/* $NetBSD: audio.c,v 1.104 1998/09/01 07:27:06 mycroft 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 "audio.h" #if NAUDIO > 0 #include #include #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 = 0; #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 __P((struct audio_softc *, struct audio_info *)); int audiogetinfo __P((struct audio_softc *, struct audio_info *)); int audio_open __P((dev_t, int, int, struct proc *)); int audio_close __P((dev_t, int, int, struct proc *)); int audio_read __P((dev_t, struct uio *, int)); int audio_write __P((dev_t, struct uio *, int)); int audio_ioctl __P((dev_t, int, caddr_t, int, struct proc *)); int audio_poll __P((dev_t, int, struct proc *)); int audio_mmap __P((dev_t, int, int)); int mixer_open __P((dev_t, int, int, struct proc *)); int mixer_close __P((dev_t, int, int, struct proc *)); int mixer_ioctl __P((dev_t, int, caddr_t, int, struct proc *)); static void mixer_remove __P((struct audio_softc *, struct proc *p)); static void mixer_signal __P((struct audio_softc *)); void audio_init_record __P((struct audio_softc *)); void audio_init_play __P((struct audio_softc *)); int audiostartr __P((struct audio_softc *)); int audiostartp __P((struct audio_softc *)); void audio_rint __P((void *)); void audio_pint __P((void *)); int audio_check_params __P((struct audio_params *)); void audio_calc_blksize __P((struct audio_softc *, int)); void audio_fill_silence __P((struct audio_params *, u_char *, int)); int audio_silence_copyout __P((struct audio_softc *, int, struct uio *)); void audio_init_ringbuffer __P((struct audio_ringbuffer *)); int audio_initbufs __P((struct audio_softc *)); void audio_calcwater __P((struct audio_softc *)); static __inline int audio_sleep_timo __P((int *, char *, int)); static __inline int audio_sleep __P((int *, char *)); static __inline void audio_wakeup __P((int *)); int audio_drain __P((struct audio_softc *)); void audio_clear __P((struct audio_softc *)); static __inline void audio_pint_silence __P((struct audio_softc *, struct audio_ringbuffer *, u_char *, int)); int audio_alloc_ring __P((struct audio_softc *, struct audio_ringbuffer *, int)); void audio_free_ring __P((struct audio_softc *, struct audio_ringbuffer *)); int audioprobe __P((struct device *, struct cfdata *, void *)); void audioattach __P((struct device *, struct device *, void *)); struct portname { char *name; int mask; }; static struct portname itable[] = { { AudioNmicrophone, AUDIO_MICROPHONE }, { AudioNline, AUDIO_LINE_IN }, { AudioNcd, AUDIO_CD }, { 0 } }; static struct portname otable[] = { { AudioNspeaker, AUDIO_SPEAKER }, { AudioNheadphone, AUDIO_HEADPHONE }, { AudioNline, AUDIO_LINE_OUT }, { 0 } }; void au_check_ports __P((struct audio_softc *, struct au_mixer_ports *, mixer_devinfo_t *, int, char *, char *, struct portname *)); int au_set_gain __P((struct audio_softc *, struct au_mixer_ports *, int, int)); void au_get_gain __P((struct audio_softc *, struct au_mixer_ports *, u_int *, u_char *)); int au_set_port __P((struct audio_softc *, struct au_mixer_ports *, u_int)); int au_get_port __P((struct audio_softc *, struct au_mixer_ports *)); int au_get_lr_value __P((struct audio_softc *, mixer_ctrl_t *, int *, int *r)); int au_set_lr_value __P((struct audio_softc *, mixer_ctrl_t *, int, int)); int au_portof __P((struct audio_softc *, char *)); /* The default audio mode: 8 kHz mono ulaw */ struct audio_params audio_default = { 8000, AUDIO_ENCODING_ULAW, 8, 1, 0, 1 }; struct cfattach audio_ca = { sizeof(struct audio_softc), audioprobe, audioattach }; extern struct cfdriver audio_cd; int audioprobe(parent, match, aux) 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(parent, self, aux) struct device *parent, *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, oclass; printf("\n"); #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("audio: missing method\n"); sc->hw_if = 0; return; } #endif sc->hw_if = hwp; sc->hw_hdl = hdlp; sc->sc_dev = parent; error = audio_alloc_ring(sc, &sc->sc_pr, 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, 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; } /* * Set default softc params */ sc->sc_pparams = audio_default; sc->sc_rparams = audio_default; /* Set up some default values */ sc->sc_blkset = 0; audio_calc_blksize(sc, AUMODE_RECORD); audio_calc_blksize(sc, AUMODE_PLAY); audio_init_ringbuffer(&sc->sc_rr); audio_init_ringbuffer(&sc->sc_pr); audio_calcwater(sc); iclass = oclass = -1; sc->sc_inports.index = -1; sc->sc_inports.nports = 0; sc->sc_inports.isenum = 0; sc->sc_inports.allports = 0; sc->sc_outports.index = -1; sc->sc_outports.nports = 0; sc->sc_outports.isenum = 0; sc->sc_outports.allports = 0; 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 && strcmp(mi.label.name, AudioCrecord) == 0) iclass = mi.index; if (mi.type == AUDIO_MIXER_CLASS && strcmp(mi.label.name, AudioCmonitor) == 0) oclass = mi.index; } for(mi.index = 0; ; mi.index++) { if (hwp->query_devinfo(hdlp, &mi) != 0) break; au_check_ports(sc, &sc->sc_inports, &mi, iclass, AudioNsource, AudioNrecord, itable); au_check_ports(sc, &sc->sc_outports, &mi, oclass, AudioNoutput, AudioNmaster, otable); if (mi.mixer_class == oclass && strcmp(mi.label.name, AudioNmonitor)) sc->sc_monitor_port = mi.index; } DPRINTF(("audio_attach: inputs ports=0x%x, output ports=0x%x\n", sc->sc_inports.allports, sc->sc_outports.allports)); } int au_portof(sc, name) struct audio_softc *sc; char *name; { mixer_devinfo_t mi; for(mi.index = 0; sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0; mi.index++) if (strcmp(mi.label.name, name) == 0) return mi.index; return -1; } void au_check_ports(sc, ports, mi, cls, name, mname, tbl) struct audio_softc *sc; struct au_mixer_ports *ports; mixer_devinfo_t *mi; int cls; char *name; char *mname; struct portname *tbl; { int i, j; if (mi->mixer_class != cls) return; if (strcmp(mi->label.name, mname) == 0) { ports->master = mi->index; return; } if (strcmp(mi->label.name, name) != 0) return; if (mi->type == AUDIO_MIXER_ENUM) { ports->index = mi->index; 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->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); ports->allports |= tbl[i].mask; } } } ports->isenum = 1; } else if (mi->type == AUDIO_MIXER_SET) { ports->index = mi->index; 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->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); ports->allports |= tbl[i].mask; } } } } } /* * Called from hardware driver. This is where the MI audio driver gets * probed/attached to the hardware driver. */ void audio_attach_mi(ahwp, hdlp, dev) struct audio_hw_if *ahwp; void *hdlp; struct device *dev; { struct audio_attach_args arg; #ifdef DIAGNOSTIC if (ahwp == NULL) { printf("audio_attach_mi: NULL\n"); return; } #endif arg.type = AUDIODEV_TYPE_AUDIO; arg.hwif = ahwp; arg.hdl = hdlp; (void)config_found(dev, &arg, audioprint); } #ifdef AUDIO_DEBUG void audio_printsc __P((struct audio_softc *)); void audio_print_params __P((char *, struct audio_params *)); void audio_printsc(sc) 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(s, p) 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(sc, r, bufsize) struct audio_softc *sc; struct audio_ringbuffer *r; int bufsize; { struct audio_hw_if *hw = sc->hw_if; void *hdl = sc->hw_hdl; /* * Alloc DMA play and record buffers */ ROUNDSIZE(bufsize); if (bufsize < AUMINBUF) bufsize = AUMINBUF; if (hw->round_buffersize) bufsize = hw->round_buffersize(hdl, bufsize); r->bufsize = bufsize; if (hw->allocm) r->start = hw->allocm(hdl, r->bufsize, M_DEVBUF, M_WAITOK); else r->start = malloc(bufsize, M_DEVBUF, M_WAITOK); if (r->start == 0) 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); } } 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 (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 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 */