/* $NetBSD: audio.c,v 1.182 2004/01/31 00:07:56 fredb Exp $ */ /* * Copyright (c) 1991-1993 Regents of the University of California. * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. All advertising materials mentioning features or use of this software * must display the following acknowledgement: * This product includes software developed by the Computer Systems * Engineering Group at Lawrence Berkeley Laboratory. * 4. Neither the name of the University nor of the Laboratory may be used * to endorse or promote products derived from this software without * specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF * SUCH DAMAGE. */ /* * This is a (partially) SunOS-compatible /dev/audio driver for NetBSD. * * This code tries to do something half-way sensible with * half-duplex hardware, such as with the SoundBlaster hardware. With * half-duplex hardware allowing O_RDWR access doesn't really make * sense. However, closing and opening the device to "turn around the * line" is relatively expensive and costs a card reset (which can * take some time, at least for the SoundBlaster hardware). Instead * we allow O_RDWR access, and provide an ioctl to set the "mode", * i.e. playing or recording. * * If you write to a half-duplex device in record mode, the data is * tossed. If you read from the device in play mode, you get silence * filled buffers at the rate at which samples are naturally * generated. * * If you try to set both play and record mode on a half-duplex * device, playing takes precedence. */ /* * Todo: * - Add softaudio() isr processing for wakeup, poll, signals, * and silence fill. */ #include __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.182 2004/01/31 00:07:56 fredb Exp $"); #include "audio.h" #if NAUDIO > 0 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #ifdef AUDIO_DEBUG #define DPRINTF(x) if (audiodebug) printf x #define DPRINTFN(n,x) if (audiodebug>(n)) printf x int audiodebug = AUDIO_DEBUG; #else #define DPRINTF(x) #define DPRINTFN(n,x) #endif #define ROUNDSIZE(x) x &= -16 /* round to nice boundary */ int audio_blk_ms = AUDIO_BLK_MS; int audiosetinfo(struct audio_softc *, struct audio_info *); int audiogetinfo(struct audio_softc *, struct audio_info *); int audio_open(dev_t, struct audio_softc *, int, int, struct proc *); int audio_close(struct audio_softc *, int, int, struct proc *); int audio_read(struct audio_softc *, struct uio *, int); int audio_write(struct audio_softc *, struct uio *, int); int audio_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *); int audio_poll(struct audio_softc *, int, struct proc *); int audio_kqfilter(struct audio_softc *, struct knote *); paddr_t audio_mmap(struct audio_softc *, off_t, int); int mixer_open(dev_t, struct audio_softc *, int, int, struct proc *); int mixer_close(struct audio_softc *, int, int, struct proc *); int mixer_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *); static void mixer_remove(struct audio_softc *, struct proc *p); static void mixer_signal(struct audio_softc *); void audio_init_record(struct audio_softc *); void audio_init_play(struct audio_softc *); int audiostartr(struct audio_softc *); int audiostartp(struct audio_softc *); void audio_rint(void *); void audio_pint(void *); int audio_check_params(struct audio_params *); void audio_calc_blksize(struct audio_softc *, int); void audio_fill_silence(struct audio_params *, u_char *, int); int audio_silence_copyout(struct audio_softc *, int, struct uio *); void audio_init_ringbuffer(struct audio_softc *, struct audio_ringbuffer *); int audio_initbufs(struct audio_softc *); void audio_calcwater(struct audio_softc *); static __inline int audio_sleep_timo(int *, char *, int); static __inline int audio_sleep(int *, char *); static __inline void audio_wakeup(int *); int audio_drain(struct audio_softc *); void audio_clear(struct audio_softc *); static __inline void audio_pint_silence (struct audio_softc *, struct audio_ringbuffer *, u_char *, int); int audio_alloc_ring (struct audio_softc *, struct audio_ringbuffer *, int, size_t); void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *); int audio_set_defaults(struct audio_softc *, u_int); int audioprobe(struct device *, struct cfdata *, void *); void audioattach(struct device *, struct device *, void *); int audiodetach(struct device *, int); int audioactivate(struct device *, enum devact); struct portname { const char *name; int mask; }; static const struct portname itable[] = { { AudioNmicrophone, AUDIO_MICROPHONE }, { AudioNline, AUDIO_LINE_IN }, { AudioNcd, AUDIO_CD }, { 0 } }; static const struct portname otable[] = { { AudioNspeaker, AUDIO_SPEAKER }, { AudioNheadphone, AUDIO_HEADPHONE }, { AudioNline, AUDIO_LINE_OUT }, { 0 } }; void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, mixer_devinfo_t *, const struct portname *); int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); void au_get_gain(struct audio_softc *, struct au_mixer_ports *, u_int *, u_char *); int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); int au_get_port(struct audio_softc *, struct au_mixer_ports *); int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *r); int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); int au_portof(struct audio_softc *, char *, int); dev_type_open(audioopen); dev_type_close(audioclose); dev_type_read(audioread); dev_type_write(audiowrite); dev_type_ioctl(audioioctl); dev_type_poll(audiopoll); dev_type_mmap(audiommap); dev_type_kqfilter(audiokqfilter); const struct cdevsw audio_cdevsw = { audioopen, audioclose, audioread, audiowrite, audioioctl, nostop, notty, audiopoll, audiommap, audiokqfilter, }; /* The default audio mode: 8 kHz mono mu-law */ const struct audio_params audio_default = { .sample_rate = 8000, .encoding = AUDIO_ENCODING_ULAW, .precision = 8, .channels = 1, .sw_code = 0, .factor = 1, .factor_denom = 1 }; CFATTACH_DECL(audio, sizeof(struct audio_softc), audioprobe, audioattach, audiodetach, audioactivate); extern struct cfdriver audio_cd; int audioprobe(struct device *parent, struct cfdata *match, void *aux) { struct audio_attach_args *sa = aux; DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n", sa->type, sa, sa->hwif)); return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; } void audioattach(struct device *parent, struct device *self, void *aux) { struct audio_softc *sc = (void *)self; struct audio_attach_args *sa = aux; struct audio_hw_if *hwp = sa->hwif; void *hdlp = sa->hdl; int error; mixer_devinfo_t mi; int iclass, mclass, oclass, props; #ifdef DIAGNOSTIC if (hwp == 0 || hwp->open == 0 || hwp->close == 0 || hwp->query_encoding == 0 || hwp->set_params == 0 || (hwp->start_output == 0 && hwp->trigger_output == 0) || (hwp->start_input == 0 && hwp->trigger_input == 0) || hwp->halt_output == 0 || hwp->halt_input == 0 || hwp->getdev == 0 || hwp->set_port == 0 || hwp->get_port == 0 || hwp->query_devinfo == 0 || hwp->get_props == 0) { printf(": missing method\n"); sc->hw_if = 0; return; } #endif props = hwp->get_props(hdlp); if (props & AUDIO_PROP_FULLDUPLEX) printf(": full duplex"); else printf(": half duplex"); if (props & AUDIO_PROP_MMAP) printf(", mmap"); if (props & AUDIO_PROP_INDEPENDENT) printf(", independent"); printf("\n"); sc->hw_if = hwp; sc->hw_hdl = hdlp; sc->sc_dev = parent; error = audio_alloc_ring(sc, &sc->sc_pr, AUMODE_PLAY, AU_RING_SIZE); if (error) { sc->hw_if = 0; printf("audio: could not allocate play buffer\n"); return; } error = audio_alloc_ring(sc, &sc->sc_rr, AUMODE_RECORD, AU_RING_SIZE); if (error) { audio_free_ring(sc, &sc->sc_pr); sc->hw_if = 0; printf("audio: could not allocate record buffer\n"); return; } sc->sc_pconvbuffer = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK); sc->sc_pconvbuffer_size = AU_RING_SIZE; sc->sc_rconvbuffer = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK); sc->sc_rconvbuffer_size = AU_RING_SIZE; if (audio_set_defaults(sc, 0)) panic("audioattach: audio_set_defaults failed"); sc->sc_input_fragment_length = 0; iclass = mclass = oclass = -1; sc->sc_inports.index = -1; sc->sc_inports.master = -1; sc->sc_inports.nports = 0; sc->sc_inports.isenum = 0; sc->sc_inports.allports = 0; sc->sc_inports.isdual = 0; sc->sc_inports.mixerout = -1; sc->sc_inports.cur_port = -1; sc->sc_outports.index = -1; sc->sc_outports.master = -1; sc->sc_outports.nports = 0; sc->sc_outports.isenum = 0; sc->sc_outports.allports = 0; sc->sc_outports.isdual = 0; sc->sc_outports.mixerout = -1; sc->sc_outports.cur_port = -1; sc->sc_monitor_port = -1; for(mi.index = 0; ; mi.index++) { if (hwp->query_devinfo(hdlp, &mi) != 0) break; if (mi.type == AUDIO_MIXER_CLASS) { if (strcmp(mi.label.name, AudioCrecord) == 0) iclass = mi.mixer_class; if (strcmp(mi.label.name, AudioCmonitor) == 0) mclass = mi.mixer_class; if (strcmp(mi.label.name, AudioCoutputs) == 0) oclass = mi.mixer_class; } } for(mi.index = 0; ; mi.index++) { if (hwp->query_devinfo(hdlp, &mi) != 0) break; if (mi.type == AUDIO_MIXER_CLASS) continue; if (mi.mixer_class == iclass) { if (strcmp(mi.label.name, AudioNmaster) == 0) sc->sc_inports.master = mi.index; #if 1 /* Deprecated. Use AudioNmaster. */ if (strcmp(mi.label.name, AudioNrecord) == 0) sc->sc_inports.master = mi.index; if (strcmp(mi.label.name, AudioNvolume) == 0) sc->sc_inports.master = mi.index; #endif if (strcmp(mi.label.name, AudioNsource) == 0) { if (mi.type == AUDIO_MIXER_ENUM) { int i; for(i = 0; i < mi.un.e.num_mem; i++) if (strcmp(mi.un.e.member[i].label.name, AudioNmixerout) == 0) sc->sc_inports.mixerout = mi.un.e.member[i].ord; } au_setup_ports(sc, &sc->sc_inports, &mi, itable); } } else if (mi.mixer_class == mclass) { if (strcmp(mi.label.name, AudioNmonitor) == 0) sc->sc_monitor_port = mi.index; } else if (mi.mixer_class == oclass) { if (strcmp(mi.label.name, AudioNmaster) == 0) sc->sc_outports.master = mi.index; if (strcmp(mi.label.name, AudioNselect) == 0) au_setup_ports(sc, &sc->sc_outports, &mi, otable); } } DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, " "output ports=0x%x, output master=%d\n", sc->sc_inports.allports, sc->sc_inports.master, sc->sc_outports.allports, sc->sc_outports.master)); } int audioactivate(struct device *self, enum devact act) { struct audio_softc *sc = (struct audio_softc *)self; switch (act) { case DVACT_ACTIVATE: return (EOPNOTSUPP); case DVACT_DEACTIVATE: sc->sc_dying = 1; break; } return (0); } int audiodetach(struct device *self, int flags) { struct audio_softc *sc = (struct audio_softc *)self; int maj, mn; int s; DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags)); sc->sc_dying = 1; wakeup(&sc->sc_wchan); wakeup(&sc->sc_rchan); s = splaudio(); if (--sc->sc_refcnt >= 0) { if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120)) printf("audiodetach: %s didn't detach\n", sc->dev.dv_xname); } splx(s); /* free resources */ audio_free_ring(sc, &sc->sc_pr); audio_free_ring(sc, &sc->sc_rr); free(sc->sc_pconvbuffer, M_DEVBUF); free(sc->sc_rconvbuffer, M_DEVBUF); /* locate the major number */ maj = cdevsw_lookup_major(&audio_cdevsw); /* Nuke the vnodes for any open instances (calls close). */ mn = self->dv_unit; vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); return (0); } int au_portof(struct audio_softc *sc, char *name, int class) { mixer_devinfo_t mi; for(mi.index = 0; sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0; mi.index++) if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) return mi.index; return -1; } void au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, mixer_devinfo_t *mi, const struct portname *tbl) { int i, j; ports->index = mi->index; if (mi->type == AUDIO_MIXER_ENUM) { ports->isenum = 1; for(i = 0; tbl[i].name; i++) for(j = 0; j < mi->un.e.num_mem; j++) if (strcmp(mi->un.e.member[j].label.name, tbl[i].name) == 0) { ports->allports |= tbl[i].mask; ports->aumask[ports->nports] = tbl[i].mask; ports->misel[ports->nports] = mi->un.e.member[j].ord; ports->miport[ports->nports] = au_portof(sc, mi->un.e.member[j].label.name, mi->mixer_class); if (ports->mixerout != -1 && ports->miport[ports->nports++] != -1) ports->isdual = 1; } } else if (mi->type == AUDIO_MIXER_SET) { for(i = 0; tbl[i].name; i++) for(j = 0; j < mi->un.s.num_mem; j++) if (strcmp(mi->un.s.member[j].label.name, tbl[i].name) == 0) { ports->allports |= tbl[i].mask; ports->aumask[ports->nports] = tbl[i].mask; ports->misel[ports->nports] = mi->un.s.member[j].mask; ports->miport[ports->nports++] = au_portof(sc, mi->un.s.member[j].label.name, mi->mixer_class); } } } /* * Called from hardware driver. This is where the MI audio driver gets * probed/attached to the hardware driver. */ struct device * audio_attach_mi(struct audio_hw_if *ahwp, void *hdlp, struct device *dev) { struct audio_attach_args arg; #ifdef DIAGNOSTIC if (ahwp == NULL) { aprint_error("audio_attach_mi: NULL\n"); return (0); } #endif arg.type = AUDIODEV_TYPE_AUDIO; arg.hwif = ahwp; arg.hdl = hdlp; return (config_found(dev, &arg, audioprint)); } #ifdef AUDIO_DEBUG void audio_printsc(struct audio_softc *); void audio_print_params(char *, struct audio_params *); void audio_printsc(struct audio_softc *sc) { printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if); printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode); printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan); printf("rring used 0x%x pring used=%d\n", sc->sc_rr.used, sc->sc_pr.used); printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus); printf("blksize %d", sc->sc_pr.blksize); printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow); } void audio_print_params(char *s, struct audio_params *p) { printf("audio: %s sr=%ld, enc=%d, chan=%d, prec=%d\n", s, p->sample_rate, p->encoding, p->channels, p->precision); } #endif int audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r, int direction, size_t bufsize) { struct audio_hw_if *hw = sc->hw_if; void *hdl = sc->hw_hdl; /* * Alloc DMA play and record buffers */ if (bufsize < AUMINBUF) bufsize = AUMINBUF; ROUNDSIZE(bufsize); if (hw->round_buffersize) bufsize = hw->round_buffersize(hdl, direction, bufsize); if (hw->allocm) r->start = hw->allocm(hdl, direction, bufsize, M_DEVBUF, M_WAITOK); else r->start = malloc(bufsize, M_DEVBUF, M_WAITOK); if (r->start == 0) return ENOMEM; r->bufsize = bufsize; return 0; } void audio_free_ring(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); r->start = 0; } int audioopen(dev_t dev, int flags, int ifmt, struct proc *p) { struct audio_softc *sc; int error; sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return (ENXIO); if (sc->sc_dying) return (EIO); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_open(dev, sc, flags, ifmt, p); break; case AUDIOCTL_DEVICE: error = 0; break; case MIXER_DEVICE: error = mixer_open(dev, sc, flags, ifmt, p); break; default: error = ENXIO; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } int audioclose(dev_t dev, int flags, int ifmt, struct proc *p) { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_cd.cd_devs[unit]; int error; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_close(sc, flags, ifmt, p); break; case MIXER_DEVICE: error = mixer_close(sc, flags, ifmt, p); break; case AUDIOCTL_DEVICE: error = 0; break; default: error = ENXIO; break; } return (error); } int audioread(dev_t dev, struct uio *uio, int ioflag) { struct audio_softc *sc; int error; sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return (ENXIO); if (sc->sc_dying) return (EIO); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_read(sc, uio, ioflag); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } int audiowrite(dev_t dev, struct uio *uio, int ioflag) { struct audio_softc *sc; int error; sc = device_lookup(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return (ENXIO); if (sc->sc_dying) return (EIO); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_write(sc, uio, ioflag); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } int audioioctl(dev_t dev, u_long cmd, caddr_t addr, int flag, struct proc *p) { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_cd.cd_devs[unit]; int error; if (sc->sc_dying) return (EIO); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: case AUDIOCTL_DEVICE: error = audio_ioctl(sc, cmd, addr, flag, p); break; case MIXER_DEVICE: error = mixer_ioctl(sc, cmd, addr, flag, p); break; default: error = ENXIO; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } int audiopoll(dev_t dev, int events, struct proc *p) { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_cd.cd_devs[unit]; int error; if (sc->sc_dying) return (EIO); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_poll(sc, events, p); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } int audiokqfilter(dev_t dev, struct knote *kn) { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_cd.cd_devs[unit]; int rv; if (sc->sc_dying) return (1); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: rv = audio_kqfilter(sc, kn); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: rv = 1; break; default: rv = 1; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (rv); } paddr_t audiommap(dev_t dev, off_t off, int prot) { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_cd.cd_devs[unit]; paddr_t error; if (sc->sc_dying) return (-1); sc->sc_refcnt++; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_mmap(sc, off, prot); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = -1; break; default: error = -1; break; } if (--sc->sc_refcnt < 0) wakeup(&sc->sc_refcnt); return (error); } /* * Audio driver */ void audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp) { int nblks; int blksize = rp->blksize; if (blksize < AUMINBLK) blksize = AUMINBLK; if (blksize > rp->bufsize / AUMINNOBLK) blksize = rp->bufsize / AUMINNOBLK; ROUNDSIZE(blksize); if (sc->hw_if->round_blocksize) blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize); if (blksize <= 0) panic("audio_init_ringbuffer: blksize"); nblks = rp->bufsize / 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(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->sc_rr); #if NAURATECONV > 0 auconv_init_context(&sc->sc_rconv, sc->sc_rparams.hw_sample_rate, sc->sc_rparams.sample_rate, sc->sc_rr.start, sc->sc_rr.end); #endif 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->sc_pr); #if NAURATECONV > 0 auconv_init_context(&sc->sc_pconv, sc->sc_pparams.sample_rate, sc->sc_pparams.hw_sample_rate, sc->sc_pr.start, sc->sc_pr.end); #endif 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(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 low 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_rr.end - sc->sc_rr.start - sc->sc_rr.blksize; sc->sc_rr.usedlow = 0; } static __inline int audio_sleep_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 && st != EINTR) DPRINTF(("audio_sleep: woke up st=%d\n", st)); #endif return (st); } static __inline int audio_sleep(int *chan, char *label) { return audio_sleep_timo(chan, label, 0); } /* call at splaudio() */ static __inline void audio_wakeup(int *chan) { DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan)); if (*chan) { wakeup(chan); *chan = 0; } } int audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, struct proc *p) { int error; u_int mode; struct audio_hw_if *hw; hw = sc->hw_if; if (!hw) return ENXIO; DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n", flags, sc, sc->hw_hdl)); 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_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)) { error = audio_set_defaults(sc, mode); } else { struct audio_info ai; AUDIO_INITINFO(&ai); ai.mode = mode; error = audiosetinfo(sc, &ai); } if (error) goto bad; #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_close() decreases sc_pr.usedlow, recalculate here */ audio_calcwater(sc); 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(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); sc->sc_rconvbuffer_begin = 0; sc->sc_rconvbuffer_end = 0; splx(s); } /* * Must be called from task context. */ void audio_init_play(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); sc->sc_input_fragment_length = 0; splx(s); } int audio_drain(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); if (sc->sc_dying) error = EIO; } splx(s); return error; } /* * Close an audio chip. */ /* ARGSUSED */ int audio_close(struct audio_softc *sc, int flags, int ifmt, struct proc *p) { struct audio_hw_if *hw = sc->hw_if; int s; DPRINTF(("audio_close: sc=%p\n", sc)); 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(struct audio_softc *sc, struct uio *uio, int ioflag) { struct audio_ringbuffer *cb = &sc->sc_rr; u_char *outp; int error, s, used, cc, n; const struct audio_params *params; int hw_bits_per_sample; if (cb->mmapped) return EINVAL; DPRINTFN(1,("audio_read: cc=%lu mode=%d\n", (unsigned long)uio->uio_resid, sc->sc_mode)); params = &sc->sc_rparams; switch (params->hw_encoding) { case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: hw_bits_per_sample = params->hw_channels * params->precision * params->factor; break; default: hw_bits_per_sample = 8 * params->factor / params->factor_denom; } 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 (sc->sc_dying) error = EIO; 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) { if (sc->sc_rconvbuffer_end - sc->sc_rconvbuffer_begin <= 0) { s = splaudio(); while (cb->used * 8 < hw_bits_per_sample) { if (!sc->sc_rbus) { error = audiostartr(sc); if (error) { splx(s); return (error); } } if (ioflag & IO_NDELAY) { splx(s); return (EWOULDBLOCK); } DPRINTFN(1, ("audio_read: sleep used=%d\n", cb->used)); error = audio_sleep(&sc->sc_rchan, "aud_rd"); if (sc->sc_dying) error = EIO; if (error) { splx(s); return error; } } /* * Move data in the ring buffer to sc_rconvbuffer as * possible with/without rate conversion. */ used = cb->used; outp = cb->outp; cb->copying = 1; splx(s); cc = used - cb->usedlow; /* maximum to read */ if (cc > sc->sc_rconvbuffer_size) cc = sc->sc_rconvbuffer_size; n = cc * params->factor / params->factor_denom; if (n < cc) cc = n; /* cc must be aligned by the sample size */ cc = (cc * 8 / hw_bits_per_sample) * hw_bits_per_sample / 8; #ifdef DIAGNOSTIC if (cc == 0) printf("audio_read: cc=0 hw_bits_per_sample=%d\n", hw_bits_per_sample); #endif /* * The format of data in the ring buffer is * [hw_sample_rate, hw_encoding, hw_precision, hw_channels] */ #if NAURATECONV > 0 sc->sc_rconvbuffer_end = auconv_record(&sc->sc_rconv, params, sc->sc_rconvbuffer, outp, cc); #else n = cb->end - outp; if (cc <= n) { memcpy(sc->sc_rconvbuffer, outp, cc); } else { memcpy(sc->sc_rconvbuffer, outp, n); memcpy(sc->sc_rconvbuffer + n, cb->start, cc - n); } sc->sc_rconvbuffer_end = cc; #endif /* !NAURATECONV */ /* * The format of data in sc_rconvbuffer is * [sample_rate, hw_encoding, hw_precision, channels] */ outp += cc; if (outp >= cb->end) outp -= cb->end - cb->start; s = splaudio(); cb->outp = outp; cb->used -= cc; cb->copying = 0; splx(s); if (params->sw_code) { cc = sc->sc_rconvbuffer_end; #ifdef DIAGNOSTIC if (cc % params->factor != 0) printf("audio_read: cc is not aligned" ": cc=%d factor=%d\n", cc, params->factor); #endif cc = cc * params->factor_denom / params->factor; #ifdef DIAGNOSTIC if (cc == 0) printf("audio_read: cc=0 " "factor=%d/%d\n", params->factor, params->factor_denom); #endif params->sw_code(sc->hw_hdl, sc->sc_rconvbuffer, cc); sc->sc_rconvbuffer_end = cc; } sc->sc_rconvbuffer_begin = 0; /* * The format of data in sc_rconvbuffer is * [sample_rate, encoding, precision, channels] */ } cc = sc->sc_rconvbuffer_end - sc->sc_rconvbuffer_begin; if (uio->uio_resid < cc) cc = uio->uio_resid; /* and no more than we want */ DPRINTFN(0,("audio_read: buffer=%p[%d] (~ %d), cc=%d\n", sc->sc_rconvbuffer, sc->sc_rconvbuffer_begin, sc->sc_rconvbuffer_end, cc)); n = uio->uio_resid; error = uiomove(sc->sc_rconvbuffer + sc->sc_rconvbuffer_begin, cc, uio); cc = n - uio->uio_resid; /* number of bytes actually moved */ sc->sc_rconvbuffer_begin += cc; } return (error); } void audio_clear(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(struct audio_softc *sc, int mode) { struct audio_params *parm; struct audio_ringbuffer *rb; 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; } rb->blksize = parm->hw_sample_rate * audio_blk_ms / 1000 * parm->hw_channels * parm->precision / NBBY * parm->factor; DPRINTF(("audio_calc_blksize: %s blksize=%d\n", mode == AUMODE_PLAY ? "play" : "record", rb->blksize)); } void audio_fill_silence(struct audio_params *params, u_char *p, int n) { u_char auzero0, auzero1 = 0; /* initialize to please gcc */ int nfill = 1; switch (params->hw_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 any number of bits */ break; case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: if (params->hw_precision > 8) { nfill = (params->hw_precision + NBBY - 1)/ NBBY; 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 no longer be 2 */ { if (params->hw_encoding == AUDIO_ENCODING_ULINEAR_LE) { int k = nfill; while (--k > 0) *p++ = auzero1; n -= nfill - 1; } while (n >= nfill) { int k = nfill; *p++ = auzero0; while (--k > 0) *p++ = auzero1; n -= nfill; } if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */ *p++ = auzero0; } } int audio_silence_copyout(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(struct audio_softc *sc, struct uio *uio, int ioflag) { struct audio_ringbuffer *cb = &sc->sc_pr; u_char *inp, *einp; int saveerror, error, s, n, cc, used; struct audio_params *params; int samples, hw_bits_per_sample, user_bits_per_sample; int input_remain, space; DPRINTFN(2,("audio_write: sc=%p count=%lu used=%d(hi=%d)\n", sc, (unsigned long)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; } params = &sc->sc_pparams; 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)); /* * For some encodings, handle data in sample unit. */ switch (params->hw_encoding) { case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: hw_bits_per_sample = params->hw_channels * params->precision * params->factor; user_bits_per_sample = params->channels * params->precision; break; default: hw_bits_per_sample = 8 * params->factor / params->factor_denom; user_bits_per_sample = 8; } #ifdef DIAGNOSTIC if (hw_bits_per_sample > MAX_SAMPLE_SIZE * 8) { printf("audio_write(): Invalid sample size: cur=%d max=%d\n", hw_bits_per_sample / 8, MAX_SAMPLE_SIZE); } #endif space = ((params->hw_sample_rate / params->sample_rate) + 1) * hw_bits_per_sample / 8; error = 0; while ((input_remain = uio->uio_resid + sc->sc_input_fragment_length) > 0 && !error) { s = splaudio(); if (input_remain < user_bits_per_sample / 8) { n = uio->uio_resid; DPRINTF(("audio_write: fragment uiomove length=%d\n", n)); error = uiomove(sc->sc_input_fragment + sc->sc_input_fragment_length, n, uio); if (!error) sc->sc_input_fragment_length += n; splx(s); return (error); } while (cb->used + space >= 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 (sc->sc_dying) error = EIO; if (error) { splx(s); return (error); } } used = cb->used; inp = cb->inp; cb->copying = 1; splx(s); cc = cb->usedhigh - used; /* maximum to write */ /* cc may be greater than the size of the ring buffer */ if (cc > cb->end - cb->start) cc = cb->end - cb->start; /* cc: # of bytes we can write to the ring buffer */ samples = cc * 8 / hw_bits_per_sample; #ifdef DIAGNOSTIC if (samples == 0) printf("audio_write: samples (cc/hw_bps) == 0\n"); #endif /* samples: # of samples we can write to the ring buffer */ samples = samples * params->sample_rate / params->hw_sample_rate; #ifdef DIAGNOSTIC if (samples == 0) printf("audio_write: samples (rate/hw_rate) == 0 " "usedhigh-used=%d cc/hw_bps=%d/%d " "rate/hw_rate=%ld/%ld space=%d\n", cb->usedhigh - cb->used, cc, hw_bits_per_sample / 8, params->sample_rate, params->hw_sample_rate, space); #endif /* samples: # of samples in source data */ cc = samples * user_bits_per_sample / 8; /* cc: # of bytes in source data */ if (input_remain < cc) /* and no more than we have */ cc = (input_remain * 8 / user_bits_per_sample) * user_bits_per_sample / 8; #ifdef DIAGNOSTIC if (cc == 0) printf("audio_write: cc == 0\n"); #endif if (cc * params->factor / params->factor_denom > sc->sc_pconvbuffer_size) { /* * cc = (pconv / factor / user_bps ) * user_bps */ cc = (sc->sc_pconvbuffer_size * params->factor_denom * 8 / params->factor / user_bits_per_sample) * user_bits_per_sample / 8; } #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 " "remain=%d u_bps=%d hw_bps=%d\n", sc->sc_pparams.sw_code, sc->sc_pparams.factor, input_remain, user_bits_per_sample, hw_bits_per_sample); cb->copying = 0; return EINVAL; } #endif DPRINTFN(1, ("audio_write: uiomove cc=%d inp=%p, left=%lu\n", cc, inp, (unsigned long)uio->uio_resid)); memcpy(sc->sc_pconvbuffer, sc->sc_input_fragment, sc->sc_input_fragment_length); cc -= sc->sc_input_fragment_length; n = uio->uio_resid; error = uiomove(sc->sc_pconvbuffer + sc->sc_input_fragment_length, cc, uio); if (cc != n - uio->uio_resid) { printf("audio_write: uiomove didn't move requested " "amount: requested=%d, actual=%ld\n", cc, (long)n - uio->uio_resid); } /* number of bytes actually moved */ cc = sc->sc_input_fragment_length + n - uio->uio_resid; sc->sc_input_fragment_length = 0; #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. */ /* * The format of data in sc_pconvbuffer is: * [sample_rate, encoding, precision, channels] */ if (sc->sc_pparams.sw_code) { sc->sc_pparams.sw_code(sc->hw_hdl, sc->sc_pconvbuffer, cc); /* Adjust count after the expansion. */ cc = cc * sc->sc_pparams.factor / sc->sc_pparams.factor_denom; DPRINTFN(1, ("audio_write: expanded cc=%d\n", cc)); } /* * The format of data in sc_pconvbuffer is: * [sample_rate, hw_encoding, hw_precision, channels] */ #if NAURATECONV > 0 cc = auconv_play(&sc->sc_pconv, params, inp, sc->sc_pconvbuffer, cc); #else n = cb->end - inp; if (cc <= n) { memcpy(inp, sc->sc_pconvbuffer, cc); } else { memcpy(inp, sc->sc_pconvbuffer, n); memcpy(cb->start, sc->sc_pconvbuffer + n, cc - n); } #endif /* !NAURATECONV */ /* * The format of data in inp is: * [hw_sample_rate, hw_encoding, hw_precision, hw_channels] * cc is the size of data actually written to inp. */ einp = cb->inp + cc; if (einp >= cb->end) einp -= cb->end - cb->start; /* not cb->bufsize */ 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 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) { saveerror = error; error = audiostartp(sc); if (saveerror != 0) { /* Report the first error that occurred. */ error = saveerror; } } splx(s); if (cc != 0) { DPRINTFN(1, ("audio_write: fill %d\n", cc)); audio_fill_silence(&sc->sc_pparams, einp, cc); } } return (error); } int audio_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag, struct proc *p) { struct audio_hw_if *hw = sc->hw_if; struct audio_offset *ao; int error = 0, s, offs, fd; u_long stamp; int rbus, pbus; DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); switch (cmd) { case FIONBIO: /* All handled in the upper FS layer. */ break; case FIONREAD: *(int *)addr = sc->sc_rr.used; 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: ao = (struct audio_offset *)addr; s = splaudio(); /* figure out where next DMA will start */ stamp = sc->sc_rr.stamp; offs = sc->sc_rr.inp - sc->sc_rr.start; splx(s); ao->samples = stamp; ao->deltablks = (stamp / sc->sc_rr.blksize) - (sc->sc_rr.stamp_last / sc->sc_rr.blksize); sc->sc_rr.stamp_last = stamp; ao->offset = offs; break; case AUDIO_GETOOFFS: ao = (struct audio_offset *)addr; s = splaudio(); /* figure out where next DMA will start */ stamp = sc->sc_pr.stamp; offs = sc->sc_pr.outp - sc->sc_pr.start + sc->sc_pr.blksize; splx(s); ao->samples = stamp; ao->deltablks = (stamp / sc->sc_pr.blksize) - (sc->sc_pr.stamp_last / sc->sc_pr.blksize); sc->sc_pr.stamp_last = stamp; if (sc->sc_pr.start + offs >= sc->sc_pr.end) offs = 0; ao->offset = offs; 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: if (hw->dev_ioctl) { error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p); } else { DPRINTF(("audio_ioctl: unknown ioctl\n")); error = EINVAL; } break; } DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); return (error); } int audio_poll(struct audio_softc *sc, int events, struct proc *p) { int revents = 0; int s = splaudio(); DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode)); if (events & (POLLIN | POLLRDNORM)) { /* * If half duplex and playing, audio_read() will generate * silence at the play rate; poll for silence being * available. Otherwise, poll for recorded sound. */ if ((!sc->sc_full_duplex && (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 half duplex and recording, audio_write() will throw * away play data, which means we are always ready to write. * Otherwise, poll for play buffer being below its low water * mark. */ if ((!sc->sc_full_duplex && (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); } static void filt_audiordetach(struct knote *kn) { struct audio_softc *sc = kn->kn_hook; int s; s = splaudio(); SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); splx(s); } static int filt_audioread(struct knote *kn, long hint) { struct audio_softc *sc = kn->kn_hook; int s; /* XXXLUKEM (thorpej): please make sure this is right */ s = splaudio(); if (sc->sc_mode & AUMODE_PLAY) kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp; else kn->kn_data = sc->sc_rr.used - sc->sc_rr.usedlow; splx(s); return (kn->kn_data > 0); } static const struct filterops audioread_filtops = { 1, NULL, filt_audiordetach, filt_audioread }; static void filt_audiowdetach(struct knote *kn) { struct audio_softc *sc = kn->kn_hook; int s; s = splaudio(); SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); splx(s); } static int filt_audiowrite(struct knote *kn, long hint) { struct audio_softc *sc = kn->kn_hook; int s; /* XXXLUKEM (thorpej): please make sure this is right */ s = splaudio(); kn->kn_data = sc->sc_pr.usedlow - sc->sc_pr.used; splx(s); return (kn->kn_data > 0); } static const struct filterops audiowrite_filtops = { 1, NULL, filt_audiowdetach, filt_audiowrite }; int audio_kqfilter(struct audio_softc *sc, struct knote *kn) { struct klist *klist; int s; switch (kn->kn_filter) { case EVFILT_READ: klist = &sc->sc_rsel.sel_klist; kn->kn_fop = &audioread_filtops; break; case EVFILT_WRITE: klist = &sc->sc_wsel.sel_klist; kn->kn_fop = &audiowrite_filtops; break; default: return (1); } kn->kn_hook = sc; s = splaudio(); SLIST_INSERT_HEAD(klist, kn, kn_selnext); splx(s); return (0); } paddr_t audio_mmap(struct audio_softc *sc, off_t off, int prot) { struct audio_hw_if *hw = sc->hw_if; struct audio_ringbuffer *cb; int s; DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)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(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(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(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 sil., 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(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); selnotify(&sc->sc_wsel, 0); 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); selnotify(&sc->sc_rsel, 0); 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(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; if (cb->outp >= cb->end) cb->outp = cb->start; 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; if (cb->outp >= cb->end) cb->outp = cb->start; 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); selnotify(&sc->sc_rsel, 0); if (sc->sc_async_audio) psignal(sc->sc_async_audio, SIGIO); } int audio_check_params(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: if (p->precision != 8) return (EINVAL); break; case AUDIO_ENCODING_ADPCM: if (p->precision != 4 && 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: /* XXX is: our zero-fill can handle any multiple of 8 */ if (p->precision != 8 && p->precision != 16 && p->precision != 24 && p->precision != 32) 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 audio_set_defaults(struct audio_softc *sc, u_int mode) { struct audio_info ai; /* default parameters */ sc->sc_rparams = audio_default; sc->sc_pparams = audio_default; sc->sc_blkset = 0; 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; return (audiosetinfo(sc, &ai)); } int au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int 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(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) { l = gain; r = (balance * gain) / AUDIO_MID_BALANCE; } else { r = gain; l = ((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; if (ports->isdual) { if (ports->cur_port == -1) ct.dev = ports->master; else ct.dev = ports->miport[ports->cur_port]; error = au_set_lr_value(sc, &ct, l, r); } else { 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(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *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(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; if (ports->isdual) { if (ports->cur_port == -1) ct.dev = ports->master; else ct.dev = ports->miport[ports->cur_port]; au_get_lr_value(sc, &ct, &lgain, &rgain); } else { 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; /* balance should be > AUDIO_MID_BALANCE */ *pbalance = AUDIO_RIGHT_BALANCE - (AUDIO_MID_BALANCE * lgain) / rgain; } else /* lgain > rgain */ { *pgain = lgain; /* balance should be < AUDIO_MID_BALANCE */ *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; } } int au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) { mixer_ctrl_t ct; int i, error, use_mixerout; use_mixerout = 1; if (port == 0) { if (ports->allports == 0) return 0; /* Allow this special case. */ else if (ports->isdual) { if (ports->cur_port == -1) { return 0; } else { port = ports->aumask[ports->cur_port]; ports->cur_port = -1; use_mixerout = 0; } } } 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) { if (ports->isdual && use_mixerout) { ct.un.ord = ports->mixerout; ports->cur_port = i; } else { 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(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) { if (ports->isdual && ports->cur_port != -1) { if (ports->mixerout == ct.un.ord) aumask = ports->aumask[ports->cur_port]; else ports->cur_port = -1; } if (aumask == 0) for(i = 0; i < ports->nports; i++) if (ports->misel[i] == ct.un.ord) 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; } #if NAURATECONV <= 0 /* dummy function for the case that aurateconv is not linked */ int auconv_check_params(const struct audio_params *params) { if (params->hw_channels == params->channels && params->hw_sample_rate == params->sample_rate) return 0; /* No conversion */ return (EINVAL); } #endif /* !NAURATECONV */ int audiosetinfo(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); cleared = 1; } modechange = 1; rp.sw_code = 0; rp.factor = 1; rp.factor_denom = 1; rp.hw_sample_rate = rp.sample_rate; rp.hw_encoding = rp.encoding; rp.hw_precision = rp.precision; rp.hw_channels = rp.channels; setmode |= AUMODE_RECORD; } if (np) { if (!cleared) { audio_clear(sc); cleared = 1; } modechange = 1; pp.sw_code = 0; pp.factor = 1; pp.factor_denom = 1; pp.hw_sample_rate = pp.sample_rate; pp.hw_encoding = pp.encoding; pp.hw_precision = pp.precision; pp.hw_channels = pp.channels; setmode |= AUMODE_PLAY; } if (ai->mode != ~0) { if (!cleared) { audio_clear(sc); cleared = 1; } modechange = 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 orig_p_channels, orig_p_rate; int orig_r_channels, orig_r_rate; int indep; 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; } /* Some device drivers change channels/sample_rate and change * no channels/sample_rate. */ orig_p_channels = pp.channels; orig_p_rate = pp.sample_rate; orig_r_channels = rp.channels; orig_r_rate = rp.sample_rate; error = hw->set_params(sc->hw_hdl, setmode, sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp); if (error) return (error); if (np) { if (orig_p_channels != pp.channels) pp.hw_channels = pp.channels; if (orig_p_rate != pp.sample_rate) pp.hw_sample_rate = pp.sample_rate; error = auconv_check_params(&pp); if (error) return (error); } if (nr) { if (orig_r_channels != rp.channels) rp.hw_channels = rp.channels; if (orig_r_rate != rp.sample_rate) rp.hw_sample_rate = rp.sample_rate; error = auconv_check_params(&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->set_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) { sc->sc_blkset = 0; audio_calc_blksize(sc, AUMODE_RECORD); audio_calc_blksize(sc, AUMODE_PLAY); } else { sc->sc_blkset = 1; sc->sc_pr.blksize = sc->sc_rr.blksize = ai->blocksize; } } 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(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_t dev, struct audio_softc *sc, int flags, int ifmt, struct proc *p) { if (!sc->hw_if) return (ENXIO); DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc)); return (0); } /* * Remove a process from those to be signalled on mixer activity. */ static void mixer_remove(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 waiting for the mixer. */ static void mixer_signal(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(struct audio_softc *sc, int flags, int ifmt, struct proc *p) { DPRINTF(("mixer_close: sc %p\n", sc)); mixer_remove(sc, p); return (0); } int mixer_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag, struct proc *p) { struct audio_hw_if *hw = sc->hw_if; int error = EINVAL; DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n", IOCPARM_LEN(cmd), (char)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")); ((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */ 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: if (hw->dev_ioctl) error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p); else error = EINVAL; break; } DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); return (error); } #endif /* NAUDIO > 0 */ #include "midi.h" #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) #include #include #include #include #include #endif #if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) int audioprint(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); } aprint_normal("%s at %s", type, pnp); } return (UNCONF); } #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */