/* $OpenBSD: audio.c,v 1.137 2015/07/28 21:04:28 ratchov Exp $ */ /* * Copyright (c) 2015 Alexandre Ratchov * * Permission to use, copy, modify, and distribute this software for any * purpose with or without fee is hereby granted, provided that the above * copyright notice and this permission notice appear in all copies. * * THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF * MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF * OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE. */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "audio.h" #include "wskbd.h" #ifdef AUDIO_DEBUG #define DPRINTF(...) \ do { \ if (audio_debug) \ printf(__VA_ARGS__); \ } while(0) #define DPRINTFN(n, ...) \ do { \ if (audio_debug > (n)) \ printf(__VA_ARGS__); \ } while(0) #else #define DPRINTF(...) do {} while(0) #define DPRINTFN(n, ...) do {} while(0) #endif #define DEVNAME(sc) ((sc)->dev.dv_xname) #define AUDIO_UNIT(n) (minor(n) & 0x0f) #define AUDIO_DEV(n) (minor(n) & 0xf0) #define AUDIO_DEV_SOUND 0 /* minor of /dev/sound0 */ #define AUDIO_DEV_MIXER 0x10 /* minor of /dev/mixer0 */ #define AUDIO_DEV_AUDIO 0x80 /* minor of /dev/audio0 */ #define AUDIO_DEV_AUDIOCTL 0xc0 /* minor of /dev/audioctl */ #define AUDIO_BUFSZ 65536 /* buffer size in bytes */ /* * dma buffer */ struct audio_buf { unsigned char *data; /* DMA memory block */ size_t datalen; /* size of DMA memory block */ size_t len; /* size of DMA FIFO */ size_t start; /* first byte used in the FIFO */ size_t used; /* bytes used in the FIFO */ size_t blksz; /* DMA block size */ struct selinfo sel; /* to record & wakeup poll(2) */ unsigned int pos; /* bytes transferred */ unsigned int xrun; /* bytes lost by xruns */ int blocking; /* read/write blocking */ }; #if NWSKBD > 0 struct wskbd_vol { int val; /* index of the value control */ int mute; /* index of the mute control */ int step; /* increment/decrement step */ int nch; /* channels in the value control */ int val_pending; /* pending change of val */ int mute_pending; /* pending mute toggles */ }; #endif /* * device structure */ struct audio_softc { struct device dev; struct audio_hw_if *ops; /* driver funcs */ void *arg; /* first arg to driver funcs */ int mode; /* bitmask of AUMODE_* */ int quiesce; /* device suspended */ struct audio_buf play, rec; unsigned int sw_enc; /* user exposed AUDIO_ENCODING_* */ unsigned int hw_enc; /* harware AUDIO_ENCODING_* */ unsigned int bits; /* bits per sample */ unsigned int bps; /* bytes-per-sample */ unsigned int msb; /* sample are MSB aligned */ unsigned int rate; /* rate in Hz */ unsigned int round; /* block size in frames */ unsigned int nblks; /* number of play blocks */ unsigned int pchan, rchan; /* number of channels */ unsigned char silence[4]; /* a sample of silence */ int pause; /* not trying to start DMA */ int active; /* DMA in process */ int offs; /* offset between play & rec dir */ void (*conv_enc)(unsigned char *, int); /* encode to native */ void (*conv_dec)(unsigned char *, int); /* decode to user */ #if NWSKBD > 0 struct wskbd_vol spkr, mic; struct task wskbd_task; int wskbd_taskset; #endif }; int audio_match(struct device *, void *, void *); void audio_attach(struct device *, struct device *, void *); int audio_activate(struct device *, int); int audio_detach(struct device *, int); void audio_pintr(void *); void audio_rintr(void *); #if NWSKBD > 0 void wskbd_mixer_init(struct audio_softc *); #endif const struct cfattach audio_ca = { sizeof(struct audio_softc), audio_match, audio_attach, audio_detach, audio_activate }; struct cfdriver audio_cd = { NULL, "audio", DV_DULL }; /* * This mutex protects data structures (including registers on the * sound-card) that are manipulated by both the interrupt handler and * syscall code-paths. * * Note that driver methods may sleep (e.g. in malloc); consequently the * audio layer calls them with the mutex unlocked. Driver methods are * responsible for locking the mutex when they manipulate data used by * the interrupt handler and interrupts may occur. * * Similarly, the driver is responsible for locking the mutex in its * interrupt handler and to call the audio layer call-backs (i.e. * audio_{p,r}int()) with the mutex locked. */ struct mutex audio_lock = MUTEX_INITIALIZER(IPL_AUDIO); #ifdef AUDIO_DEBUG /* * 0 - nothing, as if AUDIO_DEBUG isn't defined * 1 - initialisations & setup * 2 - blocks & interrupts */ int audio_debug = 1; #endif unsigned int audio_gcd(unsigned int a, unsigned int b) { unsigned int r; while (b > 0) { r = a % b; a = b; b = r; } return a; } int audio_buf_init(struct audio_softc *sc, struct audio_buf *buf, int dir) { if (sc->ops->round_buffersize) { buf->datalen = sc->ops->round_buffersize(sc->arg, dir, AUDIO_BUFSZ); } else buf->datalen = AUDIO_BUFSZ; if (sc->ops->allocm) { buf->data = sc->ops->allocm(sc->arg, dir, buf->datalen, M_DEVBUF, M_WAITOK); } else buf->data = malloc(buf->datalen, M_DEVBUF, M_WAITOK); if (buf->data == NULL) return ENOMEM; return 0; } void audio_buf_done(struct audio_softc *sc, struct audio_buf *buf) { if (sc->ops->freem) sc->ops->freem(sc->arg, buf->data, M_DEVBUF); else free(buf->data, M_DEVBUF, buf->datalen); } /* * return the reader pointer and the number of bytes available */ unsigned char * audio_buf_rgetblk(struct audio_buf *buf, size_t *rsize) { size_t count; count = buf->len - buf->start; if (count > buf->used) count = buf->used; *rsize = count; return buf->data + buf->start; } /* * discard "count" bytes at the start postion. */ void audio_buf_rdiscard(struct audio_buf *buf, size_t count) { #ifdef AUDIO_DEBUG if (count > buf->used) { panic("audio_buf_rdiscard: bad count = %zu\n", count); } #endif buf->used -= count; buf->start += count; if (buf->start >= buf->len) buf->start -= buf->len; } /* * advance the writer pointer by "count" bytes */ void audio_buf_wcommit(struct audio_buf *buf, size_t count) { #ifdef AUDIO_DEBUG if (count > (buf->len - buf->used)) { panic("audio_buf_wcommit: bad count = %zu\n", count); } #endif buf->used += count; } /* * get writer pointer and the number of bytes writable */ unsigned char * audio_buf_wgetblk(struct audio_buf *buf, size_t *rsize) { size_t end, avail, count; end = buf->start + buf->used; if (end >= buf->len) end -= buf->len; avail = buf->len - buf->used; count = buf->len - end; if (count > avail) count = avail; *rsize = count; return buf->data + end; } void audio_calc_sil(struct audio_softc *sc) { unsigned char *q; unsigned int s, i; int d, e; e = sc->sw_enc; #ifdef AUDIO_DEBUG switch (e) { case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_BE: break; default: printf("%s: unhandled play encoding %d\n", DEVNAME(sc), e); memset(sc->silence, 0, sc->bps); return; } #endif if (e == AUDIO_ENCODING_SLINEAR_BE || e == AUDIO_ENCODING_ULINEAR_BE) { d = -1; q = sc->silence + sc->bps - 1; } else { d = 1; q = sc->silence; } if (e == AUDIO_ENCODING_SLINEAR_LE || e == AUDIO_ENCODING_SLINEAR_BE) { s = 0; } else { s = 0x80000000; if (sc->msb) s >>= 32 - 8 * sc->bps; else s >>= 32 - sc->bits; } for (i = 0; i < sc->bps; i++) { *q = s; q += d; s >>= 8; } if (sc->conv_enc) sc->conv_enc(sc->silence, sc->bps); } void audio_fill_sil(struct audio_softc *sc, unsigned char *ptr, size_t count) { unsigned char *q, *p; size_t i, j; q = ptr; for (j = count / sc->bps; j > 0; j--) { p = sc->silence; for (i = sc->bps; i > 0; i--) *q++ = *p++; } } void audio_clear(struct audio_softc *sc) { if (sc->mode & AUMODE_PLAY) { sc->play.used = sc->play.start = 0; sc->play.pos = sc->play.xrun = 0; audio_fill_sil(sc, sc->play.data, sc->play.len); } if (sc->mode & AUMODE_RECORD) { sc->rec.used = sc->rec.start = 0; sc->rec.pos = sc->rec.xrun = 0; audio_fill_sil(sc, sc->rec.data, sc->rec.len); } } /* * called whenever a block is consumed by the driver */ void audio_pintr(void *addr) { struct audio_softc *sc = addr; unsigned char *ptr; size_t count; int error, nblk, todo; MUTEX_ASSERT_LOCKED(&audio_lock); if (!(sc->mode & AUMODE_PLAY) || !sc->active) { printf("%s: play interrupt but not playing\n", DEVNAME(sc)); return; } if (sc->quiesce) { DPRINTF("%s: quesced, skipping play intr\n", DEVNAME(sc)); return; } /* * check if record pointer wrapped, see explanation * in audio_rintr() */ if (sc->mode & AUMODE_RECORD) { sc->offs--; nblk = sc->rec.len / sc->rec.blksz; todo = -sc->offs; if (todo >= nblk) { todo -= todo % nblk; DPRINTFN(1, "%s: rec ptr wrapped, moving %d blocks\n", DEVNAME(sc), todo); while (todo-- > 0) audio_rintr(sc); } } sc->play.pos += sc->play.blksz; audio_fill_sil(sc, sc->play.data + sc->play.start, sc->play.blksz); audio_buf_rdiscard(&sc->play, sc->play.blksz); if (sc->play.used < sc->play.blksz) { DPRINTFN(1, "%s: play underrun\n", DEVNAME(sc)); sc->play.xrun += sc->play.blksz; audio_buf_wcommit(&sc->play, sc->play.blksz); } DPRINTFN(1, "%s: play intr, used -> %zu, start -> %zu\n", DEVNAME(sc), sc->play.used, sc->play.start); if (!sc->ops->trigger_output) { ptr = audio_buf_rgetblk(&sc->play, &count); error = sc->ops->start_output(sc->arg, ptr, sc->play.blksz, audio_pintr, (void *)sc); if (error) { printf("%s: play restart failed: %d\n", DEVNAME(sc), error); } } if (sc->play.used < sc->play.len) { DPRINTFN(1, "%s: play wakeup, chan = %d\n", DEVNAME(sc), sc->play.blocking); if (sc->play.blocking) { wakeup(&sc->play.blocking); sc->play.blocking = 0; } selwakeup(&sc->play.sel); } } /* * called whenever a block is produced by the driver */ void audio_rintr(void *addr) { struct audio_softc *sc = addr; unsigned char *ptr; size_t count; int error, nblk, todo; MUTEX_ASSERT_LOCKED(&audio_lock); if (!(sc->mode & AUMODE_RECORD) || !sc->active) { printf("%s: rec interrupt but not recording\n", DEVNAME(sc)); return; } if (sc->quiesce) { DPRINTF("%s: quesced, skipping rec intr\n", DEVNAME(sc)); return; } /* * Interrupts may be masked by other sub-systems during 320ms * and more. During such a delay the hardware doesn't stop * playing and the play buffer pointers may wrap, this can't be * detected and corrected by low level drivers. This makes the * record stream ahead of the play stream; this is detected as a * hardware anomaly by userland and cause programs to misbehave. * * We fix this by advancing play position by an integer count of * full buffers, so it reaches the record position. */ if (sc->mode & AUMODE_PLAY) { sc->offs++; nblk = sc->play.len / sc->play.blksz; todo = sc->offs; if (todo >= nblk) { todo -= todo % nblk; DPRINTFN(1, "%s: play ptr wrapped, moving %d blocks\n", DEVNAME(sc), todo); while (todo-- > 0) audio_pintr(sc); } } sc->rec.pos += sc->rec.blksz; audio_buf_wcommit(&sc->rec, sc->rec.blksz); if (sc->rec.used == sc->rec.len) { DPRINTFN(1, "%s: rec overrun\n", DEVNAME(sc)); sc->rec.xrun += sc->rec.blksz; audio_buf_rdiscard(&sc->rec, sc->rec.blksz); } DPRINTFN(1, "%s: rec intr, used -> %zu\n", DEVNAME(sc), sc->rec.used); if (!sc->ops->trigger_input) { ptr = audio_buf_wgetblk(&sc->rec, &count); error = sc->ops->start_input(sc->arg, ptr, sc->rec.blksz, audio_rintr, (void *)sc); if (error) { printf("%s: rec restart failed: %d\n", DEVNAME(sc), error); } } if (sc->rec.used > 0) { DPRINTFN(1, "%s: rec wakeup, chan = %d\n", DEVNAME(sc), sc->rec.blocking); if (sc->rec.blocking) { wakeup(&sc->rec.blocking); sc->rec.blocking = 0; } selwakeup(&sc->rec.sel); } } int audio_start_do(struct audio_softc *sc) { int error; struct audio_params p; unsigned char *ptr; size_t count; DPRINTFN(1, "%s: start play: " "start = %zu, used = %zu, " "len = %zu, blksz = %zu\n", DEVNAME(sc), sc->play.start, sc->play.used, sc->play.len, sc->play.blksz); DPRINTFN(1, "%s: start rec: " "start = %zu, used = %zu, " "len = %zu, blksz = %zu\n", DEVNAME(sc), sc->rec.start, sc->rec.used, sc->rec.len, sc->rec.blksz); error = 0; sc->offs = 0; if (sc->mode & AUMODE_PLAY) { if (sc->ops->trigger_output) { p.encoding = sc->hw_enc; p.precision = sc->bits; p.bps = sc->bps; p.msb = sc->msb; p.sample_rate = sc->rate; p.channels = sc->pchan; error = sc->ops->trigger_output(sc->arg, sc->play.data, sc->play.data + sc->play.len, sc->play.blksz, audio_pintr, (void *)sc, &p); } else { mtx_enter(&audio_lock); ptr = audio_buf_rgetblk(&sc->play, &count); error = sc->ops->start_output(sc->arg, ptr, sc->play.blksz, audio_pintr, (void *)sc); mtx_leave(&audio_lock); } if (error) printf("%s: failed to start playback\n", DEVNAME(sc)); } if (sc->mode & AUMODE_RECORD) { if (sc->ops->trigger_input) { p.encoding = sc->hw_enc; p.precision = sc->bits; p.bps = sc->bps; p.msb = sc->msb; p.sample_rate = sc->rate; p.channels = sc->rchan; error = sc->ops->trigger_input(sc->arg, sc->rec.data, sc->rec.data + sc->rec.len, sc->rec.blksz, audio_rintr, (void *)sc, &p); } else { mtx_enter(&audio_lock); ptr = audio_buf_wgetblk(&sc->rec, &count); error = sc->ops->start_input(sc->arg, ptr, sc->rec.blksz, audio_rintr, (void *)sc); mtx_leave(&audio_lock); } if (error) printf("%s: failed to start recording\n", DEVNAME(sc)); } return error; } int audio_stop_do(struct audio_softc *sc) { if (sc->mode & AUMODE_PLAY) sc->ops->halt_output(sc->arg); if (sc->mode & AUMODE_RECORD) sc->ops->halt_input(sc->arg); return 0; } int audio_start(struct audio_softc *sc) { sc->active = 1; sc->play.xrun = sc->play.pos = sc->rec.xrun = sc->rec.pos = 0; return audio_start_do(sc); } int audio_stop(struct audio_softc *sc) { int error; error = audio_stop_do(sc); if (error) return error; audio_clear(sc); sc->active = 0; return 0; } int audio_setpar(struct audio_softc *sc) { struct audio_params p, r; unsigned int nr, np, max, min, mult; int error; DPRINTF("%s: setpar: req enc=%d bits=%d, bps=%d, msb=%d " "rate=%d, pchan=%d, rchan=%d, round=%u, nblks=%d\n", DEVNAME(sc), sc->sw_enc, sc->bits, sc->bps, sc->msb, sc->rate, sc->pchan, sc->rchan, sc->round, sc->nblks); /* * AUDIO_ENCODING_SLINEAR and AUDIO_ENCODING_ULINEAR are not * used anymore, promote them to the _LE and _BE equivalents */ if (sc->sw_enc == AUDIO_ENCODING_SLINEAR) { #if BYTE_ORDER == LITTLE_ENDIAN sc->sw_enc = AUDIO_ENCODING_SLINEAR_LE; #else sc->sw_enc = AUDIO_ENCODING_SLINEAR_BE; #endif } if (sc->sw_enc == AUDIO_ENCODING_ULINEAR) { #if BYTE_ORDER == LITTLE_ENDIAN sc->sw_enc = AUDIO_ENCODING_ULINEAR_LE; #else sc->sw_enc = AUDIO_ENCODING_ULINEAR_BE; #endif } /* * check if requested parameters are in the allowed ranges */ if (sc->mode & AUMODE_PLAY) { if (sc->pchan < 1) sc->pchan = 1; if (sc->pchan > 64) sc->pchan = 64; } if (sc->mode & AUMODE_RECORD) { if (sc->rchan < 1) sc->rchan = 1; if (sc->rchan > 64) sc->rchan = 64; } switch (sc->sw_enc) { case AUDIO_ENCODING_ULAW: case AUDIO_ENCODING_ALAW: case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: break; default: sc->sw_enc = AUDIO_ENCODING_SLINEAR_LE; } if (sc->bits < 8) sc->bits = 8; if (sc->bits > 32) sc->bits = 32; if (sc->bps < 1) sc->bps = 1; if (sc->bps > 4) sc->bps = 4; if (sc->rate < 4000) sc->rate = 4000; if (sc->rate > 192000) sc->rate = 192000; /* * copy into struct audio_params, required by drivers */ p.encoding = r.encoding = sc->sw_enc; p.precision = r.precision = sc->bits; p.bps = r.bps = sc->bps; p.msb = r.msb = sc->msb; p.sample_rate = r.sample_rate = sc->rate; p.channels = sc->pchan; r.channels = sc->rchan; /* * set parameters */ error = sc->ops->set_params(sc->arg, sc->mode, sc->mode, &p, &r); if (error) return error; if (sc->mode == (AUMODE_PLAY | AUMODE_RECORD)) { if (p.encoding != r.encoding || p.precision != r.precision || p.bps != r.bps || p.msb != r.msb || p.sample_rate != r.sample_rate) { printf("%s: different play and record parameters " "returned by hardware\n", DEVNAME(sc)); return ENODEV; } } if (sc->mode & AUMODE_PLAY) { sc->hw_enc = p.encoding; sc->bits = p.precision; sc->bps = p.bps; sc->msb = p.msb; sc->rate = p.sample_rate; sc->pchan = p.channels; } if (sc->mode & AUMODE_RECORD) { sc->hw_enc = r.encoding; sc->bits = r.precision; sc->bps = r.bps; sc->msb = r.msb; sc->rate = r.sample_rate; sc->rchan = r.channels; } if (sc->rate == 0 || sc->bps == 0 || sc->bits == 0) { printf("%s: invalid parameters returned by hardware\n", DEVNAME(sc)); return ENODEV; } if (sc->ops->commit_settings) { error = sc->ops->commit_settings(sc->arg); if (error) return error; } /* * conversion from/to exotic/dead encoding, for drivers not