3773 lines
92 KiB
C
3773 lines
92 KiB
C
/* $NetBSD: audio.c,v 1.211 2006/10/12 01:30:50 christos Exp $ */
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/*
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* Copyright (c) 1991-1993 Regents of the University of California.
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. All advertising materials mentioning features or use of this software
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* must display the following acknowledgement:
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* This product includes software developed by the Computer Systems
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* Engineering Group at Lawrence Berkeley Laboratory.
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* 4. Neither the name of the University nor of the Laboratory may be used
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* to endorse or promote products derived from this software without
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* specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*/
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/*
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* This is a (partially) SunOS-compatible /dev/audio driver for NetBSD.
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*
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* This code tries to do something half-way sensible with
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* half-duplex hardware, such as with the SoundBlaster hardware. With
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* half-duplex hardware allowing O_RDWR access doesn't really make
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* sense. However, closing and opening the device to "turn around the
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* line" is relatively expensive and costs a card reset (which can
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* take some time, at least for the SoundBlaster hardware). Instead
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* we allow O_RDWR access, and provide an ioctl to set the "mode",
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* i.e. playing or recording.
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*
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* If you write to a half-duplex device in record mode, the data is
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* tossed. If you read from the device in play mode, you get silence
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* filled buffers at the rate at which samples are naturally
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* generated.
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*
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* If you try to set both play and record mode on a half-duplex
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* device, playing takes precedence.
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*/
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/*
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* Todo:
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* - Add softaudio() isr processing for wakeup, poll, signals,
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* and silence fill.
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*/
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#include <sys/cdefs.h>
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__KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.211 2006/10/12 01:30:50 christos Exp $");
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#include "audio.h"
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#if NAUDIO > 0
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#include <sys/param.h>
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#include <sys/ioctl.h>
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#include <sys/fcntl.h>
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#include <sys/vnode.h>
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#include <sys/select.h>
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#include <sys/poll.h>
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#include <sys/malloc.h>
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#include <sys/proc.h>
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#include <sys/systm.h>
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#include <sys/syslog.h>
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#include <sys/kernel.h>
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#include <sys/signalvar.h>
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#include <sys/conf.h>
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#include <sys/audioio.h>
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#include <sys/device.h>
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#include <dev/audio_if.h>
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#include <dev/audiovar.h>
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#include <machine/endian.h>
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/* #define AUDIO_DEBUG 1 */
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#ifdef AUDIO_DEBUG
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#define DPRINTF(x) if (audiodebug) printf x
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#define DPRINTFN(n,x) if (audiodebug>(n)) printf x
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int audiodebug = AUDIO_DEBUG;
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#else
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#define DPRINTF(x)
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#define DPRINTFN(n,x)
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#endif
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#define ROUNDSIZE(x) x &= -16 /* round to nice boundary */
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#define SPECIFIED(x) (x != ~0)
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#define SPECIFIED_CH(x) (x != (u_char)~0)
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int audio_blk_ms = AUDIO_BLK_MS;
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int audiosetinfo(struct audio_softc *, struct audio_info *);
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int audiogetinfo(struct audio_softc *, struct audio_info *);
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int audio_open(dev_t, struct audio_softc *, int, int, struct lwp *);
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int audio_close(struct audio_softc *, int, int, struct lwp *);
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int audio_read(struct audio_softc *, struct uio *, int);
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int audio_write(struct audio_softc *, struct uio *, int);
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int audio_ioctl(struct audio_softc *, u_long, caddr_t, int, struct lwp *);
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int audio_poll(struct audio_softc *, int, struct lwp *);
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int audio_kqfilter(struct audio_softc *, struct knote *);
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paddr_t audio_mmap(struct audio_softc *, off_t, int);
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int mixer_open(dev_t, struct audio_softc *, int, int, struct lwp *);
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int mixer_close(struct audio_softc *, int, int, struct lwp *);
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int mixer_ioctl(struct audio_softc *, u_long, caddr_t, int, struct lwp *);
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static void mixer_remove(struct audio_softc *, struct lwp *);
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static void mixer_signal(struct audio_softc *);
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void audio_init_record(struct audio_softc *);
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void audio_init_play(struct audio_softc *);
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int audiostartr(struct audio_softc *);
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int audiostartp(struct audio_softc *);
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void audio_rint(void *);
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void audio_pint(void *);
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int audio_check_params(struct audio_params *);
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void audio_calc_blksize(struct audio_softc *, int);
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void audio_fill_silence(struct audio_params *, uint8_t *, int);
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int audio_silence_copyout(struct audio_softc *, int, struct uio *);
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void audio_init_ringbuffer(struct audio_softc *,
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struct audio_ringbuffer *, int);
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int audio_initbufs(struct audio_softc *);
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void audio_calcwater(struct audio_softc *);
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static inline int audio_sleep_timo(int *, const char *, int);
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static inline int audio_sleep(int *, const char *);
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static inline void audio_wakeup(int *);
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int audio_drain(struct audio_softc *);
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void audio_clear(struct audio_softc *);
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static inline void audio_pint_silence
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(struct audio_softc *, struct audio_ringbuffer *, uint8_t *, int);
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int audio_alloc_ring
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(struct audio_softc *, struct audio_ringbuffer *, int, size_t);
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void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *);
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static int audio_setup_pfilters(struct audio_softc *, const audio_params_t *,
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stream_filter_list_t *);
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static int audio_setup_rfilters(struct audio_softc *, const audio_params_t *,
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stream_filter_list_t *);
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static void audio_destruct_pfilters(struct audio_softc *);
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static void audio_destruct_rfilters(struct audio_softc *);
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static void audio_stream_dtor(audio_stream_t *);
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static int audio_stream_ctor(audio_stream_t *, const audio_params_t *, int);
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static void stream_filter_list_append
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(stream_filter_list_t *, stream_filter_factory_t,
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const audio_params_t *);
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static void stream_filter_list_prepend
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(stream_filter_list_t *, stream_filter_factory_t,
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const audio_params_t *);
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static void stream_filter_list_set
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(stream_filter_list_t *, int, stream_filter_factory_t,
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const audio_params_t *);
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int audio_set_defaults(struct audio_softc *, u_int);
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int audioprobe(struct device *, struct cfdata *, void *);
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void audioattach(struct device *, struct device *, void *);
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int audiodetach(struct device *, int);
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int audioactivate(struct device *, enum devact);
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void audio_powerhook(int, void *);
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struct portname {
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const char *name;
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int mask;
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};
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static const struct portname itable[] = {
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{ AudioNmicrophone, AUDIO_MICROPHONE },
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{ AudioNline, AUDIO_LINE_IN },
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{ AudioNcd, AUDIO_CD },
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{ 0, 0 }
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};
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static const struct portname otable[] = {
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{ AudioNspeaker, AUDIO_SPEAKER },
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{ AudioNheadphone, AUDIO_HEADPHONE },
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{ AudioNline, AUDIO_LINE_OUT },
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{ 0, 0 }
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};
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void au_setup_ports(struct audio_softc *, struct au_mixer_ports *,
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mixer_devinfo_t *, const struct portname *);
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int au_set_gain(struct audio_softc *, struct au_mixer_ports *,
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int, int);
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void au_get_gain(struct audio_softc *, struct au_mixer_ports *,
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u_int *, u_char *);
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int au_set_port(struct audio_softc *, struct au_mixer_ports *,
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u_int);
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int au_get_port(struct audio_softc *, struct au_mixer_ports *);
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int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *,
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int *, int *);
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int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *,
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int, int);
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int au_portof(struct audio_softc *, char *, int);
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typedef struct uio_fetcher {
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stream_fetcher_t base;
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struct uio *uio;
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int usedhigh;
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int last_used;
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} uio_fetcher_t;
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static void uio_fetcher_ctor(uio_fetcher_t *, struct uio *, int);
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static int uio_fetcher_fetch_to(stream_fetcher_t *,
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audio_stream_t *, int);
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static int null_fetcher_fetch_to(stream_fetcher_t *,
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audio_stream_t *, int);
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dev_type_open(audioopen);
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dev_type_close(audioclose);
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dev_type_read(audioread);
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dev_type_write(audiowrite);
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dev_type_ioctl(audioioctl);
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dev_type_poll(audiopoll);
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dev_type_mmap(audiommap);
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dev_type_kqfilter(audiokqfilter);
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const struct cdevsw audio_cdevsw = {
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audioopen, audioclose, audioread, audiowrite, audioioctl,
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nostop, notty, audiopoll, audiommap, audiokqfilter, D_OTHER
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};
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/* The default audio mode: 8 kHz mono mu-law */
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const struct audio_params audio_default = {
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.sample_rate = 8000,
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.encoding = AUDIO_ENCODING_ULAW,
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.precision = 8,
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.validbits = 8,
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.channels = 1,
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};
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CFATTACH_DECL(audio, sizeof(struct audio_softc),
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audioprobe, audioattach, audiodetach, audioactivate);
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extern struct cfdriver audio_cd;
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int
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audioprobe(struct device *parent __unused, struct cfdata *match __unused,
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void *aux)
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{
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struct audio_attach_args *sa;
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sa = aux;
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DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n",
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sa->type, sa, sa->hwif));
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return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0;
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}
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void
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audioattach(struct device *parent, struct device *self, void *aux)
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{
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struct audio_softc *sc;
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struct audio_attach_args *sa;
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const struct audio_hw_if *hwp;
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void *hdlp;
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int error;
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mixer_devinfo_t mi;
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int iclass, mclass, oclass, rclass, props;
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int record_master_found, record_source_found;
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sc = (void *)self;
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sa = aux;
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hwp = sa->hwif;
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hdlp = sa->hdl;
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#ifdef DIAGNOSTIC
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if (hwp == 0 ||
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hwp->query_encoding == 0 ||
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hwp->set_params == 0 ||
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(hwp->start_output == 0 && hwp->trigger_output == 0) ||
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(hwp->start_input == 0 && hwp->trigger_input == 0) ||
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hwp->halt_output == 0 ||
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hwp->halt_input == 0 ||
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hwp->getdev == 0 ||
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hwp->set_port == 0 ||
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hwp->get_port == 0 ||
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hwp->query_devinfo == 0 ||
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hwp->get_props == 0) {
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printf(": missing method\n");
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sc->hw_if = 0;
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return;
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}
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#endif
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props = hwp->get_props(hdlp);
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if (props & AUDIO_PROP_FULLDUPLEX)
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printf(": full duplex");
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else
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printf(": half duplex");
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if (props & AUDIO_PROP_MMAP)
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printf(", mmap");
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if (props & AUDIO_PROP_INDEPENDENT)
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printf(", independent");
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printf("\n");
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sc->hw_if = hwp;
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sc->hw_hdl = hdlp;
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sc->sc_dev = parent;
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sc->sc_opencnt = 0;
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sc->sc_writing = sc->sc_waitcomp = 0;
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error = audio_alloc_ring(sc, &sc->sc_pr, AUMODE_PLAY, AU_RING_SIZE);
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if (error) {
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sc->hw_if = NULL;
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printf("audio: could not allocate play buffer\n");
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return;
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}
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error = audio_alloc_ring(sc, &sc->sc_rr, AUMODE_RECORD, AU_RING_SIZE);
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if (error) {
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audio_free_ring(sc, &sc->sc_pr);
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sc->hw_if = NULL;
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printf("audio: could not allocate record buffer\n");
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return;
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}
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if ((error = audio_set_defaults(sc, 0))) {
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printf("audioattach: audio_set_defaults() failed\n");
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sc->hw_if = NULL;
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return;
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}
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iclass = mclass = oclass = rclass = -1;
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sc->sc_inports.index = -1;
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sc->sc_inports.master = -1;
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sc->sc_inports.nports = 0;
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sc->sc_inports.isenum = FALSE;
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sc->sc_inports.allports = 0;
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sc->sc_inports.isdual = FALSE;
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sc->sc_inports.mixerout = -1;
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sc->sc_inports.cur_port = -1;
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sc->sc_outports.index = -1;
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sc->sc_outports.master = -1;
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sc->sc_outports.nports = 0;
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sc->sc_outports.isenum = FALSE;
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sc->sc_outports.allports = 0;
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sc->sc_outports.isdual = FALSE;
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sc->sc_outports.mixerout = -1;
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sc->sc_outports.cur_port = -1;
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sc->sc_monitor_port = -1;
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/*
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* Read through the underlying driver's list, picking out the class
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* names from the mixer descriptions. We'll need them to decode the
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* mixer descriptions on the next pass through the loop.
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*/
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for(mi.index = 0; ; mi.index++) {
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if (hwp->query_devinfo(hdlp, &mi) != 0)
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break;
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/*
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* The type of AUDIO_MIXER_CLASS merely introduces a class.
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* All the other types describe an actual mixer.
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*/
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if (mi.type == AUDIO_MIXER_CLASS) {
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if (strcmp(mi.label.name, AudioCinputs) == 0)
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iclass = mi.mixer_class;
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if (strcmp(mi.label.name, AudioCmonitor) == 0)
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mclass = mi.mixer_class;
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if (strcmp(mi.label.name, AudioCoutputs) == 0)
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oclass = mi.mixer_class;
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if (strcmp(mi.label.name, AudioCrecord) == 0)
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rclass = mi.mixer_class;
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}
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}
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/*
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* This is where we assign each control in the "audio" model, to the
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* underlying "mixer" control. We walk through the whole list once,
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* assigning likely candidates as we come across them.
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*/
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record_master_found = 0;
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record_source_found = 0;
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for(mi.index = 0; ; mi.index++) {
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if (hwp->query_devinfo(hdlp, &mi) != 0)
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break;
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if (mi.type == AUDIO_MIXER_CLASS)
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continue;
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if (mi.mixer_class == iclass) {
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/*
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* AudioCinputs is only a fallback, when we don't
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* find what we're looking for in AudioCrecord, so
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* check the flags before accepting one of these.
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*/
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if (strcmp(mi.label.name, AudioNmaster) == 0
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&& record_master_found == 0)
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sc->sc_inports.master = mi.index;
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if (strcmp(mi.label.name, AudioNsource) == 0
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&& record_source_found == 0) {
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if (mi.type == AUDIO_MIXER_ENUM) {
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int i;
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for(i = 0; i < mi.un.e.num_mem; i++)
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if (strcmp(mi.un.e.member[i].label.name,
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AudioNmixerout) == 0)
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sc->sc_inports.mixerout =
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mi.un.e.member[i].ord;
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}
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au_setup_ports(sc, &sc->sc_inports, &mi,
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itable);
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}
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} else if (mi.mixer_class == mclass) {
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if (strcmp(mi.label.name, AudioNmonitor) == 0)
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sc->sc_monitor_port = mi.index;
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} else if (mi.mixer_class == oclass) {
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if (strcmp(mi.label.name, AudioNmaster) == 0)
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sc->sc_outports.master = mi.index;
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if (strcmp(mi.label.name, AudioNselect) == 0)
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au_setup_ports(sc, &sc->sc_outports, &mi,
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otable);
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} else if (mi.mixer_class == rclass) {
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/*
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* These are the preferred mixers for the audio record
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* controls, so set the flags here, but don't check.
