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/* $NetBSD: audio.c,v 1.459 2019/02/27 02:27:38 mrg Exp $ */
/*-
* Copyright (c) 2016 Nathanial Sloss <nathanialsloss@yahoo.com.au>
* All rights reserved.
*
* Copyright (c) 2008 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by Andrew Doran.
*
* 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.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. 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 FOUNDATION 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.
*/
/*
* 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.
*/
/*
* Locking: there are two locks.
*
* - sc_lock, provided by the underlying driver. This is an adaptive lock,
* returned in the second parameter to hw_if->get_locks(). It is known
* as the "thread lock".
*
* It serializes access to state in all places except the
* driver's interrupt service routine. This lock is taken from process
* context (example: access to /dev/audio). It is also taken from soft
* interrupt handlers in this module, primarily to serialize delivery of
* wakeups. This lock may be used/provided by modules external to the
* audio subsystem, so take care not to introduce a lock order problem.
* LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD.
*
* - sc_intr_lock, provided by the underlying driver. This may be either a
* spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or
* IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It
* is known as the "interrupt lock".
*
* It provides atomic access to the device's hardware state, and to audio
* channel data that may be accessed by the hardware driver's ISR.
* In all places outside the ISR, sc_lock must be held before taking
* sc_intr_lock. This is to ensure that groups of hardware operations are
* made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD.
*
* List of hardware interface methods, and which locks are held when each
* is called by this module:
*
* METHOD INTR THREAD NOTES
* ----------------------- ------- ------- -------------------------
* open x x
* close x x
* drain x x
* query_encoding - x
* set_params - x
* round_blocksize - x
* commit_settings - x
* init_output x x
* init_input x x
* start_output x x
* start_input x x
* halt_output x x
* halt_input x x
* speaker_ctl x x
* getdev - x
* setfd - x
* set_port - x
* get_port - x
* query_devinfo - x
* allocm - - Called at attach time
* freem - - Called at attach time
* round_buffersize - x
* mappage - - Mem. unchanged after attach
* get_props - x
* trigger_output x x
* trigger_input x x
* dev_ioctl - x
* get_locks - - Called at attach time
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.459 2019/02/27 02:27:38 mrg Exp $");
#ifdef _KERNEL_OPT
#include "audio.h"
#include "midi.h"
#endif
#if NAUDIO > 0
#include <sys/types.h>
#include <sys/param.h>
#include <sys/ioctl.h>
#include <sys/fcntl.h>
#include <sys/file.h>
#include <sys/filedesc.h>
#include <sys/vnode.h>
#include <sys/select.h>
#include <sys/poll.h>
#include <sys/kauth.h>
#include <sys/kmem.h>
#include <sys/malloc.h>
#include <sys/module.h>
#include <sys/proc.h>
#include <sys/queue.h>
#include <sys/stat.h>
#include <sys/systm.h>
#include <sys/sysctl.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 <sys/kthread.h>
#include <sys/cpu.h>
#include <sys/mman.h>
#include <dev/audio_if.h>
#include <dev/audiovar.h>
#include <dev/auconv.h>
#include <dev/auvolconv.h>
#include <machine/endian.h>
#include <uvm/uvm.h>
#include "ioconf.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 PREFILL_BLOCKS 3 /* no. audioblocks required to start stream */
#define ROUNDSIZE(x) (x) &= -16 /* round to nice boundary */
#define SPECIFIED(x) ((int)(x) != ~0)
#define SPECIFIED_CH(x) ((x) != (u_char)~0)
/* #define AUDIO_PM_IDLE */
#ifdef AUDIO_PM_IDLE
int audio_idle_timeout = 30;
#endif
#define HW_LOCK(x) do { \
if ((x) == sc->sc_hwvc) \
mutex_enter(sc->sc_intr_lock); \
} while (0)
#define HW_UNLOCK(x) do { \
if ((x) == sc->sc_hwvc) \
mutex_exit(sc->sc_intr_lock); \
} while (0)
int audio_blk_ms = AUDIO_BLK_MS;
int audiosetinfo(struct audio_softc *, struct audio_info *, bool,
struct virtual_channel *);
int audiogetinfo(struct audio_softc *, struct audio_info *, int,
struct virtual_channel *);
int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *,
struct file **);
int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *,
struct file **);
int audio_close(struct audio_softc *, int, struct audio_chan *);
int audio_read(struct audio_softc *, struct uio *, int,
struct virtual_channel *);
int audio_write(struct audio_softc *, struct uio *, int,
struct virtual_channel *);
int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int,
struct lwp *, struct audio_chan *);
int audio_poll(struct audio_softc *, int, struct lwp *,
struct virtual_channel *);
int audio_kqfilter(struct audio_chan *, struct knote *);
int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *,
struct uvm_object **, int *, struct virtual_channel *);
static int audio_fop_mmap(struct file *, off_t *, size_t, int, int *, int *,
struct uvm_object **, int *);
int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *,
struct file **);
int mixer_close(struct audio_softc *, int, struct audio_chan *);
int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *);
static void mixer_remove(struct audio_softc *);
static void mixer_signal(struct audio_softc *);
static void grow_mixer_states(struct audio_softc *, int);
static void shrink_mixer_states(struct audio_softc *, int);
void audio_init_record(struct audio_softc *, struct virtual_channel *);
void audio_init_play(struct audio_softc *, struct virtual_channel *);
int audiostartr(struct audio_softc *, struct virtual_channel *);
int audiostartp(struct audio_softc *, struct virtual_channel *);
void audio_rint(void *);
void audio_pint(void *);
void audio_mix(void *);
void audio_upmix(void *);
void audio_play_thread(void *);
void audio_rec_thread(void *);
void recswvol_func(struct audio_softc *, struct audio_ringbuffer *,
size_t, struct virtual_channel *);
void mix_func(struct audio_softc *, struct audio_ringbuffer *,
struct virtual_channel *);
int mix_write(void *);
int mix_read(void *);
int audio_check_params(struct audio_params *);
static void audio_calc_latency(struct audio_softc *);
static void audio_setblksize(struct audio_softc *,
struct virtual_channel *, int, int);
int audio_calc_blksize(struct audio_softc *, const audio_params_t *);
void audio_fill_silence(const struct audio_params *, uint8_t *, int);
int audio_silence_copyout(struct audio_softc *, int, struct uio *);
static int audio_allocbufs(struct audio_softc *);
void audio_init_ringbuffer(struct audio_softc *,
struct audio_ringbuffer *, int);
int audio_initbufs(struct audio_softc *, struct virtual_channel *);
void audio_calcwater(struct audio_softc *, struct virtual_channel *);
int audio_drain(struct audio_softc *, struct virtual_channel *);
void audio_clear(struct audio_softc *, struct virtual_channel *);
void audio_clear_intr_unlocked(struct audio_softc *sc,
struct virtual_channel *);
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 *, struct virtual_channel *);
static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *,
stream_filter_list_t *, struct virtual_channel *);
static void audio_destroy_pfilters(struct virtual_channel *);
static void audio_destroy_rfilters(struct virtual_channel *);
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,
struct virtual_channel *);
static int audio_sysctl_frequency(SYSCTLFN_PROTO);
static int audio_sysctl_precision(SYSCTLFN_PROTO);
static int audio_sysctl_channels(SYSCTLFN_PROTO);
static int audio_sysctl_latency(SYSCTLFN_PROTO);
static int audio_sysctl_usemixer(SYSCTLFN_PROTO);
static int audiomatch(device_t, cfdata_t, void *);
static void audioattach(device_t, device_t, void *);
static int audiodetach(device_t, int);
static int audioactivate(device_t, enum devact);
static void audiochilddet(device_t, device_t);
static int audiorescan(device_t, const char *, const int *);
static int audio_modcmd(modcmd_t, void *);
#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 *);
static int audio_enter(dev_t, krw_t, struct audio_softc **);
static void audio_exit(struct audio_softc *);
static int audio_waitio(struct audio_softc *, kcondvar_t *,
struct virtual_channel *);
static int audioclose(struct file *);
static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int);
static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int);
static int audioioctl(struct file *, u_long, void *);
static int audiopoll(struct file *, int);
static int audiokqfilter(struct file *, struct knote *);
static int audiostat(struct file *, struct stat *);
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 *);
static int
audio_get_port(struct audio_softc *, mixer_ctrl_t *);
static int
audio_set_port(struct audio_softc *, mixer_ctrl_t *);
static int
audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *);
static int audio_set_params (struct audio_softc *, int, int,
audio_params_t *, audio_params_t *,
stream_filter_list_t *, stream_filter_list_t *,
const struct virtual_channel *);
static int
audio_query_encoding(struct audio_softc *, struct audio_encoding *);
static int audio_set_vchan_defaults(struct audio_softc *, u_int);
static int vchan_autoconfig(struct audio_softc *);
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(struct audio_softc *, stream_fetcher_t *,
audio_stream_t *, int);
static int null_fetcher_fetch_to(struct audio_softc *, stream_fetcher_t *,
audio_stream_t *, int);
static dev_type_open(audioopen);
/* XXXMRG use more dev_type_xxx */
const struct cdevsw audio_cdevsw = {
.d_open = audioopen,
.d_close = noclose,
.d_read = noread,
.d_write = nowrite,
.d_ioctl = noioctl,
.d_stop = nostop,
.d_tty = notty,
.d_poll = nopoll,
.d_mmap = nommap,
.d_kqfilter = nokqfilter,
.d_discard = nodiscard,
.d_flag = D_OTHER | D_MPSAFE
};
const struct fileops audio_fileops = {
.fo_name = "audio",
.fo_read = audioread,
.fo_write = audiowrite,
.fo_ioctl = audioioctl,
.fo_fcntl = fnullop_fcntl,
.fo_stat = audiostat,
.fo_poll = audiopoll,
.fo_close = audioclose,
.fo_mmap = audio_fop_mmap,
.fo_kqfilter = audiokqfilter,
.fo_restart = fnullop_restart
};
/* 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,
};
int auto_config_precision[] = { 16, 8, 32 };
int auto_config_channels[] = { 2, AUDIO_MAX_CHANNELS, 10, 8, 6, 4, 1 };
int auto_config_freq[] = { 48000, 44100, 96000, 192000, 32000,
22050, 16000, 11025, 8000, 4000 };
CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc),
audiomatch, audioattach, audiodetach, audioactivate, audiorescan,
audiochilddet, DVF_DETACH_SHUTDOWN);
static int
audiomatch(device_t parent, cfdata_t match, void *aux)
{
struct audio_attach_args *sa;
sa = aux;
DPRINTF(("%s: type=%d sa=%p hw=%p\n",
__func__, sa->type, sa, sa->hwif));
return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0;
}
static void
audioattach(device_t parent, device_t self, void *aux)
{
struct audio_softc *sc;
struct audio_attach_args *sa;
struct virtual_channel *vc;
const struct audio_hw_if *hwp;
const struct sysctlnode *node;
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;
sc->sc_opens = 0;
sc->sc_recopens = 0;
sc->sc_aivalid = false;
sc->sc_ready = true;
sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS;
sc->sc_format[0].mode = AUMODE_PLAY | AUMODE_RECORD;
sc->sc_format[0].encoding =
#if BYTE_ORDER == LITTLE_ENDIAN
AUDIO_ENCODING_SLINEAR_LE;
#else
AUDIO_ENCODING_SLINEAR_BE;
#endif
sc->sc_format[0].precision = 16;
sc->sc_format[0].validbits = 16;
sc->sc_format[0].channels = 2;
sc->sc_format[0].channel_mask = AUFMT_STEREO;
sc->sc_format[0].frequency_type = 1;
sc->sc_format[0].frequency[0] = 44100;
sc->sc_trigger_started = false;
sc->sc_rec_started = false;
sc->sc_dying = false;
SIMPLEQ_INIT(&sc->sc_audiochan);
vc = kmem_zalloc(sizeof(struct virtual_channel), KM_SLEEP);
sc->sc_hwvc = vc;
vc->sc_open = 0;
vc->sc_mode = 0;
vc->sc_npfilters = 0;
vc->sc_nrfilters = 0;
memset(vc->sc_pfilters, 0, sizeof(vc->sc_pfilters));
memset(vc->sc_rfilters, 0, sizeof(vc->sc_rfilters));
vc->sc_lastinfovalid = false;
vc->sc_swvol = 255;
vc->sc_recswvol = 255;
if (auconv_create_encodings(sc->sc_format, VAUDIO_NFORMATS,
&sc->sc_encodings) != 0) {
aprint_error_dev(self, "couldn't create encodings\n");
return;
}
cv_init(&sc->sc_rchan, "audiord");
cv_init(&sc->sc_wchan, "audiowr");
cv_init(&sc->sc_lchan, "audiolk");
cv_init(&sc->sc_condvar,"play");
cv_init(&sc->sc_rcondvar,"record");
if (hwp == NULL || hwp->get_locks == NULL) {
aprint_error(": missing method\n");
panic("audioattach");
}
hwp->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock);
#ifdef DIAGNOSTIC
if (hwp->query_encoding == NULL ||
hwp->set_params == NULL ||
(hwp->start_output == NULL && hwp->trigger_output == NULL) ||
(hwp->start_input == NULL && hwp->trigger_input == NULL) ||
hwp->halt_output == NULL ||
hwp->halt_input == NULL ||
hwp->getdev == NULL ||
hwp->set_port == NULL ||
hwp->get_port == NULL ||
hwp->query_devinfo == NULL ||
hwp->get_props == NULL) {
aprint_error(": missing method\n");
return;
}
#endif
sc->hw_if = hwp;
sc->hw_hdl = hdlp;
sc->sc_dev = parent;
mutex_enter(sc->sc_lock);
props = audio_get_props(sc);
mutex_exit(sc->sc_lock);
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");
mutex_enter(sc->sc_lock);
if (audio_allocbufs(sc) != 0) {
aprint_error_dev(sc->sc_dev,
"could not allocate ring buffer\n");
mutex_exit(sc->sc_lock);
return;
}
mutex_exit(sc->sc_lock);
sc->sc_lastgain = 128;
sc->sc_multiuser = false;
sc->sc_usemixer = true;
error = vchan_autoconfig(sc);
if (error != 0) {
aprint_error_dev(sc->sc_dev, "%s: audio_set_vchan_defaults() "
"failed\n", __func__);
}
sc->sc_sih_rd = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE,
audio_softintr_rd, sc);
sc->sc_sih_wr = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE,
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.
