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09b2af815f
NetBSD/sys/dev/audio.c
tsutsui e0a14f3564 Fix botches in rev 1.185 (kent-audio1 merge) which cause buffer overrun 
and "tlb_exception: no copyin/out fault hander" panic in
audio_fill_silence() after audio playback on dreamcast.

The problem is tracked and fix is provieded by Ryo SHIMIZU,
who is the auther of aica(4) audio driver for dreamcast.
2010-11-20 18:03:49 +00:00

4214 lines
102 KiB
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/* $NetBSD: audio.c,v 1.252 2010/11/20 18:03:49 tsutsui 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 <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.252 2010/11/20 18:03:49 tsutsui Exp $");
#include "audio.h"
#if NAUDIO > 0
#include <sys/param.h>
#include <sys/ioctl.h>
#include <sys/fcntl.h>
#include <sys/vnode.h>
#include <sys/select.h>
#include <sys/poll.h>
#include <sys/malloc.h>
#include <sys/proc.h>
#include <sys/systm.h>
#include <sys/syslog.h>
#include <sys/kernel.h>
#include <sys/signalvar.h>
#include <sys/conf.h>
#include <sys/audioio.h>
#include <sys/device.h>
#include <sys/intr.h>
#include <dev/audio_if.h>
#include <dev/audiovar.h>
#include <machine/endian.h>
/* #define AUDIO_DEBUG 1 */
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (audiodebug) printf x
#define DPRINTFN(n,x) if (audiodebug>(n)) printf x
int audiodebug = AUDIO_DEBUG;
#else
#define DPRINTF(x)
#define DPRINTFN(n,x)
#endif
#define ROUNDSIZE(x) x &= -16 /* round to nice boundary */
#define SPECIFIED(x) (x != ~0)
#define SPECIFIED_CH(x) (x != (u_char)~0)
/* #define AUDIO_PM_IDLE */
#ifdef AUDIO_PM_IDLE
int audio_idle_timeout = 30;
#endif
int audio_blk_ms = AUDIO_BLK_MS;
int audiosetinfo(struct audio_softc *, struct audio_info *);
int audiogetinfo(struct audio_softc *, struct audio_info *, int);
int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *);
int audio_close(struct audio_softc *, int, int, struct lwp *);
int audio_read(struct audio_softc *, struct uio *, int);
int audio_write(struct audio_softc *, struct uio *, int);
int audio_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *);
int audio_poll(struct audio_softc *, int, struct lwp *);
int audio_kqfilter(struct audio_softc *, struct knote *);
paddr_t audio_mmap(struct audio_softc *, off_t, int);
int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *);
int mixer_close(struct audio_softc *, int, int, struct lwp *);
int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *);
static void mixer_remove(struct audio_softc *, struct lwp *);
static void mixer_signal(struct audio_softc *);
void audio_init_record(struct audio_softc *);
void audio_init_play(struct audio_softc *);
int audiostartr(struct audio_softc *);
int audiostartp(struct audio_softc *);
void audio_rint(void *);
void audio_pint(void *);
int audio_check_params(struct audio_params *);
void audio_calc_blksize(struct audio_softc *, int);
void audio_fill_silence(struct audio_params *, uint8_t *, int);
int audio_silence_copyout(struct audio_softc *, int, struct uio *);
void audio_init_ringbuffer(struct audio_softc *,
struct audio_ringbuffer *, int);
int audio_initbufs(struct audio_softc *);
void audio_calcwater(struct audio_softc *);
static inline int audio_sleep_timo(int *, const char *, int);
static inline int audio_sleep(int *, const char *);
static inline void audio_wakeup(int *);
int audio_drain(struct audio_softc *);
void audio_clear(struct audio_softc *);
static inline void audio_pint_silence
(struct audio_softc *, struct audio_ringbuffer *, uint8_t *, int);
int audio_alloc_ring
(struct audio_softc *, struct audio_ringbuffer *, int, size_t);
void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *);
static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *,
stream_filter_list_t *);
static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *,
stream_filter_list_t *);
static void audio_destruct_pfilters(struct audio_softc *);
static void audio_destruct_rfilters(struct audio_softc *);
static void audio_stream_dtor(audio_stream_t *);
static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int);
static void stream_filter_list_append
(stream_filter_list_t *, stream_filter_factory_t,
const audio_params_t *);
static void stream_filter_list_prepend
(stream_filter_list_t *, stream_filter_factory_t,
const audio_params_t *);
static void stream_filter_list_set
(stream_filter_list_t *, int, stream_filter_factory_t,
const audio_params_t *);
int audio_set_defaults(struct audio_softc *, u_int);
int audioprobe(device_t, cfdata_t, void *);
void audioattach(device_t, device_t, void *);
int audiodetach(device_t, int);
int audioactivate(device_t, enum devact);
#ifdef AUDIO_PM_IDLE
static void audio_idle(void *);
static void audio_activity(device_t, devactive_t);
#endif
static bool audio_suspend(device_t dv, const pmf_qual_t *);
static bool audio_resume(device_t dv, const pmf_qual_t *);
static void audio_volume_down(device_t);
static void audio_volume_up(device_t);
static void audio_volume_toggle(device_t);
static void audio_mixer_capture(struct audio_softc *);
static void audio_mixer_restore(struct audio_softc *);
static int audio_get_props(struct audio_softc *);
static bool audio_can_playback(struct audio_softc *);
static bool audio_can_capture(struct audio_softc *);
static void audio_softintr_rd(void *);
static void audio_softintr_wr(void *);
struct portname {
const char *name;
int mask;
};
static const struct portname itable[] = {
{ AudioNmicrophone, AUDIO_MICROPHONE },
{ AudioNline, AUDIO_LINE_IN },
{ AudioNcd, AUDIO_CD },
{ 0, 0 }
};
static const struct portname otable[] = {
{ AudioNspeaker, AUDIO_SPEAKER },
{ AudioNheadphone, AUDIO_HEADPHONE },
{ AudioNline, AUDIO_LINE_OUT },
{ 0, 0 }
};
void au_setup_ports(struct audio_softc *, struct au_mixer_ports *,
mixer_devinfo_t *, const struct portname *);
int au_set_gain(struct audio_softc *, struct au_mixer_ports *,
int, int);
void au_get_gain(struct audio_softc *, struct au_mixer_ports *,
u_int *, u_char *);
int au_set_port(struct audio_softc *, struct au_mixer_ports *,
u_int);
int au_get_port(struct audio_softc *, struct au_mixer_ports *);
int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *,
int *, int *);
int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *,
int, int);
int au_portof(struct audio_softc *, char *, int);
typedef struct uio_fetcher {
stream_fetcher_t base;
struct uio *uio;
int usedhigh;
int last_used;
} uio_fetcher_t;
static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int);
static int uio_fetcher_fetch_to(stream_fetcher_t *,
audio_stream_t *, int);
static int null_fetcher_fetch_to(stream_fetcher_t *,
audio_stream_t *, int);
dev_type_open(audioopen);
dev_type_close(audioclose);
dev_type_read(audioread);
dev_type_write(audiowrite);
dev_type_ioctl(audioioctl);
dev_type_poll(audiopoll);
dev_type_mmap(audiommap);
dev_type_kqfilter(audiokqfilter);
const struct cdevsw audio_cdevsw = {
audioopen, audioclose, audioread, audiowrite, audioioctl,
nostop, notty, audiopoll, audiommap, audiokqfilter, D_OTHER
};
/* The default audio mode: 8 kHz mono mu-law */
const struct audio_params audio_default = {
.sample_rate = 8000,
.encoding = AUDIO_ENCODING_ULAW,
.precision = 8,
.validbits = 8,
.channels = 1,
};
CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc),
audioprobe, audioattach, audiodetach, audioactivate, NULL, NULL,
DVF_DETACH_SHUTDOWN);
extern struct cfdriver audio_cd;
int
audioprobe(device_t parent, cfdata_t match, void *aux)
{
struct audio_attach_args *sa;
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(device_t parent, device_t self, void *aux)
{
struct audio_softc *sc;
struct audio_attach_args *sa;
const struct audio_hw_if *hwp;
void *hdlp;
int error;
mixer_devinfo_t mi;
int iclass, mclass, oclass, rclass, props;
int record_master_found, record_source_found;
sc = device_private(self);
sc->dev = self;
sa = aux;
hwp = sa->hwif;
hdlp = sa->hdl;
#ifdef DIAGNOSTIC
if (hwp == 0 ||
hwp->query_encoding == 0 ||
hwp->set_params == 0 ||
(hwp->start_output == 0 && hwp->trigger_output == 0) ||
(hwp->start_input == 0 && hwp->trigger_input == 0) ||
hwp->halt_output == 0 ||
hwp->halt_input == 0 ||
hwp->getdev == 0 ||
hwp->set_port == 0 ||
hwp->get_port == 0 ||
hwp->query_devinfo == 0 ||
hwp->get_props == 0) {
printf(": missing method\n");
sc->hw_if = 0;
return;
}
#endif
sc->hw_if = hwp;
sc->hw_hdl = hdlp;
sc->sc_dev = parent;
sc->sc_opencnt = 0;
sc->sc_writing = sc->sc_waitcomp = 0;
sc->sc_lastinfovalid = false;
props = audio_get_props(sc);
if (props & AUDIO_PROP_FULLDUPLEX)
aprint_normal(": full duplex");
else
aprint_normal(": half duplex");
if (props & AUDIO_PROP_PLAYBACK)
aprint_normal(", playback");
if (props & AUDIO_PROP_CAPTURE)
aprint_normal(", capture");
if (props & AUDIO_PROP_MMAP)
aprint_normal(", mmap");
if (props & AUDIO_PROP_INDEPENDENT)
aprint_normal(", independent");
aprint_naive("\n");
aprint_normal("\n");
if (audio_can_playback(sc)) {
error = audio_alloc_ring(sc, &sc->sc_pr,
AUMODE_PLAY, AU_RING_SIZE);
if (error) {
sc->hw_if = NULL;
aprint_error("audio: could not allocate play buffer\n");
return;
}
}
if (audio_can_capture(sc)) {
error = audio_alloc_ring(sc, &sc->sc_rr,
AUMODE_RECORD, AU_RING_SIZE);
if (error) {
if (sc->sc_pr.s.start != 0)
audio_free_ring(sc, &sc->sc_pr);
sc->hw_if = NULL;
aprint_error("audio: could not allocate record buffer\n");
return;
}
}
sc->sc_lastgain = 128;
if ((error = audio_set_defaults(sc, 0))) {
aprint_error("audioattach: audio_set_defaults() failed\n");
sc->hw_if = NULL;
return;
}
sc->sc_sih_rd = softint_establish(SOFTINT_SERIAL,
audio_softintr_rd, sc);
sc->sc_sih_wr = softint_establish(SOFTINT_SERIAL,
audio_softintr_wr, sc);
iclass = mclass = oclass = rclass = -1;
sc->sc_inports.index = -1;
sc->sc_inports.master = -1;
sc->sc_inports.nports = 0;
sc->sc_inports.isenum = false;
sc->sc_inports.allports = 0;
sc->sc_inports.isdual = false;
sc->sc_inports.mixerout = -1;
sc->sc_inports.cur_port = -1;
sc->sc_outports.index = -1;
sc->sc_outports.master = -1;
sc->sc_outports.nports = 0;
sc->sc_outports.isenum = false;
sc->sc_outports.allports = 0;
sc->sc_outports.isdual = false;
sc->sc_outports.mixerout = -1;
sc->sc_outports.cur_port = -1;
sc->sc_monitor_port = -1;
/*
* Read through the underlying driver's list, picking out the class
* names from the mixer descriptions. We'll need them to decode the
* mixer descriptions on the next pass through the loop.
*/
for(mi.index = 0; ; mi.index++) {
if (hwp->query_devinfo(hdlp, &mi) != 0)
break;
/*
* The type of AUDIO_MIXER_CLASS merely introduces a class.
* All the other types describe an actual mixer.