supporting * linear */ switch (sc->hw_enc) { case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: sc->sw_enc = sc->hw_enc; sc->conv_dec = sc->conv_enc = NULL; break; case AUDIO_ENCODING_ULAW: #if BYTE_ORDER == LITTLE_ENDIAN sc->sw_enc = AUDIO_ENCODING_SLINEAR_LE; #else sc->sw_enc = AUDIO_ENCODING_SLINEAR_BE; #endif if (sc->bits == 8) { sc->conv_enc = slinear8_to_mulaw; sc->conv_dec = mulaw_to_slinear8; break; } else if (sc->bits == 24) { sc->conv_enc = slinear24_to_mulaw24; sc->conv_dec = mulaw24_to_slinear24; break; } sc->sw_enc = sc->hw_enc; sc->conv_dec = sc->conv_enc = NULL; break; default: printf("%s: setpar: enc = %d, bits = %d: emulation skipped\n", DEVNAME(sc), sc->hw_enc, sc->bits); sc->sw_enc = sc->hw_enc; sc->conv_dec = sc->conv_enc = NULL; } audio_calc_sil(sc); /* * get least multiplier of the number of frames per block */ if (sc->ops->round_blocksize) { mult = sc->ops->round_blocksize(sc->arg, 1); if (mult == 0) { printf("%s: 0x%x: bad block size multiplier\n", DEVNAME(sc), mult); return ENODEV; } } else mult = 1; DPRINTF("%s: hw block size multiplier: %u\n", DEVNAME(sc), mult); if (sc->mode & AUMODE_PLAY) { np = mult / audio_gcd(sc->pchan * sc->bps, mult); if (!(sc->mode & AUMODE_RECORD)) nr = np; DPRINTF("%s: play number of frames multiplier: %u\n", DEVNAME(sc), np); } if (sc->mode & AUMODE_RECORD) { nr = mult / audio_gcd(sc->rchan * sc->bps, mult); if (!(sc->mode & AUMODE_PLAY)) np = nr; DPRINTF("%s: record number of frames multiplier: %u\n", DEVNAME(sc), nr); } mult = nr * np / audio_gcd(nr, np); DPRINTF("%s: least common number of frames multiplier: %u\n", DEVNAME(sc), mult); /* * get minumum and maximum frames per block */ if (sc->mode & AUMODE_PLAY) { np = sc->play.datalen / (sc->pchan * sc->bps * 2); if (!(sc->mode & AUMODE_RECORD)) nr = np; } if (sc->mode & AUMODE_RECORD) { nr = sc->rec.datalen / (sc->rchan * sc->bps * 2); if (!(sc->mode & AUMODE_PLAY)) np = nr; } max = np < nr ? np : nr; max -= max % mult; min = sc->rate / 1000 + mult - 1; min -= min % mult; DPRINTF("%s: frame number range: %u..%u\n", DEVNAME(sc), min, max); if (max < min) { printf("%s: %u: bad max frame number\n", DEVNAME(sc), max); return EIO; } /* * adjust the frame per block to match our constraints */ sc->round += mult / 2; sc->round -= sc->round % mult; if (sc->round > max) sc->round = max; if (sc->round < min) sc->round = min; sc->round = sc->round; /* * set buffer size (number of blocks) */ if (sc->mode & AUMODE_PLAY) { sc->play.blksz = sc->round * sc->pchan * sc->bps; max = sc->play.datalen / sc->play.blksz; if (sc->nblks > max) sc->nblks = max; if (sc->nblks < 2) sc->nblks = 2; sc->play.len = sc->nblks * sc->play.blksz; sc->nblks = sc->nblks; } if (sc->mode & AUMODE_RECORD) { /* * for recording, buffer size is not the latency (it's * exactly one block), so let's get the maximum buffer * size of maximum reliability during xruns */ sc->rec.blksz = sc->round * sc->rchan * sc->bps; sc->rec.len = sc->rec.datalen; sc->rec.len -= sc->rec.datalen % sc->rec.blksz; } DPRINTF("%s: setpar: new enc=%d bits=%d, bps=%d, msb=%d " "rate=%d, pchan=%d, rchan=%d, round=%u, nblks=%d\n", DEVNAME(sc), sc->sw_enc, sc->bits, sc->bps, sc->msb, sc->rate, sc->pchan, sc->rchan, sc->round, sc->nblks); return 0; } int audio_setinfo(struct audio_softc *sc, struct audio_info *ai) { struct audio_prinfo *r = &ai->record, *p = &ai->play; int error; int set; /* * stop the device if requested to stop */ if (sc->mode != 0) { if (sc->mode & AUMODE_PLAY) { if (p->pause != (unsigned char)~0) sc->pause = p->pause; } if (sc->mode & AUMODE_RECORD) { if (r->pause != (unsigned char)~0) sc->pause = r->pause; } if (sc->pause) { if (sc->active) audio_stop(sc); } } /* * copy parameters into the softc structure */ set = 0; if (ai->play.encoding != ~0) { sc->sw_enc = ai->play.encoding; set = 1; } if (ai->play.precision != ~0) { sc->bits = ai->play.precision; set = 1; } if (ai->play.bps != ~0) { sc->bps = ai->play.bps; set = 1; } if (ai->play.msb != ~0) { sc->msb = ai->play.msb; set = 1; } if (ai->play.sample_rate != ~0) { sc->rate = ai->play.sample_rate; set = 1; } if (ai->play.channels != ~0) { sc->pchan = ai->play.channels; set = 1; } if (ai->play.block_size != ~0) { sc->round = ai->play.block_size / (sc->bps * sc->pchan); set = 1; } if (ai->hiwat != ~0) { sc->nblks = ai->hiwat; set = 1; } if (ai->record.encoding != ~0) { sc->sw_enc = ai->record.encoding; set = 1; } if (ai->record.precision != ~0) { sc->bits = ai->record.precision; set = 1; } if (ai->record.bps != ~0) { sc->bps = ai->record.bps; set = 1; } if (ai->record.msb != ~0) { sc->msb = ai->record.msb; set = 1; } if (ai->record.sample_rate != ~0) { sc->rate = ai->record.sample_rate; set = 1; } if (ai->record.channels != ~0) { sc->rchan = ai->record.channels; set = 1; } if (ai->record.block_size != ~0) { sc->round = ai->record.block_size / (sc->bps * sc->rchan); set = 1; } DPRINTF("%s: setinfo: set = %d, mode = %d, pause = %d\n", DEVNAME(sc), set, sc->mode, sc->pause); /* * if the device not opened, we're done, don't touch the hardware */ if (sc->mode == 0) return 0; /* * change parameters and recalculate buffer sizes */ if (set) { if (sc->active) { DPRINTF("%s: can't change params during dma\n", DEVNAME(sc)); return EBUSY; } error = audio_setpar(sc); if (error) return error; audio_clear(sc); if ((sc->mode & AUMODE_PLAY) && sc->ops->init_output) { error = sc->ops->init_output(sc->arg, sc->play.data, sc->play.len); if (error) return error; } if ((sc->mode & AUMODE_RECORD) && sc->ops->init_input) { error = sc->ops->init_input(sc->arg, sc->rec.data, sc->rec.len); if (error) return error; } } /* * if unpaused, start */ if (!sc->pause && !sc->active) { error = audio_start(sc); if (error) return error; } return 0; } int audio_getinfo(struct audio_softc *sc, struct audio_info *ai) { ai->play.sample_rate = ai->record.sample_rate = sc->rate; ai->play.encoding = ai->record.encoding = sc->sw_enc; ai->play.precision = ai->record.precision = sc->bits; ai->play.bps = ai->record.bps = sc->bps; ai->play.msb = ai->record.msb = sc->msb; ai->play.channels = sc->pchan; ai->record.channels = sc->rchan; /* * XXX: this is used only to display counters through audioctl * and the pos counters are more useful */ mtx_enter(&audio_lock); ai->play.samples = sc->play.pos - sc->play.xrun; ai->record.samples = sc->rec.pos - sc->rec.xrun; mtx_leave(&audio_lock); ai->play.pause = ai->record.pause = sc->pause; ai->play.active = ai->record.active = sc->active; ai->play.buffer_size = sc->play.datalen; ai->record.buffer_size = sc->rec.datalen; ai->play.block_size = sc->round * sc->bps * sc->pchan; ai->record.block_size = sc->round * sc->bps * sc->rchan; ai->hiwat = sc->nblks; ai->lowat = sc->nblks; ai->mode = sc->mode; return 0; } int audio_match(struct device *parent, void *match, void *aux) { struct audio_attach_args *sa = aux; return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; } void audio_attach(struct device *parent, struct device *self, void *aux) { struct audio_softc *sc = (void *)self; struct audio_attach_args *sa = aux; struct audio_hw_if *ops = sa->hwif; void *arg = sa->hdl; int error; printf("\n"); #ifdef DIAGNOSTIC if (ops == 0 || ops->open == 0 || ops->close == 0 || ops->query_encoding == 0 || ops->set_params == 0 || (ops->start_output == 0 && ops->trigger_output == 0) || (ops->start_input == 0 && ops->trigger_input == 0) || ops->halt_output == 0 || ops->halt_input == 0 || ops->getdev == 0 || ops->set_port == 0 || ops->get_port == 0 || ops->query_devinfo == 0 || ops->get_props == 0) { printf("%s: missing method\n", DEVNAME(sc)); sc->ops = 0; return; } #endif sc->ops = ops; sc->arg = arg; #if NWSKBD > 0 wskbd_mixer_init(sc); #endif /* NWSKBD > 0 */ error = audio_buf_init(sc, &sc->play, AUMODE_PLAY); if (error) { sc->ops = 0; printf("%s: could not allocate play buffer\n", DEVNAME(sc)); return; } error = audio_buf_init(sc, &sc->rec, AUMODE_RECORD); if (error) { audio_buf_done(sc, &sc->play); sc->ops = 0; printf("%s: could not allocate record buffer\n", DEVNAME(sc)); return; } /* set defaults */ sc->sw_enc = AUDIO_ENCODING_SLINEAR; sc->bits = 16; sc->bps = 2; sc->msb = 1; sc->rate = 48000; sc->pchan = 2; sc->rchan = 2; sc->round = 960; sc->nblks = 2; sc->play.pos = sc->play.xrun = sc->rec.pos = sc->rec.xrun = 0; } int audio_activate(struct device *self, int act) { struct audio_softc *sc = (struct audio_softc *)self; switch (act) { case DVACT_QUIESCE: /* * good drivers run play and rec handlers in a single * interrupt. Grab the lock to ensure we expose the same * sc->quiesce value to both play and rec handlers */ mtx_enter(&audio_lock); sc->quiesce = 1; mtx_leave(&audio_lock); /* * once sc->quiesce is set, interrupts may occur, but * counters are not advanced and consequently processes * keep sleeping. * * XXX: ensure read/write/ioctl don't start/stop * DMA at the same time, this needs a "ready" condvar */ if (sc->mode != 0 && sc->active) audio_stop_do(sc); DPRINTF("%s: quesce: active = %d\n", DEVNAME(sc), sc->active); break; case DVACT_WAKEUP: DPRINTF("%s: wakeup: active = %d\n", DEVNAME(sc), sc->active); /* * keep buffer usage the same, but set start pointer to * the beginning of the buffer. * * No need to grab the audio_lock as DMA is stopped and * this is the only thread running (caller ensures this) */ sc->quiesce = 0; wakeup(&sc->quiesce); if(sc->mode != 0) { if (audio_setpar(sc) != 0) break; if (sc->mode & AUMODE_PLAY) { sc->play.start = 0; audio_fill_sil(sc, sc->play.data, sc->play.len); } if (sc->mode & AUMODE_RECORD) { sc->rec.start = sc->rec.len - sc->rec.used; audio_fill_sil(sc, sc->rec.data, sc->rec.len); } if (sc->active) audio_start_do(sc); } break; } return 0; } int audio_detach(struct device *self, int flags) { struct audio_softc *sc = (struct audio_softc *)self; int maj, mn; DPRINTF("%s: audio_detach: flags = %d\n", DEVNAME(sc), flags); wakeup(&sc->quiesce); /* locate the major number */ for (maj = 0; maj < nchrdev; maj++) if (cdevsw[maj].d_open == audioopen) break; /* * Nuke the vnodes for any open instances, calls close but as * close uses device_lookup, it returns EXIO and does nothing */ mn = self->dv_unit; vdevgone(maj, mn | AUDIO_DEV_SOUND, mn | AUDIO_DEV_SOUND, VCHR); vdevgone(maj, mn | AUDIO_DEV_AUDIO, mn | AUDIO_DEV_AUDIO, VCHR); vdevgone(maj, mn | AUDIO_DEV_AUDIOCTL, mn | AUDIO_DEV_AUDIOCTL, VCHR); vdevgone(maj, mn | AUDIO_DEV_MIXER, mn | AUDIO_DEV_MIXER, VCHR); /* * The close() method did nothing, quickly halt DMA (normally * parent is already gone, and code below is no-op), and wake-up * user-land blocked in read/write/ioctl, which return EIO. */ if (sc->mode != 0) { if (sc->active) { wakeup(&sc->play.blocking); selwakeup(&sc->play.sel); wakeup(&sc->rec.blocking); selwakeup(&sc->rec.sel); audio_stop(sc); } sc->ops->close(sc->arg); sc->mode = 0; } /* free resources */ audio_buf_done(sc, &sc->play); audio_buf_done(sc, &sc->rec); return 0; } struct device * audio_attach_mi(struct audio_hw_if *ops, void *arg, struct device *dev) { struct audio_attach_args aa; aa.type = AUDIODEV_TYPE_AUDIO; aa.hwif = ops; aa.hdl = arg; /* * attach this driver to the caller (hardware driver), this * checks the kernel config and possibly calls audio_attach() */ return config_found(dev, &aa, audioprint); } int audioprint(void *aux, const char *pnp) { struct audio_attach_args *arg = aux; const char *type; if (pnp != NULL) { switch (arg->type) { case AUDIODEV_TYPE_AUDIO: type = "audio"; break; case AUDIODEV_TYPE_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; } int audio_open(struct audio_softc *sc, int flags) { int error; int props; if (sc->mode) return EBUSY; error = sc->ops->open(sc->arg, flags); if (error) return error; sc->active = 0; sc->pause = 1; sc->rec.blocking = 0; sc->play.blocking = 0; sc->mode = 0; if (flags & FWRITE) sc->mode |= AUMODE_PLAY; if (flags & FREAD) sc->mode |= AUMODE_RECORD; props = sc->ops->get_props(sc->arg); if (sc->mode == (AUMODE_PLAY | AUMODE_RECORD)) { if (!(props & AUDIO_PROP_FULLDUPLEX)) { error = ENOTTY; goto bad; } if (sc->ops->setfd) { error = sc->ops->setfd(sc->arg, 1); if (error) goto bad; } } if (sc->ops->speaker_ctl) { /* * XXX: what is this used for? */ sc->ops->speaker_ctl(sc->arg, (sc->mode & AUMODE_PLAY) ? SPKR_ON : SPKR_OFF); } error = audio_setpar(sc); if (error) goto bad; audio_clear(sc); /* * allow read(2)/write(2) to automatically start DMA, without * the need for ioctl(), to make /dev/audio usable in scripts */ sc->pause = 0; return 0; bad: sc->ops->close(sc->arg); sc->mode = 0; return error; } int audio_drain(struct audio_softc *sc) { int error, xrun; unsigned char *ptr; size_t count, bpf; DPRINTF("%s: drain: mode = %d, pause = %d, active = %d, used = %zu\n", DEVNAME(sc), sc->mode, sc->pause, sc->active, sc->play.used); if (!(sc->mode & AUMODE_PLAY) || sc->pause) return 0; /* discard partial samples, required by audio_fill_sil() */ mtx_enter(&audio_lock); bpf = sc->pchan * sc->bps; sc->play.used -= sc->play.used % bpf; if (sc->play.used == 0) { mtx_leave(&audio_lock); return 0; } if (!sc->active) { /* * dma not started yet because buffer was not full * enough to start automatically. Pad it and start now. */ for (;;) { ptr = audio_buf_wgetblk(&sc->play, &count); if (count == 0) break; audio_fill_sil(sc, ptr, count); audio_buf_wcommit(&sc->play, count); } mtx_leave(&audio_lock); error = audio_start(sc); if (error) return error; mtx_enter(&audio_lock); } xrun = sc->play.xrun; while (sc->play.xrun == xrun) { DPRINTF("%s: drain: used = %zu, xrun = %d\n", DEVNAME(sc), sc->play.used, sc->play.xrun); /* * set a 5 second timeout, in case interrupts don't * work, useful only for debugging drivers */ sc->play.blocking = 1; error = msleep(&sc->play.blocking, &audio_lock, PWAIT | PCATCH, "au_dr", 5 * hz); if (!(sc->dev.dv_flags & DVF_ACTIVE)) error = EIO; if (error) { DPRINTF("%s: drain, err = %d\n", DEVNAME(sc), error); break; } } mtx_leave(&audio_lock); return error; } int audio_close(struct audio_softc *sc) { audio_drain(sc); if (sc->active) audio_stop(sc); sc->ops->close(sc->arg); sc->mode = 0; DPRINTF("%s: close: done\n", DEVNAME(sc)); return 0; } int audio_read(struct audio_softc *sc, struct uio *uio, int ioflag) { unsigned char *ptr; size_t count; int error; DPRINTFN(1, "%s: read: resid = %zd\n", DEVNAME(sc), uio->uio_resid); /* block if quiesced */ while (sc->quiesce) tsleep(&sc->quiesce, 0, "au_qrd", 0); /* start automatically if setinfo() was never called */ mtx_enter(&audio_lock); if (!sc->active && !sc->pause && sc->rec.used == 0) { mtx_leave(&audio_lock); error = audio_start(sc); if (error) return error; mtx_enter(&audio_lock); } /* if there is no data then sleep */ while (sc->rec.used == 0) { if (ioflag & IO_NDELAY) { mtx_leave(&audio_lock); return EWOULDBLOCK; } DPRINTFN(1, "%s: read sleep\n", DEVNAME(sc)); sc->rec.blocking = 1; error = msleep(&sc->rec.blocking, &audio_lock, PWAIT | PCATCH, "au_rd", 0); if (!(sc->dev.dv_flags & DVF_ACTIVE)) error = EIO; #ifdef AUDIO_DEBUG if (error) { DPRINTF("%s: read woke up error = %d\n", DEVNAME(sc), error); } #endif if (error) { mtx_leave(&audio_lock); return error; } } /* at this stage, there is data to transfer */ while (uio->uio_resid > 0 && sc->rec.used > 0) { ptr = audio_buf_rgetblk(&sc->rec, &count); if (count > uio->uio_resid) count = uio->uio_resid; mtx_leave(&audio_lock); DPRINTFN(1, "%s: read: start = %zu, count = %zu\n", DEVNAME(sc), ptr - sc->rec.data, count); if (sc->conv_dec) sc->conv_dec(ptr, count); error = uiomove(ptr, count, uio); if (error) return error; mtx_enter(&audio_lock); audio_buf_rdiscard(&sc->rec, count); } mtx_leave(&audio_lock); return 0; } int audio_write(struct audio_softc *sc, struct uio *uio, int ioflag) { unsigned char *ptr; size_t count; int error; DPRINTFN(1, "%s: write: resid = %zd\n", DEVNAME(sc), uio->uio_resid); /* block if quiesced */ while (sc->quiesce) tsleep(&sc->quiesce, 0, "au_qwr", 0); /* * if IO_NDELAY flag is set then check if there is enough room * in the buffer to store at least one byte. If not then dont * start the write process. */ mtx_enter(&audio_lock); if (uio->uio_resid > 0 && (ioflag & IO_NDELAY)) { if (sc->play.used == sc->play.len ) { mtx_leave(&audio_lock); return EWOULDBLOCK; } } while (uio->uio_resid > 0) { while (1) { ptr = audio_buf_wgetblk(&sc->play, &count); if (count > 0) break; if (ioflag & IO_NDELAY) { /* * At this stage at least one byte is already * moved so we do not return EWOULDBLOCK */ mtx_leave(&audio_lock); return 0; } DPRINTFN(1, "%s: write sleep\n", DEVNAME(sc)); sc->play.blocking = 1; error = msleep(&sc->play.blocking, &audio_lock, PWAIT | PCATCH, "au_wr", 0); if (!