*/
mutex_enter(sc->sc_lock);
for(mi.index = 0; ; mi.index++) {
if (audio_query_devinfo(sc, &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;
}
}
mutex_exit(sc->sc_lock);
/* Allocate save area. Ensure non-zero allocation. */
sc->sc_static_nmixer_states = mi.index;
sc->sc_static_nmixer_states++;
sc->sc_nmixer_states = sc->sc_static_nmixer_states;
sc->sc_mixer_state = kmem_zalloc(sizeof(mixer_ctrl_t) *
(sc->sc_nmixer_states + 1), KM_SLEEP);
/*
* 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;
mutex_enter(sc->sc_lock);
for(mi.index = 0; ; mi.index++) {
if (audio_query_devinfo(sc, &mi) != 0)
break;
KASSERT(mi.index < sc->sc_nmixer_states);
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;
}
}
}
mutex_exit(sc->sc_lock);
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));
/* sysctl set-up for alternate configs */
sysctl_createv(&sc->sc_log, 0, NULL, &node,
0,
CTLTYPE_NODE, device_xname(sc->sc_dev),
SYSCTL_DESCR("audio format information"),
NULL, 0,
NULL, 0,
CTL_HW,
CTL_CREATE, CTL_EOL);
if (node != NULL) {
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_INT, "frequency",
SYSCTL_DESCR("intermediate frequency"),
audio_sysctl_frequency, 0,
(void *)sc, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_INT, "precision",
SYSCTL_DESCR("intermediate precision"),
audio_sysctl_precision, 0,
(void *)sc, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_INT, "channels",
SYSCTL_DESCR("intermediate channels"),
audio_sysctl_channels, 0,
(void *)sc, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_INT, "latency",
SYSCTL_DESCR("latency"),
audio_sysctl_latency, 0,
(void *)sc, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_BOOL, "multiuser",
SYSCTL_DESCR("allow multiple user acess"),
NULL, 0,
&sc->sc_multiuser, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_BOOL, "usemixer",
SYSCTL_DESCR("allow in-kernel mixing"),
audio_sysctl_usemixer, 0,
(void *)sc, 0,
CTL_HW, node->sysctl_num,
CTL_CREATE, CTL_EOL);
}
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
kthread_create(PRI_SOFTSERIAL, KTHREAD_MPSAFE | KTHREAD_MUSTJOIN, NULL,
audio_rec_thread, sc, &sc->sc_recthread, "audiorec");
kthread_create(PRI_SOFTSERIAL, KTHREAD_MPSAFE | KTHREAD_MUSTJOIN, NULL,
audio_play_thread, sc, &sc->sc_playthread, "audiomix");
audiorescan(self, "audio", NULL);
}
static int
audioactivate(device_t self, enum devact act)
{
struct audio_softc *sc = device_private(self);
switch (act) {
case DVACT_DEACTIVATE:
mutex_enter(sc->sc_lock);
sc->sc_dying = true;
mutex_enter(sc->sc_intr_lock);
cv_broadcast(&sc->sc_condvar);
cv_broadcast(&sc->sc_rcondvar);
cv_broadcast(&sc->sc_wchan);
cv_broadcast(&sc->sc_rchan);
cv_broadcast(&sc->sc_lchan);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
return 0;
default:
return EOPNOTSUPP;
}
}
static int
audiodetach(device_t self, int flags)
{
struct audio_softc *sc;
struct audio_chan *chan;
int maj, mn, rc;
sc = device_private(self);
DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags));
/* Start draining existing accessors of the device. */
if ((rc = config_detach_children(self, flags | DETACH_FORCE)) != 0)
return rc;
mutex_enter(sc->sc_lock);
sc->sc_dying = true;
cv_broadcast(&sc->sc_wchan);
cv_broadcast(&sc->sc_rchan);
mutex_enter(sc->sc_intr_lock);
cv_broadcast(&sc->sc_condvar);
cv_broadcast(&sc->sc_rcondvar);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
kthread_join(sc->sc_playthread);
kthread_join(sc->sc_recthread);
mutex_enter(sc->sc_lock);
cv_destroy(&sc->sc_condvar);
cv_destroy(&sc->sc_rcondvar);
mutex_exit(sc->sc_lock);
/* delete sysctl nodes */
sysctl_teardown(&sc->sc_log);
/* locate the major number */
maj = cdevsw_lookup_major(&audio_cdevsw);
/*
* Nuke the vnodes for any open instances (calls close).
* Will wait until any activity on the device nodes has ceased.
*/
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);
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_halt(&sc->sc_idle_counter, sc->sc_lock);
device_active_deregister(self, audio_activity);
#endif
pmf_device_deregister(self);
/* free resources */
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
audio_free_ring(sc, &chan->vc->sc_mpr);
audio_free_ring(sc, &chan->vc->sc_mrr);
}
audio_free_ring(sc, &sc->sc_hwvc->sc_mpr);
audio_free_ring(sc, &sc->sc_hwvc->sc_mrr);
audio_free_ring(sc, &sc->sc_mixring.sc_mpr);
audio_free_ring(sc, &sc->sc_mixring.sc_mrr);
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
audio_destroy_pfilters(chan->vc);
audio_destroy_rfilters(chan->vc);
}
audio_destroy_pfilters(sc->sc_hwvc);
audio_destroy_rfilters(sc->sc_hwvc);
auconv_delete_encodings(sc->sc_encodings);
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;
}
kmem_free(sc->sc_hwvc, sizeof(struct virtual_channel));
kmem_free(sc->sc_mixer_state, sizeof(mixer_ctrl_t) *
(sc->sc_nmixer_states + 1));
#ifdef AUDIO_PM_IDLE
callout_destroy(&sc->sc_idle_counter);
#endif
seldestroy(&sc->sc_rsel);
seldestroy(&sc->sc_wsel);
cv_destroy(&sc->sc_rchan);
cv_destroy(&sc->sc_wchan);
cv_destroy(&sc->sc_lchan);
return 0;
}
static void
audiochilddet(device_t self, device_t child)
{
/* we hold no child references, so do nothing */
}
static int
audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux)
{
if (config_match(parent, cf, aux))
config_attach_loc(parent, cf, locs, aux, NULL);
return 0;
}
static int
audiorescan(device_t self, const char *ifattr, const int *flags)
{
struct audio_softc *sc = device_private(self);
if (!ifattr_match(ifattr, "audio"))
return 0;
config_search_loc(audiosearch, sc->dev, "audio", NULL, NULL);
return 0;
}
int
au_portof(struct audio_softc *sc, char *name, int class)
{
mixer_devinfo_t mi;
for (mi.index = 0; audio_query_devinfo(sc, &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)
{
struct virtual_channel *vc;
vc = sc->sc_hwvc;
printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
printf("open 0x%x mode 0x%x\n", vc->sc_open, vc->sc_mode);
printf("rchan 0x%x wchan 0x%x ", cv_has_waiters(&sc->sc_rchan),
cv_has_waiters(&sc->sc_wchan));
printf("rring used 0x%x pring used=%d\n",
audio_stream_get_used(&vc->sc_mrr.s),
audio_stream_get_used(&vc->sc_mpr.s));
printf("rbus 0x%x pbus 0x%x ", vc->sc_rbus, vc->sc_pbus);
printf("blksize %d", vc->sc_mpr.blksize);
printf("hiwat %d lowat %d\n", vc->sc_mpr.usedhigh,
vc->sc_mpr.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
/* Allocate all ring buffers. called from audioattach() */
static int
audio_allocbufs(struct audio_softc *sc)
{
struct virtual_channel *vc;
int error;
vc = sc->sc_hwvc;
sc->sc_mixring.sc_mpr.s.start = NULL;
vc->sc_mpr.s.start = NULL;
sc->sc_mixring.sc_mrr.s.start = NULL;
vc->sc_mrr.s.start = NULL;
if (audio_can_playback(sc)) {
error = audio_alloc_ring(sc, &sc->sc_mixring.sc_mpr,
AUMODE_PLAY, AU_RING_SIZE);
if (error)
goto bad_play1;
error = audio_alloc_ring(sc, &vc->sc_mpr,
AUMODE_PLAY, AU_RING_SIZE);
if (error)
goto bad_play2;
}
if (audio_can_capture(sc)) {
error = audio_alloc_ring(sc, &sc->sc_mixring.sc_mrr,
AUMODE_RECORD, AU_RING_SIZE);
if (error)
goto bad_rec1;
error = audio_alloc_ring(sc, &vc->sc_mrr,
AUMODE_RECORD, AU_RING_SIZE);
if (error)
goto bad_rec2;
}
return 0;
bad_rec2:
if (sc->sc_mixring.sc_mrr.s.start != NULL)
audio_free_ring(sc, &sc->sc_mixring.sc_mrr);
bad_rec1:
if (vc->sc_mpr.s.start != NULL)
audio_free_ring(sc, &vc->sc_mpr);
bad_play2:
if (sc->sc_mixring.sc_mpr.s.start != NULL)
audio_free_ring(sc, &sc->sc_mixring.sc_mpr);
bad_play1:
return error;
}
int
audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r,
int direction, size_t bufsize)
{
const struct audio_hw_if *hw;
struct virtual_channel *vc;
void *hdl;
vaddr_t vstart;
vsize_t vsize;
int error;
vc = sc->sc_hwvc;
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 == &vc->sc_mpr || r == &vc->sc_mrr)) {
/* Hardware ringbuffer. No dedicated uvm object.*/
r->uobj = NULL;
r->s.start = hw->allocm(hdl, direction, bufsize);
if (r->s.start == NULL)
return ENOMEM;
} else {
/* Software ringbuffer. */
vstart = 0;
/* Get a nonzero multiple of PAGE_SIZE. */
vsize = roundup2(MAX(bufsize, PAGE_SIZE), PAGE_SIZE);
/* Create a uvm anonymous object. */
r->uobj = uao_create(vsize, 0);
/* Map it into the kernel virtual address space. */
error = uvm_map(kernel_map, &vstart, vsize, r->uobj, 0, 0,
UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE,
UVM_ADV_RANDOM, 0));
if (error) {
uao_detach(r->uobj); /* release reference */
r->uobj = NULL; /* paranoia */
return error;
}
error = uvm_map_pageable(kernel_map, vstart, vstart + vsize,
false, 0);
if (error) {
uvm_unmap(kernel_map, vstart, vstart + vsize);
/* uvm_unmap also detach uobj */
r->uobj = NULL; /* paranoia */
return error;
}
r->s.start = (void *)vstart;
}
r->s.bufsize = bufsize;
return 0;
}
void
audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r)
{
struct virtual_channel *vc;
vaddr_t vstart;
vsize_t vsize;
if (r->s.start == NULL)
return;
vc = sc->sc_hwvc;
if (sc->hw_if->freem && (r == &vc->sc_mpr || r == &vc->sc_mrr)) {
/* Hardware ringbuffer. */
KASSERT(r->uobj == NULL);
sc->hw_if->freem(sc->hw_hdl, r->s.start, r->s.bufsize);
} else {
/* Software ringbuffer. */
vstart = (vaddr_t)r->s.start;
vsize = roundup2(MAX(r->s.bufsize, PAGE_SIZE), PAGE_SIZE);
/*
* Unmap the kernel mapping. uvm_unmap releases the
* reference to the uvm object, and this should be the
* last virtual mapping of the uvm object, so no need
* to explicitly release (`detach') the object.
*/
uvm_unmap(kernel_map, vstart, vstart + vsize);
r->uobj = NULL; /* paranoia */
}
r->s.start = NULL;
}
static int
audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp,
stream_filter_list_t *pfilters, struct virtual_channel *vc)
{
stream_filter_t *pf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS];
audio_stream_t ps[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS];
const audio_params_t *from_param;
audio_params_t *to_param;
int i, n, onfilters;
KASSERT(mutex_owned(sc->sc_lock));
/* Construct new filters. */
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]);
}
return EINVAL;
}
/* Swap in new filters. */
HW_LOCK(vc);
memcpy(of, vc->sc_pfilters, sizeof(of));
memcpy(os, vc->sc_pstreams, sizeof(os));
onfilters = vc->sc_npfilters;
memcpy(vc->sc_pfilters, pf, sizeof(pf));
memcpy(vc->sc_pstreams, ps, sizeof(ps));
vc->sc_npfilters = pfilters->req_size;
for (i = 0; i < pfilters->req_size; i++)
pf[i]->set_inputbuffer(pf[i], &vc->sc_pstreams[i]);
/* hardware format and the buffer near to userland */
if (pfilters->req_size <= 0) {
vc->sc_mpr.s.param = *pp;
vc->sc_pustream = &vc->sc_mpr.s;
} else {
vc->sc_mpr.s.param = pfilters->filters[0].param;
vc->sc_pustream = &vc->sc_pstreams[0];
}
HW_UNLOCK(vc);
/* Destroy old filters. */
for (i = 0; i < onfilters; i++) {
of[i]->dtor(of[i]);
audio_stream_dtor(&os[i]);
}
#ifdef AUDIO_DEBUG
if (audiodebug) {
printf("%s: HW-buffer=%p pustream=%p\n",
__func__, &vc->sc_mpr.s, vc->sc_pustream);
for (i = 0; i < pfilters->req_size; i++) {
char num[100];
snprintf(num, 100, "[%d]", i);
audio_print_params(num, &vc->sc_pstreams[i].param);
}
audio_print_params("[HW]", &vc->sc_mpr.s.param);
}
#endif /* AUDIO_DEBUG */
return 0;
}
static int
audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp,
stream_filter_list_t *rfilters, struct virtual_channel *vc)
{
stream_filter_t *rf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS];
audio_stream_t rs[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS];
const audio_params_t *to_param;
audio_params_t *from_param;
int i, onfilters;
KASSERT(mutex_owned(sc->sc_lock));
/* Construct new filters. */
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], &vc->sc_mrr.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;
}
/* Swap in new filters. */
HW_LOCK(vc);
memcpy(of, vc->sc_rfilters, sizeof(of));
memcpy(os, vc->sc_rstreams, sizeof(os));
onfilters = vc->sc_nrfilters;
memcpy(vc->sc_rfilters, rf, sizeof(rf));
memcpy(vc->sc_rstreams, rs, sizeof(rs));
vc->sc_nrfilters = rfilters->req_size;
for (i = 1; i < rfilters->req_size; i++)
rf[i]->set_inputbuffer(rf[i], &vc->sc_rstreams[i - 1]);
/* hardware format and the buffer near to userland */
if (rfilters->req_size <= 0) {
vc->sc_mrr.s.param = *rp;
vc->sc_rustream = &vc->sc_mrr.s;
} else {
vc->sc_mrr.s.param = rfilters->filters[0].param;
vc->sc_rustream = &vc->sc_rstreams[rfilters->req_size - 1];
}
HW_UNLOCK(vc);
#ifdef AUDIO_DEBUG
if (audiodebug) {
printf("%s: HW-buffer=%p rustream=%p\n",
__func__, &vc->sc_mrr.s, vc->sc_rustream);
audio_print_params("[HW]", &vc->sc_mrr.s.param);
for (i = 0; i < rfilters->req_size; i++) {
char num[100];
snprintf(num, 100, "[%d]", i);
audio_print_params(num, &vc->sc_rstreams[i].param);
}
}
#endif /* AUDIO_DEBUG */
/* Destroy old filters. */
for (i = 0; i < onfilters; i++) {
of[i]->dtor(of[i]);
audio_stream_dtor(&os[i]);
}
return 0;
}
static void
audio_destroy_pfilters(struct virtual_channel *vc)
{
int i;
for (i = 0; i < vc->sc_npfilters; i++) {
vc->sc_pfilters[i]->dtor(vc->sc_pfilters[i]);
vc->sc_pfilters[i] = NULL;
audio_stream_dtor(&vc->sc_pstreams[i]);
}
vc->sc_npfilters = 0;
}
static void
audio_destroy_rfilters(struct virtual_channel *vc)
{
int i;
for (i = 0; i < vc->sc_nrfilters; i++) {
vc->sc_rfilters[i]->dtor(vc->sc_rfilters[i]);
vc->sc_rfilters[i] = NULL;
audio_stream_dtor(&vc->sc_pstreams[i]);
}
vc->sc_nrfilters = 0;
}
static void
audio_stream_dtor(audio_stream_t *stream)
{
if (stream->start != NULL)
kmem_free(stream->start, stream->bufsize);
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 = uimin(size, AU_RING_SIZE);
stream->bufsize = size;
stream->start = kmem_zalloc(size, KM_SLEEP);
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++;
}
/*
* Look up audio device and acquire locks for device access.
*/
static int
audio_enter(dev_t dev, krw_t rw, struct audio_softc **scp)
{
struct audio_softc *sc;
/* First, find the device and take sc_lock. */
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL || sc->hw_if == NULL)
return ENXIO;
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
return EIO;
}
*scp = sc;
return 0;
}
/*
* Release reference to device acquired with audio_enter().
*/
static void
audio_exit(struct audio_softc *sc)
{
cv_broadcast(&sc->sc_lchan);
mutex_exit(sc->sc_lock);
}
/*
* Wait for I/O to complete, releasing device lock.