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*/
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if (strcmp(mi.label.name, AudioNmaster) == 0) {
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sc->sc_inports.master = mi.index;
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record_master_found = 1;
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}
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#if 1 /* Deprecated. Use AudioNmaster. */
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if (strcmp(mi.label.name, AudioNrecord) == 0) {
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sc->sc_inports.master = mi.index;
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record_master_found = 1;
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}
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if (strcmp(mi.label.name, AudioNvolume) == 0) {
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sc->sc_inports.master = mi.index;
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record_master_found = 1;
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}
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#endif
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if (strcmp(mi.label.name, AudioNsource) == 0) {
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if (mi.type == AUDIO_MIXER_ENUM) {
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int i;
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for(i = 0; i < mi.un.e.num_mem; i++)
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if (strcmp(mi.un.e.member[i].label.name,
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AudioNmixerout) == 0)
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sc->sc_inports.mixerout =
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mi.un.e.member[i].ord;
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}
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au_setup_ports(sc, &sc->sc_inports, &mi,
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itable);
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record_source_found = 1;
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}
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}
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}
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DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, "
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"output ports=0x%x, output master=%d\n",
|
|
sc->sc_inports.allports, sc->sc_inports.master,
|
|
sc->sc_outports.allports, sc->sc_outports.master));
|
|
|
|
sc->sc_powerhook = powerhook_establish(sc->dev.dv_xname,
|
|
audio_powerhook, sc);
|
|
if (sc->sc_powerhook == NULL)
|
|
aprint_error("%s: can't establish powerhook\n",
|
|
sc->dev.dv_xname);
|
|
}
|
|
|
|
int
|
|
audioactivate(struct device *self, enum devact act)
|
|
{
|
|
struct audio_softc *sc;
|
|
|
|
sc = (struct audio_softc *)self;
|
|
switch (act) {
|
|
case DVACT_ACTIVATE:
|
|
return EOPNOTSUPP;
|
|
|
|
case DVACT_DEACTIVATE:
|
|
sc->sc_dying = TRUE;
|
|
break;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audiodetach(struct device *self, int flags __unused)
|
|
{
|
|
struct audio_softc *sc;
|
|
int maj, mn;
|
|
int s;
|
|
|
|
sc = (struct audio_softc *)self;
|
|
DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags));
|
|
|
|
sc->sc_dying = TRUE;
|
|
|
|
wakeup(&sc->sc_wchan);
|
|
wakeup(&sc->sc_rchan);
|
|
s = splaudio();
|
|
if (--sc->sc_refcnt >= 0) {
|
|
if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120))
|
|
printf("audiodetach: %s didn't detach\n",
|
|
sc->dev.dv_xname);
|
|
}
|
|
splx(s);
|
|
|
|
/* free resources */
|
|
audio_free_ring(sc, &sc->sc_pr);
|
|
audio_free_ring(sc, &sc->sc_rr);
|
|
audio_destruct_pfilters(sc);
|
|
audio_destruct_rfilters(sc);
|
|
|
|
/* locate the major number */
|
|
maj = cdevsw_lookup_major(&audio_cdevsw);
|
|
|
|
/* Nuke the vnodes for any open instances (calls close). */
|
|
mn = device_unit(self);
|
|
vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR);
|
|
vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR);
|
|
vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR);
|
|
vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR);
|
|
|
|
if (sc->sc_powerhook != NULL) {
|
|
powerhook_disestablish(sc->sc_powerhook);
|
|
sc->sc_powerhook = NULL;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
au_portof(struct audio_softc *sc, char *name, int class)
|
|
{
|
|
mixer_devinfo_t mi;
|
|
|
|
for(mi.index = 0;
|
|
sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0;
|
|
mi.index++)
|
|
if (mi.mixer_class == class && strcmp(mi.label.name, name) == 0)
|
|
return mi.index;
|
|
return -1;
|
|
}
|
|
|
|
void
|
|
au_setup_ports(struct audio_softc *sc, struct au_mixer_ports *ports,
|
|
mixer_devinfo_t *mi, const struct portname *tbl)
|
|
{
|
|
int i, j;
|
|
|
|
ports->index = mi->index;
|
|
if (mi->type == AUDIO_MIXER_ENUM) {
|
|
ports->isenum = TRUE;
|
|
for(i = 0; tbl[i].name; i++)
|
|
for(j = 0; j < mi->un.e.num_mem; j++)
|
|
if (strcmp(mi->un.e.member[j].label.name,
|
|
tbl[i].name) == 0) {
|
|
ports->allports |= tbl[i].mask;
|
|
ports->aumask[ports->nports] = tbl[i].mask;
|
|
ports->misel[ports->nports] =
|
|
mi->un.e.member[j].ord;
|
|
ports->miport[ports->nports] =
|
|
au_portof(sc, mi->un.e.member[j].label.name,
|
|
mi->mixer_class);
|
|
if (ports->mixerout != -1 &&
|
|
ports->miport[ports->nports++] != -1)
|
|
ports->isdual = TRUE;
|
|
}
|
|
} 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);
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from hardware driver. This is where the MI audio driver gets
|
|
* probed/attached to the hardware driver.
|
|
*/
|
|
struct device *
|
|
audio_attach_mi(const struct audio_hw_if *ahwp, void *hdlp, struct device *dev)
|
|
{
|
|
struct audio_attach_args arg;
|
|
|
|
#ifdef DIAGNOSTIC
|
|
if (ahwp == NULL) {
|
|
aprint_error("audio_attach_mi: NULL\n");
|
|
return 0;
|
|
}
|
|
#endif
|
|
arg.type = AUDIODEV_TYPE_AUDIO;
|
|
arg.hwif = ahwp;
|
|
arg.hdl = hdlp;
|
|
return config_found(dev, &arg, audioprint);
|
|
}
|
|
|
|
#ifdef AUDIO_DEBUG
|
|
void audio_printsc(struct audio_softc *);
|
|
void audio_print_params(const char *, struct audio_params *);
|
|
|
|
void
|
|
audio_printsc(struct audio_softc *sc)
|
|
{
|
|
printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
|
|
printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode);
|
|
printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan);
|
|
printf("rring used 0x%x pring used=%d\n",
|
|
audio_stream_get_used(&sc->sc_rr.s),
|
|
audio_stream_get_used(&sc->sc_pr.s));
|
|
printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus);
|
|
printf("blksize %d", sc->sc_pr.blksize);
|
|
printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow);
|
|
}
|
|
|
|
void
|
|
audio_print_params(const char *s, struct audio_params *p)
|
|
{
|
|
printf("%s enc=%u %uch %u/%ubit %uHz\n", s, p->encoding, p->channels,
|
|
p->validbits, p->precision, p->sample_rate);
|
|
}
|
|
#endif
|
|
|
|
int
|
|
audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r,
|
|
int direction, size_t bufsize)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
void *hdl;
|
|
|
|
hw = sc->hw_if;
|
|
hdl = sc->hw_hdl;
|
|
/*
|
|
* Alloc DMA play and record buffers
|
|
*/
|
|
if (bufsize < AUMINBUF)
|
|
bufsize = AUMINBUF;
|
|
ROUNDSIZE(bufsize);
|
|
if (hw->round_buffersize)
|
|
bufsize = hw->round_buffersize(hdl, direction, bufsize);
|
|
if (hw->allocm)
|
|
r->s.start = hw->allocm(hdl, direction, bufsize,
|
|
M_DEVBUF, M_WAITOK);
|
|
else
|
|
r->s.start = malloc(bufsize, M_DEVBUF, M_WAITOK);
|
|
if (r->s.start == 0)
|
|
return ENOMEM;
|
|
r->s.bufsize = bufsize;
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r)
|
|
{
|
|
|
|
if (sc->hw_if->freem)
|
|
sc->hw_if->freem(sc->hw_hdl, r->s.start, M_DEVBUF);
|
|
else
|
|
free(r->s.start, M_DEVBUF);
|
|
r->s.start = 0;
|
|
}
|
|
|
|
static int
|
|
audio_setup_pfilters(struct audio_softc *sc, const audio_params_t *pp,
|
|
stream_filter_list_t *pfilters)
|
|
{
|
|
stream_filter_t *pf[AUDIO_MAX_FILTERS];
|
|
audio_stream_t ps[AUDIO_MAX_FILTERS];
|
|
const audio_params_t *from_param;
|
|
audio_params_t *to_param;
|
|
int i, n;
|
|
|
|
while (sc->sc_writing) {
|
|
sc->sc_waitcomp = 1;
|
|
(void)tsleep(sc, 0, "audioch", 10*hz);
|
|
}
|
|
|
|
memset(pf, 0, sizeof(pf));
|
|
memset(ps, 0, sizeof(ps));
|
|
from_param = pp;
|
|
for (i = 0; i < pfilters->req_size; i++) {
|
|
n = pfilters->req_size - i - 1;
|
|
to_param = &pfilters->filters[n].param;
|
|
audio_check_params(to_param);
|
|
pf[i] = pfilters->filters[n].factory(sc, from_param, to_param);
|
|
if (pf[i] == NULL)
|
|
break;
|
|
if (audio_stream_ctor(&ps[i], from_param, AU_RING_SIZE))
|
|
break;
|
|
if (i > 0)
|
|
pf[i]->set_fetcher(pf[i], &pf[i - 1]->base);
|
|
from_param = to_param;
|
|
}
|
|
if (i < pfilters->req_size) { /* failure */
|
|
DPRINTF(("%s: pfilters failure\n", __func__));
|
|
for (; i >= 0; i--) {
|
|
if (pf[i] != NULL)
|
|
pf[i]->dtor(pf[i]);
|
|
audio_stream_dtor(&ps[i]);
|
|
}
|
|
sc->sc_waitcomp = 0;
|
|
return EINVAL;
|
|
}
|
|
|
|
audio_destruct_pfilters(sc);
|
|
memcpy(sc->sc_pfilters, pf, sizeof(pf));
|
|
memcpy(sc->sc_pstreams, ps, sizeof(ps));
|
|
sc->sc_npfilters = pfilters->req_size;
|
|
for (i = 0; i < pfilters->req_size; i++) {
|
|
pf[i]->set_inputbuffer(pf[i], &sc->sc_pstreams[i]);
|
|
}
|
|
|
|
/* hardware format and the buffer near to userland */
|
|
if (pfilters->req_size <= 0) {
|
|
sc->sc_pr.s.param = *pp;
|
|
sc->sc_pustream = &sc->sc_pr.s;
|
|
} else {
|
|
sc->sc_pr.s.param = pfilters->filters[0].param;
|
|
sc->sc_pustream = &sc->sc_pstreams[0];
|
|
}
|
|
#ifdef AUDIO_DEBUG
|
|
printf("%s: HW-buffer=%p pustream=%p\n",
|
|
__func__, &sc->sc_pr.s, sc->sc_pustream);
|
|
for (i = 0; i < pfilters->req_size; i++) {
|
|
char num[100];
|
|
snprintf(num, 100, "[%d]", i);
|
|
audio_print_params(num, &sc->sc_pstreams[i].param);
|
|
}
|
|
audio_print_params("[HW]", &sc->sc_pr.s.param);
|
|
#endif /* AUDIO_DEBUG */
|
|
|
|
sc->sc_waitcomp = 0;
|
|
return 0;
|
|
}
|
|
|
|
static int
|
|
audio_setup_rfilters(struct audio_softc *sc, const audio_params_t *rp,
|
|
stream_filter_list_t *rfilters)
|
|
{
|
|
stream_filter_t *rf[AUDIO_MAX_FILTERS];
|
|
audio_stream_t rs[AUDIO_MAX_FILTERS];
|
|
const audio_params_t *to_param;
|
|
audio_params_t *from_param;
|
|
int i;
|
|
|
|
memset(rf, 0, sizeof(rf));
|
|
memset(rs, 0, sizeof(rs));
|
|
for (i = 0; i < rfilters->req_size; i++) {
|
|
from_param = &rfilters->filters[i].param;
|
|
audio_check_params(from_param);
|
|
to_param = i + 1 < rfilters->req_size
|
|
? &rfilters->filters[i + 1].param : rp;
|
|
rf[i] = rfilters->filters[i].factory(sc, from_param, to_param);
|
|
if (rf[i] == NULL)
|
|
break;
|
|
if (audio_stream_ctor(&rs[i], to_param, AU_RING_SIZE))
|
|
break;
|
|
if (i > 0) {
|
|
rf[i]->set_fetcher(rf[i], &rf[i - 1]->base);
|
|
} else {
|
|
/* rf[0] has no previous fetcher because
|
|
* the audio hardware fills data to the
|
|
* input buffer. */
|
|
rf[0]->set_inputbuffer(rf[0], &sc->sc_rr.s);
|
|
}
|
|
}
|
|
if (i < rfilters->req_size) { /* failure */
|
|
DPRINTF(("%s: rfilters failure\n", __func__));
|
|
for (; i >= 0; i--) {
|
|
if (rf[i] != NULL)
|
|
rf[i]->dtor(rf[i]);
|
|
audio_stream_dtor(&rs[i]);
|
|
}
|
|
return EINVAL;
|
|
}
|
|
|
|
audio_destruct_rfilters(sc);
|
|
memcpy(sc->sc_rfilters, rf, sizeof(rf));
|
|
memcpy(sc->sc_rstreams, rs, sizeof(rs));
|
|
sc->sc_nrfilters = rfilters->req_size;
|
|
for (i = 1; i < rfilters->req_size; i++) {
|
|
rf[i]->set_inputbuffer(rf[i], &sc->sc_rstreams[i - 1]);
|
|
}
|
|
|
|
/* hardware format and the buffer near to userland */
|
|
if (rfilters->req_size <= 0) {
|
|
sc->sc_rr.s.param = *rp;
|
|
sc->sc_rustream = &sc->sc_rr.s;
|
|
} else {
|
|
sc->sc_rr.s.param = rfilters->filters[0].param;
|
|
sc->sc_rustream = &sc->sc_rstreams[rfilters->req_size - 1];
|
|
}
|
|
#ifdef AUDIO_DEBUG
|
|
printf("%s: HW-buffer=%p pustream=%p\n",
|
|
__func__, &sc->sc_rr.s, sc->sc_rustream);
|
|
audio_print_params("[HW]", &sc->sc_rr.s.param);
|
|
for (i = 0; i < rfilters->req_size; i++) {
|
|
char num[100];
|
|
snprintf(num, 100, "[%d]", i);
|
|
audio_print_params(num, &sc->sc_rstreams[i].param);
|
|
}
|
|
#endif /* AUDIO_DEBUG */
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
audio_destruct_pfilters(struct audio_softc *sc)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < sc->sc_npfilters; i++) {
