/* $NetBSD: audio.c,v 1.136.4.1 2024/03/26 18:09:21 martin 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. */ /* * Terminology: "sample", "channel", "frame", "block", "track": * * channel frame * | ........ * v : : \ * +------:------:------:- -+------+ : +------+-.. | * #0(L) |sample|sample|sample| .. |sample| : |sample| | * +------:------:------:- -+------+ : +------+-.. | * #1(R) |sample|sample|sample| .. |sample| : |sample| | * +------:------:------:- -+------+ : +------+-.. | track * : : : : | * +------:------:------:- -+------+ : +------+-.. | * |sample|sample|sample| .. |sample| : |sample| | * +------:------:------:- -+------+ : +------+-.. | * : : / * ........ * * \--------------------------------/ \--------.. * block * * - A "frame" is the minimum unit in the time axis direction, and consists * of samples for the number of channels. * - A "block" is basic length of processing. The audio layer basically * handles audio data stream block by block, asks underlying hardware to * process them block by block, and then the hardware raises interrupt by * each block. * - A "track" is single completed audio stream. * * For example, the hardware block is assumed to be 10 msec, and your audio * track consists of 2.1(=3) channels 44.1kHz 16bit PCM, * * "channel" = 3 * "sample" = 2 [bytes] * "frame" = 2 [bytes/sample] * 3 [channels] = 6 [bytes] * "block" = 44100 [Hz] * (10/1000) [seconds] * 6 [bytes/frame] = 2646 [bytes] * * The terminologies shown here are only for this MI audio layer. Note that * different terminologies may be used in each manufacturer's datasheet, and * each MD driver may follow it. For example, what we call a "block" is * called a "frame" in sys/dev/pci/yds.c. */ /* * 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 - - * 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 __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.136.4.1 2024/03/26 18:09:21 martin Exp $"); #ifdef _KERNEL_OPT #include "audio.h" #include "midi.h" #endif #if NAUDIO > 0 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "ioconf.h" /* * 0: No debug logs * 1: action changes like open/close/set_format/mmap... * 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; membar_acquire(); int rpage = mlog_wpage; mlog_wpage ^= 1; mlog_buf[mlog_wpage][0] = '\0'; mlog_used = 0; atomic_store_release(&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; } membar_acquire(); 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_store_release(&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 , 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 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 void 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 int audio_track_inputblk_as_usrbyte(const audio_track_t *, int); static 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 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); } #if 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); } #endif #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 */ static int audiounit(dev_t dev) { return AUDIOUNIT(dev); } 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_cfdriver = &audio_cd, .d_devtounit = audiounit, .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(sc->sc_mixer_state[0]) * (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 maj, mn; 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; /* * Prevent new opens and wait for existing opens to complete. * * At the moment there are only four bits in the minor for the * unit number, so we only revoke if the unit number could be * used in a device node. * * XXX If we want more audio units, we need to encode them * more elaborately in the minor space. */ maj = cdevsw_lookup_major(&audio_cdevsw); mn = device_unit(self); if (mn <= 0xf) { vdevgone(maj, mn|SOUND_DEVICE, mn|SOUND_DEVICE, VCHR); vdevgone(maj, mn|AUDIO_DEVICE, mn|AUDIO_DEVICE, VCHR); vdevgone(maj, mn|AUDIOCTL_DEVICE, mn|AUDIOCTL_DEVICE, VCHR); vdevgone(maj, mn|MIXER_DEVICE, mn|MIXER_DEVICE, VCHR); } /* * 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); } mutex_exit(sc->sc_lock); /* * 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); 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. */ mutex_enter(sc->sc_lock); while ((file = SLIST_FIRST(&sc->sc_files)) != NULL) { mutex_enter(sc->sc_intr_lock); SLIST_REMOVE_HEAD(&sc->sc_files, entry); mutex_exit(sc->sc_intr_lock); if (file->ptrack || file->rtrack) { mutex_exit(sc->sc_lock); audio_unlink(sc, file); mutex_enter(sc->sc_lock); } } mutex_exit(sc->sc_lock); 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, CFARGS_NONE); 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, CFARGS(.search = audiosearch)); 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, CFARGS(.iattr = "audiobus")); } /* * 