*/
if (mi.type == AUDIO_MIXER_CLASS) {
if (strcmp(mi.label.name, AudioCinputs) == 0)
iclass = mi.mixer_class;
if (strcmp(mi.label.name, AudioCmonitor) == 0)
mclass = mi.mixer_class;
if (strcmp(mi.label.name, AudioCoutputs) == 0)
oclass = mi.mixer_class;
if (strcmp(mi.label.name, AudioCrecord) == 0)
rclass = mi.mixer_class;
}
}
/*
* This is where we assign each control in the "audio" model, to the
* underlying "mixer" control. We walk through the whole list once,
* assigning likely candidates as we come across them.
*/
record_master_found = 0;
record_source_found = 0;
for(mi.index = 0; ; mi.index++) {
if (hwp->query_devinfo(hdlp, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS)
continue;
if (mi.mixer_class == iclass) {
/*
* AudioCinputs is only a fallback, when we don't
* find what we're looking for in AudioCrecord, so
* check the flags before accepting one of these.
*/
if (strcmp(mi.label.name, AudioNmaster) == 0
&& record_master_found == 0)
sc->sc_inports.master = mi.index;
if (strcmp(mi.label.name, AudioNsource) == 0
&& record_source_found == 0) {
if (mi.type == AUDIO_MIXER_ENUM) {
int i;
for(i = 0; i < mi.un.e.num_mem; i++)
if (strcmp(mi.un.e.member[i].label.name,
AudioNmixerout) == 0)
sc->sc_inports.mixerout =
mi.un.e.member[i].ord;
}
au_setup_ports(sc, &sc->sc_inports, &mi,
itable);
}
if (strcmp(mi.label.name, AudioNdac) == 0 &&
sc->sc_outports.master == -1)
sc->sc_outports.master = mi.index;
} else if (mi.mixer_class == mclass) {
if (strcmp(mi.label.name, AudioNmonitor) == 0)
sc->sc_monitor_port = mi.index;
} else if (mi.mixer_class == oclass) {
if (strcmp(mi.label.name, AudioNmaster) == 0)
sc->sc_outports.master = mi.index;
if (strcmp(mi.label.name, AudioNselect) == 0)
au_setup_ports(sc, &sc->sc_outports, &mi,
otable);
} else if (mi.mixer_class == rclass) {
/*
* These are the preferred mixers for the audio record
* controls, so set the flags here, but don't check.
*/
if (strcmp(mi.label.name, AudioNmaster) == 0) {
sc->sc_inports.master = mi.index;
record_master_found = 1;
}
#if 1 /* Deprecated. Use AudioNmaster. */
if (strcmp(mi.label.name, AudioNrecord) == 0) {
sc->sc_inports.master = mi.index;
record_master_found = 1;
}
if (strcmp(mi.label.name, AudioNvolume) == 0) {
sc->sc_inports.master = mi.index;
record_master_found = 1;
}
#endif
if (strcmp(mi.label.name, AudioNsource) == 0) {
if (mi.type == AUDIO_MIXER_ENUM) {
int i;
for(i = 0; i < mi.un.e.num_mem; i++)
if (strcmp(mi.un.e.member[i].label.name,
AudioNmixerout) == 0)
sc->sc_inports.mixerout =
mi.un.e.member[i].ord;
}
au_setup_ports(sc, &sc->sc_inports, &mi,
itable);
record_source_found = 1;
}
}
}
DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, "
"output ports=0x%x, output master=%d\n",
sc->sc_inports.allports, sc->sc_inports.master,
sc->sc_outports.allports, sc->sc_outports.master));
selinit(&sc->sc_rsel);
selinit(&sc->sc_wsel);
#ifdef AUDIO_PM_IDLE
callout_init(&sc->sc_idle_counter, 0);
callout_setfunc(&sc->sc_idle_counter, audio_idle, self);
#endif
if (!pmf_device_register(self, audio_suspend, audio_resume))
aprint_error_dev(self, "couldn't establish power handler\n");
#ifdef AUDIO_PM_IDLE
if (!device_active_register(self, audio_activity))
aprint_error_dev(self, "couldn't register activity handler\n");
#endif
if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN,
audio_volume_down, true))
aprint_error_dev(self, "couldn't add volume down handler\n");
if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP,
audio_volume_up, true))
aprint_error_dev(self, "couldn't add volume up handler\n");
if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE,
audio_volume_toggle, true))
aprint_error_dev(self, "couldn't add volume toggle handler\n");
#ifdef AUDIO_PM_IDLE
callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz);
#endif
}
int
audioactivate(device_t self, enum devact act)
{
struct audio_softc *sc = device_private(self);
switch (act) {
case DVACT_DEACTIVATE:
sc->sc_dying = true;
return 0;
default:
return EOPNOTSUPP;
}
}
int
audiodetach(device_t self, int flags)
{
struct audio_softc *sc;
int maj, mn;
int s;
sc = device_private(self);
DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags));
sc->sc_dying = true;
pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN,
audio_volume_down, true);
pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP,
audio_volume_up, true);
pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE,
audio_volume_toggle, true);
#ifdef AUDIO_PM_IDLE
callout_stop(&sc->sc_idle_counter);
device_active_deregister(self, audio_activity);
#endif
pmf_device_deregister(self);
wakeup(&sc->sc_wchan);
wakeup(&sc->sc_rchan);
s = splaudio();
if (--sc->sc_refcnt >= 0) {
if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120))
printf("audiodetach: %s didn't detach\n",
device_xname(sc->dev));
}
splx(s);
/* free resources */
audio_free_ring(sc, &sc->sc_pr);
audio_free_ring(sc, &sc->sc_rr);
audio_destruct_pfilters(sc);
audio_destruct_rfilters(sc);
/* locate the major number */
maj = cdevsw_lookup_major(&audio_cdevsw);
/* Nuke the vnodes for any open instances (calls close). */
mn = device_unit(self);
vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR);
vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR);
vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR);
vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR);
if (sc->sc_sih_rd) {
softint_disestablish(sc->sc_sih_rd);
sc->sc_sih_rd = NULL;
}
if (sc->sc_sih_wr) {
softint_disestablish(sc->sc_sih_wr);
sc->sc_sih_wr = NULL;
}
#ifdef AUDIO_PM_IDLE
callout_destroy(&sc->sc_idle_counter);
#endif
seldestroy(&sc->sc_rsel);
seldestroy(&sc->sc_wsel);
return 0;
}
int
au_portof(struct audio_softc *sc, char *name, int class)
{
mixer_devinfo_t mi;
for(mi.index = 0;
sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0;
mi.index++)
if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0)
return mi.index;
return -1;
}
void
au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports,
mixer_devinfo_t *mi, const struct portname *tbl)
{
int i, j;
ports->index = mi->index;
if (mi->type == AUDIO_MIXER_ENUM) {
ports->isenum = true;
for(i = 0; tbl[i].name; i++)
for(j = 0; j < mi->un.e.num_mem; j++)
if (strcmp(mi->un.e.member[j].label.name,
tbl[i].name) == 0) {
ports->allports |= tbl[i].mask;
ports->aumask[ports->nports] = tbl[i].mask;
ports->misel[ports->nports] =
mi->un.e.member[j].ord;
ports->miport[ports->nports] =
au_portof(sc, mi->un.e.member[j].label.name,
mi->mixer_class);
if (ports->mixerout != -1 &&
ports->miport[ports->nports] != -1)
ports->isdual = true;
++ports->nports;
}
} else if (mi->type == AUDIO_MIXER_SET) {
for(i = 0; tbl[i].name; i++)
for(j = 0; j < mi->un.s.num_mem; j++)
if (strcmp(mi->un.s.member[j].label.name,
tbl[i].name) == 0) {
ports->allports |= tbl[i].mask;
ports->aumask[ports->nports] = tbl[i].mask;
ports->misel[ports->nports] =
mi->un.s.member[j].mask;
ports->miport[ports->nports] =
au_portof(sc, mi->un.s.member[j].label.name,
mi->mixer_class);
++ports->nports;
}
}
}
/*
* Called from hardware driver. This is where the MI audio driver gets
* probed/attached to the hardware driver.
*/
device_t
audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev)
{
struct audio_attach_args arg;
#ifdef DIAGNOSTIC
if (ahwp == NULL) {
aprint_error("audio_attach_mi: NULL\n");
return 0;
}
#endif
arg.type = AUDIODEV_TYPE_AUDIO;
arg.hwif = ahwp;
arg.hdl = hdlp;
return config_found(dev, &arg, audioprint);
}
#ifdef AUDIO_DEBUG
void audio_printsc(struct audio_softc *);
void audio_print_params(const char *, struct audio_params *);
void
audio_printsc(struct audio_softc *sc)
{
printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode);
printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan);
printf("rring used 0x%x pring used=%d\n",
audio_stream_get_used(&sc->sc_rr.s),
audio_stream_get_used(&sc->sc_pr.s));
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(const char *s, struct audio_params *p)
{
printf("%s enc=%u %uch %u/%ubit %uHz\n", s, p->encoding, p->channels,
p->validbits, p->precision, p->sample_rate);
}
#endif
int
audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r,
int direction, size_t bufsize)
{
const struct audio_hw_if *hw;
void *hdl;
hw = sc->hw_if;
hdl = sc->hw_hdl;
/*
* Alloc DMA play and record buffers
*/
if (bufsize < AUMINBUF)
bufsize = AUMINBUF;
ROUNDSIZE(bufsize);
if (hw->round_buffersize)
bufsize = hw->round_buffersize(hdl, direction, bufsize);
if (hw->allocm)
r->s.start = hw->allocm(hdl, direction, bufsize,
M_DEVBUF, M_WAITOK);
else
r->s.start = malloc(bufsize, M_DEVBUF, M_WAITOK);
if (r->s.start == 0)
return ENOMEM;
r->s.bufsize = bufsize;
return 0;
}
void
audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r)
{
if (r->s.start == 0)
return;
if (sc->hw_if->freem)
sc->hw_if->freem(sc->hw_hdl, r->s.start, M_DEVBUF);
else
free(r->s.start, M_DEVBUF);
r->s.start = 0;
}
static int
audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp,
stream_filter_list_t *pfilters)
{
stream_filter_t *pf[AUDIO_MAX_FILTERS];
audio_stream_t ps[AUDIO_MAX_FILTERS];
const audio_params_t *from_param;
audio_params_t *to_param;
int i, n;
while (sc->sc_writing) {
sc->sc_waitcomp = 1;
(void)tsleep(sc, 0, "audioch", 10*hz);
}
memset(pf, 0, sizeof(pf));
memset(ps, 0, sizeof(ps));
from_param = pp;
for (i = 0; i < pfilters->req_size; i++) {
n = pfilters->req_size - i - 1;
to_param = &pfilters->filters[n].param;
audio_check_params(to_param);
pf[i] = pfilters->filters[n].factory(sc, from_param, to_param);
if (pf[i] == NULL)
break;
if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE))
break;
if (i > 0)
pf[i]->set_fetcher(pf[i], &pf[i - 1]->base);
from_param = to_param;
}
if (i < pfilters->req_size) { /* failure */
DPRINTF(("%s: pfilters failure\n", __func__));
for (; i >= 0; i--) {
if (pf[i] != NULL)
pf[i]->dtor(pf[i]);
audio_stream_dtor(&ps[i]);
}
sc->sc_waitcomp = 0;
return EINVAL;
}
audio_destruct_pfilters(sc);
memcpy(sc->sc_pfilters, pf, sizeof(pf));
memcpy(sc->sc_pstreams, ps, sizeof(ps));
sc->sc_npfilters = pfilters->req_size;
for (i = 0; i < pfilters->req_size; i++) {
pf[i]->set_inputbuffer(pf[i], &sc->sc_pstreams[i]);
}
/* hardware format and the buffer near to userland */
if (pfilters->req_size <= 0) {
sc->sc_pr.s.param = *pp;
sc->sc_pustream = &sc->sc_pr.s;
} else {
sc->sc_pr.s.param = pfilters->filters[0].param;
sc->sc_pustream = &sc->sc_pstreams[0];
}
#ifdef AUDIO_DEBUG
printf("%s: HW-buffer=%p pustream=%p\n",
__func__, &sc->sc_pr.s, sc->sc_pustream);
for (i = 0; i < pfilters->req_size; i++) {
char num[100];
snprintf(num, 100, "[%d]", i);
audio_print_params(num, &sc->sc_pstreams[i].param);
}
audio_print_params("[HW]", &sc->sc_pr.s.param);
#endif /* AUDIO_DEBUG */
sc->sc_waitcomp = 0;
return 0;
}
static int
audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp,
stream_filter_list_t *rfilters)
{
stream_filter_t *rf[AUDIO_MAX_FILTERS];
audio_stream_t rs[AUDIO_MAX_FILTERS];
const audio_params_t *to_param;
audio_params_t *from_param;
int i;
memset(rf, 0, sizeof(rf));
memset(rs, 0, sizeof(rs));
for (i = 0; i < rfilters->req_size; i++) {
from_param = &rfilters->filters[i].param;
audio_check_params(from_param);
to_param = i + 1 < rfilters->req_size
? &rfilters->filters[i + 1].param : rp;
rf[i] = rfilters->filters[i].factory(sc, from_param, to_param);
if (rf[i] == NULL)
break;
if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE))
break;
if (i > 0) {
rf[i]->set_fetcher(rf[i], &rf[i - 1]->base);
} else {
/* rf[0] has no previous fetcher because
