File: [cvs.NetBSD.org] / src / sys / dev / audio / audio.c (download)
Revision 1.91.2.7, Mon Apr 5 00:48:53 2021 UTC (3 years ago) by thorpej
Branch: thorpej-cfargs
Changes since 1.91.2.6: +3 -3
lines
config_match() -> config_probe() for the straight-forward indirect config
cases. There are still a few odd balls using config_match() which should
be sorted out later.
|
/* $NetBSD: audio.c,v 1.91.2.7 2021/04/05 00:48:53 thorpej Exp $ */
/*-
* 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.
*/
/*
* Locking: there are three locks per device.
*
* - 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.
*
* - sc_exlock, private to this module. This is a variable protected by
* sc_lock. It is known as the "critical section".
* Some operations release sc_lock in order to allocate memory, to wait
* for in-flight I/O to complete, to copy to/from user context, etc.
* sc_exlock provides a critical section even under the circumstance.
* "+" in following list indicates the interfaces which necessary to be
* protected by sc_exlock.
*
* 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 +
* query_format - x
* set_format - 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
* set_port - x +
* get_port - x +
* query_devinfo - x
* allocm - - +
* freem - - +
* round_buffersize - x
* get_props - - Called at attach time
* trigger_output x x +
* trigger_input x x +
* dev_ioctl - x
* get_locks - - Called at attach time
*
* In addition, there is an additional lock.
*
* - track->lock. This is an atomic variable and is similar to the
* "interrupt lock". This is one for each track. If any thread context
* (and software interrupt context) and hardware interrupt context who
* want to access some variables on this track, they must acquire this
* lock before. It protects track's consistency between hardware
* interrupt context and others.
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.91.2.7 2021/04/05 00:48:53 thorpej Exp $");
#ifdef _KERNEL_OPT
#include "audio.h"
#include "midi.h"
#endif
#if NAUDIO > 0
#include <sys/types.h>
#include <sys/param.h>
#include <sys/atomic.h>
#include <sys/audioio.h>
#include <sys/conf.h>
#include <sys/cpu.h>
#include <sys/device.h>
#include <sys/fcntl.h>
#include <sys/file.h>
#include <sys/filedesc.h>
#include <sys/intr.h>
#include <sys/ioctl.h>
#include <sys/kauth.h>
#include <sys/kernel.h>
#include <sys/kmem.h>
#include <sys/malloc.h>
#include <sys/mman.h>
#include <sys/module.h>
#include <sys/poll.h>
#include <sys/proc.h>
#include <sys/queue.h>
#include <sys/select.h>
#include <sys/signalvar.h>
#include <sys/stat.h>
#include <sys/sysctl.h>
#include <sys/systm.h>
#include <sys/syslog.h>
#include <sys/vnode.h>
#include <dev/audio/audio_if.h>
#include <dev/audio/audiovar.h>
#include <dev/audio/audiodef.h>
#include <dev/audio/linear.h>
#include <dev/audio/mulaw.h>
#include <machine/endian.h>
#include <uvm/uvm_extern.h>
#include "ioconf.h"
/*
* 0: No debug logs
* 1: action changes like open/close/set_format...
* 2: + normal operations like read/write/ioctl...
* 3: + TRACEs except interrupt
* 4: + TRACEs including interrupt
*/
//#define AUDIO_DEBUG 1
#if defined(AUDIO_DEBUG)
int audiodebug = AUDIO_DEBUG;
static void audio_vtrace(struct audio_softc *sc, const char *, const char *,
const char *, va_list);
static void audio_trace(struct audio_softc *sc, const char *, const char *, ...)
__printflike(3, 4);
static void audio_tracet(const char *, audio_track_t *, const char *, ...)
__printflike(3, 4);
static void audio_tracef(const char *, audio_file_t *, const char *, ...)
__printflike(3, 4);
/* XXX sloppy memory logger */
static void audio_mlog_init(void);
static void audio_mlog_free(void);
static void audio_mlog_softintr(void *);
extern void audio_mlog_flush(void);
extern void audio_mlog_printf(const char *, ...);
static int mlog_refs; /* reference counter */
static char *mlog_buf[2]; /* double buffer */
static int mlog_buflen; /* buffer length */
static int mlog_used; /* used length */
static int mlog_full; /* number of dropped lines by buffer full */
static int mlog_drop; /* number of dropped lines by busy */
static volatile uint32_t mlog_inuse; /* in-use */
static int mlog_wpage; /* active page */
static void *mlog_sih; /* softint handle */
static void
audio_mlog_init(void)
{
mlog_refs++;
if (mlog_refs > 1)
return;
mlog_buflen = 4096;
mlog_buf[0] = kmem_zalloc(mlog_buflen, KM_SLEEP);
mlog_buf[1] = kmem_zalloc(mlog_buflen, KM_SLEEP);
mlog_used = 0;
mlog_full = 0;
mlog_drop = 0;
mlog_inuse = 0;
mlog_wpage = 0;
mlog_sih = softint_establish(SOFTINT_SERIAL, audio_mlog_softintr, NULL);
if (mlog_sih == NULL)
printf("%s: softint_establish failed\n", __func__);
}
static void
audio_mlog_free(void)
{
mlog_refs--;
if (mlog_refs > 0)
return;
audio_mlog_flush();
if (mlog_sih)
softint_disestablish(mlog_sih);
kmem_free(mlog_buf[0], mlog_buflen);
kmem_free(mlog_buf[1], mlog_buflen);
}
/*
* Flush memory buffer.
* It must not be called from hardware interrupt context.
*/
void
audio_mlog_flush(void)
{
if (mlog_refs == 0)
return;
/* Nothing to do if already in use ? */
if (atomic_swap_32(&mlog_inuse, 1) == 1)
return;
int rpage = mlog_wpage;
mlog_wpage ^= 1;
mlog_buf[mlog_wpage][0] = '\0';
mlog_used = 0;
atomic_swap_32(&mlog_inuse, 0);
if (mlog_buf[rpage][0] != '\0') {
printf("%s", mlog_buf[rpage]);
if (mlog_drop > 0)
printf("mlog_drop %d\n", mlog_drop);
if (mlog_full > 0)
printf("mlog_full %d\n", mlog_full);
}
mlog_full = 0;
mlog_drop = 0;
}
static void
audio_mlog_softintr(void *cookie)
{
audio_mlog_flush();
}
void
audio_mlog_printf(const char *fmt, ...)
{
int len;
va_list ap;
if (atomic_swap_32(&mlog_inuse, 1) == 1) {
/* already inuse */
mlog_drop++;
return;
}
va_start(ap, fmt);
len = vsnprintf(
mlog_buf[mlog_wpage] + mlog_used,
mlog_buflen - mlog_used,
fmt, ap);
va_end(ap);
mlog_used += len;
if (mlog_buflen - mlog_used <= 1) {
mlog_full++;
}
atomic_swap_32(&mlog_inuse, 0);
if (mlog_sih)
softint_schedule(mlog_sih);
}
/* trace functions */
static void
audio_vtrace(struct audio_softc *sc, const char *funcname, const char *header,
const char *fmt, va_list ap)
{
char buf[256];
int n;
n = 0;
buf[0] = '\0';
n += snprintf(buf + n, sizeof(buf) - n, "%s@%d %s",
funcname, device_unit(sc->sc_dev), header);
n += vsnprintf(buf + n, sizeof(buf) - n, fmt, ap);
if (cpu_intr_p()) {
audio_mlog_printf("%s\n", buf);
} else {
audio_mlog_flush();
printf("%s\n", buf);
}
}
static void
audio_trace(struct audio_softc *sc, const char *funcname, const char *fmt, ...)
{
va_list ap;
va_start(ap, fmt);
audio_vtrace(sc, funcname, "", fmt, ap);
va_end(ap);
}
static void
audio_tracet(const char *funcname, audio_track_t *track, const char *fmt, ...)
{
char hdr[16];
va_list ap;
snprintf(hdr, sizeof(hdr), "#%d ", track->id);
va_start(ap, fmt);
audio_vtrace(track->mixer->sc, funcname, hdr, fmt, ap);
va_end(ap);
}
static void
audio_tracef(const char *funcname, audio_file_t *file, const char *fmt, ...)
{
char hdr[32];
char phdr[16], rhdr[16];
va_list ap;
phdr[0] = '\0';
rhdr[0] = '\0';
if (file->ptrack)
snprintf(phdr, sizeof(phdr), "#%d", file->ptrack->id);
if (file->rtrack)
snprintf(rhdr, sizeof(rhdr), "#%d", file->rtrack->id);
snprintf(hdr, sizeof(hdr), "{%s,%s} ", phdr, rhdr);
va_start(ap, fmt);
audio_vtrace(file->sc, funcname, hdr, fmt, ap);
va_end(ap);
}
#define DPRINTF(n, fmt...) do { \
if (audiodebug >= (n)) { \
audio_mlog_flush(); \
printf(fmt); \
} \
} while (0)
#define TRACE(n, fmt...) do { \
if (audiodebug >= (n)) audio_trace(sc, __func__, fmt); \
} while (0)
#define TRACET(n, t, fmt...) do { \
if (audiodebug >= (n)) audio_tracet(__func__, t, fmt); \
} while (0)
#define TRACEF(n, f, fmt...) do { \
if (audiodebug >= (n)) audio_tracef(__func__, f, fmt); \
} while (0)
struct audio_track_debugbuf {
char usrbuf[32];
char codec[32];
char chvol[32];
char chmix[32];
char freq[32];
char outbuf[32];
};
static void
audio_track_bufstat(audio_track_t *track, struct audio_track_debugbuf *buf)
{
memset(buf, 0, sizeof(*buf));
snprintf(buf->outbuf, sizeof(buf->outbuf), " out=%d/%d/%d",
track->outbuf.head, track->outbuf.used, track->outbuf.capacity);
if (track->freq.filter)
snprintf(buf->freq, sizeof(buf->freq), " f=%d/%d/%d",
track->freq.srcbuf.head,
track->freq.srcbuf.used,
track->freq.srcbuf.capacity);
if (track->chmix.filter)
snprintf(buf->chmix, sizeof(buf->chmix), " m=%d",
track->chmix.srcbuf.used);
if (track->chvol.filter)
snprintf(buf->chvol, sizeof(buf->chvol), " v=%d",
track->chvol.srcbuf.used);
if (track->codec.filter)
snprintf(buf->codec, sizeof(buf->codec), " e=%d",
track->codec.srcbuf.used);
snprintf(buf->usrbuf, sizeof(buf->usrbuf), " usr=%d/%d/H%d",
track->usrbuf.head, track->usrbuf.used, track->usrbuf_usedhigh);
}
#else
#define DPRINTF(n, fmt...) do { } while (0)
#define TRACE(n, fmt, ...) do { } while (0)
#define TRACET(n, t, fmt, ...) do { } while (0)
#define TRACEF(n, f, fmt, ...) do { } while (0)
#endif
#define SPECIFIED(x) ((x) != ~0)
#define SPECIFIED_CH(x) ((x) != (u_char)~0)
/*
* Default hardware blocksize in msec.
*
* We use 10 msec for most modern platforms. This period is good enough to
* play audio and video synchronizely.
* In contrast, for very old platforms, this is usually too short and too
* severe. Also such platforms usually can not play video confortably, so
* it's not so important to make the blocksize shorter. If the platform
* defines its own value as __AUDIO_BLK_MS in its <machine/param.h>, it
* uses this instead.
*
* In either case, you can overwrite AUDIO_BLK_MS by your kernel
* configuration file if you wish.
*/
#if !defined(AUDIO_BLK_MS)
# if defined(__AUDIO_BLK_MS)
# define AUDIO_BLK_MS __AUDIO_BLK_MS
# else
# define AUDIO_BLK_MS (10)
# endif
#endif
/* Device timeout in msec */
#define AUDIO_TIMEOUT (3000)
/* #define AUDIO_PM_IDLE */
#ifdef AUDIO_PM_IDLE
int audio_idle_timeout = 30;
#endif
/* Number of elements of async mixer's pid */
#define AM_CAPACITY (4)
struct portname {
const char *name;
int mask;
};
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 void audio_softintr_rd(void *);
static void audio_softintr_wr(void *);
static void audio_printf(struct audio_softc *, const char *, ...)
__printflike(2, 3);
static int audio_exlock_mutex_enter(struct audio_softc *);
static void audio_exlock_mutex_exit(struct audio_softc *);
static int audio_exlock_enter(struct audio_softc *);
static void audio_exlock_exit(struct audio_softc *);
static void audio_sc_acquire_foropen(struct audio_softc *, struct psref *);
static struct audio_softc *audio_sc_acquire_fromfile(audio_file_t *,
struct psref *);
static void audio_sc_release(struct audio_softc *, struct psref *);
static int audio_track_waitio(struct audio_softc *, audio_track_t *);
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 audiommap(struct file *, off_t *, size_t, int, int *, int *,
struct uvm_object **, int *);
static int audiostat(struct file *, struct stat *);
static void filt_audiowrite_detach(struct knote *);
static int filt_audiowrite_event(struct knote *, long);
static void filt_audioread_detach(struct knote *);
static int filt_audioread_event(struct knote *, long);
static int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *,
audio_file_t **);
static int audio_close(struct audio_softc *, audio_file_t *);
static int audio_unlink(struct audio_softc *, audio_file_t *);
static int audio_read(struct audio_softc *, struct uio *, int, audio_file_t *);
static int audio_write(struct audio_softc *, struct uio *, int, audio_file_t *);
static void audio_file_clear(struct audio_softc *, audio_file_t *);
static int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int,
struct lwp *, audio_file_t *);
static int audio_poll(struct audio_softc *, int, struct lwp *, audio_file_t *);
static int audio_kqfilter(struct audio_softc *, audio_file_t *, struct knote *);
static int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *,
struct uvm_object **, int *, audio_file_t *);
static int audioctl_open(dev_t, struct audio_softc *, int, int, struct lwp *);
static void audio_pintr(void *);
static void audio_rintr(void *);
static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *);
static __inline int audio_track_readablebytes(const audio_track_t *);
static int audio_file_setinfo(struct audio_softc *, audio_file_t *,
const struct audio_info *);
static int audio_track_setinfo_check(audio_track_t *,
audio_format2_t *, const struct audio_prinfo *);
static void audio_track_setinfo_water(audio_track_t *,
const struct audio_info *);
static int audio_hw_setinfo(struct audio_softc *, const struct audio_info *,
struct audio_info *);
static int audio_hw_set_format(struct audio_softc *, int,
const audio_format2_t *, const audio_format2_t *,
audio_filter_reg_t *, audio_filter_reg_t *);
static int audiogetinfo(struct audio_softc *, struct audio_info *, int,
audio_file_t *);
static bool audio_can_playback(struct audio_softc *);
static bool audio_can_capture(struct audio_softc *);
static int audio_check_params(audio_format2_t *);
static int audio_mixers_init(struct audio_softc *sc, int,
const audio_format2_t *, const audio_format2_t *,
const audio_filter_reg_t *, const audio_filter_reg_t *);
static int audio_select_freq(const struct audio_format *);
static int audio_hw_probe(struct audio_softc *, audio_format2_t *, int);
static int audio_hw_validate_format(struct audio_softc *, int,
const audio_format2_t *);
static int audio_mixers_set_format(struct audio_softc *,
const struct audio_info *);
static void audio_mixers_get_format(struct audio_softc *, struct audio_info *);
static int audio_sysctl_blk_ms(SYSCTLFN_PROTO);
static int audio_sysctl_multiuser(SYSCTLFN_PROTO);
#if defined(AUDIO_DEBUG)
static int audio_sysctl_debug(SYSCTLFN_PROTO);
static void audio_format2_tostr(char *, size_t, const audio_format2_t *);
static void audio_print_format2(const char *, const audio_format2_t *) __unused;
#endif
static void *audio_realloc(void *, size_t);
static int audio_realloc_usrbuf(audio_track_t *, int);
static void audio_free_usrbuf(audio_track_t *);
static audio_track_t *audio_track_create(struct audio_softc *,
audio_trackmixer_t *);
static void audio_track_destroy(audio_track_t *);
static audio_filter_t audio_track_get_codec(audio_track_t *,
const audio_format2_t *, const audio_format2_t *);
static int audio_track_set_format(audio_track_t *, audio_format2_t *);
static void audio_track_play(audio_track_t *);
static int audio_track_drain(struct audio_softc *, audio_track_t *);
static void audio_track_record(audio_track_t *);
static void audio_track_clear(struct audio_softc *, audio_track_t *);
static int audio_mixer_init(struct audio_softc *, int,
const audio_format2_t *, const audio_filter_reg_t *);
static void audio_mixer_destroy(struct audio_softc *, audio_trackmixer_t *);
static void audio_pmixer_start(struct audio_softc *, bool);
static void audio_pmixer_process(struct audio_softc *);
static void audio_pmixer_agc(audio_trackmixer_t *, int);
static int audio_pmixer_mix_track(audio_trackmixer_t *, audio_track_t *, int);
static void audio_pmixer_output(struct audio_softc *);
static int audio_pmixer_halt(struct audio_softc *);
static void audio_rmixer_start(struct audio_softc *);
static void audio_rmixer_process(struct audio_softc *);
static void audio_rmixer_input(struct audio_softc *);
static int audio_rmixer_halt(struct audio_softc *);
static void mixer_init(struct audio_softc *);
static int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *);
static int mixer_close(struct audio_softc *, audio_file_t *);
static int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *);
static void mixer_async_add(struct audio_softc *, pid_t);
static void mixer_async_remove(struct audio_softc *, pid_t);
static void mixer_signal(struct audio_softc *);
static int au_portof(struct audio_softc *, char *, int);
static void au_setup_ports(struct audio_softc *, struct au_mixer_ports *,
mixer_devinfo_t *, const struct portname *);
static int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int);
static int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *);
static int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int);
static void au_get_gain(struct audio_softc *, struct au_mixer_ports *,
u_int *, u_char *);
static int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int);
static int au_get_port(struct audio_softc *, struct au_mixer_ports *);
static int au_set_monitor_gain(struct audio_softc *, int);
static int au_get_monitor_gain(struct audio_softc *);
static int audio_get_port(struct audio_softc *, mixer_ctrl_t *);
static int audio_set_port(struct audio_softc *, mixer_ctrl_t *);
static __inline struct audio_params
format2_to_params(const audio_format2_t *f2)
{
audio_params_t p;
/* validbits/precision <-> precision/stride */
p.sample_rate = f2->sample_rate;
p.channels = f2->channels;
p.encoding = f2->encoding;
p.validbits = f2->precision;
p.precision = f2->stride;
return p;
}
static __inline audio_format2_t
params_to_format2(const struct audio_params *p)
{
audio_format2_t f2;
/* precision/stride <-> validbits/precision */
f2.sample_rate = p->sample_rate;
f2.channels = p->channels;
f2.encoding = p->encoding;
f2.precision = p->validbits;
f2.stride = p->precision;
return f2;
}
/* Return true if this track is a playback track. */
static __inline bool
audio_track_is_playback(const audio_track_t *track)
{
return ((track->mode & AUMODE_PLAY) != 0);
}
/* Return true if this track is a recording track. */
static __inline bool
audio_track_is_record(const audio_track_t *track)
{
return ((track->mode & AUMODE_RECORD) != 0);
}
#if 0 /* XXX Not used yet */
/*
* Convert 0..255 volume used in userland to internal presentation 0..256.
*/
static __inline u_int
audio_volume_to_inner(u_int v)
{
return v < 127 ? v : v + 1;
}
/*
* Convert 0..256 internal presentation to 0..255 volume used in userland.
*/
static __inline u_int
audio_volume_to_outer(u_int v)
{
return v < 127 ? v : v - 1;
}
#endif /* 0 */
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 = audiommap,
.fo_kqfilter = audiokqfilter,
.fo_restart = fnullop_restart
};
/* The default audio mode: 8 kHz mono mu-law */
static const struct audio_params audio_default = {
.sample_rate = 8000,
.encoding = AUDIO_ENCODING_ULAW,
.precision = 8,
.validbits = 8,
.channels = 1,
};
static const char *encoding_names[] = {
"none",
AudioEmulaw,
AudioEalaw,
"pcm16",
"pcm8",
AudioEadpcm,
AudioEslinear_le,
AudioEslinear_be,
AudioEulinear_le,
AudioEulinear_be,
AudioEslinear,
AudioEulinear,
AudioEmpeg_l1_stream,
AudioEmpeg_l1_packets,
AudioEmpeg_l1_system,
AudioEmpeg_l2_stream,
AudioEmpeg_l2_packets,
AudioEmpeg_l2_system,
AudioEac3,
};
/*
* Returns encoding name corresponding to AUDIO_ENCODING_*.
* Note that it may return a local buffer because it is mainly for debugging.
*/
const char *
audio_encoding_name(int encoding)
{
static char buf[16];
if (0 <= encoding && encoding < __arraycount(encoding_names)) {
return encoding_names[encoding];
} else {
snprintf(buf, sizeof(buf), "enc=%d", encoding);
return buf;
}
}
/*
* Supported encodings used by AUDIO_GETENC.
* index and flags are set by code.
* XXX is there any needs for SLINEAR_OE:>=16/ULINEAR_OE:>=16 ?
*/
static const audio_encoding_t audio_encodings[] = {
{ 0, AudioEmulaw, AUDIO_ENCODING_ULAW, 8, 0 },
{ 0, AudioEalaw, AUDIO_ENCODING_ALAW, 8, 0 },
{ 0, AudioEslinear, AUDIO_ENCODING_SLINEAR, 8, 0 },
{ 0, AudioEulinear, AUDIO_ENCODING_ULINEAR, 8, 0 },
{ 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 16, 0 },
{ 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 16, 0 },
{ 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 16, 0 },
{ 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 16, 0 },
#if defined(AUDIO_SUPPORT_LINEAR24)
{ 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 24, 0 },
{ 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 24, 0 },
{ 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 24, 0 },
{ 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 24, 0 },
#endif
{ 0, AudioEslinear_le, AUDIO_ENCODING_SLINEAR_LE, 32, 0 },
{ 0, AudioEulinear_le, AUDIO_ENCODING_ULINEAR_LE, 32, 0 },
{ 0, AudioEslinear_be, AUDIO_ENCODING_SLINEAR_BE, 32, 0 },
{ 0, AudioEulinear_be, AUDIO_ENCODING_ULINEAR_BE, 32, 0 },
};
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 }
};
static struct psref_class *audio_psref_class __read_mostly;
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(1, "%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;
const struct audio_hw_if *hw_if;
audio_format2_t phwfmt;
audio_format2_t rhwfmt;
audio_filter_reg_t pfil;
audio_filter_reg_t rfil;
const struct sysctlnode *node;
void *hdlp;
bool has_playback;
bool has_capture;
bool has_indep;
bool has_fulldup;
int mode;
int error;
sc = device_private(self);
sc->sc_dev = self;
sa = (struct audio_attach_args *)aux;
hw_if = sa->hwif;
hdlp = sa->hdl;
if (hw_if == NULL) {
panic("audioattach: missing hw_if method");
}
if (hw_if->get_locks == NULL || hw_if->get_props == NULL) {
aprint_error(": missing mandatory method\n");
return;
}
hw_if->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock);
sc->sc_props = hw_if->get_props(hdlp);
has_playback = (sc->sc_props & AUDIO_PROP_PLAYBACK);
has_capture = (sc->sc_props & AUDIO_PROP_CAPTURE);
has_indep = (sc->sc_props & AUDIO_PROP_INDEPENDENT);
has_fulldup = (sc->sc_props & AUDIO_PROP_FULLDUPLEX);
#ifdef DIAGNOSTIC
if (hw_if->query_format == NULL ||
hw_if->set_format == NULL ||
hw_if->getdev == NULL ||
hw_if->set_port == NULL ||
hw_if->get_port == NULL ||
hw_if->query_devinfo == NULL) {
aprint_error(": missing mandatory method\n");
return;
}
if (has_playback) {
if ((hw_if->start_output == NULL &&
hw_if->trigger_output == NULL) ||
hw_if->halt_output == NULL) {
aprint_error(": missing playback method\n");
}
}
if (has_capture) {
if ((hw_if->start_input == NULL &&
hw_if->trigger_input == NULL) ||
hw_if->halt_input == NULL) {
aprint_error(": missing capture method\n");
}
}
#endif
sc->hw_if = hw_if;
sc->hw_hdl = hdlp;
sc->hw_dev = parent;
sc->sc_exlock = 1;
sc->sc_blk_ms = AUDIO_BLK_MS;
SLIST_INIT(&sc->sc_files);
cv_init(&sc->sc_exlockcv, "audiolk");
sc->sc_am_capacity = 0;
sc->sc_am_used = 0;
sc->sc_am = NULL;
/* MMAP is now supported by upper layer. */
sc->sc_props |= AUDIO_PROP_MMAP;
KASSERT(has_playback || has_capture);
/* Unidirectional device must have neither FULLDUP nor INDEPENDENT. */
if (!has_playback || !has_capture) {
KASSERT(!has_indep);
KASSERT(!has_fulldup);
}
mode = 0;
if (has_playback) {
aprint_normal(": playback");
mode |= AUMODE_PLAY;
}
if (has_capture) {
aprint_normal("%c capture", has_playback ? ',' : ':');
mode |= AUMODE_RECORD;
}
if (has_playback && has_capture) {
if (has_fulldup)
aprint_normal(", full duplex");
else
aprint_normal(", half duplex");
if (has_indep)
aprint_normal(", independent");
}
aprint_naive("\n");
aprint_normal("\n");
/* probe hw params */
memset(&phwfmt, 0, sizeof(phwfmt));
memset(&rhwfmt, 0, sizeof(rhwfmt));
memset(&pfil, 0, sizeof(pfil));
memset(&rfil, 0, sizeof(rfil));
if (has_indep) {
int perror, rerror;
/* On independent devices, probe separately. */
perror = audio_hw_probe(sc, &phwfmt, AUMODE_PLAY);
rerror = audio_hw_probe(sc, &rhwfmt, AUMODE_RECORD);
if (perror && rerror) {
aprint_error_dev(self,
"audio_hw_probe failed: perror=%d, rerror=%d\n",
perror, rerror);
goto bad;
}
if (perror) {
mode &= ~AUMODE_PLAY;
aprint_error_dev(self, "audio_hw_probe failed: "
"errno=%d, playback disabled\n", perror);
}
if (rerror) {
mode &= ~AUMODE_RECORD;
aprint_error_dev(self, "audio_hw_probe failed: "
"errno=%d, capture disabled\n", rerror);
}
} else {
/*
* On non independent devices or uni-directional devices,
* probe once (simultaneously).
*/
audio_format2_t *fmt = has_playback ? &phwfmt : &rhwfmt;
error = audio_hw_probe(sc, fmt, mode);
if (error) {
aprint_error_dev(self,
"audio_hw_probe failed: errno=%d\n", error);
goto bad;
}
if (has_playback && has_capture)
rhwfmt = phwfmt;
}
/* Init hardware. */
/* hw_probe() also validates [pr]hwfmt. */
error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (error) {
aprint_error_dev(self,
"audio_hw_set_format failed: errno=%d\n", error);
goto bad;
}
/*
* Init track mixers. If at least one direction is available on
* attach time, we assume a success.
