/* $OpenBSD: audio.c,v 1.10 1997/07/10 23:06:29 provos Exp $ */ /* $NetBSD: audio.c,v 1.26 1996/05/13 02:26:15 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, select and signals. * - Allow opens for READ and WRITE (one open each) * - Setup for single isr for full-duplex * - Add SIGIO generation for changes in the mixer device * - Fixup SunOS compat for mixer device changes in ioctl. */ #include "audio.h" #if NAUDIO > 0 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #ifdef AUDIO_DEBUG #include void Dprintf __P((const char *, ...)); void #ifdef __STDC__ Dprintf(const char *fmt, ...) #else Dprintf(fmt, va_alist) char *fmt; #endif { va_list ap; va_start(ap, fmt); log(LOG_DEBUG, "%:", fmt, ap); va_end(ap); } #define DPRINTF(x) if (audiodebug) Dprintf x int audiodebug = 0; #else #define DPRINTF(x) #endif int naudio; /* Count of attached hardware drivers */ int audio_blk_ms = AUDIO_BLK_MS; int audio_backlog = AUDIO_BACKLOG; struct audio_softc *audio_softc[NAUDIO]; 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_select __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 *)); void audio_init_record __P((struct audio_softc *)); void audio_init_play __P((struct audio_softc *)); void audiostartr __P((struct audio_softc *)); void audiostartp __P((struct audio_softc *)); void audio_rint __P((void *)); void audio_pint __P((void *)); void audio_rpint __P((void *)); int audio_check_format __P((u_int *, u_int *)); int audio_calc_blksize __P((struct audio_softc *)); void audio_fill_silence __P((int, u_char *, int)); int audio_silence_copyout __P((struct audio_softc *, int, struct uio *)); void audio_alloc_auzero __P((struct audio_softc *, int)); void audio_printsc __P((struct audio_softc *)); void audioattach __P((int)); int audio_hardware_attach __P((struct audio_hw_if *, void *)); void audio_init_ring __P((struct audio_buffer *, int)); void audio_initbufs __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 *)); #ifdef AUDIO_DEBUG void audio_printsc(sc) struct audio_softc *sc; { printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if); printf("open %x mode %x\n", sc->sc_open, sc->sc_mode); printf("rchan %x wchan %x ", sc->sc_rchan, sc->sc_wchan); printf("rring blk %x pring nblk %x\n", sc->rr.nblk, sc->pr.nblk); printf("rbus %x pbus %x ", sc->sc_rbus, sc->sc_pbus); printf("blksz %d sib %d ", sc->sc_blksize, sc->sc_smpl_in_blk); printf("sp50ms %d backlog %d\n", sc->sc_50ms, sc->sc_backlog); printf("hiwat %d lowat %d rblks %d\n", sc->sc_hiwat, sc->sc_lowat, sc->sc_rblks); } #endif void audioattach(num) int num; { } /* * Called from hardware driver. */ int audio_hardware_attach(hwp, hdlp) struct audio_hw_if *hwp; void *hdlp; { struct audio_softc *sc; if (naudio >= NAUDIO) { DPRINTF(("audio_hardware_attach: not enough audio devices: %d > %d\n", naudio, NAUDIO)); return(EINVAL); } /* * Malloc a softc for the device */ /* XXX Find the first free slot */ audio_softc[naudio] = malloc(sizeof(struct audio_softc), M_DEVBUF, M_WAITOK); sc = audio_softc[naudio]; bzero(sc, sizeof(struct audio_softc)); /* XXX too paranoid? */ if (hwp->open == 0 || hwp->close == 0 || hwp->set_in_sr == 0 || hwp->get_in_sr == 0 || hwp->set_out_sr == 0 || hwp->get_out_sr == 0 || hwp->set_format == 0 || hwp->get_encoding == 0 || hwp->get_precision == 0 || hwp->set_channels == 0 || hwp->get_channels == 0 || hwp->round_blocksize == 0 || hwp->set_out_port == 0 || hwp->get_out_port == 0 || hwp->set_in_port == 0 || hwp->get_in_port == 0 || hwp->commit_settings == 0 || hwp->start_output == 0 || hwp->start_input == 0 || hwp->halt_output == 0 || hwp->halt_input == 0 || hwp->cont_output == 0 || hwp->cont_input == 0 || hwp->getdev == 0 || hwp->set_port == 0 || hwp->get_port == 0 || hwp->query_devinfo == 0) return(EINVAL); sc->hw_if = hwp; sc->hw_hdl = hdlp; /* * Alloc DMA play and record buffers */ sc->rr.bp = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK); if (sc->rr.bp == 0) { return (ENOMEM); } sc->pr.bp = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK); if (sc->pr.bp == 0) { free(sc->rr.bp, M_DEVBUF); return (ENOMEM); } /* * Set default softc params */ sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_LINEAR; /* * Return the audio unit