* the audio hardware fills data to the
* input buffer. */
rf[0]->set_inputbuffer(rf[0], &sc->sc_rr.s);
}
}
if (i < rfilters->req_size) { /* failure */
DPRINTF(("%s: rfilters failure\n", __func__));
for (; i >= 0; i--) {
if (rf[i] != NULL)
rf[i]->dtor(rf[i]);
audio_stream_dtor(&rs[i]);
}
return EINVAL;
}
audio_destruct_rfilters(sc);
memcpy(sc->sc_rfilters, rf, sizeof(rf));
memcpy(sc->sc_rstreams, rs, sizeof(rs));
sc->sc_nrfilters = rfilters->req_size;
for (i = 1; i < rfilters->req_size; i++) {
rf[i]->set_inputbuffer(rf[i], &sc->sc_rstreams[i - 1]);
}
/* hardware format and the buffer near to userland */
if (rfilters->req_size <= 0) {
sc->sc_rr.s.param = *rp;
sc->sc_rustream = &sc->sc_rr.s;
} else {
sc->sc_rr.s.param = rfilters->filters[0].param;
sc->sc_rustream = &sc->sc_rstreams[rfilters->req_size - 1];
}
#ifdef AUDIO_DEBUG
printf("%s: HW-buffer=%p pustream=%p\n",
__func__, &sc->sc_rr.s, sc->sc_rustream);
audio_print_params("[HW]", &sc->sc_rr.s.param);
for (i = 0; i < rfilters->req_size; i++) {
char num[100];
snprintf(num, 100, "[%d]", i);
audio_print_params(num, &sc->sc_rstreams[i].param);
}
#endif /* AUDIO_DEBUG */
return 0;
}
static void
audio_destruct_pfilters(struct audio_softc *sc)
{
int i;
for (i = 0; i < sc->sc_npfilters; i++) {
sc->sc_pfilters[i]->dtor(sc->sc_pfilters[i]);
sc->sc_pfilters[i] = NULL;
audio_stream_dtor(&sc->sc_pstreams[i]);
}
sc->sc_npfilters = 0;
}
static void
audio_destruct_rfilters(struct audio_softc *sc)
{
int i;
for (i = 0; i < sc->sc_nrfilters; i++) {
sc->sc_rfilters[i]->dtor(sc->sc_rfilters[i]);
sc->sc_rfilters[i] = NULL;
audio_stream_dtor(&sc->sc_rstreams[i]);
}
sc->sc_nrfilters = 0;
}
static void
audio_stream_dtor(audio_stream_t *stream)
{
if (stream->start != NULL)
free(stream->start, M_DEVBUF);
memset(stream, 0, sizeof(audio_stream_t));
}
static int
audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size)
{
int frame_size;
size = min(size, AU_RING_SIZE);
stream->bufsize = size;
stream->start = malloc(size, M_DEVBUF, M_NOWAIT);
if (stream->start == NULL)
return ENOMEM;
frame_size = (param->precision + 7) / 8 * param->channels;
size = (size / frame_size) * frame_size;
stream->end = stream->start + size;
stream->inp = stream->start;
stream->outp = stream->start;
stream->used = 0;
stream->param = *param;
stream->loop = false;
return 0;
}
static void
stream_filter_list_append(stream_filter_list_t *list,
stream_filter_factory_t factory,
const audio_params_t *param)
{
if (list->req_size >= AUDIO_MAX_FILTERS) {
printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n",
__func__);
return;
}
list->filters[list->req_size].factory = factory;
list->filters[list->req_size].param = *param;
list->req_size++;
}
static void
stream_filter_list_set(stream_filter_list_t *list, int i,
stream_filter_factory_t factory,
const audio_params_t *param)
{
if (i < 0 || i >= AUDIO_MAX_FILTERS) {
printf("%s: invalid index: %d\n", __func__, i);
return;
}
list->filters[i].factory = factory;
list->filters[i].param = *param;
if (list->req_size <= i)
list->req_size = i + 1;
}
static void
stream_filter_list_prepend(stream_filter_list_t *list,
stream_filter_factory_t factory,
const audio_params_t *param)
{
if (list->req_size >= AUDIO_MAX_FILTERS) {
printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n",
__func__);
return;
}
memmove(&list->filters[1], &list->filters[0],
sizeof(struct stream_filter_req) * list->req_size);
list->filters[0].factory = factory;
list->filters[0].param = *param;
list->req_size++;
}
int
audioopen(dev_t dev, int flags, int ifmt, struct lwp *l)
{
struct audio_softc *sc;
int error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
if (sc->sc_dying)
return EIO;
device_active(sc->dev, DVA_SYSTEM);
sc->sc_opencnt++;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_open(dev, sc, flags, ifmt, l);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
case MIXER_DEVICE:
error = mixer_open(dev, sc, flags, ifmt, l);
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return error;
}
int
audioclose(dev_t dev, int flags, int ifmt, struct lwp *l)
{
struct audio_softc *sc;
int error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
device_active(sc->dev, DVA_SYSTEM);
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_close(sc, flags, ifmt, l);
break;
case MIXER_DEVICE:
error = mixer_close(sc, flags, ifmt, l);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
default:
error = ENXIO;
break;
}
sc->sc_opencnt--;
return error;
}
int
audioread(dev_t dev, struct uio *uio, int ioflag)
{
struct audio_softc *sc;
int error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
if (sc->sc_dying)
return EIO;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_read(sc, uio, ioflag);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return error;
}
int
audiowrite(dev_t dev, struct uio *uio, int ioflag)
{
struct audio_softc *sc;
int error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
if (sc->sc_dying)
return EIO;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_write(sc, uio, ioflag);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return error;
}
int
audioioctl(dev_t dev, u_long cmd, void *addr, int flag, struct lwp *l)
{
struct audio_softc *sc;
int error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc->sc_dying)
return EIO;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
case AUDIOCTL_DEVICE:
device_active(sc->dev, DVA_SYSTEM);
if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ))
error = mixer_ioctl(sc, cmd, addr, flag, l);
else
error = audio_ioctl(sc, cmd, addr, flag, l);
break;
case MIXER_DEVICE:
error = mixer_ioctl(sc, cmd, addr, flag, l);
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return error;
}
int
audiopoll(dev_t dev, int events, struct lwp *l)
{
struct audio_softc *sc;
int revents;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc->sc_dying)
return POLLHUP;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
revents = audio_poll(sc, events, l);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
revents = 0;
break;
default:
revents = POLLERR;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return revents;
}
int
audiokqfilter(dev_t dev, struct knote *kn)
{
struct audio_softc *sc;
int rv;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc->sc_dying)
return 1;
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
rv = audio_kqfilter(sc, kn);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
rv = 1;
break;
default:
rv = 1;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return rv;
}
paddr_t
audiommap(dev_t dev, off_t off, int prot)
{
struct audio_softc *sc;
paddr_t error;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc->sc_dying)
return -1;
device_active(sc->dev, DVA_SYSTEM); /* XXXJDM */
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_mmap(sc, off, prot);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = -1;
break;
default:
error = -1;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return error;
}
/*
* Audio driver
*/
void
audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp,
int mode)
{
int nblks;
int blksize;
blksize = rp->blksize;
if (blksize < AUMINBLK)
blksize = AUMINBLK;
if (blksize > rp->s.bufsize / AUMINNOBLK)
blksize = rp->s.bufsize / AUMINNOBLK;
ROUNDSIZE(blksize);
DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize));
if (sc->hw_if->round_blocksize)
blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize,
mode, &rp->s.param);
if (blksize <= 0)
panic("audio_init_ringbuffer: blksize");
nblks = rp->s.bufsize / blksize;
DPRINTF(("audio_init_ringbuffer: final blksize=%d\n", blksize));
rp->blksize = blksize;
rp->maxblks = nblks;
rp->s.end = rp->s.start + nblks * blksize;
rp->s.outp = rp->s.inp = rp->s.start;
rp->s.used = 0;
rp->stamp = 0;
rp->stamp_last = 0;
rp->fstamp = 0;
rp->drops = 0;
rp->copying = false;
rp->needfill = false;
rp->mmapped = false;
}
int
audio_initbufs(struct audio_softc *sc)
{
const struct audio_hw_if *hw;
int error;
DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
hw = sc->hw_if;
if (audio_can_capture(sc)) {
audio_init_ringbuffer(sc, &sc->sc_rr, AUMODE_RECORD);
if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) {
error = hw->init_input(sc->hw_hdl, sc->sc_rr.s.start,
sc->sc_rr.s.end - sc->sc_rr.s.start);
if (error)
return error;
}
}
if (audio_can_playback(sc)) {
audio_init_ringbuffer(sc, &sc->sc_pr, AUMODE_PLAY);
sc->sc_sil_count = 0;
if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) {
error = hw->init_output(sc->hw_hdl, sc->sc_pr.s.start,
sc->sc_pr.s.end - sc->sc_pr.s.start);
if (error)
return error;
}
}
#ifdef AUDIO_INTR_TIME
#define double u_long
if (audio_can_playback(sc)) {
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));
}
if (audio_can_capture(sc)) {
sc->sc_rnintr = 0;
sc->sc_rblktime = (u_long)(
(double)sc->sc_rr.blksize * 100000 /
(double)(sc->sc_rparams.precision / NBBY *
sc->sc_rparams.channels *
sc->sc_rparams.sample_rate)) * 10;
DPRINTF(("audio: record blktime = %lu for %d\n",
sc->sc_rblktime, sc->sc_rr.blksize));
}
#undef double
#endif
return 0;
}
void
audio_calcwater(struct audio_softc *sc)
{
/* set high at 100% */
if (audio_can_playback(sc)) {
sc->sc_pr.usedhigh =
sc->sc_pustream->end - sc->sc_pustream->start;
/* set low at 75% of usedhigh */
sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4;
if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh)
sc->sc_pr.usedlow -= sc->sc_pr.blksize;
}
if (audio_can_capture(sc)) {
sc->sc_rr.usedhigh =
sc->sc_rustream->end - sc->sc_rustream->start -
sc->sc_rr.blksize;
sc->sc_rr.usedlow = 0;
DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__,
sc->sc_pr.usedlow, sc->sc_pr.usedhigh,
sc->sc_rr.usedlow, sc->sc_rr.usedhigh));
}
}
static inline int
audio_sleep_timo(int *chan, const char *label, int timo)
{
int st;
if (label == NULL)
label = "audio";
DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n",
chan, label, timo));
*chan = 1;
st = tsleep(chan, PWAIT | PCATCH, label, timo);
*chan = 0;
#ifdef AUDIO_DEBUG
if (st != 0 && st != EINTR)
DPRINTF(("audio_sleep: woke up st=%d\n", st));
#endif
return st;
}
static inline int
audio_sleep(int *chan, const char *label)
{
return audio_sleep_timo(chan, label, 0);
}
/* call at splaudio() */
static inline void
audio_wakeup(int *chan)
{
DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan));
if (*chan) {
wakeup(chan);
*chan = 0;
}
}
int
audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l)
{
int error;
u_int mode;
const struct audio_hw_if *hw;
hw = sc->hw_if;
if (hw == NULL)
return ENXIO;
DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n",
flags, sc, sc->hw_hdl));
if (((flags & FREAD) && (sc->sc_open & AUOPEN_READ)) ||
((flags & FWRITE) && (sc->sc_open & AUOPEN_WRITE)))
return EBUSY;
if (hw->open != NULL) {
error = hw->open(sc->hw_hdl, flags);
if (error)
return error;
}
sc->sc_async_audio = 0;
sc->sc_rchan = 0;
sc->sc_wchan = 0;
sc->sc_sil_count = 0;
sc->sc_rbus = false;
sc->sc_pbus = false;
sc->sc_eof = 0;
sc->sc_playdrop = 0;
sc->sc_full_duplex =
(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
(audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX);
mode = 0;
if (flags & FREAD) {
sc->sc_open |= AUOPEN_READ;
mode |= AUMODE_RECORD;
}
if (flags & FWRITE) {
sc->sc_open |= AUOPEN_WRITE;
mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
}
/*
* Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
* The /dev/audio is always (re)set to 8-bit MU-Law mono
* For the other devices, you get what they were last set to.