|
|
sc->sc_pfilters[i]->dtor(sc->sc_pfilters[i]);
|
|
sc->sc_pfilters[i] = NULL;
|
|
audio_stream_dtor(&sc->sc_pstreams[i]);
|
|
}
|
|
sc->sc_npfilters = 0;
|
|
}
|
|
|
|
static void
|
|
audio_destruct_rfilters(struct audio_softc *sc)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < sc->sc_nrfilters; i++) {
|
|
sc->sc_rfilters[i]->dtor(sc->sc_rfilters[i]);
|
|
sc->sc_rfilters[i] = NULL;
|
|
audio_stream_dtor(&sc->sc_rstreams[i]);
|
|
}
|
|
sc->sc_nrfilters = 0;
|
|
}
|
|
|
|
static void
|
|
audio_stream_dtor(audio_stream_t *stream)
|
|
{
|
|
|
|
if (stream->start != NULL)
|
|
free(stream->start, M_DEVBUF);
|
|
memset(stream, 0, sizeof(audio_stream_t));
|
|
}
|
|
|
|
static int
|
|
audio_stream_ctor(audio_stream_t *stream, const audio_params_t *param, int size)
|
|
{
|
|
int frame_size;
|
|
|
|
size = min(size, AU_RING_SIZE);
|
|
stream->bufsize = size;
|
|
stream->start = malloc(size, M_DEVBUF, M_WAITOK);
|
|
if (stream->start == NULL)
|
|
return ENOMEM;
|
|
frame_size = (param->precision + 7) / 8 * param->channels;
|
|
size = (size / frame_size) * frame_size;
|
|
stream->end = stream->start + size;
|
|
stream->inp = stream->start;
|
|
stream->outp = stream->start;
|
|
stream->used = 0;
|
|
stream->param = *param;
|
|
stream->loop = FALSE;
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
stream_filter_list_append(stream_filter_list_t *list,
|
|
stream_filter_factory_t factory,
|
|
const audio_params_t *param)
|
|
{
|
|
|
|
if (list->req_size >= AUDIO_MAX_FILTERS) {
|
|
printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n",
|
|
__func__);
|
|
return;
|
|
}
|
|
list->filters[list->req_size].factory = factory;
|
|
list->filters[list->req_size].param = *param;
|
|
list->req_size++;
|
|
}
|
|
|
|
static void
|
|
stream_filter_list_set(stream_filter_list_t *list, int i,
|
|
stream_filter_factory_t factory,
|
|
const audio_params_t *param)
|
|
{
|
|
|
|
if (i < 0 || i >= AUDIO_MAX_FILTERS) {
|
|
printf("%s: invalid index: %d\n", __func__, i);
|
|
return;
|
|
}
|
|
|
|
list->filters[i].factory = factory;
|
|
list->filters[i].param = *param;
|
|
if (list->req_size <= i)
|
|
list->req_size = i + 1;
|
|
}
|
|
|
|
static void
|
|
stream_filter_list_prepend(stream_filter_list_t *list,
|
|
stream_filter_factory_t factory,
|
|
const audio_params_t *param)
|
|
{
|
|
|
|
if (list->req_size >= AUDIO_MAX_FILTERS) {
|
|
printf("%s: increase AUDIO_MAX_FILTERS in sys/dev/audio_if.h\n",
|
|
__func__);
|
|
return;
|
|
}
|
|
memmove(&list->filters[1], &list->filters[0],
|
|
sizeof(struct stream_filter_req) * list->req_size);
|
|
list->filters[0].factory = factory;
|
|
list->filters[0].param = *param;
|
|
list->req_size++;
|
|
}
|
|
|
|
int
|
|
audioopen(dev_t dev, int flags, int ifmt, struct lwp *l)
|
|
{
|
|
struct audio_softc *sc;
|
|
int error;
|
|
|
|
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
|
|
if (sc == NULL)
|
|
return ENXIO;
|
|
|
|
if (sc->sc_dying)
|
|
return EIO;
|
|
|
|
if (sc->hw_if->powerstate && sc->sc_opencnt <= 0)
|
|
sc->hw_if->powerstate(sc->hw_hdl, AUDIOPOWER_ON);
|
|
sc->sc_opencnt++;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
error = audio_open(dev, sc, flags, ifmt, l);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
error = 0;
|
|
break;
|
|
case MIXER_DEVICE:
|
|
error = mixer_open(dev, sc, flags, ifmt, l);
|
|
break;
|
|
default:
|
|
error = ENXIO;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audioclose(dev_t dev, int flags, int ifmt, struct lwp *l)
|
|
{
|
|
struct audio_softc *sc;
|
|
int unit;
|
|
int error;
|
|
|
|
unit = AUDIOUNIT(dev);
|
|
sc = audio_cd.cd_devs[unit];
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
error = audio_close(sc, flags, ifmt, l);
|
|
break;
|
|
case MIXER_DEVICE:
|
|
error = mixer_close(sc, flags, ifmt, l);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
error = 0;
|
|
break;
|
|
default:
|
|
error = ENXIO;
|
|
break;
|
|
}
|
|
|
|
sc->sc_opencnt--;
|
|
if (sc->hw_if->powerstate && sc->sc_opencnt <= 0)
|
|
sc->hw_if->powerstate(sc->hw_hdl, AUDIOPOWER_OFF);
|
|
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audioread(dev_t dev, struct uio *uio, int ioflag)
|
|
{
|
|
struct audio_softc *sc;
|
|
int error;
|
|
|
|
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
|
|
if (sc == NULL)
|
|
return ENXIO;
|
|
|
|
if (sc->sc_dying)
|
|
return EIO;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
error = audio_read(sc, uio, ioflag);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
error = ENODEV;
|
|
break;
|
|
default:
|
|
error = ENXIO;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audiowrite(dev_t dev, struct uio *uio, int ioflag)
|
|
{
|
|
struct audio_softc *sc;
|
|
int error;
|
|
|
|
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
|
|
if (sc == NULL)
|
|
return ENXIO;
|
|
|
|
if (sc->sc_dying)
|
|
return EIO;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
error = audio_write(sc, uio, ioflag);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
error = ENODEV;
|
|
break;
|
|
default:
|
|
error = ENXIO;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audioioctl(dev_t dev, u_long cmd, caddr_t addr, int flag, struct lwp *l)
|
|
{
|
|
struct audio_softc *sc;
|
|
int error;
|
|
|
|
sc = audio_cd.cd_devs[AUDIOUNIT(dev)];
|
|
if (sc->sc_dying)
|
|
return EIO;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
case AUDIOCTL_DEVICE:
|
|
error = audio_ioctl(sc, cmd, addr, flag, l);
|
|
break;
|
|
case MIXER_DEVICE:
|
|
error = mixer_ioctl(sc, cmd, addr, flag, l);
|
|
break;
|
|
default:
|
|
error = ENXIO;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audiopoll(dev_t dev, int events, struct lwp *l)
|
|
{
|
|
struct audio_softc *sc;
|
|
int revents;
|
|
|
|
sc = audio_cd.cd_devs[AUDIOUNIT(dev)];
|
|
if (sc->sc_dying)
|
|
return POLLHUP;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
revents = audio_poll(sc, events, l);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
revents = 0;
|
|
break;
|
|
default:
|
|
revents = POLLERR;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return revents;
|
|
}
|
|
|
|
int
|
|
audiokqfilter(dev_t dev, struct knote *kn)
|
|
{
|
|
struct audio_softc *sc;
|
|
int rv;
|
|
|
|
sc = audio_cd.cd_devs[AUDIOUNIT(dev)];
|
|
if (sc->sc_dying)
|
|
return 1;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
rv = audio_kqfilter(sc, kn);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
rv = 1;
|
|
break;
|
|
default:
|
|
rv = 1;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return rv;
|
|
}
|
|
|
|
paddr_t
|
|
audiommap(dev_t dev, off_t off, int prot)
|
|
{
|
|
struct audio_softc *sc;
|
|
paddr_t error;
|
|
|
|
sc = audio_cd.cd_devs[AUDIOUNIT(dev)];
|
|
if (sc->sc_dying)
|
|
return -1;
|
|
|
|
sc->sc_refcnt++;
|
|
switch (AUDIODEV(dev)) {
|
|
case SOUND_DEVICE:
|
|
case AUDIO_DEVICE:
|
|
error = audio_mmap(sc, off, prot);
|
|
break;
|
|
case AUDIOCTL_DEVICE:
|
|
case MIXER_DEVICE:
|
|
error = -1;
|
|
break;
|
|
default:
|
|
error = -1;
|
|
break;
|
|
}
|
|
if (--sc->sc_refcnt < 0)
|
|
wakeup(&sc->sc_refcnt);
|
|
return error;
|
|
}
|
|
|
|
/*
|
|
* Audio driver
|
|
*/
|
|
void
|
|
audio_init_ringbuffer(struct audio_softc *sc, struct audio_ringbuffer *rp,
|
|
int mode)
|
|
{
|
|
int nblks;
|
|
int blksize;
|
|
|
|
blksize = rp->blksize;
|
|
if (blksize < AUMINBLK)
|
|
blksize = AUMINBLK;
|
|
if (blksize > rp->s.bufsize / AUMINNOBLK)
|
|
blksize = rp->s.bufsize / AUMINNOBLK;
|
|
ROUNDSIZE(blksize);
|
|
DPRINTF(("audio_init_ringbuffer: MI blksize=%d\n", blksize));
|
|
if (sc->hw_if->round_blocksize)
|
|
blksize = sc->hw_if->round_blocksize(sc->hw_hdl, blksize,
|
|
mode, &rp->s.param);
|
|
if (blksize <= 0)
|
|
panic("audio_init_ringbuffer: blksize");
|
|
nblks = rp->s.bufsize / blksize;
|
|
|
|
DPRINTF(("audio_init_ringbuffer: final blksize=%d\n", blksize));
|
|
rp->blksize = blksize;
|
|
rp->maxblks = nblks;
|
|
rp->s.end = rp->s.start + nblks * blksize;
|
|
rp->s.outp = rp->s.inp = rp->s.start;
|
|
rp->s.used = 0;
|
|
rp->stamp = 0;
|
|
rp->stamp_last = 0;
|
|
rp->fstamp = 0;
|
|
rp->drops = 0;
|
|
rp->pause = FALSE;
|
|
rp->copying = FALSE;
|
|
rp->needfill = FALSE;
|
|
rp->mmapped = FALSE;
|
|
}
|
|
|
|
int
|
|
audio_initbufs(struct audio_softc *sc)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
int error;
|
|
|
|
DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
|
|
hw = sc->hw_if;
|
|
audio_init_ringbuffer(sc, &sc->sc_rr, AUMODE_RECORD);
|
|
if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) {
|
|
error = hw->init_input(sc->hw_hdl, sc->sc_rr.s.start,
|
|
sc->sc_rr.s.end - sc->sc_rr.s.start);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
audio_init_ringbuffer(sc, &sc->sc_pr, AUMODE_PLAY);
|
|
sc->sc_sil_count = 0;
|
|
if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) {
|
|
error = hw->init_output(sc->hw_hdl, sc->sc_pr.s.start,
|
|
sc->sc_pr.s.end - sc->sc_pr.s.start);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
#define double u_long
|
|
sc->sc_pnintr = 0;
|
|
sc->sc_pblktime = (u_long)(
|
|
(double)sc->sc_pr.blksize * 100000 /
|
|
(double)(sc->sc_pparams.precision / NBBY *
|
|
sc->sc_pparams.channels *
|
|
sc->sc_pparams.sample_rate)) * 10;
|
|
DPRINTF(("audio: play blktime = %lu for %d\n",
|
|
sc->sc_pblktime, sc->sc_pr.blksize));
|
|
sc->sc_rnintr = 0;
|
|
sc->sc_rblktime = (u_long)(
|
|
(double)sc->sc_rr.blksize * 100000 /
|
|
(double)(sc->sc_rparams.precision / NBBY *
|
|
sc->sc_rparams.channels *
|
|
sc->sc_rparams.sample_rate)) * 10;
|
|
DPRINTF(("audio: record blktime = %lu for %d\n",
|
|
sc->sc_rblktime, sc->sc_rr.blksize));
|
|
#undef double
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
audio_calcwater(struct audio_softc *sc)
|
|
{
|
|
|
|
/* set high at 100% */
|
|
sc->sc_pr.usedhigh = sc->sc_pustream->end - sc->sc_pustream->start;
|
|
/* set low at 75% of usedhigh */
|
|
sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4;
|
|
if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh)
|
|
sc->sc_pr.usedlow -= sc->sc_pr.blksize;
|
|
|
|
sc->sc_rr.usedhigh = sc->sc_rustream->end - sc->sc_rustream->start
|
|
- sc->sc_rr.blksize;
|
|
sc->sc_rr.usedlow = 0;
|
|
DPRINTF(("%s: plow=%d phigh=%d rlow=%d rhigh=%d\n", __func__,
|
|
sc->sc_pr.usedlow, sc->sc_pr.usedhigh,
|
|
sc->sc_rr.usedlow, sc->sc_rr.usedhigh));
|
|
}
|
|
|
|
static inline int
|
|
audio_sleep_timo(int *chan, const char *label, int timo)
|
|
{
|
|
int st;
|
|
|
|
if (label == NULL)
|
|
label = "audio";
|
|
|
|
DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n",
|
|
chan, label, timo));
|
|
*chan = 1;
|
|
st = tsleep(chan, PWAIT | PCATCH, label, timo);
|
|
*chan = 0;
|
|
#ifdef AUDIO_DEBUG
|
|
if (st != 0 && st != EINTR)
|
|
DPRINTF(("audio_sleep: woke up st=%d\n", st));
|
|
#endif
|
|
return st;
|
|
}
|
|
|
|
static inline int
|
|
audio_sleep(int *chan, const char *label)
|
|
{
|
|
|
|
return audio_sleep_timo(chan, label, 0);
|
|
}
|
|
|
|
/* call at splaudio() */
|
|
static inline void
|
|
audio_wakeup(int *chan)
|
|
{
|
|
|
|
DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan));
|
|
if (*chan) {
|
|
wakeup(chan);
|
|
*chan = 0;
|
|
}
|
|
}
|
|
|
|
int
|
|
audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt __unused,
|
|
struct lwp *l __unused)
|
|
{
|
|
int error;
|
|
u_int mode;
|
|
const struct audio_hw_if *hw;
|
|
|
|
hw = sc->hw_if;
|
|
if (hw == NULL)
|
|
return ENXIO;
|
|
|
|
DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n",
|
|
flags, sc, sc->hw_hdl));
|
|
|
|
if (((flags & FREAD) && (sc->sc_open & AUOPEN_READ)) ||
|
|
((flags & FWRITE) && (sc->sc_open & AUOPEN_WRITE)))
|
|
return EBUSY;
|
|
|
|
if (hw->open != NULL) {
|
|
error = hw->open(sc->hw_hdl, flags);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
sc->sc_async_audio = 0;
|
|
sc->sc_rchan = 0;
|
|
sc->sc_wchan = 0;
|
|
sc->sc_sil_count = 0;
|
|
sc->sc_rbus = FALSE;
|
|
sc->sc_pbus = FALSE;
|
|
sc->sc_eof = 0;
|
|
sc->sc_playdrop = 0;
|
|
|
|
sc->sc_full_duplex =
|
|
(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
|
|
(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX);
|
|
|
|
mode = 0;
|
|
if (flags & FREAD) {
|
|
sc->sc_open |= AUOPEN_READ;
|
|
mode |= AUMODE_RECORD;
|
|
}
|
|
if (flags & FWRITE) {
|
|
sc->sc_open |= AUOPEN_WRITE;
|
|
mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
|
|
}
|
|
|
|
/*
|
|
* Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
|
|
* The /dev/audio is always (re)set to 8-bit MU-Law mono
|
|
* For the other devices, you get what they were last set to.
