/* $NetBSD: audio.c,v 1.357.2.10 2018/01/15 00:08:55 snj Exp $ */ /*- * Copyright (c) 2016 Nathanial Sloss * All rights reserved. * * Copyright (c) 2008 The NetBSD Foundation, Inc. * All rights reserved. * * This code is derived from software contributed to The NetBSD Foundation * by Andrew Doran. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE * POSSIBILITY OF SUCH DAMAGE. */ /* * Copyright (c) 1991-1993 Regents of the University of California. * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * 3. All advertising materials mentioning features or use of this software * must display the following acknowledgement: * This product includes software developed by the Computer Systems * Engineering Group at Lawrence Berkeley Laboratory. * 4. Neither the name of the University nor of the Laboratory may be used * to endorse or promote products derived from this software without * specific prior written permission. * * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF * SUCH DAMAGE. */ /* * This is a (partially) SunOS-compatible /dev/audio driver for NetBSD. * * This code tries to do something half-way sensible with * half-duplex hardware, such as with the SoundBlaster hardware. With * half-duplex hardware allowing O_RDWR access doesn't really make * sense. However, closing and opening the device to "turn around the * line" is relatively expensive and costs a card reset (which can * take some time, at least for the SoundBlaster hardware). Instead * we allow O_RDWR access, and provide an ioctl to set the "mode", * i.e. playing or recording. * * If you write to a half-duplex device in record mode, the data is * tossed. If you read from the device in play mode, you get silence * filled buffers at the rate at which samples are naturally * generated. * * If you try to set both play and record mode on a half-duplex * device, playing takes precedence. */ /* * Locking: there are two locks. * * - sc_lock, provided by the underlying driver. This is an adaptive lock, * returned in the second parameter to hw_if->get_locks(). It is known * as the "thread lock". * * It serializes access to state in all places except the * driver's interrupt service routine. This lock is taken from process * context (example: access to /dev/audio). It is also taken from soft * interrupt handlers in this module, primarily to serialize delivery of * wakeups. This lock may be used/provided by modules external to the * audio subsystem, so take care not to introduce a lock order problem. * LONG TERM SLEEPS MUST NOT OCCUR WITH THIS LOCK HELD. * * - sc_intr_lock, provided by the underlying driver. This may be either a * spinlock (at IPL_SCHED or IPL_VM) or an adaptive lock (IPL_NONE or * IPL_SOFT*), returned in the first parameter to hw_if->get_locks(). It * is known as the "interrupt lock". * * It provides atomic access to the device's hardware state, and to audio * channel data that may be accessed by the hardware driver's ISR. * In all places outside the ISR, sc_lock must be held before taking * sc_intr_lock. This is to ensure that groups of hardware operations are * made atomically. SLEEPS CANNOT OCCUR WITH THIS LOCK HELD. * * List of hardware interface methods, and which locks are held when each * is called by this module: * * METHOD INTR THREAD NOTES * ----------------------- ------- ------- ------------------------- * open x x * close x x * drain x x * query_encoding - x * set_params - x * round_blocksize - x * commit_settings - x * init_output x x * init_input x x * start_output x x * start_input x x * halt_output x x * halt_input x x * speaker_ctl x x * getdev - x * setfd - x * set_port - x * get_port - x * query_devinfo - x * allocm - - Called at attach time * freem - - Called at attach time * round_buffersize - x * mappage - - Mem. unchanged after attach * get_props - x * trigger_output x x * trigger_input x x * dev_ioctl - x * get_locks - - Called at attach time */ #include __KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.357.2.10 2018/01/15 00:08:55 snj 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 "ioconf.h" /* #define AUDIO_DEBUG 1 */ #ifdef AUDIO_DEBUG #define DPRINTF(x) if (audiodebug) printf x #define DPRINTFN(n,x) if (audiodebug>(n)) printf x int audiodebug = AUDIO_DEBUG; #else #define DPRINTF(x) #define DPRINTFN(n,x) #endif #define PREFILL_BLOCKS 3 /* no. audioblocks required to start stream */ #define ROUNDSIZE(x) (x) &= -16 /* round to nice boundary */ #define SPECIFIED(x) ((x) != ~0) #define SPECIFIED_CH(x) ((x) != (u_char)~0) /* #define AUDIO_PM_IDLE */ #ifdef AUDIO_PM_IDLE int audio_idle_timeout = 30; #endif #define HW_LOCK(x) do { \ if ((x) == sc->sc_hwvc) \ mutex_enter(sc->sc_intr_lock); \ } while (0) #define HW_UNLOCK(x) do { \ if ((x) == sc->sc_hwvc) \ mutex_exit(sc->sc_intr_lock); \ } while (0) int audio_blk_ms = AUDIO_BLK_MS; int audiosetinfo(struct audio_softc *, struct audio_info *, bool, struct virtual_channel *); int audiogetinfo(struct audio_softc *, struct audio_info *, int, struct virtual_channel *); int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *, struct file **); int audio_close(struct audio_softc *, int, struct audio_chan *); int audio_read(struct audio_softc *, struct uio *, int, struct virtual_channel *); int audio_write(struct audio_softc *, struct uio *, int, struct virtual_channel *); int audio_ioctl(dev_t, struct audio_softc *, u_long, void *, int, struct lwp *, struct audio_chan *); int audio_poll(struct audio_softc *, int, struct lwp *, struct virtual_channel *); int audio_kqfilter(struct audio_chan *, struct knote *); int audio_mmap(struct audio_softc *, off_t *, size_t, int, int *, int *, struct uvm_object **, int *, struct virtual_channel *); static int audio_fop_mmap(struct file *, off_t *, size_t, int, int *, int *, struct uvm_object **, int *); int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *, struct file **); int mixer_close(struct audio_softc *, int, struct audio_chan *); int mixer_ioctl(struct audio_softc *, u_long, void *, int, struct lwp *); static void mixer_remove(struct audio_softc *); static void mixer_signal(struct audio_softc *); static void grow_mixer_states(struct audio_softc *, int); static void shrink_mixer_states(struct audio_softc *, int); void audio_init_record(struct audio_softc *, struct virtual_channel *); void audio_init_play(struct audio_softc *, struct virtual_channel *); int audiostartr(struct audio_softc *, struct virtual_channel *); int audiostartp(struct audio_softc *, struct virtual_channel *); void audio_rint(void *); void audio_pint(void *); void audio_mix(void *); void audio_upmix(void *); void audio_play_thread(void *); void audio_rec_thread(void *); void recswvol_func(struct audio_softc *, struct audio_ringbuffer *, size_t, struct virtual_channel *); void mix_func(struct audio_softc *, struct audio_ringbuffer *, struct virtual_channel *); int mix_write(void *); int mix_read(void *); int audio_check_params(struct audio_params *); static void audio_calc_latency(struct audio_softc *); static void audio_setblksize(struct audio_softc *, struct virtual_channel *, int, int); int audio_calc_blksize(struct audio_softc *, const audio_params_t *); void audio_fill_silence(struct audio_params *, uint8_t *, int); int audio_silence_copyout(struct audio_softc *, int, struct uio *); static int audio_allocbufs(struct audio_softc *, struct virtual_channel *); void audio_init_ringbuffer(struct audio_softc *, struct audio_ringbuffer *, int); int audio_initbufs(struct audio_softc *, struct virtual_channel *); void audio_calcwater(struct audio_softc *, struct virtual_channel *); int audio_drain(struct audio_softc *, struct virtual_channel *); void audio_clear(struct audio_softc *, struct virtual_channel *); void audio_clear_intr_unlocked(struct audio_softc *sc, struct virtual_channel *); int audio_alloc_ring(struct audio_softc *, struct audio_ringbuffer *, int, size_t); void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *); static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *, stream_filter_list_t *, struct virtual_channel *); static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *, stream_filter_list_t *, struct virtual_channel *); static void audio_destroy_pfilters(struct virtual_channel *); static void audio_destroy_rfilters(struct virtual_channel *); static void audio_stream_dtor(audio_stream_t *); static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int); static void stream_filter_list_append(stream_filter_list_t *, stream_filter_factory_t, const audio_params_t *); static void stream_filter_list_prepend(stream_filter_list_t *, stream_filter_factory_t, const audio_params_t *); static void stream_filter_list_set(stream_filter_list_t *, int, stream_filter_factory_t, const audio_params_t *); int audio_set_defaults(struct audio_softc *, u_int, struct virtual_channel *); static int audio_sysctl_frequency(SYSCTLFN_PROTO); static int audio_sysctl_precision(SYSCTLFN_PROTO); static int audio_sysctl_channels(SYSCTLFN_PROTO); static int audio_sysctl_latency(SYSCTLFN_PROTO); static int audio_sysctl_usemixer(SYSCTLFN_PROTO); static int audiomatch(device_t, cfdata_t, void *); static void audioattach(device_t, device_t, void *); static int audiodetach(device_t, int); static int audioactivate(device_t, enum devact); static void audiochilddet(device_t, device_t); static int audiorescan(device_t, const char *, const int *); static int audio_modcmd(modcmd_t, void *); #ifdef AUDIO_PM_IDLE static void audio_idle(void *); static void audio_activity(device_t, devactive_t); #endif static bool audio_suspend(device_t dv, const pmf_qual_t *); static bool audio_resume(device_t dv, const pmf_qual_t *); static void audio_volume_down(device_t); static void audio_volume_up(device_t); static void audio_volume_toggle(device_t); static void audio_mixer_capture(struct audio_softc *); static void audio_mixer_restore(struct audio_softc *); static int audio_get_props(struct audio_softc *); static bool audio_can_playback(struct audio_softc *); static bool audio_can_capture(struct audio_softc *); static void audio_softintr_rd(void *); static void audio_softintr_wr(void *); static int audio_enter(dev_t, krw_t, struct audio_softc **); static void audio_exit(struct audio_softc *); static int audio_waitio(struct audio_softc *, kcondvar_t *, struct virtual_channel *); static int audioclose(struct file *); static int audioread(struct file *, off_t *, struct uio *, kauth_cred_t, int); static int audiowrite(struct file *, off_t *, struct uio *, kauth_cred_t, int); static int audioioctl(struct file *, u_long, void *); static int audiopoll(struct file *, int); static int audiokqfilter(struct file *, struct knote *); static int audiostat(struct file *, struct stat *); struct portname { const char *name; int mask; }; static const struct portname itable[] = { { AudioNmicrophone, AUDIO_MICROPHONE }, { AudioNline, AUDIO_LINE_IN }, { AudioNcd, AUDIO_CD }, { 0, 0 } }; static const struct portname otable[] = { { AudioNspeaker, AUDIO_SPEAKER }, { AudioNheadphone, AUDIO_HEADPHONE }, { AudioNline, AUDIO_LINE_OUT }, { 0, 0 } }; void au_setup_ports(struct audio_softc *, struct au_mixer_ports *, mixer_devinfo_t *, const struct portname *); int au_set_gain(struct audio_softc *, struct au_mixer_ports *, int, int); void au_get_gain(struct audio_softc *, struct au_mixer_ports *, u_int *, u_char *); int au_set_port(struct audio_softc *, struct au_mixer_ports *, u_int); int au_get_port(struct audio_softc *, struct au_mixer_ports *); static int audio_get_port(struct audio_softc *, mixer_ctrl_t *); static int audio_set_port(struct audio_softc *, mixer_ctrl_t *); static int audio_query_devinfo(struct audio_softc *, mixer_devinfo_t *); static int audio_set_params (struct audio_softc *, int, int, audio_params_t *, audio_params_t *, stream_filter_list_t *, stream_filter_list_t *, struct virtual_channel *); static int audio_query_encoding(struct audio_softc *, struct audio_encoding *); static int audio_set_vchan_defaults(struct audio_softc *, u_int); static int vchan_autoconfig(struct audio_softc *); int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *, int *, int *); int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *, int, int); int au_portof(struct audio_softc *, char *, int); typedef struct uio_fetcher { stream_fetcher_t base; struct uio *uio; int usedhigh; int last_used; } uio_fetcher_t; static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int); static int uio_fetcher_fetch_to(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int); static int null_fetcher_fetch_to(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int); static dev_type_open(audioopen); /* XXXMRG use more dev_type_xxx */ const struct cdevsw audio_cdevsw = { .d_open = audioopen, .d_close = noclose, .d_read = noread, .d_write = nowrite, .d_ioctl = noioctl, .d_stop = nostop, .d_tty = notty, .d_poll = nopoll, .d_mmap = nommap, .d_kqfilter = nokqfilter, .d_discard = nodiscard, .d_flag = D_OTHER | D_MPSAFE }; const struct fileops audio_fileops = { .fo_read = audioread, .fo_write = audiowrite, .fo_ioctl = audioioctl, .fo_fcntl = fnullop_fcntl, .fo_stat = audiostat, .fo_poll = audiopoll, .fo_close = audioclose, .fo_mmap = audio_fop_mmap, .fo_kqfilter = audiokqfilter, .fo_restart = fnullop_restart }; /* The default audio mode: 8 kHz mono mu-law */ const struct audio_params audio_default = { .sample_rate = 8000, .encoding = AUDIO_ENCODING_ULAW, .precision = 8, .validbits = 8, .channels = 1, }; int auto_config_precision[] = { 16, 8, 32 }; int auto_config_channels[] = { 2, AUDIO_MAX_CHANNELS, 10, 8, 6, 4, 1 }; int auto_config_freq[] = { 48000, 44100, 96000, 192000, 32000, 22050, 16000, 11025, 8000, 4000 }; CFATTACH_DECL3_NEW(audio, sizeof(struct audio_softc), audiomatch, audioattach, audiodetach, audioactivate, audiorescan, audiochilddet, DVF_DETACH_SHUTDOWN); static int audiomatch(device_t parent, cfdata_t match, void *aux) { struct audio_attach_args *sa; sa = aux; DPRINTF(("%s: type=%d sa=%p hw=%p\n", __func__, sa->type, sa, sa->hwif)); return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0; } static void audioattach(device_t parent, device_t self, void *aux) { struct audio_softc *sc; struct audio_attach_args *sa; struct virtual_channel *vc; const struct audio_hw_if *hwp; const struct sysctlnode *node; void *hdlp; int error; mixer_devinfo_t mi; int iclass, mclass, oclass, rclass, props; int record_master_found, record_source_found; sc = device_private(self); sc->dev = self; sa = aux; hwp = sa->hwif; hdlp = sa->hdl; sc->sc_opens = 0; sc->sc_recopens = 0; sc->sc_aivalid = false; sc->sc_ready = true; sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS; sc->sc_format[0].mode = AUMODE_PLAY | AUMODE_RECORD; sc->sc_format[0].encoding = #if BYTE_ORDER == LITTLE_ENDIAN AUDIO_ENCODING_SLINEAR_LE; #else AUDIO_ENCODING_SLINEAR_BE; #endif sc->sc_format[0].precision = 16; sc->sc_format[0].validbits = 16; sc->sc_format[0].channels = 2; sc->sc_format[0].channel_mask = AUFMT_STEREO; sc->sc_format[0].frequency_type = 1; sc->sc_format[0].frequency[0] = 44100; sc->sc_trigger_started = false; sc->sc_rec_started = false; sc->sc_dying = false; SIMPLEQ_INIT(&sc->sc_audiochan); vc = kmem_zalloc(sizeof(struct virtual_channel), KM_SLEEP); sc->sc_hwvc = vc; vc->sc_open = 0; vc->sc_mode = 0; vc->sc_npfilters = 0; memset(vc->sc_pfilters, 0, sizeof(vc->sc_pfilters)); vc->sc_lastinfovalid = false; vc->sc_swvol = 255; vc->sc_recswvol = 255; if (auconv_create_encodings(sc->sc_format, VAUDIO_NFORMATS, &sc->sc_encodings) != 0) { aprint_error_dev(self, "couldn't create encodings\n"); return; } cv_init(&sc->sc_rchan, "audiord"); cv_init(&sc->sc_wchan, "audiowr"); cv_init(&sc->sc_lchan, "audiolk"); cv_init(&sc->sc_condvar,"play"); cv_init(&sc->sc_rcondvar,"record"); if (hwp == 0 || hwp->get_locks == 0) { aprint_error(": missing method\n"); panic("audioattach"); } hwp->get_locks(hdlp, &sc->sc_intr_lock, &sc->sc_lock); #ifdef DIAGNOSTIC if (hwp->query_encoding == 0 || hwp->set_params == 0 || (hwp->start_output == 0 && hwp->trigger_output == 0) || (hwp->start_input == 0 && hwp->trigger_input == 0) || hwp->halt_output == 0 || hwp->halt_input == 0 || hwp->getdev == 0 || hwp->set_port == 0 || hwp->get_port == 0 || hwp->query_devinfo == 0 || hwp->get_props == 0) { aprint_error(": missing method\n"); return; } #endif sc->hw_if = hwp; sc->hw_hdl = hdlp; sc->sc_dev = parent; mutex_enter(sc->sc_lock); props = audio_get_props(sc); mutex_exit(sc->sc_lock); if (props & AUDIO_PROP_FULLDUPLEX) aprint_normal(": full duplex"); else aprint_normal(": half duplex"); if (props & AUDIO_PROP_PLAYBACK) aprint_normal(", playback"); if (props & AUDIO_PROP_CAPTURE) aprint_normal(", capture"); if (props & AUDIO_PROP_MMAP) aprint_normal(", mmap"); if (props & AUDIO_PROP_INDEPENDENT) aprint_normal(", independent"); aprint_naive("\n"); aprint_normal("\n"); mutex_enter(sc->sc_lock); if (audio_allocbufs(sc, vc) != 0) { aprint_error_dev(sc->sc_dev, "could not allocate ring buffer\n"); mutex_exit(sc->sc_lock); return; } mutex_exit(sc->sc_lock); sc->sc_lastgain = 128; sc->sc_multiuser = false; sc->sc_usemixer = true; error = vchan_autoconfig(sc); if (error != 0) { aprint_error_dev(sc->sc_dev, "%s: audio_set_vchan_defaults() " "failed\n", __func__); } sc->sc_sih_rd = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, audio_softintr_rd, sc); sc->sc_sih_wr = softint_establish(SOFTINT_SERIAL | SOFTINT_MPSAFE, audio_softintr_wr, sc); iclass = mclass = oclass = rclass = -1; sc->sc_inports.index = -1; sc->sc_inports.master = -1; sc->sc_inports.nports = 0; sc->sc_inports.isenum = false; sc->sc_inports.allports = 0; sc->sc_inports.isdual = false; sc->sc_inports.mixerout = -1; sc->sc_inports.cur_port = -1; sc->sc_outports.index = -1; sc->sc_outports.master = -1; sc->sc_outports.nports = 0; sc->sc_outports.isenum = false; sc->sc_outports.allports = 0; sc->sc_outports.isdual = false; sc->sc_outports.mixerout = -1; sc->sc_outports.cur_port = -1; sc->sc_monitor_port = -1; /* * Read through the underlying driver's list, picking out the class * names from the mixer descriptions. We'll need them to decode the * mixer descriptions on the next pass through the loop. */ mutex_enter(sc->sc_lock); for(mi.index = 0; ; mi.index++) { if (audio_query_devinfo(sc, &mi) != 0) break; /* * The type of AUDIO_MIXER_CLASS merely introduces a class. * All the other types describe an actual mixer. */ if (mi.type == AUDIO_MIXER_CLASS) { if (strcmp(mi.label.name, AudioCinputs) == 0) iclass = mi.mixer_class; if (strcmp(mi.label.name, AudioCmonitor) == 0) mclass = mi.mixer_class; if (strcmp(mi.label.name, AudioCoutputs) == 0) oclass = mi.mixer_class; if (strcmp(mi.label.name, AudioCrecord) == 0) rclass = mi.mixer_class; } } mutex_exit(sc->sc_lock); /* Allocate save area. Ensure non-zero allocation. */ sc->sc_static_nmixer_states = mi.index; sc->sc_static_nmixer_states++; sc->sc_nmixer_states = sc->sc_static_nmixer_states; sc->sc_mixer_state = kmem_zalloc(sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1), KM_SLEEP); /* * This is where we assign each control in the "audio" model, to the * underlying "mixer" control. We walk through the whole list once, * assigning likely candidates as we come across them. */ record_master_found = 0; record_source_found = 0; mutex_enter(sc->sc_lock); for(mi.index = 0; ; mi.index++) { if (audio_query_devinfo(sc, &mi) != 0) break; KASSERT(mi.index < sc->sc_nmixer_states); if (mi.type == AUDIO_MIXER_CLASS) continue; if (mi.mixer_class == iclass) { /* * AudioCinputs is only a fallback, when we don't * find what we're looking for in AudioCrecord, so * check the flags before accepting one of these. */ if (strcmp(mi.label.name, AudioNmaster) == 0 && record_master_found == 0) sc->sc_inports.master = mi.index; if (strcmp(mi.label.name, AudioNsource) == 0 && record_source_found == 0) { if (mi.type == AUDIO_MIXER_ENUM) { int i; for(i = 0; i < mi.un.e.num_mem; i++) if (strcmp(mi.un.e.member[i].label.name, AudioNmixerout) == 0) sc->sc_inports.mixerout = mi.un.e.member[i].ord; } au_setup_ports(sc, &sc->sc_inports, &mi, itable); } if (strcmp(mi.label.name, AudioNdac) == 0 && sc->sc_outports.master == -1) sc->sc_outports.master = mi.index; } else if (mi.mixer_class == mclass) { if (strcmp(mi.label.name, AudioNmonitor) == 0) sc->sc_monitor_port = mi.index; } else if (mi.mixer_class == oclass) { if (strcmp(mi.label.name, AudioNmaster) == 0) sc->sc_outports.master = mi.index; if (strcmp(mi.label.name, AudioNselect) == 0) au_setup_ports(sc, &sc->sc_outports, &mi, otable); } else if (mi.mixer_class == rclass) { /* * These are the preferred mixers for the audio record * controls, so set the flags here, but don't check. */ if (strcmp(mi.label.name, AudioNmaster) == 0) { sc->sc_inports.master = mi.index; record_master_found = 1; } #if 1 /* Deprecated. Use AudioNmaster. */ if (strcmp(mi.label.name, AudioNrecord) == 0) { sc->sc_inports.master = mi.index; record_master_found = 1; } if (strcmp(mi.label.name, AudioNvolume) == 0) { sc->sc_inports.master = mi.index; record_master_found = 1; } #endif if (strcmp(mi.label.name, AudioNsource) == 0) { if (mi.type == AUDIO_MIXER_ENUM) { int i; for(i = 0; i < mi.un.e.num_mem; i++) if (strcmp(mi.un.e.member[i].label.name, AudioNmixerout) == 0) sc->sc_inports.mixerout = mi.un.e.member[i].ord; } au_setup_ports(sc, &sc->sc_inports, &mi, itable); record_source_found = 1; } } } mutex_exit(sc->sc_lock); DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, " "output ports=0x%x, output master=%d\n", sc->sc_inports.allports, sc->sc_inports.master, sc->sc_outports.allports, sc->sc_outports.master)); /* sysctl set-up for alternate configs */ sysctl_createv(&sc->sc_log, 0, NULL, &node, 0, CTLTYPE_NODE, device_xname(sc->sc_dev), SYSCTL_DESCR("audio format information"), NULL, 0, NULL, 0, CTL_HW, CTL_CREATE, CTL_EOL); if (node != NULL) { sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_INT, "frequency", SYSCTL_DESCR("intermediate frequency"), audio_sysctl_frequency, 0, (void *)sc, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_INT, "precision", SYSCTL_DESCR("intermediate precision"), audio_sysctl_precision, 0, (void *)sc, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_INT, "channels", SYSCTL_DESCR("intermediate channels"), audio_sysctl_channels, 0, (void *)sc, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_INT, "latency", SYSCTL_DESCR("latency"), audio_sysctl_latency, 0, (void *)sc, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_BOOL, "multiuser", SYSCTL_DESCR("allow multiple user acess"), NULL, 0, &sc->sc_multiuser, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); sysctl_createv(&sc->sc_log, 0, NULL, NULL, CTLFLAG_READWRITE, CTLTYPE_BOOL, "usemixer", SYSCTL_DESCR("allow in-kernel mixing"), audio_sysctl_usemixer, 0, (void *)sc, 0, CTL_HW, node->sysctl_num, CTL_CREATE, CTL_EOL); } selinit(&sc->sc_rsel); selinit(&sc->sc_wsel); #ifdef AUDIO_PM_IDLE callout_init(&sc->sc_idle_counter, 0); callout_setfunc(&sc->sc_idle_counter, audio_idle, self); #endif if (!pmf_device_register(self, audio_suspend, audio_resume)) aprint_error_dev(self, "couldn't establish power handler\n"); #ifdef AUDIO_PM_IDLE if (!device_active_register(self, audio_activity)) aprint_error_dev(self, "couldn't register activity handler\n"); #endif if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_DOWN, audio_volume_down, true)) aprint_error_dev(self, "couldn't add volume down handler\n"); if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_UP, audio_volume_up, true)) aprint_error_dev(self, "couldn't add volume up handler\n"); if (!pmf_event_register(self, PMFE_AUDIO_VOLUME_TOGGLE, audio_volume_toggle, true)) aprint_error_dev(self, "couldn't add volume toggle handler\n"); #ifdef AUDIO_PM_IDLE callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); #endif kthread_create(PRI_SOFTSERIAL, KTHREAD_MPSAFE | KTHREAD_MUSTJOIN, NULL, audio_rec_thread, sc, &sc->sc_recthread, "audiorec"); kthread_create(PRI_SOFTSERIAL, KTHREAD_MPSAFE | KTHREAD_MUSTJOIN, NULL, audio_play_thread, sc, &sc->sc_playthread, "audiomix"); audiorescan(self, "audio", NULL); } static int audioactivate(device_t self, enum devact act) { struct audio_softc *sc = device_private(self); switch (act) { case DVACT_DEACTIVATE: mutex_enter(sc->sc_lock); sc->sc_dying = true; mutex_enter(sc->sc_intr_lock); cv_broadcast(&sc->sc_condvar); mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); return 0; default: return EOPNOTSUPP; } } static int audiodetach(device_t self, int flags) { struct audio_softc *sc; struct audio_chan *chan; int maj, mn, rc; sc = device_private(self); DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags)); /* Start draining existing accessors of the device. */ if ((rc = config_detach_children(self, flags | DETACH_FORCE)) != 0) return rc; mutex_enter(sc->sc_lock); sc->sc_dying = true; cv_broadcast(&sc->sc_wchan); cv_broadcast(&sc->sc_rchan); mutex_enter(sc->sc_intr_lock); cv_broadcast(&sc->sc_condvar); cv_broadcast(&sc->sc_rcondvar); mutex_exit(sc->sc_intr_lock); mutex_exit(sc->sc_lock); kthread_join(sc->sc_playthread); kthread_join(sc->sc_recthread); mutex_enter(sc->sc_lock); cv_destroy(&sc->sc_condvar); cv_destroy(&sc->sc_rcondvar); mutex_exit(sc->sc_lock); /* delete sysctl nodes */ sysctl_teardown(&sc->sc_log); /* locate the major number */ maj = cdevsw_lookup_major(&audio_cdevsw); /* * Nuke the vnodes for any open instances (calls close). * Will wait until any activity on the device nodes has ceased. * * XXXAD NOT YET. * * XXXAD NEED TO PREVENT NEW REFERENCES THROUGH AUDIO_ENTER(). */ mn = device_unit(self); vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR); vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR); vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR); vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR); pmf_event_deregister(self, PMFE_AUDIO_VOLUME_DOWN, audio_volume_down, true); pmf_event_deregister(self, PMFE_AUDIO_VOLUME_UP, audio_volume_up, true); pmf_event_deregister(self, PMFE_AUDIO_VOLUME_TOGGLE, audio_volume_toggle, true); #ifdef AUDIO_PM_IDLE callout_halt(&sc->sc_idle_counter, sc->sc_lock); device_active_deregister(self, audio_activity); #endif pmf_device_deregister(self); /* free resources */ SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { audio_free_ring(sc, &chan->vc->sc_mpr); audio_free_ring(sc, &chan->vc->sc_mrr); } audio_free_ring(sc, &sc->sc_hwvc->sc_mpr); audio_free_ring(sc, &sc->sc_hwvc->sc_mrr); audio_free_ring(sc, &sc->sc_mixring.sc_mpr); audio_free_ring(sc, &sc->sc_mixring.sc_mrr); SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { audio_destroy_pfilters(chan->vc); audio_destroy_rfilters(chan->vc); } audio_destroy_pfilters(sc->sc_hwvc); audio_destroy_rfilters(sc->sc_hwvc); auconv_delete_encodings(sc->sc_encodings); if (sc->sc_sih_rd) { softint_disestablish(sc->sc_sih_rd); sc->sc_sih_rd = NULL; } if (sc->sc_sih_wr) { softint_disestablish(sc->sc_sih_wr); sc->sc_sih_wr = NULL; } kmem_free(sc->sc_hwvc, sizeof(struct virtual_channel)); kmem_free(sc->sc_mixer_state, sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1)); #ifdef AUDIO_PM_IDLE callout_destroy(&sc->sc_idle_counter); #endif seldestroy(&sc->sc_rsel); seldestroy(&sc->sc_wsel); cv_destroy(&sc->sc_rchan); cv_destroy(&sc->sc_wchan); cv_destroy(&sc->sc_lchan); return 0; } static void audiochilddet(device_t self, device_t child) { /* we hold no child references, so do nothing */ } static int audiosearch(device_t parent, cfdata_t cf, const int *locs, void *aux) { if (config_match(parent, cf, aux)) config_attach_loc(parent, cf, locs, aux, NULL); return 0; } static int audiorescan(device_t self, const char *ifattr, const int *flags) { struct audio_softc *sc = device_private(self); if (!ifattr_match(ifattr, "audio")) return 0; config_search_loc(audiosearch, sc->dev, "audio", NULL, NULL); return 0; } int au_portof(struct audio_softc *sc, char *name, int class) { mixer_devinfo_t mi; for (mi.index = 0; audio_query_devinfo(sc, &mi) == 0; mi.index++) { if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0) return mi.index; } return -1; } void au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports, mixer_devinfo_t *mi, const struct portname *tbl) { int i, j; ports->index = mi->index; if (mi->type == AUDIO_MIXER_ENUM) { ports->isenum = true; for(i = 0; tbl[i].name; i++) for(j = 0; j < mi->un.e.num_mem; j++) if (strcmp(mi->un.e.member[j].label.name, tbl[i].name) == 0) { ports->allports |= tbl[i].mask; ports->aumask[ports->nports] = tbl[i].mask; ports->misel[ports->nports] = mi->un.e.member[j].ord; ports->miport[ports->nports] = au_portof(sc, mi->un.e.member[j].label.name, mi->mixer_class); if (ports->mixerout != -1 && ports->miport[ports->nports] != -1) ports->isdual = true; ++ports->nports; } } else if (mi->type == AUDIO_MIXER_SET) { for(i = 0; tbl[i].name; i++) for(j = 0; j < mi->un.s.num_mem; j++) if (strcmp(mi->un.s.member[j].label.name, tbl[i].name) == 0) { ports->allports |= tbl[i].mask; ports->aumask[ports->nports] = tbl[i].mask; ports->misel[ports->nports] = mi->un.s.member[j].mask; ports->miport[ports->nports] = au_portof(sc, mi->un.s.member[j].label.name, mi->mixer_class); ++ports->nports; } } } /* * Called from hardware driver. This is where the MI audio driver gets * probed/attached to the hardware driver. */ device_t audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, device_t dev) { struct audio_attach_args arg; #ifdef DIAGNOSTIC if (ahwp == NULL) { aprint_error("audio_attach_mi: NULL\n"); return 0; } #endif arg.type = AUDIODEV_TYPE_AUDIO; arg.hwif = ahwp; arg.hdl = hdlp; return config_found(dev, &arg, audioprint); } #ifdef AUDIO_DEBUG void audio_printsc(struct audio_softc *); void audio_print_params(const char *, struct audio_params *); void audio_printsc(struct audio_softc *sc) { struct virtual_channel *vc; vc = sc->sc_hwvc; printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if); printf("open 0x%x mode 0x%x\n", vc->sc_open, vc->sc_mode); printf("rchan 0x%x wchan 0x%x ", cv_has_waiters(&sc->sc_rchan), cv_has_waiters(&sc->sc_wchan)); printf("rring used 0x%x pring used=%d\n", audio_stream_get_used(&vc->sc_mrr.s), audio_stream_get_used(&vc->sc_mpr.s)); printf("rbus 0x%x pbus 0x%x ", vc->sc_rbus, vc->sc_pbus); printf("blksize %d", vc->sc_mpr.blksize); printf("hiwat %d lowat %d\n", vc->sc_mpr.usedhigh, vc->sc_mpr.usedlow); } void audio_print_params(const char *s, struct audio_params *p) { printf("%s enc=%u %uch %u/%ubit %uHz\n", s, p->encoding, p->channels, p->validbits, p->precision, p->sample_rate); } #endif /* Allocate all ring buffers. called from audioattach() */ static int audio_allocbufs(struct audio_softc *sc, struct virtual_channel *vc) { int error; sc->sc_mixring.sc_mpr.s.start = NULL; vc->sc_mpr.s.start = NULL; sc->sc_mixring.sc_mrr.s.start = NULL; vc->sc_mrr.s.start = NULL; if (audio_can_playback(sc)) { error = audio_alloc_ring(sc, &sc->sc_mixring.sc_mpr, AUMODE_PLAY, AU_RING_SIZE); if (error) goto bad_play1; error = audio_alloc_ring(sc, &vc->sc_mpr, AUMODE_PLAY, AU_RING_SIZE); if (error) goto bad_play2; } if (audio_can_capture(sc)) { error = audio_alloc_ring(sc, &sc->sc_mixring.sc_mrr, AUMODE_RECORD, AU_RING_SIZE); if (error) goto bad_rec1; error = audio_alloc_ring(sc, &vc->sc_mrr, AUMODE_RECORD, AU_RING_SIZE); if (error) goto bad_rec2; } return 0; bad_rec2: if (sc->sc_mixring.sc_mrr.s.start != NULL) audio_free_ring(sc, &sc->sc_mixring.sc_mrr); bad_rec1: if (vc->sc_mpr.s.start != NULL) audio_free_ring(sc, &vc->sc_mpr); bad_play2: if (sc->sc_mixring.sc_mpr.s.start != NULL) audio_free_ring(sc, &sc->sc_mixring.sc_mpr); bad_play1: return error; } int audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r, int direction, size_t bufsize) { const struct audio_hw_if *hw; struct virtual_channel *vc; void *hdl; vaddr_t vstart; vsize_t vsize; int error; vc = sc->sc_hwvc; hw = sc->hw_if; hdl = sc->hw_hdl; /* * Alloc DMA play and record buffers */ if (bufsize < AUMINBUF) bufsize = AUMINBUF; ROUNDSIZE(bufsize); if (hw->round_buffersize) bufsize = hw->round_buffersize(hdl, direction, bufsize); if (hw->allocm && (r == &vc->sc_mpr || r == &vc->sc_mrr)) { /* Hardware ringbuffer. No dedicated uvm object.*/ r->uobj = NULL; r->s.start = hw->allocm(hdl, direction, bufsize); if (r->s.start == NULL) return ENOMEM; } else { /* Software ringbuffer. */ vstart = 0; /* Get a nonzero multiple of PAGE_SIZE. */ vsize = roundup2(MAX(bufsize, PAGE_SIZE), PAGE_SIZE); /* Create a uvm anonymous object. */ r->uobj = uao_create(vsize, 0); /* Map it into the kernel virtual address space. */ error = uvm_map(kernel_map, &vstart, vsize, r->uobj, 0, 0, UVM_MAPFLAG(UVM_PROT_RW, UVM_PROT_RW, UVM_INH_NONE, UVM_ADV_RANDOM, 0)); if (error) { uao_detach(r->uobj); /* release reference */ r->uobj = NULL; /* paranoia */ return error; } error = uvm_map_pageable(kernel_map, vstart, vstart + vsize, false, 0); if (error) { uvm_unmap(kernel_map, vstart, vstart + vsize); uao_detach(r->uobj); r->uobj = NULL; /* paranoia */ return error; } r->s.start = (void *)vstart; } r->s.bufsize = bufsize; return 0; } void audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r) { struct virtual_channel *vc; vaddr_t vstart; vsize_t vsize; if (r->s.start == NULL) return; vc = sc->sc_hwvc; if (sc->hw_if->freem && (r == &vc->sc_mpr || r == &vc->sc_mrr)) { /* Hardware ringbuffer. */ KASSERT(r->uobj == NULL); sc->hw_if->freem(sc->hw_hdl, r->s.start, r->s.bufsize); } else { /* Software ringbuffer. */ vstart = (vaddr_t)r->s.start; vsize = roundup2(MAX(r->s.bufsize, PAGE_SIZE), PAGE_SIZE); /* * Unmap the kernel mapping. uvm_unmap releases the * reference to the uvm object, and this should be the * last virtual mapping of the uvm object, so no need * to explicitly release (`detach') the object. */ uvm_unmap(kernel_map, vstart, vstart + vsize); r->uobj = NULL; /* paranoia */ } r->s.start = NULL; } static int audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp, stream_filter_list_t *pfilters, struct virtual_channel *vc) { stream_filter_t *pf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS]; audio_stream_t ps[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS]; const audio_params_t *from_param; audio_params_t *to_param; int i, n, onfilters; KASSERT(mutex_owned(sc->sc_lock)); /* Construct new filters. */ memset(pf, 0, sizeof(pf)); memset(ps, 0, sizeof(ps)); from_param = pp; for (i = 0; i < pfilters->req_size; i++) { n = pfilters->req_size - i - 1; to_param = &pfilters->filters[n].param; audio_check_params(to_param); pf[i] = pfilters->filters[n].factory(sc, from_param, to_param); if (pf[i] == NULL) break; if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE)) break; if (i > 0) pf[i]->set_fetcher(pf[i], &pf[i - 1]->base); from_param = to_param; } if (i < pfilters->req_size) { /* failure */ DPRINTF(("%s: pfilters failure\n", __func__)); for (; i >= 0; i--) { if (pf[i] != NULL) pf[i]->dtor(pf[i]); audio_stream_dtor(&ps[i]); } return EINVAL; } /* Swap in new filters. */ HW_LOCK(vc); memcpy(of, vc->sc_pfilters, sizeof(of)); memcpy(os, vc->sc_pstreams, sizeof(os)); onfilters = vc->sc_npfilters; memcpy(vc->sc_pfilters, pf, sizeof(pf)); memcpy(vc->sc_pstreams, ps, sizeof(ps)); vc->sc_npfilters = pfilters->req_size; for (i = 0; i < pfilters->req_size; i++) pf[i]->set_inputbuffer(pf[i], &vc->sc_pstreams[i]); /* hardware format and the buffer near to userland */ if (pfilters->req_size <= 0) { vc->sc_mpr.s.param = *pp; vc->sc_pustream = &vc->sc_mpr.s; } else { vc->sc_mpr.s.param = pfilters->filters[0].param; vc->sc_pustream = &vc->sc_pstreams[0]; } HW_UNLOCK(vc); /* Destroy old filters. */ for (i = 0; i < onfilters; i++) { of[i]->dtor(of[i]); audio_stream_dtor(&os[i]); } #ifdef AUDIO_DEBUG if (audiodebug) { printf("%s: HW-buffer=%p pustream=%p\n", __func__, &vc->sc_mpr.s, vc->sc_pustream); for (i = 0; i < pfilters->req_size; i++) { char