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); } /* * 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 acquired. * 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) { if (atomic_swap_uint(&track->lock, 1) != 0) return false; membar_acquire(); return true; } /* * 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) SPINLOCK_BACKOFF_HOOK; } /* * Release track lock. */ static __inline void audio_track_lock_exit(audio_track_t *track) { atomic_store_release(&track->lock, 0); } static int audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) { struct audio_softc *sc; int error; /* * Find the device. Because we wired the cdevsw to the audio * autoconf instance, the system ensures it will not go away * until after we return. */ sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL || sc->hw_if == NULL) return ENXIO; error = audio_exlock_enter(sc); if (error) return error; 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); 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: mutex_enter(sc->sc_lock); mutex_enter(sc->sc_intr_lock); SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); error = 0; break; case MIXER_DEVICE: mutex_enter(sc->sc_lock); mutex_enter(sc->sc_intr_lock); SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); 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(len > 0); 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) { device_t audiodev = NULL; struct audio_softc *sc; bool exlock = false; int error; /* * Find the autoconf instance and make sure it doesn't go away * while we are opening it. */ audiodev = device_lookup_acquire(&audio_cd, AUDIOUNIT(dev)); if (audiodev == NULL) { error = ENXIO; goto out; } /* If attach failed, it's hopeless -- give up. */ sc = device_private(audiodev); if (sc->hw_if == NULL) { error = ENXIO; goto out; } /* Take the exclusive configuration lock. */ error = audio_exlock_enter(sc); if (error) goto out; exlock = true; /* Open the audio device. */ device_active(sc->sc_dev, DVA_SYSTEM); error = audio_open(dev, sc, FWRITE, 0, curlwp, filep); out: if (exlock) audio_exlock_exit(sc); if (audiodev) device_release(audiodev); 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(*af), 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); error = ENXIO; 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; /* * 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); } mutex_enter(sc->sc_lock); mutex_enter(sc->sc_intr_lock); SLIST_REMOVE(&sc->sc_files, file, audio_file, entry); mutex_exit(sc->sc_intr_lock); 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); } audio_unlink(sc, file); audio_exlock_exit(sc); return 0; } /* * Unlink this file, but not freeing memory here. * Must be called with sc_exlock held and without sc_lock held. */ static void audio_unlink(struct audio_softc *sc, audio_file_t *file) { kauth_cred_t cred = NULL; 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); 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); } cred = sc->sc_cred; sc->sc_cred = NULL; } mutex_exit(sc->sc_lock); if (cred) kauth_cred_free(cred); TRACE(3, "done"); } /* * 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/C%d", uio->uio_resid, input->head, input->used, input->capacity, usrbuf->head, usrbuf->used, usrbuf->capacity); /* 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); /* Convert one block if possible. */ if (usrbuf->used == 0 && input->used > 0) { audio_track_record(track); } /* uiomove from usrbuf as many bytes as possible. */ bytes = uimin(usrbuf->used, uio->uio_resid); error = uiomove((uint8_t *)usrbuf->mem + usrbuf->head, bytes, uio); if (error) { audio_track_lock_exit(track); device_printf(sc->sc_dev, "%s: uiomove(%d) failed: errno=%d\n", __func__, bytes, error); goto abort; } auring_take(usrbuf, bytes); TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", bytes, usrbuf->head, usrbuf->used, usrbuf->capacity); 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; if (track == NULL) return EPERM; /* 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 many bytes 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); TRACET(3, track, "uiomove(len=%d) usrbuf=%d/%d/C%d", len, usrbuf->head, usrbuf->used, usrbuf->capacity); bytes -= len; } /* Convert them as many blocks 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 offset; int val; 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 */ }; char pre[64]; int nameidx = (cmd & 0xff); if (21 <= nameidx && nameidx <= 21 + __arraycount(ioctlnames)) { snprintf(pre, sizeof(pre), "pid=%d.%d %s", (int)curproc->p_pid, (int)l->l_lid, ioctlnames[nameidx - 21]); } else { snprintf(pre, sizeof(pre), "pid=%d.