)
|
|
sc->sc_mode |= AUMODE_PLAY;
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex)
|
|
/* Play takes precedence */
|
|
sc->sc_mode &= ~AUMODE_RECORD;
|
|
}
|
|
|
|
if (modechange) {
|
|
int indep = hw->get_props(sc->hw_hdl) & AUDIO_PROP_INDEPENDENT;
|
|
if (!indep) {
|
|
if (setmode == AUMODE_RECORD)
|
|
pp = rp;
|
|
else if (setmode == AUMODE_PLAY)
|
|
rp = pp;
|
|
}
|
|
error = hw->set_params(sc->hw_hdl, setmode,
|
|
sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp);
|
|
if (error)
|
|
return (error);
|
|
if (!indep) {
|
|
if (setmode == AUMODE_RECORD) {
|
|
pp.sample_rate = rp.sample_rate;
|
|
pp.encoding = rp.encoding;
|
|
pp.channels = rp.channels;
|
|
pp.precision = rp.precision;
|
|
} else if (setmode == AUMODE_PLAY) {
|
|
rp.sample_rate = pp.sample_rate;
|
|
rp.encoding = pp.encoding;
|
|
rp.channels = pp.channels;
|
|
rp.precision = pp.precision;
|
|
}
|
|
}
|
|
sc->sc_rparams = rp;
|
|
sc->sc_pparams = pp;
|
|
}
|
|
|
|
oldpblksize = sc->sc_pr.blksize;
|
|
oldrblksize = sc->sc_rr.blksize;
|
|
/* Play params can affect the record params, so recalculate blksize. */
|
|
if (nr || np) {
|
|
audio_calc_blksize(sc, AUMODE_RECORD);
|
|
audio_calc_blksize(sc, AUMODE_PLAY);
|
|
}
|
|
#ifdef AUDIO_DEBUG
|
|
if (audiodebug > 1 && nr)
|
|
audio_print_params("After setting record params", &sc->sc_rparams);
|
|
if (audiodebug > 1 && np)
|
|
audio_print_params("After setting play params", &sc->sc_pparams);
|
|
#endif
|
|
|
|
if (p->port != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
|
|
error = au_set_port(sc, &sc->sc_outports, p->port);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
if (r->port != ~0) {
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
|
|
error = au_set_port(sc, &sc->sc_inports, r->port);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
if (p->gain != ~0) {
|
|
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_outports, p->gain, balance);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
if (r->gain != ~0) {
|
|
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_inports, r->gain, balance);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
|
|
if (p->balance != (u_char)~0) {
|
|
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_outports, gain, p->balance);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
if (r->balance != (u_char)~0) {
|
|
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_inports, gain, r->balance);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
|
|
if (ai->monitor_gain != ~0 &&
|
|
sc->sc_monitor_port != -1) {
|
|
mixer_ctrl_t ct;
|
|
|
|
ct.dev = sc->sc_monitor_port;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return(error);
|
|
}
|
|
|
|
if (p->pause != (u_char)~0) {
|
|
sc->sc_pr.pause = p->pause;
|
|
if (!p->pause && !sc->sc_pbus && (sc->sc_mode & AUMODE_PLAY)) {
|
|
s = splaudio();
|
|
error = audiostartp(sc);
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
}
|
|
if (r->pause != (u_char)~0) {
|
|
sc->sc_rr.pause = r->pause;
|
|
if (!r->pause && !sc->sc_rbus && (sc->sc_mode & AUMODE_RECORD)) {
|
|
s = splaudio();
|
|
error = audiostartr(sc);
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
}
|
|
|
|
if (ai->blocksize != ~0) {
|
|
/* Block size specified explicitly. */
|
|
if (!cleared)
|
|
audio_clear(sc);
|
|
cleared = 1;
|
|
|
|
if (ai->blocksize == 0) {
|
|
audio_calc_blksize(sc, AUMODE_RECORD);
|
|
audio_calc_blksize(sc, AUMODE_PLAY);
|
|
sc->sc_blkset = 0;
|
|
} else {
|
|
int bs = ai->blocksize;
|
|
if (hw->round_blocksize)
|
|
bs = hw->round_blocksize(sc->hw_hdl, bs);
|
|
sc->sc_pr.blksize = sc->sc_rr.blksize = bs;
|
|
sc->sc_blkset = 1;
|
|
}
|
|
}
|
|
|
|
if (ai->mode != ~0) {
|
|
if (sc->sc_mode & AUMODE_PLAY)