num[100]; snprintf(num, 100, "[%d]", i); audio_print_params(num, &vc->sc_pstreams[i].param); } audio_print_params("[HW]", &vc->sc_mpr.s.param); } #endif /* AUDIO_DEBUG */ return 0; } static int audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp, stream_filter_list_t *rfilters, struct virtual_channel *vc) { stream_filter_t *rf[AUDIO_MAX_FILTERS], *of[AUDIO_MAX_FILTERS]; audio_stream_t rs[AUDIO_MAX_FILTERS], os[AUDIO_MAX_FILTERS]; const audio_params_t *to_param; audio_params_t *from_param; int i, onfilters; KASSERT(mutex_owned(sc->sc_lock)); /* Construct new filters. */ memset(rf, 0, sizeof(rf)); memset(rs, 0, sizeof(rs)); for (i = 0; i < rfilters->req_size; i++) { from_param = &rfilters->filters[i].param; audio_check_params(from_param); to_param = i + 1 < rfilters->req_size ? &rfilters->filters[i + 1].param : rp; rf[i] = rfilters->filters[i].factory(sc, from_param, to_param); if (rf[i] == NULL) break; if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE)) break; if (i > 0) { rf[i]->set_fetcher(rf[i], &rf[i - 1]->base); } else { /* rf[0] has no previous fetcher because * the audio hardware fills data to the * input buffer. */ rf[0]->set_inputbuffer(rf[0], &vc->sc_mrr.s); } } if (i < rfilters->req_size) { /* failure */ DPRINTF(("%s: rfilters failure\n", __func__)); for (; i >= 0; i--) { if (rf[i] != NULL) rf[i]->dtor(rf[i]); audio_stream_dtor(&rs[i]); } return EINVAL; } /* Swap in new filters. */ HW_LOCK(vc); memcpy(of, vc->sc_rfilters, sizeof(of)); memcpy(os, vc->sc_rstreams, sizeof(os)); onfilters = vc->sc_nrfilters; memcpy(vc->sc_rfilters, rf, sizeof(rf)); memcpy(vc->sc_rstreams, rs, sizeof(rs)); vc->sc_nrfilters = rfilters->req_size; for (i = 1; i < rfilters->req_size; i++) rf[i]->set_inputbuffer(rf[i], &vc->sc_rstreams[i - 1]); /* hardware format and the buffer near to userland */ if (rfilters->req_size <= 0) { vc->sc_mrr.s.param = *rp; vc->sc_rustream = &vc->sc_mrr.s; } else { vc->sc_mrr.s.param = rfilters->filters[0].param; vc->sc_rustream = &vc->sc_rstreams[rfilters->req_size - 1]; } HW_UNLOCK(vc); #ifdef AUDIO_DEBUG if (audiodebug) { printf("%s: HW-buffer=%p pustream=%p\n", __func__, &vc->sc_mrr.s, vc->sc_rustream); audio_print_params("[HW]", &vc->sc_mrr.s.param); for (i = 0; i < rfilters->req_size; i++) { char num[100]; snprintf(num, 100, "[%d]", i); audio_print_params(num, &vc->sc_rstreams[i].param); } } #endif /* AUDIO_DEBUG */ /* Destroy old filters. */ for (i = 0; i < onfilters; i++) { of[i]->dtor(of[i]); audio_stream_dtor(&os[i]); } return 0; } static void audio_destroy_pfilters(struct virtual_channel *vc) { int i; for (i = 0; i < vc->sc_npfilters; i++) { vc->sc_pfilters[i]->dtor(vc->sc_pfilters[i]); vc->sc_pfilters[i] = NULL; audio_stream_dtor(&vc->sc_pstreams[i]); } vc->sc_npfilters = 0; } static void audio_destroy_rfilters(struct virtual_channel *vc) { int i; for (i = 0; i < vc->sc_nrfilters; i++) { vc->sc_rfilters[i]->dtor(vc->sc_rfilters[i]); vc->sc_rfilters[i] = NULL; audio_stream_dtor(&vc->sc_pstreams[i]); } vc->sc_nrfilters = 0; } static void audio_stream_dtor(audio_stream_t *stream) { if (stream->start != NULL) kmem_free(stream->start, stream->bufsize); memset(stream, 0, sizeof(audio_stream_t)); } static int audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size) { int frame_size; size = min(size, AU_RING_SIZE); stream->bufsize = size; stream->start = kmem_zalloc(size, KM_SLEEP); frame_size = (param->precision + 7) / 8 * param->channels; size = (size / frame_size) * frame_size; stream->end = stream->start + size; stream->inp = stream->start; stream->outp = stream->start; stream->used = 0; stream->param = *param; stream->loop = false; return 0; } static void stream_filter_list_append(stream_filter_list_t *list, stream_filter_factory_t factory, const audio_params_t *param) { if (list->req_size >= AUDIO_MAX_FILTERS) { printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", __func__); return; } list->filters[list->req_size].factory = factory; list->filters[list->req_size].param = *param; list->req_size++; } static void stream_filter_list_set(stream_filter_list_t *list, int i, stream_filter_factory_t factory, const audio_params_t *param) { if (i < 0 || i >= AUDIO_MAX_FILTERS) { printf("%s: invalid index: %d\n", __func__, i); return; } list->filters[i].factory = factory; list->filters[i].param = *param; if (list->req_size <= i) list->req_size = i + 1; } static void stream_filter_list_prepend(stream_filter_list_t *list, stream_filter_factory_t factory, const audio_params_t *param) { if (list->req_size >= AUDIO_MAX_FILTERS) { printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n", __func__); return; } memmove(&list->filters[1], &list->filters[0], sizeof(struct stream_filter_req) * list->req_size); list->filters[0].factory = factory; list->filters[0].param = *param; list->req_size++; } /* * Look up audio device and acquire locks for device access. */ static int audio_enter(dev_t dev, krw_t rw, struct audio_softc **scp) { struct audio_softc *sc; /* First, find the device and take sc_lock. */ sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL || sc->hw_if == NULL) return ENXIO; mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); return EIO; } *scp = sc; return 0; } /* * Release reference to device acquired with audio_enter(). */ static void audio_exit(struct audio_softc *sc) { cv_broadcast(&sc->sc_lchan); mutex_exit(sc->sc_lock); } /* * Wait for I/O to complete, releasing device lock. */ static int audio_waitio(struct audio_softc *sc, kcondvar_t *chan, struct virtual_channel *vc) { struct audio_chan *vchan; bool found = false; int error; KASSERT(mutex_owned(sc->sc_lock)); cv_broadcast(&sc->sc_lchan); /* Wait for pending I/O to complete. */ error = cv_wait_sig(chan, sc->sc_lock); if (!sc->sc_usemixer || vc == sc->sc_hwvc) return error; found = false; SIMPLEQ_FOREACH(vchan, &sc->sc_audiochan, entries) { if (vchan->vc == vc) { found = true; break; } } if (found == false) error = EIO; return error; } /* Exported interfaces for audiobell. */ int audiobellopen(dev_t dev, int flags, int ifmt, struct lwp *l, struct file **fp) { struct audio_softc *sc; int error; if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0) return error; device_active(sc->dev, DVA_SYSTEM); switch (AUDIODEV(dev)) { case AUDIO_DEVICE: error = audio_open(dev, sc, flags, ifmt, l, fp); break; default: error = EINVAL; break; } audio_exit(sc); return error; } int audiobellclose(struct file *fp) { return audioclose(fp); } int audiobellwrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, int ioflag) { return audiowrite(fp, offp, uio, cred, ioflag); } int audiobellioctl(struct file *fp, u_long cmd, void *addr) { return audioioctl(fp, cmd, addr); } static int audioopen(dev_t dev, int flags, int ifmt, struct lwp *l) { struct audio_softc *sc; struct file *fp; int error; if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0) return error; device_active(sc->dev, DVA_SYSTEM); switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: case AUDIOCTL_DEVICE: error = audio_open(dev, sc, flags, ifmt, l, &fp); break; case MIXER_DEVICE: error = mixer_open(dev, sc, flags, ifmt, l, &fp); break; default: error = ENXIO; break; } audio_exit(sc); return error; } static int audioclose(struct file *fp) { struct audio_softc *sc; struct audio_chan *chan; int error; dev_t dev; chan = fp->f_audioctx; if (chan == NULL) /* XXX:NS Why is this needed. */ return EIO; dev = chan->dev; if ((error = audio_enter(dev, RW_WRITER, &sc)) != 0) return error; device_active(sc->dev, DVA_SYSTEM); switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: case AUDIOCTL_DEVICE: error = audio_close(sc, fp->f_flag, chan); break; case MIXER_DEVICE: error = mixer_close(sc, fp->f_flag, chan); break; default: error = ENXIO; break; } if (error == 0) { kmem_free(fp->f_audioctx, sizeof(struct audio_chan)); fp->f_audioctx = NULL; } audio_exit(sc); return error; } static int audioread(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, int ioflag) { struct audio_softc *sc; struct virtual_channel *vc; int error; dev_t dev; if (fp->f_audioctx == NULL) return EIO; dev = fp->f_audioctx->dev; if ((error = audio_enter(dev, RW_READER, &sc)) != 0) return error; if (fp->f_flag & O_NONBLOCK) ioflag |= IO_NDELAY; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: vc = fp->f_audioctx->vc; error = audio_read(sc, uio, ioflag, vc); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } audio_exit(sc); return error; } static int audiowrite(struct file *fp, off_t *offp, struct uio *uio, kauth_cred_t cred, int ioflag) { struct audio_softc *sc; struct virtual_channel *vc; int error; dev_t dev; if (fp->f_audioctx == NULL) return EIO; dev = fp->f_audioctx->dev; if ((error = audio_enter(dev, RW_READER, &sc)) != 0) return error; if (fp->f_flag & O_NONBLOCK) ioflag |= IO_NDELAY; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: vc = fp->f_audioctx->vc; error = audio_write(sc, uio, ioflag, vc); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } audio_exit(sc); return error; } static int audioioctl(struct file *fp, u_long cmd, void *addr) { struct audio_softc *sc; struct audio_chan *chan; struct lwp *l = curlwp; int error; krw_t rw; dev_t dev; if (fp->f_audioctx == NULL) return EIO; chan = fp->f_audioctx; dev = chan->dev; /* Figure out which lock type we need. */ switch (cmd) { case AUDIO_FLUSH: case AUDIO_SETINFO: case AUDIO_DRAIN: case AUDIO_SETFD: rw = RW_WRITER; break; default: rw = RW_READER; break; } if ((error = audio_enter(dev, rw, &sc)) != 0) return error; chan = fp->f_audioctx; switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: case AUDIOCTL_DEVICE: device_active(sc->dev, DVA_SYSTEM); if (IOCGROUP(cmd) == IOCGROUP(AUDIO_MIXER_READ)) error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); else error = audio_ioctl(dev, sc, cmd, addr, fp->f_flag, l, chan); break; case MIXER_DEVICE: error = mixer_ioctl(sc, cmd, addr, fp->f_flag, l); break; default: error = ENXIO; break; } audio_exit(sc); return error; } static int audiostat(struct file *fp, struct stat *st) { if (fp->f_audioctx == NULL) return EIO; memset(st, 0, sizeof(*st)); st->st_dev = fp->f_audioctx->dev; st->st_uid = kauth_cred_geteuid(fp->f_cred); st->st_gid = kauth_cred_getegid(fp->f_cred); st->st_mode = S_IFCHR; return 0; } static int audiopoll(struct file *fp, int events) { struct audio_softc *sc; struct virtual_channel *vc; struct lwp *l = curlwp; int revents; dev_t dev; if (fp->f_audioctx == NULL) return POLLERR; dev = fp->f_audioctx->dev; /* Don't bother with device level lock here. */ sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return POLLERR; mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); return POLLERR; } switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: vc = fp->f_audioctx->vc; revents = audio_poll(sc, events, l, vc); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: revents = 0; break; default: revents = POLLERR; break; } mutex_exit(sc->sc_lock); return revents; } static int audiokqfilter(struct file *fp, struct knote *kn) { struct audio_softc *sc; struct audio_chan *chan; int error; dev_t dev; chan = fp->f_audioctx; dev = chan->dev; /* Don't bother with device level lock here. */ sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return ENXIO; mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); return EIO; } switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_kqfilter(chan, kn); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: error = ENODEV; break; default: error = ENXIO; break; } mutex_exit(sc->sc_lock); return error; } static int audio_fop_mmap(struct file *fp, off_t *offp, size_t len, int prot, int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp) { struct audio_softc *sc; struct audio_chan *chan; struct virtual_channel *vc; dev_t dev; int error; chan = fp->f_audioctx; dev = chan->dev; vc = chan->vc; error = 0; if ((error = audio_enter(dev, RW_READER, &sc)) != 0) return error; device_active(sc->dev, DVA_SYSTEM); /* XXXJDM */ switch (AUDIODEV(dev)) { case SOUND_DEVICE: case AUDIO_DEVICE: error = audio_mmap(sc, offp, len, prot, flagsp, advicep, uobjp, maxprotp, vc); break; case AUDIOCTL_DEVICE: case MIXER_DEVICE: default: error = ENOTSUP; break; } audio_exit(sc); return error; } /* * Audio driver */ void audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp, int mode) { int nblks; int blksize; blksize = rp->blksize; if (blksize < AUMINBLK) blksize = AUMINBLK; if (blksize > rp->s.bufsize / AUMINNOBLK) blksize = rp->s.bufsize / AUMINNOBLK; ROUNDSIZE(blksize); DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize)); struct virtual_channel *hwvc = sc->sc_hwvc; int tmpblksize = 1; /* round blocksize to a power of 2 */ while (tmpblksize < blksize) tmpblksize <<= 1; blksize = tmpblksize; if (sc->hw_if->round_blocksize && (rp == &hwvc->sc_mpr || rp == &hwvc->sc_mrr || rp == &sc->sc_mixring.sc_mpr || rp == &sc->sc_mixring.sc_mrr)) { blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize, mode, &rp->s.param); } if (blksize <= 0) panic("audio_init_ringbuffer: blksize=%d", blksize); nblks = rp->s.bufsize / blksize; DPRINTF(("audio_init_ringbuffer: final blksize=%d\n", blksize)); rp->blksize = blksize; rp->maxblks = nblks; rp->s.end = rp->s.start + nblks * blksize; rp->s.outp = rp->s.inp = rp->s.start; rp->s.used = 0; rp->stamp = 0; rp->stamp_last = 0; rp->fstamp = 0; rp->drops = 0; rp->copying = false; rp->needfill = false; rp->mmapped = false; memset(rp->s.start, 0, blksize * 2); } int audio_initbufs(struct audio_softc *sc, struct virtual_channel *vc) { const struct audio_hw_if *hw; int error; if (vc == NULL) vc = sc->sc_hwvc; DPRINTF(("audio_initbufs: mode=0x%x\n", vc->sc_mode)); hw = sc->hw_if; if (audio_can_capture(sc) && ((vc->sc_open & AUOPEN_READ) || vc == sc->sc_hwvc)) { audio_init_ringbuffer(sc, &vc->sc_mrr, AUMODE_RECORD); if (sc->sc_recopens == 0 && (vc->sc_open & AUOPEN_READ)) { if (hw->init_input) { error = hw->init_input(sc->hw_hdl, vc->sc_mrr.s.start, vc->sc_mrr.s.end - vc->sc_mrr.s.start); if (error) return error; } } } if (audio_can_playback(sc) && ((vc->sc_open & AUOPEN_WRITE) || vc == sc->sc_hwvc)) { audio_init_ringbuffer(sc, &vc->sc_mpr, AUMODE_PLAY); if (sc->sc_opens == 0 && (vc->sc_open & AUOPEN_WRITE)) { if (hw->init_output) { error = hw->init_output(sc->hw_hdl, vc->sc_mpr.s.start, vc->sc_mpr.s.end - vc->sc_mpr.s.start); if (error) return error; } } } #ifdef AUDIO_INTR_TIME if (audio_can_playback(sc)) { sc->sc_pnintr = 0; sc->sc_pblktime = (int64_t)vc->sc_mpr.blksize * 1000000 / (vc->sc_pparams.channels * vc->sc_pparams.sample_rate * vc->sc_pparams.precision / NBBY); DPRINTF(("audio: play blktime = %" PRId64 " for %d\n", sc->sc_pblktime, vc->sc_mpr.blksize)); } if (audio_can_capture(sc)) { sc->sc_rnintr = 0; sc->sc_rblktime = (int64_t)vc->sc_mrr.blksize * 1000000 / (vc->sc_rparams.channels * vc->sc_rparams.sample_rate * vc->sc_rparams.precision / NBBY); DPRINTF(("audio: record blktime = %" PRId64 " for %d\n", sc->sc_rblktime, vc->sc_mrr.blksize)); } #endif return 0; } void audio_calcwater(struct audio_softc *sc, struct virtual_channel *vc) { /* set high at 100% */ if (audio_can_playback(sc) && vc && vc->sc_pustream) { vc->sc_mpr.usedhigh = vc->sc_pustream->end - vc->sc_pustream->start; /* set low at 75% of usedhigh */ vc->sc_mpr.usedlow = vc->sc_mpr.usedhigh * 3 / 4; if (vc->sc_mpr.usedlow == vc->sc_mpr.usedhigh) vc->sc_mpr.usedlow -= vc->sc_mpr.blksize; } if (audio_can_capture(sc) && vc && vc->sc_rustream) { vc->sc_mrr.usedhigh = vc->sc_rustream->end - vc->sc_rustream->start - vc->sc_mrr.blksize; vc->sc_mrr.usedlow = 0; DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__, vc->sc_mpr.usedlow, vc->sc_mpr.usedhigh, vc->sc_mrr.usedlow, vc->sc_mrr.usedhigh)); } } int audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, struct lwp *l, struct file **nfp) { struct file *fp; int error, fd, n; u_int mode; const struct audio_hw_if *hw; struct virtual_channel *vc; struct audio_chan *chan; KASSERT(mutex_owned(sc->sc_lock)); if (sc->sc_usemixer && !sc->sc_ready) return ENXIO; hw = sc->hw_if; if (hw == NULL) return ENXIO; n = 1; SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { n = chan->chan + 1; } if (n < 0) return ENOMEM; chan = kmem_zalloc(sizeof(struct audio_chan), KM_SLEEP); if (sc->sc_usemixer) vc = kmem_zalloc(sizeof(struct virtual_channel), KM_SLEEP); else vc = sc->sc_hwvc; chan->vc = vc; if (!sc->sc_usemixer && AUDIODEV(dev) == AUDIOCTL_DEVICE) goto audioctl_dev; if (sc->sc_usemixer) { vc->sc_open = 0; vc->sc_mode = 0; vc->sc_nrfilters = 0; memset(vc->sc_rfilters, 0, sizeof(vc->sc_rfilters)); vc->sc_rbus = false; vc->sc_npfilters = 0; memset(vc->sc_pfilters, 0, sizeof(vc->sc_pfilters)); vc->sc_draining = false; vc->sc_pbus = false; vc->sc_lastinfovalid = false; vc->sc_swvol = 255; vc->sc_recswvol = 255; } else { if (sc->sc_opens > 0 || sc->sc_recopens > 0 ) { kmem_free(chan, sizeof(struct audio_chan)); return EBUSY; } } DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n", flags, sc, sc->hw_hdl)); if (sc->sc_usemixer) { error = audio_alloc_ring(sc, &vc->sc_mpr, AUMODE_PLAY, AU_RING_SIZE); if (error) goto bad; error = audio_alloc_ring(sc, &vc->sc_mrr, AUMODE_RECORD, AU_RING_SIZE); if (error) goto bad; } if (!sc->sc_usemixer || sc->sc_opens + sc->sc_recopens == 0) { sc->sc_credentials = kauth_cred_get(); kauth_cred_hold(sc->sc_credentials); if (hw->open != NULL) { mutex_enter(sc->sc_intr_lock); error = hw->open(sc->hw_hdl, flags); mutex_exit(sc->sc_intr_lock); if (error) { goto bad; } } audio_initbufs(sc, NULL); if (sc->sc_usemixer && audio_can_playback(sc)) audio_init_ringbuffer(sc, &sc->sc_mixring.sc_mpr, AUMODE_PLAY); if (sc->sc_usemixer && audio_can_capture(sc)) audio_init_ringbuffer(sc, &sc->sc_mixring.sc_mrr, AUMODE_RECORD); sc->schedule_wih = false; sc->schedule_rih = false; sc->sc_last_drops = 0; sc->sc_eof = 0; vc->sc_rbus = false; sc->sc_async_audio = 0; } else if (sc->sc_multiuser == false) { /* XXX:NS Should be handled correctly. */ /* Do we allow multi user access */ if (kauth_cred_geteuid(sc->sc_credentials) != kauth_cred_geteuid(kauth_cred_get()) && kauth_cred_geteuid(kauth_cred_get()) != 0) { error = EPERM; goto bad; } } mutex_enter(sc->sc_intr_lock); vc->sc_full_duplex = (flags & (FWRITE|FREAD)) == (FWRITE|FREAD) && (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX); mutex_exit(sc->sc_intr_lock); mode = 0; if (flags & FREAD) { vc->sc_open |= AUOPEN_READ; mode |= AUMODE_RECORD; } if (flags & FWRITE) { vc->sc_open |= AUOPEN_WRITE; mode |= AUMODE_PLAY | AUMODE_PLAY_ALL; } /* * Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear) * The /dev/audio is always (re)set to 8-bit MU-Law mono * For the other devices, you get what they were last set to. */ if (ISDEVSOUND(dev) && sc->sc_aivalid == true) { sc->sc_ai.mode = mode; sc->sc_ai.play.port = ~0; sc->sc_ai.record.port = ~0; error = audiosetinfo(sc, &sc->sc_ai, true, vc); } else error = audio_set_defaults(sc, mode, vc); if (error) goto bad; #ifdef DIAGNOSTIC /* * Sample rate and precision are supposed to be set to proper * default values by the hardware driver, so that it may give * us these values. */ if (vc->sc_rparams.precision == 0 || vc->sc_pparams.precision == 0) { printf("audio_open: 0 precision\n"); goto bad; } #endif /* audio_close() decreases sc_mpr[n].usedlow, recalculate here */ audio_calcwater(sc, vc); audioctl_dev: error = fd_allocfile(&fp, &fd); if (error) goto bad; if (!sc->sc_usemixer && AUDIODEV(dev) == AUDIOCTL_DEVICE) goto setup_chan; DPRINTF(("audio_open: done sc_mode = 0x%x\n", vc->sc_mode)); if (sc->sc_usemixer) grow_mixer_states(sc, 2); if (flags & FREAD) sc->sc_recopens++; if (flags & FWRITE) sc->sc_opens++; setup_chan: chan->dev = dev; chan->chan = n; chan->deschan = n; if (sc->sc_usemixer) SIMPLEQ_INSERT_TAIL(&sc->sc_audiochan, chan, entries); error = fd_clone(fp, fd, flags, &audio_fileops, chan); KASSERT(error == EMOVEFD); *nfp = fp; return error; bad: audio_destroy_pfilters(vc); audio_destroy_rfilters(vc); if (hw->close != NULL && sc->sc_opens == 0 && sc->sc_recopens == 0) hw->close(sc->hw_hdl); mutex_exit(sc->sc_lock); if (sc->sc_usemixer) { audio_free_ring(sc, &vc->sc_mpr); audio_free_ring(sc, &vc->sc_mrr); mutex_enter(sc->sc_lock); kmem_free(vc, sizeof(struct virtual_channel)); } else mutex_enter(sc->sc_lock); kmem_free(chan, sizeof(struct audio_chan)); return error; } /* * Must be called from task context. */ void audio_init_record(struct audio_softc *sc, struct virtual_channel *vc) { KASSERT(mutex_owned(sc->sc_lock)); if (sc->sc_recopens != 0) return; mutex_enter(sc->sc_intr_lock); if (sc->hw_if->speaker_ctl && (!vc->sc_full_duplex || (vc->sc_mode & AUMODE_PLAY) == 0)) sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF); mutex_exit(sc->sc_intr_lock); } /* * Must be called from task context. */ void audio_init_play(struct audio_softc *sc, struct virtual_channel *vc) { KASSERT(mutex_owned(sc->sc_lock)); if (sc->sc_opens != 0) return; mutex_enter(sc->sc_intr_lock); vc->sc_wstamp = vc->sc_mpr.stamp; if (sc->hw_if->speaker_ctl) sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON); mutex_exit(sc->sc_intr_lock); } int audio_drain(struct audio_softc *sc, struct virtual_channel *vc) { struct audio_ringbuffer *cb; int error, drops; int cc, i, used; bool hw = false; KASSERT(mutex_owned(sc->sc_lock)); KASSERT(mutex_owned(sc->sc_intr_lock)); error = 0; DPRINTF(("audio_drain: enter busy=%d\n", vc->sc_pbus)); cb = &vc->sc_mpr; if (cb->mmapped) return 0; used = audio_stream_get_used(&cb->s); if (vc == sc->sc_hwvc && sc->sc_usemixer) { hw = true; used += audio_stream_get_used(&sc->sc_mixring.sc_mpr.s); } for (i = 0; i < vc->sc_npfilters; i++) used += audio_stream_get_used(&vc->sc_pstreams[i]); if (used <= 0) return 0; if (hw == false && !vc->sc_pbus) { /* We've never started playing, probably because the * block was too short. Pad it and start now. */ uint8_t *inp = cb->s.inp; int blksize = sc->sc_mixring.sc_mpr.blksize; cc = blksize - (inp - cb->s.start) % blksize; audio_fill_silence(&cb->s.param, inp, cc); cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); mutex_exit(sc->sc_intr_lock); error = audiostartp(sc, vc); mutex_enter(sc->sc_intr_lock); if (error) return error; } else if (hw == true) { used = cb->blksize - (sc->sc_mixring.sc_mpr.s.inp - sc->sc_mixring.sc_mpr.s.start) % cb->blksize; while (used > 0) { cc = sc->sc_mixring.sc_mpr.s.end - sc->sc_mixring.sc_mpr.s.inp; if (cc > used) cc = used; audio_fill_silence(&cb->s.param, sc->sc_mixring.sc_mpr.s.inp, cc); sc->sc_mixring.sc_mpr.s.inp = audio_stream_add_inp(&sc->sc_mixring.sc_mpr.s, sc->sc_mixring.sc_mpr.s.inp, cc); used -= cc; } mix_write(sc); } /* * Play until a silence block has been played, then we * know all has been drained. * XXX This should be done some other way to avoid * playing silence. */ #ifdef DIAGNOSTIC if (cb->copying) { DPRINTF(("audio_drain: copying in progress!?!\n")); cb->copying = false; } #endif vc->sc_draining = true; drops = cb->drops; if (vc == sc->sc_hwvc) drops += cb->blksize; else if (sc->sc_usemixer) drops += sc->sc_mixring.sc_mpr.blksize * PREFILL_BLOCKS; error = 0; while (cb->drops <= drops && !error) { DPRINTF(("audio_drain: vc=%p used=%d, drops=%ld\n", vc, audio_stream_get_used(&vc->sc_mpr.s), cb->drops)); mutex_exit(sc->sc_intr_lock); error = audio_waitio(sc, &sc->sc_wchan, vc); mutex_enter(sc->sc_intr_lock); if (sc->sc_dying) error = EIO; } vc->sc_draining = false; return error; } /* * Close an audio chip. */ /* ARGSUSED */ int audio_close(struct audio_softc *sc, int flags, struct audio_chan *chan) { struct virtual_channel *vc; const struct audio_hw_if *hw; KASSERT(mutex_owned(sc->sc_lock)); if (!sc->sc_usemixer && AUDIODEV(chan->dev) == AUDIOCTL_DEVICE) return 0; if (sc->sc_opens == 0 && sc->sc_recopens == 0) return ENXIO; vc = chan->vc; hw = sc->hw_if; if (hw == NULL) return ENXIO; mutex_enter(sc->sc_intr_lock); DPRINTF(("audio_close: sc=%p\n", sc)); /* Stop recording. */ if (sc->sc_recopens == 1 && (flags & FREAD) && vc->sc_rbus) { /* * XXX Some drivers (e.g. SB) use the same routine * to halt input and output so don't halt input if * in full duplex mode. These drivers should be fixed. */ if (!vc->sc_full_duplex || hw->halt_input != hw->halt_output) hw->halt_input(sc->hw_hdl); vc->sc_rbus = false; } /* * Block until output drains, but allow ^C interrupt. */ vc->sc_mpr.usedlow = vc->sc_mpr.blksize; /* avoid excessive wakeups */ /* * If there is pending output, let it drain (unless * the output is paused). */ if ((flags & FWRITE) && vc->sc_pbus) { if (!vc->sc_mpr.pause) audio_drain(sc, chan->vc); vc->sc_pbus = false; } if ((flags & FWRITE) && (sc->sc_opens == 1)) { if (vc->sc_mpr.mmapped == false) audio_drain(sc, sc->sc_hwvc); if (hw->drain) (void)hw->drain(sc->hw_hdl); hw->halt_output(sc->hw_hdl); sc->sc_trigger_started = false; } if ((flags & FREAD) && (sc->sc_recopens == 1)) sc->sc_rec_started = false; if (sc->sc_opens + sc->sc_recopens == 1 && hw->close != NULL) hw->close(sc->hw_hdl); mutex_exit(sc->sc_intr_lock); if (sc->sc_opens + sc->sc_recopens == 1) { sc->sc_async_audio = 0; kauth_cred_free(sc->sc_credentials); } vc->sc_open = 0; vc->sc_mode = 0; vc->sc_full_duplex = 0; audio_destroy_pfilters(vc); audio_destroy_rfilters(vc); if (flags & FREAD) sc->sc_recopens--; if (flags & FWRITE) sc->sc_opens--; if (sc->sc_usemixer) { shrink_mixer_states(sc, 2); SIMPLEQ_REMOVE(&sc->sc_audiochan, chan, audio_chan, entries); mutex_exit(sc->sc_lock); audio_free_ring(sc, &vc->sc_mpr); audio_free_ring(sc, &vc->sc_mrr); mutex_enter(sc->sc_lock); kmem_free(vc, sizeof(struct virtual_channel)); } return 0; } int audio_read(struct audio_softc *sc, struct uio *uio, int ioflag, struct virtual_channel *vc) { struct audio_ringbuffer *cb; const uint8_t *outp; uint8_t *inp; int error, used, cc, n; KASSERT(mutex_owned(sc->sc_lock)); if (sc->hw_if == NULL) return ENXIO; cb = &vc->sc_mrr; if (cb->mmapped) return EINVAL; DPRINTFN(1,("audio_read: cc=%zu mode=%d\n", uio->uio_resid, vc->sc_mode)); #ifdef AUDIO_PM_IDLE if (device_is_active(&sc->dev) || sc->sc_idle) device_active(&sc->dev, DVA_SYSTEM); #endif error = 0; /* * If hardware is half-duplex and currently playing, return * silence blocks based on the number of blocks we have output. */ if (!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY)) { while (uio->uio_resid > 0 && !error) { for(;;) { /* * No need to lock, as any wakeup will be * held for us while holding sc_lock. */ cc = vc->sc_mpr.stamp - vc->sc_wstamp; if (cc > 0) break; DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n", vc->sc_mpr.stamp, vc->sc_wstamp)); if (ioflag & IO_NDELAY) return EWOULDBLOCK; error = audio_waitio(sc, &sc->sc_rchan, vc); if (sc->sc_dying) error = EIO; if (error) return error; } if (uio->uio_resid < cc) cc = uio->uio_resid; DPRINTFN(1,("audio_read: reading in write mode, " "cc=%d\n", cc)); error = audio_silence_copyout(sc, cc, uio); vc->sc_wstamp += cc; } return error; } while (uio->uio_resid > 0 && !error) { while ((used = audio_stream_get_used(vc->sc_rustream)) <= 0) { if (!vc->sc_rbus && !vc->sc_mrr.pause) error = audiostartr(sc, vc); if (error) return error; if (ioflag & IO_NDELAY) return EWOULDBLOCK; DPRINTFN(2, ("audio_read: sleep used=%d\n", used)); error = audio_waitio(sc, &sc->sc_rchan, vc); if (sc->sc_dying) error = EIO; if (error) return error; } outp = vc->sc_rustream->outp; inp = vc->sc_rustream->inp; cb->copying = true; /* * cc is the amount of data in the sc_rustream excluding * wrapped data. Note the tricky case of inp == outp, which * must mean the buffer is full, not empty, because used > 0. */ cc = outp < inp ? inp - outp :vc->sc_rustream->end - outp; DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc)); n = uio->uio_resid; mutex_exit(sc->sc_lock); error = uiomove(__UNCONST(outp), cc, uio); mutex_enter(sc->sc_lock); n -= uio->uio_resid; /* number of bytes actually moved */ vc->sc_rustream->outp = audio_stream_add_outp (vc->sc_rustream, outp, n); cb->copying = false; } return error; } void audio_clear(struct audio_softc *sc, struct virtual_channel *vc) { KASSERT(mutex_owned(sc->sc_intr_lock)); if (vc->sc_rbus) { cv_broadcast(&sc->sc_rchan); if (sc->sc_recopens == 1) { sc->hw_if->halt_input(sc->hw_hdl); sc->sc_rec_started = false; } vc->sc_rbus = false; vc->sc_mrr.pause = false; } if (vc->sc_pbus) { cv_broadcast(&sc->sc_wchan); vc->sc_pbus = false; vc->sc_mpr.pause = false; } } void audio_clear_intr_unlocked(struct audio_softc *sc, struct virtual_channel *vc) { mutex_enter(sc->sc_intr_lock); audio_clear(sc, vc); mutex_exit(sc->sc_intr_lock); } static void audio_calc_latency(struct audio_softc *sc) { const struct audio_params *ap = &sc->sc_vchan_params; if (ap->sample_rate == 0 || ap->channels == 0 || ap->precision == 0) return; sc->sc_latency = sc->sc_hwvc->sc_mpr.blksize * 1000 * PREFILL_BLOCKS * NBBY / ap->sample_rate / ap->channels / ap->precision; } static void audio_setblksize(struct audio_softc *sc, struct virtual_channel *vc, int blksize, int mode) { struct audio_ringbuffer *mixcb, *cb; audio_params_t *parm; audio_stream_t *stream; if (mode == AUMODE_RECORD) { mixcb = &sc->sc_mixring.sc_mrr; cb = &vc->sc_mrr; parm = &vc->sc_rparams; stream = vc->sc_rustream; } else { mixcb = &sc->sc_mixring.sc_mpr; cb = &vc->sc_mpr; parm = &vc->sc_pparams; stream = vc->sc_pustream; } if (sc->sc_usemixer && vc == sc->sc_hwvc) { mixcb->blksize = audio_calc_blksize(sc, parm); cb->blksize = audio_calc_blksize(sc, &cb->s.param); } else { cb->blksize = audio_calc_blksize(sc, &stream->param); if ((!sc->sc_usemixer && SPECIFIED(blksize)) || (SPECIFIED(blksize) && blksize > cb->blksize)) cb->blksize = blksize; } } int audio_calc_blksize(struct audio_softc *sc, const audio_params_t *parm) { int blksize; blksize = parm->sample_rate * sc->sc_latency * parm->channels / 1000 * parm->precision / NBBY / PREFILL_BLOCKS; return blksize; } void audio_fill_silence(struct audio_params *params, uint8_t *p, int n) { uint8_t auzero0, auzero1; int nfill; auzero1 = 0; /* initialize to please gcc */ nfill = 1; switch (params->encoding) { case AUDIO_ENCODING_ULAW: auzero0 = 0x7f; break; case AUDIO_ENCODING_ALAW: auzero0 = 0x55; break; case AUDIO_ENCODING_MPEG_L1_STREAM: case AUDIO_ENCODING_MPEG_L1_PACKETS: case AUDIO_ENCODING_MPEG_L1_SYSTEM: case AUDIO_ENCODING_MPEG_L2_STREAM: case AUDIO_ENCODING_MPEG_L2_PACKETS: case AUDIO_ENCODING_MPEG_L2_SYSTEM: case AUDIO_ENCODING_AC3: case AUDIO_ENCODING_ADPCM: /* is this right XXX */ case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: auzero0 = 0;/* fortunately this works for any number of bits */ break; case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: if (params->precision > 8) { nfill = (params->precision + NBBY - 1)/ NBBY; auzero0 = 0x80; auzero1 = 0; } else auzero0 = 0x80; break; default: DPRINTF(("audio: bad encoding %d\n", params->encoding)); auzero0 = 0; break; } if (nfill == 1) { while (--n >= 0) *p++ = auzero0; /* XXX memset */ } else /* nfill must no longer be 2 */ { if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) { int k = nfill; while (--k > 0) *p++ = auzero1; n -= nfill - 1; } while (n >= nfill) { int k = nfill; *p++ = auzero0; while (--k > 0) *p++ = auzero1; n -= nfill; } if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */ *p++ = auzero0; } } int audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio) { struct virtual_channel *vc; uint8_t zerobuf[128]; int error; int k; vc = sc->sc_hwvc; audio_fill_silence(&vc->sc_rparams, zerobuf, sizeof zerobuf); error = 0; while (n > 0 && uio->uio_resid > 0 && !error) { k = min(n, min(uio->uio_resid, sizeof zerobuf)); mutex_exit(sc->sc_lock); error = uiomove(zerobuf, k, uio); mutex_enter(sc->sc_lock); n -= k; } return error; } static int uio_fetcher_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, audio_stream_t *p, int max_used) { uio_fetcher_t *this; int size; int stream_space; int error; KASSERT(mutex_owned(sc->sc_lock)); KASSERT(!cpu_intr_p()); KASSERT(!cpu_softintr_p()); this = (uio_fetcher_t *)self; this->last_used = audio_stream_get_used(p); if (this->last_used >= this->usedhigh) return 0; /* * uio_fetcher ignores max_used and move the data as * much as possible in order to return the correct value * for audio_prinfo::seek and kfilters. */ stream_space = audio_stream_get_space(p); size = min(this->uio->uio_resid, stream_space); /* the first fragment of the space */ stream_space = p->end - p->inp; if (stream_space >= size) { mutex_exit(sc->sc_lock); error = uiomove(p->inp, size, this->uio); mutex_enter(sc->sc_lock); if (error) return error; p->inp = audio_stream_add_inp(p, p->inp, size); } else { mutex_exit(sc->sc_lock); error = uiomove(p->inp, stream_space, this->uio); mutex_enter(sc->sc_lock); if (error) return error; p->inp = audio_stream_add_inp(p, p->inp, stream_space); mutex_exit(sc->sc_lock); error = uiomove(p->start, size - stream_space, this->uio); mutex_enter(sc->sc_lock); if (error) return error; p->inp = audio_stream_add_inp(p, p->inp, size - stream_space); } this->last_used = audio_stream_get_used(p); return 0; } static int null_fetcher_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, audio_stream_t *p, int max_used) { return 0; } static void uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h) { this->base.fetch_to = uio_fetcher_fetch_to; this->uio = u; this->usedhigh = h; } int audio_write(struct audio_softc *sc, struct uio *uio, int ioflag, struct virtual_channel *vc) { uio_fetcher_t ufetcher; audio_stream_t stream; struct audio_ringbuffer *cb; stream_fetcher_t *fetcher; stream_filter_t *filter; uint8_t *inp, *einp; int saveerror, error, m, cc, used; KASSERT(mutex_owned(sc->sc_lock)); if (sc->hw_if == NULL) return ENXIO; cb = &vc->sc_mpr; DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n", sc, uio->uio_resid, audio_stream_get_used(vc->sc_pustream), vc->sc_mpr.usedhigh)); if (vc->sc_mpr.mmapped) return EINVAL; if (uio->uio_resid == 0) { sc->sc_eof++; return 0; } #ifdef AUDIO_PM_IDLE if (device_is_active(&sc->dev) || sc->sc_idle) device_active(&sc->dev, DVA_SYSTEM); #endif /* * If half-duplex and currently recording, throw away data. */ if (!vc->sc_full_duplex && (vc->sc_mode & AUMODE_RECORD)) { uio->uio_offset += uio->uio_resid; uio->uio_resid = 0; DPRINTF(("audio_write: half-dpx read busy\n")); return 0; } if (!