*/
static int
audio_waitio(struct audio_softc *sc, kcondvar_t *chan, struct virtual_channel *vc)
{
struct audio_chan *vchan;
bool found = false;
int error;
KASSERT(mutex_owned(sc->sc_lock));
cv_broadcast(&sc->sc_lchan);
/* Wait for pending I/O to complete. */
error = cv_wait_sig(chan, sc->sc_lock);
if (!sc->sc_usemixer || vc == sc->sc_hwvc)
return error;
found = false;
SIMPLEQ_FOREACH(vchan, &sc->sc_audiochan, entries) {
if (vchan->vc == vc) {
found = true;
break;
}
}
if (found == false)
error = EIO;
return error;
}
/* Exported interfaces for audiobell. */
int
audiobellopen(dev_t dev, int flags, int ifmt, struct lwp *l,
struct file **fp)
{
struct audio_softc *sc;
int error;
if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0)
return error;
device_active(sc->dev, DVA_SYSTEM);
switch (AUDIODEV(dev)) {
case AUDIO_DEVICE:
error = audio_open(dev, sc, flags, ifmt, l, fp);
break;
default:
error = EINVAL;
break;
}
audio_exit(sc);
return error;
}
int
audiobellclose(struct file *fp)
{
return audioclose(fp);
}
int
audiobellwrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred,
int ioflag)
{
return audiowrite(fp, offp, uio, cred, ioflag);
}
int
audiobellioctl(struct file *fp, u_long cmd, void *addr)
{
return audioioctl(fp, cmd, addr);
}
static int
audioopen(dev_t dev, int flags, int ifmt, struct lwp *l)
{
struct audio_softc *sc;
struct file *fp;
int error;
if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0)
return error;
device_active(sc->dev, DVA_SYSTEM);
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_open(dev, sc, flags, ifmt, l, &fp);
break;
case AUDIOCTL_DEVICE:
error = audioctl_open(dev, sc, flags, ifmt, l, &fp);
break;
case MIXER_DEVICE:
error = mixer_open(dev, sc, flags, ifmt, l, &fp);
break;
default:
error = ENXIO;
break;
}
audio_exit(sc);
return error;
}
static int
audioclose(struct file *fp)
{
struct audio_softc *sc;
struct audio_chan *chan;
int error;
dev_t dev;
chan = fp->f_audioctx;
if (chan == NULL) /* XXX:NS Why is this needed. */
return EIO;
dev = chan->dev;
if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0)
return error;
device_active(sc->dev, DVA_SYSTEM);
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_close(sc, fp->f_flag, chan);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
case MIXER_DEVICE:
error = mixer_close(sc, fp->f_flag, chan);
break;
default:
error = ENXIO;
break;
}
if (error == 0) {
kmem_free(fp->f_audioctx, sizeof(struct audio_chan));
fp->f_audioctx = NULL;
}
audio_exit(sc);
return error;
}
static int
audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred,
int ioflag)
{
struct audio_softc *sc;
struct virtual_channel *vc;
int error;
dev_t dev;
if (fp->f_audioctx == NULL)
return EIO;
dev = fp->f_audioctx->dev;
if ((error = audio_enter(dev, RW_READER, &sc)) != 0)
return error;
if (fp->f_flag & O_NONBLOCK)
ioflag |= IO_NDELAY;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
vc = fp->f_audioctx->vc;
error = audio_read(sc, uio, ioflag, vc);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
audio_exit(sc);
return error;
}
static int
audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred,
int ioflag)
{
struct audio_softc *sc;
struct virtual_channel *vc;
int error;
dev_t dev;
if (fp->f_audioctx == NULL)
return EIO;
dev = fp->f_audioctx->dev;
if ((error = audio_enter(dev, RW_READER, &sc)) != 0)
return error;
if (fp->f_flag & O_NONBLOCK)
ioflag |= IO_NDELAY;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
vc = fp->f_audioctx->vc;
error = audio_write(sc, uio, ioflag, vc);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
audio_exit(sc);
return error;
}
static int
audioioctl(struct file *fp, u_long cmd, void *addr)
{
struct audio_softc *sc;
struct audio_chan *chan;
struct lwp *l = curlwp;
int error;
krw_t rw;
dev_t dev;
if (fp->f_audioctx == NULL)
return EIO;
chan = fp->f_audioctx;
dev = chan->dev;
/* Figure out which lock type we need. */
switch (cmd) {
case AUDIO_FLUSH:
case AUDIO_SETINFO:
case AUDIO_DRAIN:
case AUDIO_SETFD:
rw = RW_WRITER;
break;
default:
rw = RW_READER;
break;
}
if ((error = audio_enter(dev, rw, &sc)) != 0)
return error;
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, fp->f_flag, l);
else
error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l,
chan);
break;
case MIXER_DEVICE:
error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l);
break;
default:
error = ENXIO;
break;
}
audio_exit(sc);
return error;
}
static int
audiostat(struct file *fp, struct stat *st)
{
if (fp->f_audioctx == NULL)
return EIO;
memset(st, 0, sizeof(*st));
st->st_dev = fp->f_audioctx->dev;
st->st_uid = kauth_cred_geteuid(fp->f_cred);
st->st_gid = kauth_cred_getegid(fp->f_cred);
st->st_mode = S_IFCHR;
return 0;
}
static int
audiopoll(struct file *fp, int events)
{
struct audio_softc *sc;
struct virtual_channel *vc;
struct lwp *l = curlwp;
int revents;
dev_t dev;
if (fp->f_audioctx == NULL)
return POLLERR;
dev = fp->f_audioctx->dev;
/* Don't bother with device level lock here. */
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return POLLERR;
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
return POLLERR;
}
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
vc = fp->f_audioctx->vc;
revents = audio_poll(sc, events, l, vc);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
revents = 0;
break;
default:
revents = POLLERR;
break;
}
mutex_exit(sc->sc_lock);
return revents;
}
static int
audiokqfilter(struct file *fp, struct knote *kn)
{
struct audio_softc *sc;
struct audio_chan *chan;
int error;
dev_t dev;
chan = fp->f_audioctx;
dev = chan->dev;
/* Don't bother with device level lock here. */
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
return EIO;
}
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_kqfilter(chan, kn);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
mutex_exit(sc->sc_lock);
return error;
}
static int
audio_fop_mmap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp,
int *advicep, struct uvm_object **uobjp, int *maxprotp)
{
struct audio_softc *sc;
struct audio_chan *chan;
struct virtual_channel *vc;
dev_t dev;
int error;
chan = fp->f_audioctx;
dev = chan->dev;
vc = chan->vc;
error = 0;
if ((error = audio_enter(dev, RW_READER, &sc)) != 0)
return error;
device_active(sc->dev, DVA_SYSTEM); /* XXXJDM */
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_mmap(sc, offp, len, prot, flagsp, advicep,
uobjp, maxprotp, vc);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
default:
error = ENOTSUP;
break;
}
audio_exit(sc);
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 > (int)(rp->s.bufsize / AUMINNOBLK))
blksize = rp->s.bufsize / AUMINNOBLK;
ROUNDSIZE(blksize);
DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize));
struct virtual_channel *hwvc = sc->sc_hwvc;
int tmpblksize = 1;
/* round blocksize to a power of 2 */
while (tmpblksize < blksize)
tmpblksize <<= 1;
blksize = tmpblksize;
if (sc->hw_if->round_blocksize &&
(rp == &hwvc->sc_mpr || rp == &hwvc->sc_mrr || rp ==
&sc->sc_mixring.sc_mpr || rp == &sc->sc_mixring.sc_mrr)) {
blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize,
mode, &rp->s.param);
}
if (blksize <= 0)
panic("audio_init_ringbuffer: blksize=%d", 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;
memset(rp->s.start, 0, blksize * 2);
}
int
audio_initbufs(struct audio_softc *sc, struct virtual_channel *vc)
{
const struct audio_hw_if *hw;
int error;
if (vc == NULL)
vc = sc->sc_hwvc;
DPRINTF(("audio_initbufs: mode=0x%x\n", vc->sc_mode));
hw = sc->hw_if;
if (audio_can_capture(sc) &&
((vc->sc_open & AUOPEN_READ) || vc == sc->sc_hwvc)) {
audio_init_ringbuffer(sc, &vc->sc_mrr,
AUMODE_RECORD);
if (sc->sc_recopens == 0 && (vc->sc_open & AUOPEN_READ)) {
if (hw->init_input) {
error = hw->init_input(sc->hw_hdl,
vc->sc_mrr.s.start,
vc->sc_mrr.s.end - vc->sc_mrr.s.start);
if (error)
return error;
}
}
}
if (audio_can_playback(sc) &&
((vc->sc_open & AUOPEN_WRITE) || vc == sc->sc_hwvc)) {
audio_init_ringbuffer(sc, &vc->sc_mpr,
AUMODE_PLAY);
if (sc->sc_opens == 0 && (vc->sc_open & AUOPEN_WRITE)) {
if (hw->init_output) {
error = hw->init_output(sc->hw_hdl,
vc->sc_mpr.s.start,
vc->sc_mpr.s.end - vc->sc_mpr.s.start);
if (error)
return error;
}
}
}
#ifdef AUDIO_INTR_TIME
if (audio_can_playback(sc)) {
sc->sc_pnintr = 0;
sc->sc_pblktime = (int64_t)vc->sc_mpr.blksize * 1000000 /
(vc->sc_pparams.channels *
vc->sc_pparams.sample_rate *
vc->sc_pparams.precision / NBBY);
DPRINTF(("audio: play blktime = %" PRId64 " for %d\n",
sc->sc_pblktime, vc->sc_mpr.blksize));
}
if (audio_can_capture(sc)) {
sc->sc_rnintr = 0;
sc->sc_rblktime = (int64_t)vc->sc_mrr.blksize * 1000000 /
(vc->sc_rparams.channels *
vc->sc_rparams.sample_rate *
vc->sc_rparams.precision / NBBY);
DPRINTF(("audio: record blktime = %" PRId64 " for %d\n",
sc->sc_rblktime, vc->sc_mrr.blksize));
}
#endif
return 0;
}
void
audio_calcwater(struct audio_softc *sc, struct virtual_channel *vc)
{
/* set high at 100% */
if (audio_can_playback(sc) && vc && vc->sc_pustream) {
vc->sc_mpr.usedhigh =
vc->sc_pustream->end - vc->sc_pustream->start;
/* set low at 75% of usedhigh */
vc->sc_mpr.usedlow = vc->sc_mpr.usedhigh * 3 / 4;
if (vc->sc_mpr.usedlow == vc->sc_mpr.usedhigh)
vc->sc_mpr.usedlow -= vc->sc_mpr.blksize;
}
if (audio_can_capture(sc) && vc && vc->sc_rustream) {
vc->sc_mrr.usedhigh =
vc->sc_rustream->end - vc->sc_rustream->start -
vc->sc_mrr.blksize;
vc->sc_mrr.usedlow = 0;
DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__,
vc->sc_mpr.usedlow, vc->sc_mpr.usedhigh,
vc->sc_mrr.usedlow, vc->sc_mrr.usedhigh));
}
}
int
audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l, struct file **nfp)
{
struct file *fp;
int error, fd;
const struct audio_hw_if *hw;
struct virtual_channel *vc;
struct audio_chan *chan;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_usemixer && !sc->sc_ready)
return ENXIO;
hw = sc->hw_if;
if (hw == NULL)
return ENXIO;
chan = kmem_zalloc(sizeof(struct audio_chan), KM_SLEEP);
if (sc->sc_usemixer)
vc = &sc->sc_mixring;
else
vc = sc->sc_hwvc;
chan->vc = vc;
error = fd_allocfile(&fp, &fd);
if (error)
goto bad;
chan->dev = dev;
chan->chan = 0;
chan->deschan = 0;
error = fd_clone(fp, fd, flags, &audio_fileops, chan);
KASSERT(error == EMOVEFD);
*nfp = fp;
return error;
bad:
kmem_free(chan, sizeof(struct audio_chan));
return error;
}
int
audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l, struct file **nfp)
{
struct file *fp;
int error, fd, n;
u_int mode;
const struct audio_hw_if *hw;
struct virtual_channel *vc;
struct audio_chan *chan;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_usemixer && !sc->sc_ready)
return ENXIO;
hw = sc->hw_if;
if (hw == NULL)
return ENXIO;
n = 1;
chan = SIMPLEQ_LAST(&sc->sc_audiochan, audio_chan, entries);
if (chan != NULL)
n = chan->chan + 1;
chan = kmem_zalloc(sizeof(struct audio_chan), KM_SLEEP);
if (sc->sc_usemixer)
vc = kmem_zalloc(sizeof(struct virtual_channel), KM_SLEEP);
else
vc = sc->sc_hwvc;
chan->vc = vc;
if (sc->sc_usemixer) {
vc->sc_open = 0;
vc->sc_mode = 0;
vc->sc_nrfilters = 0;
memset(vc->sc_rfilters, 0,
sizeof(vc->sc_rfilters));
vc->sc_rbus = false;
vc->sc_npfilters = 0;
memset(vc->sc_pfilters, 0,
sizeof(vc->sc_pfilters));
vc->sc_draining = false;
vc->sc_pbus = false;
vc->sc_lastinfovalid = false;
vc->sc_swvol = 255;
vc->sc_recswvol = 255;
} else {
if (sc->sc_opens > 0 || sc->sc_recopens > 0 ) {
kmem_free(chan, sizeof(struct audio_chan));
return EBUSY;
}
}
DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n",
flags, sc, sc->hw_hdl));
if (sc->sc_usemixer) {
error = audio_alloc_ring(sc, &vc->sc_mpr, AUMODE_PLAY,
AU_RING_SIZE);
if (error)
goto bad;
error = audio_alloc_ring(sc, &vc->sc_mrr, AUMODE_RECORD,
AU_RING_SIZE);
if (error)
goto bad;
}
if (!sc->sc_usemixer || sc->sc_opens + sc->sc_recopens == 0) {
sc->sc_credentials = kauth_cred_get();
kauth_cred_hold(sc->sc_credentials);
if (hw->open != NULL) {
mutex_enter(sc->sc_intr_lock);
error = hw->open(sc->hw_hdl, flags);
mutex_exit(sc->sc_intr_lock);
if (error) {
goto bad;
}
}
audio_initbufs(sc, NULL);
if (sc->sc_usemixer && audio_can_playback(sc))
audio_init_ringbuffer(sc, &sc->sc_mixring.sc_mpr,
AUMODE_PLAY);
if (sc->sc_usemixer && audio_can_capture(sc))
audio_init_ringbuffer(sc, &sc->sc_mixring.sc_mrr,
AUMODE_RECORD);
sc->schedule_wih = false;
sc->schedule_rih = false;
sc->sc_last_drops = 0;
sc->sc_eof = 0;
vc->sc_rbus = false;
sc->sc_async_audio = 0;
} else if (sc->sc_multiuser == false) {
/* XXX:NS Should be handled correctly. */
/* Do we allow multi user access */
if (kauth_cred_geteuid(sc->sc_credentials) !=
kauth_cred_geteuid(kauth_cred_get()) &&
kauth_cred_geteuid(kauth_cred_get()) != 0) {
error = EPERM;
goto bad;
}
}
mutex_enter(sc->sc_intr_lock);
vc->sc_full_duplex =
(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
(audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX);
mutex_exit(sc->sc_intr_lock);
mode = 0;
if (flags & FREAD) {
vc->sc_open |= AUOPEN_READ;
mode |= AUMODE_RECORD;
}
if (flags & FWRITE) {
vc->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 (ISDEVSOUND(dev) && sc->sc_aivalid == true) {
sc->sc_ai.mode = mode;
sc->sc_ai.play.port = ~0;
sc->sc_ai.record.port = ~0;
error = audiosetinfo(sc, &sc->sc_ai, true, vc);
} else
error = audio_set_defaults(sc, mode, vc);
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 (vc->sc_rparams.precision == 0 || vc->sc_pparams.precision == 0) {
printf("audio_open: 0 precision\n");
goto bad;
}
#endif
/* audio_close() decreases sc_mpr[n].usedlow, recalculate here */
audio_calcwater(sc, vc);
error = fd_allocfile(&fp, &fd);
if (error)
goto bad;
DPRINTF(("audio_open: done sc_mode = 0x%x\n", vc->sc_mode));
if (sc->sc_usemixer)
grow_mixer_states(sc, 2);
if (flags & FREAD)
sc->sc_recopens++;
if (flags & FWRITE)
sc->sc_opens++;
chan->dev = dev;
chan->chan = n;
chan->deschan = n;
if (sc->sc_usemixer)
SIMPLEQ_INSERT_TAIL(&sc->sc_audiochan, chan, entries);
error = fd_clone(fp, fd, flags, &audio_fileops, chan);
KASSERT(error == EMOVEFD);
*nfp = fp;
return error;
bad:
audio_destroy_pfilters(vc);
audio_destroy_rfilters(vc);
if (hw->close != NULL && sc->sc_opens == 0 && sc->sc_recopens == 0)
hw->close(sc->hw_hdl);
mutex_exit(sc->sc_lock);
if (sc->sc_usemixer) {
audio_free_ring(sc, &vc->sc_mpr);
audio_free_ring(sc, &vc->sc_mrr);
mutex_enter(sc->sc_lock);
kmem_free(vc, sizeof(struct virtual_channel));
} else
mutex_enter(sc->sc_lock);
kmem_free(chan, sizeof(struct audio_chan));
return error;
}
/*
* Must be called from task context.
*/
void
audio_init_record(struct audio_softc *sc, struct virtual_channel *vc)
{
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_recopens != 0)
return;
mutex_enter(sc->sc_intr_lock);
if (sc->hw_if->speaker_ctl &&
(!vc->sc_full_duplex || (vc->sc_mode & AUMODE_PLAY) == 0))
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF);
mutex_exit(sc->sc_intr_lock);
}
/*
* Must be called from task context.