*/
error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (sc->sc_pmixer == NULL && sc->sc_rmixer == NULL) {
aprint_error_dev(self,
"audio_mixers_init failed: errno=%d\n", error);
goto bad;
}
sc->sc_psz = pserialize_create();
psref_target_init(&sc->sc_psref, audio_psref_class);
selinit(&sc->sc_wsel);
selinit(&sc->sc_rsel);
/* Initial parameter of /dev/sound */
sc->sc_sound_pparams = params_to_format2(&audio_default);
sc->sc_sound_rparams = params_to_format2(&audio_default);
sc->sc_sound_ppause = false;
sc->sc_sound_rpause = false;
/* XXX TODO: consider about sc_ai */
mixer_init(sc);
TRACE(2, "inputs ports=0x%x, input master=%d, "
"output ports=0x%x, output master=%d",
sc->sc_inports.allports, sc->sc_inports.master,
sc->sc_outports.allports, sc->sc_outports.master);
sysctl_createv(&sc->sc_log, 0, NULL, &node,
0,
CTLTYPE_NODE, device_xname(sc->sc_dev),
SYSCTL_DESCR("audio test"),
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, "blk_ms",
SYSCTL_DESCR("blocksize in msec"),
audio_sysctl_blk_ms, 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 access"),
audio_sysctl_multiuser, 0, (void *)sc, 0,
CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL);
#if defined(AUDIO_DEBUG)
sysctl_createv(&sc->sc_log, 0, NULL, NULL,
CTLFLAG_READWRITE,
CTLTYPE_INT, "debug",
SYSCTL_DESCR("debug level (0..4)"),
audio_sysctl_debug, 0, (void *)sc, 0,
CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL);
#endif
}
#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
#if defined(AUDIO_DEBUG)
audio_mlog_init();
#endif
audiorescan(self, NULL, NULL);
sc->sc_exlock = 0;
return;
bad:
/* Clearing hw_if means that device is attached but disabled. */
sc->hw_if = NULL;
sc->sc_exlock = 0;
aprint_error_dev(sc->sc_dev, "disabled\n");
return;
}
/*
* Initialize hardware mixer.
* This function is called from audioattach().
*/
static void
mixer_init(struct audio_softc *sc)
{
mixer_devinfo_t mi;
int iclass, mclass, oclass, rclass;
int record_master_found, record_source_found;
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_nmixer_states = mi.index;
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);
}
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;
cv_broadcast(&sc->sc_exlockcv);
mutex_exit(sc->sc_lock);
return 0;
default:
return EOPNOTSUPP;
}
}
static int
audiodetach(device_t self, int flags)
{
struct audio_softc *sc;
struct audio_file *file;
int error;
sc = device_private(self);
TRACE(2, "flags=%d", flags);
/* device is not initialized */
if (sc->hw_if == NULL)
return 0;
/* Start draining existing accessors of the device. */
error = config_detach_children(self, flags);
if (error)
return error;
/*
* This waits currently running sysctls to finish if exists.
* After this, no more new sysctls will come.
*/
sysctl_teardown(&sc->sc_log);
mutex_enter(sc->sc_lock);
sc->sc_dying = true;
cv_broadcast(&sc->sc_exlockcv);
if (sc->sc_pmixer)
cv_broadcast(&sc->sc_pmixer->outcv);
if (sc->sc_rmixer)
cv_broadcast(&sc->sc_rmixer->outcv);
/* Prevent new users */
SLIST_FOREACH(file, &sc->sc_files, entry) {
atomic_store_relaxed(&file->dying, true);
}
/*
* Wait for existing users to drain.
* - pserialize_perform waits for all pserialize_read sections on
* all CPUs; after this, no more new psref_acquire can happen.
* - psref_target_destroy waits for all extant acquired psrefs to
* be psref_released.
*/
pserialize_perform(sc->sc_psz);
mutex_exit(sc->sc_lock);
psref_target_destroy(&sc->sc_psref, audio_psref_class);
/*
* We are now guaranteed that there are no calls to audio fileops
* that hold sc, and any new calls with files that were for sc will
* fail. Thus, we now have exclusive access to the softc.
*/
sc->sc_exlock = 1;
/*
* Clean up all open instances.
* Here, we no longer need any locks to traverse sc_files.
*/
while ((file = SLIST_FIRST(&sc->sc_files)) != NULL) {
audio_unlink(sc, file);
}
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 */
if (sc->sc_pmixer) {
audio_mixer_destroy(sc, sc->sc_pmixer);
kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer));
}
if (sc->sc_rmixer) {
audio_mixer_destroy(sc, sc->sc_rmixer);
kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer));
}
if (sc->sc_am)
kern_free(sc->sc_am);
seldestroy(&sc->sc_wsel);
seldestroy(&sc->sc_rsel);
#ifdef AUDIO_PM_IDLE
callout_destroy(&sc->sc_idle_counter);
#endif
cv_destroy(&sc->sc_exlockcv);
#if defined(AUDIO_DEBUG)
audio_mlog_free();
#endif
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_probe(parent, cf, aux))
config_attach(parent, cf, aux, NULL,
CFARG_LOCATORS,
CFARG_EOL);
return 0;
}
static int
audiorescan(device_t self, const char *ifattr, const int *locators)
{
struct audio_softc *sc = device_private(self);
config_search(sc->sc_dev, NULL,
CFARG_SEARCH, audiosearch,
CFARG_EOL);
return 0;
}
/*
* 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, CFARG_EOL);
}
/*
* audio_printf() outputs fmt... with the audio device name and MD device
* name prefixed. If the message is considered to be related to the MD
* driver, use this one instead of device_printf().
*/
static void
audio_printf(struct audio_softc *sc, const char *fmt, ...)
{
va_list ap;
printf("%s(%s): ", device_xname(sc->sc_dev), device_xname(sc->hw_dev));
va_start(ap, fmt);
vprintf(fmt, ap);
va_end(ap);
}
/*
* Enter critical section and also keep sc_lock.
* If successful, returns 0 with sc_lock held. Otherwise returns errno.
* Must be called without sc_lock held.
*/
static int
audio_exlock_mutex_enter(struct audio_softc *sc)
{
int error;
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
return EIO;
}
while (__predict_false(sc->sc_exlock != 0)) {
error = cv_wait_sig(&sc->sc_exlockcv, sc->sc_lock);
if (sc->sc_dying)
error = EIO;
if (error) {
mutex_exit(sc->sc_lock);
return error;
}
}
/* Acquire */
sc->sc_exlock = 1;
return 0;
}
/*
* Exit critical section and exit sc_lock.
* Must be called with sc_lock held.
*/
static void
audio_exlock_mutex_exit(struct audio_softc *sc)
{
KASSERT(mutex_owned(sc->sc_lock));
sc->sc_exlock = 0;
cv_broadcast(&sc->sc_exlockcv);
mutex_exit(sc->sc_lock);
}
/*
* Enter critical section.
* If successful, it returns 0. Otherwise returns errno.
* Must be called without sc_lock held.
* This function returns without sc_lock held.
*/
static int
audio_exlock_enter(struct audio_softc *sc)
{
int error;
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
mutex_exit(sc->sc_lock);
return 0;
}
/*
* Exit critical section.
* Must be called without sc_lock held.
*/
static void
audio_exlock_exit(struct audio_softc *sc)
{
mutex_enter(sc->sc_lock);
audio_exlock_mutex_exit(sc);
}
/*
* Increment reference counter for this sc.
* This is intended to be used for open.
*/
void
audio_sc_acquire_foropen(struct audio_softc *sc, struct psref *refp)
{
int s;
/* Block audiodetach while we acquire a reference */
s = pserialize_read_enter();
/*
* We don't examine sc_dying here. However, all open methods
* call audio_exlock_enter() right after this, so we can examine
* sc_dying in it.
*/
/* Acquire a reference */
psref_acquire(refp, &sc->sc_psref, audio_psref_class);
/* Now sc won't go away until we drop the reference count */
pserialize_read_exit(s);
}
/*
* Get sc from file, and increment reference counter for this sc.
* This is intended to be used for methods other than open.
* If successful, returns sc. Otherwise returns NULL.
*/
struct audio_softc *
audio_sc_acquire_fromfile(audio_file_t *file, struct psref *refp)
{
int s;
bool dying;
/* Block audiodetach while we acquire a reference */
s = pserialize_read_enter();
/* If close or audiodetach already ran, tough -- no more audio */
dying = atomic_load_relaxed(&file->dying);
if (dying) {
pserialize_read_exit(s);
return NULL;
}
/* Acquire a reference */
psref_acquire(refp, &file->sc->sc_psref, audio_psref_class);
/* Now sc won't go away until we drop the reference count */
pserialize_read_exit(s);
return file->sc;
}
/*
* Decrement reference counter for this sc.
*/
void
audio_sc_release(struct audio_softc *sc, struct psref *refp)
{
psref_release(refp, &sc->sc_psref, audio_psref_class);
}
/*
* Wait for I/O to complete, releasing sc_lock.
* Must be called with sc_lock held.
*/
static int
audio_track_waitio(struct audio_softc *sc, audio_track_t *track)
{
int error;
KASSERT(track);
KASSERT(mutex_owned(sc->sc_lock));
/* Wait for pending I/O to complete. */
error = cv_timedwait_sig(&track->mixer->outcv, sc->sc_lock,
mstohz(AUDIO_TIMEOUT));
if (sc->sc_suspending) {
/* If it's about to suspend, ignore timeout error. */
if (error == EWOULDBLOCK) {
TRACET(2, track, "timeout (suspending)");
return 0;
}
}
if (sc->sc_dying) {
error = EIO;
}
if (error) {
TRACET(2, track, "cv_timedwait_sig failed %d", error);
if (error == EWOULDBLOCK)
audio_printf(sc, "device timeout\n");
} else {
TRACET(3, track, "wakeup");
}
return error;
}
/*
* Try to acquire track lock.
* It doesn't block if the track lock is already aquired.
* Returns true if the track lock was acquired, or false if the track
* lock was already acquired.
*/
static __inline bool
audio_track_lock_tryenter(audio_track_t *track)
{
return (atomic_cas_uint(&track->lock, 0, 1) == 0);
}
/*
* Acquire track lock.
*/
static __inline void
audio_track_lock_enter(audio_track_t *track)
{
/* Don't sleep here. */
while (audio_track_lock_tryenter(track) == false)
;
}
/*
* Release track lock.
*/
static __inline void
audio_track_lock_exit(audio_track_t *track)
{
atomic_swap_uint(&track->lock, 0);
}
static int
audioopen(dev_t dev, int flags, int ifmt, struct lwp *l)
{
struct audio_softc *sc;
struct psref sc_ref;
int bound;
int error;
/* Find the device */
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL || sc->hw_if == NULL)
return ENXIO;
bound = curlwp_bind();
audio_sc_acquire_foropen(sc, &sc_ref);
error = audio_exlock_enter(sc);
if (error)
goto done;
device_active(sc->sc_dev, DVA_SYSTEM);
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_open(dev, sc, flags, ifmt, l, NULL);
break;
case AUDIOCTL_DEVICE:
error = audioctl_open(dev, sc, flags, ifmt, l);
break;
case MIXER_DEVICE:
error = mixer_open(dev, sc, flags, ifmt, l);
break;
default:
error = ENXIO;
break;
}
audio_exlock_exit(sc);
done:
audio_sc_release(sc, &sc_ref);
curlwp_bindx(bound);
return error;
}
static int
audioclose(struct file *fp)
{
struct audio_softc *sc;
struct psref sc_ref;
audio_file_t *file;
int bound;
int error;
dev_t dev;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
error = 0;
/*
* audioclose() must
* - unplug track from the trackmixer (and unplug anything from softc),
* if sc exists.
* - free all memory objects, regardless of sc.
*/
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc) {
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_close(sc, file);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
case MIXER_DEVICE:
error = mixer_close(sc, file);
break;
default:
error = ENXIO;
break;
}
audio_sc_release(sc, &sc_ref);
}
curlwp_bindx(bound);
/* Free memory objects anyway */
TRACEF(2, file, "free memory");
if (file->ptrack)
audio_track_destroy(file->ptrack);
if (file->rtrack)
audio_track_destroy(file->rtrack);
kmem_free(file, sizeof(*file));
fp->f_audioctx = NULL;
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 psref sc_ref;
audio_file_t *file;
int bound;
int error;
dev_t dev;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
if (fp->f_flag & O_NONBLOCK)
ioflag |= IO_NDELAY;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_read(sc, uio, ioflag, file);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
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 psref sc_ref;
audio_file_t *file;
int bound;
int error;
dev_t dev;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
if (fp->f_flag & O_NONBLOCK)
ioflag |= IO_NDELAY;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_write(sc, uio, ioflag, file);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
static int
audioioctl(struct file *fp, u_long cmd, void *addr)
{
struct audio_softc *sc;
struct psref sc_ref;
audio_file_t *file;
struct lwp *l = curlwp;
int bound;
int error;
dev_t dev;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
case AUDIOCTL_DEVICE:
mutex_enter(sc->sc_lock);
device_active(sc->sc_dev, DVA_SYSTEM);
mutex_exit(sc->sc_lock);
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,
file);
break;
case MIXER_DEVICE:
error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l);
break;
default:
error = ENXIO;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
static int
audiostat(struct file *fp, struct stat *st)
{
struct audio_softc *sc;
struct psref sc_ref;
audio_file_t *file;
int bound;
int error;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
error = 0;
memset(st, 0, sizeof(*st));
st->st_dev = file->dev;
st->st_uid = kauth_cred_geteuid(fp->f_cred);
st->st_gid = kauth_cred_getegid(fp->f_cred);
st->st_mode = S_IFCHR;
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
static int
audiopoll(struct file *fp, int events)
{
struct audio_softc *sc;
struct psref sc_ref;
audio_file_t *file;
struct lwp *l = curlwp;
int bound;
int revents;
dev_t dev;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
revents = POLLERR;
goto done;
}
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
revents = audio_poll(sc, events, l, file);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
revents = 0;
break;
default:
revents = POLLERR;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return revents;
}
static int
audiokqfilter(struct file *fp, struct knote *kn)
{
struct audio_softc *sc;
struct psref sc_ref;
audio_file_t *file;
dev_t dev;
int bound;
int error;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_kqfilter(sc, file, kn);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
static int
audiommap(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 psref sc_ref;
audio_file_t *file;
dev_t dev;
int bound;
int error;
KASSERT(fp->f_audioctx);
file = fp->f_audioctx;
dev = file->dev;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
mutex_enter(sc->sc_lock);
device_active(sc->sc_dev, DVA_SYSTEM); /* XXXJDM */
mutex_exit(sc->sc_lock);
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_mmap(sc, offp, len, prot, flagsp, advicep,
uobjp, maxprotp, file);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
default:
error = ENOTSUP;
break;
}
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
/* Exported interfaces for audiobell. */
/*
* Open for audiobell.
* It stores allocated file to *filep.
* If successful returns 0, otherwise errno.
*/
int
audiobellopen(dev_t dev, audio_file_t **filep)
{
struct audio_softc *sc;
struct psref sc_ref;
int bound;
int error;
/* Find the device */
sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL || sc->hw_if == NULL)
return ENXIO;
bound = curlwp_bind();
audio_sc_acquire_foropen(sc, &sc_ref);
error = audio_exlock_enter(sc);
if (error)
goto done;
device_active(sc->sc_dev, DVA_SYSTEM);
error = audio_open(dev, sc, FWRITE, 0, curlwp, filep);
audio_exlock_exit(sc);
done:
audio_sc_release(sc, &sc_ref);
curlwp_bindx(bound);
return error;
}
/* Close for audiobell */
int
audiobellclose(audio_file_t *file)
{
struct audio_softc *sc;
struct psref sc_ref;
int bound;
int error;
error = 0;
/*
* audiobellclose() must
* - unplug track from the trackmixer if sc exist.
* - free all memory objects, regardless of sc.
*/
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc) {
error = audio_close(sc, file);
audio_sc_release(sc, &sc_ref);
}
curlwp_bindx(bound);
/* Free memory objects anyway */
KASSERT(file->ptrack);
audio_track_destroy(file->ptrack);
KASSERT(file->rtrack == NULL);
kmem_free(file, sizeof(*file));
return error;
}
/* Set sample rate for audiobell */
int
audiobellsetrate(audio_file_t *file, u_int sample_rate)
{
struct audio_softc *sc;
struct psref sc_ref;
struct audio_info ai;
int bound;
int error;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done1;
}
AUDIO_INITINFO(&ai);
ai.play.sample_rate = sample_rate;
error = audio_exlock_enter(sc);
if (error)
goto done2;
error = audio_file_setinfo(sc, file, &ai);
audio_exlock_exit(sc);
done2:
audio_sc_release(sc, &sc_ref);
done1:
curlwp_bindx(bound);
return error;
}
/* Playback for audiobell */
int
audiobellwrite(audio_file_t *file, struct uio *uio)
{
struct audio_softc *sc;
struct psref sc_ref;
int bound;
int error;
bound = curlwp_bind();
sc = audio_sc_acquire_fromfile(file, &sc_ref);
if (sc == NULL) {
error = EIO;
goto done;
}
error = audio_write(sc, uio, 0, file);
audio_sc_release(sc, &sc_ref);
done:
curlwp_bindx(bound);
return error;
}
/*
* Audio driver
*/
/*
* Must be called with sc_exlock held and without sc_lock held.
*/
int
audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l, audio_file_t **bellfile)
{
struct audio_info ai;
struct file *fp;
audio_file_t *af;
audio_ring_t *hwbuf;
bool fullduplex;
bool cred_held;
bool hw_opened;
bool rmixer_started;
bool inserted;
int fd;
int error;
KASSERT(sc->sc_exlock);
TRACE(1, "%sdev=%s flags=0x%x po=%d ro=%d",
(audiodebug >= 3) ? "start " : "",
ISDEVSOUND(dev) ? "sound" : "audio",
flags, sc->sc_popens, sc->sc_ropens);
fp = NULL;
cred_held = false;
hw_opened = false;
rmixer_started = false;
inserted = false;
af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP);
af->sc = sc;
af->dev = dev;
if ((flags & FWRITE) != 0 && audio_can_playback(sc))
af->mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
if ((flags & FREAD) != 0 && audio_can_capture(sc))
af->mode |= AUMODE_RECORD;
if (af->mode == 0) {
error = ENXIO;
goto bad;
}
fullduplex = (sc->sc_props & AUDIO_PROP_FULLDUPLEX);
/*
* On half duplex hardware,
* 1. if mode is (PLAY | REC), let mode PLAY.
* 2. if mode is PLAY, let mode PLAY if no rec tracks, otherwise error.
* 3. if mode is REC, let mode REC if no play tracks, otherwise error.
*/
if (fullduplex == false) {
if ((af->mode & AUMODE_PLAY)) {
if (sc->sc_ropens != 0) {
TRACE(1, "record track already exists");
error = ENODEV;
goto bad;
}
/* Play takes precedence */
af->mode &= ~AUMODE_RECORD;
}
if ((af->mode & AUMODE_RECORD)) {
if (sc->sc_popens != 0) {
TRACE(1, "play track already exists");
error = ENODEV;
goto bad;
}
}
}
/* Create tracks */
if ((af->mode & AUMODE_PLAY))
af->ptrack = audio_track_create(sc, sc->sc_pmixer);
if ((af->mode & AUMODE_RECORD))
af->rtrack = audio_track_create(sc, sc->sc_rmixer);
/* Set parameters */
AUDIO_INITINFO(&ai);
if (bellfile) {
/* If audiobell, only sample_rate will be set later. */
ai.play.sample_rate = audio_default.sample_rate;
ai.play.encoding = AUDIO_ENCODING_SLINEAR_NE;
ai.play.channels = 1;
ai.play.precision = 16;
ai.play.pause = 0;
} else if (ISDEVAUDIO(dev)) {
/* If /dev/audio, initialize everytime. */
ai.play.sample_rate = audio_default.sample_rate;
ai.play.encoding = audio_default.encoding;
ai.play.channels = audio_default.channels;
ai.play.precision = audio_default.precision;
ai.play.pause = 0;
ai.record.sample_rate = audio_default.sample_rate;
ai.record.encoding = audio_default.encoding;
ai.record.channels = audio_default.channels;
ai.record.precision = audio_default.precision;
ai.record.pause = 0;
} else {
/* If /dev/sound, take over the previous parameters. */
ai.play.sample_rate = sc->sc_sound_pparams.sample_rate;
ai.play.encoding = sc->sc_sound_pparams.encoding;
ai.play.channels = sc->sc_sound_pparams.channels;
ai.play.precision = sc->sc_sound_pparams.precision;
ai.play.pause = sc->sc_sound_ppause;
ai.record.sample_rate = sc->sc_sound_rparams.sample_rate;
ai.record.encoding = sc->sc_sound_rparams.encoding;
ai.record.channels = sc->sc_sound_rparams.channels;
ai.record.precision = sc->sc_sound_rparams.precision;
ai.record.pause = sc->sc_sound_rpause;
}
error = audio_file_setinfo(sc, af, &ai);
if (error)
goto bad;
if (sc->sc_popens + sc->sc_ropens == 0) {
/* First open */
sc->sc_cred = kauth_cred_get();
kauth_cred_hold(sc->sc_cred);
cred_held = true;
if (sc->hw_if->open) {
int hwflags;
/*
* Call hw_if->open() only at first open of
* combination of playback and recording.
* On full duplex hardware, the flags passed to
* hw_if->open() is always (FREAD | FWRITE)
* regardless of this open()'s flags.
* see also dev/isa/aria.c
* On half duplex hardware, the flags passed to
* hw_if->open() is either FREAD or FWRITE.
* see also arch/evbarm/mini2440/audio_mini2440.c
*/
if (fullduplex) {
hwflags = FREAD | FWRITE;
} else {
/* Construct hwflags from af->mode. */
hwflags = 0;
if ((af->mode & AUMODE_PLAY) != 0)
hwflags |= FWRITE;
if ((af->mode & AUMODE_RECORD) != 0)
hwflags |= FREAD;
}
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->open(sc->hw_hdl, hwflags);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
if (error)
goto bad;
}
/*
* Regardless of whether we called hw_if->open (whether
* hw_if->open exists) or not, we move to the Opened phase
* here. Therefore from this point, we have to call
* hw_if->close (if exists) whenever abort.
* Note that both of hw_if->{open,close} are optional.
*/
hw_opened = true;
/*
* Set speaker mode when a half duplex.
* XXX I'm not sure this is correct.
*/
if (1/*XXX*/) {
if (sc->hw_if->speaker_ctl) {
int on;
if (af->ptrack) {
on = 1;
} else {
on = 0;
}
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->speaker_ctl(sc->hw_hdl, on);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
if (error)
goto bad;
}
}
} else if (sc->sc_multiuser == false) {
uid_t euid = kauth_cred_geteuid(kauth_cred_get());
if (euid != 0 && euid != kauth_cred_geteuid(sc->sc_cred)) {
error = EPERM;
goto bad;
}
}
/* Call init_output if this is the first playback open. */
if (af->ptrack && sc->sc_popens == 0) {
if (sc->hw_if->init_output) {
hwbuf = &sc->sc_pmixer->hwbuf;
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->init_output(sc->hw_hdl,
hwbuf->mem,
hwbuf->capacity *
hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
if (error)
goto bad;
}
}
/*
* Call init_input and start rmixer, if this is the first recording
* open. See pause consideration notes.
*/
if (af->rtrack && sc->sc_ropens == 0) {
if (sc->hw_if->init_input) {
hwbuf = &sc->sc_rmixer->hwbuf;
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->init_input(sc->hw_hdl,
hwbuf->mem,
hwbuf->capacity *
hwbuf->fmt.channels * hwbuf->fmt.stride / NBBY);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
if (error)
goto bad;
}
mutex_enter(sc->sc_lock);
audio_rmixer_start(sc);
mutex_exit(sc->sc_lock);
rmixer_started = true;
}
/*
* This is the last sc_lock section in the function, so we have to
* examine sc_dying again before starting the rest tasks. Because
* audiodeatch() may have been invoked (and it would set sc_dying)
* from the time audioopen() was executed until now. If it happens,
* audiodetach() may already have set file->dying for all sc_files
* that exist at that point, so that audioopen() must abort without
* inserting af to sc_files, in order to keep consistency.
*/
mutex_enter(sc->sc_lock);
if (sc->sc_dying) {
mutex_exit(sc->sc_lock);
goto bad;
}
/* Count up finally */
if (af->ptrack)
sc->sc_popens++;
if (af->rtrack)
sc->sc_ropens++;
mutex_enter(sc->sc_intr_lock);
SLIST_INSERT_HEAD(&sc->sc_files, af, entry);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
inserted = true;
if (bellfile) {
*bellfile = af;
} else {
error = fd_allocfile(&fp, &fd);
if (error)
goto bad;
error = fd_clone(fp, fd, flags, &audio_fileops, af);
KASSERTMSG(error == EMOVEFD, "error=%d", error);
}
/* Be nothing else after fd_clone */
TRACEF(3, af, "done");
return error;
bad:
if (inserted) {
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
SLIST_REMOVE(&sc->sc_files, af, audio_file, entry);
mutex_exit(sc->sc_intr_lock);
if (af->ptrack)
sc->sc_popens--;
if (af->rtrack)
sc->sc_ropens--;
mutex_exit(sc->sc_lock);
}
if (rmixer_started) {
mutex_enter(sc->sc_lock);
audio_rmixer_halt(sc);
mutex_exit(sc->sc_lock);
}
if (hw_opened) {
if (sc->hw_if->close) {
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
sc->hw_if->close(sc->hw_hdl);
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
}
}
if (cred_held) {
kauth_cred_free(sc->sc_cred);
}
/*
* Since track here is not yet linked to sc_files,
* you can call track_destroy() without sc_intr_lock.
*/
if (af->rtrack) {
audio_track_destroy(af->rtrack);
af->rtrack = NULL;
}
if (af->ptrack) {
audio_track_destroy(af->ptrack);
af->ptrack = NULL;
}
kmem_free(af, sizeof(*af));
return error;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_close(struct audio_softc *sc, audio_file_t *file)
{
int error;
/* Protect entering new fileops to this file */
atomic_store_relaxed(&file->dying, true);
/*
* Drain first.
* It must be done before unlinking(acquiring exlock).
*/
if (file->ptrack) {
mutex_enter(sc->sc_lock);
audio_track_drain(sc, file->ptrack);
mutex_exit(sc->sc_lock);
}
error = audio_exlock_enter(sc);
if (error) {
/*
* If EIO, this sc is about to detach. In this case, even if
* we don't do subsequent _unlink(), audiodetach() will do it.
*/
if (error == EIO)
return error;
/* XXX This should not happen but what should I do ? */
panic("%s: can't acquire exlock: errno=%d", __func__, error);
}
error = audio_unlink(sc, file);
audio_exlock_exit(sc);
return error;
}
/*
* Unlink this file, but not freeing memory here.
* Must be called with sc_exlock held and without sc_lock held.