|
|
*/
|
|
if (ISDEVAUDIO(dev)) {
|
|
error = audio_set_defaults(sc, mode);
|
|
} else {
|
|
struct audio_info ai;
|
|
|
|
AUDIO_INITINFO(&ai);
|
|
ai.mode = mode;
|
|
error = audiosetinfo(sc, &ai);
|
|
}
|
|
if (error)
|
|
goto bad;
|
|
|
|
#ifdef DIAGNOSTIC
|
|
/*
|
|
* Sample rate and precision are supposed to be set to proper
|
|
* default values by the hardware driver, so that it may give
|
|
* us these values.
|
|
*/
|
|
if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) {
|
|
printf("audio_open: 0 precision\n");
|
|
return EINVAL;
|
|
}
|
|
#endif
|
|
|
|
/* audio_close() decreases sc_pr.usedlow, recalculate here */
|
|
audio_calcwater(sc);
|
|
|
|
DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode));
|
|
|
|
return 0;
|
|
|
|
bad:
|
|
if (hw->close != NULL)
|
|
hw->close(sc->hw_hdl);
|
|
sc->sc_open = 0;
|
|
sc->sc_mode = 0;
|
|
sc->sc_full_duplex = 0;
|
|
return error;
|
|
}
|
|
|
|
/*
|
|
* Must be called from task context.
|
|
*/
|
|
void
|
|
audio_init_record(struct audio_softc *sc)
|
|
{
|
|
int s;
|
|
|
|
s = splaudio();
|
|
if (sc->hw_if->speaker_ctl &&
|
|
(!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0))
|
|
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF);
|
|
splx(s);
|
|
}
|
|
|
|
/*
|
|
* Must be called from task context.
|
|
*/
|
|
void
|
|
audio_init_play(struct audio_softc *sc)
|
|
{
|
|
int s;
|
|
|
|
s = splaudio();
|
|
sc->sc_wstamp = sc->sc_pr.stamp;
|
|
if (sc->hw_if->speaker_ctl)
|
|
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
|
|
splx(s);
|
|
}
|
|
|
|
int
|
|
audio_drain(struct audio_softc *sc)
|
|
{
|
|
struct audio_ringbuffer *cb;
|
|
int error, drops;
|
|
int s;
|
|
int i, used;
|
|
|
|
DPRINTF(("audio_drain: enter busy=%d\n", sc->sc_pbus));
|
|
cb = &sc->sc_pr;
|
|
if (cb->mmapped)
|
|
return 0;
|
|
|
|
used = audio_stream_get_used(&sc->sc_pr.s);
|
|
s = splaudio();
|
|
for (i = 0; i < sc->sc_npfilters; i++)
|
|
used += audio_stream_get_used(&sc->sc_pstreams[i]);
|
|
splx(s);
|
|
if (used <= 0)
|
|
return 0;
|
|
|
|
if (!sc->sc_pbus) {
|
|
/* We've never started playing, probably because the
|
|
* block was too short. Pad it and start now.
|
|
*/
|
|
int cc;
|
|
uint8_t *inp = cb->s.inp;
|
|
|
|
cc = cb->blksize - (inp - cb->s.start) % cb->blksize;
|
|
audio_fill_silence(&cb->s.param, inp, cc);
|
|
s = splaudio();
|
|
cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc);
|
|
error = audiostartp(sc);
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
/*
|
|
* Play until a silence block has been played, then we
|
|
* know all has been drained.
|
|
* XXX This should be done some other way to avoid
|
|
* playing silence.
|
|
*/
|
|
#ifdef DIAGNOSTIC
|
|
if (cb->copying) {
|
|
printf("audio_drain: copying in progress!?!\n");
|
|
cb->copying = FALSE;
|
|
}
|
|
#endif
|
|
drops = cb->drops;
|
|
error = 0;
|
|
s = splaudio();
|
|
while (cb->drops == drops && !error) {
|
|
DPRINTF(("audio_drain: used=%d, drops=%ld\n",
|
|
audio_stream_get_used(&sc->sc_pr.s), cb->drops));
|
|
/*
|
|
* When the process is exiting, it ignores all signals and
|
|
* we can't interrupt this sleep, so we set a timeout
|
|
* just in case.
|
|
*/
|
|
error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz);
|
|
if (sc->sc_dying)
|
|
error = EIO;
|
|
}
|
|
splx(s);
|
|
return error;
|
|
}
|
|
|
|
/*
|
|
* Close an audio chip.
|
|
*/
|
|
/* ARGSUSED */
|
|
int
|
|
audio_close(struct audio_softc *sc, int flags, int ifmt __unused,
|
|
struct lwp *l __unused)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
int s;
|
|
|
|
DPRINTF(("audio_close: sc=%p\n", sc));
|
|
hw = sc->hw_if;
|
|
s = splaudio();
|
|
/* Stop recording. */
|
|
if ((flags & FREAD) && sc->sc_rbus) {
|
|
/*
|
|
* XXX Some drivers (e.g. SB) use the same routine
|
|
* to halt input and output so don't halt input if
|
|
* in full duplex mode. These drivers should be fixed.
|
|
*/
|
|
if (!sc->sc_full_duplex || hw->halt_input != hw->halt_output)
|
|
hw->halt_input(sc->hw_hdl);
|
|
sc->sc_rbus = FALSE;
|
|
}
|
|
/*
|
|
* Block until output drains, but allow ^C interrupt.
|
|
*/
|
|
sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */
|
|
/*
|
|
* If there is pending output, let it drain (unless
|
|
* the output is paused).
|
|
*/
|
|
if ((flags & FWRITE) && sc->sc_pbus) {
|
|
if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain)
|
|
(void)hw->drain(sc->hw_hdl);
|
|
hw->halt_output(sc->hw_hdl);
|
|
sc->sc_pbus = FALSE;
|
|
}
|
|
|
|
if (hw->close != NULL)
|
|
hw->close(sc->hw_hdl);
|
|
|
|
sc->sc_open = 0;
|
|
sc->sc_mode = 0;
|
|
sc->sc_async_audio = 0;
|
|
sc->sc_full_duplex = 0;
|
|
splx(s);
|
|
DPRINTF(("audio_close: done\n"));
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audio_read(struct audio_softc *sc, struct uio *uio, int ioflag)
|
|
{
|
|
struct audio_ringbuffer *cb;
|
|
const uint8_t *outp;
|
|
uint8_t *inp;
|
|
int error, s, used, cc, n;
|
|
|
|
cb = &sc->sc_rr;
|
|
if (cb->mmapped)
|
|
return EINVAL;
|
|
|
|
DPRINTFN(1,("audio_read: cc=%zu mode=%d\n",
|
|
uio->uio_resid, sc->sc_mode));
|
|
|
|
error = 0;
|
|
/*
|
|
* If hardware is half-duplex and currently playing, return
|
|
* silence blocks based on the number of blocks we have output.
|
|
*/
|
|
if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) {
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
for(;;) {
|
|
cc = sc->sc_pr.stamp - sc->sc_wstamp;
|
|
if (cc > 0)
|
|
break;
|
|
DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n",
|
|
sc->sc_pr.stamp, sc->sc_wstamp));
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return EWOULDBLOCK;
|
|
}
|
|
error = audio_sleep(&sc->sc_rchan, "aud_hr");
|
|
if (sc->sc_dying)
|
|
error = EIO;
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
splx(s);
|
|
|
|
if (uio->uio_resid < cc)
|
|
cc = uio->uio_resid;
|
|
DPRINTFN(1,("audio_read: reading in write mode, "
|
|
"cc=%d\n", cc));
|
|
error = audio_silence_copyout(sc, cc, uio);
|
|
sc->sc_wstamp += cc;
|
|
}
|
|
return error;
|
|
}
|
|
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
while ((used = audio_stream_get_used(sc->sc_rustream)) <= 0) {
|
|
if (!sc->sc_rbus) {
|
|
error = audiostartr(sc);
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return EWOULDBLOCK;
|
|
}
|
|
DPRINTFN(2, ("audio_read: sleep used=%d\n", used));
|
|
error = audio_sleep(&sc->sc_rchan, "aud_rd");
|
|
if (sc->sc_dying)
|
|
error = EIO;
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
|
|
outp = sc->sc_rustream->outp;
|
|
inp = sc->sc_rustream->inp;
|
|
cb->copying = TRUE;
|
|
splx(s);
|
|
|
|
/*
|
|
* cc is the amount of data in the sc_rustream excluding
|
|
* wrapped data
|
|
*/
|
|
cc = outp <= inp ? inp - outp :sc->sc_rustream->end - outp;
|
|
DPRINTFN(1,("audio_read: outp=%p, cc=%d\n", outp, cc));
|
|
|
|
n = uio->uio_resid;
|
|
error = uiomove(__UNCONST(outp), cc, uio);
|
|
n -= uio->uio_resid; /* number of bytes actually moved */
|
|
|
|
s = splaudio();
|
|
sc->sc_rustream->outp = audio_stream_add_outp
|
|
(sc->sc_rustream, outp, n);
|
|
cb->copying = FALSE;
|
|
splx(s);
|
|
}
|
|
return error;
|
|
}
|
|
|
|
void
|
|
audio_clear(struct audio_softc *sc)
|
|
{
|
|
int s;
|
|
|
|
s = splaudio();
|
|
if (sc->sc_rbus) {
|
|
audio_wakeup(&sc->sc_rchan);
|
|
sc->hw_if->halt_input(sc->hw_hdl);
|
|
sc->sc_rbus = FALSE;
|
|
}
|
|
if (sc->sc_pbus) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
sc->hw_if->halt_output(sc->hw_hdl);
|
|
sc->sc_pbus = FALSE;
|
|
}
|
|
splx(s);
|
|
}
|
|
|
|
void
|
|
audio_calc_blksize(struct audio_softc *sc, int mode)
|
|
{
|
|
const audio_params_t *parm;
|
|
struct audio_ringbuffer *rb;
|
|
|
|
if (sc->sc_blkset)
|
|
return;
|
|
|
|
if (mode == AUMODE_PLAY) {
|
|
rb = &sc->sc_pr;
|
|
parm = &rb->s.param;
|
|
} else {
|
|
rb = &sc->sc_rr;
|
|
parm = &rb->s.param;
|
|
}
|
|
|
|
rb->blksize = parm->sample_rate * audio_blk_ms / 1000 *
|
|
parm->channels * parm->precision / NBBY;
|
|
|
|
DPRINTF(("audio_calc_blksize: %s blksize=%d\n",
|
|
mode == AUMODE_PLAY ? "play" : "record", rb->blksize));
|
|
}
|
|
|
|
void
|
|
audio_fill_silence(struct audio_params *params, uint8_t *p, int n)
|
|
{
|
|
uint8_t auzero0, auzero1;
|
|
int nfill;
|
|
|
|
auzero1 = 0; /* initialize to please gcc */
|
|
nfill = 1;
|
|
switch (params->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
auzero0 = 0x7f;
|
|
break;
|
|
case AUDIO_ENCODING_ALAW:
|
|
auzero0 = 0x55;
|
|
break;
|
|
case AUDIO_ENCODING_MPEG_L1_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L1_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
|
|
case AUDIO_ENCODING_MPEG_L2_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L2_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
|
|
case AUDIO_ENCODING_ADPCM: /* is this right XXX */
|
|
case AUDIO_ENCODING_SLINEAR_LE:
|
|
case AUDIO_ENCODING_SLINEAR_BE:
|
|
auzero0 = 0;/* fortunately this works for any number of bits */
|
|
break;
|
|
case AUDIO_ENCODING_ULINEAR_LE:
|
|
case AUDIO_ENCODING_ULINEAR_BE:
|
|
if (params->precision > 8) {
|
|
nfill = (params->precision + NBBY - 1)/ NBBY;
|
|
auzero0 = 0x80;
|
|
auzero1 = 0;
|
|
} else
|
|
auzero0 = 0x80;
|
|
break;
|
|
default:
|
|
DPRINTF(("audio: bad encoding %d\n", params->encoding));
|
|
auzero0 = 0;
|
|
break;
|
|
}
|
|
if (nfill == 1) {
|
|
while (--n >= 0)
|
|
*p++ = auzero0; /* XXX memset */
|
|
} else /* nfill must no longer be 2 */ {
|
|
if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) {
|
|
int k = nfill;
|
|
while (--k > 0)
|
|
*p++ = auzero1;
|
|
n -= nfill - 1;
|
|
}
|
|
while (n >= nfill) {
|
|
int k = nfill;
|
|
*p++ = auzero0;
|
|
while (--k > 0)
|
|
*p++ = auzero1;
|
|
|
|
n -= nfill;
|
|
}
|
|
if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */
|
|
*p++ = auzero0;
|
|
}
|
|
}
|
|
|
|
int
|
|
audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio)
|
|
{
|
|
uint8_t zerobuf[128];
|
|
int error;
|
|
int k;
|
|
|
|
audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf);
|
|
|
|
error = 0;
|
|
while (n > 0 && uio->uio_resid > 0 && !error) {
|
|
k = min(n, min(uio->uio_resid, sizeof zerobuf));
|
|
error = uiomove(zerobuf, k, uio);
|
|
n -= k;
|
|
}
|
|
return error;
|
|
}
|
|
|
|
static int
|
|
uio_fetcher_fetch_to(stream_fetcher_t *self, audio_stream_t *p,
|
|
int max_used __unused)
|
|
{
|
|
uio_fetcher_t *this;
|
|
int size;
|
|
int stream_space;
|
|
int error;
|
|
|
|
this = (uio_fetcher_t *)self;
|
|
this->last_used = audio_stream_get_used(p);
|
|
if (this->last_used >= this->usedhigh)
|
|
return 0;
|
|
/*
|
|
* uio_fetcher ignores max_used and move the data as
|
|
* much as possible in order to return the correct value
|
|
* for audio_prinfo::seek and kfilters.
|
|
*/
|
|
stream_space = audio_stream_get_space(p);
|
|
size = min(this->uio->uio_resid, stream_space);
|
|
|
|
/* the first fragment of the space */
|
|
stream_space = p->end - p->inp;
|
|
if (stream_space >= size) {
|
|
error = uiomove(p->inp, size, this->uio);
|
|
if (error)
|
|
return error;
|
|
p->inp = audio_stream_add_inp(p, p->inp, size);
|
|
} else {
|
|
error = uiomove(p->inp, stream_space, this->uio);
|
|
if (error)
|
|
return error;
|
|
p->inp = audio_stream_add_inp(p, p->inp, stream_space);
|
|
error = uiomove(p->start, size - stream_space, this->uio);
|
|
if (error)
|
|
return error;
|
|
p->inp = audio_stream_add_inp(p, p->inp, size - stream_space);
|
|
}
|
|
this->last_used = audio_stream_get_used(p);
|
|
return 0;
|
|
}
|
|
|
|
static int
|
|
null_fetcher_fetch_to(stream_fetcher_t *self __unused,
|
|
audio_stream_t *p __unused, int max_used __unused)
|
|
{
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
uio_fetcher_ctor(uio_fetcher_t *this, struct uio *u, int h)
|
|
{
|
|
|
|
this->base.fetch_to = uio_fetcher_fetch_to;
|
|
this->uio = u;
|
|
this->usedhigh = h;
|
|
}
|
|
|
|
int
|
|
audio_write(struct audio_softc *sc, struct uio *uio, int ioflag)
|
|
{
|
|
uio_fetcher_t ufetcher;
|
|
audio_stream_t stream;
|
|
struct audio_ringbuffer *cb;
|
|
stream_fetcher_t *fetcher;
|
|
stream_filter_t *filter;
|
|
uint8_t *inp, *einp;
|
|
int saveerror, error, s, n, cc, used;
|
|
|
|
DPRINTFN(2,("audio_write: sc=%p count=%zu used=%d(hi=%d)\n",
|
|
sc, uio->uio_resid, audio_stream_get_used(sc->sc_pustream),
|
|
sc->sc_pr.usedhigh));
|
|
cb = &sc->sc_pr;
|
|
if (cb->mmapped)
|
|
return EINVAL;
|
|
|
|
if (uio->uio_resid == 0) {
|
|
sc->sc_eof++;
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* If half-duplex and currently recording, throw away data.