(vc->sc_mode & AUMODE_PLAY_ALL) && vc->sc_playdrop > 0) { m = min(vc->sc_playdrop, uio->uio_resid); DPRINTF(("audio_write: playdrop %d\n", m)); uio->uio_offset += m; uio->uio_resid -= m; vc->sc_playdrop -= m; if (uio->uio_resid == 0) return 0; } /** * setup filter pipeline */ uio_fetcher_ctor(&ufetcher, uio, vc->sc_mpr.usedhigh); if (vc->sc_npfilters > 0) { fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base; } else { fetcher = &ufetcher.base; } error = 0; while (uio->uio_resid > 0 && !error) { /* wait if the first buffer is occupied */ while ((used = audio_stream_get_used(vc->sc_pustream)) >= cb->usedhigh) { DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d " "hiwat=%d\n", used, cb->usedlow, cb->usedhigh)); if (ioflag & IO_NDELAY) return EWOULDBLOCK; error = audio_waitio(sc, &sc->sc_wchan, vc); if (sc->sc_dying) error = EIO; if (error) return error; } inp = cb->s.inp; cb->copying = true; stream = cb->s; used = stream.used; /* Write to the sc_pustream as much as possible. */ if (vc->sc_npfilters > 0) { filter = vc->sc_pfilters[0]; filter->set_fetcher(filter, &ufetcher.base); fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base; if (sc->sc_usemixer) cc = sc->sc_mixring.sc_mpr.blksize * 2; else cc = vc->sc_mpr.blksize * 2; error = fetcher->fetch_to(sc, fetcher, &stream, cc); if (error != 0) { fetcher = &ufetcher.base; cc = vc->sc_pustream->end - vc->sc_pustream->start; error = fetcher->fetch_to(sc, fetcher, vc->sc_pustream, cc); } } else { fetcher = &ufetcher.base; cc = stream.end - stream.start; error = fetcher->fetch_to(sc, fetcher, &stream, cc); } if (vc->sc_npfilters > 0) { cb->fstamp += ufetcher.last_used - audio_stream_get_used(vc->sc_pustream); } cb->s.used += stream.used - used; cb->s.inp = stream.inp; einp = cb->s.inp; /* * If the interrupt routine wants the last block filled AND * the copy did not fill the last block completely it needs to * be padded. */ if (cb->needfill && inp < einp && (inp - cb->s.start) / cb->blksize == (einp - cb->s.start) / cb->blksize) { /* Figure out how many bytes to a block boundary. */ cc = cb->blksize - (einp - cb->s.start) % cb->blksize; DPRINTF(("audio_write: partial fill %d\n", cc)); } else cc = 0; cb->needfill = false; cb->copying = false; if (!vc->sc_pbus && !cb->pause) { saveerror = error; error = audiostartp(sc, vc); if (saveerror != 0) { /* Report the first error that occurred. */ error = saveerror; } } if (cc != 0) { DPRINTFN(1, ("audio_write: fill %d\n", cc)); audio_fill_silence(&cb->s.param, einp, cc); } } return error; } int audio_ioctl(dev_t dev, struct audio_softc *sc, u_long cmd, void *addr, int flag, struct lwp *l, struct audio_chan *chan) { const struct audio_hw_if *hw; struct audio_chan *pchan; struct virtual_channel *vc; struct audio_offset *ao; u_long stamp; int error, offs, fd; bool rbus, pbus; KASSERT(mutex_owned(sc->sc_lock)); if (sc->sc_usemixer) { SIMPLEQ_FOREACH(pchan, &sc->sc_audiochan, entries) { if (pchan->chan == chan->deschan) break; } if (pchan == NULL) return ENXIO; } else pchan = chan; vc = pchan->vc; DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); hw = sc->hw_if; if (hw == NULL) return ENXIO; error = 0; switch (cmd) { case AUDIO_GETCHAN: if ((int *)addr != NULL) *(int*)addr = chan->chan; break; case AUDIO_SETCHAN: if ((int *)addr != NULL && *(int*)addr > 0) chan->deschan = *(int*)addr; break; case FIONBIO: /* All handled in the upper FS layer. */ break; case FIONREAD: *(int *)addr = audio_stream_get_used(vc->sc_rustream); break; case FIOASYNC: if (*(int *)addr) { if (sc->sc_async_audio != 0) error = EBUSY; else sc->sc_async_audio = pchan->chan; DPRINTF(("audio_ioctl: FIOASYNC chan %d\n", pchan->chan)); } else sc->sc_async_audio = 0; break; case AUDIO_FLUSH: DPRINTF(("AUDIO_FLUSH\n")); rbus = vc->sc_rbus; pbus = vc->sc_pbus; mutex_enter(sc->sc_intr_lock); audio_clear(sc, vc); error = audio_initbufs(sc, vc); if (error) { mutex_exit(sc->sc_intr_lock); return error; } mutex_exit(sc->sc_intr_lock); if ((vc->sc_mode & AUMODE_PLAY) && !vc->sc_pbus && pbus) error = audiostartp(sc, vc); if (!error && (vc->sc_mode & AUMODE_RECORD) && !vc->sc_rbus && rbus) error = audiostartr(sc, vc); break; /* * Number of read (write) samples dropped. We don't know where or * when they were dropped. */ case AUDIO_RERROR: *(int *)addr = vc->sc_mrr.drops; break; case AUDIO_PERROR: *(int *)addr = vc->sc_mpr.drops; break; /* * Offsets into buffer. */ case AUDIO_GETIOFFS: ao = (struct audio_offset *)addr; HW_LOCK(vc); /* figure out where next DMA will start */ stamp = vc->sc_rustream == &vc->sc_mrr.s ? vc->sc_mrr.stamp : vc->sc_mrr.fstamp; offs = vc->sc_rustream->inp - vc->sc_rustream->start; HW_UNLOCK(vc); ao->samples = stamp; ao->deltablks = (stamp / vc->sc_mrr.blksize) - (vc->sc_mrr.stamp_last / vc->sc_mrr.blksize); vc->sc_mrr.stamp_last = stamp; ao->offset = offs; break; case AUDIO_GETOOFFS: ao = (struct audio_offset *)addr; HW_LOCK(vc); /* figure out where next DMA will start */ stamp = vc->sc_pustream == &vc->sc_mpr.s ? vc->sc_mpr.stamp : vc->sc_mpr.fstamp; offs = vc->sc_pustream->outp - vc->sc_pustream->start + vc->sc_mpr.blksize; HW_UNLOCK(vc); ao->samples = stamp; ao->deltablks = (stamp / vc->sc_mpr.blksize) - (vc->sc_mpr.stamp_last / vc->sc_mpr.blksize); vc->sc_mpr.stamp_last = stamp; if (vc->sc_pustream->start + offs >= vc->sc_pustream->end) offs = 0; ao->offset = offs; break; /* * How many bytes will elapse until mike hears the first * sample of what we write next? */ case AUDIO_WSEEK: *(u_long *)addr = audio_stream_get_used(vc->sc_pustream); break; case AUDIO_SETINFO: DPRINTF(("AUDIO_SETINFO mode=0x%x\n", vc->sc_mode)); error = audiosetinfo(sc, (struct audio_info *)addr, false, vc); if (!error && ISDEVSOUND(dev)) { error = audiogetinfo(sc, &sc->sc_ai, 0, vc); sc->sc_aivalid = true; } break; case AUDIO_GETINFO: DPRINTF(("AUDIO_GETINFO\n")); error = audiogetinfo(sc, (struct audio_info *)addr, 0, vc); break; case AUDIO_GETBUFINFO: DPRINTF(("AUDIO_GETBUFINFO\n")); error = audiogetinfo(sc, (struct audio_info *)addr, 1, vc); break; case AUDIO_DRAIN: DPRINTF(("AUDIO_DRAIN\n")); mutex_enter(sc->sc_intr_lock); error = audio_drain(sc, pchan->vc); if (!error && sc->sc_opens == 1 && hw->drain) error = hw->drain(sc->hw_hdl); mutex_exit(sc->sc_intr_lock); break; case AUDIO_GETDEV: DPRINTF(("AUDIO_GETDEV\n")); error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_GETENC: DPRINTF(("AUDIO_GETENC\n")); error = audio_query_encoding(sc, (struct audio_encoding *)addr); break; case AUDIO_GETFD: DPRINTF(("AUDIO_GETFD\n")); *(int *)addr = vc->sc_full_duplex; break; case AUDIO_SETFD: DPRINTF(("AUDIO_SETFD\n")); fd = *(int *)addr; if (audio_get_props(sc) & AUDIO_PROP_FULLDUPLEX) { if (hw->setfd) error = hw->setfd(sc->hw_hdl, fd); else error = 0; if (!error) vc->sc_full_duplex = fd; } else { if (fd) error = ENOTTY; else error = 0; } break; case AUDIO_GETPROPS: DPRINTF(("AUDIO_GETPROPS\n")); *(int *)addr = audio_get_props(sc); break; default: if (hw->dev_ioctl) { error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); } else { DPRINTF(("audio_ioctl: unknown ioctl\n")); error = EINVAL; } break; } DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); return error; } int audio_poll(struct audio_softc *sc, int events, struct lwp *l, struct virtual_channel *vc) { int revents; int used; KASSERT(mutex_owned(sc->sc_lock)); DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, vc->sc_mode)); revents = 0; HW_LOCK(vc); if (events & (POLLIN | POLLRDNORM)) { used = audio_stream_get_used(vc->sc_rustream); /* * If half duplex and playing, audio_read() will generate * silence at the play rate; poll for silence being * available. Otherwise, poll for recorded sound. */ if ((!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY)) ? vc->sc_mpr.stamp > vc->sc_wstamp : used > vc->sc_mrr.usedlow) revents |= events & (POLLIN | POLLRDNORM); } if (events & (POLLOUT | POLLWRNORM)) { used = audio_stream_get_used(vc->sc_pustream); /* * If half duplex and recording, audio_write() will throw * away play data, which means we are always ready to write. * Otherwise, poll for play buffer being below its low water * mark. */ if ((!vc->sc_full_duplex && (vc->sc_mode & AUMODE_RECORD)) || (!(vc->sc_mode & AUMODE_PLAY_ALL) && vc->sc_playdrop > 0) || (used <= vc->sc_mpr.usedlow)) revents |= events & (POLLOUT | POLLWRNORM); } HW_UNLOCK(vc); if (revents == 0) { if (events & (POLLIN | POLLRDNORM)) selrecord(l, &sc->sc_rsel); if (events & (POLLOUT | POLLWRNORM)) selrecord(l, &sc->sc_wsel); } return revents; } static void filt_audiordetach(struct knote *kn) { struct audio_softc *sc; struct audio_chan *chan; dev_t dev; chan = kn->kn_hook; dev = chan->dev; sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return; mutex_enter(sc->sc_intr_lock); SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext); mutex_exit(sc->sc_intr_lock); } static int filt_audioread(struct knote *kn, long hint) { struct audio_softc *sc; struct audio_chan *chan; struct virtual_channel *vc; dev_t dev; chan = kn->kn_hook; dev = chan->dev; sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return ENXIO; vc = chan->vc; mutex_enter(sc->sc_intr_lock); if (!vc->sc_full_duplex && (vc->sc_mode & AUMODE_PLAY)) kn->kn_data = vc->sc_mpr.stamp - vc->sc_wstamp; else kn->kn_data = audio_stream_get_used(vc->sc_rustream) - vc->sc_mrr.usedlow; mutex_exit(sc->sc_intr_lock); return kn->kn_data > 0; } static const struct filterops audioread_filtops = { 1, NULL, filt_audiordetach, filt_audioread }; static void filt_audiowdetach(struct knote *kn) { struct audio_softc *sc; struct audio_chan *chan; dev_t dev; chan = kn->kn_hook; dev = chan->dev; sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return; mutex_enter(sc->sc_intr_lock); SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext); mutex_exit(sc->sc_intr_lock); } static int filt_audiowrite(struct knote *kn, long hint) { struct audio_softc *sc; struct audio_chan *chan; audio_stream_t *stream; dev_t dev; chan = kn->kn_hook; dev = chan->dev; sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return ENXIO; mutex_enter(sc->sc_intr_lock); stream = chan->vc->sc_pustream; kn->kn_data = (stream->end - stream->start) - audio_stream_get_used(stream); mutex_exit(sc->sc_intr_lock); return kn->kn_data > 0; } static const struct filterops audiowrite_filtops = { 1, NULL, filt_audiowdetach, filt_audiowrite }; int audio_kqfilter(struct audio_chan *chan, struct knote *kn) { struct audio_softc *sc; struct klist *klist; dev_t dev; dev = chan->dev; sc = device_lookup_private(&audio_cd, AUDIOUNIT(dev)); if (sc == NULL) return ENXIO; switch (kn->kn_filter) { case EVFILT_READ: klist = &sc->sc_rsel.sel_klist; kn->kn_fop = &audioread_filtops; break; case EVFILT_WRITE: klist = &sc->sc_wsel.sel_klist; kn->kn_fop = &audiowrite_filtops; break; default: return EINVAL; } kn->kn_hook = chan; mutex_enter(sc->sc_intr_lock); SLIST_INSERT_HEAD(klist, kn, kn_selnext); mutex_exit(sc->sc_intr_lock); return 0; } int audio_mmap(struct audio_softc *sc, off_t *offp, size_t len, int prot, int *flagsp, int *advicep, struct uvm_object **uobjp, int *maxprotp, struct virtual_channel *vc) { struct audio_ringbuffer *cb; KASSERT(mutex_owned(sc->sc_lock)); if (sc->hw_if == NULL) return ENXIO; DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)(*offp), prot)); if (!(audio_get_props(sc) & AUDIO_PROP_MMAP)) return ENOTSUP; if (*offp < 0) return EINVAL; #if 0 /* XXX * The idea here was to use the protection to determine if * we are mapping the read or write buffer, but it fails. * The VM system is broken in (at least) two ways. * 1) If you map memory VM_PROT_WRITE you SIGSEGV * when writing to it, so VM_PROT_READ|VM_PROT_WRITE * has to be used for mmapping the play buffer. * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE * audio_mmap will get called at some point with VM_PROT_READ * only. * So, alas, we always map the play buffer for now. */ if (prot == (VM_PROT_READ|VM_PROT_WRITE) || prot == VM_PROT_WRITE) cb = &vc->sc_mpr; else if (prot == VM_PROT_READ) cb = &vc->sc_mrr; else return EINVAL; #else cb = &vc->sc_mpr; #endif if (len > cb->s.bufsize || *offp > cb->s.bufsize - len) return EOVERFLOW; if (!cb->mmapped) { cb->mmapped = true; if (cb == &vc->sc_mpr) { audio_fill_silence(&cb->s.param, cb->s.start, cb->s.bufsize); vc->sc_pustream = &cb->s; if (!vc->sc_pbus && !vc->sc_mpr.pause) (void)audiostartp(sc, vc); } else if (cb == &vc->sc_mrr) { vc->sc_rustream = &cb->s; if (!vc->sc_rbus && !sc->sc_mixring.sc_mrr.pause) (void)audiostartr(sc, vc); } } /* get ringbuffer */ *uobjp = cb->uobj; /* Acquire a reference for the mmap. munmap will release.*/ uao_reference(*uobjp); *maxprotp = prot; *advicep = UVM_ADV_RANDOM; *flagsp = MAP_SHARED; return 0; } int audiostartr(struct audio_softc *sc, struct virtual_channel *vc) { int error; KASSERT(mutex_owned(sc->sc_lock)); DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n", vc->sc_mrr.s.start, audio_stream_get_used(&vc->sc_mrr.s), vc->sc_mrr.usedhigh, vc->sc_mrr.mmapped)); if (!audio_can_capture(sc)) return EINVAL; if (vc == sc->sc_hwvc && sc->sc_usemixer) return 0; error = 0; if (sc->sc_rec_started == false) { mutex_enter(sc->sc_intr_lock); error = mix_read(sc); if (sc->sc_usemixer) cv_broadcast(&sc->sc_rcondvar); mutex_exit(sc->sc_intr_lock); } vc->sc_rbus = true; return error; } int audiostartp(struct audio_softc *sc, struct virtual_channel *vc) { int error, used; KASSERT(mutex_owned(sc->sc_lock)); error = 0; used = audio_stream_get_used(&vc->sc_mpr.s); DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n", vc->sc_mpr.s.start, used, vc->sc_mpr.usedhigh, vc->sc_mpr.blksize, vc->sc_mpr.mmapped)); if (!audio_can_playback(sc)) return EINVAL; if (vc == sc->sc_hwvc && sc->sc_usemixer) return 0; int blksize; if (sc->sc_usemixer) blksize = sc->sc_mixring.sc_mpr.blksize; else blksize = vc->sc_mpr.blksize; if (!vc->sc_mpr.mmapped && used < blksize) { cv_broadcast(&sc->sc_wchan); DPRINTF(("%s: wakeup and return\n", __func__)); return 0; } vc->sc_pbus = true; if (sc->sc_trigger_started == false) { if (sc->sc_usemixer) { audio_mix(sc); audio_mix(sc); audio_mix(sc); } mutex_enter(sc->sc_intr_lock); error = mix_write(sc); if (error) goto done; if (sc->sc_usemixer) { vc = sc->sc_hwvc; vc->sc_mpr.s.outp = audio_stream_add_outp(&vc->sc_mpr.s, vc->sc_mpr.s.outp, vc->sc_mpr.blksize); error = mix_write(sc); cv_broadcast(&sc->sc_condvar); } done: mutex_exit(sc->sc_intr_lock); } return error; } static void audio_softintr_rd(void *cookie) { struct audio_softc *sc = cookie; proc_t *p; pid_t pid; mutex_enter(sc->sc_lock); cv_broadcast(&sc->sc_rchan); selnotify(&sc->sc_rsel, 0, NOTE_SUBMIT); if ((pid = sc->sc_async_audio) != 0) { DPRINTFN(3, ("audio_softintr_rd: sending SIGIO %d\n", pid)); mutex_enter(proc_lock); if ((p = proc_find(pid)) != NULL) psignal(p, SIGIO); mutex_exit(proc_lock); } mutex_exit(sc->sc_lock); } static void audio_softintr_wr(void *cookie) { struct audio_softc *sc = cookie; proc_t *p; pid_t pid; mutex_enter(sc->sc_lock); cv_broadcast(&sc->sc_wchan); selnotify(&sc->sc_wsel, 0, NOTE_SUBMIT); if ((pid = sc->sc_async_audio) != 0) { DPRINTFN(3, ("audio_softintr_wr: sending SIGIO %d\n", pid)); mutex_enter(proc_lock); if ((p = proc_find(pid)) != NULL) psignal(p, SIGIO); mutex_exit(proc_lock); } mutex_exit(sc->sc_lock); } /* * Called from HW driver module on completion of DMA output. * Start output of new block, wrap in ring buffer if needed. * If no more buffers to play, output zero instead. * Do a wakeup if necessary. */ void audio_pint(void *v) { struct audio_softc *sc; struct audio_ringbuffer *cb; struct virtual_channel *vc; int blksize, cc, used; sc = v; vc = sc->sc_hwvc; blksize = vc->sc_mpr.blksize; if (sc->sc_dying == true || sc->sc_trigger_started == false) return; if (sc->sc_usemixer) cb = &sc->sc_mixring.sc_mpr; else cb = &vc->sc_mpr; if (vc->sc_draining && cb->drops != sc->sc_last_drops) { vc->sc_mpr.drops += blksize; cv_broadcast(&sc->sc_wchan); } sc->sc_last_drops = cb->drops; vc->sc_mpr.s.outp = audio_stream_add_outp(&vc->sc_mpr.s, vc->sc_mpr.s.outp, blksize); if (audio_stream_get_used(&cb->s) < blksize) { DPRINTFN(3, ("HW RING - INSERT SILENCE\n")); used = blksize; while (used > 0) { cc = cb->s.end - cb->s.inp; if (cc > used) cc = used; audio_fill_silence(&cb->s.param, cb->s.inp, cc); cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, cc); used -= cc; } vc->sc_mpr.drops += blksize; } mix_write(sc); if (sc->sc_usemixer) cv_broadcast(&sc->sc_condvar); else cv_broadcast(&sc->sc_wchan); } void audio_mix(void *v) { stream_fetcher_t null_fetcher; struct audio_softc *sc; struct audio_chan *chan; struct virtual_channel *vc; struct audio_ringbuffer *cb; stream_fetcher_t *fetcher; uint8_t *inp; int cc, cc1, used, blksize; sc = v; DPRINTF(("PINT MIX\n")); sc->schedule_rih = false; sc->schedule_wih = false; sc->sc_writeme = false; if (sc->sc_dying == true) return; blksize = sc->sc_mixring.sc_mpr.blksize; SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { if (!sc->sc_opens) break; /* ignore interrupt if not open */ vc = chan->vc; if (!vc->sc_open) continue; if (!vc->sc_pbus) continue; cb = &vc->sc_mpr; sc->sc_writeme = true; inp = cb->s.inp; cb->stamp += blksize; if (cb->mmapped) { DPRINTF(("audio_pint: vc=%p mmapped outp=%p cc=%d " "inp=%p\n", vc, cb->s.outp, blksize, cb->s.inp)); mutex_enter(sc->sc_intr_lock); mix_func(sc, cb, vc); cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); mutex_exit(sc->sc_intr_lock); continue; } #ifdef AUDIO_INTR_TIME { struct timeval tv; int64_t t; microtime(&tv); t = (int64_t)tv.tv_sec * 1000000 + tv.tv_usec; if (sc->sc_pnintr) { int64_t lastdelta, totdelta; lastdelta = t - sc->sc_plastintr - sc->sc_pblktime; if (lastdelta > sc->sc_pblktime / 3) { printf("audio: play interrupt(%d) off " "relative by %" PRId64 " us " "(%" PRId64 ")\n", sc->sc_pnintr, lastdelta, sc->sc_pblktime); } totdelta = t - sc->sc_pfirstintr - sc->sc_pblktime * sc->sc_pnintr; if (totdelta > sc->sc_pblktime) { printf("audio: play interrupt(%d) " "off absolute by %" PRId64 " us " "(%" PRId64 ") (LOST)\n", sc->sc_pnintr, totdelta, sc->sc_pblktime); sc->sc_pnintr++; /* avoid repeated messages */ } } else sc->sc_pfirstintr = t; sc->sc_plastintr = t; sc->sc_pnintr++; } #endif used = audio_stream_get_used(&cb->s); /* * "used <= cb->usedlow" should be "used < blksize" ideally. * Some HW drivers such as uaudio(4) does not call audio_pint() * at accurate timing. If used < blksize, uaudio(4) already * request transfer of garbage data. */ if (used <= sc->sc_hwvc->sc_mpr.usedlow && !cb->copying && vc->sc_npfilters > 0) { /* we might have data in filter pipeline */ null_fetcher.fetch_to = null_fetcher_fetch_to; fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base; vc->sc_pfilters[0]->set_fetcher(vc->sc_pfilters[0], &null_fetcher); used = audio_stream_get_used(vc->sc_pustream); cc = cb->s.end - cb->s.start; if (blksize * 2 < cc) cc = blksize * 2; fetcher->fetch_to(sc, fetcher, &cb->s, cc); cb->fstamp += used - audio_stream_get_used(vc->sc_pustream); used = audio_stream_get_used(&cb->s); } if (used < blksize) { /* we don't have a full block to use */ if (cb->copying) { /* writer is in progress, don't disturb */ cb->needfill = true; DPRINTFN(1, ("audio_pint: copying in " "progress\n")); } else { DPRINTF(("audio_pint: used < blksize vc=%p " "used=%d blksize=%d\n", vc, used, blksize)); inp = cb->s.inp; cc = blksize - (inp - cb->s.start) % blksize; if (cb->pause) cb->pdrops += cc; else { cb->drops += cc; vc->sc_playdrop += cc; } audio_fill_silence(&cb->s.param, inp, cc); cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc); /* Clear next block to keep ahead of the DMA. */ used = audio_stream_get_used(&cb->s); if (used + blksize < cb->s.end - cb->s.start) { audio_fill_silence(&cb->s.param, cb->s.inp, blksize); } } } DPRINTFN(5, ("audio_pint: vc=%p outp=%p used=%d cc=%d\n", vc, cb->s.outp, used, blksize)); mutex_enter(sc->sc_intr_lock); mix_func(sc, cb, vc); mutex_exit(sc->sc_intr_lock); cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); DPRINTFN(2, ("audio_pint: vc=%p mode=%d pause=%d used=%d " "lowat=%d\n", vc, vc->sc_mode, cb->pause, audio_stream_get_used(&cb->s), cb->usedlow)); if ((vc->sc_mode & AUMODE_PLAY) && !cb->pause) { if (audio_stream_get_used(vc->sc_pustream) <= cb->usedlow) sc->schedule_wih = true; } /* Possible to return one or more "phantom blocks" now. */ if (!vc->sc_full_duplex && vc->sc_mode & AUMODE_RECORD) sc->schedule_rih = true; } mutex_enter(sc->sc_intr_lock); vc = sc->sc_hwvc; cb = &sc->sc_mixring.sc_mpr; inp = cb->s.inp; cc = blksize - (inp - cb->s.start) % blksize; if (sc->sc_writeme == false) { DPRINTFN(3, ("MIX RING EMPTY - INSERT SILENCE\n")); audio_fill_silence(&vc->sc_pustream->param, inp, cc); sc->sc_mixring.sc_mpr.drops += cc; } else cc = blksize; cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, cc); cc = blksize; cc1 = sc->sc_mixring.sc_mpr.s.end - sc->sc_mixring.sc_mpr.s.inp; if (cc1 < cc) { audio_fill_silence(&vc->sc_pustream->param, sc->sc_mixring.sc_mpr.s.inp, cc1); cc -= cc1; audio_fill_silence(&vc->sc_pustream->param, sc->sc_mixring.sc_mpr.s.start, cc); } else audio_fill_silence(&vc->sc_pustream->param, sc->sc_mixring.sc_mpr.s.inp, cc); mutex_exit(sc->sc_intr_lock); kpreempt_disable(); if (sc->schedule_wih == true) softint_schedule(sc->sc_sih_wr); if (sc->schedule_rih == true) softint_schedule(sc->sc_sih_rd); kpreempt_enable(); } /* * Called from HW driver module on completion of DMA input. * Mark it as input in the ring buffer (fiddle pointers). * Do a wakeup if necessary. */ void audio_rint(void *v) { struct audio_softc *sc; int blksize; sc = v; KASSERT(mutex_owned(sc->sc_intr_lock)); if (sc->sc_dying == true || sc->sc_rec_started == false) return; blksize = audio_stream_get_used(&sc->sc_mixring.sc_mrr.s); sc->sc_mixring.sc_mrr.s.outp = audio_stream_add_outp(&sc->sc_mixring.sc_mrr.s, sc->sc_mixring.sc_mrr.s.outp, blksize); mix_read(sc); if (sc->sc_usemixer) cv_broadcast(&sc->sc_rcondvar); else cv_broadcast(&sc->sc_rchan); } void audio_upmix(void *v) { stream_fetcher_t null_fetcher; struct audio_softc *sc; struct audio_chan *chan; struct audio_ringbuffer *cb; stream_fetcher_t *last_fetcher; struct virtual_channel *vc; int cc, used, blksize, cc1; sc = v; blksize = sc->sc_mixring.sc_mrr.blksize; SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { if (!sc->sc_opens) break; /* ignore interrupt if not open */ vc = chan->vc; if (!(vc->sc_open & AUOPEN_READ)) continue; if (!vc->sc_rbus) continue; cb = &vc->sc_mrr; blksize = audio_stream_get_used(&sc->sc_mixring.sc_mrr.s); if (audio_stream_get_space(&cb->s) < blksize) { cb->drops += blksize; cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, sc->sc_mixring.sc_mrr.blksize); continue; } cc = blksize; if (cb->s.inp + blksize > cb->s.end) cc = cb->s.end - cb->s.inp; mutex_enter(sc->sc_intr_lock); memcpy(cb->s.inp, sc->sc_mixring.sc_mrr.s.start, cc); if (cc < blksize && cc != 0) { cc1 = cc; cc = blksize - cc; memcpy(cb->s.start, sc->sc_mixring.sc_mrr.s.start + cc1, cc); } mutex_exit(sc->sc_intr_lock); cc = blksize; recswvol_func(sc, cb, blksize, vc); cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize); cb->stamp += blksize; if (cb->mmapped) { DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n", cb->s.inp, blksize)); continue; } #ifdef AUDIO_INTR_TIME { struct timeval tv; int64_t t; microtime(&tv); t = (int64_t)tv.tv_sec * 1000000 + tv.tv_usec; if (sc->sc_rnintr) { int64_t lastdelta, totdelta; lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime; if (lastdelta > sc->sc_rblktime / 5) { printf("audio: record interrupt(%d) " "off relative by %" PRId64 " us " "(%" PRId64 ")\n", sc->sc_rnintr, lastdelta, sc->sc_rblktime); } totdelta = t - sc->sc_rfirstintr - sc->sc_rblktime * sc->sc_rnintr; if (totdelta > sc->sc_rblktime / 2) { sc->sc_rnintr++; printf("audio: record interrupt(%d) " "off absolute by %" PRId64 " us " "(%" PRId64 ")\n", sc->sc_rnintr, totdelta, sc->sc_rblktime); sc->sc_rnintr++; /* avoid repeated messages */ } } else sc->sc_rfirstintr = t; sc->sc_rlastintr = t; sc->sc_rnintr++; } #endif if (!cb->pause && vc->sc_nrfilters > 0) { null_fetcher.fetch_to = null_fetcher_fetch_to; last_fetcher = &vc->sc_rfilters[vc->sc_nrfilters - 1]->base; vc->sc_rfilters[0]->set_fetcher(vc->sc_rfilters[0], &null_fetcher); used = audio_stream_get_used(vc->sc_rustream); cc = vc->sc_rustream->end - vc->sc_rustream->start; last_fetcher->fetch_to (sc, last_fetcher, vc->sc_rustream, cc); cb->fstamp += audio_stream_get_used(vc->sc_rustream) - used; /* XXX what should do for error? */ } used = audio_stream_get_used(&vc->sc_mrr.s); if (cb->pause) { DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops)); cb->pdrops += blksize; cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); } else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) { DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops)); cb->drops += blksize; cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize); } } kpreempt_disable(); softint_schedule(sc->sc_sih_rd); kpreempt_enable(); } int audio_check_params(struct audio_params *p) { if (p->encoding == AUDIO_ENCODING_PCM16) { if (p->precision == 8) p->encoding = AUDIO_ENCODING_ULINEAR; else p->encoding = AUDIO_ENCODING_SLINEAR; } else if (p->encoding == AUDIO_ENCODING_PCM8) { if (p->precision == 8) p->encoding = AUDIO_ENCODING_ULINEAR; else return EINVAL; } if (p->encoding == AUDIO_ENCODING_SLINEAR) #if BYTE_ORDER == LITTLE_ENDIAN p->encoding = AUDIO_ENCODING_SLINEAR_LE; #else p->encoding = AUDIO_ENCODING_SLINEAR_BE; #endif if (p->encoding == AUDIO_ENCODING_ULINEAR) #if BYTE_ORDER == LITTLE_ENDIAN p->encoding = AUDIO_ENCODING_ULINEAR_LE; #else p->encoding = AUDIO_ENCODING_ULINEAR_BE; #endif switch (p->encoding) { case AUDIO_ENCODING_ULAW: case AUDIO_ENCODING_ALAW: if (p->precision != 8) return EINVAL; break; case AUDIO_ENCODING_ADPCM: if (p->precision != 4 && p->precision != 8) return EINVAL; break; case AUDIO_ENCODING_SLINEAR_LE: case AUDIO_ENCODING_SLINEAR_BE: case AUDIO_ENCODING_ULINEAR_LE: case AUDIO_ENCODING_ULINEAR_BE: /* XXX is: our zero-fill can handle any multiple of 8 */ if (p->precision != 8 && p->precision != 16 && p->precision != 24 && p->precision != 32) return EINVAL; if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE) p->encoding = AUDIO_ENCODING_SLINEAR_LE; if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE) p->encoding = AUDIO_ENCODING_ULINEAR_LE; if (p->validbits > p->precision) return EINVAL; break; case AUDIO_ENCODING_MPEG_L1_STREAM: case AUDIO_ENCODING_MPEG_L1_PACKETS: case AUDIO_ENCODING_MPEG_L1_SYSTEM: case AUDIO_ENCODING_MPEG_L2_STREAM: case AUDIO_ENCODING_MPEG_L2_PACKETS: case AUDIO_ENCODING_MPEG_L2_SYSTEM: case AUDIO_ENCODING_AC3: break; default: return EINVAL; } /* sanity check # of channels*/ if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS) return EINVAL; return 0; } /* * set some parameters from sc->sc_vchan_params. */ static int audio_set_vchan_defaults(struct audio_softc *sc, u_int mode) { struct virtual_channel *vc; struct audio_info ai; int error; KASSERT(mutex_owned(sc->sc_lock)); vc = sc->sc_hwvc; /* default parameters */ vc->sc_rparams = sc->sc_vchan_params; vc->sc_pparams = sc->sc_vchan_params; AUDIO_INITINFO(&ai); ai.record.sample_rate = sc->sc_vchan_params.sample_rate; ai.record.encoding = sc->sc_vchan_params.encoding; ai.record.channels = sc->sc_vchan_params.channels; ai.record.precision = sc->sc_vchan_params.precision; ai.record.pause = false; ai.play.sample_rate = sc->sc_vchan_params.sample_rate; ai.play.encoding = sc->sc_vchan_params.encoding; ai.play.channels = sc->sc_vchan_params.channels; ai.play.precision = sc->sc_vchan_params.precision; ai.play.pause = false; ai.mode = mode; sc->sc_format->encoding = sc->sc_vchan_params.encoding; sc->sc_format->channels = sc->sc_vchan_params.channels; sc->sc_format->precision = sc->sc_vchan_params.precision; sc->sc_format->validbits = sc->sc_vchan_params.precision; sc->sc_format->frequency_type = 1; sc->sc_format->frequency[0] = sc->sc_vchan_params.sample_rate; auconv_delete_encodings(sc->sc_encodings); error = auconv_create_encodings(sc->sc_format, VAUDIO_NFORMATS, &sc->sc_encodings); if (error == 0) error = audiosetinfo(sc, &ai, true, vc); return error; } int audio_set_defaults(struct audio_softc *sc, u_int mode, struct virtual_channel *vc) { struct audio_info ai; KASSERT(mutex_owned(sc->sc_lock)); /* default parameters */ vc->sc_rparams = audio_default; vc->sc_pparams = audio_default; AUDIO_INITINFO(&ai); ai.record.sample_rate = vc->sc_rparams.sample_rate; ai.record.encoding = vc->sc_rparams.encoding; ai.record.channels = vc->sc_rparams.channels; ai.record.precision = vc->sc_rparams.precision; ai.record.pause = false; ai.play.sample_rate = vc->sc_pparams.sample_rate; ai.play.encoding = vc->sc_pparams.encoding; ai.play.channels = vc->sc_pparams.channels; ai.play.precision = vc->sc_pparams.precision; ai.play.pause = false; ai.mode = mode; return audiosetinfo(sc, &ai, true, vc); } int au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r) { KASSERT(mutex_owned(sc->sc_lock)); ct->type = AUDIO_MIXER_VALUE; ct->un.value.num_channels = 2; ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; if (audio_set_port(sc, ct) == 0) return 0; ct->un.value.num_channels = 1; ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2; return audio_set_port(sc, ct); } int au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports, int gain, int balance) { mixer_ctrl_t ct; int i, error; int l, r; u_int mask; int nset; KASSERT(mutex_owned(sc->sc_lock)); if (balance == AUDIO_MID_BALANCE) { l = r = gain; } else if (balance < AUDIO_MID_BALANCE) { l = gain; r = (balance * gain) / AUDIO_MID_BALANCE; } else { r = gain; l = ((AUDIO_RIGHT_BALANCE - balance) * gain) / AUDIO_MID_BALANCE; } DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n", gain, balance, l, r)); if (ports->index == -1) { usemaster: if (ports->master == -1) return 0; /* just ignore it silently */ ct.dev = ports->master; error = au_set_lr_value(sc, &ct, l, r); } else { ct.dev = ports->index; if (ports->isenum) { ct.type = AUDIO_MIXER_ENUM; error = audio_get_port(sc, &ct); if (error) return error; if (ports->isdual) { if (ports->cur_port == -1) ct.dev = ports->master; else ct.dev = ports->miport[ports->cur_port]; error = au_set_lr_value(sc, &ct, l, r); } else { for(i = 0; i < ports->nports; i++) if (ports->misel[i] == ct.un.ord) { ct.dev = ports->miport[i]; if (ct.dev == -1 || au_set_lr_value(sc, &ct, l, r)) goto usemaster; else break; } } } else { ct.type = AUDIO_MIXER_SET; error = audio_get_port(sc, &ct); if (error) return error; mask = ct.un.mask; nset = 0; for(i = 0; i < ports->nports; i++) { if (ports->misel[i] & mask) { ct.dev = ports->miport[i]; if (ct.dev != -1 && au_set_lr_value(sc, &ct, l, r) == 0) nset++; } } if (nset == 0) goto usemaster; } } if (!error) mixer_signal(sc); return error; } int au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r) { int error; KASSERT(mutex_owned(sc->sc_lock)); ct->un.value.num_channels = 2; if (audio_get_port(sc, ct) == 0) { *l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT]; *r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; } else { ct->un.value.num_channels = 1; error = audio_get_port(sc, ct); if (error) return error; *r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO]; } return 0; } void au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports, u_int *pgain, u_char *pbalance) { mixer_ctrl_t ct; int i, l, r, n; int lgain, rgain; KASSERT(mutex_owned(sc->sc_lock)); lgain = AUDIO_MAX_GAIN / 2; rgain = AUDIO_MAX_GAIN / 2; if (ports->index == -1) { usemaster: if (ports->master == -1) goto bad; ct.dev = ports->master; ct.type = AUDIO_MIXER_VALUE; if (au_get_lr_value(sc, &ct, &lgain, &rgain)) goto bad; } else { ct.dev = ports->index; if (ports->isenum) { ct.type = AUDIO_MIXER_ENUM; if (audio_get_port(sc, &ct)) goto bad; ct.type = AUDIO_MIXER_VALUE; if (ports->isdual) { if (ports->cur_port == -1) ct.dev = ports->master; else ct.dev = ports->miport[ports->cur_port]; au_get_lr_value(sc, &ct, &lgain, &rgain); } else { for(i = 0; i < ports->nports; i++) if (ports->misel[i] == ct.un.ord) { ct.dev = ports->miport[i]; if (ct.dev == -1 || au_get_lr_value(sc, &ct, &lgain, &rgain)) goto usemaster; else break; } } } else { ct.type = AUDIO_MIXER_SET; if (audio_get_port(sc, &ct)) goto bad; ct.type = AUDIO_MIXER_VALUE; lgain = rgain = n = 0; for(i = 0; i < ports->nports; i++) { if (ports->misel[i] & ct.un.mask) { ct.dev = ports->miport[i]; if (ct.dev == -1 || au_get_lr_value(sc, &ct, &l, &r)) goto usemaster; else { lgain += l; rgain += r; n++; } } } if (n != 0) { lgain /= n; rgain /= n; } } } bad: if (lgain == rgain) { /* handles lgain==rgain==0 */ *pgain = lgain; *pbalance = AUDIO_MID_BALANCE; } else if (lgain < rgain) { *pgain = rgain; /* balance should be > AUDIO_MID_BALANCE */ *pbalance = AUDIO_RIGHT_BALANCE - (AUDIO_MID_BALANCE * lgain) / rgain; } else /* lgain > rgain */ { *pgain = lgain; /* balance should be < AUDIO_MID_BALANCE */ *pbalance = (AUDIO_MID_BALANCE * rgain) / lgain; } } int au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port) { mixer_ctrl_t ct; int i, error, use_mixerout; KASSERT(mutex_owned(sc->sc_lock)); use_mixerout = 1; if (port == 0) { if (ports->allports == 0) return 0; /* Allow this special case. */ else if (ports->isdual) { if (ports->cur_port == -1) { return 0; } else { port = ports->aumask[ports->cur_port]; ports->cur_port = -1; use_mixerout = 0; } } } if (ports->index == -1) return EINVAL; ct.dev = ports->index; if (ports->isenum) { if (port & (port-1)) return EINVAL; /* Only one port allowed */ ct.type = AUDIO_MIXER_ENUM; error = EINVAL; for(i = 0; i < ports->nports; i++) if (ports->aumask[i] == port) { if (ports->isdual && use_mixerout) { ct.un.ord = ports->mixerout; ports->cur_port = i; } else { ct.un.ord = ports->misel[i]; } error = audio_set_port(sc, &ct); break; } } else { ct.type = AUDIO_MIXER_SET; ct.un.mask = 0; for(i = 0; i < ports->nports; i++) if (ports->aumask[i] & port) ct.un.mask |= ports->misel[i]; if (port != 0 && ct.un.mask == 0) error = EINVAL; else error = audio_set_port(sc, &ct); } if (!error) mixer_signal(sc); return error; } int au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports) { mixer_ctrl_t ct; int i, aumask; KASSERT(mutex_owned(sc->sc_lock)); if (ports->index == -1) return 0; ct.dev = ports->index; ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET; if (audio_get_port(sc, &ct)) return 0; aumask = 0; if (ports->isenum) { if (ports->isdual && ports->cur_port != -1) { if (ports->mixerout == ct.un.ord) aumask = ports->aumask[ports->cur_port]; else ports->cur_port = -1; } if (aumask == 0) for(i = 0; i < ports->nports; i++) if (ports->misel[i] == ct.un.ord) aumask = ports->aumask[i]; } else { for(i = 0; i < ports->nports; i++) if (ct.un.mask & ports->misel[i]) aumask |= ports->aumask[i]; } return aumask; } int audiosetinfo(struct audio_softc *sc, struct audio_info *ai, bool reset, struct virtual_channel *vc) { stream_filter_list_t pfilters, rfilters; audio_params_t pp, rp; struct audio_prinfo *r, *p; const struct audio_hw_if *hw; audio_stream_t *oldpus, *oldrus; int setmode; int error; int np, nr; unsigned int blks; u_int gain; bool rbus, pbus; bool cleared, modechange, pausechange; u_char balance; KASSERT(mutex_owned(sc->sc_lock)); hw = sc->hw_if; if (hw == NULL) /* HW has not attached */ return ENXIO; DPRINTF(("%s sc=%p ai=%p\n", __func__, sc, ai)); r = &ai->record; p = &ai->play; rbus = vc->sc_rbus; pbus = vc->sc_pbus; error = 0; cleared = false; modechange = false; pausechange = false; pp = vc->sc_pparams; /* Temporary encoding storage in */ rp = vc->sc_rparams; /* case setting the modes fails. */ nr = np = 0; if (SPECIFIED(p->sample_rate)) { pp.sample_rate = p->sample_rate; np++; } if (SPECIFIED(r->sample_rate)) { rp.sample_rate = r->sample_rate; nr++; } if (SPECIFIED(p->encoding)) { pp.encoding = p->encoding; np++; } if (SPECIFIED(r->encoding)) { rp.encoding = r->encoding; nr++; } if (SPECIFIED(p->precision)) { pp.precision = p->precision; /* we don't have API to specify validbits */ pp.validbits = p->precision; np++; } if (SPECIFIED(r->precision)) { rp.precision = r->precision; /* we don't have API to specify validbits */ rp.validbits = r->precision; nr++; } if (SPECIFIED(p->channels)) { pp.channels = p->channels; np++; } if (SPECIFIED(r->channels)) { rp.channels = r->channels; nr++; } if (!audio_can_capture(sc)) nr = 0; if (!audio_can_playback(sc)) np = 0; #ifdef AUDIO_DEBUG if (audiodebug && nr > 0) audio_print_params("audiosetinfo() Setting record params:", &rp); if (audiodebug && np > 0) audio_print_params("audiosetinfo() Setting play params:", &pp); #endif if (nr > 0 && (error = audio_check_params(&rp))) return error; if (np > 0 && (error = audio_check_params(&pp))) return error; setmode = 0; if (nr > 0) { if (!cleared) { audio_clear_intr_unlocked(sc, vc); cleared = true; } modechange = true; setmode |= AUMODE_RECORD; } if (np > 0) { if (!cleared) { audio_clear_intr_unlocked(sc, vc); cleared = true; } modechange = true; setmode |= AUMODE_PLAY; } if (SPECIFIED(ai->mode)) { if (!cleared) { audio_clear_intr_unlocked(sc, vc); cleared = true; } modechange = true; vc->sc_mode = ai->mode; if (vc->sc_mode & AUMODE_PLAY_ALL) vc->sc_mode |= AUMODE_PLAY; if ((vc->sc_mode & AUMODE_PLAY) && !vc->sc_full_duplex) /* Play takes precedence */ vc->sc_mode &= ~AUMODE_RECORD; } oldpus = vc->sc_pustream; oldrus = vc->sc_rustream; if (modechange || reset) { int indep; indep = audio_get_props(sc) & AUDIO_PROP_INDEPENDENT; if (!indep) { if (setmode == AUMODE_RECORD) pp = rp; else if (setmode == AUMODE_PLAY) rp = pp; } memset(&pfilters, 0, sizeof(pfilters)); memset(&rfilters, 0, sizeof(rfilters)); pfilters.append = stream_filter_list_append; pfilters.prepend = stream_filter_list_prepend; pfilters.set = stream_filter_list_set; rfilters.append = stream_filter_list_append; rfilters.prepend = stream_filter_list_prepend; rfilters.set = stream_filter_list_set; /* Some device drivers change channels/sample_rate and change * no channels/sample_rate. */ error = audio_set_params(sc, setmode, vc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp, &pfilters, &rfilters, vc); if (error) { DPRINTF(("%s: audio_set_params() failed with %d\n", __func__, error)); goto cleanup; } audio_check_params(&pp); audio_check_params(&rp); if (!indep) { /* XXX for !indep device, we have to use the same * parameters for the hardware, not userland */ if (setmode == AUMODE_RECORD) { pp = rp; } else if (setmode == AUMODE_PLAY) { rp = pp; } } if (vc->sc_mpr.mmapped && pfilters.req_size > 0) { DPRINTF(("%s: mmapped, and filters are requested.