*/
int
audio_unlink(struct audio_softc *sc, audio_file_t *file)
{
int error;
mutex_enter(sc->sc_lock);
TRACEF(1, file, "%spid=%d.%d po=%d ro=%d",
(audiodebug >= 3) ? "start " : "",
(int)curproc->p_pid, (int)curlwp->l_lid,
sc->sc_popens, sc->sc_ropens);
KASSERTMSG(sc->sc_popens + sc->sc_ropens > 0,
"sc->sc_popens=%d, sc->sc_ropens=%d",
sc->sc_popens, sc->sc_ropens);
device_active(sc->sc_dev, DVA_SYSTEM);
mutex_enter(sc->sc_intr_lock);
SLIST_REMOVE(&sc->sc_files, file, audio_file, entry);
mutex_exit(sc->sc_intr_lock);
if (file->ptrack) {
TRACET(3, file->ptrack, "dropframes=%" PRIu64,
file->ptrack->dropframes);
KASSERT(sc->sc_popens > 0);
sc->sc_popens--;
/* Call hw halt_output if this is the last playback track. */
if (sc->sc_popens == 0 && sc->sc_pbusy) {
error = audio_pmixer_halt(sc);
if (error) {
audio_printf(sc,
"halt_output failed: errno=%d (ignored)\n",
error);
}
}
/* Restore mixing volume if all tracks are gone. */
if (sc->sc_popens == 0) {
/* intr_lock is not necessary, but just manners. */
mutex_enter(sc->sc_intr_lock);
sc->sc_pmixer->volume = 256;
sc->sc_pmixer->voltimer = 0;
mutex_exit(sc->sc_intr_lock);
}
}
if (file->rtrack) {
TRACET(3, file->rtrack, "dropframes=%" PRIu64,
file->rtrack->dropframes);
KASSERT(sc->sc_ropens > 0);
sc->sc_ropens--;
/* Call hw halt_input if this is the last recording track. */
if (sc->sc_ropens == 0 && sc->sc_rbusy) {
error = audio_rmixer_halt(sc);
if (error) {
audio_printf(sc,
"halt_input failed: errno=%d (ignored)\n",
error);
}
}
}
/* Call hw close if this is the last track. */
if (sc->sc_popens + sc->sc_ropens == 0) {
if (sc->hw_if->close) {
TRACE(2, "hw_if close");
mutex_enter(sc->sc_intr_lock);
sc->hw_if->close(sc->hw_hdl);
mutex_exit(sc->sc_intr_lock);
}
}
mutex_exit(sc->sc_lock);
if (sc->sc_popens + sc->sc_ropens == 0)
kauth_cred_free(sc->sc_cred);
TRACE(3, "done");
return 0;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_read(struct audio_softc *sc, struct uio *uio, int ioflag,
audio_file_t *file)
{
audio_track_t *track;
audio_ring_t *usrbuf;
audio_ring_t *input;
int error;
/*
* On half-duplex hardware, O_RDWR is treated as O_WRONLY.
* However read() system call itself can be called because it's
* opened with O_RDWR. So in this case, deny this read().
*/
track = file->rtrack;
if (track == NULL) {
return EBADF;
}
/* I think it's better than EINVAL. */
if (track->mmapped)
return EPERM;
TRACET(2, track, "resid=%zd ioflag=0x%x", uio->uio_resid, ioflag);
#ifdef AUDIO_PM_IDLE
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
if (device_is_active(&sc->sc_dev) || sc->sc_idle)
device_active(&sc->sc_dev, DVA_SYSTEM);
/* In recording, unlike playback, read() never operates rmixer. */
audio_exlock_mutex_exit(sc);
#endif
usrbuf = &track->usrbuf;
input = track->input;
error = 0;
while (uio->uio_resid > 0 && error == 0) {
int bytes;
TRACET(3, track,
"while resid=%zd input=%d/%d/%d usrbuf=%d/%d/H%d",
uio->uio_resid,
input->head, input->used, input->capacity,
usrbuf->head, usrbuf->used, track->usrbuf_usedhigh);
/* Wait when buffers are empty. */
mutex_enter(sc->sc_lock);
for (;;) {
bool empty;
audio_track_lock_enter(track);
empty = (input->used == 0 && usrbuf->used == 0);
audio_track_lock_exit(track);
if (!empty)
break;
if ((ioflag & IO_NDELAY)) {
mutex_exit(sc->sc_lock);
return EWOULDBLOCK;
}
TRACET(3, track, "sleep");
error = audio_track_waitio(sc, track);
if (error) {
mutex_exit(sc->sc_lock);
return error;
}
}
mutex_exit(sc->sc_lock);
audio_track_lock_enter(track);
audio_track_record(track);
/* uiomove from usrbuf as much as possible. */
bytes = uimin(usrbuf->used, uio->uio_resid);
while (bytes > 0) {
int head = usrbuf->head;
int len = uimin(bytes, usrbuf->capacity - head);
error = uiomove((uint8_t *)usrbuf->mem + head, len,
uio);
if (error) {
audio_track_lock_exit(track);
device_printf(sc->sc_dev,
"%s: uiomove(%d) failed: errno=%d\n",
__func__, len, error);
goto abort;
}
auring_take(usrbuf, len);
track->useriobytes += len;
TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d",
len,
usrbuf->head, usrbuf->used, usrbuf->capacity);
bytes -= len;
}
audio_track_lock_exit(track);
}
abort:
return error;
}
/*
* Clear file's playback and/or record track buffer immediately.
*/
static void
audio_file_clear(struct audio_softc *sc, audio_file_t *file)
{
if (file->ptrack)
audio_track_clear(sc, file->ptrack);
if (file->rtrack)
audio_track_clear(sc, file->rtrack);
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_write(struct audio_softc *sc, struct uio *uio, int ioflag,
audio_file_t *file)
{
audio_track_t *track;
audio_ring_t *usrbuf;
audio_ring_t *outbuf;
int error;
track = file->ptrack;
KASSERT(track);
/* I think it's better than EINVAL. */
if (track->mmapped)
return EPERM;
TRACET(2, track, "%sresid=%zd pid=%d.%d ioflag=0x%x",
audiodebug >= 3 ? "begin " : "",
uio->uio_resid, (int)curproc->p_pid, (int)curlwp->l_lid, ioflag);
if (uio->uio_resid == 0) {
track->eofcounter++;
return 0;
}
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
#ifdef AUDIO_PM_IDLE
if (device_is_active(&sc->sc_dev) || sc->sc_idle)
device_active(&sc->sc_dev, DVA_SYSTEM);
#endif
/*
* The first write starts pmixer.
*/
if (sc->sc_pbusy == false)
audio_pmixer_start(sc, false);
audio_exlock_mutex_exit(sc);
usrbuf = &track->usrbuf;
outbuf = &track->outbuf;
track->pstate = AUDIO_STATE_RUNNING;
error = 0;
while (uio->uio_resid > 0 && error == 0) {
int bytes;
TRACET(3, track, "while resid=%zd usrbuf=%d/%d/H%d",
uio->uio_resid,
usrbuf->head, usrbuf->used, track->usrbuf_usedhigh);
/* Wait when buffers are full. */
mutex_enter(sc->sc_lock);
for (;;) {
bool full;
audio_track_lock_enter(track);
full = (usrbuf->used >= track->usrbuf_usedhigh &&
outbuf->used >= outbuf->capacity);
audio_track_lock_exit(track);
if (!full)
break;
if ((ioflag & IO_NDELAY)) {
error = EWOULDBLOCK;
mutex_exit(sc->sc_lock);
goto abort;
}
TRACET(3, track, "sleep usrbuf=%d/H%d",
usrbuf->used, track->usrbuf_usedhigh);
error = audio_track_waitio(sc, track);
if (error) {
mutex_exit(sc->sc_lock);
goto abort;
}
}
mutex_exit(sc->sc_lock);
audio_track_lock_enter(track);
/* uiomove to usrbuf as much as possible. */
bytes = uimin(track->usrbuf_usedhigh - usrbuf->used,
uio->uio_resid);
while (bytes > 0) {
int tail = auring_tail(usrbuf);
int len = uimin(bytes, usrbuf->capacity - tail);
error = uiomove((uint8_t *)usrbuf->mem + tail, len,
uio);
if (error) {
audio_track_lock_exit(track);
device_printf(sc->sc_dev,
"%s: uiomove(%d) failed: errno=%d\n",
__func__, len, error);
goto abort;
}
auring_push(usrbuf, len);
track->useriobytes += len;
TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d",
len,
usrbuf->head, usrbuf->used, usrbuf->capacity);
bytes -= len;
}
/* Convert them as much as possible. */
while (usrbuf->used >= track->usrbuf_blksize &&
outbuf->used < outbuf->capacity) {
audio_track_play(track);
}
audio_track_lock_exit(track);
}
abort:
TRACET(3, track, "done error=%d", error);
return error;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag,
struct lwp *l, audio_file_t *file)
{
struct audio_offset *ao;
struct audio_info ai;
audio_track_t *track;
audio_encoding_t *ae;
audio_format_query_t *query;
u_int stamp;
u_int offs;
int fd;
int index;
int error;
#if defined(AUDIO_DEBUG)
const char *ioctlnames[] = {
" AUDIO_GETINFO", /* 21 */
" AUDIO_SETINFO", /* 22 */
" AUDIO_DRAIN", /* 23 */
" AUDIO_FLUSH", /* 24 */
" AUDIO_WSEEK", /* 25 */
" AUDIO_RERROR", /* 26 */
" AUDIO_GETDEV", /* 27 */
" AUDIO_GETENC", /* 28 */
" AUDIO_GETFD", /* 29 */
" AUDIO_SETFD", /* 30 */
" AUDIO_PERROR", /* 31 */
" AUDIO_GETIOFFS", /* 32 */
" AUDIO_GETOOFFS", /* 33 */
" AUDIO_GETPROPS", /* 34 */
" AUDIO_GETBUFINFO", /* 35 */
" AUDIO_SETCHAN", /* 36 */
" AUDIO_GETCHAN", /* 37 */
" AUDIO_QUERYFORMAT", /* 38 */
" AUDIO_GETFORMAT", /* 39 */
" AUDIO_SETFORMAT", /* 40 */
};
int nameidx = (cmd & 0xff);
const char *ioctlname = "";
if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames))
ioctlname = ioctlnames[nameidx - 21];
TRACEF(2, file, "(%lu,'%c',%lu)%s pid=%d.%d",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname,
(int)curproc->p_pid, (int)l->l_lid);
#endif
error = 0;
switch (cmd) {
case FIONBIO:
/* All handled in the upper FS layer. */
break;
case FIONREAD:
/* Get the number of bytes that can be read. */
if (file->rtrack) {
*(int *)addr = audio_track_readablebytes(file->rtrack);
} else {
*(int *)addr = 0;
}
break;
case FIOASYNC:
/* Set/Clear ASYNC I/O. */
if (*(int *)addr) {
file->async_audio = curproc->p_pid;
TRACEF(2, file, "FIOASYNC pid %d", file->async_audio);
} else {
file->async_audio = 0;
TRACEF(2, file, "FIOASYNC off");
}
break;
case AUDIO_FLUSH:
/* XXX TODO: clear errors and restart? */
audio_file_clear(sc, file);
break;
case AUDIO_RERROR:
/*
* Number of read bytes dropped. We don't know where
* or when they were dropped (including conversion stage).
* Therefore, the number of accurate bytes or samples is
* also unknown.
*/
track = file->rtrack;
if (track) {
*(int *)addr = frametobyte(&track->usrbuf.fmt,
track->dropframes);
}
break;
case AUDIO_PERROR:
/*
* Number of write bytes dropped. We don't know where
* or when they were dropped (including conversion stage).
* Therefore, the number of accurate bytes or samples is
* also unknown.
*/
track = file->ptrack;
if (track) {
*(int *)addr = frametobyte(&track->usrbuf.fmt,
track->dropframes);
}
break;
case AUDIO_GETIOFFS:
/* XXX TODO */
ao = (struct audio_offset *)addr;
ao->samples = 0;
ao->deltablks = 0;
ao->offset = 0;
break;
case AUDIO_GETOOFFS:
ao = (struct audio_offset *)addr;
track = file->ptrack;
if (track == NULL) {
ao->samples = 0;
ao->deltablks = 0;
ao->offset = 0;
break;
}
mutex_enter(sc->sc_lock);
mutex_enter(sc->sc_intr_lock);
/* figure out where next DMA will start */
stamp = track->usrbuf_stamp;
offs = track->usrbuf.head;
mutex_exit(sc->sc_intr_lock);
mutex_exit(sc->sc_lock);
ao->samples = stamp;
ao->deltablks = (stamp / track->usrbuf_blksize) -
(track->usrbuf_stamp_last / track->usrbuf_blksize);
track->usrbuf_stamp_last = stamp;
offs = rounddown(offs, track->usrbuf_blksize)
+ track->usrbuf_blksize;
if (offs >= track->usrbuf.capacity)
offs -= track->usrbuf.capacity;
ao->offset = offs;
TRACET(3, track, "GETOOFFS: samples=%u deltablks=%u offset=%u",
ao->samples, ao->deltablks, ao->offset);
break;
case AUDIO_WSEEK:
/* XXX return value does not include outbuf one. */
if (file->ptrack)
*(u_long *)addr = file->ptrack->usrbuf.used;
break;
case AUDIO_SETINFO:
error = audio_exlock_enter(sc);
if (error)
break;
error = audio_file_setinfo(sc, file, (struct audio_info *)addr);
if (error) {
audio_exlock_exit(sc);
break;
}
/* XXX TODO: update last_ai if /dev/sound ? */
if (ISDEVSOUND(dev))
error = audiogetinfo(sc, &sc->sc_ai, 0, file);
audio_exlock_exit(sc);
break;
case AUDIO_GETINFO:
error = audio_exlock_enter(sc);
if (error)
break;
error = audiogetinfo(sc, (struct audio_info *)addr, 1, file);
audio_exlock_exit(sc);
break;
case AUDIO_GETBUFINFO:
error = audio_exlock_enter(sc);
if (error)
break;
error = audiogetinfo(sc, (struct audio_info *)addr, 0, file);
audio_exlock_exit(sc);
break;
case AUDIO_DRAIN:
if (file->ptrack) {
mutex_enter(sc->sc_lock);
error = audio_track_drain(sc, file->ptrack);
mutex_exit(sc->sc_lock);
}
break;
case AUDIO_GETDEV:
mutex_enter(sc->sc_lock);
error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr);
mutex_exit(sc->sc_lock);
break;
case AUDIO_GETENC:
ae = (audio_encoding_t *)addr;
index = ae->index;
if (index < 0 || index >= __arraycount(audio_encodings)) {
error = EINVAL;
break;
}
*ae = audio_encodings[index];
ae->index = index;
/*
* EMULATED always.
* EMULATED flag at that time used to mean that it could
* not be passed directly to the hardware as-is. But
* currently, all formats including hardware native is not
* passed directly to the hardware. So I set EMULATED
* flag for all formats.
*/
ae->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case AUDIO_GETFD:
/*
* Returns the current setting of full duplex mode.
* If HW has full duplex mode and there are two mixers,
* it is full duplex. Otherwise half duplex.
*/
error = audio_exlock_enter(sc);
if (error)
break;
fd = (sc->sc_props & AUDIO_PROP_FULLDUPLEX)
&& (sc->sc_pmixer && sc->sc_rmixer);
audio_exlock_exit(sc);
*(int *)addr = fd;
break;
case AUDIO_GETPROPS:
*(int *)addr = sc->sc_props;
break;
case AUDIO_QUERYFORMAT:
query = (audio_format_query_t *)addr;
mutex_enter(sc->sc_lock);
error = sc->hw_if->query_format(sc->hw_hdl, query);
mutex_exit(sc->sc_lock);
/* Hide internal information */
query->fmt.driver_data = NULL;
break;
case AUDIO_GETFORMAT:
error = audio_exlock_enter(sc);
if (error)
break;
audio_mixers_get_format(sc, (struct audio_info *)addr);
audio_exlock_exit(sc);
break;
case AUDIO_SETFORMAT:
error = audio_exlock_enter(sc);
audio_mixers_get_format(sc, &ai);
error = audio_mixers_set_format(sc, (struct audio_info *)addr);
if (error) {
/* Rollback */
audio_mixers_set_format(sc, &ai);
}
audio_exlock_exit(sc);
break;
case AUDIO_SETFD:
case AUDIO_SETCHAN:
case AUDIO_GETCHAN:
/* Obsoleted */
break;
default:
if (sc->hw_if->dev_ioctl) {
mutex_enter(sc->sc_lock);
error = sc->hw_if->dev_ioctl(sc->hw_hdl,
cmd, addr, flag, l);
mutex_exit(sc->sc_lock);
} else {
TRACEF(2, file, "unknown ioctl");
error = EINVAL;
}
break;
}
TRACEF(2, file, "(%lu,'%c',%lu)%s result %d",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, ioctlname,
error);
return error;
}
/*
* Returns the number of bytes that can be read on recording buffer.
*/
static __inline int
audio_track_readablebytes(const audio_track_t *track)
{
int bytes;
KASSERT(track);
KASSERT(track->mode == AUMODE_RECORD);
/*
* Although usrbuf is primarily readable data, recorded data
* also stays in track->input until reading. So it is necessary
* to add it. track->input is in frame, usrbuf is in byte.
*/
bytes = track->usrbuf.used +
track->input->used * frametobyte(&track->usrbuf.fmt, 1);
return bytes;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_poll(struct audio_softc *sc, int events, struct lwp *l,
audio_file_t *file)
{
audio_track_t *track;
int revents;
bool in_is_valid;
bool out_is_valid;
#if defined(AUDIO_DEBUG)
#define POLLEV_BITMAP "\177\020" \
"b\10WRBAND\0" \
"b\7RDBAND\0" "b\6RDNORM\0" "b\5NVAL\0" "b\4HUP\0" \
"b\3ERR\0" "b\2OUT\0" "b\1PRI\0" "b\0IN\0"
char evbuf[64];
snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, events);
TRACEF(2, file, "pid=%d.%d events=%s",
(int)curproc->p_pid, (int)l->l_lid, evbuf);
#endif
revents = 0;
in_is_valid = false;
out_is_valid = false;
if (events & (POLLIN | POLLRDNORM)) {
track = file->rtrack;
if (track) {
int used;
in_is_valid = true;
used = audio_track_readablebytes(track);
if (used > 0)
revents |= events & (POLLIN | POLLRDNORM);
}
}
if (events & (POLLOUT | POLLWRNORM)) {
track = file->ptrack;
if (track) {
out_is_valid = true;
if (track->usrbuf.used <= track->usrbuf_usedlow)
revents |= events & (POLLOUT | POLLWRNORM);
}
}
if (revents == 0) {
mutex_enter(sc->sc_lock);
if (in_is_valid) {
TRACEF(3, file, "selrecord rsel");
selrecord(l, &sc->sc_rsel);
}
if (out_is_valid) {
TRACEF(3, file, "selrecord wsel");
selrecord(l, &sc->sc_wsel);
}
mutex_exit(sc->sc_lock);
}
#if defined(AUDIO_DEBUG)
snprintb(evbuf, sizeof(evbuf), POLLEV_BITMAP, revents);
TRACEF(2, file, "revents=%s", evbuf);
#endif
return revents;
}
static const struct filterops audioread_filtops = {
.f_isfd = 1,
.f_attach = NULL,
.f_detach = filt_audioread_detach,
.f_event = filt_audioread_event,
};
static void
filt_audioread_detach(struct knote *kn)
{
struct audio_softc *sc;
audio_file_t *file;
file = kn->kn_hook;
sc = file->sc;
TRACEF(3, file, "called");
mutex_enter(sc->sc_lock);
selremove_knote(&sc->sc_rsel, kn);
mutex_exit(sc->sc_lock);
}
static int
filt_audioread_event(struct knote *kn, long hint)
{
audio_file_t *file;
audio_track_t *track;
file = kn->kn_hook;
track = file->rtrack;
/*
* kn_data must contain the number of bytes can be read.
* The return value indicates whether the event occurs or not.
*/
if (track == NULL) {
/* can not read with this descriptor. */
kn->kn_data = 0;
return 0;
}
kn->kn_data = audio_track_readablebytes(track);
TRACEF(3, file, "data=%" PRId64, kn->kn_data);
return kn->kn_data > 0;
}
static const struct filterops audiowrite_filtops = {
.f_isfd = 1,
.f_attach = NULL,
.f_detach = filt_audiowrite_detach,
.f_event = filt_audiowrite_event,
};
static void
filt_audiowrite_detach(struct knote *kn)
{
struct audio_softc *sc;
audio_file_t *file;
file = kn->kn_hook;
sc = file->sc;
TRACEF(3, file, "called");
mutex_enter(sc->sc_lock);
selremove_knote(&sc->sc_wsel, kn);
mutex_exit(sc->sc_lock);
}
static int
filt_audiowrite_event(struct knote *kn, long hint)
{
audio_file_t *file;
audio_track_t *track;
file = kn->kn_hook;
track = file->ptrack;
/*
* kn_data must contain the number of bytes can be write.
* The return value indicates whether the event occurs or not.
*/
if (track == NULL) {
/* can not write with this descriptor. */
kn->kn_data = 0;
return 0;
}
kn->kn_data = track->usrbuf_usedhigh - track->usrbuf.used;
TRACEF(3, file, "data=%" PRId64, kn->kn_data);
return (track->usrbuf.used < track->usrbuf_usedlow);
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
audio_kqfilter(struct audio_softc *sc, audio_file_t *file, struct knote *kn)
{
struct selinfo *sip;
TRACEF(3, file, "kn=%p kn_filter=%x", kn, (int)kn->kn_filter);
switch (kn->kn_filter) {
case EVFILT_READ:
sip = &sc->sc_rsel;
kn->kn_fop = &audioread_filtops;
break;
case EVFILT_WRITE:
sip = &sc->sc_wsel;
kn->kn_fop = &audiowrite_filtops;
break;
default:
return EINVAL;
}
kn->kn_hook = file;
mutex_enter(sc->sc_lock);
selrecord_knote(sip, kn);
mutex_exit(sc->sc_lock);
return 0;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
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,
audio_file_t *file)
{
audio_track_t *track;
vsize_t vsize;
int error;
TRACEF(2, file, "off=%lld, prot=%d", (long long)(*offp), prot);
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)
track = file->ptrack;
else if (prot == VM_PROT_READ)
track = file->rtrack;
else
return EINVAL;
#else
track = file->ptrack;
#endif
if (track == NULL)
return EACCES;
vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE);
if (len > vsize)
return EOVERFLOW;
if (*offp > (uint)(vsize - len))
return EOVERFLOW;
/* XXX TODO: what happens when mmap twice. */
if (!track->mmapped) {
track->mmapped = true;
if (!track->is_pause) {
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
if (sc->sc_pbusy == false)
audio_pmixer_start(sc, true);
audio_exlock_mutex_exit(sc);
}
/* XXX mmapping record buffer is not supported */
}
/* get ringbuffer */
*uobjp = track->uobj;
/* Acquire a reference for the mmap. munmap will release. */
uao_reference(*uobjp);
*maxprotp = prot;
*advicep = UVM_ADV_RANDOM;
*flagsp = MAP_SHARED;
return 0;
}
/*
* /dev/audioctl has to be able to open at any time without interference
* with any /dev/audio or /dev/sound.
* Must be called with sc_exlock held and without sc_lock held.
*/
static int
audioctl_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l)
{
struct file *fp;
audio_file_t *af;
int fd;
int error;
KASSERT(sc->sc_exlock);
TRACE(1, "called");
error = fd_allocfile(&fp, &fd);
if (error)
return error;
af = kmem_zalloc(sizeof(audio_file_t), KM_SLEEP);
af->sc = sc;
af->dev = dev;
/* Not necessary to insert sc_files. */
error = fd_clone(fp, fd, flags, &audio_fileops, af);
KASSERTMSG(error == EMOVEFD, "error=%d", error);
return error;
}
/*
* Free 'mem' if available, and initialize the pointer.
* For this reason, this is implemented as macro.
*/
#define audio_free(mem) do { \
if (mem != NULL) { \
kern_free(mem); \
mem = NULL; \
} \
} while (0)
/*
* (Re)allocate 'memblock' with specified 'bytes'.
* bytes must not be 0.
* This function never returns NULL.
*/
static void *
audio_realloc(void *memblock, size_t bytes)
{
KASSERT(bytes != 0);
audio_free(memblock);
return kern_malloc(bytes, M_WAITOK);
}
/*
* (Re)allocate usrbuf with 'newbufsize' bytes.
* Use this function for usrbuf because only usrbuf can be mmapped.
* If successful, it updates track->usrbuf.mem, track->usrbuf.capacity and
* returns 0. Otherwise, it clears track->usrbuf.mem, track->usrbuf.capacity
* and returns errno.
* It must be called before updating usrbuf.capacity.
*/
static int
audio_realloc_usrbuf(audio_track_t *track, int newbufsize)
{
struct audio_softc *sc;
vaddr_t vstart;
vsize_t oldvsize;
vsize_t newvsize;
int error;
KASSERT(newbufsize > 0);
sc = track->mixer->sc;
/* Get a nonzero multiple of PAGE_SIZE */
newvsize = roundup2(MAX(newbufsize, PAGE_SIZE), PAGE_SIZE);
if (track->usrbuf.mem != NULL) {
oldvsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE),
PAGE_SIZE);
if (oldvsize == newvsize) {
track->usrbuf.capacity = newbufsize;
return 0;
}
vstart = (vaddr_t)track->usrbuf.mem;
uvm_unmap(kernel_map, vstart, vstart + oldvsize);
/* uvm_unmap also detach uobj */
track->uobj = NULL; /* paranoia */
track->usrbuf.mem = NULL;
}
/* Create a uvm anonymous object */
track->uobj = uao_create(newvsize, 0);
/* Map it into the kernel virtual address space */
vstart = 0;
error = uvm_map(kernel_map, &vstart, newvsize, track->uobj, 0, 0,
UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE,
UVM_ADV_RANDOM, 0));
if (error) {
device_printf(sc->sc_dev, "uvm_map failed: errno=%d\n", error);
uao_detach(track->uobj); /* release reference */
goto abort;
}
error = uvm_map_pageable(kernel_map, vstart, vstart + newvsize,
false, 0);
if (error) {
device_printf(sc->sc_dev, "uvm_map_pageable failed: errno=%d\n",
error);
uvm_unmap(kernel_map, vstart, vstart + newvsize);
/* uvm_unmap also detach uobj */
goto abort;
}
track->usrbuf.mem = (void *)vstart;
track->usrbuf.capacity = newbufsize;
memset(track->usrbuf.mem, 0, newvsize);
return 0;
/* failure */
abort:
track->uobj = NULL; /* paranoia */
track->usrbuf.mem = NULL;
track->usrbuf.capacity = 0;
return error;
}
/*
* Free usrbuf (if available).
*/
static void
audio_free_usrbuf(audio_track_t *track)
{
vaddr_t vstart;
vsize_t vsize;
vstart = (vaddr_t)track->usrbuf.mem;
vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE);
if (track->usrbuf.mem != NULL) {
/*
* 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);
track->uobj = NULL;
track->usrbuf.mem = NULL;
track->usrbuf.capacity = 0;
}
}
/*
* This filter changes the volume for each channel.