|
|
*/
|
|
if (!sc->sc_full_duplex &&
|
|
(sc->sc_mode & AUMODE_RECORD)) {
|
|
uio->uio_offset += uio->uio_resid;
|
|
uio->uio_resid = 0;
|
|
DPRINTF(("audio_write: half-dpx read busy\n"));
|
|
return 0;
|
|
}
|
|
|
|
if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) {
|
|
n = min(sc->sc_playdrop, uio->uio_resid);
|
|
DPRINTF(("audio_write: playdrop %d\n", n));
|
|
uio->uio_offset += n;
|
|
uio->uio_resid -= n;
|
|
sc->sc_playdrop -= n;
|
|
if (uio->uio_resid == 0)
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* setup filter pipeline
|
|
*/
|
|
uio_fetcher_ctor(&ufetcher, uio, cb->usedhigh);
|
|
if (sc->sc_npfilters > 0) {
|
|
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
|
|
} else {
|
|
fetcher = &ufetcher.base;
|
|
}
|
|
|
|
error = 0;
|
|
while (uio->uio_resid > 0 && !error) {
|
|
s = splaudio();
|
|
/* wait if the first buffer is occupied */
|
|
while ((used = audio_stream_get_used(sc->sc_pustream))
|
|
>= cb->usedhigh) {
|
|
DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d "
|
|
"hiwat=%d\n", used,
|
|
cb->usedlow, cb->usedhigh));
|
|
if (ioflag & IO_NDELAY) {
|
|
splx(s);
|
|
return EWOULDBLOCK;
|
|
}
|
|
error = audio_sleep(&sc->sc_wchan, "aud_wr");
|
|
if (sc->sc_dying)
|
|
error = EIO;
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
}
|
|
inp = cb->s.inp;
|
|
cb->copying = TRUE;
|
|
stream = cb->s;
|
|
used = stream.used;
|
|
splx(s);
|
|
|
|
/*
|
|
* write to the sc_pustream as much as possible
|
|
*
|
|
* work with a temporary audio_stream_t to narrow
|
|
* splaudio() enclosure
|
|
*/
|
|
|
|
sc->sc_writing = 1;
|
|
|
|
if (sc->sc_npfilters > 0) {
|
|
filter = sc->sc_pfilters[0];
|
|
filter->set_fetcher(filter, &ufetcher.base);
|
|
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
|
|
cc = cb->blksize * 2;
|
|
error = fetcher->fetch_to(fetcher, &stream, cc);
|
|
if (error != 0) {
|
|
fetcher = &ufetcher.base;
|
|
cc = sc->sc_pustream->end - sc->sc_pustream->start;
|
|
error = fetcher->fetch_to(fetcher, sc->sc_pustream, cc);
|
|
}
|
|
} else {
|
|
fetcher = &ufetcher.base;
|
|
cc = stream.end - stream.start;
|
|
error = fetcher->fetch_to(fetcher, &stream, cc);
|
|
}
|
|
sc->sc_writing = 0;
|
|
if (sc->sc_waitcomp)
|
|
wakeup(sc);
|
|
|
|
s = splaudio();
|
|
if (sc->sc_npfilters > 0) {
|
|
cb->fstamp += ufetcher.last_used
|
|
- audio_stream_get_used(sc->sc_pustream);
|
|
}
|
|
cb->s.used += stream.used - used;
|
|
cb->s.inp = stream.inp;
|
|
einp = cb->s.inp;
|
|
|
|
/*
|
|
* This is a very suboptimal way of keeping track of
|
|
* silence in the buffer, but it is simple.
|
|
*/
|
|
sc->sc_sil_count = 0;
|
|
|
|
/*
|
|
* If the interrupt routine wants the last block filled AND
|
|
* the copy did not fill the last block completely it needs to
|
|
* be padded.
|
|
*/
|
|
if (cb->needfill && inp < einp &&
|
|
(inp - cb->s.start) / cb->blksize ==
|
|
(einp - cb->s.start) / cb->blksize) {
|
|
/* Figure out how many bytes to a block boundary. */
|
|
cc = cb->blksize - (einp - cb->s.start) % cb->blksize;
|
|
DPRINTF(("audio_write: partial fill %d\n", cc));
|
|
} else
|
|
cc = 0;
|
|
cb->needfill = FALSE;
|
|
cb->copying = FALSE;
|
|
if (!sc->sc_pbus && !cb->pause) {
|
|
saveerror = error;
|
|
error = audiostartp(sc);
|
|
if (saveerror != 0) {
|
|
/* Report the first error that occurred. */
|
|
error = saveerror;
|
|
}
|
|
}
|
|
splx(s);
|
|
if (cc != 0) {
|
|
DPRINTFN(1, ("audio_write: fill %d\n", cc));
|
|
audio_fill_silence(&cb->s.param, einp, cc);
|
|
}
|
|
}
|
|
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audio_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag,
|
|
struct lwp *l)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
struct audio_offset *ao;
|
|
u_long stamp;
|
|
int error, s, offs, fd;
|
|
boolean_t rbus, pbus;
|
|
|
|
DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n",
|
|
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
|
|
hw = sc->hw_if;
|
|
error = 0;
|
|
switch (cmd) {
|
|
case FIONBIO:
|
|
/* All handled in the upper FS layer. */
|
|
break;
|
|
|
|
case FIONREAD:
|
|
*(int *)addr = audio_stream_get_used(sc->sc_rustream);
|
|
break;
|
|
|
|
case FIOASYNC:
|
|
if (*(int *)addr) {
|
|
if (sc->sc_async_audio)
|
|
return EBUSY;
|
|
sc->sc_async_audio = l->l_proc;
|
|
DPRINTF(("audio_ioctl: FIOASYNC %p\n", l->l_proc));
|
|
} else
|
|
sc->sc_async_audio = 0;
|
|
break;
|
|
|
|
case AUDIO_FLUSH:
|
|
DPRINTF(("AUDIO_FLUSH\n"));
|
|
rbus = sc->sc_rbus;
|
|
pbus = sc->sc_pbus;
|
|
audio_clear(sc);
|
|
s = splaudio();
|
|
error = audio_initbufs(sc);
|
|
if (error) {
|
|
splx(s);
|
|
return error;
|
|
}
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus)
|
|
error = audiostartp(sc);
|
|
if (!error &&
|
|
(sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus)
|
|
error = audiostartr(sc);
|
|
splx(s);
|
|
break;
|
|
|
|
/*
|
|
* Number of read (write) samples dropped. We don't know where or
|
|
* when they were dropped.
|
|
*/
|
|
case AUDIO_RERROR:
|
|
*(int *)addr = sc->sc_rr.drops;
|
|
break;
|
|
|
|
case AUDIO_PERROR:
|
|
*(int *)addr = sc->sc_pr.drops;
|
|
break;
|
|
|
|
/*
|
|
* Offsets into buffer.
|
|
*/
|
|
case AUDIO_GETIOFFS:
|
|
ao = (struct audio_offset *)addr;
|
|
s = splaudio();
|
|
/* figure out where next DMA will start */
|
|
stamp = sc->sc_rustream == &sc->sc_rr.s
|
|
? sc->sc_rr.stamp : sc->sc_rr.fstamp;
|
|
offs = sc->sc_rustream->inp - sc->sc_rustream->start;
|
|
splx(s);
|
|
ao->samples = stamp;
|
|
ao->deltablks =
|
|
(stamp / sc->sc_rr.blksize) -
|
|
(sc->sc_rr.stamp_last / sc->sc_rr.blksize);
|
|
sc->sc_rr.stamp_last = stamp;
|
|
ao->offset = offs;
|
|
break;
|
|
|
|
case AUDIO_GETOOFFS:
|
|
ao = (struct audio_offset *)addr;
|
|
s = splaudio();
|
|
/* figure out where next DMA will start */
|
|
stamp = sc->sc_pustream == &sc->sc_pr.s
|
|
? sc->sc_pr.stamp : sc->sc_pr.fstamp;
|
|
offs = sc->sc_pustream->outp - sc->sc_pustream->start
|
|
+ sc->sc_pr.blksize;
|
|
splx(s);
|
|
ao->samples = stamp;
|
|
ao->deltablks =
|
|
(stamp / sc->sc_pr.blksize) -
|
|
(sc->sc_pr.stamp_last / sc->sc_pr.blksize);
|
|
sc->sc_pr.stamp_last = stamp;
|
|
if (sc->sc_pustream->start + offs >= sc->sc_pustream->end)
|
|
offs = 0;
|
|
ao->offset = offs;
|
|
break;
|
|
|
|
/*
|
|
* How many bytes will elapse until mike hears the first
|
|
* sample of what we write next?
|
|
*/
|
|
case AUDIO_WSEEK:
|
|
*(u_long *)addr = audio_stream_get_used(sc->sc_rustream);
|
|
break;
|
|
|
|
case AUDIO_SETINFO:
|
|
DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode));
|
|
error = audiosetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETINFO:
|
|
DPRINTF(("AUDIO_GETINFO\n"));
|
|
error = audiogetinfo(sc, (struct audio_info *)addr);
|
|
break;
|
|
|
|
case AUDIO_DRAIN:
|
|
DPRINTF(("AUDIO_DRAIN\n"));
|
|
error = audio_drain(sc);
|
|
if (!error && hw->drain)
|
|
error = hw->drain(sc->hw_hdl);
|
|
break;
|
|
|
|
case AUDIO_GETDEV:
|
|
DPRINTF(("AUDIO_GETDEV\n"));
|
|
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETENC:
|
|
DPRINTF(("AUDIO_GETENC\n"));
|
|
error = hw->query_encoding(sc->hw_hdl,
|
|
(struct audio_encoding *)addr);
|
|
break;
|
|
|
|
case AUDIO_GETFD:
|
|
DPRINTF(("AUDIO_GETFD\n"));
|
|
*(int *)addr = sc->sc_full_duplex;
|
|
break;
|
|
|
|
case AUDIO_SETFD:
|
|
DPRINTF(("AUDIO_SETFD\n"));
|
|
fd = *(int *)addr;
|
|
if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) {
|
|
if (hw->setfd)
|
|
error = hw->setfd(sc->hw_hdl, fd);
|
|
else
|
|
error = 0;
|
|
if (!error)
|
|
sc->sc_full_duplex = fd;
|
|
} else {
|
|
if (fd)
|
|
error = ENOTTY;
|
|
else
|
|
error = 0;
|
|
}
|
|
break;
|
|
|
|
case AUDIO_GETPROPS:
|
|
DPRINTF(("AUDIO_GETPROPS\n"));
|
|
*(int *)addr = hw->get_props(sc->hw_hdl);
|
|
break;
|
|
|
|
default:
|
|
if (hw->dev_ioctl) {
|
|
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l);
|
|
} else {
|
|
DPRINTF(("audio_ioctl: unknown ioctl\n"));
|
|
error = EINVAL;
|
|
}
|
|
break;
|
|
}
|
|
DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n",
|
|
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
|
|
return error;
|
|
}
|
|
|
|
int
|
|
audio_poll(struct audio_softc *sc, int events, struct lwp *l)
|
|
{
|
|
int revents;
|
|
int s;
|
|
int used;
|
|
|
|
DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode));
|
|
|
|
revents = 0;
|
|
s = splaudio();
|
|
if (events & (POLLIN | POLLRDNORM)) {
|
|
used = audio_stream_get_used(sc->sc_rustream);
|
|
/*
|
|
* If half duplex and playing, audio_read() will generate
|
|
* silence at the play rate; poll for silence being
|
|
* available. Otherwise, poll for recorded sound.
|
|
*/
|
|
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ?
|
|
sc->sc_pr.stamp > sc->sc_wstamp :
|
|
used > sc->sc_rr.usedlow)
|
|
revents |= events & (POLLIN | POLLRDNORM);
|
|
}
|
|
|
|
if (events & (POLLOUT | POLLWRNORM)) {
|
|
used = audio_stream_get_used(sc->sc_pustream);
|
|
/*
|
|
* If half duplex and recording, audio_write() will throw
|
|
* away play data, which means we are always ready to write.
|
|
* Otherwise, poll for play buffer being below its low water
|
|
* mark.
|
|
*/
|
|
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) ||
|
|
used <= sc->sc_pr.usedlow)
|
|
revents |= events & (POLLOUT | POLLWRNORM);
|
|
}
|
|
|
|
if (revents == 0) {
|
|
if (events & (POLLIN | POLLRDNORM))
|
|
selrecord(l, &sc->sc_rsel);
|
|
|
|
if (events & (POLLOUT | POLLWRNORM))
|
|
selrecord(l, &sc->sc_wsel);
|
|
}
|
|
|
|
splx(s);
|
|
return revents;
|
|
}
|
|
|
|
static void
|
|
filt_audiordetach(struct knote *kn)
|
|
{
|
|
struct audio_softc *sc;
|
|
int s;
|
|
|
|
sc = kn->kn_hook;
|
|
s = splaudio();
|
|
SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext);
|
|
splx(s);
|
|
}
|
|
|
|
static int
|
|
filt_audioread(struct knote *kn, long hint __unused)
|
|
{
|
|
struct audio_softc *sc;
|
|
int s;
|
|
|
|
sc = kn->kn_hook;
|
|
s = splaudio();
|
|
if (!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY))
|
|
kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp;
|
|
else
|
|
kn->kn_data = audio_stream_get_used(sc->sc_rustream)
|
|
- sc->sc_rr.usedlow;
|
|
splx(s);
|
|
|
|
return kn->kn_data > 0;
|
|
}
|
|
|
|
static const struct filterops audioread_filtops =
|
|
{ 1, NULL, filt_audiordetach, filt_audioread };
|
|
|
|
static void
|
|
filt_audiowdetach(struct knote *kn)
|
|
{
|
|
struct audio_softc *sc;
|
|
int s;
|
|
|
|
sc = kn->kn_hook;
|
|
s = splaudio();
|
|
SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext);
|
|
splx(s);
|
|
}
|
|
|
|
static int
|
|
filt_audiowrite(struct knote *kn, long hint __unused)
|
|
{
|
|
struct audio_softc *sc;
|
|
audio_stream_t *stream;
|
|
int s;
|
|
|
|
sc = kn->kn_hook;
|
|
s = splaudio();
|
|
stream = sc->sc_pustream;
|
|
kn->kn_data = (stream->end - stream->start)
|
|
- audio_stream_get_used(stream);
|
|
splx(s);
|
|
|
|
return kn->kn_data > 0;
|
|
}
|
|
|
|
static const struct filterops audiowrite_filtops =
|
|
{ 1, NULL, filt_audiowdetach, filt_audiowrite };
|
|
|
|
int
|
|
audio_kqfilter(struct audio_softc *sc, struct knote *kn)
|
|
{
|
|
struct klist *klist;
|
|
int s;
|
|
|
|
switch (kn->kn_filter) {
|
|
case EVFILT_READ:
|
|
klist = &sc->sc_rsel.sel_klist;
|
|
kn->kn_fop = &audioread_filtops;
|
|
break;
|
|
|
|
case EVFILT_WRITE:
|
|
klist = &sc->sc_wsel.sel_klist;
|
|
kn->kn_fop = &audiowrite_filtops;
|
|
break;
|
|
|
|
default:
|
|
return 1;
|
|
}
|
|
|
|
kn->kn_hook = sc;
|
|
|
|
s = splaudio();
|
|
SLIST_INSERT_HEAD(klist, kn, kn_selnext);
|
|
splx(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
paddr_t
|
|
audio_mmap(struct audio_softc *sc, off_t off, int prot)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
struct audio_ringbuffer *cb;
|
|
int s;
|
|
|
|
DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)off, prot));
|
|
hw = sc->hw_if;
|
|
if (!(hw->get_props(sc->hw_hdl) & AUDIO_PROP_MMAP) || !hw->mappage)
|
|
return -1;
|
|
#if 0
|
|
/* XXX
|
|
* The idea here was to use the protection to determine if
|
|
* we are mapping the read or write buffer, but it fails.