%d (%lu,'%c',%u)", (int)curproc->p_pid, (int)l->l_lid, IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), nameidx); } #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. */ track = file->rtrack; if (track) { val = audio_track_readablebytes(track); *(int *)addr = val; TRACET(2, track, "pid=%d.%d FIONREAD bytes=%d", (int)curproc->p_pid, (int)l->l_lid, val); } else { TRACEF(2, file, "pid=%d.%d FIONREAD no track", (int)curproc->p_pid, (int)l->l_lid); } break; case FIOASYNC: /* Set/Clear ASYNC I/O. */ if (*(int *)addr) { file->async_audio = curproc->p_pid; } else { file->async_audio = 0; } TRACEF(2, file, "pid=%d.%d FIOASYNC %s", (int)curproc->p_pid, (int)l->l_lid, file->async_audio ? "on" : "off"); break; case AUDIO_FLUSH: /* XXX TODO: clear errors and restart? */ TRACEF(2, file, "%s", pre); audio_file_clear(sc, file); break; case AUDIO_PERROR: case AUDIO_RERROR: /* * Number of dropped bytes during playback/record. 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 = (cmd == AUDIO_PERROR) ? file->ptrack : file->rtrack; if (track) { val = frametobyte(&track->usrbuf.fmt, track->dropframes); *(int *)addr = val; TRACET(2, track, "%s bytes=%d", pre, val); } else { TRACEF(2, file, "%s no track", pre); } break; case AUDIO_GETIOFFS: ao = (struct audio_offset *)addr; track = file->rtrack; if (track == NULL) { ao->samples = 0; ao->deltablks = 0; ao->offset = 0; TRACEF(2, file, "%s no rtrack", pre); break; } mutex_enter(sc->sc_lock); mutex_enter(sc->sc_intr_lock); /* figure out where next transfer will start */ stamp = track->stamp; offset = auring_tail(track->input); mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); /* samples will overflow soon but is as per spec. */ ao->samples = stamp * track->usrbuf_blksize; ao->deltablks = stamp - track->last_stamp; ao->offset = audio_track_inputblk_as_usrbyte(track, offset); TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", pre, ao->samples, ao->deltablks, ao->offset); track->last_stamp = stamp; break; case AUDIO_GETOOFFS: ao = (struct audio_offset *)addr; track = file->ptrack; if (track == NULL) { ao->samples = 0; ao->deltablks = 0; ao->offset = 0; TRACEF(2, file, "%s no ptrack", pre); break; } mutex_enter(sc->sc_lock); mutex_enter(sc->sc_intr_lock); /* figure out where next transfer will start */ stamp = track->stamp; offset = track->usrbuf.head; mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); /* samples will overflow soon but is as per spec. */ ao->samples = stamp * track->usrbuf_blksize; ao->deltablks = stamp - track->last_stamp; ao->offset = offset; TRACET(2, track, "%s samples=%u deltablks=%u offset=%u", pre, ao->samples, ao->deltablks, ao->offset); track->last_stamp = stamp; break; case AUDIO_WSEEK: track = file->ptrack; if (track) { val = track->usrbuf.used; *(u_long *)addr = val; TRACET(2, track, "%s bytes=%d", pre, val); } else { TRACEF(2, file, "%s no ptrack", pre); } break; case AUDIO_SETINFO: TRACEF(2, file, "%s", pre); 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; } if (ISDEVSOUND(dev)) error = audiogetinfo(sc, &sc->sc_ai, 0, file); audio_exlock_exit(sc); break; case AUDIO_GETINFO: TRACEF(2, file, "%s", pre); 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: TRACEF(2, file, "%s", pre); 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: track = file->ptrack; if (track) { TRACET(2, track, "%s", pre); mutex_enter(sc->sc_lock); error = audio_track_drain(sc, track); mutex_exit(sc->sc_lock); } else { TRACEF(2, file, "%s no ptrack", pre); } break; case AUDIO_GETDEV: TRACEF(2, file, "%s", pre); error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_GETENC: ae = (audio_encoding_t *)addr; index = ae->index; TRACEF(2, file, "%s index=%d", pre, 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; val = (sc->sc_props & AUDIO_PROP_FULLDUPLEX) && (sc->sc_pmixer && sc->sc_rmixer); audio_exlock_exit(sc); *(int *)addr = val; TRACEF(2, file, "%s fulldup=%d", pre, val); break; case AUDIO_GETPROPS: val = sc->sc_props; *(int *)addr = val; #if defined(AUDIO_DEBUG) char pbuf[64]; snprintb(pbuf, sizeof(pbuf), "\x10" "\6CAPTURE" "\5PLAY" "\3INDEP" "\2MMAP" "\1FULLDUP", val); TRACEF(2, file, "%s %s", pre, pbuf); #endif break; case AUDIO_QUERYFORMAT: query = (audio_format_query_t *)addr; TRACEF(2, file, "%s index=%u", pre, query->index); 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: TRACEF(2, file, "%s", pre); 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: TRACEF(2, file, "%s", pre); 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 */ TRACEF(2, file, "%s", pre); break; default: TRACEF(2, file, "%s", pre); 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; } if (error) TRACEF(2, file, "%s error=%d", pre, error); return error; } /* * Convert n [frames] of the input buffer to bytes in the usrbuf format. * n is in frames but should be a multiple of frame/block. Note that the * usrbuf's frame/block and the