* arg->context points track->ch_volume[].
*/
static void
audio_track_chvol(audio_filter_arg_t *arg)
{
int16_t *ch_volume;
const aint_t *s;
aint_t *d;
u_int i;
u_int ch;
u_int channels;
DIAGNOSTIC_filter_arg(arg);
KASSERTMSG(arg->srcfmt->channels == arg->dstfmt->channels,
"arg->srcfmt->channels=%d, arg->dstfmt->channels=%d",
arg->srcfmt->channels, arg->dstfmt->channels);
KASSERT(arg->context != NULL);
KASSERTMSG(arg->srcfmt->channels <= AUDIO_MAX_CHANNELS,
"arg->srcfmt->channels=%d", arg->srcfmt->channels);
s = arg->src;
d = arg->dst;
ch_volume = arg->context;
channels = arg->srcfmt->channels;
for (i = 0; i < arg->count; i++) {
for (ch = 0; ch < channels; ch++) {
aint2_t val;
val = *s++;
val = AUDIO_SCALEDOWN(val * ch_volume[ch], 8);
*d++ = (aint_t)val;
}
}
}
/*
* This filter performs conversion from stereo (or more channels) to mono.
*/
static void
audio_track_chmix_mixLR(audio_filter_arg_t *arg)
{
const aint_t *s;
aint_t *d;
u_int i;
DIAGNOSTIC_filter_arg(arg);
s = arg->src;
d = arg->dst;
for (i = 0; i < arg->count; i++) {
*d++ = AUDIO_SCALEDOWN(s[0], 1) + AUDIO_SCALEDOWN(s[1], 1);
s += arg->srcfmt->channels;
}
}
/*
* This filter performs conversion from mono to stereo (or more channels).
*/
static void
audio_track_chmix_dupLR(audio_filter_arg_t *arg)
{
const aint_t *s;
aint_t *d;
u_int i;
u_int ch;
u_int dstchannels;
DIAGNOSTIC_filter_arg(arg);
s = arg->src;
d = arg->dst;
dstchannels = arg->dstfmt->channels;
for (i = 0; i < arg->count; i++) {
d[0] = s[0];
d[1] = s[0];
s++;
d += dstchannels;
}
if (dstchannels > 2) {
d = arg->dst;
for (i = 0; i < arg->count; i++) {
for (ch = 2; ch < dstchannels; ch++) {
d[ch] = 0;
}
d += dstchannels;
}
}
}
/*
* This filter shrinks M channels into N channels.
* Extra channels are discarded.
*/
static void
audio_track_chmix_shrink(audio_filter_arg_t *arg)
{
const aint_t *s;
aint_t *d;
u_int i;
u_int ch;
DIAGNOSTIC_filter_arg(arg);
s = arg->src;
d = arg->dst;
for (i = 0; i < arg->count; i++) {
for (ch = 0; ch < arg->dstfmt->channels; ch++) {
*d++ = s[ch];
}
s += arg->srcfmt->channels;
}
}
/*
* This filter expands M channels into N channels.
* Silence is inserted for missing channels.
*/
static void
audio_track_chmix_expand(audio_filter_arg_t *arg)
{
const aint_t *s;
aint_t *d;
u_int i;
u_int ch;
u_int srcchannels;
u_int dstchannels;
DIAGNOSTIC_filter_arg(arg);
s = arg->src;
d = arg->dst;
srcchannels = arg->srcfmt->channels;
dstchannels = arg->dstfmt->channels;
for (i = 0; i < arg->count; i++) {
for (ch = 0; ch < srcchannels; ch++) {
*d++ = *s++;
}
for (; ch < dstchannels; ch++) {
*d++ = 0;
}
}
}
/*
* This filter performs frequency conversion (up sampling).
* It uses linear interpolation.
*/
static void
audio_track_freq_up(audio_filter_arg_t *arg)
{
audio_track_t *track;
audio_ring_t *src;
audio_ring_t *dst;
const aint_t *s;
aint_t *d;
aint_t prev[AUDIO_MAX_CHANNELS];
aint_t curr[AUDIO_MAX_CHANNELS];
aint_t grad[AUDIO_MAX_CHANNELS];
u_int i;
u_int t;
u_int step;
u_int channels;
u_int ch;
int srcused;
track = arg->context;
KASSERT(track);
src = &track->freq.srcbuf;
dst = track->freq.dst;
DIAGNOSTIC_ring(dst);
DIAGNOSTIC_ring(src);
KASSERT(src->used > 0);
KASSERTMSG(src->fmt.channels == dst->fmt.channels,
"src->fmt.channels=%d dst->fmt.channels=%d",
src->fmt.channels, dst->fmt.channels);
KASSERTMSG(src->head % track->mixer->frames_per_block == 0,
"src->head=%d track->mixer->frames_per_block=%d",
src->head, track->mixer->frames_per_block);
s = arg->src;
d = arg->dst;
/*
* In order to faciliate interpolation for each block, slide (delay)
* input by one sample. As a result, strictly speaking, the output
* phase is delayed by 1/dstfreq. However, I believe there is no
* observable impact.
*
* Example)
* srcfreq:dstfreq = 1:3
*
* A - -
* |
* |
* | B - -
* +-----+-----> input timeframe
* 0 1
*
* 0 1
* +-----+-----> input timeframe
* | A
* | x x
* | x x
* x (B)
* +-+-+-+-+-+-> output timeframe
* 0 1 2 3 4 5
*/
/* Last samples in previous block */
channels = src->fmt.channels;
for (ch = 0; ch < channels; ch++) {
prev[ch] = track->freq_prev[ch];
curr[ch] = track->freq_curr[ch];
grad[ch] = curr[ch] - prev[ch];
}
step = track->freq_step;
t = track->freq_current;
//#define FREQ_DEBUG
#if defined(FREQ_DEBUG)
#define PRINTF(fmt...) printf(fmt)
#else
#define PRINTF(fmt...) do { } while (0)
#endif
srcused = src->used;
PRINTF("upstart step=%d leap=%d", step, track->freq_leap);
PRINTF(" srcused=%d arg->count=%u", src->used, arg->count);
PRINTF(" prev=%d curr=%d grad=%d", prev[0], curr[0], grad[0]);
PRINTF(" t=%d\n", t);
for (i = 0; i < arg->count; i++) {
PRINTF("i=%d t=%5d", i, t);
if (t >= 65536) {
for (ch = 0; ch < channels; ch++) {
prev[ch] = curr[ch];
curr[ch] = *s++;
grad[ch] = curr[ch] - prev[ch];
}
PRINTF(" prev=%d s[%d]=%d",
prev[0], src->used - srcused, curr[0]);
/* Update */
t -= 65536;
srcused--;
if (srcused < 0) {
PRINTF(" break\n");
break;
}
}
for (ch = 0; ch < channels; ch++) {
*d++ = prev[ch] + (aint2_t)grad[ch] * t / 65536;
#if defined(FREQ_DEBUG)
if (ch == 0)
printf(" t=%5d *d=%d", t, d[-1]);
#endif
}
t += step;
PRINTF("\n");
}
PRINTF("end prev=%d curr=%d\n", prev[0], curr[0]);
auring_take(src, src->used);
auring_push(dst, i);
/* Adjust */
t += track->freq_leap;
track->freq_current = t;
for (ch = 0; ch < channels; ch++) {
track->freq_prev[ch] = prev[ch];
track->freq_curr[ch] = curr[ch];
}
}
/*
* This filter performs frequency conversion (down sampling).
* It uses simple thinning.
*/
static void
audio_track_freq_down(audio_filter_arg_t *arg)
{
audio_track_t *track;
audio_ring_t *src;
audio_ring_t *dst;
const aint_t *s0;
aint_t *d;
u_int i;
u_int t;
u_int step;
u_int ch;
u_int channels;
track = arg->context;
KASSERT(track);
src = &track->freq.srcbuf;
dst = track->freq.dst;
DIAGNOSTIC_ring(dst);
DIAGNOSTIC_ring(src);
KASSERT(src->used > 0);
KASSERTMSG(src->fmt.channels == dst->fmt.channels,
"src->fmt.channels=%d dst->fmt.channels=%d",
src->fmt.channels, dst->fmt.channels);
KASSERTMSG(src->head % track->mixer->frames_per_block == 0,
"src->head=%d track->mixer->frames_per_block=%d",
src->head, track->mixer->frames_per_block);
s0 = arg->src;
d = arg->dst;
t = track->freq_current;
step = track->freq_step;
channels = dst->fmt.channels;
PRINTF("downstart step=%d leap=%d", step, track->freq_leap);
PRINTF(" srcused=%d arg->count=%u", src->used, arg->count);
PRINTF(" t=%d\n", t);
for (i = 0; i < arg->count && t / 65536 < src->used; i++) {
const aint_t *s;
PRINTF("i=%4d t=%10d", i, t);
s = s0 + (t / 65536) * channels;
PRINTF(" s=%5ld", (s - s0) / channels);
for (ch = 0; ch < channels; ch++) {
if (ch == 0) PRINTF(" *s=%d", s[ch]);
*d++ = s[ch];
}
PRINTF("\n");
t += step;
}
t += track->freq_leap;
PRINTF("end t=%d\n", t);
auring_take(src, src->used);
auring_push(dst, i);
track->freq_current = t % 65536;
}
/*
* Creates track and returns it.
* Must be called without sc_lock held.
*/
audio_track_t *
audio_track_create(struct audio_softc *sc, audio_trackmixer_t *mixer)
{
audio_track_t *track;
static int newid = 0;
track = kmem_zalloc(sizeof(*track), KM_SLEEP);
track->id = newid++;
track->mixer = mixer;
track->mode = mixer->mode;
/* Do TRACE after id is assigned. */
TRACET(3, track, "for %s",
mixer->mode == AUMODE_PLAY ? "playback" : "recording");
#if defined(AUDIO_SUPPORT_TRACK_VOLUME)
track->volume = 256;
#endif
for (int i = 0; i < AUDIO_MAX_CHANNELS; i++) {
track->ch_volume[i] = 256;
}
return track;
}
/*
* Release all resources of the track and track itself.
* track must not be NULL. Don't specify the track within the file
* structure linked from sc->sc_files.
*/
static void
audio_track_destroy(audio_track_t *track)
{
KASSERT(track);
audio_free_usrbuf(track);
audio_free(track->codec.srcbuf.mem);
audio_free(track->chvol.srcbuf.mem);
audio_free(track->chmix.srcbuf.mem);
audio_free(track->freq.srcbuf.mem);
audio_free(track->outbuf.mem);
kmem_free(track, sizeof(*track));
}
/*
* It returns encoding conversion filter according to src and dst format.
* If it is not a convertible pair, it returns NULL. Either src or dst
* must be internal format.
*/
static audio_filter_t
audio_track_get_codec(audio_track_t *track, const audio_format2_t *src,
const audio_format2_t *dst)
{
if (audio_format2_is_internal(src)) {
if (dst->encoding == AUDIO_ENCODING_ULAW) {
return audio_internal_to_mulaw;
} else if (dst->encoding == AUDIO_ENCODING_ALAW) {
return audio_internal_to_alaw;
} else if (audio_format2_is_linear(dst)) {
switch (dst->stride) {
case 8:
return audio_internal_to_linear8;
case 16:
return audio_internal_to_linear16;
#if defined(AUDIO_SUPPORT_LINEAR24)
case 24:
return audio_internal_to_linear24;
#endif
case 32:
return audio_internal_to_linear32;
default:
TRACET(1, track, "unsupported %s stride %d",
"dst", dst->stride);
goto abort;
}
}
} else if (audio_format2_is_internal(dst)) {
if (src->encoding == AUDIO_ENCODING_ULAW) {
return audio_mulaw_to_internal;
} else if (src->encoding == AUDIO_ENCODING_ALAW) {
return audio_alaw_to_internal;
} else if (audio_format2_is_linear(src)) {
switch (src->stride) {
case 8:
return audio_linear8_to_internal;
case 16:
return audio_linear16_to_internal;
#if defined(AUDIO_SUPPORT_LINEAR24)
case 24:
return audio_linear24_to_internal;
#endif
case 32:
return audio_linear32_to_internal;
default:
TRACET(1, track, "unsupported %s stride %d",
"src", src->stride);
goto abort;
}
}
}
TRACET(1, track, "unsupported encoding");
abort:
#if defined(AUDIO_DEBUG)
if (audiodebug >= 2) {
char buf[100];
audio_format2_tostr(buf, sizeof(buf), src);
TRACET(2, track, "src %s", buf);
audio_format2_tostr(buf, sizeof(buf), dst);
TRACET(2, track, "dst %s", buf);
}
#endif
return NULL;
}
/*
* Initialize the codec stage of this track as necessary.
* If successful, it initializes the codec stage as necessary, stores updated
* last_dst in *last_dstp in any case, and returns 0.
* Otherwise, it returns errno without modifying *last_dstp.
*/
static int
audio_track_init_codec(audio_track_t *track, audio_ring_t **last_dstp)
{
audio_ring_t *last_dst;
audio_ring_t *srcbuf;
audio_format2_t *srcfmt;
audio_format2_t *dstfmt;
audio_filter_arg_t *arg;
u_int len;
int error;
KASSERT(track);
last_dst = *last_dstp;
dstfmt = &last_dst->fmt;
srcfmt = &track->inputfmt;
srcbuf = &track->codec.srcbuf;
error = 0;
if (srcfmt->encoding != dstfmt->encoding
|| srcfmt->precision != dstfmt->precision
|| srcfmt->stride != dstfmt->stride) {
track->codec.dst = last_dst;
srcbuf->fmt = *dstfmt;
srcbuf->fmt.encoding = srcfmt->encoding;
srcbuf->fmt.precision = srcfmt->precision;
srcbuf->fmt.stride = srcfmt->stride;
track->codec.filter = audio_track_get_codec(track,
&srcbuf->fmt, dstfmt);
if (track->codec.filter == NULL) {
error = EINVAL;
goto abort;
}
srcbuf->head = 0;
srcbuf->used = 0;
srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt);
len = auring_bytelen(srcbuf);
srcbuf->mem = audio_realloc(srcbuf->mem, len);
arg = &track->codec.arg;
arg->srcfmt = &srcbuf->fmt;
arg->dstfmt = dstfmt;
arg->context = NULL;
*last_dstp = srcbuf;
return 0;
}
abort:
track->codec.filter = NULL;
audio_free(srcbuf->mem);
return error;
}
/*
* Initialize the chvol stage of this track as necessary.
* If successful, it initializes the chvol stage as necessary, stores updated
* last_dst in *last_dstp in any case, and returns 0.
* Otherwise, it returns errno without modifying *last_dstp.
*/
static int
audio_track_init_chvol(audio_track_t *track, audio_ring_t **last_dstp)
{
audio_ring_t *last_dst;
audio_ring_t *srcbuf;
audio_format2_t *srcfmt;
audio_format2_t *dstfmt;
audio_filter_arg_t *arg;
u_int len;
int error;
KASSERT(track);
last_dst = *last_dstp;
dstfmt = &last_dst->fmt;
srcfmt = &track->inputfmt;
srcbuf = &track->chvol.srcbuf;
error = 0;
/* Check whether channel volume conversion is necessary. */
bool use_chvol = false;
for (int ch = 0; ch < srcfmt->channels; ch++) {
if (track->ch_volume[ch] != 256) {
use_chvol = true;
break;
}
}
if (use_chvol == true) {
track->chvol.dst = last_dst;
track->chvol.filter = audio_track_chvol;
srcbuf->fmt = *dstfmt;
/* no format conversion occurs */
srcbuf->head = 0;
srcbuf->used = 0;
srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt);
len = auring_bytelen(srcbuf);
srcbuf->mem = audio_realloc(srcbuf->mem, len);
arg = &track->chvol.arg;
arg->srcfmt = &srcbuf->fmt;
arg->dstfmt = dstfmt;
arg->context = track->ch_volume;
*last_dstp = srcbuf;
return 0;
}
track->chvol.filter = NULL;
audio_free(srcbuf->mem);
return error;
}
/*
* Initialize the chmix stage of this track as necessary.
* If successful, it initializes the chmix stage as necessary, stores updated
* last_dst in *last_dstp in any case, and returns 0.
* Otherwise, it returns errno without modifying *last_dstp.
*/
static int
audio_track_init_chmix(audio_track_t *track, audio_ring_t **last_dstp)
{
audio_ring_t *last_dst;
audio_ring_t *srcbuf;
audio_format2_t *srcfmt;
audio_format2_t *dstfmt;
audio_filter_arg_t *arg;
u_int srcch;
u_int dstch;
u_int len;
int error;
KASSERT(track);
last_dst = *last_dstp;
dstfmt = &last_dst->fmt;
srcfmt = &track->inputfmt;
srcbuf = &track->chmix.srcbuf;
error = 0;
srcch = srcfmt->channels;
dstch = dstfmt->channels;
if (srcch != dstch) {
track->chmix.dst = last_dst;
if (srcch >= 2 && dstch == 1) {
track->chmix.filter = audio_track_chmix_mixLR;
} else if (srcch == 1 && dstch >= 2) {
track->chmix.filter = audio_track_chmix_dupLR;
} else if (srcch > dstch) {
track->chmix.filter = audio_track_chmix_shrink;
} else {
track->chmix.filter = audio_track_chmix_expand;
}
srcbuf->fmt = *dstfmt;
srcbuf->fmt.channels = srcch;
srcbuf->head = 0;
srcbuf->used = 0;
/* XXX The buffer size should be able to calculate. */
srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt);
len = auring_bytelen(srcbuf);
srcbuf->mem = audio_realloc(srcbuf->mem, len);
arg = &track->chmix.arg;
arg->srcfmt = &srcbuf->fmt;
arg->dstfmt = dstfmt;
arg->context = NULL;
*last_dstp = srcbuf;
return 0;
}
track->chmix.filter = NULL;
audio_free(srcbuf->mem);
return error;
}
/*
* Initialize the freq stage of this track as necessary.
* If successful, it initializes the freq stage as necessary, stores updated
* last_dst in *last_dstp in any case, and returns 0.
* Otherwise, it returns errno without modifying *last_dstp.
*/
static int
audio_track_init_freq(audio_track_t *track, audio_ring_t **last_dstp)
{
audio_ring_t *last_dst;
audio_ring_t *srcbuf;
audio_format2_t *srcfmt;
audio_format2_t *dstfmt;
audio_filter_arg_t *arg;
uint32_t srcfreq;
uint32_t dstfreq;
u_int dst_capacity;
u_int mod;
u_int len;
int error;
KASSERT(track);
last_dst = *last_dstp;
dstfmt = &last_dst->fmt;
srcfmt = &track->inputfmt;
srcbuf = &track->freq.srcbuf;
error = 0;
srcfreq = srcfmt->sample_rate;
dstfreq = dstfmt->sample_rate;
if (srcfreq != dstfreq) {
track->freq.dst = last_dst;
memset(track->freq_prev, 0, sizeof(track->freq_prev));
memset(track->freq_curr, 0, sizeof(track->freq_curr));
/* freq_step is the ratio of src/dst when let dst 65536. */
track->freq_step = (uint64_t)srcfreq * 65536 / dstfreq;
dst_capacity = frame_per_block(track->mixer, dstfmt);
mod = (uint64_t)srcfreq * 65536 % dstfreq;
track->freq_leap = (mod * dst_capacity + dstfreq / 2) / dstfreq;
if (track->freq_step < 65536) {
track->freq.filter = audio_track_freq_up;
/* In order to carry at the first time. */
track->freq_current = 65536;
} else {
track->freq.filter = audio_track_freq_down;
track->freq_current = 0;
}
srcbuf->fmt = *dstfmt;
srcbuf->fmt.sample_rate = srcfreq;
srcbuf->head = 0;
srcbuf->used = 0;
srcbuf->capacity = frame_per_block(track->mixer, &srcbuf->fmt);
len = auring_bytelen(srcbuf);
srcbuf->mem = audio_realloc(srcbuf->mem, len);
arg = &track->freq.arg;
arg->srcfmt = &srcbuf->fmt;
arg->dstfmt = dstfmt;/*&last_dst->fmt;*/
arg->context = track;
*last_dstp = srcbuf;
return 0;
}
track->freq.filter = NULL;
audio_free(srcbuf->mem);
return error;
}
/*
* When playing back: (e.g. if codec and freq stage are valid)
*
* write
* | uiomove
* v
* usrbuf [...............] byte ring buffer (mmap-able)
* | memcpy
* v
* codec.srcbuf[....] 1 block (ring) buffer <-- stage input
* .dst ----+
* | convert
* v
* freq.srcbuf [....] 1 block (ring) buffer
* .dst ----+
* | convert
* v
* outbuf [...............] NBLKOUT blocks ring buffer
*
*
* When recording:
*
* freq.srcbuf [...............] NBLKOUT blocks ring buffer <-- stage input
* .dst ----+
* | convert
* v
* codec.srcbuf[.....] 1 block (ring) buffer
* .dst ----+
* | convert
* v
* outbuf [.....] 1 block (ring) buffer
* | memcpy
* v
* usrbuf [...............] byte ring buffer (mmap-able *)
* | uiomove
* v
* read
*
* *: usrbuf for recording is also mmap-able due to symmetry with
* playback buffer, but for now mmap will never happen for recording.
*/
/*
* Set the userland format of this track.
* usrfmt argument should have been previously verified by
* audio_track_setinfo_check().
* This function may release and reallocate all internal conversion buffers.
* It returns 0 if successful. Otherwise it returns errno with clearing all
* internal buffers.
* It must be called without sc_intr_lock since uvm_* routines require non
* intr_lock state.
* It must be called with track lock held since it may release and reallocate
* outbuf.
*/
static int
audio_track_set_format(audio_track_t *track, audio_format2_t *usrfmt)
{
struct audio_softc *sc;
u_int newbufsize;
u_int oldblksize;
u_int len;
int error;
KASSERT(track);
sc = track->mixer->sc;
/* usrbuf is the closest buffer to the userland. */
track->usrbuf.fmt = *usrfmt;
/*
* For references, one block size (in 40msec) is:
* 320 bytes = 204 blocks/64KB for mulaw/8kHz/1ch
* 7680 bytes = 8 blocks/64KB for s16/48kHz/2ch
* 30720 bytes = 90 KB/3blocks for s16/48kHz/8ch
* 61440 bytes = 180 KB/3blocks for s16/96kHz/8ch
* 245760 bytes = 720 KB/3blocks for s32/192kHz/8ch
*
* For example,
* 1) If usrbuf_blksize = 7056 (s16/44.1k/2ch) and PAGE_SIZE = 8192,
* newbufsize = rounddown(65536 / 7056) = 63504
* newvsize = roundup2(63504, PAGE_SIZE) = 65536
* Therefore it maps 8 * 8K pages and usrbuf->capacity = 63504.
*
* 2) If usrbuf_blksize = 7680 (s16/48k/2ch) and PAGE_SIZE = 4096,
* newbufsize = rounddown(65536 / 7680) = 61440
* newvsize = roundup2(61440, PAGE_SIZE) = 61440 (= 15 pages)
* Therefore it maps 15 * 4K pages and usrbuf->capacity = 61440.
*/
oldblksize = track->usrbuf_blksize;
track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt,
frame_per_block(track->mixer, &track->usrbuf.fmt));
track->usrbuf.head = 0;
track->usrbuf.used = 0;
newbufsize = MAX(track->usrbuf_blksize * AUMINNOBLK, 65536);
newbufsize = rounddown(newbufsize, track->usrbuf_blksize);
error = audio_realloc_usrbuf(track, newbufsize);
if (error) {
device_printf(sc->sc_dev, "malloc usrbuf(%d) failed\n",
newbufsize);
goto error;
}
/* Recalc water mark. */
if (track->usrbuf_blksize != oldblksize) {
if (audio_track_is_playback(track)) {
/* Set high at 100%, low at 75%. */
track->usrbuf_usedhigh = track->usrbuf.capacity;
track->usrbuf_usedlow = track->usrbuf.capacity * 3 / 4;
} else {
/* Set high at 100% minus 1block(?), low at 0% */
track->usrbuf_usedhigh = track->usrbuf.capacity -
track->usrbuf_blksize;
track->usrbuf_usedlow = 0;
}
}
/* Stage buffer */
audio_ring_t *last_dst = &track->outbuf;
if (audio_track_is_playback(track)) {
/* On playback, initialize from the mixer side in order. */
track->inputfmt = *usrfmt;
track->outbuf.fmt = track->mixer->track_fmt;
if ((error = audio_track_init_freq(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_chmix(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_chvol(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_codec(track, &last_dst)) != 0)
goto error;
} else {
/* On recording, initialize from userland side in order. */
track->inputfmt = track->mixer->track_fmt;
track->outbuf.fmt = *usrfmt;
if ((error = audio_track_init_codec(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_chvol(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_chmix(track, &last_dst)) != 0)
goto error;
if ((error = audio_track_init_freq(track, &last_dst)) != 0)
goto error;
}
#if 0
/* debug */
if (track->freq.filter) {
audio_print_format2("freq src", &track->freq.srcbuf.fmt);
audio_print_format2("freq dst", &track->freq.dst->fmt);
}
if (track->chmix.filter) {
audio_print_format2("chmix src", &track->chmix.srcbuf.fmt);
audio_print_format2("chmix dst", &track->chmix.dst->fmt);
}
if (track->chvol.filter) {
audio_print_format2("chvol src", &track->chvol.srcbuf.fmt);
audio_print_format2("chvol dst", &track->chvol.dst->fmt);
}
if (track->codec.filter) {
audio_print_format2("codec src", &track->codec.srcbuf.fmt);
audio_print_format2("codec dst", &track->codec.dst->fmt);
}
#endif
/* Stage input buffer */
track->input = last_dst;
/*
* On the recording track, make the first stage a ring buffer.
* XXX is there a better way?
*/
if (audio_track_is_record(track)) {
track->input->capacity = NBLKOUT *
frame_per_block(track->mixer, &track->input->fmt);
len = auring_bytelen(track->input);
track->input->mem = audio_realloc(track->input->mem, len);
}
/*
* Output buffer.
* On the playback track, its capacity is NBLKOUT blocks.
* On the recording track, its capacity is 1 block.