*/
if (ISDEVAUDIO(dev)) {
error = audio_set_defaults(sc, mode);
} else {
struct audio_info ai;
AUDIO_INITINFO(&ai);
ai.mode = mode;
error = audiosetinfo(sc, &ai);
}
if (error)
goto bad;
#ifdef DIAGNOSTIC
/*
* Sample rate and precision are supposed to be set to proper
* default values by the hardware driver, so that it may give
* us these values.
*/
if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) {
printf("audio_open: 0 precision\n");
return EINVAL;
}
#endif
/* audio_close() decreases sc_pr.usedlow, recalculate here */
audio_calcwater(sc);
DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode));
return 0;
bad:
if (hw->close != NULL)
hw->close(sc->hw_hdl);
sc->sc_open = 0;
sc->sc_mode = 0;
sc->sc_full_duplex = 0;
return error;
}
/*
* Must be called from task context.
*/
void
audio_init_record(struct audio_softc *sc)
{
int s;
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(struct audio_softc *sc)
{
int s;
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(struct audio_softc *sc)
{
struct audio_ringbuffer *cb;
int error, drops;
int s;
int i, used;
DPRINTF(("audio_drain: enter busy=%d\n", sc->sc_pbus));
cb = &sc->sc_pr;
if (cb->mmapped)
return 0;
used = audio_stream_get_used(&sc->sc_pr.s);
s = splaudio();
for (i = 0; i < sc->sc_npfilters; i++)
used += audio_stream_get_used(&sc->sc_pstreams[i]);
splx(s);
if (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;
uint8_t *inp = cb->s.inp;
cc = cb->blksize - (inp - cb->s.start) % cb->blksize;
audio_fill_silence(&cb->s.param, inp, cc);
s = splaudio();
cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc);
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 = false;
}
#endif
drops = cb->drops;
error = 0;
s = splaudio();
while (cb->drops == drops && !error) {
DPRINTF(("audio_drain: used=%d, drops=%ld\n",
audio_stream_get_used(&sc->sc_pr.s), cb->drops));
/*
* When the process is exiting, it ignores all signals and
* we can't interrupt this sleep, so we set a timeout
* just in case.
*/
error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz);
if (sc->sc_dying)
error = EIO;
}
splx(s);
return error;
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
audio_close(struct audio_softc *sc, int flags, int ifmt,
struct lwp *l)
{
const struct audio_hw_if *hw;
int s;
DPRINTF(("audio_close: sc=%p\n", sc));
hw = sc->hw_if;
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 || hw->halt_input != hw->halt_output)
hw->halt_input(sc->hw_hdl);
sc->sc_rbus = false;
}
/*
* 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);
hw->halt_output(sc->hw_hdl);
sc->sc_pbus = false;
}
if (hw->close != NULL)
hw->close(sc->hw_hdl);
sc->sc_open = 0;
sc->sc_mode = 0;
sc->sc_async_audio = 0;
sc->sc_full_duplex = 0;
splx(s);
DPRINTF(("audio_close: done\n"));
return 0;
}
int
audio_read(struct audio_softc *sc, struct uio *uio, int ioflag)
{
struct audio_ringbuffer *cb;
const uint8_t *outp;
uint8_t *inp;
int error, s, used, cc, n;
cb = &sc->sc_rr;
if (cb->mmapped)
return EINVAL;
DPRINTFN(1,("audio_read: cc=%zu mode=%d\n",
uio->uio_resid, sc->sc_mode));
#ifdef AUDIO_PM_IDLE
if (device_is_active(&sc->dev) || sc->sc_idle)
device_active(&sc->dev, DVA_SYSTEM);
#endif
error = 0;
/*
* If hardware is half-duplex and currently playing, return
* silence blocks based on the number of blocks we have output.
*/
if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) {
while (uio->uio_resid > 0 && !error) {
s = splaudio();
for(;;) {
cc = sc->sc_pr.stamp - sc->sc_wstamp;
if (cc > 0)
break;
DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n",
sc->sc_pr.stamp, sc->sc_wstamp));
if (ioflag & IO_NDELAY) {
splx(s);
return EWOULDBLOCK;
}
error = audio_sleep(&sc->sc_rchan, "aud_hr");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return error;
}
}
splx(s);
if (uio->uio_resid < cc)
cc = uio->uio_resid;
DPRINTFN(1,("audio_read: reading in write mode, "
"cc=%d\n", cc));
error = audio_silence_copyout(sc, cc, uio);
sc->sc_wstamp += cc;
}
return error;
}
while (uio->uio_resid > 0 && !error) {
s = splaudio();
while ((used = audio_stream_get_used(sc->sc_rustream)) <= 0) {
if (!sc->sc_rbus && !sc->sc_rr.pause) {
error = audiostartr(sc);
if (error) {
splx(s);
return error;
}
}
if (ioflag & IO_NDELAY) {
splx(s);
return EWOULDBLOCK;
}
DPRINTFN(2, ("audio_read: sleep used=%d\n", used));
error = audio_sleep(&sc->sc_rchan, "aud_rd");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return error;
}
}
outp = sc->sc_rustream->outp;
inp = sc->sc_rustream->inp;
cb->copying = true;
splx(s);
/*
* cc is the amount of data in the sc_rustream excluding
* wrapped data. Note the tricky case of inp == outp, which
* must mean the buffer is full, not empty, because used > 0.
*/
cc = outp < inp ? inp - outp :sc->sc_rustream->end - outp;
DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc));
n = uio->uio_resid;
error = uiomove(__UNCONST(outp), cc, uio);
n -= uio->uio_resid; /* number of bytes actually moved */
s = splaudio();
sc->sc_rustream->outp = audio_stream_add_outp
(sc->sc_rustream, outp, n);
cb->copying = false;
splx(s);
}
return error;
}
void
audio_clear(struct audio_softc *sc)
{
int s;
s = splaudio();
if (sc->sc_rbus) {
audio_wakeup(&sc->sc_rchan);
sc->hw_if->halt_input(sc->hw_hdl);
sc->sc_rbus = false;
sc->sc_rr.pause = false;
}
if (sc->sc_pbus) {
audio_wakeup(&sc->sc_wchan);
sc->hw_if->halt_output(sc->hw_hdl);
sc->sc_pbus = false;
sc->sc_pr.pause = false;
}
splx(s);
}
void
audio_calc_blksize(struct audio_softc *sc, int mode)
{
const audio_params_t *parm;
struct audio_ringbuffer *rb;
if (sc->sc_blkset)
return;
if (mode == AUMODE_PLAY) {
rb = &sc->sc_pr;
parm = &rb->s.param;
} else {
rb = &sc->sc_rr;
parm = &rb->s.param;
}
rb->blksize = parm->sample_rate * audio_blk_ms / 1000 *
parm->channels * parm->precision / NBBY;
DPRINTF(("audio_calc_blksize: %s blksize=%d\n",
mode == AUMODE_PLAY ? "play" : "record", rb->blksize));
}
void
audio_fill_silence(struct audio_params *params, uint8_t *p, int n)
{
uint8_t auzero0, auzero1;
int nfill;
auzero1 = 0; /* initialize to please gcc */
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 any number of bits */
break;
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
if (params->precision > 8) {
nfill = (params->precision + NBBY - 1)/ NBBY;
auzero0 = 0x80;
auzero1 = 0;
} else
auzero0 = 0x80;
break;
default:
DPRINTF(("audio: bad encoding %d\n", params->encoding));
auzero0 = 0;
break;
}
if (nfill == 1) {
while (--n >= 0)
*p++ = auzero0; /* XXX memset */
} else /* nfill must no longer be 2 */ {
if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) {
int k = nfill;
while (--k > 0)
*p++ = auzero1;
n -= nfill - 1;
}
while (n >= nfill) {
int k = nfill;
*p++ = auzero0;
while (--k > 0)
*p++ = auzero1;
n -= nfill;
}
if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */
*p++ = auzero0;
}
}
int
audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio)
{
uint8_t zerobuf[128];
int error;
int k;
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;
}
static int
uio_fetcher_fetch_to(stream_fetcher_t *self, audio_stream_t *p,
int max_used)
{
uio_fetcher_t *this;
int size;
int stream_space;
int error;
this = (uio_fetcher_t *)self;
this->last_used = audio_stream_get_used(p);
if (this->last_used >= this->usedhigh)
return 0;
/*
* uio_fetcher ignores max_used and move the data as
* much as possible in order to return the correct value
* for audio_prinfo::seek and kfilters.
*/
stream_space = audio_stream_get_space(p);
size = min(this->uio->uio_resid, stream_space);
/* the first fragment of the space */
stream_space = p->end - p->inp;
if (stream_space >= size) {
error = uiomove(p->inp, size, this->uio);
if (error)
return error;
p->inp = audio_stream_add_inp(p, p->inp, size);
} else {
error = uiomove(p->inp, stream_space, this->uio);
if (error)
return error;
p->inp = audio_stream_add_inp(p, p->inp, stream_space);
error = uiomove(p->start, size - stream_space, this->uio);
if (error)
return error;
p->inp = audio_stream_add_inp(p, p->inp, size - stream_space);
}
this->last_used = audio_stream_get_used(p);
return 0;
}
static int
null_fetcher_fetch_to(stream_fetcher_t *self,
audio_stream_t *p, int max_used)
{
return 0;
}
static void
uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h)
{
this->base.fetch_to = uio_fetcher_fetch_to;
this->uio = u;
this->usedhigh = h;
}
int
audio_write(struct audio_softc *sc, struct uio *uio, int ioflag)
{
uio_fetcher_t ufetcher;
audio_stream_t stream;
struct audio_ringbuffer *cb;
stream_fetcher_t *fetcher;
stream_filter_t *filter;
uint8_t *inp, *einp;
int saveerror, error, s, n, cc, used;
DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n",
sc, uio->uio_resid, audio_stream_get_used(sc->sc_pustream),
sc->sc_pr.usedhigh));
cb = &sc->sc_pr;
if (cb->mmapped)
return EINVAL;
if (uio->uio_resid == 0) {
sc->sc_eof++;
return 0;
}
#ifdef AUDIO_PM_IDLE
if (device_is_active(&sc->dev) || sc->sc_idle)
device_active(&sc->dev, DVA_SYSTEM);
#endif
/*
* 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;
}
/**
* setup filter pipeline
*/
uio_fetcher_ctor(&ufetcher, uio, cb->usedhigh);
if (sc->sc_npfilters > 0) {
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
} else {
fetcher = &ufetcher.base;
}
error = 0;
while (uio->uio_resid > 0 && !error) {
s = splaudio();
/* wait if the first buffer is occupied */
while ((used = audio_stream_get_used(sc->sc_pustream))
>= cb->usedhigh) {
DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d "
"hiwat=%d\n", used,
cb->usedlow, cb->usedhigh));
if (ioflag & IO_NDELAY) {
splx(s);
return EWOULDBLOCK;
}
error = audio_sleep(&sc->sc_wchan, "aud_wr");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return error;
}
}
inp = cb->s.inp;
cb->copying = true;
stream = cb->s;
used = stream.used;
splx(s);
/*
* write to the sc_pustream as much as possible
*
* work with a temporary audio_stream_t to narrow
* splaudio() enclosure
*/
sc->sc_writing = 1;
if (sc->sc_npfilters > 0) {
filter = sc->sc_pfilters[0];
filter->set_fetcher(filter, &ufetcher.base);
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
cc = cb->blksize * 2;
error = fetcher->fetch_to(fetcher, &stream, cc);
if (error != 0) {
fetcher = &ufetcher.base;
cc = sc->sc_pustream->end - sc->sc_pustream->start;
error = fetcher->fetch_to(fetcher, sc->sc_pustream, cc);
}
} else {
fetcher = &ufetcher.base;
cc = stream.end - stream.start;
error = fetcher->fetch_to(fetcher, &stream, cc);
}
sc->sc_writing = 0;
if (sc->sc_waitcomp)
wakeup(sc);
s = splaudio();
if (sc->sc_npfilters > 0) {
cb->fstamp += ufetcher.last_used
- audio_stream_get_used(sc->sc_pustream);
}
cb->s.used += stream.used - used;
cb->s.inp = stream.inp;
einp = cb->s.inp;
/*
* This is a very suboptimal way of keeping track of
* silence in the buffer, but it is simple.