number */ hwp->audio_unit = naudio++; #ifdef AUDIO_DEBUG printf("audio: unit %d attached\n", hwp->audio_unit); #endif return(0); } int audio_hardware_detach(hwp) struct audio_hw_if *hwp; { struct audio_softc *sc; #ifdef DIAGNOSTIC if (!hwp) panic("audio_hardware_detach: null hwp"); if (hwp->audio_unit > naudio) panic("audio_hardware_detach: invalid audio unit"); #endif sc = audio_softc[hwp->audio_unit]; if (hwp != sc->hw_if) return(EINVAL); if (sc->sc_open != 0) return(EBUSY); sc->hw_if = 0; /* Free audio buffers */ free(sc->rr.bp, M_DEVBUF); free(sc->pr.bp, M_DEVBUF); free(sc, M_DEVBUF); audio_softc[hwp->audio_unit] = NULL; return(0); } int audioopen(dev, flags, ifmt, p) dev_t dev; int flags, ifmt; struct proc *p; { switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_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)); 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 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 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: return (audio_ioctl(dev, cmd, addr, flag, p)); case MIXER_DEVICE: return (mixer_ioctl(dev, cmd, addr, flag, p)); default: return (ENXIO); } } int audioselect(dev, events, p) dev_t dev; int events; struct proc *p; { switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: return (audio_select(dev, events, p)); 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 MIXER_DEVICE: return (ENODEV); default: return (ENXIO); } } /* * Audio driver */ void audio_init_ring(rp, blksize) struct audio_buffer *rp; int blksize; { int nblk = AU_RING_SIZE / blksize; rp->ep = rp->bp + nblk * blksize; rp->hp = rp->tp = rp->bp; rp->maxblk = nblk; rp->nblk = 0; rp->cb_drops = 0; rp->cb_pdrops = 0; } void audio_initbufs(sc) struct audio_softc *sc; { int nblk = AU_RING_SIZE / sc->sc_blksize; audio_init_ring(&sc->rr, sc->sc_blksize); audio_init_ring(&sc->pr, sc->sc_blksize); sc->sc_lowat = nblk / 2; sc->sc_hiwat = nblk; } static __inline int audio_sleep_timo(chan, label, timo) int *chan; char *label; int timo; { int st; if (!label) label = "audio"; *chan = 1; st = (tsleep(chan, PWAIT | PCATCH, label, timo)); *chan = 0; if (st != 0) { DPRINTF(("audio_sleep: %d\n", st)); } return (st); } static __inline int audio_sleep(chan, label) int *chan; char *label; { return audio_sleep_timo(chan, label, 0); } static __inline void audio_wakeup(chan) int *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 s, error; struct audio_hw_if *hw; if (unit >= NAUDIO || !audio_softc[unit]) { DPRINTF(("audio_open: invalid device unit - %d\n", unit)); return (ENODEV); } sc = audio_softc[unit]; hw = sc->hw_if; DPRINTF(("audio_open: dev=0x%x flags=0x%x sc=0x%x hdl=0x%x\n", dev, flags, sc, sc->hw_hdl)); if (hw == 0) /* Hardware has not attached to us... */ return (ENXIO); if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0) /* XXX use flags */ return (EBUSY); if ((error = hw->open(dev, flags)) != 0) return (error); if (flags&FREAD) sc->sc_open |= AUOPEN_READ; if (flags&FWRITE) sc->sc_open |= AUOPEN_WRITE; /* * 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 */ hw->set_format(sc->hw_hdl, AUDIO_ENCODING_ULAW, 8); hw->set_in_sr(sc->hw_hdl, 8000); hw->set_out_sr(sc->hw_hdl, 8000); hw->set_channels(sc->hw_hdl, 1); sc->sc_pencoding = sc->sc_rencoding = AUDIO_ENCODING_ULAW; } /* * 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. */ #ifdef DIAGNOSTIC if (hw->get_precision(sc->hw_hdl) == 0) panic("audio_open: hardware driver returned 0 for get_precision"); #endif sc->sc_50ms = 50 * hw->get_out_sr(sc->hw_hdl) / 1000; sc->sc_blksize = audio_calc_blksize(sc); audio_alloc_auzero(sc, sc->sc_blksize); sc->sc_smpl_in_blk = sc->sc_blksize / (hw->get_precision(sc->hw_hdl) / NBBY); audio_initbufs(sc); sc->sc_backlog = audio_backlog; DPRINTF(("audio_open: rr.bp=%x-%x pr.bp=%x-%x\n", sc->rr.bp, sc->rr.ep, sc->pr.bp, sc->pr.ep)); hw->commit_settings(sc->hw_hdl); s = splaudio(); /* nothing read or written yet */ sc->sc_rseek = 0; sc->sc_wseek = 0; sc->sc_rchan = 0; sc->sc_wchan = 0; sc->sc_rbus = 0; sc->sc_pbus = 0; if ((flags & FWRITE) != 0) { audio_init_play(sc); /* audio_pint(sc); ??? */ } if ((flags & FREAD) != 0) { /* Play takes precedence if HW is half-duplex */ if (hw->full_duplex || ((flags & FWRITE) == 0)) { audio_init_record(sc); /* audiostartr(sc); don't start recording until read */ } } if (ISDEVAUDIO(dev)) { /* if open only for read or only for write, then set specific mode */ if ((flags & (FWRITE|FREAD)) == FWRITE) { sc->sc_mode = AUMODE_PLAY; sc->pr.cb_pause = 0; sc->rr.cb_pause = 1; audiostartp(sc); } else if ((flags & (FWRITE|FREAD)) == FREAD) { sc->sc_mode = AUMODE_RECORD; sc->rr.cb_pause = 0; sc->pr.cb_pause = 1; audiostartr(sc); } } /* Play all sample, and don't pad short writes by default */ sc->sc_mode |= AUMODE_PLAY_ALL; splx(s); return (0); } /* * Must be called from task context. */ void audio_init_record(sc) struct audio_softc *sc; { int s = splaudio(); sc->sc_mode |= AUMODE_RECORD; if (sc->hw_if->speaker_ctl && (!sc->hw_if->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_mode |= AUMODE_PLAY; sc->sc_rblks = sc->sc_wblks = 0; if (sc->hw_if->speaker_ctl) sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON); splx(s); } int audio_drain(sc) struct audio_softc *sc; { int error; while (sc->pr.nblk > 0) { DPRINTF(("audio_drain: nblk=%d\n", sc->pr.nblk)); /* * XXX * When the process is exiting, it ignores all signals and * we can't interrupt this sleep, so we set a 1-minute * timeout. */ error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 60*hz); if (error) return (error); } return (0); } /* * Close an audio chip. */ /* ARGSUSED */ int audio_close(dev, flags, ifmt, p) dev_t dev; int flags, ifmt; struct proc *p; { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_softc[unit]; struct audio_hw_if *hw = sc->hw_if; int s; DPRINTF(("audio_close: unit=%d\n", unit)); /* * Block until output drains, but allow ^C interrupt. */ sc->sc_lowat = 0; /* avoid excessive wakeups */ s = splaudio(); /* * If there is pending output, let it drain (unless * the output is paused). */ if (sc->sc_pbus && sc->pr.nblk > 0 && !sc->pr.cb_pause) { if (!audio_drain(sc) && hw->drain) (void)hw->drain(sc->hw_hdl); } hw->close(sc->hw_hdl); if (flags&FREAD) sc->sc_open &= ~AUOPEN_READ; if (flags&FWRITE) sc->sc_open &= ~AUOPEN_WRITE; sc->sc_async = 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_softc[unit]; struct audio_hw_if *hw = sc->hw_if; struct audio_buffer *cb = &sc->rr; u_char *hp; int blocksize = sc->sc_blksize; int error, s; DPRINTF(("audio_read: cc=%d mode=%d rblks=%d\n", uio->uio_resid, sc->sc_mode, sc->sc_rblks)); if (uio->uio_resid == 0) return (0); if (uio->uio_resid < blocksize) return (EINVAL); /* First sample we'll read in sample space */ sc->sc_rseek = cb->au_stamp - AU_RING_LEN(cb); /* * If hardware is half-duplex and currently playing, return * silence blocks based on the number of blocks we have output. */ if ((!hw->full_duplex) && (sc->sc_mode & AUMODE_PLAY)) { do { s = splaudio(); while (sc->sc_rblks <= 0) { DPRINTF(("audio_read: sc_rblks=%d\n", sc->sc_rblks)); if (ioflag & IO_NDELAY) { splx(s); return (EWOULDBLOCK); } error = audio_sleep(&sc->sc_rchan, "aud hr"); if (error) { splx(s); return (error); } } splx(s); error = audio_silence_copyout(sc, blocksize, uio); if (error) break; s = splaudio(); --sc->sc_rblks; splx(s); } while (uio->uio_resid >= blocksize); return (error); } error = 0; do { while (cb->nblk <= 0) { if (ioflag & IO_NDELAY) { error = EWOULDBLOCK; return (error); } s = splaudio(); if (!sc->sc_rbus) audiostartr(sc); error = audio_sleep(&sc->sc_rchan, "aud rd"); splx(s); if (error) return (error); } hp = cb->hp; if (hw->sw_decode) hw->sw_decode(sc->hw_hdl, sc->sc_rencoding, hp, blocksize); error = uiomove(hp, blocksize, uio); if (error) break; hp += blocksize; if (hp >= cb->ep) hp = cb->bp; cb->hp = hp; --cb->nblk; } while (uio->uio_resid >= blocksize); return (error); } void audio_clear(sc) struct audio_softc *sc; { int s = splaudio(); if (sc->sc_rbus || sc->sc_pbus) { sc->hw_if->halt_output(sc->hw_hdl); sc->hw_if->halt_input(sc->hw_hdl); sc->sc_rbus = 0; sc->sc_pbus = 0; } AU_RING_INIT(&sc->rr); AU_RING_INIT(&sc->pr); sc->sc_rblks = sc->sc_wblks = 0; splx(s); } int audio_calc_blksize(sc) struct audio_softc *sc; { struct audio_hw_if *hw = sc->hw_if; int bs; bs = hw->get_out_sr(sc->hw_hdl) * audio_blk_ms / 1000; if (bs == 0) bs = 1; bs *= hw->get_channels(sc->hw_hdl); bs *= hw->get_precision(sc->hw_hdl) / NBBY; if (bs > AU_RING_SIZE/2) bs = AU_RING_SIZE/2; bs = hw->round_blocksize(sc->hw_hdl, bs); if (bs > AU_RING_SIZE) bs = AU_RING_SIZE; return(bs); } void audio_fill_silence(encoding, p, n) int encoding; u_char *p; int n; { u_int auzero; u_char *q; switch (encoding) { case AUDIO_ENCODING_ULAW: auzero = 0x7f; break; case AUDIO_ENCODING_ALAW: auzero = 0x55; break; case AUDIO_ENCODING_ADPCM: /* is this right XXX */ case AUDIO_ENCODING_PCM8: case AUDIO_ENCODING_PCM16: default: auzero = 0; /* fortunately this works for both 8 and 16 bits */ break; } q = p; while (--n >= 0) *q++ = auzero; } #define NSILENCE 128 /* An arbitrary even constant >= 2 */ int audio_silence_copyout(sc, n, uio) struct audio_softc *sc; int n; struct uio *uio; { struct iovec *iov; int error = 0; u_char zerobuf[NSILENCE]; int k; audio_fill_silence(sc->sc_rencoding, zerobuf, NSILENCE); while (n > 0 && uio->uio_resid) { iov = uio->uio_iov; if (iov->iov_len == 0) { uio->uio_iov++; uio->uio_iovcnt--; continue; } k = min(min(n, iov->iov_len), NSILENCE); switch (uio->uio_segflg) { case UIO_USERSPACE: error = copyout(zerobuf, iov->iov_base, k); if (error) return (error); break; case UIO_SYSSPACE: bcopy(zerobuf, iov->iov_base, k); break; } iov->iov_base += k; iov->iov_len -= k; uio->uio_resid -= k; uio->uio_offset += k; n -= k; } return (error); } void audio_alloc_auzero(sc, bs) struct audio_softc *sc; int bs; { struct audio_hw_if *hw = sc->hw_if; if (sc->auzero_block) free(sc->auzero_block, M_DEVBUF); sc->auzero_block = malloc(bs, M_DEVBUF, M_WAITOK); #ifdef DIAGNOSTIC if (sc->auzero_block == 0) { panic("audio_alloc_auzero: malloc auzero_block failed"); } #endif audio_fill_silence(sc->sc_pencoding, sc->auzero_block, bs); if (hw->sw_encode) hw->sw_encode(sc->hw_hdl, sc->sc_pencoding, sc->auzero_block, bs); } int audio_write(dev, uio, ioflag) dev_t dev; struct uio *uio; int ioflag; { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_softc[unit]; struct audio_hw_if *hw = sc->hw_if; struct audio_buffer *cb = &sc->pr; u_char *tp; int error, s, cc; int blocksize = sc->sc_blksize; DPRINTF(("audio_write: cc=%d hiwat=%d\n", uio->uio_resid, sc->sc_hiwat)); /* * If half-duplex and currently recording, throw away data. */ if (!hw->full_duplex && (sc->sc_mode & AUMODE_RECORD)) { uio->uio_offset += uio->uio_resid; uio->uio_resid = 0; DPRINTF(("audio_write: half-dpx read busy\n")); return (0); } error = 0; while (uio->uio_resid > 0) { if (cb->fill > 0) { if (sc->sc_pbus == 0) { /* playing has stopped, ignore fill */ cb->fill = 0; } else { /* Write samples in the silence fill space. * We don't know where the DMA is * happening in the buffer, but if we * are lucky we will fill the buffer before * playing has reached the point we move to. * If we are unlucky some sample will * not be played. */ cc = min(cb->fill, uio->uio_resid); error = uiomove(cb->otp, cc, uio); if (error == 0) { if (hw->sw_encode) hw->sw_encode(sc->hw_hdl, sc->sc_pencoding, cb->otp, cc); cb->fill -= cc; cb->otp += cc; } continue; } } if (cb->nblk >= sc->sc_hiwat) { do { DPRINTF(("audio_write: nblk=%d hiwat=%d lowat=%d\n", cb->nblk, sc->sc_hiwat, sc->sc_lowat)); if (ioflag & IO_NDELAY) return (EWOULDBLOCK); error = audio_sleep(&sc->sc_wchan, "aud wr"); if (error) return (error); } while (cb->nblk >= sc->sc_lowat); } #if 0 if (cb->nblk == 0 && cb->maxblk > sc->sc_backlog && uio->uio_resid <= blocksize && (cb->au_stamp - sc->sc_wseek) > sc->sc_50ms) { /* * the write is 'small', the buffer is empty * and we have been silent for at least 50ms * so we might be dealing with an application * that writes frames synchronously with * reading them. If so, we need an output * backlog to cover scheduling delays or * there will be gaps in the sound output. * Also take this opportunity to reset the * buffer pointers in case we ended up on * a bad boundary (odd byte, blksize bytes * from end, etc.). */ DPRINTF(("audiowrite: set backlog %d\n", sc->sc_backlog)); s = splaudio(); cb->hp = cb->bp; cb->nblk = sc->sc_backlog; cb->tp = cb->hp + sc->sc_backlog * blocksize; splx(s); audio_fill_silence(sc->sc_pencoding, cb->hp, sc->sc_backlog * blocksize); } #endif /* Calculate sample number of first sample in block we