input buffer's frame/block may be different * (i.e., if frequencies are different). * * This function is for recording track only. */ static int audio_track_inputblk_as_usrbyte(const audio_track_t *track, int n) { int input_fpb; /* * In the input buffer on recording track, these are the same. * input_fpb = frame_per_block(track->mixer, &track->input->fmt); */ input_fpb = track->mixer->frames_per_block; return (n / input_fpb) * track->usrbuf_blksize; } /* * Returns the number of bytes that can be read on recording buffer. */ static int audio_track_readablebytes(const audio_track_t *track) { int bytes; KASSERT(track); KASSERT(track->mode == AUMODE_RECORD); /* * For recording, track->input is the main block-unit buffer and * track->usrbuf holds less than one block of byte data ("fragment"). * Note that the input buffer is in frames and the usrbuf is in bytes. * * Actual total capacity of these two buffers is * input->capacity [frames] + usrbuf.capacity [bytes], * but only input->capacity is reported to userland as buffer_size. * So, even if the total used bytes exceed input->capacity, report it * as input->capacity for consistency. */ bytes = audio_track_inputblk_as_usrbyte(track, track->input->used); if (track->input->used < track->input->capacity) { bytes += track->usrbuf.used; } 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_flags = FILTEROP_ISFD, .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_flags = FILTEROP_ISFD, .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; struct uvm_object *uobj; vaddr_t vstart; vsize_t vsize; int error; TRACEF(1, file, "off=%jd, len=%ju, prot=%d", (intmax_t)(*offp), (uintmax_t)len, prot); KASSERT(len > 0); 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; /* XXX TODO: what happens when mmap twice. */ if (track->mmapped) return EIO; /* Create a uvm anonymous object */ vsize = roundup2(MAX(track->usrbuf.capacity, PAGE_SIZE), PAGE_SIZE); if (*offp + len > vsize) return EOVERFLOW; uobj = uao_create(vsize, 0); /* Map it into the kernel virtual address space */ vstart = 0; error = uvm_map(kernel_map, &vstart, vsize, 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(uobj); /* release reference */ return error; } error = uvm_map_pageable(kernel_map, vstart, vstart + vsize, false, 0); if (error) { device_printf(sc->sc_dev, "uvm_map_pageable failed: errno=%d\n", error); goto abort; } error = audio_exlock_mutex_enter(sc); if (error) goto abort; /* * mmap() will start playing immediately. XXX Maybe we lack API... * If no one has played yet, start pmixer here. */ if (sc->sc_pbusy == false) audio_pmixer_start(sc, true); audio_exlock_mutex_exit(sc); /* Finally, replace the usrbuf from kmem to uvm. */ audio_track_lock_enter(track); kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); track->usrbuf.mem = (void *)vstart; track->usrbuf_allocsize = vsize; memset(track->usrbuf.mem, 0, vsize); track->mmapped = true; audio_track_lock_exit(track); /* Acquire a reference for the mmap. munmap will release. */ uao_reference(uobj); *uobjp = uobj; *maxprotp = prot; *advicep = UVM_ADV_RANDOM; *flagsp = MAP_SHARED; return 0; abort: uvm_unmap(kernel_map, vstart, vstart + vsize); /* uvm_unmap also detach uobj */ return error; } /* * /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(*af), KM_SLEEP); af->sc = sc; af->dev = dev; mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); kmem_free(af, sizeof(*af)); fd_abort(curproc, fp, fd); return ENXIO; } 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); 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); if (memblock) kern_free(memblock); return kern_malloc(bytes, M_WAITOK); } /* * Free usrbuf (if available). */ static void audio_free_usrbuf(audio_track_t *track) { vaddr_t vstart; vsize_t vsize; if (track->usrbuf_allocsize != 0) { if (track->mmapped) { /* * 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. */ vstart = (vaddr_t)track->usrbuf.mem; vsize = track->usrbuf_allocsize; uvm_unmap(kernel_map, vstart, vstart + vsize); track->mmapped = false; } else { kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); } } track->usrbuf.mem = NULL; track->usrbuf.capacity = 0; track->usrbuf_allocsize = 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 facilitate 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; arg->context = track; *last_dstp = srcbuf; return 0; } track->freq.filter = NULL; audio_free(srcbuf->mem); return error; } /* * There are two unit of buffers; A block buffer and a byte buffer. Both use * audio_ring_t. Internally, audio data is always handled in block unit. * Converting format, sythesizing tracks, transferring from/to the hardware, * and etc. Only one exception is usrbuf. To transfer with userland, usrbuf * is buffered in byte unit. * For playing back, write(2) writes