*/
void
audio_init_play(struct audio_softc *sc, struct virtual_channel *vc)
{
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_opens != 0)
return;
mutex_enter(sc->sc_intr_lock);
vc->sc_wstamp = vc->sc_mpr.stamp;
if (sc->hw_if->speaker_ctl)
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
mutex_exit(sc->sc_intr_lock);
}
int
audio_drain(struct audio_softc *sc, struct virtual_channel *vc)
{
struct audio_ringbuffer *cb;
int error, cc, i, used;
uint drops;
bool hw = false;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(mutex_owned(sc->sc_intr_lock));
error = 0;
DPRINTF(("audio_drain: enter busy=%d\n", vc->sc_pbus));
cb = &vc->sc_mpr;
if (cb->mmapped)
return 0;
used = audio_stream_get_used(&cb->s);
if (vc == sc->sc_hwvc && sc->sc_usemixer) {
hw = true;
used += audio_stream_get_used(&sc->sc_mixring.sc_mpr.s);
}
for (i = 0; i < vc->sc_npfilters; i++)
used += audio_stream_get_used(&vc->sc_pstreams[i]);
if (used <= 0)
return 0;
if (hw == false && !vc->sc_pbus) {
/* We've never started playing, probably because the
* block was too short. Pad it and start now.
*/
uint8_t *inp = cb->s.inp;
int blksize = sc->sc_mixring.sc_mpr.blksize;
cc = blksize - (inp - cb->s.start) % blksize;
audio_fill_silence(&cb->s.param, inp, cc);
cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc);
mutex_exit(sc->sc_intr_lock);
error = audiostartp(sc, vc);
mutex_enter(sc->sc_intr_lock);
if (error)
return error;
} else if (hw == true) {
used = cb->blksize - (sc->sc_mixring.sc_mpr.s.inp -
sc->sc_mixring.sc_mpr.s.start) % cb->blksize;
while (used > 0) {
cc = sc->sc_mixring.sc_mpr.s.end -
sc->sc_mixring.sc_mpr.s.inp;
if (cc > used)
cc = used;
audio_fill_silence(&cb->s.param,
sc->sc_mixring.sc_mpr.s.inp, cc);
sc->sc_mixring.sc_mpr.s.inp =
audio_stream_add_inp(&sc->sc_mixring.sc_mpr.s,
sc->sc_mixring.sc_mpr.s.inp, cc);
used -= cc;
}
mix_write(sc);
}
/*
* 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) {
DPRINTF(("audio_drain: copying in progress!?!\n"));
cb->copying = false;
}
#endif
vc->sc_draining = true;
drops = cb->drops;
if (vc == sc->sc_hwvc)
drops += cb->blksize;
else if (sc->sc_usemixer)
drops += sc->sc_mixring.sc_mpr.blksize * PREFILL_BLOCKS;
error = 0;
while (cb->drops <= drops && !error) {
DPRINTF(("audio_drain: vc=%p used=%d, drops=%ld\n",
vc,
audio_stream_get_used(&vc->sc_mpr.s),
cb->drops));
mutex_exit(sc->sc_intr_lock);
error = audio_waitio(sc, &sc->sc_wchan, vc);
mutex_enter(sc->sc_intr_lock);
if (sc->sc_dying)
error = EIO;
}
vc->sc_draining = false;
return error;
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
audio_close(struct audio_softc *sc, int flags, struct audio_chan *chan)
{
struct virtual_channel *vc;
const struct audio_hw_if *hw;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_opens == 0 && sc->sc_recopens == 0)
return ENXIO;
vc = chan->vc;
hw = sc->hw_if;
if (hw == NULL)
return ENXIO;
mutex_enter(sc->sc_intr_lock);
DPRINTF(("audio_close: sc=%p\n", sc));
/* Stop recording. */
if (sc->sc_recopens == 1 && (flags & FREAD) && vc->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 (!vc->sc_full_duplex || hw->halt_input != hw->halt_output)
hw->halt_input(sc->hw_hdl);
vc->sc_rbus = false;
}
/*
* Block until output drains, but allow ^C interrupt.
*/
vc->sc_mpr.usedlow = vc->sc_mpr.blksize; /* avoid excessive wakeups */
/*
* If there is pending output, let it drain (unless
* the output is paused).
*/
if ((flags & FWRITE) && vc->sc_pbus) {
if (!vc->sc_mpr.pause)
audio_drain(sc, chan->vc);
vc->sc_pbus = false;
}
if ((flags & FWRITE) && (sc->sc_opens == 1)) {
if (vc->sc_mpr.mmapped == false)
audio_drain(sc, sc->sc_hwvc);
if (hw->drain)
(void)hw->drain(sc->hw_hdl);
hw->halt_output(sc->hw_hdl);
sc->sc_trigger_started = false;
}
if ((flags & FREAD) && (sc->sc_recopens == 1))
sc->sc_rec_started = false;
if (sc->sc_opens + sc->sc_recopens == 1 && hw->close != NULL)
hw->close(sc->hw_hdl);
mutex_exit(sc->sc_intr_lock);
if (sc->sc_opens + sc->sc_recopens == 1) {
sc->sc_async_audio = 0;
kauth_cred_free(sc->sc_credentials);
}
vc->sc_open = 0;
vc->sc_mode = 0;
vc->sc_full_duplex = 0;
audio_destroy_pfilters(vc);
audio_destroy_rfilters(vc);
if (flags & FREAD)
sc->sc_recopens--;
if (flags & FWRITE)
sc->sc_opens--;
if (sc->sc_usemixer) {
shrink_mixer_states(sc, 2);
SIMPLEQ_REMOVE(&sc->sc_audiochan, chan, audio_chan, entries);
mutex_exit(sc->sc_lock);
audio_free_ring(sc, &vc->sc_mpr);
audio_free_ring(sc, &vc->sc_mrr);
mutex_enter(sc->sc_lock);
kmem_free(vc, sizeof(struct virtual_channel));
}
return 0;
}
int
audio_read(struct audio_softc *sc, struct uio *uio, int ioflag,
struct virtual_channel *vc)
{
struct audio_ringbuffer *cb;
const uint8_t *outp;
uint8_t *inp;
int error, used, n;
uint cc;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->hw_if == NULL)
return ENXIO;
cb = &vc->sc_mrr;
if (cb->mmapped)
return EINVAL;
DPRINTFN(1,("audio_read: cc=%zu mode=%d\n",
uio->uio_resid, vc->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 (!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY)) {
while (uio->uio_resid > 0 && !error) {
for(;;) {
/*
* No need to lock, as any wakeup will be
* held for us while holding sc_lock.
*/
cc = vc->sc_mpr.stamp - vc->sc_wstamp;
if (cc > 0)
break;
DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n",
vc->sc_mpr.stamp, vc->sc_wstamp));
if (ioflag & IO_NDELAY)
return EWOULDBLOCK;
error = audio_waitio(sc, &sc->sc_rchan, vc);
if (sc->sc_dying)
error = EIO;
if (error)
return error;
}
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);
vc->sc_wstamp += cc;
}
return error;
}
while (uio->uio_resid > 0 && !error) {
while ((used = audio_stream_get_used(vc->sc_rustream)) <= 0) {
if (!vc->sc_rbus && !vc->sc_mrr.pause)
error = audiostartr(sc, vc);
if (error)
return error;
if (ioflag & IO_NDELAY)
return EWOULDBLOCK;
DPRINTFN(2, ("audio_read: sleep used=%d\n", used));
error = audio_waitio(sc, &sc->sc_rchan, vc);
if (sc->sc_dying)
error = EIO;
if (error)
return error;
}
outp = vc->sc_rustream->outp;
inp = vc->sc_rustream->inp;
cb->copying = true;
/*
* 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 :vc->sc_rustream->end - outp;
DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc));
n = uio->uio_resid;
mutex_exit(sc->sc_lock);
error = uiomove(__UNCONST(outp), cc, uio);
mutex_enter(sc->sc_lock);
n -= uio->uio_resid; /* number of bytes actually moved */
vc->sc_rustream->outp = audio_stream_add_outp
(vc->sc_rustream, outp, n);
cb->copying = false;
}
return error;
}
void
audio_clear(struct audio_softc *sc, struct virtual_channel *vc)
{
KASSERT(mutex_owned(sc->sc_intr_lock));
if (vc->sc_rbus) {
cv_broadcast(&sc->sc_rchan);
if (sc->sc_recopens == 1) {
sc->hw_if->halt_input(sc->hw_hdl);
sc->sc_rec_started = false;
}
vc->sc_rbus = false;
vc->sc_mrr.pause = false;
}
if (vc->sc_pbus) {
cv_broadcast(&sc->sc_wchan);
vc->sc_pbus = false;
vc->sc_mpr.pause = false;
}
}
void
audio_clear_intr_unlocked(struct audio_softc *sc, struct virtual_channel *vc)
{
mutex_enter(sc->sc_intr_lock);
audio_clear(sc, vc);
mutex_exit(sc->sc_intr_lock);
}
static void
audio_calc_latency(struct audio_softc *sc)
{
const struct audio_params *ap = &sc->sc_vchan_params;
if (ap->sample_rate == 0 || ap->channels == 0 || ap->precision == 0)
return;
sc->sc_latency = sc->sc_hwvc->sc_mpr.blksize * 1000 * PREFILL_BLOCKS
* NBBY / ap->sample_rate / ap->channels / ap->precision;
}
static void
audio_setblksize(struct audio_softc *sc, struct virtual_channel *vc,
int blksize, int mode)
{
struct audio_ringbuffer *mixcb, *cb;
audio_params_t *parm;
audio_stream_t *stream;
if (mode == AUMODE_RECORD) {
mixcb = &sc->sc_mixring.sc_mrr;
cb = &vc->sc_mrr;
parm = &vc->sc_rparams;
stream = vc->sc_rustream;
} else {
mixcb = &sc->sc_mixring.sc_mpr;
cb = &vc->sc_mpr;
parm = &vc->sc_pparams;
stream = vc->sc_pustream;
}
if (sc->sc_usemixer && vc == sc->sc_hwvc) {
mixcb->blksize = audio_calc_blksize(sc, parm);
cb->blksize = audio_calc_blksize(sc, &cb->s.param);
} else {
cb->blksize = audio_calc_blksize(sc, &stream->param);
if ((!sc->sc_usemixer && SPECIFIED(blksize)) ||
(SPECIFIED(blksize) && blksize > cb->blksize))
cb->blksize = blksize;
}
}
int
audio_calc_blksize(struct audio_softc *sc, const audio_params_t *parm)
{
int blksize;
blksize = parm->sample_rate * sc->sc_latency * parm->channels /
1000 * parm->precision / NBBY / PREFILL_BLOCKS;
return blksize;
}
void
audio_fill_silence(const 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_AC3:
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)
{
struct virtual_channel *vc;
uint8_t zerobuf[128];
int error;
int k;
vc = sc->sc_hwvc;
audio_fill_silence(&vc->sc_rparams, zerobuf, sizeof zerobuf);
error = 0;
while (n > 0 && uio->uio_resid > 0 && !error) {
k = uimin(n, uimin(uio->uio_resid, sizeof zerobuf));
mutex_exit(sc->sc_lock);
error = uiomove(zerobuf, k, uio);
mutex_enter(sc->sc_lock);
n -= k;
}
return error;
}
static int
uio_fetcher_fetch_to(struct audio_softc *sc, stream_fetcher_t *self,
audio_stream_t *p, int max_used)
{
uio_fetcher_t *this;
int size;
int stream_space;
int error;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(!cpu_intr_p());
KASSERT(!cpu_softintr_p());
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 = uimin(this->uio->uio_resid, stream_space);
/* the first fragment of the space */
stream_space = p->end - p->inp;
if (stream_space >= size) {
mutex_exit(sc->sc_lock);
error = uiomove(p->inp, size, this->uio);
mutex_enter(sc->sc_lock);
if (error)
return error;
p->inp = audio_stream_add_inp(p, p->inp, size);
} else {
mutex_exit(sc->sc_lock);
error = uiomove(p->inp, stream_space, this->uio);
mutex_enter(sc->sc_lock);
if (error)
return error;
p->inp = audio_stream_add_inp(p, p->inp, stream_space);
mutex_exit(sc->sc_lock);
error = uiomove(p->start, size - stream_space, this->uio);
mutex_enter(sc->sc_lock);
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(struct audio_softc *sc, 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,
struct virtual_channel *vc)
{
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, m, cc, used;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->hw_if == NULL)
return ENXIO;
cb = &vc->sc_mpr;
DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n",
sc, uio->uio_resid, audio_stream_get_used(vc->sc_pustream),
vc->sc_mpr.usedhigh));
if (vc->sc_mpr.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 (!vc->sc_full_duplex &&
(vc->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 (!(vc->sc_mode & AUMODE_PLAY_ALL) && vc->sc_playdrop > 0) {
m = uimin(vc->sc_playdrop, uio->uio_resid);
DPRINTF(("audio_write: playdrop %d\n", m));
uio->uio_offset += m;
uio->uio_resid -= m;
vc->sc_playdrop -= m;
if (uio->uio_resid == 0)
return 0;
}
/**
* setup filter pipeline
*/
uio_fetcher_ctor(&ufetcher, uio, vc->sc_mpr.usedhigh);
if (vc->sc_npfilters > 0) {
fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base;
} else {
fetcher = &ufetcher.base;
}
error = 0;
while (uio->uio_resid > 0 && !error) {
/* wait if the first buffer is occupied */
while ((used = audio_stream_get_used(vc->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)
return EWOULDBLOCK;
error = audio_waitio(sc, &sc->sc_wchan, vc);
if (sc->sc_dying)
error = EIO;
if (error)
return error;
}
inp = cb->s.inp;
cb->copying = true;
stream = cb->s;
used = stream.used;
/* Write to the sc_pustream as much as possible. */
if (vc->sc_npfilters > 0) {
filter = vc->sc_pfilters[0];
filter->set_fetcher(filter, &ufetcher.base);
fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base;
if (sc->sc_usemixer)
cc = sc->sc_mixring.sc_mpr.blksize * 2;
else
cc = vc->sc_mpr.blksize * 2;
error = fetcher->fetch_to(sc, fetcher, &stream, cc);
if (error != 0) {
fetcher = &ufetcher.base;
cc = vc->sc_pustream->end -
vc->sc_pustream->start;
error = fetcher->fetch_to(sc, fetcher,
vc->sc_pustream, cc);
}
} else {
fetcher = &ufetcher.base;
cc = stream.end - stream.start;
error = fetcher->fetch_to(sc, fetcher, &stream, cc);
}
if (vc->sc_npfilters > 0) {
cb->fstamp += ufetcher.last_used
- audio_stream_get_used(vc->sc_pustream);
}
cb->s.used += stream.used - used;
cb->s.inp = stream.inp;
einp = cb->s.inp;
/*
* 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 (!vc->sc_pbus && !cb->pause) {
saveerror = error;
error = audiostartp(sc, vc);
if (saveerror != 0) {
/* Report the first error that occurred. */
error = saveerror;
}
}
if (cc != 0) {
DPRINTFN(1, ("audio_write: fill %d\n", cc));
audio_fill_silence(&cb->s.param, einp, cc);
}
}
return error;
}
int
audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag,
struct lwp *l, struct audio_chan *chan)
{
const struct audio_hw_if *hw;
struct audio_chan *pchan;
struct virtual_channel *vc;
struct audio_offset *ao;
u_long stamp;
int error, offs, fd;
bool rbus, pbus;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_usemixer && chan->deschan != 0) {
SIMPLEQ_FOREACH(pchan, &sc->sc_audiochan, entries) {
if (pchan->chan == chan->deschan)
break;
}
if (pchan == NULL)
return ENXIO;
} else
pchan = chan;
if (!sc->sc_usemixer || chan->deschan != 0)
vc = pchan->vc;
else
vc = &sc->sc_mixring;
DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
hw = sc->hw_if;
if (hw == NULL)
return ENXIO;
error = 0;
switch (cmd) {
case AUDIO_GETCHAN:
if ((int *)addr != NULL)
*(int*)addr = chan->chan;
break;
case AUDIO_SETCHAN:
if ((int *)addr != NULL && *(int*)addr >= 0)
chan->deschan = *(int*)addr;
break;
case FIONBIO:
/* All handled in the upper FS layer. */
break;
case FIONREAD:
*(int *)addr = audio_stream_get_used(vc->sc_rustream);
break;
case FIOASYNC:
if (*(int *)addr) {
if (sc->sc_async_audio != 0)
error = EBUSY;
else
sc->sc_async_audio = pchan->chan;
DPRINTF(("audio_ioctl: FIOASYNC chan %d\n",
pchan->chan));
} else
sc->sc_async_audio = 0;
break;
case AUDIO_FLUSH:
DPRINTF(("AUDIO_FLUSH\n"));
rbus = vc->sc_rbus;
pbus = vc->sc_pbus;
mutex_enter(sc->sc_intr_lock);
audio_clear(sc, vc);
error = audio_initbufs(sc, vc);
if (error) {
mutex_exit(sc->sc_intr_lock);
return error;
}
mutex_exit(sc->sc_intr_lock);
if ((vc->sc_mode & AUMODE_PLAY) && !vc->sc_pbus && pbus)
error = audiostartp(sc, vc);
if (!error &&
(vc->sc_mode & AUMODE_RECORD) && !vc->sc_rbus && rbus)
error = audiostartr(sc, vc);
break;
/*
* Number of read (write) samples dropped. We don't know where or
* when they were dropped.
*/
case AUDIO_RERROR:
*(int *)addr = vc->sc_mrr.drops;
break;
case AUDIO_PERROR:
*(int *)addr = vc->sc_mpr.drops;
break;
/*
* Offsets into buffer.
*/
case AUDIO_GETIOFFS:
ao = (struct audio_offset *)addr;
HW_LOCK(vc);
/* figure out where next DMA will start */
stamp = vc->sc_rustream == &vc->sc_mrr.s
? vc->sc_mrr.stamp : vc->sc_mrr.fstamp;
offs = vc->sc_rustream->inp - vc->sc_rustream->start;
HW_UNLOCK(vc);
ao->samples = stamp;
ao->deltablks =
(stamp / vc->sc_mrr.blksize) -
(vc->sc_mrr.stamp_last / vc->sc_mrr.blksize);
vc->sc_mrr.stamp_last = stamp;
ao->offset = offs;
break;
case AUDIO_GETOOFFS:
ao = (struct audio_offset *)addr;
HW_LOCK(vc);
/* figure out where next DMA will start */
stamp = vc->sc_pustream == &vc->sc_mpr.s
? vc->sc_mpr.stamp : vc->sc_mpr.fstamp;
offs = vc->sc_pustream->outp - vc->sc_pustream->start
+ vc->sc_mpr.blksize;
HW_UNLOCK(vc);
ao->samples = stamp;
ao->deltablks =
(stamp / vc->sc_mpr.blksize) -
(vc->sc_mpr.stamp_last / vc->sc_mpr.blksize);
vc->sc_mpr.stamp_last = stamp;
if (vc->sc_pustream->start + offs >= vc->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(vc->sc_pustream);
break;
case AUDIO_SETINFO:
DPRINTF(("AUDIO_SETINFO mode=0x%x\n", vc->sc_mode));
error = audiosetinfo(sc, (struct audio_info *)addr, false, vc);
if (!error && ISDEVSOUND(dev)) {
error = audiogetinfo(sc, &sc->sc_ai, 0, vc);
sc->sc_aivalid = true;
}
break;
case AUDIO_GETINFO:
DPRINTF(("AUDIO_GETINFO\n"));
error = audiogetinfo(sc, (struct audio_info *)addr, 0, vc);
break;
case AUDIO_GETBUFINFO:
DPRINTF(("AUDIO_GETBUFINFO\n"));
error = audiogetinfo(sc, (struct audio_info *)addr, 1, vc);
break;
case AUDIO_DRAIN:
DPRINTF(("AUDIO_DRAIN\n"));
mutex_enter(sc->sc_intr_lock);
error = audio_drain(sc, pchan->vc);
if (!error && sc->sc_opens == 1 && hw->drain)
error = hw->drain(sc->hw_hdl);
mutex_exit(sc->sc_intr_lock);
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 = audio_query_encoding(sc,
(struct audio_encoding *)addr);
break;
case AUDIO_GETFD:
DPRINTF(("AUDIO_GETFD\n"));
*(int *)addr = vc->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)
vc->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,
struct virtual_channel *vc)
{
int revents;
int used;
KASSERT(mutex_owned(sc->sc_lock));
DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, vc->sc_mode));
revents = 0;
HW_LOCK(vc);
if (events & (POLLIN | POLLRDNORM)) {
used = audio_stream_get_used(vc->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 ((!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY))
? vc->sc_mpr.stamp > vc->sc_wstamp :
used > vc->sc_mrr.usedlow)
revents |= events & (POLLIN | POLLRDNORM);
}
if (events & (POLLOUT | POLLWRNORM)) {
used = audio_stream_get_used(vc->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 ((!vc->sc_full_duplex && (vc->sc_mode & AUMODE_RECORD)) ||
(!(vc->sc_mode & AUMODE_PLAY_ALL) && vc->sc_playdrop > 0) ||
(used <= vc->sc_mpr.usedlow))
revents |= events & (POLLOUT | POLLWRNORM);
}
HW_UNLOCK(vc);
if (revents == 0) {
if (events & (POLLIN | POLLRDNORM))
selrecord(l, &sc->sc_rsel);
if (events & (POLLOUT | POLLWRNORM))
selrecord(l, &sc->sc_wsel);
}
return revents;
}
static void
filt_audiordetach(struct knote *kn)
{
struct audio_softc *sc;
struct audio_chan *chan;
dev_t dev;
chan = kn->kn_hook;
dev = chan->dev;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return;
mutex_enter(sc->sc_intr_lock);
SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext);
mutex_exit(sc->sc_intr_lock);
}
static int
filt_audioread(struct knote *kn, long hint)
{
struct audio_softc *sc;
struct audio_chan *chan;
struct virtual_channel *vc;
dev_t dev;
chan = kn->kn_hook;
dev = chan->dev;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
vc = chan->vc;
mutex_enter(sc->sc_intr_lock);
if (!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY))
kn->kn_data = vc->sc_mpr.stamp - vc->sc_wstamp;
else
kn->kn_data = audio_stream_get_used(vc->sc_rustream)
- vc->sc_mrr.usedlow;
mutex_exit(sc->sc_intr_lock);
return kn->kn_data > 0;
}
static const struct filterops audioread_filtops = {
.f_isfd = 1,
.f_attach = NULL,
.f_detach = filt_audiordetach,
.f_event = filt_audioread,
};
static void
filt_audiowdetach(struct knote *kn)
{
struct audio_softc *sc;
struct audio_chan *chan;
dev_t dev;
chan = kn->kn_hook;
dev = chan->dev;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return;
mutex_enter(sc->sc_intr_lock);
SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext);
mutex_exit(sc->sc_intr_lock);
}
static int
filt_audiowrite(struct knote *kn, long hint)
{
struct audio_softc *sc;
struct audio_chan *chan;
audio_stream_t *stream;
dev_t dev;
chan = kn->kn_hook;
dev = chan->dev;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
mutex_enter(sc->sc_intr_lock);
stream = chan->vc->sc_pustream;
kn->kn_data = (stream->end - stream->start)
- audio_stream_get_used(stream);
mutex_exit(sc->sc_intr_lock);
return kn->kn_data > 0;
}
static const struct filterops audiowrite_filtops = {
.f_isfd = 1,
.f_attach = NULL,
.f_detach = filt_audiowdetach,
.f_event = filt_audiowrite,
};
int
audio_kqfilter(struct audio_chan *chan, struct knote *kn)
{
struct audio_softc *sc;
struct klist *klist;
dev_t dev;
dev = chan->dev;
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return ENXIO;
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 = chan;
mutex_enter(sc->sc_intr_lock);
SLIST_INSERT_HEAD(klist, kn, kn_selnext);
mutex_exit(sc->sc_intr_lock);
return 0;
}
int
audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot,
int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp,
struct virtual_channel *vc)
{
struct audio_ringbuffer *cb;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->hw_if == NULL)
return ENXIO;
DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)(*offp), prot));
if (!(audio_get_props(sc) & AUDIO_PROP_MMAP))
return ENOTSUP;
if (*offp < 0)
return EINVAL;
#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 = &vc->sc_mpr;
else if (prot == VM_PROT_READ)
cb = &vc->sc_mrr;
else
return EINVAL;
#else
cb = &vc->sc_mpr;
#endif
if (len > cb->s.bufsize || *offp > (uint)(cb->s.bufsize - len))
return EOVERFLOW;
if (!cb->mmapped) {
cb->mmapped = true;
if (cb == &vc->sc_mpr) {
audio_fill_silence(&cb->s.param, cb->s.start,
cb->s.bufsize);
vc->sc_pustream = &cb->s;
if (!vc->sc_pbus && !vc->sc_mpr.pause)
(void)audiostartp(sc, vc);
} else if (cb == &vc->sc_mrr) {
vc->sc_rustream = &cb->s;
if (!vc->sc_rbus && !sc->sc_mixring.sc_mrr.pause)
(void)audiostartr(sc, vc);
}
}
/* get ringbuffer */
*uobjp = cb->uobj;
/* Acquire a reference for the mmap. munmap will release.*/
uao_reference(*uobjp);
*maxprotp = prot;
*advicep = UVM_ADV_RANDOM;
*flagsp = MAP_SHARED;
return 0;
}
int
audiostartr(struct audio_softc *sc, struct virtual_channel *vc)
{
int error;
KASSERT(mutex_owned(sc->sc_lock));
DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n",
vc->sc_mrr.s.start, audio_stream_get_used(&vc->sc_mrr.s),
vc->sc_mrr.usedhigh, vc->sc_mrr.mmapped));
if (!audio_can_capture(sc))
return EINVAL;
if (vc == sc->sc_hwvc && sc->sc_usemixer)
return 0;
error = 0;
if (sc->sc_rec_started == false) {
mutex_enter(sc->sc_intr_lock);
error = mix_read(sc);
if (sc->sc_usemixer)
cv_broadcast(&sc->sc_rcondvar);
mutex_exit(sc->sc_intr_lock);
}
vc->sc_rbus = true;
return error;
}
int
audiostartp(struct audio_softc *sc, struct virtual_channel *vc)
{
int error, used;
KASSERT(mutex_owned(sc->sc_lock));
error = 0;
used = audio_stream_get_used(&vc->sc_mpr.s);
DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n",
vc->sc_mpr.s.start, used, vc->sc_mpr.usedhigh,
vc->sc_mpr.blksize, vc->sc_mpr.mmapped));
if (!audio_can_playback(sc))
return EINVAL;
if (vc == sc->sc_hwvc && sc->sc_usemixer)
return 0;
int blksize;
if (sc->sc_usemixer)
blksize = sc->sc_mixring.sc_mpr.blksize;
else
blksize = vc->sc_mpr.blksize;
if (!vc->sc_mpr.mmapped && used < blksize) {
cv_broadcast(&sc->sc_wchan);
DPRINTF(("%s: wakeup and return\n", __func__));
return 0;
}
vc->sc_pbus = true;
if (sc->sc_trigger_started == false) {
if (sc->sc_usemixer) {
audio_mix(sc);
audio_mix(sc);
audio_mix(sc);
}
mutex_enter(sc->sc_intr_lock);
error = mix_write(sc);
if (error)
goto done;
if (sc->sc_usemixer) {
vc = sc->sc_hwvc;
vc->sc_mpr.s.outp =
audio_stream_add_outp(&vc->sc_mpr.s,
vc->sc_mpr.s.outp, vc->sc_mpr.blksize);
error = mix_write(sc);
cv_broadcast(&sc->sc_condvar);
}
done:
mutex_exit(sc->sc_intr_lock);
}
return error;
}
static void
audio_softintr_rd(void *cookie)
{
struct audio_softc *sc = cookie;
proc_t *p;
pid_t pid;
mutex_enter(sc->sc_lock);
cv_broadcast(&sc->sc_rchan);
selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT);
if ((pid = sc->sc_async_audio) != 0) {
DPRINTFN(3, ("audio_softintr_rd: sending SIGIO %d\n", pid));
mutex_enter(proc_lock);
if ((p = proc_find(pid)) != NULL)
psignal(p, SIGIO);
mutex_exit(proc_lock);
}
mutex_exit(sc->sc_lock);
}
static void
audio_softintr_wr(void *cookie)
{
struct audio_softc *sc = cookie;
proc_t *p;
pid_t pid;
mutex_enter(sc->sc_lock);
cv_broadcast(&sc->sc_wchan);
selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT);
if ((pid = sc->sc_async_audio) != 0) {
DPRINTFN(3, ("audio_softintr_wr: sending SIGIO %d\n", pid));
mutex_enter(proc_lock);
if ((p = proc_find(pid)) != NULL)
psignal(p, SIGIO);
mutex_exit(proc_lock);
}
mutex_exit(sc->sc_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)
{
struct audio_softc *sc;
struct audio_ringbuffer *cb;
struct virtual_channel *vc;
int blksize, cc, used;
sc = v;
vc = sc->sc_hwvc;
blksize = vc->sc_mpr.blksize;
if (sc->sc_dying == true || sc->sc_trigger_started == false)
return;
if (sc->sc_usemixer)
cb = &sc->sc_mixring.sc_mpr;
else
cb = &vc->sc_mpr;
if (vc->sc_draining && cb->drops != sc->sc_last_drops) {
vc->sc_mpr.drops += blksize;
cv_broadcast(&sc->sc_wchan);
}
sc->sc_last_drops = cb->drops;
vc->sc_mpr.s.outp = audio_stream_add_outp(&vc->sc_mpr.s,
vc->sc_mpr.s.outp, blksize);
if (audio_stream_get_used(&cb->s) < blksize) {
DPRINTFN(3, ("HW RING - INSERT SILENCE\n"));
used = blksize;
while (used > 0) {
cc = cb->s.end - cb->s.inp;
if (cc > used)
cc = used;
audio_fill_silence(&cb->s.param, cb->s.inp, cc);
cb->s.inp =
audio_stream_add_inp(&cb->s, cb->s.inp, cc);
used -= cc;
}
vc->sc_mpr.drops += blksize;
}
mix_write(sc);
if (sc->sc_usemixer)
cv_broadcast(&sc->sc_condvar);
else
cv_broadcast(&sc->sc_wchan);
}
void
audio_mix(void *v)
{
stream_fetcher_t null_fetcher;
struct audio_softc *sc;
struct audio_chan *chan;
struct virtual_channel *vc;
struct audio_ringbuffer *cb;
stream_fetcher_t *fetcher;
uint8_t *inp;
int cc, cc1, used, blksize;
sc = v;
DPRINTF(("PINT MIX\n"));
sc->schedule_rih = false;
sc->schedule_wih = false;
sc->sc_writeme = false;
if (sc->sc_dying == true)
return;
blksize = sc->sc_mixring.sc_mpr.blksize;
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
vc = chan->vc;
if (!vc->sc_open)
continue;
if (!vc->sc_pbus)
continue;
cb = &vc->sc_mpr;
sc->sc_writeme = true;
inp = cb->s.inp;
cb->stamp += blksize;
if (cb->mmapped) {
DPRINTF(("audio_pint: vc=%p mmapped outp=%p cc=%d "
"inp=%p\n", vc, cb->s.outp, blksize,
cb->s.inp));
mutex_enter(sc->sc_intr_lock);
mix_func(sc, cb, vc);
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp,
blksize);
mutex_exit(sc->sc_intr_lock);
continue;
}
#ifdef AUDIO_INTR_TIME
{
struct timeval tv;
int64_t t;
microtime(&tv);
t = (int64_t)tv.tv_sec * 1000000 + tv.tv_usec;
if (sc->sc_pnintr) {
int64_t lastdelta, totdelta;
lastdelta = t - sc->sc_plastintr -
sc->sc_pblktime;
if (lastdelta > sc->sc_pblktime / 3) {
printf("audio: play interrupt(%d) off "
"relative by %" PRId64 " us "
"(%" PRId64 ")\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 %" PRId64 " us "
"(%" PRId64 ") (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 <= sc->sc_hwvc->sc_mpr.usedlow && !cb->copying &&
vc->sc_npfilters > 0) {
/* we might have data in filter pipeline */
null_fetcher.fetch_to = null_fetcher_fetch_to;
fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base;
vc->sc_pfilters[0]->set_fetcher(vc->sc_pfilters[0],
&null_fetcher);
used = audio_stream_get_used(vc->sc_pustream);
cc = cb->s.end - cb->s.start;
if (blksize * 2 < cc)
cc = blksize * 2;
fetcher->fetch_to(sc, fetcher, &cb->s, cc);
cb->fstamp += used -
audio_stream_get_used(vc->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 {
DPRINTF(("audio_pint: used < blksize vc=%p "
"used=%d blksize=%d\n", vc, used,
blksize));
inp = cb->s.inp;
cc = blksize - (inp - cb->s.start) % blksize;
if (cb->pause)
cb->pdrops += cc;
else {
cb->drops += cc;
vc->sc_playdrop += cc;
}
audio_fill_silence(&cb->s.param, inp, cc);
cb->s.inp = audio_stream_add_inp(&cb->s, inp,
cc);
/* Clear next block to keep ahead of the DMA. */
used = audio_stream_get_used(&cb->s);
if (used + blksize < cb->s.end - cb->s.start) {
audio_fill_silence(&cb->s.param, cb->s.inp,
blksize);
}
}
}
DPRINTFN(5, ("audio_pint: vc=%p outp=%p used=%d cc=%d\n", vc,
cb->s.outp, used, blksize));
mutex_enter(sc->sc_intr_lock);
mix_func(sc, cb, vc);
mutex_exit(sc->sc_intr_lock);
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
DPRINTFN(2, ("audio_pint: vc=%p mode=%d pause=%d used=%d "
"lowat=%d\n", vc, vc->sc_mode, cb->pause,
audio_stream_get_used(&cb->s), cb->usedlow));
if ((vc->sc_mode & AUMODE_PLAY) && !cb->pause) {
if (audio_stream_get_used(vc->sc_pustream) <= cb->usedlow)
sc->schedule_wih = true;
}
/* Possible to return one or more "phantom blocks" now. */
if (!vc->sc_full_duplex && vc->sc_mode & AUMODE_RECORD)
sc->schedule_rih = true;
}
mutex_enter(sc->sc_intr_lock);
vc = sc->sc_hwvc;
cb = &sc->sc_mixring.sc_mpr;
inp = cb->s.inp;
cc = blksize - (inp - cb->s.start) % blksize;
if (sc->sc_writeme == false) {
DPRINTFN(3, ("MIX RING EMPTY - INSERT SILENCE\n"));
audio_fill_silence(&vc->sc_pustream->param, inp, cc);
sc->sc_mixring.sc_mpr.drops += cc;
} else
cc = blksize;
cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, cc);
cc = blksize;
cc1 = sc->sc_mixring.sc_mpr.s.end - sc->sc_mixring.sc_mpr.s.inp;
if (cc1 < cc) {
audio_fill_silence(&vc->sc_pustream->param,
sc->sc_mixring.sc_mpr.s.inp, cc1);
cc -= cc1;
audio_fill_silence(&vc->sc_pustream->param,
sc->sc_mixring.sc_mpr.s.start, cc);
} else
audio_fill_silence(&vc->sc_pustream->param,
sc->sc_mixring.sc_mpr.s.inp, cc);
mutex_exit(sc->sc_intr_lock);
kpreempt_disable();
if (sc->schedule_wih == true)
softint_schedule(sc->sc_sih_wr);
if (sc->schedule_rih == true)
softint_schedule(sc->sc_sih_rd);
kpreempt_enable();
}
/*
* Called from HW driver module on completion of DMA input.
* Mark it as input in the ring buffer (fiddle pointers).
* Do a wakeup if necessary.
*/
void
audio_rint(void *v)
{
struct audio_softc *sc;
int blksize;
sc = v;
KASSERT(mutex_owned(sc->sc_intr_lock));
if (sc->sc_dying == true || sc->sc_rec_started == false)
return;
blksize = audio_stream_get_used(&sc->sc_mixring.sc_mrr.s);
sc->sc_mixring.sc_mrr.s.outp =
audio_stream_add_outp(&sc->sc_mixring.sc_mrr.s,
sc->sc_mixring.sc_mrr.s.outp, blksize);
mix_read(sc);
if (sc->sc_usemixer)
cv_broadcast(&sc->sc_rcondvar);
else
cv_broadcast(&sc->sc_rchan);
}
void
audio_upmix(void *v)
{
stream_fetcher_t null_fetcher;
struct audio_softc *sc;
struct audio_chan *chan;
struct audio_ringbuffer *cb;
stream_fetcher_t *last_fetcher;
struct virtual_channel *vc;
int cc, used, blksize, cc1;
sc = v;
blksize = sc->sc_mixring.sc_mrr.blksize;
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
vc = chan->vc;
if (!(vc->sc_open & AUOPEN_READ))
continue;
if (!vc->sc_rbus)
continue;
cb = &vc->sc_mrr;
blksize = audio_stream_get_used(&sc->sc_mixring.sc_mrr.s);
if (audio_stream_get_space(&cb->s) < blksize) {
cb->drops += blksize;
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp,
sc->sc_mixring.sc_mrr.blksize);
continue;
}
cc = blksize;
if (cb->s.inp + blksize > cb->s.end)
cc = cb->s.end - cb->s.inp;
mutex_enter(sc->sc_intr_lock);
memcpy(cb->s.inp, sc->sc_mixring.sc_mrr.s.start, cc);
if (cc < blksize && cc != 0) {
cc1 = cc;
cc = blksize - cc;
memcpy(cb->s.start,
sc->sc_mixring.sc_mrr.s.start + cc1, cc);
}
mutex_exit(sc->sc_intr_lock);
cc = blksize;
recswvol_func(sc, cb, blksize, vc);
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));
continue;
}
#ifdef AUDIO_INTR_TIME
{
struct timeval tv;
int64_t t;
microtime(&tv);
t = (int64_t)tv.tv_sec * 1000000 + tv.tv_usec;
if (sc->sc_rnintr) {
int64_t lastdelta, totdelta;
lastdelta = t - sc->sc_rlastintr -
sc->sc_rblktime;
if (lastdelta > sc->sc_rblktime / 5) {
printf("audio: record interrupt(%d) "
"off relative by %" PRId64 " us "
"(%" PRId64 ")\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 %" PRId64 " us "
"(%" PRId64 ")\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 && vc->sc_nrfilters > 0) {
null_fetcher.fetch_to = null_fetcher_fetch_to;
last_fetcher =
&vc->sc_rfilters[vc->sc_nrfilters - 1]->base;
vc->sc_rfilters[0]->set_fetcher(vc->sc_rfilters[0],
&null_fetcher);
used = audio_stream_get_used(vc->sc_rustream);
cc = vc->sc_rustream->end - vc->sc_rustream->start;
last_fetcher->fetch_to
(sc, last_fetcher, vc->sc_rustream, cc);
cb->fstamp += audio_stream_get_used(vc->sc_rustream) -
used;
/* XXX what should do for error? */
}
used = audio_stream_get_used(&vc->sc_mrr.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);
}
}
kpreempt_disable();
softint_schedule(sc->sc_sih_rd);
kpreempt_enable();
}
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:
case AUDIO_ENCODING_AC3:
break;
default:
return EINVAL;
}
/* sanity check # of channels*/
if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS)
return EINVAL;
return 0;
}
/*
* set some parameters from sc->sc_vchan_params.
*/
static int
audio_set_vchan_defaults(struct audio_softc *sc, u_int mode)
{
struct virtual_channel *vc;
struct audio_info ai;
int error;
KASSERT(mutex_owned(sc->sc_lock));
vc = sc->sc_hwvc;
/* default parameters */
vc->sc_rparams = sc->sc_vchan_params;
vc->sc_pparams = sc->sc_vchan_params;
AUDIO_INITINFO(&ai);
ai.record.sample_rate = sc->sc_vchan_params.sample_rate;
ai.record.encoding = sc->sc_vchan_params.encoding;
ai.record.channels = sc->sc_vchan_params.channels;
ai.record.precision = sc->sc_vchan_params.precision;
ai.record.pause = false;
ai.play.sample_rate = sc->sc_vchan_params.sample_rate;
ai.play.encoding = sc->sc_vchan_params.encoding;
ai.play.channels = sc->sc_vchan_params.channels;
ai.play.precision = sc->sc_vchan_params.precision;
ai.play.pause = false;
ai.mode = mode;
sc->sc_format[0].encoding = sc->sc_vchan_params.encoding;
sc->sc_format[0].channels = sc->sc_vchan_params.channels;
sc->sc_format[0].precision = sc->sc_vchan_params.precision;
sc->sc_format[0].validbits = sc->sc_vchan_params.precision;
sc->sc_format[0].frequency_type = 1;
sc->sc_format[0].frequency[0] = sc->sc_vchan_params.sample_rate;
auconv_delete_encodings(sc->sc_encodings);
error = auconv_create_encodings(sc->sc_format, VAUDIO_NFORMATS,
&sc->sc_encodings);
if (error == 0)
error = audiosetinfo(sc, &ai, true, vc);
if (error == 0) {
vc = &sc->sc_mixring;
vc->sc_rparams = sc->sc_vchan_params;
vc->sc_pparams = sc->sc_vchan_params;
}
return error;
}
int
audio_set_defaults(struct audio_softc *sc, u_int mode,
struct virtual_channel *vc)
{
struct audio_info ai;
KASSERT(mutex_owned(sc->sc_lock));
/* default parameters */
vc->sc_rparams = audio_default;
vc->sc_pparams = audio_default;
AUDIO_INITINFO(&ai);
ai.record.sample_rate = vc->sc_rparams.sample_rate;
ai.record.encoding = vc->sc_rparams.encoding;
ai.record.channels = vc->sc_rparams.channels;
ai.record.precision = vc->sc_rparams.precision;
ai.record.pause = false;
ai.play.sample_rate = vc->sc_pparams.sample_rate;
ai.play.encoding = vc->sc_pparams.encoding;
ai.play.channels = vc->sc_pparams.channels;
ai.play.precision = vc->sc_pparams.precision;
ai.play.pause = false;
ai.mode = mode;
return audiosetinfo(sc, &ai, true, vc);
}
int
au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r)
{
KASSERT(mutex_owned(sc->sc_lock));
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 (audio_set_port(sc, ct) == 0)
return 0;
ct->un.value.num_channels = 1;
ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
return audio_set_port(sc, 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;
KASSERT(mutex_owned(sc->sc_lock));
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 = audio_get_port(sc, &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 = audio_get_port(sc, &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;
KASSERT(mutex_owned(sc->sc_lock));
ct->un.value.num_channels = 2;
if (audio_get_port(sc, 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 = audio_get_port(sc, 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;
KASSERT(mutex_owned(sc->sc_lock));
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 (audio_get_port(sc, &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 (audio_get_port(sc, &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;
KASSERT(mutex_owned(sc->sc_lock));
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 = audio_set_port(sc, &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 = audio_set_port(sc, &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;
KASSERT(mutex_owned(sc->sc_lock));
if (ports->index == -1)
return 0;
ct.dev = ports->index;
ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET;
if (audio_get_port(sc, &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, bool reset,
struct virtual_channel *vc)
{
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 setmode;
int error;
int np, nr;
int blks;
u_int gain;
bool rbus, pbus;
bool cleared, modechange, pausechange;
u_char balance;
KASSERT(mutex_owned(sc->sc_lock));
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 = vc->sc_rbus;
pbus = vc->sc_pbus;
error = 0;
cleared = false;
modechange = false;
pausechange = false;
pp = vc->sc_pparams; /* Temporary encoding storage in */
rp = vc->sc_rparams; /* case setting the modes fails. */
nr = np = 0;
setmode = 0;
if (vc == &sc->sc_mixring)
goto done;
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;
if (nr > 0) {
if (!cleared) {
audio_clear_intr_unlocked(sc, vc);
cleared = true;
}
modechange = true;
setmode |= AUMODE_RECORD;
}
if (np > 0) {
if (!cleared) {
audio_clear_intr_unlocked(sc, vc);
cleared = true;
}
modechange = true;
setmode |= AUMODE_PLAY;
}
if (SPECIFIED(ai->mode)) {
if (!cleared) {
audio_clear_intr_unlocked(sc, vc);
cleared = true;
}
modechange = true;
vc->sc_mode = ai->mode;
if (vc->sc_mode & AUMODE_PLAY_ALL)
vc->sc_mode |= AUMODE_PLAY;
if ((vc->sc_mode & AUMODE_PLAY) && !vc->sc_full_duplex)
/* Play takes precedence */
vc->sc_mode &= ~AUMODE_RECORD;
}
done:
oldpus = vc->sc_pustream;
oldrus = vc->sc_rustream;
if (vc != &sc->sc_mixring && (modechange || reset)) {
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 = audio_set_params(sc, setmode,
vc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp,
&pfilters, &rfilters, vc);
if (error) {
DPRINTF(("%s: audio_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 (vc->sc_mpr.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, vc);
if (error)
goto cleanup;
}
if (setmode & AUMODE_RECORD) {
error = audio_setup_rfilters(sc, &rp, &rfilters, vc);
if (error)
goto cleanup;
}
DPRINTF(("%s: filter setup is completed.\n", __func__));
/* userland formats */
vc->sc_pparams = pp;
vc->sc_rparams = rp;
}
#ifdef AUDIO_DEBUG
if (audiodebug > 1 && nr > 0) {
audio_print_params("audiosetinfo() After setting record params:",
&vc->sc_rparams);
}
if (audiodebug > 1 && np > 0) {
audio_print_params("audiosetinfo() After setting play params:",
&vc->sc_pparams);
}
#endif
if (SPECIFIED(p->port)) {
if (!cleared) {
audio_clear_intr_unlocked(sc, vc);
cleared = true;
}
error = au_set_port(sc, &sc->sc_outports, p->port);
if (error)
goto cleanup;
}
if (SPECIFIED(r->port)) {
if (!cleared) {
audio_clear_intr_unlocked(sc, vc);
cleared = true;
}
error = au_set_port(sc, &sc->sc_inports, r->port);
if (error)
goto cleanup;
}
if (SPECIFIED(p->gain)) {
if (!sc->sc_usemixer || vc == &sc->sc_mixring) {
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_outports, p->gain, balance);
if (error)
goto cleanup;
} else
vc->sc_swvol = p->gain;
}
if (SPECIFIED(r->gain)) {
if (!sc->sc_usemixer || vc == &sc->sc_mixring) {
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_inports, r->gain, balance);
if (error)
goto cleanup;
} else
vc->sc_recswvol = r->gain;
}
if (SPECIFIED_CH(p->balance)) {
if (!sc->sc_usemixer || vc == &sc->sc_mixring) {
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)) {
if (!sc->sc_usemixer || vc == &sc->sc_mixring) {
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 = audio_set_port(sc, &ct);
if (error)
goto cleanup;
}
if (SPECIFIED_CH(p->pause)) {
pbus = !p->pause;
if (pbus != !vc->sc_mpr.pause) {
vc->sc_mpr.pause = p->pause;
pausechange = true;
}
}
if (SPECIFIED_CH(r->pause)) {
rbus = !r->pause;
if (rbus != !vc->sc_mrr.pause) {
vc->sc_mrr.pause = r->pause;
pausechange = true;
}
}
if (SPECIFIED(ai->mode)) {
if (vc->sc_mode & AUMODE_PLAY)
audio_init_play(sc, vc);
if (vc->sc_mode & AUMODE_RECORD)
audio_init_record(sc, vc);
}
if (nr > 0)
audio_setblksize(sc, vc, ai->blocksize, AUMODE_RECORD);
if (np > 0)
audio_setblksize(sc, vc, ai->blocksize, AUMODE_PLAY);
if (hw->commit_settings && sc->sc_opens + sc->sc_recopens == 0) {
error = hw->commit_settings(sc->hw_hdl);
if (error)
goto cleanup;
}
vc->sc_lastinfo = *ai;
vc->sc_lastinfovalid = true;
cleanup:
if (error == 0 && (cleared || pausechange || reset)) {
int init_error;
init_error = (pausechange == 1 && reset == 0) ? 0 :
audio_initbufs(sc, vc);
if (init_error) goto err;
if (reset || vc->sc_pustream != oldpus ||
vc->sc_rustream != oldrus)
audio_calcwater(sc, vc);
if ((vc->sc_mode & AUMODE_PLAY) &&
pbus && !vc->sc_pbus)
init_error = audiostartp(sc, vc);
if (!init_error &&
(vc->sc_mode & AUMODE_RECORD) &&
rbus && !vc->sc_rbus)
init_error = audiostartr(sc, vc);
err:
if (init_error)
return init_error;
}
/* Change water marks after initializing the buffers. */
if (SPECIFIED(ai->hiwat)) {
blks = ai->hiwat;
if (blks > vc->sc_mpr.maxblks)
blks = vc->sc_mpr.maxblks;
if (blks < PREFILL_BLOCKS + 1)
blks = PREFILL_BLOCKS + 1;
vc->sc_mpr.usedhigh = blks * vc->sc_mpr.blksize;
}
if (SPECIFIED(ai->lowat)) {
blks = ai->lowat;
if (blks > vc->sc_mpr.maxblks - 1)
blks = vc->sc_mpr.maxblks - 1;
if (blks < PREFILL_BLOCKS)
blks = PREFILL_BLOCKS;
vc->sc_mpr.usedlow = blks * vc->sc_mpr.blksize;
}
if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) {
if (vc->sc_mpr.usedlow > vc->sc_mpr.usedhigh -
vc->sc_mpr.blksize) {
vc->sc_mpr.usedlow =
vc->sc_mpr.usedhigh - vc->sc_mpr.blksize;
}
}
return error;
}
int
audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int buf_only_mode,
struct virtual_channel *vc)
{
struct audio_prinfo *r, *p;
const struct audio_hw_if *hw;
KASSERT(mutex_owned(sc->sc_lock));
r = &ai->record;
p = &ai->play;
hw = sc->hw_if;
if (hw == NULL) /* HW has not attached */
return ENXIO;
p->sample_rate = vc->sc_pparams.sample_rate;
r->sample_rate = vc->sc_rparams.sample_rate;
p->channels = vc->sc_pparams.channels;
r->channels = vc->sc_rparams.channels;
p->precision = vc->sc_pparams.precision;
r->precision = vc->sc_rparams.precision;
p->encoding = vc->sc_pparams.encoding;
r->encoding = vc->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;
if (!sc->sc_usemixer || vc == &sc->sc_mixring) {
au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance);
au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance);
} else {
p->gain = vc->sc_swvol;
r->gain = vc->sc_recswvol;
}
}
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 (audio_get_port(sc, &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(vc->sc_pustream);
r->seek = audio_stream_get_used(vc->sc_rustream);
/*
* XXX samples should be a value for userland data.