*/
track->outbuf.head = 0;
track->outbuf.used = 0;
track->outbuf.capacity = frame_per_block(track->mixer,
&track->outbuf.fmt);
if (audio_track_is_playback(track))
track->outbuf.capacity *= NBLKOUT;
len = auring_bytelen(&track->outbuf);
track->outbuf.mem = audio_realloc(track->outbuf.mem, len);
if (track->outbuf.mem == NULL) {
device_printf(sc->sc_dev, "malloc outbuf(%d) failed\n", len);
error = ENOMEM;
goto error;
}
#if defined(AUDIO_DEBUG)
if (audiodebug >= 3) {
struct audio_track_debugbuf m;
memset(&m, 0, sizeof(m));
snprintf(m.outbuf, sizeof(m.outbuf), " out=%d",
track->outbuf.capacity * frametobyte(&track->outbuf.fmt,1));
if (track->freq.filter)
snprintf(m.freq, sizeof(m.freq), " freq=%d",
track->freq.srcbuf.capacity *
frametobyte(&track->freq.srcbuf.fmt, 1));
if (track->chmix.filter)
snprintf(m.chmix, sizeof(m.chmix), " chmix=%d",
track->chmix.srcbuf.capacity *
frametobyte(&track->chmix.srcbuf.fmt, 1));
if (track->chvol.filter)
snprintf(m.chvol, sizeof(m.chvol), " chvol=%d",
track->chvol.srcbuf.capacity *
frametobyte(&track->chvol.srcbuf.fmt, 1));
if (track->codec.filter)
snprintf(m.codec, sizeof(m.codec), " codec=%d",
track->codec.srcbuf.capacity *
frametobyte(&track->codec.srcbuf.fmt, 1));
snprintf(m.usrbuf, sizeof(m.usrbuf),
" usr=%d", track->usrbuf.capacity);
if (audio_track_is_playback(track)) {
TRACET(0, track, "bufsize%s%s%s%s%s%s",
m.outbuf, m.freq, m.chmix,
m.chvol, m.codec, m.usrbuf);
} else {
TRACET(0, track, "bufsize%s%s%s%s%s%s",
m.freq, m.chmix, m.chvol,
m.codec, m.outbuf, m.usrbuf);
}
}
#endif
return 0;
error:
audio_free_usrbuf(track);
audio_free(track->codec.srcbuf.mem);
audio_free(track->chvol.srcbuf.mem);
audio_free(track->chmix.srcbuf.mem);
audio_free(track->freq.srcbuf.mem);
audio_free(track->outbuf.mem);
return error;
}
/*
* Fill silence frames (as the internal format) up to 1 block
* if the ring is not empty and less than 1 block.
* It returns the number of appended frames.
*/
static int
audio_append_silence(audio_track_t *track, audio_ring_t *ring)
{
int fpb;
int n;
KASSERT(track);
KASSERT(audio_format2_is_internal(&ring->fmt));
/* XXX is n correct? */
/* XXX memset uses frametobyte()? */
if (ring->used == 0)
return 0;
fpb = frame_per_block(track->mixer, &ring->fmt);
if (ring->used >= fpb)
return 0;
n = (ring->capacity - ring->used) % fpb;
KASSERTMSG(auring_get_contig_free(ring) >= n,
"auring_get_contig_free(ring)=%d n=%d",
auring_get_contig_free(ring), n);
memset(auring_tailptr_aint(ring), 0,
n * ring->fmt.channels * sizeof(aint_t));
auring_push(ring, n);
return n;
}
/*
* Execute the conversion stage.
* It prepares arg from this stage and executes stage->filter.
* It must be called only if stage->filter is not NULL.
*
* For stages other than frequency conversion, the function increments
* src and dst counters here. For frequency conversion stage, on the
* other hand, the function does not touch src and dst counters and
* filter side has to increment them.
*/
static void
audio_apply_stage(audio_track_t *track, audio_stage_t *stage, bool isfreq)
{
audio_filter_arg_t *arg;
int srccount;
int dstcount;
int count;
KASSERT(track);
KASSERT(stage->filter);
srccount = auring_get_contig_used(&stage->srcbuf);
dstcount = auring_get_contig_free(stage->dst);
if (isfreq) {
KASSERTMSG(srccount > 0, "freq but srccount=%d", srccount);
count = uimin(dstcount, track->mixer->frames_per_block);
} else {
count = uimin(srccount, dstcount);
}
if (count > 0) {
arg = &stage->arg;
arg->src = auring_headptr(&stage->srcbuf);
arg->dst = auring_tailptr(stage->dst);
arg->count = count;
stage->filter(arg);
if (!isfreq) {
auring_take(&stage->srcbuf, count);
auring_push(stage->dst, count);
}
}
}
/*
* Produce output buffer for playback from user input buffer.
* It must be called only if usrbuf is not empty and outbuf is
* available at least one free block.
*/
static void
audio_track_play(audio_track_t *track)
{
audio_ring_t *usrbuf;
audio_ring_t *input;
int count;
int framesize;
int bytes;
KASSERT(track);
KASSERT(track->lock);
TRACET(4, track, "start pstate=%d", track->pstate);
/* At this point usrbuf must not be empty. */
KASSERT(track->usrbuf.used > 0);
/* Also, outbuf must be available at least one block. */
count = auring_get_contig_free(&track->outbuf);
KASSERTMSG(count >= frame_per_block(track->mixer, &track->outbuf.fmt),
"count=%d fpb=%d",
count, frame_per_block(track->mixer, &track->outbuf.fmt));
/* XXX TODO: is this necessary for now? */
int track_count_0 = track->outbuf.used;
usrbuf = &track->usrbuf;
input = track->input;
/*
* framesize is always 1 byte or more since all formats supported as
* usrfmt(=input) have 8bit or more stride.
*/
framesize = frametobyte(&input->fmt, 1);
KASSERT(framesize >= 1);
/* The next stage of usrbuf (=input) must be available. */
KASSERT(auring_get_contig_free(input) > 0);
/*
* Copy usrbuf up to 1block to input buffer.
* count is the number of frames to copy from usrbuf.
* bytes is the number of bytes to copy from usrbuf. However it is
* not copied less than one frame.
*/
count = uimin(usrbuf->used, track->usrbuf_blksize) / framesize;
bytes = count * framesize;
track->usrbuf_stamp += bytes;
if (usrbuf->head + bytes < usrbuf->capacity) {
memcpy((uint8_t *)input->mem + auring_tail(input) * framesize,
(uint8_t *)usrbuf->mem + usrbuf->head,
bytes);
auring_push(input, count);
auring_take(usrbuf, bytes);
} else {
int bytes1;
int bytes2;
bytes1 = auring_get_contig_used(usrbuf);
KASSERTMSG(bytes1 % framesize == 0,
"bytes1=%d framesize=%d", bytes1, framesize);
memcpy((uint8_t *)input->mem + auring_tail(input) * framesize,
(uint8_t *)usrbuf->mem + usrbuf->head,
bytes1);
auring_push(input, bytes1 / framesize);
auring_take(usrbuf, bytes1);
bytes2 = bytes - bytes1;
memcpy((uint8_t *)input->mem + auring_tail(input) * framesize,
(uint8_t *)usrbuf->mem + usrbuf->head,
bytes2);
auring_push(input, bytes2 / framesize);
auring_take(usrbuf, bytes2);
}
/* Encoding conversion */
if (track->codec.filter)
audio_apply_stage(track, &track->codec, false);
/* Channel volume */
if (track->chvol.filter)
audio_apply_stage(track, &track->chvol, false);
/* Channel mix */
if (track->chmix.filter)
audio_apply_stage(track, &track->chmix, false);
/* Frequency conversion */
/*
* Since the frequency conversion needs correction for each block,
* it rounds up to 1 block.
*/
if (track->freq.filter) {
int n;
n = audio_append_silence(track, &track->freq.srcbuf);
if (n > 0) {
TRACET(4, track,
"freq.srcbuf add silence %d -> %d/%d/%d",
n,
track->freq.srcbuf.head,
track->freq.srcbuf.used,
track->freq.srcbuf.capacity);
}
if (track->freq.srcbuf.used > 0) {
audio_apply_stage(track, &track->freq, true);
}
}
if (bytes < track->usrbuf_blksize) {
/*
* Clear all conversion buffer pointer if the conversion was
* not exactly one block. These conversion stage buffers are
* certainly circular buffers because of symmetry with the
* previous and next stage buffer. However, since they are
* treated as simple contiguous buffers in operation, so head
* always should point 0. This may happen during drain-age.
*/
TRACET(4, track, "reset stage");
if (track->codec.filter) {
KASSERT(track->codec.srcbuf.used == 0);
track->codec.srcbuf.head = 0;
}
if (track->chvol.filter) {
KASSERT(track->chvol.srcbuf.used == 0);
track->chvol.srcbuf.head = 0;
}
if (track->chmix.filter) {
KASSERT(track->chmix.srcbuf.used == 0);
track->chmix.srcbuf.head = 0;
}
if (track->freq.filter) {
KASSERT(track->freq.srcbuf.used == 0);
track->freq.srcbuf.head = 0;
}
}
if (track->input == &track->outbuf) {
track->outputcounter = track->inputcounter;
} else {
track->outputcounter += track->outbuf.used - track_count_0;
}
#if defined(AUDIO_DEBUG)
if (audiodebug >= 3) {
struct audio_track_debugbuf m;
audio_track_bufstat(track, &m);
TRACET(0, track, "end%s%s%s%s%s%s",
m.outbuf, m.freq, m.chvol, m.chmix, m.codec, m.usrbuf);
}
#endif
}
/*
* Produce user output buffer for recording from input buffer.
*/
static void
audio_track_record(audio_track_t *track)
{
audio_ring_t *outbuf;
audio_ring_t *usrbuf;
int count;
int bytes;
int framesize;
KASSERT(track);
KASSERT(track->lock);
/* Number of frames to process */
count = auring_get_contig_used(track->input);
count = uimin(count, track->mixer->frames_per_block);
if (count == 0) {
TRACET(4, track, "count == 0");
return;
}
/* Frequency conversion */
if (track->freq.filter) {
if (track->freq.srcbuf.used > 0) {
audio_apply_stage(track, &track->freq, true);
/* XXX should input of freq be from beginning of buf? */
}
}
/* Channel mix */
if (track->chmix.filter)
audio_apply_stage(track, &track->chmix, false);
/* Channel volume */
if (track->chvol.filter)
audio_apply_stage(track, &track->chvol, false);
/* Encoding conversion */
if (track->codec.filter)
audio_apply_stage(track, &track->codec, false);
/* Copy outbuf to usrbuf */
outbuf = &track->outbuf;
usrbuf = &track->usrbuf;
/*
* framesize is always 1 byte or more since all formats supported
* as usrfmt(=output) have 8bit or more stride.
*/
framesize = frametobyte(&outbuf->fmt, 1);
KASSERT(framesize >= 1);
/*
* count is the number of frames to copy to usrbuf.
* bytes is the number of bytes to copy to usrbuf.
*/
count = outbuf->used;
count = uimin(count,
(track->usrbuf_usedhigh - usrbuf->used) / framesize);
bytes = count * framesize;
if (auring_tail(usrbuf) + bytes < usrbuf->capacity) {
memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf),
(uint8_t *)outbuf->mem + outbuf->head * framesize,
bytes);
auring_push(usrbuf, bytes);
auring_take(outbuf, count);
} else {
int bytes1;
int bytes2;
bytes1 = auring_get_contig_free(usrbuf);
KASSERTMSG(bytes1 % framesize == 0,
"bytes1=%d framesize=%d", bytes1, framesize);
memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf),
(uint8_t *)outbuf->mem + outbuf->head * framesize,
bytes1);
auring_push(usrbuf, bytes1);
auring_take(outbuf, bytes1 / framesize);
bytes2 = bytes - bytes1;
memcpy((uint8_t *)usrbuf->mem + auring_tail(usrbuf),
(uint8_t *)outbuf->mem + outbuf->head * framesize,
bytes2);
auring_push(usrbuf, bytes2);
auring_take(outbuf, bytes2 / framesize);
}
/* XXX TODO: any counters here? */
#if defined(AUDIO_DEBUG)
if (audiodebug >= 3) {
struct audio_track_debugbuf m;
audio_track_bufstat(track, &m);
TRACET(0, track, "end%s%s%s%s%s%s",
m.freq, m.chvol, m.chmix, m.codec, m.outbuf, m.usrbuf);
}
#endif
}
/*
* Calculate blktime [msec] from mixer(.hwbuf.fmt).
* Must be called with sc_exlock held.
*/
static u_int
audio_mixer_calc_blktime(struct audio_softc *sc, audio_trackmixer_t *mixer)
{
audio_format2_t *fmt;
u_int blktime;
u_int frames_per_block;
KASSERT(sc->sc_exlock);
fmt = &mixer->hwbuf.fmt;
blktime = sc->sc_blk_ms;
/*
* If stride is not multiples of 8, special treatment is necessary.
* For now, it is only x68k's vs(4), 4 bit/sample ADPCM.
*/
if (fmt->stride == 4) {
frames_per_block = fmt->sample_rate * blktime / 1000;
if ((frames_per_block & 1) != 0)
blktime *= 2;
}
#ifdef DIAGNOSTIC
else if (fmt->stride % NBBY != 0) {
panic("unsupported HW stride %d", fmt->stride);
}
#endif
return blktime;
}
/*
* Initialize the mixer corresponding to the mode.
* Set AUMODE_PLAY to the 'mode' for playback or AUMODE_RECORD for recording.
* sc->sc_[pr]mixer (corresponding to the 'mode') must be zero-filled.
* This function returns 0 on successful. Otherwise returns errno.
* Must be called with sc_exlock held and without sc_lock held.
*/
static int
audio_mixer_init(struct audio_softc *sc, int mode,
const audio_format2_t *hwfmt, const audio_filter_reg_t *reg)
{
char codecbuf[64];
char blkdmsbuf[8];
audio_trackmixer_t *mixer;
void (*softint_handler)(void *);
int len;
int blksize;
int capacity;
size_t bufsize;
int hwblks;
int blkms;
int blkdms;
int error;
KASSERT(hwfmt != NULL);
KASSERT(reg != NULL);
KASSERT(sc->sc_exlock);
error = 0;
if (mode == AUMODE_PLAY)
mixer = sc->sc_pmixer;
else
mixer = sc->sc_rmixer;
mixer->sc = sc;
mixer->mode = mode;
mixer->hwbuf.fmt = *hwfmt;
mixer->volume = 256;
mixer->blktime_d = 1000;
mixer->blktime_n = audio_mixer_calc_blktime(sc, mixer);
sc->sc_blk_ms = mixer->blktime_n;
hwblks = NBLKHW;
mixer->frames_per_block = frame_per_block(mixer, &mixer->hwbuf.fmt);
blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block);
if (sc->hw_if->round_blocksize) {
int rounded;
audio_params_t p = format2_to_params(&mixer->hwbuf.fmt);
mutex_enter(sc->sc_lock);
rounded = sc->hw_if->round_blocksize(sc->hw_hdl, blksize,
mode, &p);
mutex_exit(sc->sc_lock);
TRACE(1, "round_blocksize %d -> %d", blksize, rounded);
if (rounded != blksize) {
if ((rounded * NBBY) % (mixer->hwbuf.fmt.stride *
mixer->hwbuf.fmt.channels) != 0) {
audio_printf(sc,
"round_blocksize returned blocksize "
"indivisible by framesize: "
"blksize=%d rounded=%d "
"stride=%ubit channels=%u\n",
blksize, rounded,
mixer->hwbuf.fmt.stride,
mixer->hwbuf.fmt.channels);
return EINVAL;
}
/* Recalculation */
blksize = rounded;
mixer->frames_per_block = blksize * NBBY /
(mixer->hwbuf.fmt.stride *
mixer->hwbuf.fmt.channels);
}
}
mixer->blktime_n = mixer->frames_per_block;
mixer->blktime_d = mixer->hwbuf.fmt.sample_rate;
capacity = mixer->frames_per_block * hwblks;
bufsize = frametobyte(&mixer->hwbuf.fmt, capacity);
if (sc->hw_if->round_buffersize) {
size_t rounded;
mutex_enter(sc->sc_lock);
rounded = sc->hw_if->round_buffersize(sc->hw_hdl, mode,
bufsize);
mutex_exit(sc->sc_lock);
TRACE(1, "round_buffersize %zd -> %zd", bufsize, rounded);
if (rounded < bufsize) {
/* buffersize needs NBLKHW blocks at least. */
audio_printf(sc,
"round_buffersize returned too small buffersize: "
"buffersize=%zd blksize=%d\n",
rounded, blksize);
return EINVAL;
}
if (rounded % blksize != 0) {
/* buffersize/blksize constraint mismatch? */
audio_printf(sc,
"round_buffersize returned buffersize indivisible "
"by blksize: buffersize=%zu blksize=%d\n",
rounded, blksize);
return EINVAL;
}
if (rounded != bufsize) {
/* Recalculation */
bufsize = rounded;
hwblks = bufsize / blksize;
capacity = mixer->frames_per_block * hwblks;
}
}
TRACE(1, "buffersize for %s = %zu",
(mode == AUMODE_PLAY) ? "playback" : "recording",
bufsize);
mixer->hwbuf.capacity = capacity;
if (sc->hw_if->allocm) {
/* sc_lock is not necessary for allocm */
mixer->hwbuf.mem = sc->hw_if->allocm(sc->hw_hdl, mode, bufsize);
if (mixer->hwbuf.mem == NULL) {
audio_printf(sc, "allocm(%zu) failed\n", bufsize);
return ENOMEM;
}
} else {
mixer->hwbuf.mem = kmem_alloc(bufsize, KM_SLEEP);
}
/* From here, audio_mixer_destroy is necessary to exit. */
if (mode == AUMODE_PLAY) {
cv_init(&mixer->outcv, "audiowr");
} else {
cv_init(&mixer->outcv, "audiord");
}
if (mode == AUMODE_PLAY) {
softint_handler = audio_softintr_wr;
} else {
softint_handler = audio_softintr_rd;
}
mixer->sih = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE,
softint_handler, sc);
if (mixer->sih == NULL) {
device_printf(sc->sc_dev, "softint_establish failed\n");
goto abort;
}
mixer->track_fmt.encoding = AUDIO_ENCODING_SLINEAR_NE;
mixer->track_fmt.precision = AUDIO_INTERNAL_BITS;
mixer->track_fmt.stride = AUDIO_INTERNAL_BITS;
mixer->track_fmt.channels = mixer->hwbuf.fmt.channels;
mixer->track_fmt.sample_rate = mixer->hwbuf.fmt.sample_rate;
if (mixer->hwbuf.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE &&
mixer->hwbuf.fmt.precision == AUDIO_INTERNAL_BITS) {
mixer->swap_endian = true;
TRACE(1, "swap_endian");
}
if (mode == AUMODE_PLAY) {
/* Mixing buffer */
mixer->mixfmt = mixer->track_fmt;
mixer->mixfmt.precision *= 2;
mixer->mixfmt.stride *= 2;
/* XXX TODO: use some macros? */
len = mixer->frames_per_block * mixer->mixfmt.channels *
mixer->mixfmt.stride / NBBY;
mixer->mixsample = audio_realloc(mixer->mixsample, len);
} else {
/* No mixing buffer for recording */
}
if (reg->codec) {
mixer->codec = reg->codec;
mixer->codecarg.context = reg->context;
if (mode == AUMODE_PLAY) {
mixer->codecarg.srcfmt = &mixer->track_fmt;
mixer->codecarg.dstfmt = &mixer->hwbuf.fmt;
} else {
mixer->codecarg.srcfmt = &mixer->hwbuf.fmt;
mixer->codecarg.dstfmt = &mixer->track_fmt;
}
mixer->codecbuf.fmt = mixer->track_fmt;
mixer->codecbuf.capacity = mixer->frames_per_block;
len = auring_bytelen(&mixer->codecbuf);
mixer->codecbuf.mem = audio_realloc(mixer->codecbuf.mem, len);
if (mixer->codecbuf.mem == NULL) {
device_printf(sc->sc_dev,
"malloc codecbuf(%d) failed\n", len);
error = ENOMEM;
goto abort;
}
}
/* Succeeded so display it. */
codecbuf[0] = '\0';
if (mixer->codec || mixer->swap_endian) {
snprintf(codecbuf, sizeof(codecbuf), " %s %s:%d",
(mode == AUMODE_PLAY) ? "->" : "<-",
audio_encoding_name(mixer->hwbuf.fmt.encoding),
mixer->hwbuf.fmt.precision);
}
blkms = mixer->blktime_n * 1000 / mixer->blktime_d;
blkdms = (mixer->blktime_n * 10000 / mixer->blktime_d) % 10;
blkdmsbuf[0] = '\0';
if (blkdms != 0) {
snprintf(blkdmsbuf, sizeof(blkdmsbuf), ".%1d", blkdms);
}
aprint_normal_dev(sc->sc_dev,
"%s:%d%s %dch %dHz, blk %d bytes (%d%sms) for %s\n",
audio_encoding_name(mixer->track_fmt.encoding),
mixer->track_fmt.precision,
codecbuf,
mixer->track_fmt.channels,
mixer->track_fmt.sample_rate,
blksize,
blkms, blkdmsbuf,
(mode == AUMODE_PLAY) ? "playback" : "recording");
return 0;
abort:
audio_mixer_destroy(sc, mixer);
return error;
}
/*
* Releases all resources of 'mixer'.
* Note that it does not release the memory area of 'mixer' itself.
* Must be called with sc_exlock held and without sc_lock held.
*/
static void
audio_mixer_destroy(struct audio_softc *sc, audio_trackmixer_t *mixer)
{
int bufsize;
KASSERT(sc->sc_exlock == 1);
bufsize = frametobyte(&mixer->hwbuf.fmt, mixer->hwbuf.capacity);
if (mixer->hwbuf.mem != NULL) {
if (sc->hw_if->freem) {
/* sc_lock is not necessary for freem */
sc->hw_if->freem(sc->hw_hdl, mixer->hwbuf.mem, bufsize);
} else {
kmem_free(mixer->hwbuf.mem, bufsize);
}
mixer->hwbuf.mem = NULL;
}
audio_free(mixer->codecbuf.mem);
audio_free(mixer->mixsample);
cv_destroy(&mixer->outcv);
if (mixer->sih) {
softint_disestablish(mixer->sih);
mixer->sih = NULL;
}
}
/*
* Starts playback mixer.
* Must be called only if sc_pbusy is false.
* Must be called with sc_lock && sc_exlock held.
* Must not be called from the interrupt context.
*/
static void
audio_pmixer_start(struct audio_softc *sc, bool force)
{
audio_trackmixer_t *mixer;
int minimum;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
KASSERT(sc->sc_pbusy == false);
mutex_enter(sc->sc_intr_lock);
mixer = sc->sc_pmixer;
TRACE(2, "%smixseq=%d hwseq=%d hwbuf=%d/%d/%d%s",
(audiodebug >= 3) ? "begin " : "",
(int)mixer->mixseq, (int)mixer->hwseq,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity,
force ? " force" : "");
/* Need two blocks to start normally. */
minimum = (force) ? 1 : 2;
while (mixer->hwbuf.used < mixer->frames_per_block * minimum) {
audio_pmixer_process(sc);
}
/* Start output */
audio_pmixer_output(sc);
sc->sc_pbusy = true;
TRACE(3, "end mixseq=%d hwseq=%d hwbuf=%d/%d/%d",
(int)mixer->mixseq, (int)mixer->hwseq,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity);
mutex_exit(sc->sc_intr_lock);
}
/*
* When playing back with MD filter:
*
* track track ...
* v v
* + mix (with aint2_t)
* | master volume (with aint2_t)
* v
* mixsample [::::] wide-int 1 block (ring) buffer
* |
* | convert aint2_t -> aint_t
* v
* codecbuf [....] 1 block (ring) buffer
* |
* | convert to hw format
* v
* hwbuf [............] NBLKHW blocks ring buffer
*
* When playing back without MD filter:
*
* mixsample [::::] wide-int 1 block (ring) buffer
* |
* | convert aint2_t -> aint_t
* | (with byte swap if necessary)
* v
* hwbuf [............] NBLKHW blocks ring buffer
*
* mixsample: slinear_NE, wide internal precision, HW ch, HW freq.
* codecbuf: slinear_NE, internal precision, HW ch, HW freq.
* hwbuf: HW encoding, HW precision, HW ch, HW freq.
*/
/*
* Performs track mixing and converts it to hwbuf.
* Note that this function doesn't transfer hwbuf to hardware.
* Must be called with sc_intr_lock held.
*/
static void
audio_pmixer_process(struct audio_softc *sc)
{
audio_trackmixer_t *mixer;
audio_file_t *f;
int frame_count;
int sample_count;
int mixed;
int i;
aint2_t *m;
aint_t *h;
mixer = sc->sc_pmixer;
frame_count = mixer->frames_per_block;
KASSERTMSG(auring_get_contig_free(&mixer->hwbuf) >= frame_count,
"auring_get_contig_free()=%d frame_count=%d",
auring_get_contig_free(&mixer->hwbuf), frame_count);
sample_count = frame_count * mixer->mixfmt.channels;
mixer->mixseq++;
/* Mix all tracks */
mixed = 0;
SLIST_FOREACH(f, &sc->sc_files, entry) {
audio_track_t *track = f->ptrack;
if (track == NULL)
continue;
if (track->is_pause) {
TRACET(4, track, "skip; paused");
continue;
}
/* Skip if the track is used by process context. */
if (audio_track_lock_tryenter(track) == false) {
TRACET(4, track, "skip; in use");
continue;
}
/* Emulate mmap'ped track */
if (track->mmapped) {
auring_push(&track->usrbuf, track->usrbuf_blksize);
TRACET(4, track, "mmap; usr=%d/%d/C%d",
track->usrbuf.head,
track->usrbuf.used,
track->usrbuf.capacity);
}
if (track->outbuf.used < mixer->frames_per_block &&
track->usrbuf.used > 0) {
TRACET(4, track, "process");
audio_track_play(track);
}
if (track->outbuf.used > 0) {
mixed = audio_pmixer_mix_track(mixer, track, mixed);
} else {
TRACET(4, track, "skip; empty");
}
audio_track_lock_exit(track);
}
if (mixed == 0) {
/* Silence */
memset(mixer->mixsample, 0,
frametobyte(&mixer->mixfmt, frame_count));
} else {
if (mixed > 1) {
/* If there are multiple tracks, do auto gain control */
audio_pmixer_agc(mixer, sample_count);
}
/* Apply master volume */
if (mixer->volume < 256) {
m = mixer->mixsample;
for (i = 0; i < sample_count; i++) {
*m = AUDIO_SCALEDOWN(*m * mixer->volume, 8);
m++;
}
/*
* Recover the volume gradually at the pace of
* several times per second. If it's too fast, you
* can recognize that the volume changes up and down
* quickly and it's not so comfortable.
*/
mixer->voltimer += mixer->blktime_n;
if (mixer->voltimer * 4 >= mixer->blktime_d) {
mixer->volume++;
mixer->voltimer = 0;
#if defined(AUDIO_DEBUG_AGC)
TRACE(1, "volume recover: %d", mixer->volume);
#endif
}
}
}
/*
* The rest is the hardware part.
*/
if (mixer->codec) {
h = auring_tailptr_aint(&mixer->codecbuf);
} else {
h = auring_tailptr_aint(&mixer->hwbuf);
}
m = mixer->mixsample;
if (mixer->swap_endian) {
for (i = 0; i < sample_count; i++) {
*h++ = bswap16(*m++);
}
} else {
for (i = 0; i < sample_count; i++) {
*h++ = *m++;
}
}
/* Hardware driver's codec */
if (mixer->codec) {
auring_push(&mixer->codecbuf, frame_count);
mixer->codecarg.src = auring_headptr(&mixer->codecbuf);
mixer->codecarg.dst = auring_tailptr(&mixer->hwbuf);
mixer->codecarg.count = frame_count;
mixer->codec(&mixer->codecarg);
auring_take(&mixer->codecbuf, mixer->codecarg.count);
}
auring_push(&mixer->hwbuf, frame_count);
TRACE(4, "done mixseq=%d hwbuf=%d/%d/%d%s",
(int)mixer->mixseq,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity,
(mixed == 0) ? " silent" : "");
}
/*
* Do auto gain control.