*/
sc->sc_sil_count = 0;
/*
* 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 < einp &&
(inp - cb->s.start) / cb->blksize ==
(einp - cb->s.start) / cb->blksize) {
/* Figure out how many bytes to a block boundary. */
cc = cb->blksize - (einp - cb->s.start) % cb->blksize;
DPRINTF(("audio_write: partial fill %d\n", cc));
} else
cc = 0;
cb->needfill = false;
cb->copying = false;
if (!sc->sc_pbus && !cb->pause) {
saveerror = error;
error = audiostartp(sc);
if (saveerror != 0) {
/* Report the first error that occurred. */
error = saveerror;
}
}
splx(s);
if (cc != 0) {
DPRINTFN(1, ("audio_write: fill %d\n", cc));
audio_fill_silence(&cb->s.param, einp, cc);
}
}
return error;
}
int
audio_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag,
struct lwp *l)
{
const struct audio_hw_if *hw;
struct audio_offset *ao;
u_long stamp;
int error, s, offs, fd;
bool rbus, pbus;
DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
hw = sc->hw_if;
error = 0;
switch (cmd) {
case FIONBIO:
/* All handled in the upper FS layer. */
break;
case FIONREAD:
*(int *)addr = audio_stream_get_used(sc->sc_rustream);
break;
case FIOASYNC:
if (*(int *)addr) {
if (sc->sc_async_audio)
return EBUSY;
sc->sc_async_audio = l->l_proc;
DPRINTF(("audio_ioctl: FIOASYNC %p\n", l->l_proc));
} else
sc->sc_async_audio = 0;
break;
case AUDIO_FLUSH:
DPRINTF(("AUDIO_FLUSH\n"));
rbus = sc->sc_rbus;
pbus = sc->sc_pbus;
audio_clear(sc);
s = splaudio();
error = audio_initbufs(sc);
if (error) {
splx(s);
return error;
}
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus)
error = audiostartp(sc);
if (!error &&
(sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus)
error = audiostartr(sc);
splx(s);
break;
/*
* Number of read (write) samples dropped. We don't know where or
* when they were dropped.
*/
case AUDIO_RERROR:
*(int *)addr = sc->sc_rr.drops;
break;
case AUDIO_PERROR:
*(int *)addr = sc->sc_pr.drops;
break;
/*
* Offsets into buffer.
*/
case AUDIO_GETIOFFS:
ao = (struct audio_offset *)addr;
s = splaudio();
/* figure out where next DMA will start */
stamp = sc->sc_rustream == &sc->sc_rr.s
? sc->sc_rr.stamp : sc->sc_rr.fstamp;
offs = sc->sc_rustream->inp - sc->sc_rustream->start;
splx(s);
ao->samples = stamp;
ao->deltablks =
(stamp / sc->sc_rr.blksize) -
(sc->sc_rr.stamp_last / sc->sc_rr.blksize);
sc->sc_rr.stamp_last = stamp;
ao->offset = offs;
break;
case AUDIO_GETOOFFS:
ao = (struct audio_offset *)addr;
s = splaudio();
/* figure out where next DMA will start */
stamp = sc->sc_pustream == &sc->sc_pr.s
? sc->sc_pr.stamp : sc->sc_pr.fstamp;
offs = sc->sc_pustream->outp - sc->sc_pustream->start
+ sc->sc_pr.blksize;
splx(s);
ao->samples = stamp;
ao->deltablks =
(stamp / sc->sc_pr.blksize) -
(sc->sc_pr.stamp_last / sc->sc_pr.blksize);
sc->sc_pr.stamp_last = stamp;
if (sc->sc_pustream->start + offs >= sc->sc_pustream->end)
offs = 0;
ao->offset = offs;
break;
/*
* How many bytes will elapse until mike hears the first
* sample of what we write next?
*/
case AUDIO_WSEEK:
*(u_long *)addr = audio_stream_get_used(sc->sc_pustream);
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, 0);
break;
case AUDIO_GETBUFINFO:
DPRINTF(("AUDIO_GETBUFINFO\n"));
error = audiogetinfo(sc, (struct audio_info *)addr, 1);
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 (audio_get_props(sc) & 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 = audio_get_props(sc);
break;
default:
if (hw->dev_ioctl) {
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l);
} else {
DPRINTF(("audio_ioctl: unknown ioctl\n"));
error = EINVAL;
}
break;
}
DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
return error;
}
int
audio_poll(struct audio_softc *sc, int events, struct lwp *l)
{
int revents;
int s;
int used;
DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode));
revents = 0;
s = splaudio();
if (events & (POLLIN | POLLRDNORM)) {
used = audio_stream_get_used(sc->sc_rustream);
/*
* If half duplex and playing, audio_read() will generate
* silence at the play rate; poll for silence being
* available. Otherwise, poll for recorded sound.
*/
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ?
sc->sc_pr.stamp > sc->sc_wstamp :
used > sc->sc_rr.usedlow)
revents |= events & (POLLIN | POLLRDNORM);
}
if (events & (POLLOUT | POLLWRNORM)) {
used = audio_stream_get_used(sc->sc_pustream);
/*
* If half duplex and recording, audio_write() will throw
* away play data, which means we are always ready to write.
* Otherwise, poll for play buffer being below its low water
* mark.
*/
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) ||
(!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) ||
(used <= sc->sc_pr.usedlow))
revents |= events & (POLLOUT | POLLWRNORM);
}
if (revents == 0) {
if (events & (POLLIN | POLLRDNORM))
selrecord(l, &sc->sc_rsel);
if (events & (POLLOUT | POLLWRNORM))
selrecord(l, &sc->sc_wsel);
}
splx(s);
return revents;
}
static void
filt_audiordetach(struct knote *kn)
{
struct audio_softc *sc;
int s;
sc = kn->kn_hook;
s = splaudio();
SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext);
splx(s);
}
static int
filt_audioread(struct knote *kn, long hint)
{
struct audio_softc *sc;
int s;
sc = kn->kn_hook;
s = splaudio();
if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY))
kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp;
else
kn->kn_data = audio_stream_get_used(sc->sc_rustream)
- sc->sc_rr.usedlow;
splx(s);
return kn->kn_data > 0;
}
static const struct filterops audioread_filtops =
{ 1, NULL, filt_audiordetach, filt_audioread };
static void
filt_audiowdetach(struct knote *kn)
{
struct audio_softc *sc;
int s;
sc = kn->kn_hook;
s = splaudio();
SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext);
splx(s);
}
static int
filt_audiowrite(struct knote *kn, long hint)
{
struct audio_softc *sc;
audio_stream_t *stream;
int s;
sc = kn->kn_hook;
s = splaudio();
stream = sc->sc_pustream;
kn->kn_data = (stream->end - stream->start)
- audio_stream_get_used(stream);
splx(s);
return kn->kn_data > 0;
}
static const struct filterops audiowrite_filtops =
{ 1, NULL, filt_audiowdetach, filt_audiowrite };
int
audio_kqfilter(struct audio_softc *sc, struct knote *kn)
{
struct klist *klist;
int s;
switch (kn->kn_filter) {
case EVFILT_READ:
klist = &sc->sc_rsel.sel_klist;
kn->kn_fop = &audioread_filtops;
break;
case EVFILT_WRITE:
klist = &sc->sc_wsel.sel_klist;
kn->kn_fop = &audiowrite_filtops;
break;
default:
return EINVAL;
}
kn->kn_hook = sc;
s = splaudio();
SLIST_INSERT_HEAD(klist, kn, kn_selnext);
splx(s);
return 0;
}
paddr_t
audio_mmap(struct audio_softc *sc, off_t off, int prot)
{
const struct audio_hw_if *hw;
struct audio_ringbuffer *cb;
int s;
DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)off, prot));
hw = sc->hw_if;
if (!(audio_get_props(sc) & AUDIO_PROP_MMAP) || !hw->mappage)
return -1;
#if 0
/* XXX
* The idea here was to use the protection to determine if
* we are mapping the read or write buffer, but it fails.
* The VM system is broken in (at least) two ways.
* 1) If you map memory VM_PROT_WRITE you SIGSEGV
* when writing to it, so VM_PROT_READ|VM_PROT_WRITE
* has to be used for mmapping the play buffer.
* 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE
* audio_mmap will get called at some point with VM_PROT_READ
* only.
* So, alas, we always map the play buffer for now.
*/
if (prot == (VM_PROT_READ|VM_PROT_WRITE) ||
prot == VM_PROT_WRITE)
cb = &sc->sc_pr;
else if (prot == VM_PROT_READ)
cb = &sc->sc_rr;
else
return -1;
#else
cb = &sc->sc_pr;
#endif
if ((u_int)off >= cb->s.bufsize)
return -1;
if (!cb->mmapped) {
cb->mmapped = true;
if (cb == &sc->sc_pr) {
audio_fill_silence(&cb->s.param, cb->s.start,
cb->s.bufsize);
s = splaudio();
sc->sc_pustream = &cb->s;
if (!sc->sc_pbus && !sc->sc_pr.pause)
(void)audiostartp(sc);
splx(s);
} else {
s = splaudio();
sc->sc_rustream = &cb->s;
if (!sc->sc_rbus && !sc->sc_rr.pause)
(void)audiostartr(sc);
splx(s);
}
}
return hw->mappage(sc->hw_hdl, cb->s.start, off, prot);
}
int
audiostartr(struct audio_softc *sc)
{
int error;
DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n",
sc->sc_rr.s.start, audio_stream_get_used(&sc->sc_rr.s),
sc->sc_rr.usedhigh, sc->sc_rr.mmapped));
if (!audio_can_capture(sc))
return EINVAL;
if (sc->hw_if->trigger_input)
error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.s.start,
sc->sc_rr.s.end, sc->sc_rr.blksize,
audio_rint, (void *)sc, &sc->sc_rr.s.param);
else
error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.s.start,
sc->sc_rr.blksize, audio_rint, (void *)sc);
if (error) {
DPRINTF(("audiostartr failed: %d\n", error));
return error;
}
sc->sc_rbus = true;
return 0;
}
int
audiostartp(struct audio_softc *sc)
{
int error;
int used;
used = audio_stream_get_used(&sc->sc_pr.s);
DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n",
sc->sc_pr.s.start, used, sc->sc_pr.usedhigh,
sc->sc_pr.blksize, sc->sc_pr.mmapped));
if (!audio_can_playback(sc))
return EINVAL;
if (!sc->sc_pr.mmapped && used < sc->sc_pr.blksize) {
wakeup(&sc->sc_wchan);
DPRINTF(("%s: wakeup and return\n", __func__));
return 0;
}
if (sc->hw_if->trigger_output) {
DPRINTF(("%s: call trigger_output\n", __func__));
error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.s.start,
sc->sc_pr.s.end, sc->sc_pr.blksize,
audio_pint, (void *)sc, &sc->sc_pr.s.param);
} else {
DPRINTF(("%s: call start_output\n", __func__));
error = sc->hw_if->start_output(sc->hw_hdl,
__UNCONST(sc->sc_pr.s.outp), sc->sc_pr.blksize,
audio_pint, (void *)sc);
}
if (error) {
DPRINTF(("audiostartp failed: %d\n", error));
return error;
}
sc->sc_pbus = true;
return 0;
}
/*
* When the play interrupt routine finds that the write isn't keeping
* the buffer filled it will insert silence in the buffer to make up
* for this. The part of the buffer that is filled with silence
* is kept track of in a very approximate way: it starts at sc_sil_start
* and extends sc_sil_count bytes. If there is already silence in
* the requested area nothing is done; so when the whole buffer is
* silent nothing happens. When the writer starts again sc_sil_count
* is set to 0.