* But drops is a value for HW data.
*/
p->samples = (vc->sc_pustream == &vc->sc_mpr.s
? vc->sc_mpr.stamp : vc->sc_mpr.fstamp) - vc->sc_mpr.drops;
r->samples = (vc->sc_rustream == &vc->sc_mrr.s
? vc->sc_mrr.stamp : vc->sc_mrr.fstamp) - vc->sc_mrr.drops;
p->eof = sc->sc_eof;
r->eof = 0;
p->pause = vc->sc_mpr.pause;
r->pause = vc->sc_mrr.pause;
p->error = vc->sc_mpr.drops != 0;
r->error = vc->sc_mrr.drops != 0;
p->waiting = r->waiting = 0; /* open never hangs */
p->open = (vc->sc_open & AUOPEN_WRITE) != 0;
r->open = (vc->sc_open & AUOPEN_READ) != 0;
p->active = vc->sc_pbus;
r->active = vc->sc_rbus;
p->buffer_size = vc->sc_pustream ? vc->sc_pustream->bufsize : 0;
r->buffer_size = vc->sc_rustream ? vc->sc_rustream->bufsize : 0;
ai->blocksize = vc->sc_mpr.blksize;
if (vc->sc_mpr.blksize > 0) {
ai->hiwat = vc->sc_mpr.usedhigh / vc->sc_mpr.blksize;
ai->lowat = vc->sc_mpr.usedlow / vc->sc_mpr.blksize;
} else
ai->hiwat = ai->lowat = 0;
ai->mode = vc->sc_mode;
return 0;
}
/*
* Mixer driver
*/
int
mixer_open(dev_t dev, struct audio_softc *sc, int flags,
int ifmt, struct lwp *l, struct file **nfp)
{
struct file *fp;
struct audio_chan *chan;
int error, fd;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->hw_if == NULL)
return ENXIO;
DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc));
error = fd_allocfile(&fp, &fd);
if (error)
return error;
chan = kmem_zalloc(sizeof(struct audio_chan), KM_SLEEP);
chan->dev = dev;
error = fd_clone(fp, fd, flags, &audio_fileops, chan);
KASSERT(error == EMOVEFD);
*nfp = fp;
return error;
}
/*
* Remove a process from those to be signalled on mixer activity.
*/
static void
mixer_remove(struct audio_softc *sc)
{
struct mixer_asyncs **pm, *m;
pid_t pid;
KASSERT(mutex_owned(sc->sc_lock));
pid = curproc->p_pid;
for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
if ((*pm)->pid == pid) {
m = *pm;
*pm = m->next;
kmem_free(m, sizeof(*m));
return;
}
}
}
/*
* Signal all processes waiting for the mixer.
*/
static void
mixer_signal(struct audio_softc *sc)
{
struct mixer_asyncs *m;
proc_t *p;
for (m = sc->sc_async_mixer; m; m = m->next) {
mutex_enter(proc_lock);
if ((p = proc_find(m->pid)) != NULL)
psignal(p, SIGIO);
mutex_exit(proc_lock);
}
}
/*
* Close a mixer device
*/
/* ARGSUSED */
int
mixer_close(struct audio_softc *sc, int flags, struct audio_chan *chan)
{
KASSERT(mutex_owned(sc->sc_lock));
if (sc->hw_if == NULL)
return ENXIO;
DPRINTF(("mixer_close: sc %p\n", sc));
mixer_remove(sc);
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;
struct mixer_asyncs *ma;
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;
if (hw == NULL)
return ENXIO;
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:
if (*(int *)addr) {
ma = kmem_alloc(sizeof(struct mixer_asyncs), KM_SLEEP);
} else {
ma = NULL;
}
mixer_remove(sc); /* remove old entry */
if (ma != NULL) {
ma->next = sc->sc_async_mixer;
ma->pid = curproc->p_pid;
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 = audio_query_devinfo(sc, (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))
error = audio_get_port(sc, mc);
else if (mc->dev < 0 || 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 = audio_set_port(sc, (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 */
#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
device_t
audio_get_device(struct audio_softc *sc)
{
return sc->sc_dev;
}
#endif
#if NAUDIO > 0
static void
audio_mixer_capture(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
KASSERT(mutex_owned(sc->sc_lock));
for (mi.index = 0;; mi.index++) {
if (audio_query_devinfo(sc, &mi) != 0)
break;
KASSERT(mi.index < sc->sc_nmixer_states);
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)audio_get_port(sc, mc);
}
return;
}
static void
audio_mixer_restore(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
KASSERT(mutex_owned(sc->sc_lock));
for (mi.index = 0; ; mi.index++) {
if (audio_query_devinfo(sc, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS)
continue;
mc = &sc->sc_mixer_state[mi.index];
(void)audio_set_port(sc, 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);
struct audio_chan *chan;
const struct audio_hw_if *hwp = sc->hw_if;
struct virtual_channel *vc;
bool pbus, rbus;
pbus = rbus = false;
mutex_enter(sc->sc_lock);
audio_mixer_capture(sc);
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
vc = chan->vc;
if (vc->sc_pbus && !pbus)
pbus = true;
if (vc->sc_rbus && !rbus)
rbus = true;
}
mutex_enter(sc->sc_intr_lock);
if (pbus == true)
hwp->halt_output(sc->hw_hdl);
if (rbus == true)
hwp->halt_input(sc->hw_hdl);
mutex_exit(sc->sc_intr_lock);
#ifdef AUDIO_PM_IDLE
callout_halt(&sc->sc_idle_counter, sc->sc_lock);
#endif
mutex_exit(sc->sc_lock);
return true;
}
static bool
audio_resume(device_t dv, const pmf_qual_t *qual)
{
struct audio_softc *sc = device_private(dv);
struct audio_chan *chan;
struct virtual_channel *vc;
mutex_enter(sc->sc_lock);
sc->sc_trigger_started = false;
sc->sc_rec_started = false;
audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
audio_mixer_restore(sc);
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
vc = chan->vc;
if (vc->sc_lastinfovalid == true)
audiosetinfo(sc, &vc->sc_lastinfo, true, vc);
if (vc->sc_pbus == true && !vc->sc_mpr.pause)
audiostartp(sc, vc);
if (vc->sc_rbus == true && !vc->sc_mrr.pause)
audiostartr(sc, vc);
}
mutex_exit(sc->sc_lock);
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;
mutex_enter(sc->sc_lock);
if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) {
mi.index = sc->sc_outports.master;
mi.un.v.delta = 0;
if (audio_query_devinfo(sc, &mi) == 0) {
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);
}
}
mutex_exit(sc->sc_lock);
}
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;
mutex_enter(sc->sc_lock);
if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) {
mi.index = sc->sc_outports.master;
mi.un.v.delta = 0;
if (audio_query_devinfo(sc, &mi) == 0) {
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);
}
}
mutex_exit(sc->sc_lock);
}
static void
audio_volume_toggle(device_t dv)
{
struct audio_softc *sc = device_private(dv);
u_int gain, newgain;
u_char balance;
mutex_enter(sc->sc_lock);
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);
mutex_exit(sc->sc_lock);
}
static int
audio_get_props(struct audio_softc *sc)
{
const struct audio_hw_if *hw;
int props;
KASSERT(mutex_owned(sc->sc_lock));
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);
props |= AUDIO_PROP_MMAP;
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;
}
int
mix_read(void *arg)
{
struct audio_softc *sc = arg;
struct virtual_channel *vc;
stream_filter_t *filter;
stream_fetcher_t *fetcher;
stream_fetcher_t null_fetcher;
int cc, cc1, blksize, error;
uint8_t *inp;
vc = sc->sc_hwvc;
blksize = vc->sc_mrr.blksize;
cc = blksize;
error = 0;
if (sc->hw_if->trigger_input && sc->sc_rec_started == false) {
DPRINTF(("%s: call trigger_input\n", __func__));
sc->sc_rec_started = true;
error = sc->hw_if->trigger_input(sc->hw_hdl, vc->sc_mrr.s.start,
vc->sc_mrr.s.end, vc->sc_mrr.blksize,
audio_rint, (void *)sc, &vc->sc_mrr.s.param);
} else if (sc->hw_if->start_input) {
DPRINTF(("%s: call start_input\n", __func__));
sc->sc_rec_started = true;
error = sc->hw_if->start_input(sc->hw_hdl,
vc->sc_mrr.s.inp, vc->sc_mrr.blksize,
audio_rint, (void *)sc);
}
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_upmix restart failed: %d\n", error));
audio_clear(sc, sc->sc_hwvc);
sc->sc_rec_started = false;
return error;
}
inp = vc->sc_mrr.s.inp;
vc->sc_mrr.s.inp = audio_stream_add_inp(&vc->sc_mrr.s, inp, cc);
if (vc->sc_nrfilters > 0) {
cc = vc->sc_rustream->end - vc->sc_rustream->start;
null_fetcher.fetch_to = null_fetcher_fetch_to;
filter = vc->sc_rfilters[0];
filter->set_fetcher(filter, &null_fetcher);
fetcher = &vc->sc_rfilters[vc->sc_nrfilters - 1]->base;
fetcher->fetch_to(sc, fetcher, vc->sc_rustream, cc);
}
blksize = audio_stream_get_used(vc->sc_rustream);
cc1 = blksize;
if (vc->sc_rustream->outp + blksize > vc->sc_rustream->end)
cc1 = vc->sc_rustream->end - vc->sc_rustream->outp;
memcpy(sc->sc_mixring.sc_mrr.s.start, vc->sc_rustream->outp, cc1);
if (cc1 < blksize) {
memcpy(sc->sc_mixring.sc_mrr.s.start + cc1,
vc->sc_rustream->start, blksize - cc1);
}
sc->sc_mixring.sc_mrr.s.inp =
audio_stream_add_inp(&sc->sc_mixring.sc_mrr.s,
sc->sc_mixring.sc_mrr.s.inp, blksize);
vc->sc_rustream->outp = audio_stream_add_outp(vc->sc_rustream,
vc->sc_rustream->outp, blksize);
return error;
}
int
mix_write(void *arg)
{
struct audio_softc *sc = arg;
struct virtual_channel *vc;
stream_filter_t *filter;
stream_fetcher_t *fetcher;
stream_fetcher_t null_fetcher;
int cc, cc1, cc2, error, used;
const uint8_t *orig;
uint8_t *tocopy;
vc = sc->sc_hwvc;
error = 0;
if (sc->sc_usemixer &&
audio_stream_get_used(vc->sc_pustream) <=
sc->sc_mixring.sc_mpr.blksize) {
tocopy = vc->sc_pustream->inp;
orig = sc->sc_mixring.sc_mpr.s.outp;
used = sc->sc_mixring.sc_mpr.blksize;
while (used > 0) {
cc = used;
cc1 = vc->sc_pustream->end - tocopy;
cc2 = sc->sc_mixring.sc_mpr.s.end - orig;
if (cc > cc1)
cc = cc1;
if (cc > cc2)
cc = cc2;
memcpy(tocopy, orig, cc);
orig += cc;
tocopy += cc;
if (tocopy >= vc->sc_pustream->end)
tocopy = vc->sc_pustream->start;
if (orig >= sc->sc_mixring.sc_mpr.s.end)
orig = sc->sc_mixring.sc_mpr.s.start;
used -= cc;
}
vc->sc_pustream->inp = audio_stream_add_inp(vc->sc_pustream,
vc->sc_pustream->inp, sc->sc_mixring.sc_mpr.blksize);
sc->sc_mixring.sc_mpr.s.outp =
audio_stream_add_outp(&sc->sc_mixring.sc_mpr.s,
sc->sc_mixring.sc_mpr.s.outp,
sc->sc_mixring.sc_mpr.blksize);
}
if (vc->sc_npfilters > 0 &&
(sc->sc_usemixer || sc->sc_trigger_started)) {
null_fetcher.fetch_to = null_fetcher_fetch_to;
filter = vc->sc_pfilters[0];
filter->set_fetcher(filter, &null_fetcher);
fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base;
fetcher->fetch_to(sc, fetcher, &vc->sc_mpr.s,
vc->sc_mpr.blksize * 2);
}
if (sc->hw_if->trigger_output && sc->sc_trigger_started == false) {
DPRINTF(("%s: call trigger_output\n", __func__));
sc->sc_trigger_started = true;
error = sc->hw_if->trigger_output(sc->hw_hdl,
vc->sc_mpr.s.start, vc->sc_mpr.s.end, vc->sc_mpr.blksize,
audio_pint, (void *)sc, &vc->sc_mpr.s.param);
} else if (sc->hw_if->start_output) {
DPRINTF(("%s: call start_output\n", __func__));
sc->sc_trigger_started = true;
error = sc->hw_if->start_output(sc->hw_hdl,
__UNCONST(vc->sc_mpr.s.outp), vc->sc_mpr.blksize,
audio_pint, (void *)sc);
}
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_mix restart failed: %d\n", error));
audio_clear(sc, sc->sc_hwvc);
sc->sc_trigger_started = false;
}
return error;
}
#define DEF_MIX_FUNC(bits, type, bigger_type, MINVAL, MAXVAL) \
static void \
mix_func##bits(struct audio_softc *sc, struct audio_ringbuffer *cb, \
struct virtual_channel *vc) \
{ \
int blksize, cc, cc1, cc2, m, resid; \
bigger_type product; \
bigger_type result; \
type *orig, *tomix; \
\
blksize = sc->sc_mixring.sc_mpr.blksize; \
resid = blksize; \
\
tomix = __UNCONST(cb->s.outp); \
orig = (type *)(sc->sc_mixring.sc_mpr.s.inp); \
\
while (resid > 0) { \
cc = resid; \
cc1 = sc->sc_mixring.sc_mpr.s.end - \
(uint8_t *)orig; \
cc2 = cb->s.end - (uint8_t *)tomix; \
if (cc > cc1) \
cc = cc1; \
if (cc > cc2) \
cc = cc2; \
\
for (m = 0; m < (cc / (bits / NBBY)); m++) { \
if (vc->sc_swvol == 255) \
goto vol_done; \
tomix[m] = (bigger_type)tomix[m] * \
(bigger_type)(vc->sc_swvol) / 255; \
vol_done: \
result = (bigger_type)orig[m] + tomix[m]; \
if (sc->sc_opens == 1) \
goto adj_done; \
product = (bigger_type)orig[m] * tomix[m]; \
if (orig[m] > 0 && tomix[m] > 0) \
result -= product / MAXVAL; \
else if (orig[m] < 0 && tomix[m] < 0) \
result -= product / MINVAL; \
adj_done: \
orig[m] = result; \
} \
\
if (&orig[m] >= \
(type *)sc->sc_mixring.sc_mpr.s.end) \
orig = \
(type *)sc->sc_mixring.sc_mpr.s.start; \
if (&tomix[m] >= (type *)cb->s.end) \
tomix = (type *)cb->s.start; \
\
resid -= cc; \
} \
} \
DEF_MIX_FUNC(8, int8_t, int32_t, INT8_MIN, INT8_MAX);
DEF_MIX_FUNC(16, int16_t, int32_t, INT16_MIN, INT16_MAX);
DEF_MIX_FUNC(32, int32_t, int64_t, INT32_MIN, INT32_MAX);
void
mix_func(struct audio_softc *sc, struct audio_ringbuffer *cb,
struct virtual_channel *vc)
{
switch (sc->sc_vchan_params.precision) {
case 8:
mix_func8(sc, cb, vc);
break;
case 16:
mix_func16(sc, cb, vc);
break;
case 24:
case 32:
mix_func32(sc, cb, vc);
break;
default:
break;
}
}
#define DEF_RECSWVOL_FUNC(bits, type, bigger_type) \
static void \
recswvol_func##bits(struct audio_softc *sc, \
struct audio_ringbuffer *cb, size_t blksize, \
struct virtual_channel *vc) \
{ \
int cc, cc1, m, resid; \
type *orig; \
\
orig = (type *) cb->s.inp; \
resid = blksize; \
\
while (resid > 0) { \
cc = resid; \
cc1 = cb->s.end - (uint8_t *)orig; \
if (cc > cc1) \
cc = cc1; \
\
for (m = 0; m < (cc / (bits / 8)); m++) { \
orig[m] = (bigger_type)(orig[m] * \
(bigger_type)(vc->sc_recswvol) / 256);\
} \
orig = (type *) cb->s.start; \
\
resid -= cc; \
} \
} \
DEF_RECSWVOL_FUNC(8, int8_t, int16_t);
DEF_RECSWVOL_FUNC(16, int16_t, int32_t);
DEF_RECSWVOL_FUNC(32, int32_t, int64_t);
void
recswvol_func(struct audio_softc *sc, struct audio_ringbuffer *cb,
size_t blksize, struct virtual_channel *vc)
{
switch (sc->sc_vchan_params.precision) {
case 8:
recswvol_func8(sc, cb, blksize, vc);
break;
case 16:
recswvol_func16(sc, cb, blksize, vc);
break;
case 24:
case 32:
recswvol_func32(sc, cb, blksize, vc);
break;
default:
break;
}
}
static uint8_t *
find_vchan_vol(struct audio_softc *sc, int d)
{
struct audio_chan *chan;
size_t j, n = (size_t)d / 2;
j = 0;
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
if (j == n)
break;
j++;
}
return (d & 1) == 0 ?