* Must be called sc_intr_lock held.
*/
static void
audio_pmixer_agc(audio_trackmixer_t *mixer, int sample_count)
{
struct audio_softc *sc __unused;
aint2_t val;
aint2_t maxval;
aint2_t minval;
aint2_t over_plus;
aint2_t over_minus;
aint2_t *m;
int newvol;
int i;
sc = mixer->sc;
/* Overflow detection */
maxval = AINT_T_MAX;
minval = AINT_T_MIN;
m = mixer->mixsample;
for (i = 0; i < sample_count; i++) {
val = *m++;
if (val > maxval)
maxval = val;
else if (val < minval)
minval = val;
}
/* Absolute value of overflowed amount */
over_plus = maxval - AINT_T_MAX;
over_minus = AINT_T_MIN - minval;
if (over_plus > 0 || over_minus > 0) {
if (over_plus > over_minus) {
newvol = (int)((aint2_t)AINT_T_MAX * 256 / maxval);
} else {
newvol = (int)((aint2_t)AINT_T_MIN * 256 / minval);
}
/*
* Change the volume only if new one is smaller.
* Reset the timer even if the volume isn't changed.
*/
if (newvol <= mixer->volume) {
mixer->volume = newvol;
mixer->voltimer = 0;
#if defined(AUDIO_DEBUG_AGC)
TRACE(1, "auto volume adjust: %d", mixer->volume);
#endif
}
}
}
/*
* Mix one track.
* 'mixed' specifies the number of tracks mixed so far.
* It returns the number of tracks mixed. In other words, it returns
* mixed + 1 if this track is mixed.
*/
static int
audio_pmixer_mix_track(audio_trackmixer_t *mixer, audio_track_t *track,
int mixed)
{
int count;
int sample_count;
int remain;
int i;
const aint_t *s;
aint2_t *d;
/* XXX TODO: Is this necessary for now? */
if (mixer->mixseq < track->seq)
return mixed;
count = auring_get_contig_used(&track->outbuf);
count = uimin(count, mixer->frames_per_block);
s = auring_headptr_aint(&track->outbuf);
d = mixer->mixsample;
/*
* Apply track volume with double-sized integer and perform
* additive synthesis.
*
* XXX If you limit the track volume to 1.0 or less (<= 256),
* it would be better to do this in the track conversion stage
* rather than here. However, if you accept the volume to
* be greater than 1.0 (> 256), it's better to do it here.
* Because the operation here is done by double-sized integer.
*/
sample_count = count * mixer->mixfmt.channels;
if (mixed == 0) {
/* If this is the first track, assignment can be used. */
#if defined(AUDIO_SUPPORT_TRACK_VOLUME)
if (track->volume != 256) {
for (i = 0; i < sample_count; i++) {
aint2_t v;
v = *s++;
*d++ = AUDIO_SCALEDOWN(v * track->volume, 8)
}
} else
#endif
{
for (i = 0; i < sample_count; i++) {
*d++ = ((aint2_t)*s++);
}
}
/* Fill silence if the first track is not filled. */
for (; i < mixer->frames_per_block * mixer->mixfmt.channels; i++)
*d++ = 0;
} else {
/* If this is the second or later, add it. */
#if defined(AUDIO_SUPPORT_TRACK_VOLUME)
if (track->volume != 256) {
for (i = 0; i < sample_count; i++) {
aint2_t v;
v = *s++;
*d++ += AUDIO_SCALEDOWN(v * track->volume, 8);
}
} else
#endif
{
for (i = 0; i < sample_count; i++) {
*d++ += ((aint2_t)*s++);
}
}
}
auring_take(&track->outbuf, count);
/*
* The counters have to align block even if outbuf is less than
* one block. XXX Is this still necessary?
*/
remain = mixer->frames_per_block - count;
if (__predict_false(remain != 0)) {
auring_push(&track->outbuf, remain);
auring_take(&track->outbuf, remain);
}
/*
* Update track sequence.
* mixseq has previous value yet at this point.
*/
track->seq = mixer->mixseq + 1;
return mixed + 1;
}
/*
* Output one block from hwbuf to HW.
* Must be called with sc_intr_lock held.
*/
static void
audio_pmixer_output(struct audio_softc *sc)
{
audio_trackmixer_t *mixer;
audio_params_t params;
void *start;
void *end;
int blksize;
int error;
mixer = sc->sc_pmixer;
TRACE(4, "pbusy=%d hwbuf=%d/%d/%d",
sc->sc_pbusy,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity);
KASSERTMSG(mixer->hwbuf.used >= mixer->frames_per_block,
"mixer->hwbuf.used=%d mixer->frames_per_block=%d",
mixer->hwbuf.used, mixer->frames_per_block);
blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block);
if (sc->hw_if->trigger_output) {
/* trigger (at once) */
if (!sc->sc_pbusy) {
start = mixer->hwbuf.mem;
end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf);
params = format2_to_params(&mixer->hwbuf.fmt);
error = sc->hw_if->trigger_output(sc->hw_hdl,
start, end, blksize, audio_pintr, sc, ¶ms);
if (error) {
audio_printf(sc,
"trigger_output failed: errno=%d\n",
error);
return;
}
}
} else {
/* start (everytime) */
start = auring_headptr(&mixer->hwbuf);
error = sc->hw_if->start_output(sc->hw_hdl,
start, blksize, audio_pintr, sc);
if (error) {
audio_printf(sc,
"start_output failed: errno=%d\n", error);
return;
}
}
}
/*
* This is an interrupt handler for playback.
* It is called with sc_intr_lock held.
*
* It is usually called from hardware interrupt. However, note that
* for some drivers (e.g. uaudio) it is called from software interrupt.
*/
static void
audio_pintr(void *arg)
{
struct audio_softc *sc;
audio_trackmixer_t *mixer;
sc = arg;
KASSERT(mutex_owned(sc->sc_intr_lock));
if (sc->sc_dying)
return;
if (sc->sc_pbusy == false) {
#if defined(DIAGNOSTIC)
audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n",
device_xname(sc->hw_dev));
#endif
return;
}
mixer = sc->sc_pmixer;
mixer->hw_complete_counter += mixer->frames_per_block;
mixer->hwseq++;
auring_take(&mixer->hwbuf, mixer->frames_per_block);
TRACE(4,
"HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d",
mixer->hwseq, mixer->hw_complete_counter,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity);
#if defined(AUDIO_HW_SINGLE_BUFFER)
/*
* Create a new block here and output it immediately.
* It makes a latency lower but needs machine power.
*/
audio_pmixer_process(sc);
audio_pmixer_output(sc);
#else
/*
* It is called when block N output is done.
* Output immediately block N+1 created by the last interrupt.
* And then create block N+2 for the next interrupt.
* This method makes playback robust even on slower machines.
* Instead the latency is increased by one block.
*/
/* At first, output ready block. */
if (mixer->hwbuf.used >= mixer->frames_per_block) {
audio_pmixer_output(sc);
}
bool later = false;
if (mixer->hwbuf.used < mixer->frames_per_block) {
later = true;
}
/* Then, process next block. */
audio_pmixer_process(sc);
if (later) {
audio_pmixer_output(sc);
}
#endif
/*
* When this interrupt is the real hardware interrupt, disabling
* preemption here is not necessary. But some drivers (e.g. uaudio)
* emulate it by software interrupt, so kpreempt_disable is necessary.
*/
kpreempt_disable();
softint_schedule(mixer->sih);
kpreempt_enable();
}
/*
* Starts record mixer.
* Must be called only if sc_rbusy is false.
* Must be called with sc_lock && sc_exlock held.
* Must not be called from the interrupt context.
*/
static void
audio_rmixer_start(struct audio_softc *sc)
{
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
KASSERT(sc->sc_rbusy == false);
mutex_enter(sc->sc_intr_lock);
TRACE(2, "%s", (audiodebug >= 3) ? "begin" : "");
audio_rmixer_input(sc);
sc->sc_rbusy = true;
TRACE(3, "end");
mutex_exit(sc->sc_intr_lock);
}
/*
* When recording with MD filter:
*
* hwbuf [............] NBLKHW blocks ring buffer
* |
* | convert from hw format
* v
* codecbuf [....] 1 block (ring) buffer
* | |
* v v
* track track ...
*
* When recording without MD filter:
*
* hwbuf [............] NBLKHW blocks ring buffer
* | |
* v v
* track track ...
*
* hwbuf: HW encoding, HW precision, HW ch, HW freq.
* codecbuf: slinear_NE, internal precision, HW ch, HW freq.
*/
/*
* Distribute a recorded block to all recording tracks.
*/
static void
audio_rmixer_process(struct audio_softc *sc)
{
audio_trackmixer_t *mixer;
audio_ring_t *mixersrc;
audio_file_t *f;
aint_t *p;
int count;
int bytes;
int i;
mixer = sc->sc_rmixer;
/*
* count is the number of frames to be retrieved this time.
* count should be one block.
*/
count = auring_get_contig_used(&mixer->hwbuf);
count = uimin(count, mixer->frames_per_block);
if (count <= 0) {
TRACE(4, "count %d: too short", count);
return;
}
bytes = frametobyte(&mixer->track_fmt, count);
/* Hardware driver's codec */
if (mixer->codec) {
mixer->codecarg.src = auring_headptr(&mixer->hwbuf);
mixer->codecarg.dst = auring_tailptr(&mixer->codecbuf);
mixer->codecarg.count = count;
mixer->codec(&mixer->codecarg);
auring_take(&mixer->hwbuf, mixer->codecarg.count);
auring_push(&mixer->codecbuf, mixer->codecarg.count);
mixersrc = &mixer->codecbuf;
} else {
mixersrc = &mixer->hwbuf;
}
if (mixer->swap_endian) {
/* inplace conversion */
p = auring_headptr_aint(mixersrc);
for (i = 0; i < count * mixer->track_fmt.channels; i++, p++) {
*p = bswap16(*p);
}
}
/* Distribute to all tracks. */
SLIST_FOREACH(f, &sc->sc_files, entry) {
audio_track_t *track = f->rtrack;
audio_ring_t *input;
if (track == NULL)
continue;
if (track->is_pause) {
TRACET(4, track, "skip; paused");
continue;
}
if (audio_track_lock_tryenter(track) == false) {
TRACET(4, track, "skip; in use");
continue;
}
/* If the track buffer is full, discard the oldest one? */
input = track->input;
if (input->capacity - input->used < mixer->frames_per_block) {
int drops = mixer->frames_per_block -
(input->capacity - input->used);
track->dropframes += drops;
TRACET(4, track, "drop %d frames: inp=%d/%d/%d",
drops,
input->head, input->used, input->capacity);
auring_take(input, drops);
}
KASSERTMSG(input->used % mixer->frames_per_block == 0,
"input->used=%d mixer->frames_per_block=%d",
input->used, mixer->frames_per_block);
memcpy(auring_tailptr_aint(input),
auring_headptr_aint(mixersrc),
bytes);
auring_push(input, count);
/* XXX sequence counter? */
audio_track_lock_exit(track);
}
auring_take(mixersrc, count);
}
/*
* Input one block from HW to hwbuf.
* Must be called with sc_intr_lock held.
*/
static void
audio_rmixer_input(struct audio_softc *sc)
{
audio_trackmixer_t *mixer;
audio_params_t params;
void *start;
void *end;
int blksize;
int error;
mixer = sc->sc_rmixer;
blksize = frametobyte(&mixer->hwbuf.fmt, mixer->frames_per_block);
if (sc->hw_if->trigger_input) {
/* trigger (at once) */
if (!sc->sc_rbusy) {
start = mixer->hwbuf.mem;
end = (uint8_t *)start + auring_bytelen(&mixer->hwbuf);
params = format2_to_params(&mixer->hwbuf.fmt);
error = sc->hw_if->trigger_input(sc->hw_hdl,
start, end, blksize, audio_rintr, sc, ¶ms);
if (error) {
audio_printf(sc,
"trigger_input failed: errno=%d\n",
error);
return;
}
}
} else {
/* start (everytime) */
start = auring_tailptr(&mixer->hwbuf);
error = sc->hw_if->start_input(sc->hw_hdl,
start, blksize, audio_rintr, sc);
if (error) {
audio_printf(sc,
"start_input failed: errno=%d\n", error);
return;
}
}
}
/*
* This is an interrupt handler for recording.
* It is called with sc_intr_lock.
*
* It is usually called from hardware interrupt. However, note that
* for some drivers (e.g. uaudio) it is called from software interrupt.
*/
static void
audio_rintr(void *arg)
{
struct audio_softc *sc;
audio_trackmixer_t *mixer;
sc = arg;
KASSERT(mutex_owned(sc->sc_intr_lock));
if (sc->sc_dying)
return;
if (sc->sc_rbusy == false) {
#if defined(DIAGNOSTIC)
audio_printf(sc, "DIAGNOSTIC: %s raised stray interrupt\n",
device_xname(sc->hw_dev));
#endif
return;
}
mixer = sc->sc_rmixer;
mixer->hw_complete_counter += mixer->frames_per_block;
mixer->hwseq++;
auring_push(&mixer->hwbuf, mixer->frames_per_block);
TRACE(4,
"HW_INT ++hwseq=%" PRIu64 " cmplcnt=%" PRIu64 " hwbuf=%d/%d/%d",
mixer->hwseq, mixer->hw_complete_counter,
mixer->hwbuf.head, mixer->hwbuf.used, mixer->hwbuf.capacity);
/* Distrubute recorded block */
audio_rmixer_process(sc);
/* Request next block */
audio_rmixer_input(sc);
/*
* When this interrupt is the real hardware interrupt, disabling
* preemption here is not necessary. But some drivers (e.g. uaudio)
* emulate it by software interrupt, so kpreempt_disable is necessary.
*/
kpreempt_disable();
softint_schedule(mixer->sih);
kpreempt_enable();
}
/*
* Halts playback mixer.
* This function also clears related parameters, so call this function
* instead of calling halt_output directly.
* Must be called only if sc_pbusy is true.
* Must be called with sc_lock && sc_exlock held.
*/
static int
audio_pmixer_halt(struct audio_softc *sc)
{
int error;
TRACE(2, "called");
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->halt_output(sc->hw_hdl);
/* Halts anyway even if some error has occurred. */
sc->sc_pbusy = false;
sc->sc_pmixer->hwbuf.head = 0;
sc->sc_pmixer->hwbuf.used = 0;
sc->sc_pmixer->mixseq = 0;
sc->sc_pmixer->hwseq = 0;
mutex_exit(sc->sc_intr_lock);
return error;
}
/*
* Halts recording mixer.
* This function also clears related parameters, so call this function
* instead of calling halt_input directly.
* Must be called only if sc_rbusy is true.
* Must be called with sc_lock && sc_exlock held.
*/
static int
audio_rmixer_halt(struct audio_softc *sc)
{
int error;
TRACE(2, "called");
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
mutex_enter(sc->sc_intr_lock);
error = sc->hw_if->halt_input(sc->hw_hdl);
/* Halts anyway even if some error has occurred. */
sc->sc_rbusy = false;
sc->sc_rmixer->hwbuf.head = 0;
sc->sc_rmixer->hwbuf.used = 0;
sc->sc_rmixer->mixseq = 0;
sc->sc_rmixer->hwseq = 0;
mutex_exit(sc->sc_intr_lock);
return error;
}
/*
* Flush this track.
* Halts all operations, clears all buffers, reset error counters.
* XXX I'm not sure...
*/
static void
audio_track_clear(struct audio_softc *sc, audio_track_t *track)
{
KASSERT(track);
TRACET(3, track, "clear");
audio_track_lock_enter(track);
track->usrbuf.used = 0;
/* Clear all internal parameters. */
if (track->codec.filter) {
track->codec.srcbuf.used = 0;
track->codec.srcbuf.head = 0;
}
if (track->chvol.filter) {
track->chvol.srcbuf.used = 0;
track->chvol.srcbuf.head = 0;
}
if (track->chmix.filter) {
track->chmix.srcbuf.used = 0;
track->chmix.srcbuf.head = 0;
}
if (track->freq.filter) {
track->freq.srcbuf.used = 0;
track->freq.srcbuf.head = 0;
if (track->freq_step < 65536)
track->freq_current = 65536;
else
track->freq_current = 0;
memset(track->freq_prev, 0, sizeof(track->freq_prev));
memset(track->freq_curr, 0, sizeof(track->freq_curr));
}
/* Clear buffer, then operation halts naturally. */
track->outbuf.used = 0;
/* Clear counters. */
track->dropframes = 0;
audio_track_lock_exit(track);
}
/*
* Drain the track.
* track must be present and for playback.
* If successful, it returns 0. Otherwise returns errno.
* Must be called with sc_lock held.
*/
static int
audio_track_drain(struct audio_softc *sc, audio_track_t *track)
{
audio_trackmixer_t *mixer;
int done;
int error;
KASSERT(track);
TRACET(3, track, "start");
mixer = track->mixer;
KASSERT(mutex_owned(sc->sc_lock));
/* Ignore them if pause. */
if (track->is_pause) {
TRACET(3, track, "pause -> clear");
track->pstate = AUDIO_STATE_CLEAR;
}
/* Terminate early here if there is no data in the track. */
if (track->pstate == AUDIO_STATE_CLEAR) {
TRACET(3, track, "no need to drain");
return 0;
}
track->pstate = AUDIO_STATE_DRAINING;
for (;;) {
/* I want to display it before condition evaluation. */
TRACET(3, track, "pid=%d.%d trkseq=%d hwseq=%d out=%d/%d/%d",
(int)curproc->p_pid, (int)curlwp->l_lid,
(int)track->seq, (int)mixer->hwseq,
track->outbuf.head, track->outbuf.used,
track->outbuf.capacity);
/* Condition to terminate */
audio_track_lock_enter(track);
done = (track->usrbuf.used < frametobyte(&track->inputfmt, 1) &&
track->outbuf.used == 0 &&
track->seq <= mixer->hwseq);
audio_track_lock_exit(track);
if (done)
break;
TRACET(3, track, "sleep");
error = audio_track_waitio(sc, track);
if (error)
return error;
/* XXX call audio_track_play here ? */
}
track->pstate = AUDIO_STATE_CLEAR;
TRACET(3, track, "done trk_inp=%d trk_out=%d",
(int)track->inputcounter, (int)track->outputcounter);
return 0;
}
/*
* Send signal to process.
* This is intended to be called only from audio_softintr_{rd,wr}.
* Must be called without sc_intr_lock held.
*/
static inline void
audio_psignal(struct audio_softc *sc, pid_t pid, int signum)
{
proc_t *p;
KASSERT(pid != 0);
/*
* psignal() must be called without spin lock held.
*/
mutex_enter(&proc_lock);
p = proc_find(pid);
if (p)
psignal(p, signum);
mutex_exit(&proc_lock);
}
/*
* This is software interrupt handler for record.
* It is called from recording hardware interrupt everytime.
* It does:
* - Deliver SIGIO for all async processes.
* - Notify to audio_read() that data has arrived.
* - selnotify() for select/poll-ing processes.
*/
/*
* XXX If a process issues FIOASYNC between hardware interrupt and
* software interrupt, (stray) SIGIO will be sent to the process
* despite the fact that it has not receive recorded data yet.
*/
static void
audio_softintr_rd(void *cookie)
{
struct audio_softc *sc = cookie;
audio_file_t *f;
pid_t pid;
mutex_enter(sc->sc_lock);
SLIST_FOREACH(f, &sc->sc_files, entry) {
audio_track_t *track = f->rtrack;
if (track == NULL)
continue;
TRACET(4, track, "broadcast; inp=%d/%d/%d",
track->input->head,
track->input->used,
track->input->capacity);
pid = f->async_audio;
if (pid != 0) {
TRACEF(4, f, "sending SIGIO %d", pid);
audio_psignal(sc, pid, SIGIO);
}
}
/* Notify that data has arrived. */
selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT);
cv_broadcast(&sc->sc_rmixer->outcv);
mutex_exit(sc->sc_lock);
}
/*
* This is software interrupt handler for playback.
* It is called from playback hardware interrupt everytime.
* It does:
* - Deliver SIGIO for all async and writable (used < lowat) processes.
* - Notify to audio_write() that outbuf block available.
* - selnotify() for select/poll-ing processes if there are any writable
* (used < lowat) processes. Checking each descriptor will be done by
* filt_audiowrite_event().
*/
static void
audio_softintr_wr(void *cookie)
{
struct audio_softc *sc = cookie;
audio_file_t *f;
bool found;
pid_t pid;
TRACE(4, "called");
found = false;
mutex_enter(sc->sc_lock);
SLIST_FOREACH(f, &sc->sc_files, entry) {
audio_track_t *track = f->ptrack;
if (track == NULL)
continue;
TRACET(4, track, "broadcast; trkseq=%d out=%d/%d/%d",
(int)track->seq,
track->outbuf.head,
track->outbuf.used,
track->outbuf.capacity);
/*
* Send a signal if the process is async mode and
* used is lower than lowat.
*/
if (track->usrbuf.used <= track->usrbuf_usedlow &&
!track->is_pause) {
/* For selnotify */
found = true;
/* For SIGIO */
pid = f->async_audio;
if (pid != 0) {
TRACEF(4, f, "sending SIGIO %d", pid);
audio_psignal(sc, pid, SIGIO);
}
}
}
/*
* Notify for select/poll when someone become writable.
* It needs sc_lock (and not sc_intr_lock).
*/
if (found) {
TRACE(4, "selnotify");
selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT);
}
/* Notify to audio_write() that outbuf available. */
cv_broadcast(&sc->sc_pmixer->outcv);
mutex_exit(sc->sc_lock);
}
/*
* Check (and convert) the format *p came from userland.
* If successful, it writes back the converted format to *p if necessary and
* returns 0. Otherwise returns errno (*p may be changed even in this case).
*/
static int
audio_check_params(audio_format2_t *p)
{
/*
* Convert obsolete AUDIO_ENCODING_PCM encodings.
*
* AUDIO_ENCODING_PCM16 == AUDIO_ENCODING_LINEAR
* So, it's always signed, as in SunOS.
*
* AUDIO_ENCODING_PCM8 == AUDIO_ENCODING_LINEAR8
* So, it's always unsigned, as in SunOS.
*/
if (p->encoding == AUDIO_ENCODING_PCM16) {
p->encoding = AUDIO_ENCODING_SLINEAR;
} else if (p->encoding == AUDIO_ENCODING_PCM8) {
if (p->precision == 8)
p->encoding = AUDIO_ENCODING_ULINEAR;
else
return EINVAL;
}
/*
* Convert obsoleted AUDIO_ENCODING_[SU]LINEAR without endianness
* suffix.
*/
if (p->encoding == AUDIO_ENCODING_SLINEAR)
p->encoding = AUDIO_ENCODING_SLINEAR_NE;
if (p->encoding == AUDIO_ENCODING_ULINEAR)
p->encoding = AUDIO_ENCODING_ULINEAR_NE;
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:
if (p->precision != 8 && p->precision != 16 &&
p->precision != 24 && p->precision != 32)
return EINVAL;
/* 8bit format does not have endianness. */
if (p->precision == 8) {
if (p->encoding == AUDIO_ENCODING_SLINEAR_OE)
p->encoding = AUDIO_ENCODING_SLINEAR_NE;
if (p->encoding == AUDIO_ENCODING_ULINEAR_OE)
p->encoding = AUDIO_ENCODING_ULINEAR_NE;
}
if (p->precision > p->stride)
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;
}
/*
* Initialize playback and record mixers.
* mode (AUMODE_{PLAY,RECORD}) indicates the mixer to be initialized.
* phwfmt and rhwfmt indicate the hardware format. pfil and rfil indicate
* the filter registration information. These four must not be NULL.
* If successful returns 0. Otherwise returns errno.
* Must be called with sc_exlock held and without sc_lock held.
* Must not be called if there are any tracks.
* Caller should check that the initialization succeed by whether
* sc_[pr]mixer is not NULL.
*/
static int
audio_mixers_init(struct audio_softc *sc, int mode,
const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt,
const audio_filter_reg_t *pfil, const audio_filter_reg_t *rfil)
{
int error;
KASSERT(phwfmt != NULL);
KASSERT(rhwfmt != NULL);
KASSERT(pfil != NULL);
KASSERT(rfil != NULL);
KASSERT(sc->sc_exlock);
if ((mode & AUMODE_PLAY)) {
if (sc->sc_pmixer == NULL) {
sc->sc_pmixer = kmem_zalloc(sizeof(*sc->sc_pmixer),
KM_SLEEP);
} else {
/* destroy() doesn't free memory. */
audio_mixer_destroy(sc, sc->sc_pmixer);
memset(sc->sc_pmixer, 0, sizeof(*sc->sc_pmixer));
}
error = audio_mixer_init(sc, AUMODE_PLAY, phwfmt, pfil);
if (error) {
/* audio_mixer_init already displayed error code */
audio_printf(sc, "configuring playback mode failed\n");
kmem_free(sc->sc_pmixer, sizeof(*sc->sc_pmixer));
sc->sc_pmixer = NULL;
return error;
}
}
if ((mode & AUMODE_RECORD)) {
if (sc->sc_rmixer == NULL) {
sc->sc_rmixer = kmem_zalloc(sizeof(*sc->sc_rmixer),
KM_SLEEP);
} else {
/* destroy() doesn't free memory. */
audio_mixer_destroy(sc, sc->sc_rmixer);
memset(sc->sc_rmixer, 0, sizeof(*sc->sc_rmixer));
}
error = audio_mixer_init(sc, AUMODE_RECORD, rhwfmt, rfil);
if (error) {
/* audio_mixer_init already displayed error code */
audio_printf(sc, "configuring record mode failed\n");
kmem_free(sc->sc_rmixer, sizeof(*sc->sc_rmixer));
sc->sc_rmixer = NULL;
return error;
}
}
return 0;
}
/*
* Select a frequency.
* Prioritize 48kHz and 44.1kHz. Otherwise choose the highest one.
* XXX Better algorithm?
*/
static int
audio_select_freq(const struct audio_format *fmt)
{
int freq;
int high;
int low;
int j;
if (fmt->frequency_type == 0) {
low = fmt->frequency[0];
high = fmt->frequency[1];
freq = 48000;
if (low <= freq && freq <= high) {
return freq;
}
freq = 44100;
if (low <= freq && freq <= high) {
return freq;
}
return high;
} else {
for (j = 0; j < fmt->frequency_type; j++) {
if (fmt->frequency[j] == 48000) {
return fmt->frequency[j];
}
}
high = 0;
for (j = 0; j < fmt->frequency_type; j++) {
if (fmt->frequency[j] == 44100) {
return fmt->frequency[j];
}
if (fmt->frequency[j] > high) {
high = fmt->frequency[j];
}
}
return high;
}
}
/*
* Choose the most preferred hardware format.