|
|
audio_init_play(sc);
|
|
if (sc->sc_mode & AUMODE_RECORD)
|
|
audio_init_record(sc);
|
|
}
|
|
|
|
if (hw->commit_settings) {
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
if (error)
|
|
return (error);
|
|
}
|
|
|
|
if (cleared) {
|
|
s = splaudio();
|
|
error = audio_initbufs(sc);
|
|
if (error) goto err;
|
|
if (sc->sc_pr.blksize != oldpblksize ||
|
|
sc->sc_rr.blksize != oldrblksize)
|
|
audio_calcwater(sc);
|
|
if ((sc->sc_mode & AUMODE_PLAY) &&
|
|
pbus && !sc->sc_pbus)
|
|
error = audiostartp(sc);
|
|
if (!error &&
|
|
(sc->sc_mode & AUMODE_RECORD) &&
|
|
rbus && !sc->sc_rbus)
|
|
error = audiostartr(sc);
|
|
err:
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
/* Change water marks after initializing the buffers. */
|
|
if (ai->hiwat != ~0) {
|
|
blks = ai->hiwat;
|
|
if (blks > sc->sc_pr.maxblks)
|
|
blks = sc->sc_pr.maxblks;
|
|
if (blks < 2)
|
|
blks = 2;
|
|
sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize;
|
|
}
|
|
if (ai->lowat != ~0) {
|
|
blks = ai->lowat;
|
|
if (blks > sc->sc_pr.maxblks - 1)
|
|
blks = sc->sc_pr.maxblks - 1;
|
|
sc->sc_pr.usedlow = blks * sc->sc_pr.blksize;
|
|
}
|
|
if (ai->hiwat != ~0 || ai->lowat != ~0) {
|
|
if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize)
|
|
sc->sc_pr.usedlow = sc->sc_pr.usedhigh - sc->sc_pr.blksize;
|
|
}
|
|
|
|
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;
|
|
|
|
if (hw == 0) /* HW has not attached */
|
|
return(ENXIO);
|
|
|
|
p->sample_rate = sc->sc_pparams.sample_rate;
|
|
r->sample_rate = sc->sc_rparams.sample_rate;
|
|
p->channels = sc->sc_pparams.channels;
|
|
r->channels = sc->sc_rparams.channels;
|
|
p->precision = sc->sc_pparams.precision;
|
|
r->precision = sc->sc_rparams.precision;
|
|
p->encoding = sc->sc_pparams.encoding;
|
|
r->encoding = sc->sc_rparams.encoding;
|
|
|
|
r->port = au_get_port(sc, &sc->sc_inports);
|
|
p->port = au_get_port(sc, &sc->sc_outports);
|
|
|
|
r->avail_ports = sc->sc_inports.allports;
|
|
p->avail_ports = sc->sc_outports.allports;
|
|
|
|
au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance);
|
|
au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance);
|
|
|
|
if (sc->sc_monitor_port != -1) {
|
|
mixer_ctrl_t ct;
|
|
|
|
ct.dev = sc->sc_monitor_port;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
ai->monitor_gain = 0;
|
|
else
|
|
ai->monitor_gain =
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
} else
|
|
ai->monitor_gain = 0;
|
|
|
|
p->seek = sc->sc_pr.used;
|
|
r->seek = sc->sc_rr.used;
|
|
|
|
p->samples = sc->sc_pr.stamp - sc->sc_pr.drops;
|
|
r->samples = sc->sc_rr.stamp - sc->sc_rr.drops;
|
|
|
|
p->eof = sc->sc_eof;
|
|
r->eof = 0;
|
|
|
|
p->pause = sc->sc_pr.pause;
|
|
r->pause = sc->sc_rr.pause;
|
|
|
|
p->error = sc->sc_pr.drops != 0;
|
|
r->error = sc->sc_rr.drops != 0;
|
|
|
|
p->waiting = r->waiting = 0; /* open never hangs */
|
|
|
|
p->open = (sc->sc_open & AUOPEN_WRITE) != 0;
|
|
r->open = (sc->sc_open & AUOPEN_READ) != 0;
|
|
|
|
p->active = sc->sc_pbus;
|
|
r->active = sc->sc_rbus;
|
|
|
|
p->buffer_size = sc->sc_pr.bufsize;
|
|
r->buffer_size = sc->sc_rr.bufsize;
|
|
|
|
ai->blocksize = sc->sc_pr.blksize;
|
|
ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize;
|
|
ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize;
|
|
ai->mode = sc->sc_mode;
|
|
|
|
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;
|
|
|
|
if (unit >= audio_cd.cd_ndevs ||
|
|
(sc = audio_cd.cd_devs[unit]) == NULL)
|
|
return ENXIO;
|
|
|
|
if (!sc->hw_if)
|
|
return (ENXIO);
|
|
|
|
DPRINTF(("mixer_open: dev=0x%x flags=0x%x sc=%p\n", dev, flags, sc));
|
|
|
|
return (0);
|
|
}
|
|
|
|
/*
|
|
* Remove a process from those to be signalled on mixer activity.