write */ s = splaudio(); sc->sc_wseek = AU_RING_LEN(cb) + cb->au_stamp; splx(s); tp = cb->tp; cc = uio->uio_resid; #ifdef AUDIO_DEBUG if (audiodebug > 1) { int left = cb->ep - tp; Dprintf("audio_write: cc=%d tp=%p bs=%d nblk=%d left=%d\n", cc, tp, blocksize, cb->nblk, left); } #endif #ifdef DIAGNOSTIC { int towrite = (cc < blocksize)?cc:blocksize; /* check for an overwrite. Should never happen */ if ((tp + towrite) > cb->ep) { DPRINTF(("audio_write: overwrite tp=%p towrite=%d ep=0x%x bs=%d\n", tp, towrite, cb->ep, blocksize)); printf("audio_write: overwrite tp=%p towrite=%d ep=%p\n", tp, towrite, cb->ep); tp = cb->bp; } } #endif if (cc < blocksize) { error = uiomove(tp, cc, uio); if (error == 0) { /* fill with audio silence */ tp += cc; cc = blocksize - cc; cb->fill = cc; cb->otp = tp; audio_fill_silence(sc->sc_pencoding, tp, cc); DPRINTF(("audio_write: auzero 0x%x %d 0x%x\n", tp, cc, *(int *)tp)); tp += cc; } } else { error = uiomove(tp, blocksize, uio); if (error == 0) { tp += blocksize; } } if (error) { #ifdef AUDIO_DEBUG printf("audio_write:(1) uiomove failed %d; cc=%d tp=%p bs=%d\n", error, cc, tp, blocksize); #endif break; } if (hw->sw_encode) hw->sw_encode(sc->hw_hdl, sc->sc_pencoding, cb->tp, blocksize); /* wrap the ring buffer if at end */ s = splaudio(); if ((sc->sc_mode & AUMODE_PLAY_ALL) == 0 && sc->sc_wblks) /* * discard the block if we sent out a silence * packet that hasn't yet been countered * by user data. (They must supply enough * data to catch up to "real time"). */ sc->sc_wblks--; else { if (tp >= cb->ep) tp = cb->bp; cb->tp = tp; ++cb->nblk; /* account for buffer filled */ /* * If output isn't active, start it up. */ if (sc->sc_pbus == 0) audiostartp(sc); } splx(s); } return (error); } int audio_ioctl(dev, cmd, addr, flag, p) dev_t dev; int cmd; caddr_t addr; int flag; struct proc *p; { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_softc[unit]; struct audio_hw_if *hw = sc->hw_if; int error = 0, s; DPRINTF(("audio_ioctl(%d,'%c',%d)\n", IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff)); switch (cmd) { case FIOASYNC: if (*(int *)addr) { if (sc->sc_async) return (EBUSY); sc->sc_async = p; } else sc->sc_async = 0; break; case AUDIO_FLUSH: DPRINTF(("AUDIO_FLUSH\n")); audio_clear(sc); s = splaudio(); if ((sc->sc_mode & AUMODE_PLAY) && (sc->sc_pbus == 0)) audiostartp(sc); /* Again, play takes precedence on half-duplex hardware */ if ((sc->sc_mode & AUMODE_RECORD) && (hw->full_duplex || ((sc->sc_mode & AUMODE_PLAY) == 0))) 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->rr.cb_drops; break; case AUDIO_PERROR: *(int *)addr = sc->pr.cb_drops; break; /* * How many samples will elapse until mike hears the first * sample of what we last wrote? */ case AUDIO_WSEEK: s = splaudio(); *(u_long *)addr = sc->sc_wseek - sc->pr.au_stamp + AU_RING_LEN(&sc->rr); splx(s); break; case AUDIO_SETINFO: DPRINTF(("AUDIO_SETINFO\n")); error = audiosetinfo(sc, (struct audio_info *)addr); break; case AUDIO_GETINFO: DPRINTF(("AUDIO_GETINFO\n")); error = audiogetinfo(sc, (struct audio_info *)addr); break; case AUDIO_DRAIN: DPRINTF(("AUDIO_DRAIN\n")); error = audio_drain(sc); if (!error && hw->drain) error = hw->drain(sc->hw_hdl); break; case AUDIO_GETDEV: DPRINTF(("AUDIO_GETDEV\n")); error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_GETENC: DPRINTF(("AUDIO_GETENC\n")); error = hw->query_encoding(sc->hw_hdl, (struct audio_encoding *)addr); break; case AUDIO_GETFD: DPRINTF(("AUDIO_GETFD\n")); *(int *)addr = hw->full_duplex; break; case AUDIO_SETFD: DPRINTF(("AUDIO_SETFD\n")); error = hw->setfd(sc->hw_hdl, *(int *)addr); break; default: DPRINTF(("audio_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_select(dev, rw, p) dev_t dev; int rw; struct proc *p; { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_softc[unit]; int s = splaudio(); #if 0 DPRINTF(("audio_select: rw=%d mode=%d rblks=%d rr.nblk=%d\n", rw, sc->sc_mode, sc->sc_rblks, sc->rr.nblk)); #endif switch (rw) { case FREAD: if (sc->sc_mode & AUMODE_PLAY) { if (sc->sc_rblks > 0) { splx(s); return (1); } } else if (sc->rr.nblk > 0) { splx(s); return (1); } selrecord(p, &sc->sc_rsel); break; case FWRITE: /* * Can write if we're recording because it gets