(sc->dev.dv_flags & DVF_ACTIVE)) error = EIO; #ifdef AUDIO_DEBUG if (error) { DPRINTF("%s: write woke up error = %d\n", DEVNAME(sc), error); } #endif if (error) { mtx_leave(&audio_lock); return error; } } if (count > uio->uio_resid) count = uio->uio_resid; mtx_leave(&audio_lock); error = uiomove(ptr, count, uio); if (error) return 0; if (sc->conv_enc) { sc->conv_enc(ptr, count); DPRINTFN(1, "audio_write: converted count = %zu\n", count); } mtx_enter(&audio_lock); audio_buf_wcommit(&sc->play, count); /* start automatically if setinfo() was never called */ if (!sc->active && !sc->pause && sc->play.used == sc->play.len) { mtx_leave(&audio_lock); error = audio_start(sc); if (error) return error; mtx_enter(&audio_lock); } } mtx_leave(&audio_lock); return 0; } int audio_ioctl(struct audio_softc *sc, unsigned long cmd, void *addr) { struct audio_offset *ao; struct audio_pos *ap; int error = 0, fd; /* block if quiesced */ while (sc->quiesce) tsleep(&sc->quiesce, 0, "au_qio", 0); switch (cmd) { case FIONBIO: /* All handled in the upper FS layer. */ break; case AUDIO_PERROR: mtx_enter(&audio_lock); *(int *)addr = sc->play.xrun / (sc->pchan * sc->bps); mtx_leave(&audio_lock); break; case AUDIO_RERROR: mtx_enter(&audio_lock); *(int *)addr = sc->rec.xrun / (sc->rchan * sc->bps); mtx_leave(&audio_lock); break; case AUDIO_GETOOFFS: mtx_enter(&audio_lock); ao = (struct audio_offset *)addr; ao->samples = sc->play.pos; mtx_leave(&audio_lock); break; case AUDIO_GETIOFFS: mtx_enter(&audio_lock); ao = (struct audio_offset *)addr; ao->samples = sc->rec.pos; mtx_leave(&audio_lock); break; case AUDIO_GETPOS: mtx_enter(&audio_lock); ap = (struct audio_pos *)addr; ap->play_pos = sc->play.pos; ap->play_xrun = sc->play.xrun; ap->rec_pos = sc->rec.pos; ap->rec_xrun = sc->rec.xrun; mtx_leave(&audio_lock); break; case AUDIO_SETINFO: error = audio_setinfo(sc, (struct audio_info *)addr); break; case AUDIO_GETINFO: error = audio_getinfo(sc, (struct audio_info *)addr); break; case AUDIO_GETDEV: error = sc->ops->getdev(sc->arg, (audio_device_t *)addr); break; case AUDIO_GETENC: error = sc->ops->query_encoding(sc->arg, (struct audio_encoding *)addr); break; case AUDIO_GETFD: *(int *)addr = (sc->mode & (AUMODE_PLAY | AUMODE_RECORD)) == (AUMODE_PLAY | AUMODE_RECORD); break; case AUDIO_SETFD: fd = *(int *)addr; if ((sc->mode & (AUMODE_PLAY | AUMODE_RECORD)) != (AUMODE_PLAY | AUMODE_RECORD) || !fd) return EINVAL; break; case AUDIO_GETPROPS: *(int *)addr = sc->ops->get_props(sc->arg); break; default: DPRINTF("%s: unknown ioctl 0x%lx\n", DEVNAME(sc), cmd); error = ENOTTY; break; } return error; } int audio_ioctl_mixer(struct audio_softc *sc, unsigned long cmd, void *addr) { int error; /* block if quiesced */ while (sc->quiesce) tsleep(&sc->quiesce, 0, "mix_qio", 0); switch (cmd) { case FIONBIO: /* All handled in the upper FS layer. */ break; case AUDIO_MIXER_DEVINFO: ((mixer_devinfo_t *)addr)->un.v.delta = 0; return sc->ops->query_devinfo(sc->arg, (mixer_devinfo_t *)addr); case AUDIO_MIXER_READ: return sc->ops->get_port(sc->arg, (mixer_ctrl_t *)addr); case AUDIO_MIXER_WRITE: error = sc->ops->set_port(sc->arg, (mixer_ctrl_t *)addr); if (error) return error; if (sc->ops->commit_settings) return sc->ops->commit_settings(sc->arg); break; default: return ENOTTY; } return 0; } int audio_poll(struct audio_softc *sc, int events, struct proc *p) { int revents = 0; mtx_enter(&audio_lock); if ((sc->mode & AUMODE_RECORD) && sc->rec.used > 0) revents |= events & (POLLIN | POLLRDNORM); if ((sc->mode & AUMODE_PLAY) && sc->play.used < sc->play.len) revents |= events & (POLLOUT | POLLWRNORM); if (revents == 0) { if (events & (POLLIN | POLLRDNORM)) selrecord(p, &sc->rec.sel); if (events & (POLLOUT | POLLWRNORM)) selrecord(p, &sc->play.sel); } mtx_leave(&audio_lock); return revents; } int audioopen(dev_t dev, int flags, int mode, struct proc *p) { struct audio_softc *sc; int error; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return ENXIO; if (sc->ops == NULL) error = ENXIO; else { switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: error = audio_open(sc, flags); break; case AUDIO_DEV_AUDIOCTL: case AUDIO_DEV_MIXER: error = 0; break; default: error = ENXIO; } } device_unref(&sc->dev); return error; } int audioclose(dev_t dev, int flags, int ifmt, struct proc *p) { struct audio_softc *sc; int error; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return ENXIO; switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: error = audio_close(sc); break; case AUDIO_DEV_MIXER: case AUDIO_DEV_AUDIOCTL: error = 0; default: error = ENXIO; } device_unref(&sc->dev); return error; } int audioread(dev_t dev, struct uio *uio, int ioflag) { struct audio_softc *sc; int error; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return ENXIO; switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: error = audio_read(sc, uio, ioflag); break; case AUDIO_DEV_AUDIOCTL: case AUDIO_DEV_MIXER: error = ENODEV; break; default: error = ENXIO; } device_unref(&sc->dev); return error; } int audiowrite(dev_t dev, struct uio *uio, int ioflag) { struct audio_softc *sc; int error; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return ENXIO; switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: error = audio_write(sc, uio, ioflag); break; case AUDIO_DEV_AUDIOCTL: case AUDIO_DEV_MIXER: error = ENODEV; break; default: error = ENXIO; } device_unref(&sc->dev); return error; } int audioioctl(dev_t dev, u_long cmd, caddr_t addr, int flag, struct proc *p) { struct audio_softc *sc; int error; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return ENXIO; switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: error = audio_ioctl(sc, cmd, addr); break; case AUDIO_DEV_AUDIOCTL: if (cmd == AUDIO_SETINFO && sc->mode != 0) { error = EBUSY; break; } error = audio_ioctl(sc, cmd, addr); break; case AUDIO_DEV_MIXER: error = audio_ioctl_mixer(sc, cmd, addr); break; default: error = ENXIO; } device_unref(&sc->dev); return error; } int audiopoll(dev_t dev, int events, struct proc *p) { struct audio_softc *sc; int revents; sc = (struct audio_softc *)device_lookup(&audio_cd, AUDIO_UNIT(dev)); if (sc == NULL) return POLLERR; switch (AUDIO_DEV(dev)) { case AUDIO_DEV_SOUND: case AUDIO_DEV_AUDIO: revents = audio_poll(sc, events, p); break; case AUDIO_DEV_AUDIOCTL: case AUDIO_DEV_MIXER: revents = 0; break; default: revents = 0; } device_unref(&sc->dev); return revents; } #if NWSKBD > 0 int wskbd_initmute(struct audio_softc *sc, struct mixer_devinfo *vol) { struct mixer_devinfo mi; mi.index = vol->next; for (mi.index = vol->next; mi.index != -1; mi.index = mi.next) { if (sc->ops->query_devinfo(sc->arg, &mi) != 0) break; if (strcmp(mi.label.name, AudioNmute) == 0) return mi.index; } return -1; } int wskbd_initvol(struct audio_softc *sc, struct wskbd_vol *vol, char *cn, char *dn) { struct mixer_devinfo dev, cls; for (dev.index = 0; ; dev.index++) { if (sc->ops->query_devinfo(sc->arg, &dev) != 0) break; cls.index = dev.mixer_class; if (sc->ops->query_devinfo(sc->arg, &cls) != 0) continue; if (strcmp(cls.label.name, cn) == 0 && strcmp(dev.label.name, dn) == 0) { vol->val = dev.index; vol->nch = dev.un.v.num_channels; vol->step = dev.un.v.delta > 8 ? dev.un.v.delta : 8; vol->mute = wskbd_initmute(sc, &dev); vol->val_pending = vol->mute_pending = 0; DPRINTF("%s: wskbd using %s.%s, %s\n", DEVNAME(sc), cn, dn, vol->mute >= -1 ? "mute control" : ""); return 1; } } vol->val = vol->mute = -1; return 0; } void wskbd_mixer_init(struct audio_softc *sc) { static struct { char *cn, *dn; } spkr_names[] = { {AudioCoutputs, AudioNmaster}, {AudioCinputs, AudioNdac}, {AudioCoutputs, AudioNdac}, {AudioCoutputs, AudioNoutput} }, mic_names[] = { {AudioCrecord, AudioNrecord}, {AudioCrecord, AudioNvolume}, {AudioCinputs, AudioNrecord}, {AudioCinputs, AudioNvolume}, {AudioCinputs, AudioNinput} }; int i; if (sc->dev.dv_unit != 0) { DPRINTF("%s: not configuring wskbd keys\n", DEVNAME(sc)); return; } for (i = 0; i < sizeof(spkr_names) / sizeof(spkr_names[0]); i++) { if (wskbd_initvol(sc, &sc->spkr, spkr_names[i].cn, spkr_names[i].dn)) break; } for (i = 0; i < sizeof(mic_names) / sizeof(mic_names[0]); i++) { if (wskbd_initvol(sc, &sc->mic, mic_names[i].cn, mic_names[i].dn)) break; } } void wskbd_mixer_update(struct audio_softc *sc, struct wskbd_vol *vol) { struct mixer_ctrl ctrl; int val_pending, mute_pending, i, gain, error, s; s = spltty(); val_pending = vol->val_pending; vol->val_pending = 0; mute_pending = vol->mute_pending; vol->mute_pending = 0; splx(s); if (sc->ops == NULL) return; if (vol->mute >= 0 && mute_pending) { ctrl.dev = vol->mute; ctrl.type = AUDIO_MIXER_ENUM; error = sc->ops->get_port(sc->arg, &ctrl); if (error) { DPRINTF("%s: get mute err = %d\n", DEVNAME(sc), error); return; } ctrl.un.ord = ctrl.un.ord ^ mute_pending; DPRINTFN(1, "%s: wskbd mute setting to %d\n", DEVNAME(sc), ctrl.un.ord); error = sc->ops->set_port(sc->arg, &ctrl); if (error) { DPRINTF("%s: set mute err = %d\n", DEVNAME(sc), error); return; } } if (vol->val >= 0 && val_pending) { ctrl.dev = vol->val; ctrl.type = AUDIO_MIXER_VALUE; ctrl.un.value.num_channels = vol->nch; error = sc->ops->get_port(sc->arg, &ctrl); if (error) { DPRINTF("%s: get mute err = %d\n", DEVNAME(sc), error); return; } for (i = 0; i < vol->nch; i++) { gain = ctrl.un.value.level[i] + vol->step * val_pending; if (gain > AUDIO_MAX_GAIN) gain = AUDIO_MAX_GAIN; if (gain < AUDIO_MIN_GAIN) gain = AUDIO_MIN_GAIN; ctrl.un.value.level[i] = gain; DPRINTFN(1, "%s: wskbd level %d set to %d\n", DEVNAME(sc), i, gain); } error = sc->ops->set_port(sc->arg, &ctrl); if (error) { DPRINTF("%s: set vol err = %d\n", DEVNAME(sc), error); return; } } } void wskbd_mixer_cb(void *addr) { struct audio_softc *sc = addr; int s; wskbd_mixer_update(sc, &sc->spkr); wskbd_mixer_update(sc, &sc->mic); s = spltty(); sc->wskbd_taskset = 0; splx(s); device_unref(&sc->dev); } int wskbd_set_mixervolume(long dir, long out) { struct audio_softc *sc; struct wskbd_vol *vol; sc = (struct audio_softc *)device_lookup(&audio_cd, 0); if (sc == NULL) return ENODEV; vol = out ? &sc->spkr : &sc->mic; if (dir == 0) vol->mute_pending ^= 1; else vol->val_pending += dir; if (!sc->wskbd_taskset) { task_set(&sc->wskbd_task, wskbd_mixer_cb, sc); task_add(systq, &sc->wskbd_task); sc->wskbd_taskset = 1; } return 0; } #endif /* NWSKBD > 0 */