\n", __func__)); error = EINVAL; goto cleanup; } /* construct new filter chain */ if (setmode & AUMODE_PLAY) { error = audio_setup_pfilters(sc, &pp, &pfilters, vc); if (error) goto cleanup; } if (setmode & AUMODE_RECORD) { error = audio_setup_rfilters(sc, &rp, &rfilters, vc); if (error) goto cleanup; } DPRINTF(("%s: filter setup is completed.\n", __func__)); /* userland formats */ vc->sc_pparams = pp; vc->sc_rparams = rp; } #ifdef AUDIO_DEBUG if (audiodebug > 1 && nr > 0) { audio_print_params("audiosetinfo() After setting record params:", &vc->sc_rparams); } if (audiodebug > 1 && np > 0) { audio_print_params("audiosetinfo() After setting play params:", &vc->sc_pparams); } #endif if (SPECIFIED(p->port)) { if (!cleared) { audio_clear_intr_unlocked(sc, vc); cleared = true; } error = au_set_port(sc, &sc->sc_outports, p->port); if (error) goto cleanup; } if (SPECIFIED(r->port)) { if (!cleared) { audio_clear_intr_unlocked(sc, vc); cleared = true; } error = au_set_port(sc, &sc->sc_inports, r->port); if (error) goto cleanup; } if (SPECIFIED(p->gain)) vc->sc_swvol = p->gain; if (SPECIFIED(r->gain)) vc->sc_recswvol = r->gain; if (SPECIFIED_CH(p->balance)) { au_get_gain(sc, &sc->sc_outports, &gain, &balance); error = au_set_gain(sc, &sc->sc_outports, gain, p->balance); if (error) goto cleanup; } if (SPECIFIED_CH(r->balance)) { au_get_gain(sc, &sc->sc_inports, &gain, &balance); error = au_set_gain(sc, &sc->sc_inports, gain, r->balance); if (error) goto cleanup; } if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) { mixer_ctrl_t ct; ct.dev = sc->sc_monitor_port; ct.type = AUDIO_MIXER_VALUE; ct.un.value.num_channels = 1; ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain; error = audio_set_port(sc, &ct); if (error) goto cleanup; } if (SPECIFIED_CH(p->pause)) { pbus = !p->pause; if (pbus != !vc->sc_mpr.pause) { vc->sc_mpr.pause = p->pause; pausechange = true; } } if (SPECIFIED_CH(r->pause)) { rbus = !r->pause; if (rbus != !vc->sc_mrr.pause) { vc->sc_mrr.pause = r->pause; pausechange = true; } } if (SPECIFIED(ai->mode)) { if (vc->sc_mode & AUMODE_PLAY) audio_init_play(sc, vc); if (vc->sc_mode & AUMODE_RECORD) audio_init_record(sc, vc); } if (nr > 0) audio_setblksize(sc, vc, ai->blocksize, AUMODE_RECORD); if (np > 0) audio_setblksize(sc, vc, ai->blocksize, AUMODE_PLAY); if (hw->commit_settings && sc->sc_opens + sc->sc_recopens == 0) { error = hw->commit_settings(sc->hw_hdl); if (error) goto cleanup; } vc->sc_lastinfo = *ai; vc->sc_lastinfovalid = true; cleanup: if (error == 0 && (cleared || pausechange || reset)) { int init_error; init_error = (pausechange == 1 && reset == 0) ? 0 : audio_initbufs(sc, vc); if (init_error) goto err; if (reset || vc->sc_pustream != oldpus || vc->sc_rustream != oldrus) audio_calcwater(sc, vc); if ((vc->sc_mode & AUMODE_PLAY) && pbus && !vc->sc_pbus) init_error = audiostartp(sc, vc); if (!init_error && (vc->sc_mode & AUMODE_RECORD) && rbus && !vc->sc_rbus) init_error = audiostartr(sc, vc); err: if (init_error) return init_error; } /* Change water marks after initializing the buffers. */ if (SPECIFIED(ai->hiwat)) { blks = ai->hiwat; if (blks > vc->sc_mpr.maxblks) blks = vc->sc_mpr.maxblks; if (blks < PREFILL_BLOCKS + 1) blks = PREFILL_BLOCKS + 1; vc->sc_mpr.usedhigh = blks * vc->sc_mpr.blksize; } if (SPECIFIED(ai->lowat)) { blks = ai->lowat; if (blks > vc->sc_mpr.maxblks - 1) blks = vc->sc_mpr.maxblks - 1; if (blks < PREFILL_BLOCKS) blks = PREFILL_BLOCKS; vc->sc_mpr.usedlow = blks * vc->sc_mpr.blksize; } if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) { if (vc->sc_mpr.usedlow > vc->sc_mpr.usedhigh - vc->sc_mpr.blksize) { vc->sc_mpr.usedlow = vc->sc_mpr.usedhigh - vc->sc_mpr.blksize; } } return error; } int audiogetinfo(struct audio_softc *sc, struct audio_info *ai, int buf_only_mode, struct virtual_channel *vc) { struct audio_prinfo *r, *p; const struct audio_hw_if *hw; KASSERT(mutex_owned(sc->sc_lock)); r = &ai->record; p = &ai->play; hw = sc->hw_if; if (hw == NULL) /* HW has not attached */ return ENXIO; p->sample_rate = vc->sc_pparams.sample_rate; r->sample_rate = vc->sc_rparams.sample_rate; p->channels = vc->sc_pparams.channels; r->channels = vc->sc_rparams.channels; p->precision = vc->sc_pparams.precision; r->precision = vc->sc_rparams.precision; p->encoding = vc->sc_pparams.encoding; r->encoding = vc->sc_rparams.encoding; if (buf_only_mode) { r->port = 0; p->port = 0; r->avail_ports = 0; p->avail_ports = 0; r->gain = 0; r->balance = 0; p->gain = 0; p->balance = 0; } else { r->port = au_get_port(sc, &sc->sc_inports); p->port = au_get_port(sc, &sc->sc_outports); r->avail_ports = sc->sc_inports.allports; p->avail_ports = sc->sc_outports.allports; au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance); au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance); } if (sc->sc_monitor_port != -1 && buf_only_mode == 0) { mixer_ctrl_t ct; ct.dev = sc->sc_monitor_port; ct.type = AUDIO_MIXER_VALUE; ct.un.value.num_channels = 1; if (audio_get_port(sc, &ct)) ai->monitor_gain = 0; else ai->monitor_gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO]; } else ai->monitor_gain = 0; p->seek = audio_stream_get_used(vc->sc_pustream); r->seek = audio_stream_get_used(vc->sc_rustream); /* * XXX samples should be a value for userland data. * But drops is a value for HW data. */ p->samples = (vc->sc_pustream == &vc->sc_mpr.s ? vc->sc_mpr.stamp : vc->sc_mpr.fstamp) - vc->sc_mpr.drops; r->samples = (vc->sc_rustream == &vc->sc_mrr.s ? vc->sc_mrr.stamp : vc->sc_mrr.fstamp) - vc->sc_mrr.drops; p->eof = sc->sc_eof; r->eof = 0; p->pause = vc->sc_mpr.pause; r->pause = vc->sc_mrr.pause; p->error = vc->sc_mpr.drops != 0; r->error = vc->sc_mrr.drops != 0; p->waiting = r->waiting = 0; /* open never hangs */ p->open = (vc->sc_open & AUOPEN_WRITE) != 0; r->open = (vc->sc_open & AUOPEN_READ) != 0; p->active = vc->sc_pbus; r->active = vc->sc_rbus; p->buffer_size = vc->sc_pustream ? vc->sc_pustream->bufsize : 0; r->buffer_size = vc->sc_rustream ? vc->sc_rustream->bufsize : 0; ai->blocksize = vc->sc_mpr.blksize; if (vc->sc_mpr.blksize > 0) { ai->hiwat = vc->sc_mpr.usedhigh / vc->sc_mpr.blksize; ai->lowat = vc->sc_mpr.usedlow / vc->sc_mpr.blksize; } else ai->hiwat = ai->lowat = 0; ai->mode = vc->sc_mode; return 0; } /* * Mixer driver */ int mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt, struct lwp *l, struct file **nfp) { struct file *fp; struct audio_chan *chan; int error, fd; KASSERT(mutex_owned(sc->sc_lock)); if (sc->hw_if == NULL) return ENXIO; DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc)); error = fd_allocfile(&fp, &fd); if (error) return error; chan = kmem_zalloc(sizeof(struct audio_chan), KM_SLEEP); chan->dev = dev; error = fd_clone(fp, fd, flags, &audio_fileops, chan); KASSERT(error == EMOVEFD); *nfp = fp; return error; } /* * Remove a process from those to be signalled on mixer activity. */ static void mixer_remove(struct audio_softc *sc) { struct mixer_asyncs **pm, *m; pid_t pid; KASSERT(mutex_owned(sc->sc_lock)); pid = curproc->p_pid; for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) { if ((*pm)->pid == pid) { m = *pm; *pm = m->next; kmem_free(m, sizeof(*m)); return; } } } /* * Signal all processes waiting for the mixer. */ static void mixer_signal(struct audio_softc *sc) { struct mixer_asyncs *m; proc_t *p; for (m = sc->sc_async_mixer; m; m = m->next) { mutex_enter(proc_lock); if ((p = proc_find(m->pid)) != NULL) psignal(p, SIGIO); mutex_exit(proc_lock); } } /* * Close a mixer device */ /* ARGSUSED */ int mixer_close(struct audio_softc *sc, int flags, struct audio_chan *chan) { KASSERT(mutex_owned(sc->sc_lock)); if (sc->hw_if == NULL) return ENXIO; DPRINTF(("mixer_close: sc %p\n", sc)); mixer_remove(sc); return 0; } int mixer_ioctl(struct audio_softc *sc, u_long cmd, void *addr, int flag, struct lwp *l) { const struct audio_hw_if *hw; struct mixer_asyncs *ma; mixer_ctrl_t *mc; int error; DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff)); hw = sc->hw_if; if (hw == NULL) return ENXIO; error = EINVAL; /* we can return cached values if we are sleeping */ if (cmd != AUDIO_MIXER_READ) device_active(sc->dev, DVA_SYSTEM); switch (cmd) { case FIOASYNC: if (*(int *)addr) { ma = kmem_alloc(sizeof(struct mixer_asyncs), KM_SLEEP); } else { ma = NULL; } mixer_remove(sc); /* remove old entry */ if (ma != NULL) { ma->next = sc->sc_async_mixer; ma->pid = curproc->p_pid; sc->sc_async_mixer = ma; } error = 0; break; case AUDIO_GETDEV: DPRINTF(("AUDIO_GETDEV\n")); error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr); break; case AUDIO_MIXER_DEVINFO: DPRINTF(("AUDIO_MIXER_DEVINFO\n")); ((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */ error = audio_query_devinfo(sc, (mixer_devinfo_t *)addr); break; case AUDIO_MIXER_READ: DPRINTF(("AUDIO_MIXER_READ\n")); mc = (mixer_ctrl_t *)addr; if (device_is_active(sc->sc_dev)) error = audio_get_port(sc, mc); else if (mc->dev < 0 || mc->dev >= sc->sc_nmixer_states) error = ENXIO; else { int dev = mc->dev; memcpy(mc, &sc->sc_mixer_state[dev], sizeof(mixer_ctrl_t)); error = 0; } break; case AUDIO_MIXER_WRITE: DPRINTF(("AUDIO_MIXER_WRITE\n")); error = audio_set_port(sc, (mixer_ctrl_t *)addr); if (!error && hw->commit_settings) error = hw->commit_settings(sc->hw_hdl); if (!error) mixer_signal(sc); break; default: if (hw->dev_ioctl) { error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l); } else error = EINVAL; break; } DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n", IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error)); return error; } #endif /* NAUDIO > 0 */ #if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0) #include #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; default: panic("audioprint: unknown type %d", arg->type); } aprint_normal("%s at %s", type, pnp); } return UNCONF; } #endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */ #if NAUDIO > 0 device_t audio_get_device(struct audio_softc *sc) { return sc->sc_dev; } #endif #if NAUDIO > 0 static void audio_mixer_capture(struct audio_softc *sc) { mixer_devinfo_t mi; mixer_ctrl_t *mc; KASSERT(mutex_owned(sc->sc_lock)); for (mi.index = 0;; mi.index++) { if (audio_query_devinfo(sc, &mi) != 0) break; KASSERT(mi.index < sc->sc_nmixer_states); if (mi.type == AUDIO_MIXER_CLASS) continue; mc = &sc->sc_mixer_state[mi.index]; mc->dev = mi.index; mc->type = mi.type; mc->un.value.num_channels = mi.un.v.num_channels; (void)audio_get_port(sc, mc); } return; } static void audio_mixer_restore(struct audio_softc *sc) { mixer_devinfo_t mi; mixer_ctrl_t *mc; KASSERT(mutex_owned(sc->sc_lock)); for (mi.index = 0; ; mi.index++) { if (audio_query_devinfo(sc, &mi) != 0) break; if (mi.type == AUDIO_MIXER_CLASS) continue; mc = &sc->sc_mixer_state[mi.index]; (void)audio_set_port(sc, mc); } if (sc->hw_if->commit_settings) sc->hw_if->commit_settings(sc->hw_hdl); return; } #ifdef AUDIO_PM_IDLE static void audio_idle(void *arg) { device_t dv = arg; struct audio_softc *sc = device_private(dv); #ifdef PNP_DEBUG extern int pnp_debug_idle; if (pnp_debug_idle) printf("%s: idle handler called\n", device_xname(dv)); #endif sc->sc_idle = true; /* XXX joerg Make pmf_device_suspend handle children? */ if (!pmf_device_suspend(dv, PMF_Q_SELF)) return; if (!pmf_device_suspend(sc->sc_dev, PMF_Q_SELF)) pmf_device_resume(dv, PMF_Q_SELF); } static void audio_activity(device_t dv, devactive_t type) { struct audio_softc *sc = device_private(dv); if (type != DVA_SYSTEM) return; callout_schedule(&sc->sc_idle_counter, audio_idle_timeout * hz); sc->sc_idle = false; if (!device_is_active(dv)) { /* XXX joerg How to deal with a failing resume... */ pmf_device_resume(sc->sc_dev, PMF_Q_SELF); pmf_device_resume(dv, PMF_Q_SELF); } } #endif static bool audio_suspend(device_t dv, const pmf_qual_t *qual) { struct audio_softc *sc = device_private(dv); struct audio_chan *chan; const struct audio_hw_if *hwp = sc->hw_if; struct virtual_channel *vc; bool pbus, rbus; pbus = rbus = false; mutex_enter(sc->sc_lock); audio_mixer_capture(sc); SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { vc = chan->vc; if (vc->sc_pbus && !pbus) pbus = true; if (vc->sc_rbus && !rbus) rbus = true; } mutex_enter(sc->sc_intr_lock); if (pbus == true) hwp->halt_output(sc->hw_hdl); if (rbus == true) hwp->halt_input(sc->hw_hdl); mutex_exit(sc->sc_intr_lock); #ifdef AUDIO_PM_IDLE callout_halt(&sc->sc_idle_counter, sc->sc_lock); #endif mutex_exit(sc->sc_lock); return true; } static bool audio_resume(device_t dv, const pmf_qual_t *qual) { struct audio_softc *sc = device_private(dv); struct audio_chan *chan; struct virtual_channel *vc; mutex_enter(sc->sc_lock); sc->sc_trigger_started = false; sc->sc_rec_started = false; audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); audio_mixer_restore(sc); SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { vc = chan->vc; if (vc->sc_lastinfovalid == true) audiosetinfo(sc, &vc->sc_lastinfo, true, vc); if (vc->sc_pbus == true && !vc->sc_mpr.pause) audiostartp(sc, vc); if (vc->sc_rbus == true && !vc->sc_mrr.pause) audiostartr(sc, vc); } mutex_exit(sc->sc_lock); return true; } static void audio_volume_down(device_t dv) { struct audio_softc *sc = device_private(dv); mixer_devinfo_t mi; int newgain; u_int gain; u_char balance; mutex_enter(sc->sc_lock); if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { mi.index = sc->sc_outports.master; mi.un.v.delta = 0; if (audio_query_devinfo(sc, &mi) == 0) { au_get_gain(sc, &sc->sc_outports, &gain, &balance); newgain = gain - mi.un.v.delta; if (newgain < AUDIO_MIN_GAIN) newgain = AUDIO_MIN_GAIN; au_set_gain(sc, &sc->sc_outports, newgain, balance); } } mutex_exit(sc->sc_lock); } static void audio_volume_up(device_t dv) { struct audio_softc *sc = device_private(dv); mixer_devinfo_t mi; u_int gain, newgain; u_char balance; mutex_enter(sc->sc_lock); if (sc->sc_outports.index == -1 && sc->sc_outports.master != -1) { mi.index = sc->sc_outports.master; mi.un.v.delta = 0; if (audio_query_devinfo(sc, &mi) == 0) { au_get_gain(sc, &sc->sc_outports, &gain, &balance); newgain = gain + mi.un.v.delta; if (newgain > AUDIO_MAX_GAIN) newgain = AUDIO_MAX_GAIN; au_set_gain(sc, &sc->sc_outports, newgain, balance); } } mutex_exit(sc->sc_lock); } static void audio_volume_toggle(device_t dv) { struct audio_softc *sc = device_private(dv); u_int gain, newgain; u_char balance; mutex_enter(sc->sc_lock); au_get_gain(sc, &sc->sc_outports, &gain, &balance); if (gain != 0) { sc->sc_lastgain = gain; newgain = 0; } else newgain = sc->sc_lastgain; au_set_gain(sc, &sc->sc_outports, newgain, balance); mutex_exit(sc->sc_lock); } static int audio_get_props(struct audio_softc *sc) { const struct audio_hw_if *hw; int props; KASSERT(mutex_owned(sc->sc_lock)); hw = sc->hw_if; props = hw->get_props(sc->hw_hdl); /* * if neither playback nor capture properties are reported, * assume both are supported by the device driver */ if ((props & (AUDIO_PROP_PLAYBACK|AUDIO_PROP_CAPTURE)) == 0) props |= (AUDIO_PROP_PLAYBACK | AUDIO_PROP_CAPTURE); props |= AUDIO_PROP_MMAP; return props; } static bool audio_can_playback(struct audio_softc *sc) { return audio_get_props(sc) & AUDIO_PROP_PLAYBACK ? true : false; } static bool audio_can_capture(struct audio_softc *sc) { return audio_get_props(sc) & AUDIO_PROP_CAPTURE ? true : false; } int mix_read(void *arg) { struct audio_softc *sc = arg; struct virtual_channel *vc; stream_filter_t *filter; stream_fetcher_t *fetcher; stream_fetcher_t null_fetcher; int cc, cc1, blksize, error; uint8_t *inp; vc = sc->sc_hwvc; blksize = vc->sc_mrr.blksize; cc = blksize; error = 0; if (sc->hw_if->trigger_input && sc->sc_rec_started == false) { DPRINTF(("%s: call trigger_input\n", __func__)); sc->sc_rec_started = true; error = sc->hw_if->trigger_input(sc->hw_hdl, vc->sc_mrr.s.start, vc->sc_mrr.s.end, vc->sc_mrr.blksize, audio_rint, (void *)sc, &vc->sc_mrr.s.param); } else if (sc->hw_if->start_input) { DPRINTF(("%s: call start_input\n", __func__)); sc->sc_rec_started = true; error = sc->hw_if->start_input(sc->hw_hdl, vc->sc_mrr.s.inp, vc->sc_mrr.blksize, audio_rint, (void *)sc); } if (error) { /* XXX does this really help? */ DPRINTF(("audio_upmix restart failed: %d\n", error)); audio_clear(sc, sc->sc_hwvc); sc->sc_rec_started = false; return error; } inp = vc->sc_mrr.s.inp; vc->sc_mrr.s.inp = audio_stream_add_inp(&vc->sc_mrr.s, inp, cc); if (vc->sc_nrfilters > 0) { cc = vc->sc_rustream->end - vc->sc_rustream->start; null_fetcher.fetch_to = null_fetcher_fetch_to; filter = vc->sc_rfilters[0]; filter->set_fetcher(filter, &null_fetcher); fetcher = &vc->sc_rfilters[vc->sc_nrfilters - 