&chan->vc->sc_swvol : &chan->vc->sc_recswvol;
}
static int
audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc)
{
KASSERT(mutex_owned(sc->sc_lock));
int d = mc->dev - sc->sc_static_nmixer_states;
int u = sc->sc_nmixer_states - sc->sc_static_nmixer_states;
if (d == -1 || d >= u)
return 0;
if (d < 0)
return sc->hw_if->set_port(sc->hw_hdl, mc);
uint8_t *level = &mc->un.value.level[AUDIO_MIXER_LEVEL_MONO];
uint8_t *vol = find_vchan_vol(sc, d);
*vol = *level;
return 0;
}
static int
audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc)
{
KASSERT(mutex_owned(sc->sc_lock));
int d = mc->dev - sc->sc_static_nmixer_states;
int u = sc->sc_nmixer_states - sc->sc_static_nmixer_states;
if (d == -1 || d >= u)
return 0;
if (d < 0)
return sc->hw_if->get_port(sc->hw_hdl, mc);
u_char *level = &mc->un.value.level[AUDIO_MIXER_LEVEL_MONO];
uint8_t *vol = find_vchan_vol(sc, d);
*level = *vol;
return 0;
}
static void
unitscopy(mixer_devinfo_t *di, const char *name)
{
strlcpy(di->un.v.units.name, name, sizeof(di->un.v.units.name));
}
static int
audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di)
{
struct audio_chan *chan;
unsigned int j;
KASSERT(mutex_owned(sc->sc_lock));
if (sc->sc_static_nmixer_states == 0 || sc->sc_nmixer_states == 0)
goto hardware;
if (di->index >= sc->sc_static_nmixer_states - 1 &&
di->index < sc->sc_nmixer_states) {
if (di->index == sc->sc_static_nmixer_states - 1) {
di->mixer_class = sc->sc_static_nmixer_states -1;
di->next = di->prev = AUDIO_MIXER_LAST;
strlcpy(di->label.name, AudioCvirtchan,
sizeof(di->label.name));
di->type = AUDIO_MIXER_CLASS;
} else if ((di->index - sc->sc_static_nmixer_states) % 2 == 0) {
di->mixer_class = sc->sc_static_nmixer_states -1;
j = 0;
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
if (j == (di->index -
sc->sc_static_nmixer_states) / 2)
break;
j++;
}
if (j != (di->index - sc->sc_static_nmixer_states) / 2)
return 0;
j = chan->deschan;
snprintf(di->label.name, sizeof(di->label.name),
AudioNdac"%d", j);
di->type = AUDIO_MIXER_VALUE;
di->next = di->prev = AUDIO_MIXER_LAST;
di->un.v.num_channels = 1;
unitscopy(di, AudioNvolume);
} else {
di->mixer_class = sc->sc_static_nmixer_states -1;
j = 0;
SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) {
if (j == (di->index -
sc->sc_static_nmixer_states) / 2)
break;
j++;
}
if (j != (di->index - sc->sc_static_nmixer_states) / 2)
return 0;
j = chan->deschan;
snprintf(di->label.name, sizeof(di->label.name),
AudioNmicrophone "%d", j);
di->type = AUDIO_MIXER_VALUE;
di->next = di->prev = AUDIO_MIXER_LAST;
di->un.v.num_channels = 1;
unitscopy(di, AudioNvolume);
}
return 0;
}
hardware:
return sc->hw_if->query_devinfo(sc->hw_hdl, di);
}
static int
audio_set_params(struct audio_softc *sc, int setmode, int usemode,
audio_params_t *play, audio_params_t *rec,
stream_filter_list_t *pfil, stream_filter_list_t *rfil,
const struct virtual_channel *vc)
{
int error = 0;
KASSERT(mutex_owned(sc->sc_lock));
if (vc == sc->sc_hwvc) {
sc->sc_ready = true;
if (sc->sc_usemixer && sc->sc_vchan_params.precision == 8)
play->encoding = rec->encoding = AUDIO_ENCODING_SLINEAR;
error = sc->hw_if->set_params(sc->hw_hdl, setmode, usemode,
play, rec, pfil, rfil);
if (error != 0)
sc->sc_ready = false;
return error;
}
if (setmode & AUMODE_PLAY && auconv_set_converter(sc->sc_format,
VAUDIO_NFORMATS, AUMODE_PLAY, play, true, pfil) < 0)
return EINVAL;
if (setmode & AUMODE_RECORD && auconv_set_converter(sc->sc_format,
VAUDIO_NFORMATS, AUMODE_RECORD, rec, true, rfil) < 0)
return EINVAL;
return 0;
}
static int
audio_query_encoding(struct audio_softc *sc, struct audio_encoding *ae)
{
KASSERT(mutex_owned(sc->sc_lock));
return auconv_query_encoding(sc->sc_encodings, ae);
}
void
audio_play_thread(void *v)
{
struct audio_softc *sc;
sc = (struct audio_softc *)v;
for (;;) {
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
kthread_exit(0);
}
if (!sc->sc_trigger_started)
goto play_wait;
while (!sc->sc_dying && sc->sc_usemixer &&
audio_stream_get_used(&sc->sc_mixring.sc_mpr.s) <
sc->sc_mixring.sc_mpr.blksize)
audio_mix(sc);
play_wait:
mutex_exit(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
cv_wait_sig(&sc->sc_condvar, sc->sc_intr_lock);
mutex_exit(sc->sc_intr_lock);
}
}
void
audio_rec_thread(void *v)
{
struct audio_softc *sc;
sc = (struct audio_softc *)v;
for (;;) {
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
kthread_exit(0);
}
if (!sc->sc_rec_started)
goto rec_wait;
audio_upmix(sc);
rec_wait:
mutex_exit(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
cv_wait_sig(&sc->sc_rcondvar, sc->sc_intr_lock);
mutex_exit(sc->sc_intr_lock);
}
}
/* sysctl helper to set common audio frequency */
static int
audio_sysctl_frequency(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
int t, error;
node = *rnode;
sc = node.sysctl_data;
t = sc->sc_vchan_params.sample_rate;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
mutex_enter(sc->sc_lock);
/* This may not change when a virtual channel is open */
if (sc->sc_opens || sc->sc_recopens) {
mutex_exit(sc->sc_lock);
return EBUSY;
}
if (t <= 0) {
mutex_exit(sc->sc_lock);
return EINVAL;
}
sc->sc_vchan_params.sample_rate = t;
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
if (error)
aprint_error_dev(sc->sc_dev, "Error setting frequency, "
"please check hardware capabilities\n");
if (error == 0)
audio_calc_latency(sc);
mutex_exit(sc->sc_lock);
return error;
}
/* sysctl helper to set common audio precision */
static int
audio_sysctl_precision(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
int t, error;
node = *rnode;
sc = node.sysctl_data;
t = sc->sc_vchan_params.precision;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
mutex_enter(sc->sc_lock);
/* This may not change when a virtual channel is open */
if (sc->sc_opens || sc->sc_recopens) {
mutex_exit(sc->sc_lock);
return EBUSY;
}
if (t == 0 || (t != 8 && t != 16 && t != 24 && t != 32)) {
mutex_exit(sc->sc_lock);
return EINVAL;
}
sc->sc_vchan_params.precision = t;
if (sc->sc_vchan_params.precision != 8) {
sc->sc_vchan_params.encoding =
#if BYTE_ORDER == LITTLE_ENDIAN
AUDIO_ENCODING_SLINEAR_LE;
#else
AUDIO_ENCODING_SLINEAR_BE;
#endif
} else
sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_LE;
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
if (error)
aprint_error_dev(sc->sc_dev, "Error setting precision, "
"please check hardware capabilities\n");
if (error == 0)
audio_calc_latency(sc);
mutex_exit(sc->sc_lock);
return error;
}
/* sysctl helper to enable/disable channel mixing */
static int
audio_sysctl_usemixer(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
bool t;
int error;
node = *rnode;
sc = node.sysctl_data;
t = sc->sc_usemixer;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
mutex_enter(sc->sc_lock);
/* This may not change when a virtual channel is open */
if (sc->sc_opens) {
mutex_exit(sc->sc_lock);
return EBUSY;
}
sc->sc_usemixer = t;
audio_destroy_pfilters(sc->sc_hwvc);
audio_destroy_rfilters(sc->sc_hwvc);
if (t) {
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
if (error)
aprint_error_dev(sc->sc_dev, "Error setting precision, "
"please check hardware capabilities\n");
}
if (sc->sc_usemixer) {
if (error == 0)
audio_calc_latency(sc);
} else
sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS;
mutex_exit(sc->sc_lock);
return error;
}
/* sysctl helper to set common audio channels */
static int
audio_sysctl_channels(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
int t, error;
node = *rnode;
sc = node.sysctl_data;
t = sc->sc_vchan_params.channels;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
mutex_enter(sc->sc_lock);
/* This may not change when a virtual channel is open */
if (sc->sc_opens || sc->sc_recopens) {
mutex_exit(sc->sc_lock);
return EBUSY;
}
if (t <= 0 || (t !=1 && t % 2 != 0)) {
mutex_exit(sc->sc_lock);
return EINVAL;
}
sc->sc_vchan_params.channels = t;
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
if (error)
aprint_error_dev(sc->sc_dev, "Error setting channels, "
"please check hardware capabilities\n");
if (error == 0)
audio_calc_latency(sc);
mutex_exit(sc->sc_lock);
return error;
}
/* sysctl helper to set audio latency */
static int
audio_sysctl_latency(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
int t, error;
node = *rnode;
sc = node.sysctl_data;
t = sc->sc_latency;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
mutex_enter(sc->sc_lock);
/* This may not change when a virtual channel is open */
if (sc->sc_opens || sc->sc_recopens) {
mutex_exit(sc->sc_lock);
return EBUSY;
}
if (t < 0 || t > 4000) {
mutex_exit(sc->sc_lock);
return EINVAL;
}
if (t == 0)
sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS;
else
sc->sc_latency = (unsigned int)t;
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
if (error) {
aprint_error_dev(sc->sc_dev, "Error setting latency, "
"latency restored to default\n");
sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS;
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD);
}
audio_calc_latency(sc);
mutex_exit(sc->sc_lock);
return error;
}
static int
vchan_autoconfig(struct audio_softc *sc)
{
struct virtual_channel *vc;
uint i, j, k;
int error;
vc = sc->sc_hwvc;
error = 0;
mutex_enter(sc->sc_lock);
#if BYTE_ORDER == LITTLE_ENDIAN
sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_LE;
#else
sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_BE;
#endif
for (i = 0; i < __arraycount(auto_config_precision); i++) {
sc->sc_vchan_params.precision = auto_config_precision[i];
sc->sc_vchan_params.validbits = auto_config_precision[i];
for (j = 0; j < __arraycount(auto_config_channels); j++) {
sc->sc_vchan_params.channels = auto_config_channels[j];
for (k = 0; k < __arraycount(auto_config_freq); k++) {
sc->sc_vchan_params.sample_rate =
auto_config_freq[k];
error = audio_set_vchan_defaults(sc,
AUMODE_PLAY | AUMODE_PLAY_ALL |
AUMODE_RECORD);
if (vc->sc_npfilters > 0 &&
(vc->sc_mpr.s.param.sample_rate !=
sc->sc_vchan_params.sample_rate ||
vc->sc_mpr.s.param.channels !=
sc->sc_vchan_params.channels))
error = EINVAL;
if (error == 0) {
aprint_normal_dev(sc->sc_dev,
"Virtual format configured - "
"Format SLINEAR, precision %d, "
"channels %d, frequency %d\n",
sc->sc_vchan_params.precision,
sc->sc_vchan_params.channels,
sc->sc_vchan_params.sample_rate);
goto found;
}
}
}
}
found:
if (error == 0) {
audio_calc_latency(sc);
aprint_normal_dev(sc->sc_dev, "Latency: %d milliseconds\n",
sc->sc_latency);
} else {
aprint_error_dev(sc->sc_dev, "Virtual format auto config failed!\n");
aprint_error_dev(sc->sc_dev, "Please check hardware capabilities\n");
}
mutex_exit(sc->sc_lock);
return error;
}
static void
grow_mixer_states(struct audio_softc *sc, int count)
{
mixer_ctrl_t *tmp_mixer_state;
size_t origlen = sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1);
size_t newlen = sizeof(mixer_ctrl_t) * count + origlen;
tmp_mixer_state = kmem_zalloc(newlen, KM_SLEEP);
memcpy(tmp_mixer_state, sc->sc_mixer_state, origlen);
sc->sc_nmixer_states += count;
kmem_free(sc->sc_mixer_state, origlen);
sc->sc_mixer_state = tmp_mixer_state;
}
static void
shrink_mixer_states(struct audio_softc *sc, int count)
{
mixer_ctrl_t *tmp_mixer_state;
size_t origlen = sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1);
size_t newlen = origlen - sizeof(mixer_ctrl_t) * count;
tmp_mixer_state = kmem_zalloc(newlen, KM_SLEEP);
memcpy(tmp_mixer_state, sc->sc_mixer_state, newlen);
sc->sc_nmixer_states -= count;
kmem_free(sc->sc_mixer_state, origlen);
sc->sc_mixer_state = tmp_mixer_state;
}
#endif /* NAUDIO > 0 */
#ifdef _MODULE
devmajor_t audio_bmajor = -1, audio_cmajor = -1;
#include "ioconf.c"
#endif
MODULE(MODULE_CLASS_DRIVER, audio, NULL);
static int
audio_modcmd(modcmd_t cmd, void *arg)
{
int error = 0;
#ifdef _MODULE
switch (cmd) {
case MODULE_CMD_INIT:
error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor,
&audio_cdevsw, &audio_cmajor);
if (error)
break;
error = config_init_component(cfdriver_ioconf_audio,
cfattach_ioconf_audio, cfdata_ioconf_audio);
if (error) {
devsw_detach(NULL, &audio_cdevsw);
}
break;
case MODULE_CMD_FINI:
devsw_detach(NULL, &audio_cdevsw);
error = config_fini_component(cfdriver_ioconf_audio,
cfattach_ioconf_audio, cfdata_ioconf_audio);
if (error)
devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor,
&audio_cdevsw, &audio_cmajor);
break;
default:
error = ENOTTY;
break;
}
#endif
return error;
}
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