preempted. * Otherwise, can write when below low water. * XXX this won't work right if we're in * record mode -- we need to note that a write * select has happed and flip the speaker. * * XXX The above XXX-comment is SoundBlaster-dependent, * right? Or maybe specific to half-duplex devices? */ if (sc->sc_mode & AUMODE_RECORD || sc->pr.nblk < sc->sc_lowat) { splx(s); return (1); } selrecord(p, &sc->sc_wsel); break; } splx(s); return (0); } int audio_mmap(dev, off, prot) dev_t dev; int off, prot; { /* XXX placeholder */ return (-1); } void audiostartr(sc) struct audio_softc *sc; { int error; DPRINTF(("audiostartr: tp=%p\n", sc->rr.tp)); error = sc->hw_if->start_input(sc->hw_hdl, sc->rr.tp, sc->sc_blksize, audio_rint, (void *)sc); if (error) { DPRINTF(("audiostartr failed: %d\n", error)); audio_clear(sc); } else sc->sc_rbus = 1; } void audiostartp(sc) struct audio_softc *sc; { int error; DPRINTF(("audiostartp: hp=0x%x nblk=%d\n", sc->pr.hp, sc->pr.nblk)); if (sc->pr.nblk > 0) { u_char *hp = sc->pr.hp; error = sc->hw_if->start_output(sc->hw_hdl, hp, sc->sc_blksize, audio_rpint, (void *)sc); if (error) { DPRINTF(("audiostartp: failed: %d\n", error)); } else { sc->sc_pbus = 1; hp += sc->sc_blksize; if (hp >= sc->pr.ep) hp = sc->pr.bp; sc->pr.hp = hp; } } } /* * Use this routine as DMA callback if we played user data. We need to * account for user data and silence separately. */ void audio_rpint(v) void *v; { struct audio_softc *sc = v; sc->pr.nblk--; audio_pint(v); /* 'twas a real audio block */ } /* * 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; u_char *hp; int cc = sc->sc_blksize; struct audio_hw_if *hw = sc->hw_if; struct audio_buffer *cb = &sc->pr; int error; /* * XXX * if there is only one buffer in the ring, this test * always fails and the output is always silence after the * first block. */ if (cb->nblk > 0) { hp = cb->hp; if (cb->cb_pause) { cb->cb_pdrops++; #ifdef AUDIO_DEBUG if (audiodebug > 1) Dprintf("audio_pint: paused %d\n", cb->cb_pdrops); #endif goto psilence; } else { #ifdef AUDIO_DEBUG if (audiodebug > 1) Dprintf("audio_pint: hp=0x%x cc=%d\n", hp, cc); #endif error = hw->start_output(sc->hw_hdl, hp, cc, audio_rpint, (void *)sc); if (error) { DPRINTF(("audio_pint restart failed: %d\n", error)); audio_clear(sc); } else { hp += cc; if (hp >= cb->ep) hp = cb->bp; cb->hp = hp; cb->au_stamp += sc->sc_smpl_in_blk; ++sc->sc_rblks; } } } else { cb->cb_drops++; #ifdef AUDIO_DEBUG if (audiodebug > 1) Dprintf("audio_pint: drops=%d auzero %d 0x%x\n", cb->cb_drops, cc, *(int *)sc->auzero_block); #endif psilence: error = hw->start_output(sc->hw_hdl, sc->auzero_block, cc, audio_pint, (void *)sc); if (error) { DPRINTF(("audio_pint zero failed: %d\n", error)); audio_clear(sc); } else ++sc->sc_wblks; } #ifdef AUDIO_DEBUG DPRINTF(("audio_pint: mode=%d pause=%d nblk=%d lowat=%d\n", sc->sc_mode, cb->cb_pause, cb->nblk, sc->sc_lowat)); #endif if ((sc->sc_mode & AUMODE_PLAY) && !cb->cb_pause) { if (cb->nblk <= sc->sc_lowat) { audio_wakeup(&sc->sc_wchan); selwakeup(&sc->sc_wsel); if (sc->sc_async) psignal(sc->sc_async, SIGIO); } } /* * XXX * possible to return one or more "phantom blocks" now. * Only in half duplex? */ if (hw->full_duplex) { audio_wakeup(&sc->sc_rchan); selwakeup(&sc->sc_rsel); if (sc->sc_async) psignal(sc->sc_async, 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; u_char *tp; int cc = sc->sc_blksize; struct audio_hw_if *hw = sc->hw_if; struct audio_buffer *cb = &sc->rr; int error; tp = cb->tp; if (cb->cb_pause) { cb->cb_pdrops++; DPRINTF(("audio_rint: pdrops %d\n", cb->cb_pdrops)); } else { tp += cc; if (tp >= cb->ep) tp = cb->bp; if (++cb->nblk < cb->maxblk) { #ifdef AUDIO_DEBUG if (audiodebug > 1) Dprintf("audio_rint: tp=%p cc=%d\n", tp, cc); #endif error = hw->start_input(sc->hw_hdl, tp, cc, audio_rint, (void *)sc); if (error) { DPRINTF(("audio_rint: start failed: %d\n", error)); audio_clear(sc); } cb->au_stamp += sc->sc_smpl_in_blk; } else { /* * XXX * How do we count dropped input samples due to overrun? * Start a "dummy DMA transfer" when the input ring buffer * is full and count # of these? Seems pretty lame to * me, but how else are we going to do this? */ cb->cb_drops++; sc->sc_rbus = 0; DPRINTF(("audio_rint: drops %d\n", cb->cb_drops)); } cb->tp = tp; audio_wakeup(&sc->sc_rchan); selwakeup(&sc->sc_rsel); if (sc->sc_async) psignal(sc->sc_async, SIGIO); } } int audio_check_format(encodingp, precisionp) u_int *encodingp, *precisionp; { if (*encodingp == AUDIO_ENCODING_LINEAR) switch (*precisionp) { case 8: *encodingp = AUDIO_ENCODING_PCM8; return (0); case 16: *encodingp = AUDIO_ENCODING_PCM16; return (0); default: return (EINVAL); } switch (*encodingp) { case AUDIO_ENCODING_ULAW: case AUDIO_ENCODING_ALAW: case AUDIO_ENCODING_PCM8: case AUDIO_ENCODING_ADPCM: if (*precisionp != 8) return (EINVAL); break; case AUDIO_ENCODING_PCM16: if (*precisionp != 16) return (EINVAL); break; default: return (EINVAL); } return (0); } int audiosetinfo(sc, ai) struct audio_softc *sc; struct audio_info *ai; { struct audio_prinfo *r = &ai->record, *p = &ai->play; int cleared = 0, init = 0; int bsize, error = 0; struct audio_hw_if *hw = sc->hw_if; mixer_ctrl_t ct; int s; if (hw == 0) /* HW has not attached */ return(ENXIO); if (p->sample_rate != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_out_sr(sc->hw_hdl, p->sample_rate); if (error) return(error); sc->sc_50ms = 50 * hw->get_out_sr(sc->hw_hdl) / 1000; init = 1; } if (r->sample_rate != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_in_sr(sc->hw_hdl, r->sample_rate); if (error) return(error); sc->sc_50ms = 50 * hw->get_in_sr(sc->hw_hdl) / 1000; init = 1; } if (p->encoding != ~0 || p->precision != ~0) { if (!cleared) audio_clear(sc); cleared = 1; if (p->encoding == ~0) p->encoding = hw->get_encoding(sc->hw_hdl); if (p->precision == ~0) p->precision = hw->get_precision(sc->hw_hdl); error = audio_check_format(&p->encoding, &p->precision); if (error) return(error); error = hw->set_format(sc->hw_hdl, p->encoding, p->precision); if (error) return(error); sc->sc_pencoding = p->encoding; init = 1; } if (r->encoding != ~0 || r->precision != ~0) { if (!cleared) audio_clear(sc); cleared = 1; if (r->encoding == ~0) r->encoding = hw->get_encoding(sc->hw_hdl); if (r->precision == ~0) r->precision = hw->get_precision(sc->hw_hdl); error = audio_check_format(&r->encoding, &r->precision); if (error) return(error); error = hw->set_format(sc->hw_hdl, r->encoding, r->precision); if (error) return(error); sc->sc_rencoding = r->encoding; init = 1; } if (p->channels != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_channels(sc->hw_hdl, p->channels); if (error) return(error); init = 1; } if (r->channels != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_channels(sc->hw_hdl, r->channels); if (error) return(error); init = 1; } if (p->port != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_out_port(sc->hw_hdl, p->port); if (error) return(error); } if (r->port != ~0) { if (!cleared) audio_clear(sc); cleared = 1; error = hw->set_in_port(sc->hw_hdl, r->port); if (error) return(error); } if (p->gain != ~0) { ct.dev = hw->get_out_port(sc->hw_hdl); ct.type = AUDIO_MIXER_VALUE; ct.un.value.num_channels = 1; ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = p->gain; error = hw->set_port(sc->hw_hdl, &ct); if (error) return(error); } if (r->gain != ~0) { ct.dev = hw->get_in_port(sc->hw_hdl); ct.type = AUDIO_MIXER_VALUE; ct.un.value.num_channels = 1; ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = r->gain; error = hw->set_port(sc->hw_hdl, &ct); if (error) return(error); } if (p->pause != (u_char)~0) { sc->pr.cb_pause = p->pause; if (!p->pause) { s = splaudio(); audiostartp(sc); splx(s); } } if (r->pause != (u_char)~0) { sc->rr.cb_pause = r->pause; if (!r->pause) { s = splaudio(); audiostartr(sc); splx(s); } } if (ai->blocksize != ~0) { /* Block size specified explicitly. */ if (!cleared) audio_clear(sc); cleared = 1; if (ai->blocksize == 0) bsize = sc->sc_blksize; else if (ai->blocksize > AU_RING_SIZE/2) bsize = AU_RING_SIZE/2; else bsize = ai->blocksize; bsize = hw->round_blocksize(sc->hw_hdl, bsize); if (bsize > AU_RING_SIZE) bsize = AU_RING_SIZE; sc->sc_blksize = bsize; init = 1; } else if (init) { /* Block size calculated from other parameter changes. */ sc->sc_blksize = audio_calc_blksize(sc); } if (init) { audio_alloc_auzero(sc, sc->sc_blksize); sc->sc_smpl_in_blk = sc->sc_blksize / (hw->get_precision(sc->hw_hdl) / NBBY); audio_initbufs(sc); } if (ai->hiwat != ~0) { if ((unsigned)ai->hiwat > sc->pr.maxblk) ai->hiwat = sc->pr.maxblk; if (sc->sc_hiwat != 0) sc->sc_hiwat = ai->hiwat; } if (ai->lowat != ~0) { if ((unsigned)ai->lowat > sc->pr.maxblk) ai->lowat = sc->pr.maxblk; sc->sc_lowat = ai->lowat; } if (ai->backlog != ~0) { if ((unsigned)ai->backlog > (sc->pr.maxblk/2)) ai->backlog = sc->pr.maxblk/2; sc->sc_backlog = ai->backlog; } if (ai->mode != ~0) { if (!cleared) audio_clear(sc); cleared = 1; sc->sc_mode = ai->mode; if (sc->sc_mode & AUMODE_PLAY) { audio_init_play(sc); if (!hw->full_duplex) /* Play takes precedence */ sc->sc_mode &= ~(AUMODE_RECORD); } if (sc->sc_mode & AUMODE_RECORD) audio_init_record(sc); } error = hw->commit_settings(sc->hw_hdl); if (error) return (error); if (cleared) { s = splaudio(); if (sc->sc_mode & AUMODE_PLAY) audiostartp(sc); if (sc->sc_mode & AUMODE_RECORD) audiostartr(sc); splx(s); } return (0); } int audiogetinfo(sc, ai) struct audio_softc *sc; struct audio_info *ai; { struct audio_prinfo *r = &ai->record, *p = &ai->play; struct audio_hw_if *hw = sc->hw_if; mixer_ctrl_t ct; if (hw == 0) /* HW has not attached */ return(ENXIO); p->sample_rate = hw->get_out_sr(sc->hw_hdl); r->sample_rate = hw->get_in_sr(sc->hw_hdl); p->channels = r->channels = hw->get_channels(sc->hw_hdl); p->precision = r->precision = hw->get_precision(sc->hw_hdl); p->encoding = hw->get_encoding(sc->hw_hdl); r->encoding = hw->get_encoding(sc->hw_hdl); r->port = hw->get_in_port(sc->hw_hdl); p->port = hw->get_out_port(sc->hw_hdl); ct.dev = r->port; ct.type = AUDIO_MIXER_VALUE; ct.un.value.num_channels = 1; if (hw->get_port(sc->hw_hdl, &ct) == 0) r->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; else r->gain = AUDIO_MAX_GAIN/2; ct.dev = p->port; ct.un.value.num_channels = 1; if (hw->get_port(sc->hw_hdl, &ct) == 0) p->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; else p->gain = AUDIO_MAX_GAIN/2; p->pause = sc->pr.cb_pause; r->pause = sc->rr.cb_pause; p->error = sc->pr.cb_drops != 0; r->error = sc->rr.cb_drops != 0; p->open = ((sc->sc_open & AUOPEN_WRITE) != 0); r->open = ((sc->sc_open & AUOPEN_READ) != 0); p->samples = sc->pr.au_stamp - sc->pr.cb_pdrops; r->samples = sc->rr.au_stamp - sc->rr.cb_pdrops; p->seek = sc->sc_wseek; r->seek = sc->sc_rseek; ai->blocksize = sc->sc_blksize; ai->hiwat = sc->sc_hiwat; ai->lowat = sc->sc_lowat; ai->backlog = sc->sc_backlog; ai->mode = sc->sc_mode; return (0); } /* * Mixer driver */ int mixer_open(dev, flags, ifmt, p) dev_t dev; int flags, ifmt; struct proc *p; { int unit = AUDIOUNIT(dev); struct audio_softc *sc; struct audio_hw_if *hw; if (unit >= NAUDIO || !audio_softc[unit]) { DPRINTF(("mixer_open: invalid device unit - %d\n", unit)); return (ENODEV); } sc = audio_softc[unit]; hw = sc->hw_if; DPRINTF(("mixer_open: dev=%x flags=0x%x sc=0x%x\n", dev, flags, sc)); if (hw == 0) /* Hardware has not attached to us... */ return (ENXIO); return (0); } /* * Close a mixer device */ /* ARGSUSED */ int mixer_close(dev, flags, ifmt, p) dev_t dev; int flags, ifmt; struct proc *p; { DPRINTF(("mixer_close: unit %d\n", AUDIOUNIT(dev))); return (0); } int mixer_ioctl(dev, cmd, addr, flag, p) dev_t dev; int cmd; caddr_t addr; int flag; struct proc *p; { int unit = AUDIOUNIT(dev); struct audio_softc *sc = audio_softc[unit]; struct audio_hw_if *hw = sc->hw_if; int error = EINVAL; DPRINTF(("mixer_ioctl(%d,'%c',%d)\n", IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff)); switch (cmd) { case AUDIO_GETDEV: DPRINTF(("AUDIO_GETDEV\n")); error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_MIXER_DEVINFO: DPRINTF(("AUDIO_MIXER_DEVINFO\n")); error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr); break; case AUDIO_MIXER_READ: DPRINTF(("AUDIO_MIXER_READ\n")); error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr); break; case AUDIO_MIXER_WRITE: DPRINTF(("AUDIO_MIXER_WRITE\n")); error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr); if (error == 0) error = hw->commit_settings(sc->hw_hdl); break; default: error = EINVAL; break; } DPRINTF(("mixer_ioctl(%d,'%c',%d) result %d\n", IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error)); return (error); } #endif