|
|
* The VM system is broken in (at least) two ways.
|
|
* 1) If you map memory VM_PROT_WRITE you SIGSEGV
|
|
* when writing to it, so VM_PROT_READ|VM_PROT_WRITE
|
|
* has to be used for mmapping the play buffer.
|
|
* 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE
|
|
* audio_mmap will get called at some point with VM_PROT_READ
|
|
* only.
|
|
* So, alas, we always map the play buffer for now.
|
|
*/
|
|
if (prot == (VM_PROT_READ|VM_PROT_WRITE) ||
|
|
prot == VM_PROT_WRITE)
|
|
cb = &sc->sc_pr;
|
|
else if (prot == VM_PROT_READ)
|
|
cb = &sc->sc_rr;
|
|
else
|
|
return -1;
|
|
#else
|
|
cb = &sc->sc_pr;
|
|
#endif
|
|
|
|
if ((u_int)off >= cb->s.bufsize)
|
|
return -1;
|
|
if (!cb->mmapped) {
|
|
cb->mmapped = TRUE;
|
|
if (cb == &sc->sc_pr) {
|
|
audio_fill_silence(&cb->s.param, cb->s.start,
|
|
cb->s.bufsize);
|
|
s = splaudio();
|
|
sc->sc_pustream = &cb->s;
|
|
if (!sc->sc_pbus)
|
|
(void)audiostartp(sc);
|
|
splx(s);
|
|
} else {
|
|
s = splaudio();
|
|
sc->sc_rustream = &cb->s;
|
|
if (!sc->sc_rbus)
|
|
(void)audiostartr(sc);
|
|
splx(s);
|
|
}
|
|
}
|
|
|
|
return hw->mappage(sc->hw_hdl, cb->s.start, off, prot);
|
|
}
|
|
|
|
int
|
|
audiostartr(struct audio_softc *sc)
|
|
{
|
|
int error;
|
|
|
|
DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n",
|
|
sc->sc_rr.s.start, audio_stream_get_used(&sc->sc_rr.s),
|
|
sc->sc_rr.usedhigh, sc->sc_rr.mmapped));
|
|
|
|
if (sc->hw_if->trigger_input)
|
|
error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.s.start,
|
|
sc->sc_rr.s.end, sc->sc_rr.blksize,
|
|
audio_rint, (void *)sc, &sc->sc_rr.s.param);
|
|
else
|
|
error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.s.start,
|
|
sc->sc_rr.blksize, audio_rint, (void *)sc);
|
|
if (error) {
|
|
DPRINTF(("audiostartr failed: %d\n", error));
|
|
return error;
|
|
}
|
|
sc->sc_rbus = TRUE;
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audiostartp(struct audio_softc *sc)
|
|
{
|
|
int error;
|
|
int used;
|
|
|
|
used = audio_stream_get_used(&sc->sc_pr.s);
|
|
DPRINTF(("audiostartp: start=%p used=%d(hi=%d blk=%d) mmapped=%d\n",
|
|
sc->sc_pr.s.start, used, sc->sc_pr.usedhigh,
|
|
sc->sc_pr.blksize, sc->sc_pr.mmapped));
|
|
|
|
if (!sc->sc_pr.mmapped && used < sc->sc_pr.blksize) {
|
|
wakeup(&sc->sc_wchan);
|
|
DPRINTF(("%s: wakeup and return\n", __func__));
|
|
return 0;
|
|
}
|
|
|
|
if (sc->hw_if->trigger_output) {
|
|
DPRINTF(("%s: call trigger_output\n", __func__));
|
|
error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.s.start,
|
|
sc->sc_pr.s.end, sc->sc_pr.blksize,
|
|
audio_pint, (void *)sc, &sc->sc_pr.s.param);
|
|
} else {
|
|
DPRINTF(("%s: call start_output\n", __func__));
|
|
error = sc->hw_if->start_output(sc->hw_hdl,
|
|
__UNCONST(sc->sc_pr.s.outp), sc->sc_pr.blksize,
|
|
audio_pint, (void *)sc);
|
|
}
|
|
if (error) {
|
|
DPRINTF(("audiostartp failed: %d\n", error));
|
|
return error;
|
|
}
|
|
sc->sc_pbus = TRUE;
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* When the play interrupt routine finds that the write isn't keeping
|
|
* the buffer filled it will insert silence in the buffer to make up
|
|
* for this. The part of the buffer that is filled with silence
|
|
* is kept track of in a very approximate way: it starts at sc_sil_start
|
|
* and extends sc_sil_count bytes. If there is already silence in
|
|
* the requested area nothing is done; so when the whole buffer is
|
|
* silent nothing happens. When the writer starts again sc_sil_count
|
|
* is set to 0.
|
|
*/
|
|
/* XXX
|
|
* Putting silence into the output buffer should not really be done
|
|
* at splaudio, but there is no softaudio level to do it at yet.
|
|
*/
|
|
static inline void
|
|
audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb,
|
|
uint8_t *inp, int cc)
|
|
{
|
|
uint8_t *s, *e, *p, *q;
|
|
|
|
if (sc->sc_sil_count > 0) {
|
|
s = sc->sc_sil_start; /* start of silence */
|
|
e = s + sc->sc_sil_count; /* end of sil., may be beyond end */
|
|
p = inp; /* adjusted pointer to area to fill */
|
|
if (p < s)
|
|
p += cb->s.end - cb->s.start;
|
|
q = p + cc;
|
|
/* Check if there is already silence. */
|
|
if (!(s <= p && p < e &&
|
|
s <= q && q <= e)) {
|
|
if (s <= p)
|
|
sc->sc_sil_count = max(sc->sc_sil_count, q-s);
|
|
DPRINTFN(5,("audio_pint_silence: fill cc=%d inp=%p, "
|
|
"count=%d size=%d\n",
|
|
cc, inp, sc->sc_sil_count,
|
|
(int)(cb->s.end - cb->s.start)));
|
|
audio_fill_silence(&cb->s.param, inp, cc);
|
|
} else {
|
|
DPRINTFN(5,("audio_pint_silence: already silent "
|
|
"cc=%d inp=%p\n", cc, inp));
|
|
|
|
}
|
|
} else {
|
|
sc->sc_sil_start = inp;
|
|
sc->sc_sil_count = cc;
|
|
DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n",
|
|
inp, cc));
|
|
audio_fill_silence(&cb->s.param, inp, cc);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from HW driver module on completion of DMA output.
|
|
* Start output of new block, wrap in ring buffer if needed.
|
|
* If no more buffers to play, output zero instead.
|
|
* Do a wakeup if necessary.
|
|
*/
|
|
void
|
|
audio_pint(void *v)
|
|
{
|
|
stream_fetcher_t null_fetcher;
|
|
struct audio_softc *sc;
|
|
const struct audio_hw_if *hw;
|
|
struct audio_ringbuffer *cb;
|
|
stream_fetcher_t *fetcher;
|
|
uint8_t *inp;
|
|
int cc, used;
|
|
int blksize;
|
|
int error;
|
|
|
|
sc = v;
|
|
if (!sc->sc_open)
|
|
return; /* ignore interrupt if not open */
|
|
|
|
hw = sc->hw_if;
|
|
cb = &sc->sc_pr;
|
|
blksize = cb->blksize;
|
|
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
|
|
cb->stamp += blksize;
|
|
if (cb->mmapped) {
|
|
DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n",
|
|
cb->s.outp, blksize, cb->s.inp));
|
|
if (hw->trigger_output == NULL)
|
|
(void)hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp),
|
|
blksize, audio_pint, (void *)sc);
|
|
return;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
{
|
|
struct timeval tv;
|
|
u_long t;
|
|
microtime(&tv);
|
|
t = tv.tv_usec + 1000000 * tv.tv_sec;
|
|
if (sc->sc_pnintr) {
|
|
long lastdelta, totdelta;
|
|
lastdelta = t - sc->sc_plastintr - sc->sc_pblktime;
|
|
if (lastdelta > sc->sc_pblktime / 3) {
|
|
printf("audio: play interrupt(%d) off "
|
|
"relative by %ld us (%lu)\n",
|
|
sc->sc_pnintr, lastdelta,
|
|
sc->sc_pblktime);
|
|
}
|
|
totdelta = t - sc->sc_pfirstintr -
|
|
sc->sc_pblktime * sc->sc_pnintr;
|
|
if (totdelta > sc->sc_pblktime) {
|
|
printf("audio: play interrupt(%d) off "
|
|
"absolute by %ld us (%lu) (LOST)\n",
|
|
sc->sc_pnintr, totdelta,
|
|
sc->sc_pblktime);
|
|
sc->sc_pnintr++; /* avoid repeated messages */
|
|
}
|
|
} else
|
|
sc->sc_pfirstintr = t;
|
|
sc->sc_plastintr = t;
|
|
sc->sc_pnintr++;
|
|
}
|
|
#endif
|
|
|
|
used = audio_stream_get_used(&cb->s);
|
|
/*
|
|
* "used <= cb->usedlow" should be "used < blksize" ideally.
|
|
* Some HW drivers such as uaudio(4) does not call audio_pint()
|
|
* at accurate timing. If used < blksize, uaudio(4) already
|
|
* request transfer of garbage data.
|
|
*/
|
|
if (used <= cb->usedlow && !cb->copying && sc->sc_npfilters > 0) {
|
|
/* we might have data in filter pipeline */
|
|
null_fetcher.fetch_to = null_fetcher_fetch_to;
|
|
fetcher = &sc->sc_pfilters[sc->sc_npfilters - 1]->base;
|
|
sc->sc_pfilters[0]->set_fetcher(sc->sc_pfilters[0],
|
|
&null_fetcher);
|
|
used = audio_stream_get_used(sc->sc_pustream);
|
|
cc = cb->s.end - cb->s.start;
|
|
if (blksize * 2 < cc)
|
|
cc = blksize * 2;
|
|
fetcher->fetch_to(fetcher, &cb->s, cc);
|
|
cb->fstamp += used - audio_stream_get_used(sc->sc_pustream);
|
|
used = audio_stream_get_used(&cb->s);
|
|
}
|
|
if (used < blksize) {
|
|
/* we don't have a full block to use */
|
|
if (cb->copying) {
|
|
/* writer is in progress, don't disturb */
|
|
cb->needfill = TRUE;
|
|
DPRINTFN(1, ("audio_pint: copying in progress\n"));
|
|
} else {
|
|
inp = cb->s.inp;
|
|
cc = blksize - (inp - cb->s.start) % blksize;
|
|
if (cb->pause)
|
|
cb->pdrops += cc;
|
|
else {
|
|
cb->drops += cc;
|
|
sc->sc_playdrop += cc;
|
|
}
|
|
audio_pint_silence(sc, cb, inp, cc);
|
|
cb->s.inp = audio_stream_add_inp(&cb->s, inp, cc);
|
|
|
|
/* Clear next block so we keep ahead of the DMA. */
|
|
used = audio_stream_get_used(&cb->s);
|
|
if (used + cc < cb->s.end - cb->s.start)
|
|
audio_pint_silence(sc, cb, inp, blksize);
|
|
}
|
|
}
|
|
|
|
DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->s.outp, blksize));
|
|
if (hw->trigger_output == NULL) {
|
|
error = hw->start_output(sc->hw_hdl, __UNCONST(cb->s.outp),
|
|
blksize, audio_pint, (void *)sc);
|
|
if (error) {
|
|
/* XXX does this really help? */
|
|
DPRINTF(("audio_pint restart failed: %d\n", error));
|
|
audio_clear(sc);
|
|
}
|
|
}
|
|
|
|
DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n",
|
|
sc->sc_mode, cb->pause,
|
|
audio_stream_get_used(sc->sc_pustream), cb->usedlow));
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
|
|
if (audio_stream_get_used(sc->sc_pustream) <= cb->usedlow) {
|
|
audio_wakeup(&sc->sc_wchan);
|
|
selnotify(&sc->sc_wsel, 0);
|
|
if (sc->sc_async_audio) {
|
|
DPRINTFN(3, ("audio_pint: sending SIGIO %p\n",
|
|
sc->sc_async_audio));
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Possible to return one or more "phantom blocks" now. */
|
|
if (!sc->sc_full_duplex && sc->sc_rchan) {
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selnotify(&sc->sc_rsel, 0);
|
|
if (sc->sc_async_audio)
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Called from HW driver module on completion of DMA input.
|
|
* Mark it as input in the ring buffer (fiddle pointers).
|
|
* Do a wakeup if necessary.