|
|
*/
|
|
static void
|
|
mixer_remove(sc, p)
|
|
struct audio_softc *sc;
|
|
struct proc *p;
|
|
{
|
|
struct mixer_asyncs **pm, *m;
|
|
|
|
for(pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
|
|
if ((*pm)->proc == p) {
|
|
m = *pm;
|
|
*pm = m->next;
|
|
free(m, M_DEVBUF);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Signal all processes waitinf for the mixer.
|
|
*/
|
|
static void
|
|
mixer_signal(sc)
|
|
struct audio_softc *sc;
|
|
{
|
|
struct mixer_asyncs *m;
|
|
|
|
for(m = sc->sc_async_mixer; m; m = m->next)
|
|
psignal(m->proc, SIGIO);
|
|
}
|
|
|
|
/*
|
|
* Close a mixer device
|
|
*/
|
|
/* ARGSUSED */
|
|
int
|
|
mixer_close(dev, flags, ifmt, p)
|
|
dev_t dev;
|
|
int flags, ifmt;
|
|
struct proc *p;
|
|
{
|
|
int unit = AUDIOUNIT(dev);
|
|
struct audio_softc *sc = audio_cd.cd_devs[unit];
|
|
|
|
DPRINTF(("mixer_close: unit %d\n", AUDIOUNIT(dev)));
|
|
|
|
mixer_remove(sc, p);
|
|
|
|
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_cd.cd_devs[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 FIOASYNC:
|
|
mixer_remove(sc, p); /* remove old entry */
|
|
if (*(int *)addr) {
|
|
struct mixer_asyncs *ma;
|
|
ma = malloc(sizeof (struct mixer_asyncs),
|
|
M_DEVBUF, M_WAITOK);
|
|
ma->next = sc->sc_async_mixer;
|
|
ma->proc = p;
|
|
sc->sc_async_mixer = ma;
|
|
}
|
|
error = 0;
|
|
break;
|
|
|
|
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 && hw->commit_settings)
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
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 /* NAUDIO > 0 */
|
|
|
|
#include "midi.h"
|
|
|
|
#if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0)
|
|
#include <sys/param.h>
|
|
#include <sys/systm.h>
|
|
#include <sys/device.h>
|
|
#include <sys/audioio.h>
|
|
#include <dev/audio_if.h>
|
|
#endif
|
|
|
|
#if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0)
|
|
int
|
|
audioprint(aux, pnp)
|
|
void *aux;
|
|
const char *pnp;
|
|
{
|
|
struct audio_attach_args *arg = aux;
|
|
const char *type;
|
|
|
|
if (pnp != NULL) {
|
|
switch (arg->type) {
|
|
case AUDIODEV_TYPE_AUDIO:
|
|
type = "audio";
|
|
break;
|
|
case AUDIODEV_TYPE_MIDI:
|
|
type = "midi";
|
|
break;
|
|
case AUDIODEV_TYPE_OPL:
|
|
type = "opl";
|
|
break;
|
|
case AUDIODEV_TYPE_MPU:
|
|
type = "mpu";
|
|
break;
|
|
default:
|
|
panic("audioprint: unknown type %d", arg->type);
|
|
}
|
|
printf("%s at %s", type, pnp);
|
|
}
|
|
return (UNCONF);
|
|
}
|
|
|
|
#endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */
|