* If successful, it will store the chosen format into *cand and return 0.
* Otherwise, return errno.
* Must be called without sc_lock held.
*/
static int
audio_hw_probe(struct audio_softc *sc, audio_format2_t *cand, int mode)
{
audio_format_query_t query;
int cand_score;
int score;
int i;
int error;
/*
* Score each formats and choose the highest one.
*
* +---- priority(0-3)
* |+--- encoding/precision
* ||+-- channels
* score = 0x000000PEC
*/
cand_score = 0;
for (i = 0; ; i++) {
memset(&query, 0, sizeof(query));
query.index = i;
mutex_enter(sc->sc_lock);
error = sc->hw_if->query_format(sc->hw_hdl, &query);
mutex_exit(sc->sc_lock);
if (error == EINVAL)
break;
if (error)
return error;
#if defined(AUDIO_DEBUG)
DPRINTF(1, "fmt[%d] %c%c pri=%d %s,%d/%dbit,%dch,", i,
(query.fmt.mode & AUMODE_PLAY) ? 'P' : '-',
(query.fmt.mode & AUMODE_RECORD) ? 'R' : '-',
query.fmt.priority,
audio_encoding_name(query.fmt.encoding),
query.fmt.validbits,
query.fmt.precision,
query.fmt.channels);
if (query.fmt.frequency_type == 0) {
DPRINTF(1, "{%d-%d",
query.fmt.frequency[0], query.fmt.frequency[1]);
} else {
int j;
for (j = 0; j < query.fmt.frequency_type; j++) {
DPRINTF(1, "%c%d",
(j == 0) ? '{' : ',',
query.fmt.frequency[j]);
}
}
DPRINTF(1, "}\n");
#endif
if ((query.fmt.mode & mode) == 0) {
DPRINTF(1, "fmt[%d] skip; mode not match %d\n", i,
mode);
continue;
}
if (query.fmt.priority < 0) {
DPRINTF(1, "fmt[%d] skip; unsupported encoding\n", i);
continue;
}
/* Score */
score = (query.fmt.priority & 3) * 0x100;
if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_NE &&
query.fmt.validbits == AUDIO_INTERNAL_BITS &&
query.fmt.precision == AUDIO_INTERNAL_BITS) {
score += 0x20;
} else if (query.fmt.encoding == AUDIO_ENCODING_SLINEAR_OE &&
query.fmt.validbits == AUDIO_INTERNAL_BITS &&
query.fmt.precision == AUDIO_INTERNAL_BITS) {
score += 0x10;
}
score += query.fmt.channels;
if (score < cand_score) {
DPRINTF(1, "fmt[%d] skip; score 0x%x < 0x%x\n", i,
score, cand_score);
continue;
}
/* Update candidate */
cand_score = score;
cand->encoding = query.fmt.encoding;
cand->precision = query.fmt.validbits;
cand->stride = query.fmt.precision;
cand->channels = query.fmt.channels;
cand->sample_rate = audio_select_freq(&query.fmt);
DPRINTF(1, "fmt[%d] candidate (score=0x%x)"
" pri=%d %s,%d/%d,%dch,%dHz\n", i,
cand_score, query.fmt.priority,
audio_encoding_name(query.fmt.encoding),
cand->precision, cand->stride,
cand->channels, cand->sample_rate);
}
if (cand_score == 0) {
DPRINTF(1, "%s no fmt\n", __func__);
return ENXIO;
}
DPRINTF(1, "%s selected: %s,%d/%d,%dch,%dHz\n", __func__,
audio_encoding_name(cand->encoding),
cand->precision, cand->stride, cand->channels, cand->sample_rate);
return 0;
}
/*
* Validate fmt with query_format.
* If fmt is included in the result of query_format, returns 0.
* Otherwise returns EINVAL.
* Must be called without sc_lock held.
*/
static int
audio_hw_validate_format(struct audio_softc *sc, int mode,
const audio_format2_t *fmt)
{
audio_format_query_t query;
struct audio_format *q;
int index;
int error;
int j;
for (index = 0; ; index++) {
query.index = index;
mutex_enter(sc->sc_lock);
error = sc->hw_if->query_format(sc->hw_hdl, &query);
mutex_exit(sc->sc_lock);
if (error == EINVAL)
break;
if (error)
return error;
q = &query.fmt;
/*
* Note that fmt is audio_format2_t (precision/stride) but
* q is audio_format_t (validbits/precision).
*/
if ((q->mode & mode) == 0) {
continue;
}
if (fmt->encoding != q->encoding) {
continue;
}
if (fmt->precision != q->validbits) {
continue;
}
if (fmt->stride != q->precision) {
continue;
}
if (fmt->channels != q->channels) {
continue;
}
if (q->frequency_type == 0) {
if (fmt->sample_rate < q->frequency[0] ||
fmt->sample_rate > q->frequency[1]) {
continue;
}
} else {
for (j = 0; j < q->frequency_type; j++) {
if (fmt->sample_rate == q->frequency[j])
break;
}
if (j == query.fmt.frequency_type) {
continue;
}
}
/* Matched. */
return 0;
}
return EINVAL;
}
/*
* Set track mixer's format depending on ai->mode.
* If AUMODE_PLAY is set in ai->mode, it set up the playback mixer
* with ai.play.*.
* If AUMODE_RECORD is set in ai->mode, it set up the recording mixer
* with ai.record.*.
* All other fields in ai are ignored.
* If successful returns 0. Otherwise returns errno.
* This function does not roll back even if it fails.
* Must be called with sc_exlock held and without sc_lock held.
*/
static int
audio_mixers_set_format(struct audio_softc *sc, const struct audio_info *ai)
{
audio_format2_t phwfmt;
audio_format2_t rhwfmt;
audio_filter_reg_t pfil;
audio_filter_reg_t rfil;
int mode;
int error;
KASSERT(sc->sc_exlock);
/*
* Even when setting either one of playback and recording,
* both must be halted.
*/
if (sc->sc_popens + sc->sc_ropens > 0)
return EBUSY;
if (!SPECIFIED(ai->mode) || ai->mode == 0)
return ENOTTY;
mode = ai->mode;
if ((mode & AUMODE_PLAY)) {
phwfmt.encoding = ai->play.encoding;
phwfmt.precision = ai->play.precision;
phwfmt.stride = ai->play.precision;
phwfmt.channels = ai->play.channels;
phwfmt.sample_rate = ai->play.sample_rate;
}
if ((mode & AUMODE_RECORD)) {
rhwfmt.encoding = ai->record.encoding;
rhwfmt.precision = ai->record.precision;
rhwfmt.stride = ai->record.precision;
rhwfmt.channels = ai->record.channels;
rhwfmt.sample_rate = ai->record.sample_rate;
}
/* On non-independent devices, use the same format for both. */
if ((sc->sc_props & AUDIO_PROP_INDEPENDENT) == 0) {
if (mode == AUMODE_RECORD) {
phwfmt = rhwfmt;
} else {
rhwfmt = phwfmt;
}
mode = AUMODE_PLAY | AUMODE_RECORD;
}
/* Then, unset the direction not exist on the hardware. */
if ((sc->sc_props & AUDIO_PROP_PLAYBACK) == 0)
mode &= ~AUMODE_PLAY;
if ((sc->sc_props & AUDIO_PROP_CAPTURE) == 0)
mode &= ~AUMODE_RECORD;
/* debug */
if ((mode & AUMODE_PLAY)) {
TRACE(1, "play=%s/%d/%d/%dch/%dHz",
audio_encoding_name(phwfmt.encoding),
phwfmt.precision,
phwfmt.stride,
phwfmt.channels,
phwfmt.sample_rate);
}
if ((mode & AUMODE_RECORD)) {
TRACE(1, "rec =%s/%d/%d/%dch/%dHz",
audio_encoding_name(rhwfmt.encoding),
rhwfmt.precision,
rhwfmt.stride,
rhwfmt.channels,
rhwfmt.sample_rate);
}
/* Check the format */
if ((mode & AUMODE_PLAY)) {
if (audio_hw_validate_format(sc, AUMODE_PLAY, &phwfmt)) {
TRACE(1, "invalid format");
return EINVAL;
}
}
if ((mode & AUMODE_RECORD)) {
if (audio_hw_validate_format(sc, AUMODE_RECORD, &rhwfmt)) {
TRACE(1, "invalid format");
return EINVAL;
}
}
/* Configure the mixers. */
memset(&pfil, 0, sizeof(pfil));
memset(&rfil, 0, sizeof(rfil));
error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (error)
return error;
error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (error)
return error;
/*
* Reinitialize the sticky parameters for /dev/sound.
* If the number of the hardware channels becomes less than the number
* of channels that sticky parameters remember, subsequent /dev/sound
* open will fail. To prevent this, reinitialize the sticky
* parameters whenever the hardware format is changed.
*/
sc->sc_sound_pparams = params_to_format2(&audio_default);
sc->sc_sound_rparams = params_to_format2(&audio_default);
sc->sc_sound_ppause = false;
sc->sc_sound_rpause = false;
return 0;
}
/*
* Store current mixers format into *ai.
* Must be called with sc_exlock held.
*/
static void
audio_mixers_get_format(struct audio_softc *sc, struct audio_info *ai)
{
KASSERT(sc->sc_exlock);
/*
* There is no stride information in audio_info but it doesn't matter.
* trackmixer always treats stride and precision as the same.
*/
AUDIO_INITINFO(ai);
ai->mode = 0;
if (sc->sc_pmixer) {
audio_format2_t *fmt = &sc->sc_pmixer->track_fmt;
ai->play.encoding = fmt->encoding;
ai->play.precision = fmt->precision;
ai->play.channels = fmt->channels;
ai->play.sample_rate = fmt->sample_rate;
ai->mode |= AUMODE_PLAY;
}
if (sc->sc_rmixer) {
audio_format2_t *fmt = &sc->sc_rmixer->track_fmt;
ai->record.encoding = fmt->encoding;
ai->record.precision = fmt->precision;
ai->record.channels = fmt->channels;
ai->record.sample_rate = fmt->sample_rate;
ai->mode |= AUMODE_RECORD;
}
}
/*
* audio_info details:
*
* ai.{play,record}.sample_rate (R/W)
* ai.{play,record}.encoding (R/W)
* ai.{play,record}.precision (R/W)
* ai.{play,record}.channels (R/W)
* These specify the playback or recording format.
* Ignore members within an inactive track.
*
* ai.mode (R/W)
* It specifies the playback or recording mode, AUMODE_*.
* Currently, a mode change operation by ai.mode after opening is
* prohibited. In addition, AUMODE_PLAY_ALL no longer makes sense.
* However, it's possible to get or to set for backward compatibility.
*
* ai.{hiwat,lowat} (R/W)
* These specify the high water mark and low water mark for playback
* track. The unit is block.
*
* ai.{play,record}.gain (R/W)
* It specifies the HW mixer volume in 0-255.
* It is historical reason that the gain is connected to HW mixer.
*
* ai.{play,record}.balance (R/W)
* It specifies the left-right balance of HW mixer in 0-64.
* 32 means the center.
* It is historical reason that the balance is connected to HW mixer.
*
* ai.{play,record}.port (R/W)
* It specifies the input/output port of HW mixer.
*
* ai.monitor_gain (R/W)
* It specifies the recording monitor gain(?) of HW mixer.
*
* ai.{play,record}.pause (R/W)
* Non-zero means the track is paused.
*
* ai.play.seek (R/-)
* It indicates the number of bytes written but not processed.
* ai.record.seek (R/-)
* It indicates the number of bytes to be able to read.
*
* ai.{play,record}.avail_ports (R/-)
* Mixer info.
*
* ai.{play,record}.buffer_size (R/-)
* It indicates the buffer size in bytes. Internally it means usrbuf.
*
* ai.{play,record}.samples (R/-)
* It indicates the total number of bytes played or recorded.
*
* ai.{play,record}.eof (R/-)
* It indicates the number of times reached EOF(?).
*
* ai.{play,record}.error (R/-)
* Non-zero indicates overflow/underflow has occured.
*
* ai.{play,record}.waiting (R/-)
* Non-zero indicates that other process waits to open.
* It will never happen anymore.
*
* ai.{play,record}.open (R/-)
* Non-zero indicates the direction is opened by this process(?).
* XXX Is this better to indicate that "the device is opened by
* at least one process"?
*
* ai.{play,record}.active (R/-)
* Non-zero indicates that I/O is currently active.
*
* ai.blocksize (R/-)
* It indicates the block size in bytes.
* XXX The blocksize of playback and recording may be different.
*/
/*
* Pause consideration:
*
* Pausing/unpausing never affect [pr]mixer. This single rule makes
* operation simple. Note that playback and recording are asymmetric.
*
* For playback,
* 1. Any playback open doesn't start pmixer regardless of initial pause
* state of this track.
* 2. The first write access among playback tracks only starts pmixer
* regardless of this track's pause state.
* 3. Even a pause of the last playback track doesn't stop pmixer.
* 4. The last close of all playback tracks only stops pmixer.
*
* For recording,
* 1. The first recording open only starts rmixer regardless of initial
* pause state of this track.
* 2. Even a pause of the last track doesn't stop rmixer.
* 3. The last close of all recording tracks only stops rmixer.
*/
/*
* Set both track's parameters within a file depending on ai.
* Update sc_sound_[pr]* if set.
* Must be called with sc_exlock held and without sc_lock held.
*/
static int
audio_file_setinfo(struct audio_softc *sc, audio_file_t *file,
const struct audio_info *ai)
{
const struct audio_prinfo *pi;
const struct audio_prinfo *ri;
audio_track_t *ptrack;
audio_track_t *rtrack;
audio_format2_t pfmt;
audio_format2_t rfmt;
int pchanges;
int rchanges;
int mode;
struct audio_info saved_ai;
audio_format2_t saved_pfmt;
audio_format2_t saved_rfmt;
int error;
KASSERT(sc->sc_exlock);
pi = &ai->play;
ri = &ai->record;
pchanges = 0;
rchanges = 0;
ptrack = file->ptrack;
rtrack = file->rtrack;
#if defined(AUDIO_DEBUG)
if (audiodebug >= 2) {
char buf[256];
char p[64];
int buflen;
int plen;
#define SPRINTF(var, fmt...) do { \
var##len += snprintf(var + var##len, sizeof(var) - var##len, fmt); \
} while (0)
buflen = 0;
plen = 0;
if (SPECIFIED(pi->encoding))
SPRINTF(p, "/%s", audio_encoding_name(pi->encoding));
if (SPECIFIED(pi->precision))
SPRINTF(p, "/%dbit", pi->precision);
if (SPECIFIED(pi->channels))
SPRINTF(p, "/%dch", pi->channels);
if (SPECIFIED(pi->sample_rate))
SPRINTF(p, "/%dHz", pi->sample_rate);
if (plen > 0)
SPRINTF(buf, ",play.param=%s", p + 1);
plen = 0;
if (SPECIFIED(ri->encoding))
SPRINTF(p, "/%s", audio_encoding_name(ri->encoding));
if (SPECIFIED(ri->precision))
SPRINTF(p, "/%dbit", ri->precision);
if (SPECIFIED(ri->channels))
SPRINTF(p, "/%dch", ri->channels);
if (SPECIFIED(ri->sample_rate))
SPRINTF(p, "/%dHz", ri->sample_rate);
if (plen > 0)
SPRINTF(buf, ",record.param=%s", p + 1);
if (SPECIFIED(ai->mode))
SPRINTF(buf, ",mode=%d", ai->mode);
if (SPECIFIED(ai->hiwat))
SPRINTF(buf, ",hiwat=%d", ai->hiwat);
if (SPECIFIED(ai->lowat))
SPRINTF(buf, ",lowat=%d", ai->lowat);
if (SPECIFIED(ai->play.gain))
SPRINTF(buf, ",play.gain=%d", ai->play.gain);
if (SPECIFIED(ai->record.gain))
SPRINTF(buf, ",record.gain=%d", ai->record.gain);
if (SPECIFIED_CH(ai->play.balance))
SPRINTF(buf, ",play.balance=%d", ai->play.balance);
if (SPECIFIED_CH(ai->record.balance))
SPRINTF(buf, ",record.balance=%d", ai->record.balance);
if (SPECIFIED(ai->play.port))
SPRINTF(buf, ",play.port=%d", ai->play.port);
if (SPECIFIED(ai->record.port))
SPRINTF(buf, ",record.port=%d", ai->record.port);
if (SPECIFIED(ai->monitor_gain))
SPRINTF(buf, ",monitor_gain=%d", ai->monitor_gain);
if (SPECIFIED_CH(ai->play.pause))
SPRINTF(buf, ",play.pause=%d", ai->play.pause);
if (SPECIFIED_CH(ai->record.pause))
SPRINTF(buf, ",record.pause=%d", ai->record.pause);
if (buflen > 0)
TRACE(2, "specified %s", buf + 1);
}
#endif
AUDIO_INITINFO(&saved_ai);
/* XXX shut up gcc */
memset(&saved_pfmt, 0, sizeof(saved_pfmt));
memset(&saved_rfmt, 0, sizeof(saved_rfmt));
/*
* Set default value and save current parameters.
* For backward compatibility, use sticky parameters for nonexistent
* track.
*/
if (ptrack) {
pfmt = ptrack->usrbuf.fmt;
saved_pfmt = ptrack->usrbuf.fmt;
saved_ai.play.pause = ptrack->is_pause;
} else {
pfmt = sc->sc_sound_pparams;
}
if (rtrack) {
rfmt = rtrack->usrbuf.fmt;
saved_rfmt = rtrack->usrbuf.fmt;
saved_ai.record.pause = rtrack->is_pause;
} else {
rfmt = sc->sc_sound_rparams;
}
saved_ai.mode = file->mode;
/*
* Overwrite if specified.
*/
mode = file->mode;
if (SPECIFIED(ai->mode)) {
/*
* Setting ai->mode no longer does anything because it's
* prohibited to change playback/recording mode after open
* and AUMODE_PLAY_ALL is obsoleted. However, it still
* keeps the state of AUMODE_PLAY_ALL itself for backward
* compatibility.
* In the internal, only file->mode has the state of
* AUMODE_PLAY_ALL flag and track->mode in both track does
* not have.
*/
if ((file->mode & AUMODE_PLAY)) {
mode = (file->mode & (AUMODE_PLAY | AUMODE_RECORD))
| (ai->mode & AUMODE_PLAY_ALL);
}
}
pchanges = audio_track_setinfo_check(ptrack, &pfmt, pi);
if (pchanges == -1) {
#if defined(AUDIO_DEBUG)
TRACEF(1, file, "check play.params failed: "
"%s %ubit %uch %uHz",
audio_encoding_name(pi->encoding),
pi->precision,
pi->channels,
pi->sample_rate);
#endif
return EINVAL;
}
rchanges = audio_track_setinfo_check(rtrack, &rfmt, ri);
if (rchanges == -1) {
#if defined(AUDIO_DEBUG)
TRACEF(1, file, "check record.params failed: "
"%s %ubit %uch %uHz",
audio_encoding_name(ri->encoding),
ri->precision,
ri->channels,
ri->sample_rate);
#endif
return EINVAL;
}
if (SPECIFIED(ai->mode)) {
pchanges = 1;
rchanges = 1;
}
/*
* Even when setting either one of playback and recording,
* both track must be halted.
*/
if (pchanges || rchanges) {
audio_file_clear(sc, file);
#if defined(AUDIO_DEBUG)
char nbuf[16];
char fmtbuf[64];
if (pchanges) {
if (ptrack) {
snprintf(nbuf, sizeof(nbuf), "%d", ptrack->id);
} else {
snprintf(nbuf, sizeof(nbuf), "-");
}
audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &pfmt);
DPRINTF(1, "audio track#%s play mode: %s\n",
nbuf, fmtbuf);
}
if (rchanges) {
if (rtrack) {
snprintf(nbuf, sizeof(nbuf), "%d", rtrack->id);
} else {
snprintf(nbuf, sizeof(nbuf), "-");
}
audio_format2_tostr(fmtbuf, sizeof(fmtbuf), &rfmt);
DPRINTF(1, "audio track#%s rec mode: %s\n",
nbuf, fmtbuf);
}
#endif
}
/* Set mixer parameters */
mutex_enter(sc->sc_lock);
error = audio_hw_setinfo(sc, ai, &saved_ai);
mutex_exit(sc->sc_lock);
if (error)
goto abort1;
/*
* Set to track and update sticky parameters.
*/
error = 0;
file->mode = mode;
if (SPECIFIED_CH(pi->pause)) {
if (ptrack)
ptrack->is_pause = pi->pause;
sc->sc_sound_ppause = pi->pause;
}
if (pchanges) {
if (ptrack) {
audio_track_lock_enter(ptrack);
error = audio_track_set_format(ptrack, &pfmt);
audio_track_lock_exit(ptrack);
if (error) {
TRACET(1, ptrack, "set play.params failed");
goto abort2;
}
}
sc->sc_sound_pparams = pfmt;
}
/* Change water marks after initializing the buffers. */
if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) {
if (ptrack)
audio_track_setinfo_water(ptrack, ai);
}
if (SPECIFIED_CH(ri->pause)) {
if (rtrack)
rtrack->is_pause = ri->pause;
sc->sc_sound_rpause = ri->pause;
}
if (rchanges) {
if (rtrack) {
audio_track_lock_enter(rtrack);
error = audio_track_set_format(rtrack, &rfmt);
audio_track_lock_exit(rtrack);
if (error) {
TRACET(1, rtrack, "set record.params failed");
goto abort3;
}
}
sc->sc_sound_rparams = rfmt;
}
return 0;
/* Rollback */
abort3:
if (error != ENOMEM) {
rtrack->is_pause = saved_ai.record.pause;
audio_track_lock_enter(rtrack);
audio_track_set_format(rtrack, &saved_rfmt);
audio_track_lock_exit(rtrack);
}
sc->sc_sound_rpause = saved_ai.record.pause;
sc->sc_sound_rparams = saved_rfmt;
abort2:
if (ptrack && error != ENOMEM) {
ptrack->is_pause = saved_ai.play.pause;
audio_track_lock_enter(ptrack);
audio_track_set_format(ptrack, &saved_pfmt);
audio_track_lock_exit(ptrack);
}
sc->sc_sound_ppause = saved_ai.play.pause;
sc->sc_sound_pparams = saved_pfmt;
file->mode = saved_ai.mode;
abort1:
mutex_enter(sc->sc_lock);
audio_hw_setinfo(sc, &saved_ai, NULL);
mutex_exit(sc->sc_lock);
return error;
}
/*
* Write SPECIFIED() parameters within info back to fmt.
* Note that track can be NULL here.
* Return value of 1 indicates that fmt is modified.
* Return value of 0 indicates that fmt is not modified.
* Return value of -1 indicates that error EINVAL has occurred.
*/
static int
audio_track_setinfo_check(audio_track_t *track,
audio_format2_t *fmt, const struct audio_prinfo *info)
{
const audio_format2_t *hwfmt;
int changes;
changes = 0;
if (SPECIFIED(info->sample_rate)) {
if (info->sample_rate < AUDIO_MIN_FREQUENCY)
return -1;
if (info->sample_rate > AUDIO_MAX_FREQUENCY)
return -1;
fmt->sample_rate = info->sample_rate;
changes = 1;
}
if (SPECIFIED(info->encoding)) {
fmt->encoding = info->encoding;
changes = 1;
}
if (SPECIFIED(info->precision)) {
fmt->precision = info->precision;
/* we don't have API to specify stride */
fmt->stride = info->precision;
changes = 1;
}
if (SPECIFIED(info->channels)) {
/*
* We can convert between monaural and stereo each other.
* We can reduce than the number of channels that the hardware
* supports.
*/
if (info->channels > 2) {
if (track) {
hwfmt = &track->mixer->hwbuf.fmt;
if (info->channels > hwfmt->channels)
return -1;
} else {
/*
* This should never happen.
* If track == NULL, channels should be <= 2.
*/
return -1;
}
}
fmt->channels = info->channels;
changes = 1;
}
if (changes) {
if (audio_check_params(fmt) != 0)
return -1;
}
return changes;
}
/*
* Change water marks for playback track if specfied.
*/
static void
audio_track_setinfo_water(audio_track_t *track, const struct audio_info *ai)
{
u_int blks;
u_int maxblks;
u_int blksize;
KASSERT(audio_track_is_playback(track));
blksize = track->usrbuf_blksize;
maxblks = track->usrbuf.capacity / blksize;
if (SPECIFIED(ai->hiwat)) {
blks = ai->hiwat;
if (blks > maxblks)
blks = maxblks;
if (blks < 2)
blks = 2;
track->usrbuf_usedhigh = blks * blksize;
}
if (SPECIFIED(ai->lowat)) {
blks = ai->lowat;
if (blks > maxblks - 1)
blks = maxblks - 1;
track->usrbuf_usedlow = blks * blksize;
}
if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) {
if (track->usrbuf_usedlow > track->usrbuf_usedhigh - blksize) {
track->usrbuf_usedlow = track->usrbuf_usedhigh -
blksize;
}
}
}
/*
* Set hardware part of *newai.
* The parameters handled here are *.port, *.gain, *.balance and monitor_gain.
* If oldai is specified, previous parameters are stored.
* This function itself does not roll back if error occurred.
* Must be called with sc_lock && sc_exlock held.
*/
static int
audio_hw_setinfo(struct audio_softc *sc, const struct audio_info *newai,
struct audio_info *oldai)
{
const struct audio_prinfo *newpi;
const struct audio_prinfo *newri;
struct audio_prinfo *oldpi;
struct audio_prinfo *oldri;
u_int pgain;
u_int rgain;
u_char pbalance;
u_char rbalance;
int error;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
/* XXX shut up gcc */
oldpi = NULL;
oldri = NULL;
newpi = &newai->play;
newri = &newai->record;
if (oldai) {
oldpi = &oldai->play;
oldri = &oldai->record;
}
error = 0;
/*
* It looks like unnecessary to halt HW mixers to set HW mixers.
* mixer_ioctl(MIXER_WRITE) also doesn't halt.