|
|
*/
|
|
void
|
|
audio_rint(void *v)
|
|
{
|
|
stream_fetcher_t null_fetcher;
|
|
struct audio_softc *sc;
|
|
const struct audio_hw_if *hw;
|
|
struct audio_ringbuffer *cb;
|
|
stream_fetcher_t *last_fetcher;
|
|
int cc;
|
|
int used;
|
|
int blksize;
|
|
int error;
|
|
|
|
sc = v;
|
|
cb = &sc->sc_rr;
|
|
if (!sc->sc_open)
|
|
return; /* ignore interrupt if not open */
|
|
|
|
hw = sc->hw_if;
|
|
blksize = cb->blksize;
|
|
cb->s.inp = audio_stream_add_inp(&cb->s, cb->s.inp, blksize);
|
|
cb->stamp += blksize;
|
|
if (cb->mmapped) {
|
|
DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n",
|
|
cb->s.inp, blksize));
|
|
if (hw->trigger_input == NULL)
|
|
(void)hw->start_input(sc->hw_hdl, cb->s.inp, blksize,
|
|
audio_rint, (void *)sc);
|
|
return;
|
|
}
|
|
|
|
#ifdef AUDIO_INTR_TIME
|
|
{
|
|
struct timeval tv;
|
|
u_long t;
|
|
microtime(&tv);
|
|
t = tv.tv_usec + 1000000 * tv.tv_sec;
|
|
if (sc->sc_rnintr) {
|
|
long lastdelta, totdelta;
|
|
lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime;
|
|
if (lastdelta > sc->sc_rblktime / 5) {
|
|
printf("audio: record interrupt(%d) off "
|
|
"relative by %ld us (%lu)\n",
|
|
sc->sc_rnintr, lastdelta,
|
|
sc->sc_rblktime);
|
|
}
|
|
totdelta = t - sc->sc_rfirstintr -
|
|
sc->sc_rblktime * sc->sc_rnintr;
|
|
if (totdelta > sc->sc_rblktime / 2) {
|
|
sc->sc_rnintr++;
|
|
printf("audio: record interrupt(%d) off "
|
|
"absolute by %ld us (%lu)\n",
|
|
sc->sc_rnintr, totdelta,
|
|
sc->sc_rblktime);
|
|
sc->sc_rnintr++; /* avoid repeated messages */
|
|
}
|
|
} else
|
|
sc->sc_rfirstintr = t;
|
|
sc->sc_rlastintr = t;
|
|
sc->sc_rnintr++;
|
|
}
|
|
#endif
|
|
|
|
if (!cb->pause && sc->sc_nrfilters > 0) {
|
|
null_fetcher.fetch_to = null_fetcher_fetch_to;
|
|
last_fetcher = &sc->sc_rfilters[sc->sc_nrfilters - 1]->base;
|
|
sc->sc_rfilters[0]->set_fetcher(sc->sc_rfilters[0],
|
|
&null_fetcher);
|
|
used = audio_stream_get_used(sc->sc_rustream);
|
|
cc = sc->sc_rustream->end - sc->sc_rustream->start;
|
|
error = last_fetcher->fetch_to
|
|
(last_fetcher, sc->sc_rustream, cc);
|
|
cb->fstamp += audio_stream_get_used(sc->sc_rustream) - used;
|
|
/* XXX what should do for error? */
|
|
}
|
|
used = audio_stream_get_used(&sc->sc_rr.s);
|
|
if (cb->pause) {
|
|
DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops));
|
|
cb->pdrops += blksize;
|
|
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
|
|
} else if (used + blksize > cb->s.end - cb->s.start && !cb->copying) {
|
|
DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops));
|
|
cb->drops += blksize;
|
|
cb->s.outp = audio_stream_add_outp(&cb->s, cb->s.outp, blksize);
|
|
}
|
|
|
|
DPRINTFN(2, ("audio_rint: inp=%p cc=%d\n", cb->s.inp, blksize));
|
|
if (hw->trigger_input == NULL) {
|
|
error = hw->start_input(sc->hw_hdl, cb->s.inp, blksize,
|
|
audio_rint, (void *)sc);
|
|
if (error) {
|
|
/* XXX does this really help? */
|
|
DPRINTF(("audio_rint: restart failed: %d\n", error));
|
|
audio_clear(sc);
|
|
}
|
|
}
|
|
|
|
audio_wakeup(&sc->sc_rchan);
|
|
selnotify(&sc->sc_rsel, 0);
|
|
if (sc->sc_async_audio)
|
|
psignal(sc->sc_async_audio, SIGIO);
|
|
}
|
|
|
|
int
|
|
audio_check_params(struct audio_params *p)
|
|
{
|
|
|
|
if (p->encoding == AUDIO_ENCODING_PCM16) {
|
|
if (p->precision == 8)
|
|
p->encoding = AUDIO_ENCODING_ULINEAR;
|
|
else
|
|
p->encoding = AUDIO_ENCODING_SLINEAR;
|
|
} else if (p->encoding == AUDIO_ENCODING_PCM8) {
|
|
if (p->precision == 8)
|
|
p->encoding = AUDIO_ENCODING_ULINEAR;
|
|
else
|
|
return EINVAL;
|
|
}
|
|
|
|
if (p->encoding == AUDIO_ENCODING_SLINEAR)
|
|
#if BYTE_ORDER == LITTLE_ENDIAN
|
|
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
|
|
#else
|
|
p->encoding = AUDIO_ENCODING_SLINEAR_BE;
|
|
#endif
|
|
if (p->encoding == AUDIO_ENCODING_ULINEAR)
|
|
#if BYTE_ORDER == LITTLE_ENDIAN
|
|
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
|
|
#else
|
|
p->encoding = AUDIO_ENCODING_ULINEAR_BE;
|
|
#endif
|
|
|
|
switch (p->encoding) {
|
|
case AUDIO_ENCODING_ULAW:
|
|
case AUDIO_ENCODING_ALAW:
|
|
if (p->precision != 8)
|
|
return EINVAL;
|
|
break;
|
|
case AUDIO_ENCODING_ADPCM:
|
|
if (p->precision != 4 && p->precision != 8)
|
|
return EINVAL;
|
|
break;
|
|
case AUDIO_ENCODING_SLINEAR_LE:
|
|
case AUDIO_ENCODING_SLINEAR_BE:
|
|
case AUDIO_ENCODING_ULINEAR_LE:
|
|
case AUDIO_ENCODING_ULINEAR_BE:
|
|
/* XXX is: our zero-fill can handle any multiple of 8 */
|
|
if (p->precision != 8 && p->precision != 16 &&
|
|
p->precision != 24 && p->precision != 32)
|
|
return EINVAL;
|
|
if (p->precision == 8 && p->encoding == AUDIO_ENCODING_SLINEAR_BE)
|
|
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
|
|
if (p->precision == 8 && p->encoding == AUDIO_ENCODING_ULINEAR_BE)
|
|
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
|
|
if (p->validbits > p->precision)
|
|
return EINVAL;
|
|
break;
|
|
case AUDIO_ENCODING_MPEG_L1_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L1_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
|
|
case AUDIO_ENCODING_MPEG_L2_STREAM:
|
|
case AUDIO_ENCODING_MPEG_L2_PACKETS:
|
|
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
|
|
break;
|
|
default:
|
|
return EINVAL;
|
|
}
|
|
|
|
/* sanity check # of channels*/
|
|
if (p->channels < 1 || p->channels > AUDIO_MAX_CHANNELS)
|
|
return EINVAL;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audio_set_defaults(struct audio_softc *sc, u_int mode)
|
|
{
|
|
struct audio_info ai;
|
|
|
|
/* default parameters */
|
|
sc->sc_rparams = audio_default;
|
|
sc->sc_pparams = audio_default;
|
|
sc->sc_blkset = FALSE;
|
|
|
|
AUDIO_INITINFO(&ai);
|
|
ai.record.sample_rate = sc->sc_rparams.sample_rate;
|
|
ai.record.encoding = sc->sc_rparams.encoding;
|
|
ai.record.channels = sc->sc_rparams.channels;
|
|
ai.record.precision = sc->sc_rparams.precision;
|
|
ai.play.sample_rate = sc->sc_pparams.sample_rate;
|
|
ai.play.encoding = sc->sc_pparams.encoding;
|
|
ai.play.channels = sc->sc_pparams.channels;
|
|
ai.play.precision = sc->sc_pparams.precision;
|
|
ai.mode = mode;
|
|
|
|
return audiosetinfo(sc, &ai);
|
|
}
|
|
|
|
int
|
|
au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r)
|
|
{
|
|
|
|
ct->type = AUDIO_MIXER_VALUE;
|
|
ct->un.value.num_channels = 2;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
|
|
if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0)
|
|
return 0;
|
|
ct->un.value.num_channels = 1;
|
|
ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
|
|
return sc->hw_if->set_port(sc->hw_hdl, ct);
|
|
}
|
|
|
|
int
|
|
au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
|
|
int gain, int balance)
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, error;
|
|
int l, r;
|
|
u_int mask;
|
|
int nset;
|
|
|
|
if (balance == AUDIO_MID_BALANCE) {
|
|
l = r = gain;
|
|
} else if (balance < AUDIO_MID_BALANCE) {
|
|
l = gain;
|
|
r = (balance * gain) / AUDIO_MID_BALANCE;
|
|
} else {
|
|
r = gain;
|
|
l = ((AUDIO_RIGHT_BALANCE - balance) * gain)
|
|
/ AUDIO_MID_BALANCE;
|
|
}
|
|
DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n",
|
|
gain, balance, l, r));
|
|
|
|
if (ports->index == -1) {
|
|
usemaster:
|
|
if (ports->master == -1)
|
|
return 0; /* just ignore it silently */
|
|
ct.dev = ports->master;
|
|
error = au_set_lr_value(sc, &ct, l, r);
|
|
} else {
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return error;
|
|
if (ports->isdual) {
|
|
if (ports->cur_port == -1)
|
|
ct.dev = ports->master;
|
|
else
|
|
ct.dev = ports->miport[ports->cur_port];
|
|
error = au_set_lr_value(sc, &ct, l, r);
|
|
} else {
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->misel[i] == ct.un.ord) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_set_lr_value(sc, &ct, l, r))
|
|
goto usemaster;
|
|
else
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return error;
|
|
mask = ct.un.mask;
|
|
nset = 0;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] & mask) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev != -1 &&
|
|
au_set_lr_value(sc, &ct, l, r) == 0)
|
|
nset++;
|
|
}
|
|
}
|
|
if (nset == 0)
|
|
goto usemaster;
|
|
}
|
|
}
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r)
|
|
{
|
|
int error;
|
|
|
|
ct->un.value.num_channels = 2;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) {
|
|
*l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
|
|
*r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
|
|
} else {
|
|
ct->un.value.num_channels = 1;
|
|
error = sc->hw_if->get_port(sc->hw_hdl, ct);
|
|
if (error)
|
|
return error;
|
|
*r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void
|
|
au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
|
|
u_int *pgain, u_char *pbalance)
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, l, r, n;
|
|
int lgain, rgain;
|
|
|
|
lgain = AUDIO_MAX_GAIN / 2;
|
|
rgain = AUDIO_MAX_GAIN / 2;
|
|
if (ports->index == -1) {
|
|
usemaster:
|
|
if (ports->master == -1)
|
|
goto bad;
|
|
ct.dev = ports->master;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
if (au_get_lr_value(sc, &ct, &lgain, &rgain))
|
|
goto bad;
|
|
} else {
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
goto bad;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
if (ports->isdual) {
|
|
if (ports->cur_port == -1)
|
|
ct.dev = ports->master;
|
|
else
|
|
ct.dev = ports->miport[ports->cur_port];
|
|
au_get_lr_value(sc, &ct, &lgain, &rgain);
|
|
} else {
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->misel[i] == ct.un.ord) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_get_lr_value(sc, &ct,
|
|
&lgain, &rgain))
|
|
goto usemaster;
|
|
else
|
|
break;
|
|
}
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
goto bad;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
lgain = rgain = n = 0;
|
|
for(i = 0; i < ports->nports; i++) {
|
|
if (ports->misel[i] & ct.un.mask) {
|
|
ct.dev = ports->miport[i];
|
|
if (ct.dev == -1 ||
|
|
au_get_lr_value(sc, &ct, &l, &r))
|
|
goto usemaster;
|
|
else {
|
|
lgain += l;
|
|
rgain += r;
|
|
n++;
|
|
}
|
|
}
|
|
}
|
|
if (n != 0) {
|
|
lgain /= n;
|
|
rgain /= n;
|
|
}
|
|
}
|
|
}
|
|
bad:
|
|
if (lgain == rgain) { /* handles lgain==rgain==0 */
|
|
*pgain = lgain;
|
|
*pbalance = AUDIO_MID_BALANCE;
|
|
} else if (lgain < rgain) {
|
|
*pgain = rgain;
|
|
/* balance should be > AUDIO_MID_BALANCE */
|
|
*pbalance = AUDIO_RIGHT_BALANCE -
|
|
(AUDIO_MID_BALANCE * lgain) / rgain;
|
|
} else /* lgain > rgain */ {
|
|
*pgain = lgain;
|
|
/* balance should be < AUDIO_MID_BALANCE */
|
|
*pbalance = (AUDIO_MID_BALANCE * rgain) / lgain;
|
|
}
|
|
}
|
|
|
|
int
|
|
au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port)
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, error, use_mixerout;
|
|
|
|
use_mixerout = 1;
|
|
if (port == 0) {
|
|
if (ports->allports == 0)
|
|
return 0; /* Allow this special case. */
|
|
else if (ports->isdual) {
|
|
if (ports->cur_port == -1) {
|
|
return 0;
|
|
} else {
|
|
port = ports->aumask[ports->cur_port];
|
|
ports->cur_port = -1;
|
|
use_mixerout = 0;
|
|
}
|
|
}
|
|
}
|
|
if (ports->index == -1)
|
|
return EINVAL;
|
|
ct.dev = ports->index;
|
|
if (ports->isenum) {
|
|
if (port & (port-1))
|
|
return EINVAL; /* Only one port allowed */
|
|
ct.type = AUDIO_MIXER_ENUM;
|
|
error = EINVAL;
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->aumask[i] == port) {
|
|
if (ports->isdual && use_mixerout) {
|
|
ct.un.ord = ports->mixerout;
|
|
ports->cur_port = i;
|
|
} else {
|
|
ct.un.ord = ports->misel[i];
|
|
}
|
|
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
|
|
break;
|
|
}
|
|
} else {
|
|
ct.type = AUDIO_MIXER_SET;
|
|
ct.un.mask = 0;
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->aumask[i] & port)
|
|
ct.un.mask |= ports->misel[i];
|
|
if (port != 0 && ct.un.mask == 0)
|
|
error = EINVAL;
|
|
else
|
|
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
|
|
}
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
return error;
|
|
}
|
|
|
|
int
|
|
au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports)
|
|
{
|
|
mixer_ctrl_t ct;
|
|
int i, aumask;
|
|
|
|
if (ports->index == -1)
|
|
return 0;
|
|
ct.dev = ports->index;
|
|
ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
return 0;
|
|
aumask = 0;
|
|
if (ports->isenum) {
|
|
if (ports->isdual && ports->cur_port != -1) {
|
|
if (ports->mixerout == ct.un.ord)
|
|
aumask = ports->aumask[ports->cur_port];
|
|
else
|
|
ports->cur_port = -1;