*/
/* XXX
* Putting silence into the output buffer should not really be done
* at splaudio, but there is no softaudio level to do it at yet.
*/
static inline void
audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb,
uint8_t *inp, int cc)
{
uint8_t *s, *e, *p, *q;
if (sc->sc_sil_count > 0) {
s = sc->sc_sil_start; /* start of silence */
e = s + sc->sc_sil_count; /* end of sil., may be beyond end */
p = inp; /* adjusted pointer to area to fill */
if (p < s)
p += cb->s.end - cb->s.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->s.end - cb->s.start)));
audio_fill_silence(&cb->s.param, 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(&cb->s.param, inp, cc);
}
}
static void
audio_softintr_rd(void *cookie)
{
struct audio_softc *sc = cookie;
struct proc *p;
audio_wakeup(&sc->sc_rchan);
selnotify(&sc->sc_rsel, 0, 0);
if (sc->sc_async_audio != NULL) {
DPRINTFN(3, ("audio_softintr_rd: sending SIGIO %p\n",
sc->sc_async_audio));
mutex_enter(proc_lock);
if ((p = sc->sc_async_audio) != NULL)
psignal(p, SIGIO);
mutex_exit(proc_lock);
}
}
static void
audio_softintr_wr(void *cookie)
{
struct audio_softc *sc = cookie;
struct proc *p;
audio_wakeup(&sc->sc_wchan);
selnotify(&sc->sc_wsel, 0, 0);
if (sc->sc_async_audio != NULL) {
DPRINTFN(3, ("audio_softintr_wr: sending SIGIO %p\n",
sc->sc_async_audio));
mutex_enter(proc_lock);
if ((p = sc->sc_async_audio) != NULL)
psignal(p, SIGIO);
mutex_exit(proc_lock);
}
}
/*
* Called from HW driver module on completion of DMA output.
* Start output of new block, wrap in ring buffer if needed.
* If no more buffers to play, output zero instead.
* Do a wakeup if necessary.
*/
void
audio_pint(void *v)
{
stream_fetcher_t null_fetcher;
struct audio_softc *sc;
const struct audio_hw_if *hw;
struct audio_ringbuffer *cb;
stream_fetcher_t *fetcher;
uint8_t *inp;
int cc, used;
int blksize;
int error;
sc = v;
if (!sc->sc_open)
return; /* ignore interrupt if not open */
hw = sc->hw_if;
cb = &sc->sc_pr;
blksize = cb->blksize;
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
cb->stamp += blksize;
if (cb->mmapped) {
DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n",
cb->s.outp, blksize, cb->s.inp));
if (hw->trigger_output == NULL)
(void)hw->start_output(sc->hw_hdl, __UNCONST(cb->s.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
used = audio_stream_get_used(&cb->s);
/*
* "used <= cb->usedlow" should be "used < blksize" ideally.
* Some HW drivers such as uaudio(4) does not call audio_pint()
* at accurate timing. If used < blksize, uaudio(4) already
* request transfer of garbage data.
*/
if (used <= cb->usedlow && !cb->copying && sc->sc_npfilters > 0) {
/* we might have data in filter pipeline */
null_fetcher.fetch_to = null_fetcher_fetch_to;
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
sc->sc_pfilters[0]->set_fetcher(sc->sc_pfilters[0],
&null_fetcher);
used = audio_stream_get_used(sc->sc_pustream);
cc = cb->s.end - cb->s.start;
if (blksize * 2 < cc)
cc = blksize * 2;
fetcher->fetch_to(fetcher, &cb->s, cc);
cb->fstamp += used - audio_stream_get_used(sc->sc_pustream);
used = audio_stream_get_used(&cb->s);
}
if (used < blksize) {
/* we don't have a full block to use */
if (cb->copying) {
/* writer is in progress, don't disturb */
cb->needfill = true;
DPRINTFN(1, ("audio_pint: copying in progress\n"));
} else {
inp = cb->s.inp;
cc = blksize - (inp - cb->s.start) % blksize;
if (cb->pause)
cb->pdrops += cc;
else {
cb->drops += cc;
sc->sc_playdrop += cc;
}
audio_pint_silence(sc, cb, inp, cc);
cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc);
/* Clear next block so we keep ahead of the DMA. */
used = audio_stream_get_used(&cb->s);
if (used + blksize < cb->s.end - cb->s.start)
audio_pint_silence(sc, cb, cb->s.inp, blksize);
}
}
DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->s.outp, blksize));
if (hw->trigger_output == NULL) {
error = hw->start_output(sc->hw_hdl, __UNCONST(cb->s.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,
audio_stream_get_used(sc->sc_pustream), cb->usedlow));
if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
if (audio_stream_get_used(sc->sc_pustream) <= cb->usedlow)
softint_schedule(sc->sc_sih_wr);
}
/* Possible to return one or more "phantom blocks" now. */
if (!sc->sc_full_duplex && sc->sc_rchan)
softint_schedule(sc->sc_sih_rd);
}
/*
* Called from HW driver module on completion of DMA input.
* Mark it as input in the ring buffer (fiddle pointers).
* Do a wakeup if necessary.
*/
void
audio_rint(void *v)
{
stream_fetcher_t null_fetcher;
struct audio_softc *sc;
const struct audio_hw_if *hw;
struct audio_ringbuffer *cb;
stream_fetcher_t *last_fetcher;
int cc;
int used;
int blksize;
int error;
sc = v;
cb = &sc->sc_rr;
if (!sc->sc_open)
return; /* ignore interrupt if not open */
hw = sc->hw_if;
blksize = cb->blksize;
cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize);
cb->stamp += blksize;
if (cb->mmapped) {
DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n",
cb->s.inp, blksize));
if (hw->trigger_input == NULL)
(void)hw->start_input(sc->hw_hdl, cb->s.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
if (!cb->pause && sc->sc_nrfilters > 0) {
null_fetcher.fetch_to = null_fetcher_fetch_to;
last_fetcher = &sc->sc_rfilters[sc->sc_nrfilters - 1]->base;
sc->sc_rfilters[0]->set_fetcher(sc->sc_rfilters[0],
&null_fetcher);
used = audio_stream_get_used(sc->sc_rustream);
cc = sc->sc_rustream->end - sc->sc_rustream->start;
error = last_fetcher->fetch_to
(last_fetcher, sc->sc_rustream, cc);
cb->fstamp += audio_stream_get_used(sc->sc_rustream) - used;
/* XXX what should do for error? */
}
used = audio_stream_get_used(&sc->sc_rr.s);
if (cb->pause) {
DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops));
cb->pdrops += blksize;
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
} else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) {
DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops));
cb->drops += blksize;
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
}
DPRINTFN(2, ("audio_rint: inp=%p cc=%d\n", cb->s.inp, blksize));
if (hw->trigger_input == NULL) {
error = hw->start_input(sc->hw_hdl, cb->s.inp, blksize,
audio_rint, (void *)sc);
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_rint: restart failed: %d\n", error));
audio_clear(sc);
}
}
softint_schedule(sc->sc_sih_rd);
}
int
audio_check_params(struct audio_params *p)
{
if (p->encoding == AUDIO_ENCODING_PCM16) {
if (p->precision == 8)
p->encoding = AUDIO_ENCODING_ULINEAR;
else
p->encoding = AUDIO_ENCODING_SLINEAR;
} else if (p->encoding == AUDIO_ENCODING_PCM8) {
if (p->precision == 8)
p->encoding = AUDIO_ENCODING_ULINEAR;
else
return EINVAL;
}
if (p->encoding == AUDIO_ENCODING_SLINEAR)
#if BYTE_ORDER == LITTLE_ENDIAN
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
#else
p->encoding = AUDIO_ENCODING_SLINEAR_BE;
#endif
if (p->encoding == AUDIO_ENCODING_ULINEAR)
#if BYTE_ORDER == LITTLE_ENDIAN
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
#else
p->encoding = AUDIO_ENCODING_ULINEAR_BE;
#endif
switch (p->encoding) {
case AUDIO_ENCODING_ULAW:
case AUDIO_ENCODING_ALAW:
if (p->precision != 8)
return EINVAL;
break;
case AUDIO_ENCODING_ADPCM:
if (p->precision != 4 && p->precision != 8)
return EINVAL;
break;
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
/* XXX is: our zero-fill can handle any multiple of 8 */
if (p->precision != 8 && p->precision != 16 &&
p->precision != 24 && p->precision != 32)
return EINVAL;
if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE)
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE)
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
if (p->validbits > p->precision)
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;
}
/* sanity check # of channels*/
if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS)
return EINVAL;
return 0;
}
int
audio_set_defaults(struct audio_softc *sc, u_int mode)
{
struct audio_info ai;
/* default parameters */
sc->sc_rparams = audio_default;
sc->sc_pparams = audio_default;
sc->sc_blkset = false;
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.record.pause = false;
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.play.pause = false;
ai.mode = mode;
return audiosetinfo(sc, &ai);
}
int
au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r)
{
ct->type = AUDIO_MIXER_VALUE;
ct->un.value.num_channels = 2;
ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0)
return 0;
ct->un.value.num_channels = 1;
ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
return sc->hw_if->set_port(sc->hw_hdl, ct);
}
int
au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
int gain, int balance)
{
mixer_ctrl_t ct;
int i, error;
int l, r;
u_int mask;
int nset;
if (balance == AUDIO_MID_BALANCE) {
l = r = gain;
} else if (balance < AUDIO_MID_BALANCE) {
l = gain;
r = (balance * gain) / AUDIO_MID_BALANCE;
} else {
r = gain;
l = ((AUDIO_RIGHT_BALANCE - balance) * gain)
/ AUDIO_MID_BALANCE;
}
DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n",
gain, balance, l, r));
if (ports->index == -1) {
usemaster:
if (ports->master == -1)
return 0; /* just ignore it silently */
ct.dev = ports->master;
error = au_set_lr_value(sc, &ct, l, r);
} else {
ct.dev = ports->index;
if (ports->isenum) {
ct.type = AUDIO_MIXER_ENUM;
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
if (error)
return error;
if (ports->isdual) {
if (ports->cur_port == -1)
ct.dev = ports->master;
else
ct.dev = ports->miport[ports->cur_port];
error = au_set_lr_value(sc, &ct, l, r);
} else {
for(i = 0; i < ports->nports; i++)
if (ports->misel[i] == ct.un.ord) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_set_lr_value(sc, &ct, l, r))
goto usemaster;
else
break;
}
}
} else {
ct.type = AUDIO_MIXER_SET;
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
if (error)
return error;
mask = ct.un.mask;
nset = 0;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] & mask) {
ct.dev = ports->miport[i];
if (ct.dev != -1 &&
au_set_lr_value(sc, &ct, l, r) == 0)
nset++;
}
}
if (nset == 0)
goto usemaster;
}
}
if (!error)
mixer_signal(sc);
return error;
}
int
au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r)
{
int error;
ct->un.value.num_channels = 2;
if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) {
*l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
*r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
} else {
ct->un.value.num_channels = 1;
error = sc->hw_if->get_port(sc->hw_hdl, ct);
if (error)
return error;
*r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO];
}
return 0;
}
void
au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
u_int *pgain, u_char *pbalance)
{
mixer_ctrl_t ct;
int i, l, r, n;
int lgain, rgain;
lgain = AUDIO_MAX_GAIN / 2;
rgain = AUDIO_MAX_GAIN / 2;
if (ports->index == -1) {
usemaster:
if (ports->master == -1)
goto bad;
ct.dev = ports->master;
ct.type = AUDIO_MIXER_VALUE;
if (au_get_lr_value(sc, &ct, &lgain, &rgain))
goto bad;
} else {
ct.dev = ports->index;
if (ports->isenum) {
ct.type = AUDIO_MIXER_ENUM;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
goto bad;
ct.type = AUDIO_MIXER_VALUE;
if (ports->isdual) {
if (ports->cur_port == -1)
ct.dev = ports->master;
else
ct.dev = ports->miport[ports->cur_port];