*/
if (SPECIFIED(newpi->port)) {
if (oldai)
oldpi->port = au_get_port(sc, &sc->sc_outports);
error = au_set_port(sc, &sc->sc_outports, newpi->port);
if (error) {
audio_printf(sc,
"setting play.port=%d failed: errno=%d\n",
newpi->port, error);
goto abort;
}
}
if (SPECIFIED(newri->port)) {
if (oldai)
oldri->port = au_get_port(sc, &sc->sc_inports);
error = au_set_port(sc, &sc->sc_inports, newri->port);
if (error) {
audio_printf(sc,
"setting record.port=%d failed: errno=%d\n",
newri->port, error);
goto abort;
}
}
/* Backup play.{gain,balance} */
if (SPECIFIED(newpi->gain) || SPECIFIED_CH(newpi->balance)) {
au_get_gain(sc, &sc->sc_outports, &pgain, &pbalance);
if (oldai) {
oldpi->gain = pgain;
oldpi->balance = pbalance;
}
}
/* Backup record.{gain,balance} */
if (SPECIFIED(newri->gain) || SPECIFIED_CH(newri->balance)) {
au_get_gain(sc, &sc->sc_inports, &rgain, &rbalance);
if (oldai) {
oldri->gain = rgain;
oldri->balance = rbalance;
}
}
if (SPECIFIED(newpi->gain)) {
error = au_set_gain(sc, &sc->sc_outports,
newpi->gain, pbalance);
if (error) {
audio_printf(sc,
"setting play.gain=%d failed: errno=%d\n",
newpi->gain, error);
goto abort;
}
}
if (SPECIFIED(newri->gain)) {
error = au_set_gain(sc, &sc->sc_inports,
newri->gain, rbalance);
if (error) {
audio_printf(sc,
"setting record.gain=%d failed: errno=%d\n",
newri->gain, error);
goto abort;
}
}
if (SPECIFIED_CH(newpi->balance)) {
error = au_set_gain(sc, &sc->sc_outports,
pgain, newpi->balance);
if (error) {
audio_printf(sc,
"setting play.balance=%d failed: errno=%d\n",
newpi->balance, error);
goto abort;
}
}
if (SPECIFIED_CH(newri->balance)) {
error = au_set_gain(sc, &sc->sc_inports,
rgain, newri->balance);
if (error) {
audio_printf(sc,
"setting record.balance=%d failed: errno=%d\n",
newri->balance, error);
goto abort;
}
}
if (SPECIFIED(newai->monitor_gain) && sc->sc_monitor_port != -1) {
if (oldai)
oldai->monitor_gain = au_get_monitor_gain(sc);
error = au_set_monitor_gain(sc, newai->monitor_gain);
if (error) {
audio_printf(sc,
"setting monitor_gain=%d failed: errno=%d\n",
newai->monitor_gain, error);
goto abort;
}
}
/* XXX TODO */
/* sc->sc_ai = *ai; */
error = 0;
abort:
return error;
}
/*
* Setup the hardware with mixer format phwfmt, rhwfmt.
* The arguments have following restrictions:
* - setmode is the direction you want to set, AUMODE_PLAY or AUMODE_RECORD,
* or both.
* - phwfmt and rhwfmt must not be NULL regardless of setmode.
* - On non-independent devices, phwfmt and rhwfmt must have the same
* parameters.
* - pfil and rfil must be zero-filled.
* If successful,
* - pfil, rfil will be filled with filter information specified by the
* hardware driver if necessary.
* and then returns 0. Otherwise returns errno.
* Must be called without sc_lock held.
*/
static int
audio_hw_set_format(struct audio_softc *sc, int setmode,
const audio_format2_t *phwfmt, const audio_format2_t *rhwfmt,
audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
{
audio_params_t pp, rp;
int error;
KASSERT(phwfmt != NULL);
KASSERT(rhwfmt != NULL);
pp = format2_to_params(phwfmt);
rp = format2_to_params(rhwfmt);
mutex_enter(sc->sc_lock);
error = sc->hw_if->set_format(sc->hw_hdl, setmode,
&pp, &rp, pfil, rfil);
if (error) {
mutex_exit(sc->sc_lock);
audio_printf(sc, "set_format failed: errno=%d\n", error);
return error;
}
if (sc->hw_if->commit_settings) {
error = sc->hw_if->commit_settings(sc->hw_hdl);
if (error) {
mutex_exit(sc->sc_lock);
audio_printf(sc,
"commit_settings failed: errno=%d\n", error);
return error;
}
}
mutex_exit(sc->sc_lock);
return 0;
}
/*
* Fill audio_info structure. If need_mixerinfo is true, it will also
* fill the hardware mixer information.
* Must be called with sc_exlock held and without sc_lock held.
*/
static int
audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int need_mixerinfo,
audio_file_t *file)
{
struct audio_prinfo *ri, *pi;
audio_track_t *track;
audio_track_t *ptrack;
audio_track_t *rtrack;
int gain;
KASSERT(sc->sc_exlock);
ri = &ai->record;
pi = &ai->play;
ptrack = file->ptrack;
rtrack = file->rtrack;
memset(ai, 0, sizeof(*ai));
if (ptrack) {
pi->sample_rate = ptrack->usrbuf.fmt.sample_rate;
pi->channels = ptrack->usrbuf.fmt.channels;
pi->precision = ptrack->usrbuf.fmt.precision;
pi->encoding = ptrack->usrbuf.fmt.encoding;
pi->pause = ptrack->is_pause;
} else {
/* Use sticky parameters if the track is not available. */
pi->sample_rate = sc->sc_sound_pparams.sample_rate;
pi->channels = sc->sc_sound_pparams.channels;
pi->precision = sc->sc_sound_pparams.precision;
pi->encoding = sc->sc_sound_pparams.encoding;
pi->pause = sc->sc_sound_ppause;
}
if (rtrack) {
ri->sample_rate = rtrack->usrbuf.fmt.sample_rate;
ri->channels = rtrack->usrbuf.fmt.channels;
ri->precision = rtrack->usrbuf.fmt.precision;
ri->encoding = rtrack->usrbuf.fmt.encoding;
ri->pause = rtrack->is_pause;
} else {
/* Use sticky parameters if the track is not available. */
ri->sample_rate = sc->sc_sound_rparams.sample_rate;
ri->channels = sc->sc_sound_rparams.channels;
ri->precision = sc->sc_sound_rparams.precision;
ri->encoding = sc->sc_sound_rparams.encoding;
ri->pause = sc->sc_sound_rpause;
}
if (ptrack) {
pi->seek = ptrack->usrbuf.used;
pi->samples = ptrack->usrbuf_stamp;
pi->eof = ptrack->eofcounter;
pi->error = (ptrack->dropframes != 0) ? 1 : 0;
pi->open = 1;
pi->buffer_size = ptrack->usrbuf.capacity;
}
pi->waiting = 0; /* open never hangs */
pi->active = sc->sc_pbusy;
if (rtrack) {
ri->seek = rtrack->usrbuf.used;
ri->samples = rtrack->usrbuf_stamp;
ri->eof = 0;
ri->error = (rtrack->dropframes != 0) ? 1 : 0;
ri->open = 1;
ri->buffer_size = rtrack->usrbuf.capacity;
}
ri->waiting = 0; /* open never hangs */
ri->active = sc->sc_rbusy;
/*
* XXX There may be different number of channels between playback
* and recording, so that blocksize also may be different.
* But struct audio_info has an united blocksize...
* Here, I use play info precedencely if ptrack is available,
* otherwise record info.
*
* XXX hiwat/lowat is a playback-only parameter. What should I
* return for a record-only descriptor?
*/
track = ptrack ? ptrack : rtrack;
if (track) {
ai->blocksize = track->usrbuf_blksize;
ai->hiwat = track->usrbuf_usedhigh / track->usrbuf_blksize;
ai->lowat = track->usrbuf_usedlow / track->usrbuf_blksize;
}
ai->mode = file->mode;
/*
* For backward compatibility, we have to pad these five fields
* a fake non-zero value even if there are no tracks.
*/
if (ptrack == NULL)
pi->buffer_size = 65536;
if (rtrack == NULL)
ri->buffer_size = 65536;
if (ptrack == NULL && rtrack == NULL) {
ai->blocksize = 2048;
ai->hiwat = ai->play.buffer_size / ai->blocksize;
ai->lowat = ai->hiwat * 3 / 4;
}
if (need_mixerinfo) {
mutex_enter(sc->sc_lock);
pi->port = au_get_port(sc, &sc->sc_outports);
ri->port = au_get_port(sc, &sc->sc_inports);
pi->avail_ports = sc->sc_outports.allports;
ri->avail_ports = sc->sc_inports.allports;
au_get_gain(sc, &sc->sc_outports, &pi->gain, &pi->balance);
au_get_gain(sc, &sc->sc_inports, &ri->gain, &ri->balance);
if (sc->sc_monitor_port != -1) {
gain = au_get_monitor_gain(sc);
if (gain != -1)
ai->monitor_gain = gain;
}
mutex_exit(sc->sc_lock);
}
return 0;
}
/*
* Return true if playback is configured.
* This function can be used after audioattach.
*/
static bool
audio_can_playback(struct audio_softc *sc)
{
return (sc->sc_pmixer != NULL);
}
/*
* Return true if recording is configured.
* This function can be used after audioattach.
*/
static bool
audio_can_capture(struct audio_softc *sc)
{
return (sc->sc_rmixer != NULL);
}
/*
* Get the afp->index'th item from the valid one of format[].
* If found, stores it to afp->fmt and returns 0. Otherwise return EINVAL.
*
* This is common routines for query_format.
* If your hardware driver has struct audio_format[], the simplest case
* you can write your query_format interface as follows:
*
* struct audio_format foo_format[] = { ... };
*
* int
* foo_query_format(void *hdl, audio_format_query_t *afp)
* {
* return audio_query_format(foo_format, __arraycount(foo_format), afp);
* }
*/
int
audio_query_format(const struct audio_format *format, int nformats,
audio_format_query_t *afp)
{
const struct audio_format *f;
int idx;
int i;
idx = 0;
for (i = 0; i < nformats; i++) {
f = &format[i];
if (!AUFMT_IS_VALID(f))
continue;
if (afp->index == idx) {
afp->fmt = *f;
return 0;
}
idx++;
}
return EINVAL;
}
/*
* This function is provided for the hardware driver's set_format() to
* find index matches with 'param' from array of audio_format_t 'formats'.
* 'mode' is either of AUMODE_PLAY or AUMODE_RECORD.
* It returns the matched index and never fails. Because param passed to
* set_format() is selected from query_format().
* This function will be an alternative to auconv_set_converter() to
* find index.
*/
int
audio_indexof_format(const struct audio_format *formats, int nformats,
int mode, const audio_params_t *param)
{
const struct audio_format *f;
int index;
int j;
for (index = 0; index < nformats; index++) {
f = &formats[index];
if (!AUFMT_IS_VALID(f))
continue;
if ((f->mode & mode) == 0)
continue;
if (f->encoding != param->encoding)
continue;
if (f->validbits != param->precision)
continue;
if (f->channels != param->channels)
continue;
if (f->frequency_type == 0) {
if (param->sample_rate < f->frequency[0] ||
param->sample_rate > f->frequency[1])
continue;
} else {
for (j = 0; j < f->frequency_type; j++) {
if (param->sample_rate == f->frequency[j])
break;
}
if (j == f->frequency_type)
continue;
}
/* Then, matched */
return index;
}
/* Not matched. This should not be happened. */
panic("%s: cannot find matched format\n", __func__);
}
/*
* Get or set hardware blocksize in msec.
* XXX It's for debug.
*/
static int
audio_sysctl_blk_ms(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
audio_format2_t phwfmt;
audio_format2_t rhwfmt;
audio_filter_reg_t pfil;
audio_filter_reg_t rfil;
int t;
int old_blk_ms;
int mode;
int error;
node = *rnode;
sc = node.sysctl_data;
error = audio_exlock_enter(sc);
if (error)
return error;
old_blk_ms = sc->sc_blk_ms;
t = old_blk_ms;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
goto abort;
if (t < 0) {
error = EINVAL;
goto abort;
}
if (sc->sc_popens + sc->sc_ropens > 0) {
error = EBUSY;
goto abort;
}
sc->sc_blk_ms = t;
mode = 0;
if (sc->sc_pmixer) {
mode |= AUMODE_PLAY;
phwfmt = sc->sc_pmixer->hwbuf.fmt;
}
if (sc->sc_rmixer) {
mode |= AUMODE_RECORD;
rhwfmt = sc->sc_rmixer->hwbuf.fmt;
}
/* re-init hardware */
memset(&pfil, 0, sizeof(pfil));
memset(&rfil, 0, sizeof(rfil));
error = audio_hw_set_format(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (error) {
goto abort;
}
/* re-init track mixer */
error = audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
if (error) {
/* Rollback */
sc->sc_blk_ms = old_blk_ms;
audio_mixers_init(sc, mode, &phwfmt, &rhwfmt, &pfil, &rfil);
goto abort;
}
error = 0;
abort:
audio_exlock_exit(sc);
return error;
}
/*
* Get or set multiuser mode.
*/
static int
audio_sysctl_multiuser(SYSCTLFN_ARGS)
{
struct sysctlnode node;
struct audio_softc *sc;
bool t;
int error;
node = *rnode;
sc = node.sysctl_data;
error = audio_exlock_enter(sc);
if (error)
return error;
t = sc->sc_multiuser;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
goto abort;
sc->sc_multiuser = t;
error = 0;
abort:
audio_exlock_exit(sc);
return error;
}
#if defined(AUDIO_DEBUG)
/*
* Get or set debug verbose level. (0..4)
* XXX It's for debug.
* XXX It is not separated per device.
*/
static int
audio_sysctl_debug(SYSCTLFN_ARGS)
{
struct sysctlnode node;
int t;
int error;
node = *rnode;
t = audiodebug;
node.sysctl_data = &t;
error = sysctl_lookup(SYSCTLFN_CALL(&node));
if (error || newp == NULL)
return error;
if (t < 0 || t > 4)
return EINVAL;
audiodebug = t;
printf("audio: audiodebug = %d\n", audiodebug);
return 0;
}
#endif /* AUDIO_DEBUG */
#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->hw_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->hw_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);
int error;
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
sc->sc_suspending = true;
audio_mixer_capture(sc);
if (sc->sc_pbusy) {
audio_pmixer_halt(sc);
/* Reuse this as need-to-restart flag while suspending */
sc->sc_pbusy = true;
}
if (sc->sc_rbusy) {
audio_rmixer_halt(sc);
/* Reuse this as need-to-restart flag while suspending */
sc->sc_rbusy = true;
}
#ifdef AUDIO_PM_IDLE
callout_halt(&sc->sc_idle_counter, sc->sc_lock);
#endif
audio_exlock_mutex_exit(sc);
return true;
}
static bool
audio_resume(device_t dv, const pmf_qual_t *qual)
{
struct audio_softc *sc = device_private(dv);
struct audio_info ai;
int error;
error = audio_exlock_mutex_enter(sc);
if (error)
return error;
sc->sc_suspending = false;
audio_mixer_restore(sc);
/* XXX ? */
AUDIO_INITINFO(&ai);
audio_hw_setinfo(sc, &ai, NULL);
/*
* During from suspend to resume here, sc_[pr]busy is used as
* need-to-restart flag temporarily. After this point,
* sc_[pr]busy is returned to its original usage (busy flag).
* And note that sc_[pr]busy must be false to call [pr]mixer_start().
*/
if (sc->sc_pbusy) {
/* pmixer_start() requires pbusy is false */
sc->sc_pbusy = false;
audio_pmixer_start(sc, true);
}
if (sc->sc_rbusy) {
/* rmixer_start() requires rbusy is false */
sc->sc_rbusy = false;
audio_rmixer_start(sc);
}
audio_exlock_mutex_exit(sc);
return true;
}
#if defined(AUDIO_DEBUG)
static void
audio_format2_tostr(char *buf, size_t bufsize, const audio_format2_t *fmt)
{
int n;
n = 0;
n += snprintf(buf + n, bufsize - n, "%s",
audio_encoding_name(fmt->encoding));
if (fmt->precision == fmt->stride) {
n += snprintf(buf + n, bufsize - n, " %dbit", fmt->precision);
} else {
n += snprintf(buf + n, bufsize - n, " %d/%dbit",
fmt->precision, fmt->stride);
}
snprintf(buf + n, bufsize - n, " %uch %uHz",
fmt->channels, fmt->sample_rate);
}
#endif
#if defined(AUDIO_DEBUG)
static void
audio_print_format2(const char *s, const audio_format2_t *fmt)
{
char fmtstr[64];
audio_format2_tostr(fmtstr, sizeof(fmtstr), fmt);
printf("%s %s\n", s, fmtstr);
}
#endif
#ifdef DIAGNOSTIC
void
audio_diagnostic_format2(const char *where, const audio_format2_t *fmt)
{
KASSERTMSG(fmt, "called from %s", where);
/* XXX MSM6258 vs(4) only has 4bit stride format. */
if (fmt->encoding == AUDIO_ENCODING_ADPCM) {
KASSERTMSG(fmt->stride == 4 || fmt->stride == 8,
"called from %s: fmt->stride=%d", where, fmt->stride);
} else {
KASSERTMSG(fmt->stride % NBBY == 0,
"called from %s: fmt->stride=%d", where, fmt->stride);
}
KASSERTMSG(fmt->precision <= fmt->stride,
"called from %s: fmt->precision=%d fmt->stride=%d",
where, fmt->precision, fmt->stride);
KASSERTMSG(1 <= fmt->channels && fmt->channels <= AUDIO_MAX_CHANNELS,
"called from %s: fmt->channels=%d", where, fmt->channels);
/* XXX No check for encodings? */
}
void
audio_diagnostic_filter_arg(const char *where, const audio_filter_arg_t *arg)
{
KASSERT(arg != NULL);
KASSERT(arg->src != NULL);
KASSERT(arg->dst != NULL);
audio_diagnostic_format2(where, arg->srcfmt);
audio_diagnostic_format2(where, arg->dstfmt);
KASSERT(arg->count > 0);
}
void
audio_diagnostic_ring(const char *where, const audio_ring_t *ring)
{
KASSERTMSG(ring, "called from %s", where);
audio_diagnostic_format2(where, &ring->fmt);
KASSERTMSG(0 <= ring->capacity && ring->capacity < INT_MAX / 2,
"called from %s: ring->capacity=%d", where, ring->capacity);
KASSERTMSG(0 <= ring->used && ring->used <= ring->capacity,
"called from %s: ring->used=%d ring->capacity=%d",
where, ring->used, ring->capacity);
if (ring->capacity == 0) {
KASSERTMSG(ring->mem == NULL,
"called from %s: capacity == 0 but mem != NULL", where);
} else {
KASSERTMSG(ring->mem != NULL,
"called from %s: capacity != 0 but mem == NULL", where);
KASSERTMSG(0 <= ring->head && ring->head < ring->capacity,
"called from %s: ring->head=%d ring->capacity=%d",
where, ring->head, ring->capacity);
}
}
#endif /* DIAGNOSTIC */
/*
* Mixer driver
*/
/*
* Must be called without sc_lock held.
*/
int
mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct lwp *l)
{
struct file *fp;
audio_file_t *af;
int error, fd;
TRACE(1, "flags=0x%x", flags);
error = fd_allocfile(&fp, &fd);
if (error)
return error;
af = kmem_zalloc(sizeof(*af), KM_SLEEP);
af->sc = sc;
af->dev = dev;
error = fd_clone(fp, fd, flags, &audio_fileops, af);
KASSERT(error == EMOVEFD);
return error;
}
/*
* Add a process to those to be signalled on mixer activity.
* If the process has already been added, do nothing.
* Must be called with sc_exlock held and without sc_lock held.
*/
static void
mixer_async_add(struct audio_softc *sc, pid_t pid)
{
int i;
KASSERT(sc->sc_exlock);
/* If already exists, returns without doing anything. */
for (i = 0; i < sc->sc_am_used; i++) {
if (sc->sc_am[i] == pid)
return;
}
/* Extend array if necessary. */
if (sc->sc_am_used >= sc->sc_am_capacity) {
sc->sc_am_capacity += AM_CAPACITY;
sc->sc_am = kern_realloc(sc->sc_am,
sc->sc_am_capacity * sizeof(pid_t), M_WAITOK);
TRACE(2, "realloc am_capacity=%d", sc->sc_am_capacity);
}
TRACE(2, "am[%d]=%d", sc->sc_am_used, (int)pid);
sc->sc_am[sc->sc_am_used++] = pid;
}
/*
* Remove a process from those to be signalled on mixer activity.
* If the process has not been added, do nothing.
* Must be called with sc_exlock held and without sc_lock held.
*/
static void
mixer_async_remove(struct audio_softc *sc, pid_t pid)
{
int i;
KASSERT(sc->sc_exlock);
for (i = 0; i < sc->sc_am_used; i++) {
if (sc->sc_am[i] == pid) {
sc->sc_am[i] = sc->sc_am[--sc->sc_am_used];
TRACE(2, "am[%d](%d) removed, used=%d",
i, (int)pid, sc->sc_am_used);
/* Empty array if no longer necessary. */
if (sc->sc_am_used == 0) {
kern_free(sc->sc_am);
sc->sc_am = NULL;
sc->sc_am_capacity = 0;
TRACE(2, "released");
}
return;
}
}
}
/*
* Signal all processes waiting for the mixer.
* Must be called with sc_exlock held.
*/
static void
mixer_signal(struct audio_softc *sc)
{
proc_t *p;
int i;
KASSERT(sc->sc_exlock);
for (i = 0; i < sc->sc_am_used; i++) {
mutex_enter(&proc_lock);
p = proc_find(sc->sc_am[i]);
if (p)
psignal(p, SIGIO);
mutex_exit(&proc_lock);
}
}
/*
* Close a mixer device
*/
int
mixer_close(struct audio_softc *sc, audio_file_t *file)
{
int error;
error = audio_exlock_enter(sc);
if (error)
return error;
TRACE(1, "called");
mixer_async_remove(sc, curproc->p_pid);
audio_exlock_exit(sc);
return 0;
}
/*
* Must be called without sc_lock nor sc_exlock held.
*/
int
mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag,
struct lwp *l)
{
mixer_devinfo_t *mi;
mixer_ctrl_t *mc;
int error;
TRACE(2, "(%lu,'%c',%lu)",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff);
error = EINVAL;
/* we can return cached values if we are sleeping */
if (cmd != AUDIO_MIXER_READ) {
mutex_enter(sc->sc_lock);
device_active(sc->sc_dev, DVA_SYSTEM);
mutex_exit(sc->sc_lock);
}
switch (cmd) {
case FIOASYNC:
error = audio_exlock_enter(sc);
if (error)
break;
if (*(int *)addr) {
mixer_async_add(sc, curproc->p_pid);
} else {
mixer_async_remove(sc, curproc->p_pid);
}
audio_exlock_exit(sc);
break;
case AUDIO_GETDEV:
TRACE(2, "AUDIO_GETDEV");
mutex_enter(sc->sc_lock);
error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr);
mutex_exit(sc->sc_lock);
break;
case AUDIO_MIXER_DEVINFO:
TRACE(2, "AUDIO_MIXER_DEVINFO");
mi = (mixer_devinfo_t *)addr;
mi->un.v.delta = 0; /* default */
mutex_enter(sc->sc_lock);
error = audio_query_devinfo(sc, mi);
mutex_exit(sc->sc_lock);
break;
case AUDIO_MIXER_READ:
TRACE(2, "AUDIO_MIXER_READ");
mc = (mixer_ctrl_t *)addr;
error = audio_exlock_mutex_enter(sc);
if (error)
break;
if (device_is_active(sc->hw_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;
}
audio_exlock_mutex_exit(sc);
break;
case AUDIO_MIXER_WRITE:
TRACE(2, "AUDIO_MIXER_WRITE");
error = audio_exlock_mutex_enter(sc);
if (error)
break;
error = audio_set_port(sc, (mixer_ctrl_t *)addr);
if (error) {
audio_exlock_mutex_exit(sc);
break;
}
if (sc->hw_if->commit_settings) {
error = sc->hw_if->commit_settings(sc->hw_hdl);
if (error) {
audio_exlock_mutex_exit(sc);
break;
}
}
mutex_exit(sc->sc_lock);
mixer_signal(sc);
audio_exlock_exit(sc);
break;
default:
if (sc->hw_if->dev_ioctl) {
mutex_enter(sc->sc_lock);
error = sc->hw_if->dev_ioctl(sc->hw_hdl,
cmd, addr, flag, l);
mutex_exit(sc->sc_lock);
} else
error = EINVAL;
break;
}
TRACE(2, "(%lu,'%c',%lu) result %d",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff, error);
return error;
}
/*
* Must be called with sc_lock held.
*/
int
au_portof(struct audio_softc *sc, char *name, int class)
{
mixer_devinfo_t mi;
KASSERT(mutex_owned(sc->sc_lock));
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;
}
/*
* Must be called with sc_lock held.
*/
void
au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports,
mixer_devinfo_t *mi, const struct portname *tbl)
{
int i, j;
KASSERT(mutex_owned(sc->sc_lock));
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;
}
}
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
int
au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r)
{
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
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);
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
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));
KASSERT(sc->sc_exlock);
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;
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
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));
KASSERT(sc->sc_exlock);
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;
}
TRACE(2, "gain=%d balance=%d, l=%d r=%d", 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;
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
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));
KASSERT(sc->sc_exlock);
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;
}
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
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));
KASSERT(sc->sc_exlock);
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;
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
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));
KASSERT(sc->sc_exlock);
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;
}
/*
* It returns 0 if success, otherwise errno.
* Must be called only if sc->sc_monitor_port != -1.
* Must be called with sc_lock && sc_exlock held.
*/
static int
au_set_monitor_gain(struct audio_softc *sc, int monitor_gain)
{
mixer_ctrl_t ct;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
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] = monitor_gain;
return audio_set_port(sc, &ct);
}
/*
* It returns monitor gain if success, otherwise -1.
* Must be called only if sc->sc_monitor_port != -1.
* Must be called with sc_lock && sc_exlock held.
*/
static int
au_get_monitor_gain(struct audio_softc *sc)
{
mixer_ctrl_t ct;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
ct.dev = sc->sc_monitor_port;
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
if (audio_get_port(sc, &ct))
return -1;
return ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
static int
audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc)
{
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
return sc->hw_if->set_port(sc->hw_hdl, mc);
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
static int
audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc)
{
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
return sc->hw_if->get_port(sc->hw_hdl, mc);
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
static void
audio_mixer_capture(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
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;
}
/*
* Must be called with sc_lock && sc_exlock held.
*/
static void
audio_mixer_restore(struct audio_softc *sc)
{
mixer_devinfo_t mi;
mixer_ctrl_t *mc;
KASSERT(mutex_owned(sc->sc_lock));
KASSERT(sc->sc_exlock);
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;
}
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;
if (audio_exlock_mutex_enter(sc) != 0)
return;
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);
}
}
audio_exlock_mutex_exit(sc);
}
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;
if (audio_exlock_mutex_enter(sc) != 0)
return;
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);
}
}
audio_exlock_mutex_exit(sc);
}
static void
audio_volume_toggle(device_t dv)
{
struct audio_softc *sc = device_private(dv);
u_int gain, newgain;
u_char balance;
if (audio_exlock_mutex_enter(sc) != 0)
return;
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);
audio_exlock_mutex_exit(sc);
}
/*
* Must be called with sc_lock held.
*/
static int
audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di)
{
KASSERT(mutex_owned(sc->sc_lock));
return sc->hw_if->query_devinfo(sc->hw_hdl, di);
}
#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/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) */
#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;
switch (cmd) {
case MODULE_CMD_INIT:
/* XXX interrupt level? */
audio_psref_class = psref_class_create("audio", IPL_SOFTSERIAL);
#ifdef _MODULE
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);
}
#endif
break;
case MODULE_CMD_FINI:
#ifdef _MODULE
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);
#endif
psref_class_destroy(audio_psref_class);
break;
default:
error = ENOTTY;
break;
}
return error;
}
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