arbitrary length of data to usrbuf. * When one block is filled, it is sent to the next stage (converting and/or * synthesizing). * For recording, the rmixer writes one block length of data to input buffer * (the bottom stage buffer) each time. read(2) (converts one block if usrbuf * is empty and then) reads arbitrary length of data from usrbuf. * * The following charts show the data flow and buffer types for playback and * recording track. In this example, both have two conversion stages, codec * and freq. Every [**] represents a buffer described below. * * On playback track: * * write(2) * | * | uiomove * v * usrbuf [BB|BB ... BB|BB] .. Byte ring buffer * | * | memcpy one block * v * codec.srcbuf [FF] .. 1 block (ring) buffer * .dst ----+ * | * | convert * v * freq.srcbuf [FF] .. 1 block (ring) buffer * .dst ----+ * | * | convert * v * outbuf [FF|FF|FF|FF] .. NBLKOUT blocks ring buffer * | * v * pmixer * * There are three different types of buffers: * * [BB|BB ... BB|BB] usrbuf. Is the buffer closest to userland. Mandatory. * This is a byte buffer and its length is basically less * than or equal to 64KB or at least AUMINNOBLK blocks. * * [FF] Interim conversion stage's srcbuf if necessary. * This is one block (ring) buffer counted in frames. * * [FF|FF|FF|FF] outbuf. Is the buffer closest to pmixer. Mandatory. * This is NBLKOUT blocks ring buffer counted in frames. * * * On recording track: * * read(2) * ^ * | uiomove * | * usrbuf [BB] .. Byte (ring) buffer * ^ * | memcpy one block * | * outbuf [FF] .. 1 block (ring) buffer * ^ * | convert * | * codec.dst ----+ * .srcbuf [FF] .. 1 block (ring) buffer * ^ * | convert * | * freq.dst ----+ * .srcbuf [FF|FF ... FF|FF] .. NBLKIN blocks ring buffer * ^ * | * rmixer * * There are also three different types of buffers. * * [BB] usrbuf. Is the buffer closest to userland. Mandatory. * This is a byte buffer and its length is one block. * This buffer holds only "fragment". * * [FF] Interim conversion stage's srcbuf (or outbuf). * This is one block (ring) buffer counted in frames. * * [FF|FF ... FF|FF] The bottom conversion stage's srcbuf (or outbuf). * This is the buffer closest to rmixer, and mandatory. * This is NBLKIN blocks ring buffer counted in frames. * Also pointed by *input. */ /* * 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) { audio_ring_t *last_dst; int is_playback; u_int newbufsize; u_int newvsize; u_int len; int error; KASSERT(track); is_playback = audio_track_is_playback(track); /* Once mmap is called, the track format cannot be changed. */ if (track->mmapped) return EIO; /* usrbuf is the closest buffer to the userland. */ track->usrbuf.fmt = *usrfmt; /* * Usrbuf. * On the playback track, its capacity is less than or equal to 64KB * (for historical reason) and must be a multiple of a block * (constraint in this implementation). But at least AUMINNOBLK * blocks. * On the recording track, its capacity is one block. */ /* * 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. */ track->usrbuf_blksize = frametobyte(&track->usrbuf.fmt, frame_per_block(track->mixer, &track->usrbuf.fmt)); track->usrbuf.head = 0; track->usrbuf.used = 0; if (is_playback) { newbufsize = track->usrbuf_blksize * AUMINNOBLK; if (newbufsize < 65536) newbufsize = rounddown(65536, track->usrbuf_blksize); newvsize = roundup2(newbufsize, PAGE_SIZE); } else { newbufsize = track->usrbuf_blksize; newvsize = track->usrbuf_blksize; } /* * Reallocate only if the number of pages changes. * This is because we expect kmem to allocate memory on per page * basis if the request size is about 64KB. */ if (newvsize != track->usrbuf_allocsize) { if (track->usrbuf_allocsize != 0) { kmem_free(track->usrbuf.mem, track->usrbuf_allocsize); } TRACET(2, track, "usrbuf_allocsize %d -> %d", track->usrbuf_allocsize, newvsize); track->usrbuf.mem = kmem_alloc(newvsize, KM_SLEEP); track->usrbuf_allocsize = newvsize; } track->usrbuf.capacity = newbufsize; /* Recalc water mark. */ if (is_playback) { /* 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%, low at 0%. (But not used) */ track->usrbuf_usedhigh = track->usrbuf.capacity; track->usrbuf_usedlow = 0; } /* Stage buffer */ last_dst = &track->outbuf; if (is_playback) { /* 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; /* * 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 (is_playback) track->outbuf.capacity *= NBLKOUT; len = auring_bytelen(&track->outbuf); track->outbuf.mem = audio_realloc(track->outbuf.mem, len); /* * On the recording track, expand the input stage buffer, which is * the closest buffer to rmixer, to NBLKIN blocks. * Note that input buffer may point to outbuf. */ if (!is_playback) { int input_fpb; input_fpb = frame_per_block(track->mixer, &track->input->fmt); track->input->capacity = input_fpb * NBLKIN; len = auring_bytelen(track->input); track->input->mem = audio_realloc(track->input->mem, len); } #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 (is_playback) { 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)); 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; 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; } } track->stamp++; #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); if (auring_get_contig_used(track->input) == 0) { TRACET(4, track, "input->used == 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; /* usrbuf should be empty. */ KASSERT(usrbuf->used == 0); /* * 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_blksize / 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); } #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); } /* 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->codec && 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->codec && 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 has less than one block of free space, * make one block free. */ 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(auring_tail(input) % mixer->frames_per_block == 0, "inputtail=%d mixer->frames_per_block=%d", auring_tail(input), mixer->frames_per_block); memcpy(auring_tailptr_aint(input), auring_headptr_aint(mixersrc), bytes); auring_push(input, count); track->stamp++; 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); /* Clear all internal parameters. */ track->usrbuf.used = 0; track->usrbuf.head = 0; 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->stamp = 0; track->last_stamp = 0; 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"); 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; } /* Do not prefer surround formats */ if (query.fmt.channels <= 2) 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 occurred. * * 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 specified. */ 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; } } /* 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; } if (SPECIFIED(newpi->gain)) pgain = newpi->gain; if (SPECIFIED_CH(newpi->balance)) pbalance = newpi->balance; error = au_set_gain(sc, &sc->sc_outports, pgain, pbalance); if (error) { audio_printf(sc, "setting play.gain=%d/balance=%d failed: " "errno=%d\n", pgain, pbalance, error); goto abort; } } /* 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(newri->gain)) rgain = newri->gain; if (SPECIFIED_CH(newri->balance)) rbalance = newri->balance; error = au_set_gain(sc, &sc->sc_inports, rgain, rbalance); if (error) { audio_printf(sc, "setting record.gain=%d/balance=%d failed: " "errno=%d\n", rgain, rbalance, 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->stamp * ptrack->usrbuf_blksize; 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 = audio_track_readablebytes(rtrack); ri->samples = rtrack->stamp * rtrack->usrbuf_blksize; ri->eof = 0; ri->error = (rtrack->dropframes != 0) ? 1 : 0; ri->open = 1; ri->buffer_size = audio_track_inputblk_as_usrbyte(rtrack, rtrack->input->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; mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); kmem_free(af, sizeof(*af)); fd_abort(curproc, fp, fd); return ENXIO; } 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); 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 val; int error; #if defined(AUDIO_DEBUG) char pre[64]; snprintf(pre, sizeof(pre), "pid=%d.%d", (int)curproc->p_pid, (int)l->l_lid); #endif 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: val = *(int *)addr; TRACE(2, "%s FIOASYNC %s", pre, val ? "on" : "off"); error = audio_exlock_enter(sc); if (error) break; if (val) { 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, "%s AUDIO_GETDEV", pre); error = sc->hw_if->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_MIXER_DEVINFO: TRACE(2, "%s AUDIO_MIXER_DEVINFO", pre); 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, "%s AUDIO_MIXER_READ", pre); 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, "%s AUDIO_MIXER_WRITE", pre); 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: TRACE(2, "(%lu,'%c',%lu)", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd & 0xff); 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; } if (error) TRACE(2, "error=%d", 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 #include #include #include #include #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; case AUDIODEV_TYPE_AUX: type = "aux"; 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 error = config_fini_component(cfdriver_ioconf_audio, cfattach_ioconf_audio, cfdata_ioconf_audio); if (error == 0) devsw_detach(NULL, &audio_cdevsw); #endif if (error == 0) psref_class_destroy(audio_psref_class); break; default: error = ENOTTY; break; } return error; }