au_get_lr_value(sc, &ct, &lgain, &rgain);
} else {
for(i = 0; i < ports->nports; i++)
if (ports->misel[i] == ct.un.ord) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_get_lr_value(sc, &ct,
&lgain, &rgain))
goto usemaster;
else
break;
}
}
} else {
ct.type = AUDIO_MIXER_SET;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
goto bad;
ct.type = AUDIO_MIXER_VALUE;
lgain = rgain = n = 0;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] & ct.un.mask) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_get_lr_value(sc, &ct, &l, &r))
goto usemaster;
else {
lgain += l;
rgain += r;
n++;
}
}
}
if (n != 0) {
lgain /= n;
rgain /= n;
}
}
}
bad:
if (lgain == rgain) { /* handles lgain==rgain==0 */
*pgain = lgain;
*pbalance = AUDIO_MID_BALANCE;
} else if (lgain < rgain) {
*pgain = rgain;
/* balance should be > AUDIO_MID_BALANCE */
*pbalance = AUDIO_RIGHT_BALANCE -
(AUDIO_MID_BALANCE * lgain) / rgain;
} else /* lgain > rgain */ {
*pgain = lgain;
/* balance should be < AUDIO_MID_BALANCE */
*pbalance = (AUDIO_MID_BALANCE * rgain) / lgain;
}
}
int
au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port)
{
mixer_ctrl_t ct;
int i, error, use_mixerout;
use_mixerout = 1;
if (port == 0) {
if (ports->allports == 0)
return 0; /* Allow this special case. */
else if (ports->isdual) {
if (ports->cur_port == -1) {
return 0;
} else {
port = ports->aumask[ports->cur_port];
ports->cur_port = -1;
use_mixerout = 0;
}
}
}
if (ports->index == -1)
return EINVAL;
ct.dev = ports->index;
if (ports->isenum) {
if (port & (port-1))
return EINVAL; /* Only one port allowed */
ct.type = AUDIO_MIXER_ENUM;
error = EINVAL;
for(i = 0; i < ports->nports; i++)
if (ports->aumask[i] == port) {
if (ports->isdual && use_mixerout) {
ct.un.ord = ports->mixerout;
ports->cur_port = i;
} else {
ct.un.ord = ports->misel[i];
}
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
break;
}
} else {
ct.type = AUDIO_MIXER_SET;
ct.un.mask = 0;
for(i = 0; i < ports->nports; i++)
if (ports->aumask[i] & port)
ct.un.mask |= ports->misel[i];
if (port != 0 && ct.un.mask == 0)
error = EINVAL;
else
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
}
if (!error)
mixer_signal(sc);
return error;
}
int
au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports)
{
mixer_ctrl_t ct;
int i, aumask;
if (ports->index == -1)
return 0;
ct.dev = ports->index;
ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
return 0;
aumask = 0;
if (ports->isenum) {
if (ports->isdual && ports->cur_port != -1) {
if (ports->mixerout == ct.un.ord)
aumask = ports->aumask[ports->cur_port];
else
ports->cur_port = -1;
}
if (aumask == 0)
for(i = 0; i < ports->nports; i++)
if (ports->misel[i] == ct.un.ord)
aumask = ports->aumask[i];
} else {
for(i = 0; i < ports->nports; i++)
if (ct.un.mask & ports->misel[i])
aumask |= ports->aumask[i];
}
return aumask;
}
int
audiosetinfo(struct audio_softc *sc, struct audio_info *ai)
{
stream_filter_list_t pfilters, rfilters;
audio_params_t pp, rp;
struct audio_prinfo *r, *p;
const struct audio_hw_if *hw;
audio_stream_t *oldpus, *oldrus;
int s, setmode;
int error;
int np, nr;
unsigned int blks;
int oldpblksize, oldrblksize;
u_int gain;
bool rbus, pbus;
bool cleared, modechange, pausechange;
u_char balance;
hw = sc->hw_if;
if (hw == NULL) /* HW has not attached */
return ENXIO;
DPRINTF(("%s sc=%p ai=%p\n", __func__, sc, ai));
r = &ai->record;
p = &ai->play;
rbus = sc->sc_rbus;
pbus = sc->sc_pbus;
error = 0;
cleared = false;
modechange = false;
pausechange = false;
pp = sc->sc_pparams; /* Temporary encoding storage in */
rp = sc->sc_rparams; /* case setting the modes fails. */
nr = np = 0;
if (SPECIFIED(p->sample_rate)) {
pp.sample_rate = p->sample_rate;
np++;
}
if (SPECIFIED(r->sample_rate)) {
rp.sample_rate = r->sample_rate;
nr++;
}
if (SPECIFIED(p->encoding)) {
pp.encoding = p->encoding;
np++;
}
if (SPECIFIED(r->encoding)) {
rp.encoding = r->encoding;
nr++;
}
if (SPECIFIED(p->precision)) {
pp.precision = p->precision;
/* we don't have API to specify validbits */
pp.validbits = p->precision;
np++;
}
if (SPECIFIED(r->precision)) {
rp.precision = r->precision;
/* we don't have API to specify validbits */
rp.validbits = r->precision;
nr++;
}
if (SPECIFIED(p->channels)) {
pp.channels = p->channels;
np++;
}
if (SPECIFIED(r->channels)) {
rp.channels = r->channels;
nr++;
}
if (!audio_can_capture(sc))
nr = 0;
if (!audio_can_playback(sc))
np = 0;
#ifdef AUDIO_DEBUG
if (audiodebug && nr > 0)
audio_print_params("audiosetinfo() Setting record params:", &rp);
if (audiodebug && np > 0)
audio_print_params("audiosetinfo() Setting play params:", &pp);
#endif
if (nr > 0 && (error = audio_check_params(&rp)))
return error;
if (np > 0 && (error = audio_check_params(&pp)))
return error;
oldpblksize = sc->sc_pr.blksize;
oldrblksize = sc->sc_rr.blksize;
setmode = 0;
if (nr > 0) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
modechange = true;
setmode |= AUMODE_RECORD;
}
if (np > 0) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
modechange = true;
setmode |= AUMODE_PLAY;
}
if (SPECIFIED(ai->mode)) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
modechange = true;
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;
}
oldpus = sc->sc_pustream;
oldrus = sc->sc_rustream;
if (modechange) {
int indep;
indep = audio_get_props(sc) & AUDIO_PROP_INDEPENDENT;
if (!indep) {
if (setmode == AUMODE_RECORD)
pp = rp;
else if (setmode == AUMODE_PLAY)
rp = pp;
}
memset(&pfilters, 0, sizeof(pfilters));
memset(&rfilters, 0, sizeof(rfilters));
pfilters.append = stream_filter_list_append;
pfilters.prepend = stream_filter_list_prepend;
pfilters.set = stream_filter_list_set;
rfilters.append = stream_filter_list_append;
rfilters.prepend = stream_filter_list_prepend;
rfilters.set = stream_filter_list_set;
/* Some device drivers change channels/sample_rate and change
* no channels/sample_rate. */
error = hw->set_params(sc->hw_hdl, setmode,
sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp,
&pfilters, &rfilters);
if (error) {
DPRINTF(("%s: hw->set_params() failed with %d\n",
__func__, error));
goto cleanup;
}
audio_check_params(&pp);
audio_check_params(&rp);
if (!indep) {
/* XXX for !indep device, we have to use the same
* parameters for the hardware, not userland */
if (setmode == AUMODE_RECORD) {
pp = rp;
} else if (setmode == AUMODE_PLAY) {
rp = pp;
}
}
if (sc->sc_pr.mmapped && pfilters.req_size > 0) {
DPRINTF(("%s: mmapped, and filters are requested.\n",
__func__));
error = EINVAL;
goto cleanup;
}
/* construct new filter chain */
if (setmode & AUMODE_PLAY) {
error = audio_setup_pfilters(sc, &pp, &pfilters);
if (error)
goto cleanup;
}
if (setmode & AUMODE_RECORD) {
error = audio_setup_rfilters(sc, &rp, &rfilters);
if (error)
goto cleanup;
}
DPRINTF(("%s: filter setup is completed.\n", __func__));
/* userland formats */
sc->sc_pparams = pp;
sc->sc_rparams = rp;
}
/* Play params can affect the record params, so recalculate blksize. */
if (nr > 0 || np > 0) {
audio_calc_blksize(sc, AUMODE_RECORD);
audio_calc_blksize(sc, AUMODE_PLAY);
}
#ifdef AUDIO_DEBUG
if (audiodebug > 1 && nr > 0)
audio_print_params("audiosetinfo() After setting record params:", &sc->sc_rparams);
if (audiodebug > 1 && np > 0)
audio_print_params("audiosetinfo() After setting play params:", &sc->sc_pparams);
#endif
if (SPECIFIED(p->port)) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
error = au_set_port(sc, &sc->sc_outports, p->port);
if (error)
goto cleanup;
}
if (SPECIFIED(r->port)) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
error = au_set_port(sc, &sc->sc_inports, r->port);
if (error)
goto cleanup;
}
if (SPECIFIED(p->gain)) {
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_outports, p->gain, balance);
if (error)
goto cleanup;
}
if (SPECIFIED(r->gain)) {
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_inports, r->gain, balance);
if (error)
goto cleanup;
}
if (SPECIFIED_CH(p->balance)) {
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_outports, gain, p->balance);
if (error)
goto cleanup;
}
if (SPECIFIED_CH(r->balance)) {
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_inports, gain, r->balance);
if (error)
goto cleanup;
}
if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) {
mixer_ctrl_t ct;
ct.dev = sc->sc_monitor_port;
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain;
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
if (error)
goto cleanup;
}
if (SPECIFIED_CH(p->pause)) {
sc->sc_pr.pause = p->pause;
pbus = !p->pause;
pausechange = true;
}
if (SPECIFIED_CH(r->pause)) {
sc->sc_rr.pause = r->pause;
rbus = !r->pause;
pausechange = true;
}
if (SPECIFIED(ai->blocksize)) {
int pblksize, rblksize;
/* Block size specified explicitly. */
if (ai->blocksize == 0) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
sc->sc_blkset = false;
audio_calc_blksize(sc, AUMODE_RECORD);
audio_calc_blksize(sc, AUMODE_PLAY);
} else {
sc->sc_blkset = true;
/* check whether new blocksize changes actually */
if (hw->round_blocksize == NULL) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
sc->sc_pr.blksize = ai->blocksize;
sc->sc_rr.blksize = ai->blocksize;
} else {
pblksize = hw->round_blocksize(sc->hw_hdl,
ai->blocksize, AUMODE_PLAY, &sc->sc_pr.s.param);
rblksize = hw->round_blocksize(sc->hw_hdl,
ai->blocksize, AUMODE_RECORD, &sc->sc_rr.s.param);
if (pblksize != sc->sc_pr.blksize ||
rblksize != sc->sc_rr.blksize) {
if (!cleared) {
audio_clear(sc);
cleared = true;
}
sc->sc_pr.blksize = ai->blocksize;
sc->sc_rr.blksize = ai->blocksize;
}
}
}
}
if (SPECIFIED(ai->mode)) {
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)
goto cleanup;
}
sc->sc_lastinfo = *ai;
sc->sc_lastinfovalid = true;
cleanup:
if (cleared || pausechange) {
int init_error;
s = splaudio();
init_error = audio_initbufs(sc);
if (init_error) goto err;
if (sc->sc_pr.blksize != oldpblksize ||
sc->sc_rr.blksize != oldrblksize ||
sc->sc_pustream != oldpus ||
sc->sc_rustream != oldrus)
audio_calcwater(sc);
if ((sc->sc_mode & AUMODE_PLAY) &&
pbus && !sc->sc_pbus)
init_error = audiostartp(sc);
if (!init_error &&
(sc->sc_mode & AUMODE_RECORD) &&
rbus && !sc->sc_rbus)
init_error = audiostartr(sc);
err:
splx(s);
if (init_error)
return init_error;
}
/* Change water marks after initializing the buffers. */
if (SPECIFIED(ai->hiwat)) {
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 (SPECIFIED(ai->lowat)) {
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 (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) {
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 error;
}
int
audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int buf_only_mode)
{
struct audio_prinfo *r, *p;
const struct audio_hw_if *hw;
r = &ai->record;
p = &ai->play;
hw = sc->hw_if;
if (hw == NULL) /* 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;
if (buf_only_mode) {
r->port = 0;
p->port = 0;
r->avail_ports = 0;
p->avail_ports = 0;
r->gain = 0;
r->balance = 0;
p->gain = 0;
p->balance = 0;
} else {
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 && buf_only_mode == 0) {
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 = audio_stream_get_used(sc->sc_pustream);
r->seek = audio_stream_get_used(sc->sc_rustream);
/*
* XXX samples should be a value for userland data.