1]->base; fetcher->fetch_to(sc, fetcher, vc->sc_rustream, cc); } blksize = audio_stream_get_used(vc->sc_rustream); cc1 = blksize; if (vc->sc_rustream->outp + blksize > vc->sc_rustream->end) cc1 = vc->sc_rustream->end - vc->sc_rustream->outp; memcpy(sc->sc_mixring.sc_mrr.s.start, vc->sc_rustream->outp, cc1); if (cc1 < blksize) { memcpy(sc->sc_mixring.sc_mrr.s.start + cc1, vc->sc_rustream->start, blksize - cc1); } sc->sc_mixring.sc_mrr.s.inp = audio_stream_add_inp(&sc->sc_mixring.sc_mrr.s, sc->sc_mixring.sc_mrr.s.inp, blksize); vc->sc_rustream->outp = audio_stream_add_outp(vc->sc_rustream, vc->sc_rustream->outp, blksize); return error; } int mix_write(void *arg) { struct audio_softc *sc = arg; struct virtual_channel *vc; stream_filter_t *filter; stream_fetcher_t *fetcher; stream_fetcher_t null_fetcher; int cc, cc1, cc2, error, used; const uint8_t *orig; uint8_t *tocopy; vc = sc->sc_hwvc; error = 0; if (sc->sc_usemixer && audio_stream_get_used(vc->sc_pustream) <= sc->sc_mixring.sc_mpr.blksize) { tocopy = vc->sc_pustream->inp; orig = sc->sc_mixring.sc_mpr.s.outp; used = sc->sc_mixring.sc_mpr.blksize; while (used > 0) { cc = used; cc1 = vc->sc_pustream->end - tocopy; cc2 = sc->sc_mixring.sc_mpr.s.end - orig; if (cc > cc1) cc = cc1; if (cc > cc2) cc = cc2; memcpy(tocopy, orig, cc); orig += cc; tocopy += cc; if (tocopy >= vc->sc_pustream->end) tocopy = vc->sc_pustream->start; if (orig >= sc->sc_mixring.sc_mpr.s.end) orig = sc->sc_mixring.sc_mpr.s.start; used -= cc; } vc->sc_pustream->inp = audio_stream_add_inp(vc->sc_pustream, vc->sc_pustream->inp, sc->sc_mixring.sc_mpr.blksize); sc->sc_mixring.sc_mpr.s.outp = audio_stream_add_outp(&sc->sc_mixring.sc_mpr.s, sc->sc_mixring.sc_mpr.s.outp, sc->sc_mixring.sc_mpr.blksize); } if (vc->sc_npfilters > 0 && (sc->sc_usemixer || sc->sc_trigger_started)) { null_fetcher.fetch_to = null_fetcher_fetch_to; filter = vc->sc_pfilters[0]; filter->set_fetcher(filter, &null_fetcher); fetcher = &vc->sc_pfilters[vc->sc_npfilters - 1]->base; fetcher->fetch_to(sc, fetcher, &vc->sc_mpr.s, vc->sc_mpr.blksize * 2); } if (sc->hw_if->trigger_output && sc->sc_trigger_started == false) { DPRINTF(("%s: call trigger_output\n", __func__)); sc->sc_trigger_started = true; error = sc->hw_if->trigger_output(sc->hw_hdl, vc->sc_mpr.s.start, vc->sc_mpr.s.end, vc->sc_mpr.blksize, audio_pint, (void *)sc, &vc->sc_mpr.s.param); } else if (sc->hw_if->start_output) { DPRINTF(("%s: call start_output\n", __func__)); sc->sc_trigger_started = true; error = sc->hw_if->start_output(sc->hw_hdl, __UNCONST(vc->sc_mpr.s.outp), vc->sc_mpr.blksize, audio_pint, (void *)sc); } if (error) { /* XXX does this really help? */ DPRINTF(("audio_mix restart failed: %d\n", error)); audio_clear(sc, sc->sc_hwvc); sc->sc_trigger_started = false; } return error; } #define DEF_MIX_FUNC(name, type, bigger_type, MINVAL, MAXVAL) \ static void \ mix_func##name(struct audio_softc *sc, struct audio_ringbuffer *cb, \ struct virtual_channel *vc) \ { \ int blksize, cc, cc1, cc2, m, resid; \ bigger_type product; \ bigger_type result; \ type *orig, *tomix; \ \ blksize = sc->sc_mixring.sc_mpr.blksize; \ resid = blksize; \ \ tomix = __UNCONST(cb->s.outp); \ orig = (type *)(sc->sc_mixring.sc_mpr.s.inp); \ \ while (resid > 0) { \ cc = resid; \ cc1 = sc->sc_mixring.sc_mpr.s.end - \ (uint8_t *)orig; \ cc2 = cb->s.end - (uint8_t *)tomix; \ if (cc > cc1) \ cc = cc1; \ if (cc > cc2) \ cc = cc2; \ \ for (m = 0; m < (cc / (name / NBBY)); m++) { \ if (vc->sc_swvol == 255) \ goto vol_done; \ tomix[m] = (bigger_type)tomix[m] * \ (bigger_type)(vc->sc_swvol) / 255; \ vol_done: \ result = (bigger_type)orig[m] + tomix[m]; \ if (sc->sc_opens == 1) \ goto adj_done; \ product = (bigger_type)orig[m] * tomix[m]; \ if (orig[m] > 0 && tomix[m] > 0) \ result -= product / MAXVAL; \ else if (orig[m] < 0 && tomix[m] < 0) \ result -= product / MINVAL; \ adj_done: \ orig[m] = result; \ } \ \ if (&orig[m] >= \ (type *)sc->sc_mixring.sc_mpr.s.end) \ orig = \ (type *)sc->sc_mixring.sc_mpr.s.start; \ if (&tomix[m] >= (type *)cb->s.end) \ tomix = (type *)cb->s.start; \ \ resid -= cc; \ } \ } \ DEF_MIX_FUNC(8, int8_t, int32_t, INT8_MIN, INT8_MAX); DEF_MIX_FUNC(16, int16_t, int32_t, INT16_MIN, INT16_MAX); DEF_MIX_FUNC(32, int32_t, int64_t, INT32_MIN, INT32_MAX); void mix_func(struct audio_softc *sc, struct audio_ringbuffer *cb, struct virtual_channel *vc) { switch (sc->sc_vchan_params.precision) { case 8: mix_func8(sc, cb, vc); break; case 16: mix_func16(sc, cb, vc); break; case 24: case 32: mix_func32(sc, cb, vc); break; default: break; } } #define DEF_RECSWVOL_FUNC(name, type, bigger_type) \ static void \ recswvol_func##name(struct audio_softc *sc, \ struct audio_ringbuffer *cb, size_t blksize, \ struct virtual_channel *vc) \ { \ int cc, cc1, m, resid; \ type *orig; \ \ orig = (type *) cb->s.inp; \ resid = blksize; \ \ while (resid > 0) { \ cc = resid; \ cc1 = cb->s.end - (uint8_t *)orig; \ if (cc > cc1) \ cc = cc1; \ \ for (m = 0; m < (cc / (name / 8)); m++) { \ orig[m] = (bigger_type)(orig[m] * \ (bigger_type)(vc->sc_recswvol) / 256);\ } \ orig = (type *) cb->s.start; \ \ resid -= cc; \ } \ } \ DEF_RECSWVOL_FUNC(8, int8_t, int16_t); DEF_RECSWVOL_FUNC(16, int16_t, int32_t); DEF_RECSWVOL_FUNC(32, int32_t, int64_t); void recswvol_func(struct audio_softc *sc, struct audio_ringbuffer *cb, size_t blksize, struct virtual_channel *vc) { switch (sc->sc_vchan_params.precision) { case 8: recswvol_func8(sc, cb, blksize, vc); break; case 16: recswvol_func16(sc, cb, blksize, vc); break; case 24: case 32: recswvol_func32(sc, cb, blksize, vc); break; default: break; } } static uint8_t * find_vchan_vol(struct audio_softc *sc, int d) { struct audio_chan *chan; size_t j, n = (size_t)d / 2; j = 0; SIMPLEQ_FOREACH(chan, &sc->sc_audiochan, entries) { if (j == n) break; j++; } return (d & 1) == 0 ? &chan->vc->sc_swvol : &chan->vc->sc_recswvol; } static int audio_set_port(struct audio_softc *sc, mixer_ctrl_t *mc) { KASSERT(mutex_owned(sc->sc_lock)); int d = mc->dev - sc->sc_static_nmixer_states; int u = sc->sc_nmixer_states - sc->sc_static_nmixer_states; if (d == -1 || d >= u) return 0; if (d < 0) return sc->hw_if->set_port(sc->hw_hdl, mc); uint8_t *level = &mc->un.value.level[AUDIO_MIXER_LEVEL_MONO]; uint8_t *vol = find_vchan_vol(sc, d); *vol = *level; return 0; } static int audio_get_port(struct audio_softc *sc, mixer_ctrl_t *mc) { KASSERT(mutex_owned(sc->sc_lock)); int d = mc->dev - sc->sc_static_nmixer_states; int u = sc->sc_nmixer_states - sc->sc_static_nmixer_states; if (d == -1 || d >= u) return 0; if (d < 0) return sc->hw_if->get_port(sc->hw_hdl, mc); u_char *level = &mc->un.value.level[AUDIO_MIXER_LEVEL_MONO]; uint8_t *vol = find_vchan_vol(sc, d); *level = *vol; return 0; } static int audio_query_devinfo(struct audio_softc *sc, mixer_devinfo_t *di) { char mixLabel[255]; KASSERT(mutex_owned(sc->sc_lock)); if (di->index >= sc->sc_static_nmixer_states - 1 && di->index < sc->sc_nmixer_states) { if (di->index == sc->sc_static_nmixer_states - 1) { di->mixer_class = sc->sc_static_nmixer_states -1; di->next = di->prev = AUDIO_MIXER_LAST; strcpy(di->label.name, AudioCvirtchan); di->type = AUDIO_MIXER_CLASS; } else if ((di->index - sc->sc_static_nmixer_states) % 2 == 0) { di->mixer_class = sc->sc_static_nmixer_states -1; snprintf(mixLabel, sizeof(mixLabel), AudioNdac"%d", (di->index - sc->sc_static_nmixer_states) / 2); strcpy(di->label.name, mixLabel); di->type = AUDIO_MIXER_VALUE; di->next = di->prev = AUDIO_MIXER_LAST; di->un.v.num_channels = 1; strcpy(di->un.v.units.name, AudioNvolume); } else { di->mixer_class = sc->sc_static_nmixer_states -1; snprintf(mixLabel, sizeof(mixLabel), AudioNmicrophone "%d", (di->index - sc->sc_static_nmixer_states) / 2); strcpy(di->label.name, mixLabel); di->type = AUDIO_MIXER_VALUE; di->next = di->prev = AUDIO_MIXER_LAST; di->un.v.num_channels = 1; strcpy(di->un.v.units.name, AudioNvolume); } return 0; } return sc->hw_if->query_devinfo(sc->hw_hdl, di); } static int audio_set_params(struct audio_softc *sc, int setmode, int usemode, audio_params_t *play, audio_params_t *rec, stream_filter_list_t *pfil, stream_filter_list_t *rfil, struct virtual_channel *vc) { int error = 0; KASSERT(mutex_owned(sc->sc_lock)); if (vc == sc->sc_hwvc && sc->hw_if->set_params != NULL) { sc->sc_ready = true; if (sc->sc_usemixer && sc->sc_vchan_params.precision == 8) play->encoding = rec->encoding = AUDIO_ENCODING_SLINEAR; error = sc->hw_if->set_params(sc->hw_hdl, setmode, usemode, play, rec, pfil, rfil); if (error != 0) sc->sc_ready = false; return error; } if (setmode & AUMODE_PLAY && auconv_set_converter(sc->sc_format, VAUDIO_NFORMATS, AUMODE_PLAY, play, true, pfil) < 0) return EINVAL; if (setmode & AUMODE_RECORD && auconv_set_converter(sc->sc_format, VAUDIO_NFORMATS, AUMODE_RECORD, rec, true, rfil) < 0) return EINVAL; return 0; } static int audio_query_encoding(struct audio_softc *sc, struct audio_encoding *ae) { KASSERT(mutex_owned(sc->sc_lock)); return auconv_query_encoding(sc->sc_encodings, ae); } void audio_play_thread(void *v) { struct audio_softc *sc; sc = (struct audio_softc *)v; for (;;) { mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); kthread_exit(0); } if (!sc->sc_trigger_started) goto play_wait; while (!sc->sc_dying && sc->sc_usemixer && audio_stream_get_used(&sc->sc_mixring.sc_mpr.s) < sc->sc_mixring.sc_mpr.blksize) audio_mix(sc); play_wait: mutex_exit(sc->sc_lock); mutex_enter(sc->sc_intr_lock); cv_wait_sig(&sc->sc_condvar, sc->sc_intr_lock); mutex_exit(sc->sc_intr_lock); } } void audio_rec_thread(void *v) { struct audio_softc *sc; sc = (struct audio_softc *)v; for (;;) { mutex_enter(sc->sc_lock); if (sc->sc_dying) { mutex_exit(sc->sc_lock); kthread_exit(0); } if (!sc->sc_rec_started) goto rec_wait; audio_upmix(sc); rec_wait: mutex_exit(sc->sc_lock); mutex_enter(sc->sc_intr_lock); cv_wait_sig(&sc->sc_rcondvar, sc->sc_intr_lock); mutex_exit(sc->sc_intr_lock); } } /* sysctl helper to set common audio frequency */ static int audio_sysctl_frequency(SYSCTLFN_ARGS) { struct sysctlnode node; struct audio_softc *sc; int t, error; node = *rnode; sc = node.sysctl_data; t = sc->sc_vchan_params.sample_rate; node.sysctl_data = &t; error = sysctl_lookup(SYSCTLFN_CALL(&node)); if (error || newp == NULL) return error; mutex_enter(sc->sc_lock); /* This may not change when a virtual channel is open */ if (sc->sc_opens || sc->sc_recopens) { mutex_exit(sc->sc_lock); return EBUSY; } if (t <= 0) { mutex_exit(sc->sc_lock); return EINVAL; } sc->sc_vchan_params.sample_rate = t; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (error) aprint_error_dev(sc->sc_dev, "Error setting frequency, " "please check hardware capabilities\n"); if (error == 0) audio_calc_latency(sc); mutex_exit(sc->sc_lock); return error; } /* sysctl helper to set common audio precision */ static int audio_sysctl_precision(SYSCTLFN_ARGS) { struct sysctlnode node; struct audio_softc *sc; int t, error; node = *rnode; sc = node.sysctl_data; t = sc->sc_vchan_params.precision; node.sysctl_data = &t; error = sysctl_lookup(SYSCTLFN_CALL(&node)); if (error || newp == NULL) return error; mutex_enter(sc->sc_lock); /* This may not change when a virtual channel is open */ if (sc->sc_opens || sc->sc_recopens) { mutex_exit(sc->sc_lock); return EBUSY; } if (t == 0 || (t != 8 && t != 16 && t != 24 && t != 32)) { mutex_exit(sc->sc_lock); return EINVAL; } sc->sc_vchan_params.precision = t; if (sc->sc_vchan_params.precision != 8) { sc->sc_vchan_params.encoding = #if BYTE_ORDER == LITTLE_ENDIAN AUDIO_ENCODING_SLINEAR_LE; #else AUDIO_ENCODING_SLINEAR_BE; #endif } else sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_LE; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (error) aprint_error_dev(sc->sc_dev, "Error setting precision, " "please check hardware capabilities\n"); if (error == 0) audio_calc_latency(sc); mutex_exit(sc->sc_lock); return error; } /* sysctl helper to enable/disable channel mixing */ static int audio_sysctl_usemixer(SYSCTLFN_ARGS) { struct sysctlnode node; struct audio_softc *sc; bool t; int error; node = *rnode; sc = node.sysctl_data; t = sc->sc_usemixer; node.sysctl_data = &t; error = sysctl_lookup(SYSCTLFN_CALL(&node)); if (error || newp == NULL) return error; mutex_enter(sc->sc_lock); /* This may not change when a virtual channel is open */ if (sc->sc_opens) { mutex_exit(sc->sc_lock); return EBUSY; } sc->sc_usemixer = t; audio_destroy_pfilters(sc->sc_hwvc); audio_destroy_rfilters(sc->sc_hwvc); if (t) { error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (error) aprint_error_dev(sc->sc_dev, "Error setting precision, " "please check hardware capabilities\n"); } if (sc->sc_usemixer) { if (error == 0) audio_calc_latency(sc); } else sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS; mutex_exit(sc->sc_lock); return error; } /* sysctl helper to set common audio channels */ static int audio_sysctl_channels(SYSCTLFN_ARGS) { struct sysctlnode node; struct audio_softc *sc; int t, error; node = *rnode; sc = node.sysctl_data; t = sc->sc_vchan_params.channels; node.sysctl_data = &t; error = sysctl_lookup(SYSCTLFN_CALL(&node)); if (error || newp == NULL) return error; mutex_enter(sc->sc_lock); /* This may not change when a virtual channel is open */ if (sc->sc_opens || sc->sc_recopens) { mutex_exit(sc->sc_lock); return EBUSY; } if (t <= 0 || (t !=1 && t % 2 != 0)) { mutex_exit(sc->sc_lock); return EINVAL; } sc->sc_vchan_params.channels = t; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (error) aprint_error_dev(sc->sc_dev, "Error setting channels, " "please check hardware capabilities\n"); if (error == 0) audio_calc_latency(sc); mutex_exit(sc->sc_lock); return error; } /* sysctl helper to set audio latency */ static int audio_sysctl_latency(SYSCTLFN_ARGS) { struct sysctlnode node; struct audio_softc *sc; int t, error; node = *rnode; sc = node.sysctl_data; t = sc->sc_latency; node.sysctl_data = &t; error = sysctl_lookup(SYSCTLFN_CALL(&node)); if (error || newp == NULL) return error; mutex_enter(sc->sc_lock); /* This may not change when a virtual channel is open */ if (sc->sc_opens || sc->sc_recopens) { mutex_exit(sc->sc_lock); return EBUSY; } if (t < 0 || t > 4000) { mutex_exit(sc->sc_lock); return EINVAL; } if (t == 0) sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS; else sc->sc_latency = (unsigned int)t; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (error) { aprint_error_dev(sc->sc_dev, "Error setting latency, " "latency restored to default\n"); sc->sc_latency = audio_blk_ms * PREFILL_BLOCKS; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); } audio_calc_latency(sc); mutex_exit(sc->sc_lock); return error; } static int vchan_autoconfig(struct audio_softc *sc) { struct virtual_channel *vc; int error, i, j, k; vc = sc->sc_hwvc; error = 0; mutex_enter(sc->sc_lock); #if BYTE_ORDER == LITTLE_ENDIAN sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_LE; #else sc->sc_vchan_params.encoding = AUDIO_ENCODING_SLINEAR_BE; #endif for (i = 0; i < __arraycount(auto_config_precision); i++) { sc->sc_vchan_params.precision = auto_config_precision[i]; sc->sc_vchan_params.validbits = auto_config_precision[i]; for (j = 0; j < __arraycount(auto_config_channels); j++) { sc->sc_vchan_params.channels = auto_config_channels[j]; for (k = 0; k < __arraycount(auto_config_freq); k++) { sc->sc_vchan_params.sample_rate = auto_config_freq[k]; error = audio_set_vchan_defaults(sc, AUMODE_PLAY | AUMODE_PLAY_ALL | AUMODE_RECORD); if (vc->sc_npfilters > 0 && (vc->sc_mpr.s.param.sample_rate != sc->sc_vchan_params.sample_rate || vc->sc_mpr.s.param.channels != sc->sc_vchan_params.channels)) error = EINVAL; if (error == 0) { aprint_normal_dev(sc->sc_dev, "Virtual format configured - " "Format SLINEAR, precision %d, " "channels %d, frequency %d\n", sc->sc_vchan_params.precision, sc->sc_vchan_params.channels, sc->sc_vchan_params.sample_rate); goto found; } } } } found: if (error == 0) { audio_calc_latency(sc); aprint_normal_dev(sc->sc_dev, "Latency: %d milliseconds\n", sc->sc_latency); } else { aprint_error_dev(sc->sc_dev, "Virtual format auto config failed!\n"); aprint_error_dev(sc->sc_dev, "Please check hardware capabilities\n"); } mutex_exit(sc->sc_lock); return error; } static void grow_mixer_states(struct audio_softc *sc, int count) { mixer_ctrl_t *tmp_mixer_state; size_t origlen = sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1); size_t newlen = sizeof(mixer_ctrl_t) * count + origlen; tmp_mixer_state = kmem_zalloc(newlen, KM_SLEEP); memcpy(tmp_mixer_state, sc->sc_mixer_state, origlen); sc->sc_nmixer_states += count; kmem_free(sc->sc_mixer_state, origlen); sc->sc_mixer_state = tmp_mixer_state; } static void shrink_mixer_states(struct audio_softc *sc, int count) { mixer_ctrl_t *tmp_mixer_state; size_t origlen = sizeof(mixer_ctrl_t) * (sc->sc_nmixer_states + 1); size_t newlen = origlen - sizeof(mixer_ctrl_t) * count; tmp_mixer_state = kmem_zalloc(newlen, KM_SLEEP); memcpy(tmp_mixer_state, sc->sc_mixer_state, newlen); sc->sc_nmixer_states -= count; kmem_free(sc->sc_mixer_state, origlen); sc->sc_mixer_state = tmp_mixer_state; } #endif /* NAUDIO > 0 */ #ifdef _MODULE devmajor_t audio_bmajor = -1, audio_cmajor = -1; #include "ioconf.c" #endif MODULE(MODULE_CLASS_DRIVER, audio, NULL); static int audio_modcmd(modcmd_t cmd, void *arg) { int error = 0; #ifdef _MODULE switch (cmd) { case MODULE_CMD_INIT: error = devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, &audio_cdevsw, &audio_cmajor); if (error) break; error = config_init_component(cfdriver_ioconf_audio, cfattach_ioconf_audio, cfdata_ioconf_audio); if (error) { devsw_detach(NULL, &audio_cdevsw); } break; case MODULE_CMD_FINI: devsw_detach(NULL, &audio_cdevsw); error = config_fini_component(cfdriver_ioconf_audio, cfattach_ioconf_audio, cfdata_ioconf_audio); if (error) devsw_attach(audio_cd.cd_name, NULL, &audio_bmajor, &audio_cdevsw, &audio_cmajor); break; default: error = ENOTTY; break; } #endif return error; }