* But drops is a value for HW data.
*/
p->samples = (sc->sc_pustream == &sc->sc_pr.s
? sc->sc_pr.stamp : sc->sc_pr.fstamp) - sc->sc_pr.drops;
r->samples = (sc->sc_rustream == &sc->sc_rr.s
? sc->sc_rr.stamp : sc->sc_rr.fstamp) - 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_pustream ? sc->sc_pustream->bufsize : 0;
r->buffer_size = sc->sc_rustream ? sc->sc_rustream->bufsize : 0;
ai->blocksize = sc->sc_pr.blksize;
if (sc->sc_pr.blksize > 0) {
ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize;
ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize;
} else
ai->hiwat = ai->lowat = 0;
ai->mode = sc->sc_mode;
return 0;
}
/*
* Mixer driver
*/
int
mixer_open(dev_t dev, struct audio_softc *sc, int flags,
int ifmt, struct lwp *l)
{
if (sc->hw_if == NULL)
return ENXIO;
DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc));
return 0;
}
/*
* Remove a process from those to be signalled on mixer activity.
*/
static void
mixer_remove(struct audio_softc *sc, struct lwp *l)
{
struct mixer_asyncs **pm, *m;
struct proc *p;
if (l == NULL)
return;
p = l->l_proc;
for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
if ((*pm)->proc == p) {
m = *pm;
*pm = m->next;
free(m, M_DEVBUF);
return;
}
}
}
/*
* Signal all processes waiting for the mixer.
*/
static void
mixer_signal(struct audio_softc *sc)
{
struct mixer_asyncs *m;
for (m = sc->sc_async_mixer; m; m = m->next) {
mutex_enter(proc_lock);
psignal(m->proc, SIGIO);
mutex_exit(proc_lock);
}
}
/*
* Close a mixer device
*/
/* ARGSUSED */
int
mixer_close(struct audio_softc *sc, int flags, int ifmt,
struct lwp *l)
{
DPRINTF(("mixer_close: sc %p\n", sc));
mixer_remove(sc, l);
return 0;
}
int
mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag,
struct lwp *l)
{
const struct audio_hw_if *hw;
mixer_ctrl_t *mc;
int error;
DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
hw = sc->hw_if;
error = EINVAL;
/* we can return cached values if we are sleeping */
if (cmd != AUDIO_MIXER_READ)
device_active(sc->dev, DVA_SYSTEM);
switch (cmd) {
case FIOASYNC:
mixer_remove(sc, l); /* 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 = l->l_proc;
sc->sc_async_mixer = ma;
}
error = 0;
break;
case AUDIO_GETDEV:
DPRINTF(("AUDIO_GETDEV\n"));
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
break;
case AUDIO_MIXER_DEVINFO:
DPRINTF(("AUDIO_MIXER_DEVINFO\n"));
((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */
error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr);
break;
case AUDIO_MIXER_READ:
DPRINTF(("AUDIO_MIXER_READ\n"));
mc = (mixer_ctrl_t *)addr;
if (device_is_active(sc->sc_dev) ||
sc->sc_mixer_state == NULL)
error = hw->get_port(sc->hw_hdl, mc);
else if (mc->dev >= sc->sc_nmixer_states)
error = ENXIO;
else {
int dev = mc->dev;
memcpy(mc, &sc->sc_mixer_state[dev],
sizeof(mixer_ctrl_t));
error = 0;
}
break;
case AUDIO_MIXER_WRITE:
DPRINTF(("AUDIO_MIXER_WRITE\n"));
error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
if (!error && hw->commit_settings)
error = hw->commit_settings(sc->hw_hdl);
if (!error)
mixer_signal(sc);
break;
default:
if (hw->dev_ioctl)
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l);
else
error = EINVAL;
break;
}
DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
return error;
}
#endif /* NAUDIO > 0 */
#include "midi.h"
#if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0)
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/device.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#endif
#if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0)
int
audioprint(void *aux, const char *pnp)
{
struct audio_attach_args *arg;
const char *type;
if (pnp != NULL) {
arg = aux;
switch (arg->type) {
case AUDIODEV_TYPE_AUDIO:
type = "audio";
break;
case AUDIODEV_TYPE_MIDI:
type = "midi";
break;
case AUDIODEV_TYPE_OPL:
type = "opl";
break;
case AUDIODEV_TYPE_MPU:
type = "mpu";
break;
default:
panic("audioprint: unknown type %d", arg->type);
}
aprint_normal("%s at %s", type, pnp);
}
return UNCONF;
}
#endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */
#if NAUDIO > 0
static void
audio_mixer_capture(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
for (mi.index = 0; ; mi.index++)
if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0)
break;
#ifdef DIAGNOSTIC
if (sc->sc_mixer_state != NULL && sc->sc_nmixer_states != mi.index) {
free(sc->sc_mixer_state, M_DEVBUF);
sc->sc_mixer_state = NULL;
}
#endif
sc->sc_nmixer_states = mi.index;
if (sc->sc_mixer_state == NULL)
sc->sc_mixer_state = malloc(
sizeof(mixer_ctrl_t) * sc->sc_nmixer_states,
M_DEVBUF, M_NOWAIT);
if (sc->sc_mixer_state == NULL) {
aprint_error("%s: couldn't allocate memory for mixer state\n",
device_xname(sc->dev));
return;
}
for (mi.index = 0; ; mi.index++) {
if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS)
continue;
mc = &sc->sc_mixer_state[mi.index];
mc->dev = mi.index;
mc->type = mi.type;
mc->un.value.num_channels = mi.un.v.num_channels;
(void)sc->hw_if->get_port(sc->hw_hdl, mc);
}
return;
}
static void
audio_mixer_restore(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
if (sc->sc_mixer_state == NULL)
return;
for (mi.index = 0; ; mi.index++) {
if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS)
continue;
mc = &sc->sc_mixer_state[mi.index];
(void)sc->hw_if->set_port(sc->hw_hdl, mc);
}
if (sc->hw_if->commit_settings)
sc->hw_if->commit_settings(sc->hw_hdl);
return;
}
#ifdef AUDIO_PM_IDLE
static void
audio_idle(void *arg)
{
device_t dv = arg;
struct audio_softc *sc = device_private(dv);
#ifdef PNP_DEBUG
extern int pnp_debug_idle;
if (pnp_debug_idle)
printf("%s: idle handler called\n", device_xname(dv));
#endif
sc->sc_idle = true;
/* XXX joerg Make pmf_device_suspend handle children? */
if (!pmf_device_suspend(dv, PMF_Q_SELF))
return;
if (!pmf_device_suspend(sc->sc_dev, PMF_Q_SELF))
pmf_device_resume(dv, PMF_Q_SELF);
}
static void
audio_activity(device_t dv, devactive_t type)
{
struct audio_softc *sc = device_private(dv);
if (type != DVA_SYSTEM)
return;
callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz);
sc->sc_idle = false;
if (!device_is_active(dv)) {
/* XXX joerg How to deal with a failing resume... */
pmf_device_resume(sc->sc_dev, PMF_Q_SELF);
pmf_device_resume(dv, PMF_Q_SELF);
}
}
#endif
static bool
audio_suspend(device_t dv, const pmf_qual_t *qual)
{
struct audio_softc *sc = device_private(dv);
const struct audio_hw_if *hwp = sc->hw_if;
int (s);
s = splaudio();
audio_mixer_capture(sc);
if (sc->sc_pbus == true)
hwp->halt_output(sc->hw_hdl);
if (sc->sc_rbus == true)
hwp->halt_input(sc->hw_hdl);
#ifdef AUDIO_PM_IDLE
callout_stop(&sc->sc_idle_counter);
#endif
splx(s);
return true;
}
static bool
audio_resume(device_t dv, const pmf_qual_t *qual)
{
struct audio_softc *sc = device_private(dv);
int s;
s = splaudio();
if (sc->sc_lastinfovalid)
audiosetinfo(sc, &sc->sc_lastinfo);
audio_mixer_restore(sc);
if ((sc->sc_pbus == true) && !sc->sc_pr.pause)
audiostartp(sc);
if ((sc->sc_rbus == true) && !sc->sc_rr.pause)
audiostartr(sc);
splx(s);
return true;
}
static void
audio_volume_down(device_t dv)
{
struct audio_softc *sc = device_private(dv);
mixer_devinfo_t mi;
int newgain;
u_int gain;
u_char balance;
int s;
if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) {
mi.index = sc->sc_outports.master;
mi.un.v.delta = 0;
if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0)
return;
s = splaudio();
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
newgain = gain - mi.un.v.delta;
if (newgain < AUDIO_MIN_GAIN)
newgain = AUDIO_MIN_GAIN;
au_set_gain(sc, &sc->sc_outports, newgain, balance);
splx(s);
}
}
static void
audio_volume_up(device_t dv)
{
struct audio_softc *sc = device_private(dv);
mixer_devinfo_t mi;
u_int gain, newgain;
u_char balance;
int s;
if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) {
mi.index = sc->sc_outports.master;
mi.un.v.delta = 0;
if (sc->hw_if->query_devinfo(sc->hw_hdl, &mi) != 0)
return;
s = splaudio();
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
newgain = gain + mi.un.v.delta;
if (newgain > AUDIO_MAX_GAIN)
newgain = AUDIO_MAX_GAIN;
au_set_gain(sc, &sc->sc_outports, newgain, balance);
splx(s);
}
}
static void
audio_volume_toggle(device_t dv)
{
struct audio_softc *sc = device_private(dv);
u_int gain, newgain;
u_char balance;
int s;
s = splaudio();
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
if (gain != 0) {
sc->sc_lastgain = gain;
newgain = 0;
} else
newgain = sc->sc_lastgain;
au_set_gain(sc, &sc->sc_outports, newgain, balance);
splx(s);
}
static int
audio_get_props(struct audio_softc *sc)
{
const struct audio_hw_if *hw;
int props;
hw = sc->hw_if;
props = hw->get_props(sc->hw_hdl);
/*
* if neither playback nor capture properties are reported,
* assume both are supported by the device driver
*/
if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0)
props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE);
return props;
}
static bool
audio_can_playback(struct audio_softc *sc)
{
return audio_get_props(sc) & AUDIO_PROP_PLAYBACK ? true : false;
}
static bool
audio_can_capture(struct audio_softc *sc)
{
return audio_get_props(sc) & AUDIO_PROP_CAPTURE ? true : false;
}
#endif /* NAUDIO > 0 */
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