|
|
}
|
|
if (aumask == 0)
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ports->misel[i] == ct.un.ord)
|
|
aumask = ports->aumask[i];
|
|
} else {
|
|
for(i = 0; i < ports->nports; i++)
|
|
if (ct.un.mask & ports->misel[i])
|
|
aumask |= ports->aumask[i];
|
|
}
|
|
return aumask;
|
|
}
|
|
|
|
int
|
|
audiosetinfo(struct audio_softc *sc, struct audio_info *ai)
|
|
{
|
|
stream_filter_list_t pfilters, rfilters;
|
|
audio_params_t pp, rp;
|
|
struct audio_prinfo *r, *p;
|
|
const struct audio_hw_if *hw;
|
|
audio_stream_t *oldpus, *oldrus;
|
|
int s, setmode;
|
|
int error;
|
|
int np, nr;
|
|
unsigned int blks;
|
|
int oldpblksize, oldrblksize;
|
|
u_int gain;
|
|
boolean_t rbus, pbus;
|
|
boolean_t cleared, modechange, pausechange;
|
|
u_char balance;
|
|
|
|
hw = sc->hw_if;
|
|
if (hw == NULL) /* HW has not attached */
|
|
return ENXIO;
|
|
|
|
DPRINTF(("%s sc=%p ai=%p\n", __func__, sc, ai));
|
|
r = &ai->record;
|
|
p = &ai->play;
|
|
rbus = sc->sc_rbus;
|
|
pbus = sc->sc_pbus;
|
|
error = 0;
|
|
cleared = FALSE;
|
|
modechange = FALSE;
|
|
pausechange = FALSE;
|
|
|
|
pp = sc->sc_pparams; /* Temporary encoding storage in */
|
|
rp = sc->sc_rparams; /* case setting the modes fails. */
|
|
nr = np = 0;
|
|
|
|
if (SPECIFIED(p->sample_rate)) {
|
|
pp.sample_rate = p->sample_rate;
|
|
np++;
|
|
}
|
|
if (SPECIFIED(r->sample_rate)) {
|
|
rp.sample_rate = r->sample_rate;
|
|
nr++;
|
|
}
|
|
if (SPECIFIED(p->encoding)) {
|
|
pp.encoding = p->encoding;
|
|
np++;
|
|
}
|
|
if (SPECIFIED(r->encoding)) {
|
|
rp.encoding = r->encoding;
|
|
nr++;
|
|
}
|
|
if (SPECIFIED(p->precision)) {
|
|
pp.precision = p->precision;
|
|
/* we don't have API to specify validbits */
|
|
pp.validbits = p->precision;
|
|
np++;
|
|
}
|
|
if (SPECIFIED(r->precision)) {
|
|
rp.precision = r->precision;
|
|
/* we don't have API to specify validbits */
|
|
rp.validbits = r->precision;
|
|
nr++;
|
|
}
|
|
if (SPECIFIED(p->channels)) {
|
|
pp.channels = p->channels;
|
|
np++;
|
|
}
|
|
if (SPECIFIED(r->channels)) {
|
|
rp.channels = r->channels;
|
|
nr++;
|
|
}
|
|
#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(sc);
|
|
cleared = TRUE;
|
|
}
|
|
modechange = TRUE;
|
|
setmode |= AUMODE_RECORD;
|
|
}
|
|
if (np > 0) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
modechange = TRUE;
|
|
setmode |= AUMODE_PLAY;
|
|
}
|
|
|
|
if (SPECIFIED(ai->mode)) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
modechange = TRUE;
|
|
sc->sc_mode = ai->mode;
|
|
if (sc->sc_mode & AUMODE_PLAY_ALL)
|
|
sc->sc_mode |= AUMODE_PLAY;
|
|
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex)
|
|
/* Play takes precedence */
|
|
sc->sc_mode &= ~AUMODE_RECORD;
|
|
}
|
|
|
|
oldpus = sc->sc_pustream;
|
|
oldrus = sc->sc_rustream;
|
|
if (modechange) {
|
|
int indep;
|
|
|
|
indep = hw->get_props(sc->hw_hdl) & AUDIO_PROP_INDEPENDENT;
|
|
if (!indep) {
|
|
if (setmode == AUMODE_RECORD)
|
|
pp = rp;
|
|
else if (setmode == AUMODE_PLAY)
|
|
rp = pp;
|
|
}
|
|
memset(&pfilters, 0, sizeof(pfilters));
|
|
memset(&rfilters, 0, sizeof(rfilters));
|
|
pfilters.append = stream_filter_list_append;
|
|
pfilters.prepend = stream_filter_list_prepend;
|
|
pfilters.set = stream_filter_list_set;
|
|
rfilters.append = stream_filter_list_append;
|
|
rfilters.prepend = stream_filter_list_prepend;
|
|
rfilters.set = stream_filter_list_set;
|
|
/* Some device drivers change channels/sample_rate and change
|
|
* no channels/sample_rate. */
|
|
error = hw->set_params(sc->hw_hdl, setmode,
|
|
sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp,
|
|
&pfilters, &rfilters);
|
|
if (error) {
|
|
DPRINTF(("%s: hw->set_params() failed with %d\n",
|
|
__func__, error));
|
|
return error;
|
|
}
|
|
|
|
audio_check_params(&pp);
|
|
audio_check_params(&rp);
|
|
if (!indep) {
|
|
/* XXX for !indep device, we have to use the same
|
|
* parameters for the hardware, not userland */
|
|
if (setmode == AUMODE_RECORD) {
|
|
pp = rp;
|
|
} else if (setmode == AUMODE_PLAY) {
|
|
rp = pp;
|
|
}
|
|
}
|
|
|
|
if (sc->sc_pr.mmapped && pfilters.req_size > 0) {
|
|
DPRINTF(("%s: mmapped, and filters are requested.\n",
|
|
__func__));
|
|
return EINVAL;
|
|
}
|
|
|
|
/* construct new filter chain */
|
|
if (setmode & AUMODE_PLAY) {
|
|
error = audio_setup_pfilters(sc, &pp, &pfilters);
|
|
if (error)
|
|
return error;
|
|
}
|
|
if (setmode & AUMODE_RECORD) {
|
|
error = audio_setup_rfilters(sc, &rp, &rfilters);
|
|
if (error)
|
|
return error;
|
|
}
|
|
DPRINTF(("%s: filter setup is completed.\n", __func__));
|
|
|
|
/* userland formats */
|
|
sc->sc_pparams = pp;
|
|
sc->sc_rparams = rp;
|
|
}
|
|
|
|
oldpblksize = sc->sc_pr.blksize;
|
|
oldrblksize = sc->sc_rr.blksize;
|
|
/* Play params can affect the record params, so recalculate blksize. */
|
|
if (nr > 0 || np > 0) {
|
|
audio_calc_blksize(sc, AUMODE_RECORD);
|
|
audio_calc_blksize(sc, AUMODE_PLAY);
|
|
}
|
|
#ifdef AUDIO_DEBUG
|
|
if (audiodebug > 1 && nr > 0)
|
|
audio_print_params("audiosetinfo() After setting record params:", &sc->sc_rparams);
|
|
if (audiodebug > 1 && np > 0)
|
|
audio_print_params("audiosetinfo() After setting play params:", &sc->sc_pparams);
|
|
#endif
|
|
|
|
if (SPECIFIED(p->port)) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
error = au_set_port(sc, &sc->sc_outports, p->port);
|
|
if (error)
|
|
return error;
|
|
}
|
|
if (SPECIFIED(r->port)) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
error = au_set_port(sc, &sc->sc_inports, r->port);
|
|
if (error)
|
|
return error;
|
|
}
|
|
if (SPECIFIED(p->gain)) {
|
|
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_outports, p->gain, balance);
|
|
if (error)
|
|
return error;
|
|
}
|
|
if (SPECIFIED(r->gain)) {
|
|
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
|
|
error = au_set_gain(sc, &sc->sc_inports, r->gain, balance);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
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)
|
|
return error;
|
|
}
|
|
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)
|
|
return error;
|
|
}
|
|
|
|
if (SPECIFIED(ai->monitor_gain) && sc->sc_monitor_port != -1) {
|
|
mixer_ctrl_t ct;
|
|
|
|
ct.dev = sc->sc_monitor_port;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain;
|
|
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
if (SPECIFIED_CH(p->pause)) {
|
|
sc->sc_pr.pause = p->pause;
|
|
pbus = !p->pause;
|
|
pausechange = TRUE;
|
|
}
|
|
if (SPECIFIED_CH(r->pause)) {
|
|
sc->sc_rr.pause = r->pause;
|
|
rbus = !r->pause;
|
|
pausechange = TRUE;
|
|
}
|
|
|
|
if (SPECIFIED(ai->blocksize)) {
|
|
int pblksize, rblksize;
|
|
|
|
/* Block size specified explicitly. */
|
|
if (ai->blocksize == 0) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
sc->sc_blkset = FALSE;
|
|
audio_calc_blksize(sc, AUMODE_RECORD);
|
|
audio_calc_blksize(sc, AUMODE_PLAY);
|
|
} else {
|
|
sc->sc_blkset = TRUE;
|
|
/* check whether new blocksize changes actually */
|
|
if (hw->round_blocksize == NULL) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
sc->sc_pr.blksize = ai->blocksize;
|
|
sc->sc_rr.blksize = ai->blocksize;
|
|
} else {
|
|
pblksize = hw->round_blocksize(sc->hw_hdl,
|
|
ai->blocksize, AUMODE_PLAY, &sc->sc_pr.s.param);
|
|
rblksize = hw->round_blocksize(sc->hw_hdl,
|
|
ai->blocksize, AUMODE_RECORD, &sc->sc_rr.s.param);
|
|
if (pblksize != sc->sc_pr.blksize ||
|
|
rblksize != sc->sc_rr.blksize) {
|
|
if (!cleared) {
|
|
audio_clear(sc);
|
|
cleared = TRUE;
|
|
}
|
|
sc->sc_pr.blksize = ai->blocksize;
|
|
sc->sc_rr.blksize = ai->blocksize;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (SPECIFIED(ai->mode)) {
|
|
if (sc->sc_mode & AUMODE_PLAY)
|
|
audio_init_play(sc);
|
|
if (sc->sc_mode & AUMODE_RECORD)
|
|
audio_init_record(sc);
|
|
}
|
|
|
|
if (hw->commit_settings) {
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
if (cleared || pausechange) {
|
|
s = splaudio();
|
|
error = audio_initbufs(sc);
|
|
if (error) goto err;
|
|
if (sc->sc_pr.blksize != oldpblksize ||
|
|
sc->sc_rr.blksize != oldrblksize ||
|
|
sc->sc_pustream != oldpus ||
|
|
sc->sc_rustream != oldrus)
|
|
audio_calcwater(sc);
|
|
if ((sc->sc_mode & AUMODE_PLAY) &&
|
|
pbus && !sc->sc_pbus)
|
|
error = audiostartp(sc);
|
|
if (!error &&
|
|
(sc->sc_mode & AUMODE_RECORD) &&
|
|
rbus && !sc->sc_rbus)
|
|
error = audiostartr(sc);
|
|
err:
|
|
splx(s);
|
|
if (error)
|
|
return error;
|
|
}
|
|
|
|
/* Change water marks after initializing the buffers. */
|
|
if (SPECIFIED(ai->hiwat)) {
|
|
blks = ai->hiwat;
|
|
if (blks > sc->sc_pr.maxblks)
|
|
blks = sc->sc_pr.maxblks;
|
|
if (blks < 2)
|
|
blks = 2;
|
|
sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize;
|
|
}
|
|
if (SPECIFIED(ai->lowat)) {
|
|
blks = ai->lowat;
|
|
if (blks > sc->sc_pr.maxblks - 1)
|
|
blks = sc->sc_pr.maxblks - 1;
|
|
sc->sc_pr.usedlow = blks * sc->sc_pr.blksize;
|
|
}
|
|
if (SPECIFIED(ai->hiwat) || SPECIFIED(ai->lowat)) {
|
|
if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize)
|
|
sc->sc_pr.usedlow =
|
|
sc->sc_pr.usedhigh - sc->sc_pr.blksize;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
audiogetinfo(struct audio_softc *sc, struct audio_info *ai)
|
|
{
|
|
struct audio_prinfo *r, *p;
|
|
const struct audio_hw_if *hw;
|
|
|
|
r = &ai->record;
|
|
p = &ai->play;
|
|
hw = sc->hw_if;
|
|
if (hw == NULL) /* HW has not attached */
|
|
return ENXIO;
|
|
|
|
p->sample_rate = sc->sc_pparams.sample_rate;
|
|
r->sample_rate = sc->sc_rparams.sample_rate;
|
|
p->channels = sc->sc_pparams.channels;
|
|
r->channels = sc->sc_rparams.channels;
|
|
p->precision = sc->sc_pparams.precision;
|
|
r->precision = sc->sc_rparams.precision;
|
|
p->encoding = sc->sc_pparams.encoding;
|
|
r->encoding = sc->sc_rparams.encoding;
|
|
|
|
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) {
|
|
mixer_ctrl_t ct;
|
|
|
|
ct.dev = sc->sc_monitor_port;
|
|
ct.type = AUDIO_MIXER_VALUE;
|
|
ct.un.value.num_channels = 1;
|
|
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
|
|
ai->monitor_gain = 0;
|
|
else
|
|
ai->monitor_gain =
|
|
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
} else
|
|
ai->monitor_gain = 0;
|
|
|
|
p->seek = audio_stream_get_used(sc->sc_pustream);
|
|
r->seek = audio_stream_get_used(sc->sc_rustream);
|
|
|
|
/*
|
|
* XXX samples should be a value for userland data.
|
|
* But drops is a value for HW data.
|
|
*/
|
|
p->samples = (sc->sc_pustream == &sc->sc_pr.s
|
|
? sc->sc_pr.stamp : sc->sc_pr.fstamp) - sc->sc_pr.drops;
|
|
r->samples = (sc->sc_rustream == &sc->sc_rr.s
|
|
? sc->sc_rr.stamp : sc->sc_rr.fstamp) - sc->sc_rr.drops;
|
|
|
|
p->eof = sc->sc_eof;
|
|
r->eof = 0;
|
|
|
|
p->pause = sc->sc_pr.pause;
|
|
r->pause = sc->sc_rr.pause;
|
|
|
|
p->error = sc->sc_pr.drops != 0;
|
|
r->error = sc->sc_rr.drops != 0;
|
|
|
|
p->waiting = r->waiting = 0; /* open never hangs */
|
|
|
|
p->open = (sc->sc_open & AUOPEN_WRITE) != 0;
|
|
r->open = (sc->sc_open & AUOPEN_READ) != 0;
|
|
|
|
p->active = sc->sc_pbus;
|
|
r->active = sc->sc_rbus;
|
|
|
|
p->buffer_size = sc->sc_pustream->bufsize;
|
|
r->buffer_size = sc->sc_rustream->bufsize;
|
|
|
|
ai->blocksize = sc->sc_pr.blksize;
|
|
ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize;
|
|
ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize;
|
|
ai->mode = sc->sc_mode;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Mixer driver
|
|
*/
|
|
int
|
|
mixer_open(dev_t dev __unused, struct audio_softc *sc, int flags __unused,
|
|
int ifmt __unused, struct lwp *l __unused)
|
|
{
|
|
if (sc->hw_if == NULL)
|
|
return ENXIO;
|
|
|
|
DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc));
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Remove a process from those to be signalled on mixer activity.
|
|
*/
|
|
static void
|
|
mixer_remove(struct audio_softc *sc, struct lwp *l)
|
|
{
|
|
struct mixer_asyncs **pm, *m;
|
|
struct proc *p;
|
|
|
|
if (l == NULL)
|
|
return;
|
|
|
|
p = l->l_proc;
|
|
|
|
for (pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
|
|
if ((*pm)->proc == p) {
|
|
m = *pm;
|
|
*pm = m->next;
|
|
free(m, M_DEVBUF);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Signal all processes waiting for the mixer.
|
|
*/
|
|
static void
|
|
mixer_signal(struct audio_softc *sc)
|
|
{
|
|
struct mixer_asyncs *m;
|
|
|
|
for (m = sc->sc_async_mixer; m; m = m->next)
|
|
psignal(m->proc, SIGIO);
|
|
}
|
|
|
|
/*
|
|
* Close a mixer device
|
|
*/
|
|
/* ARGSUSED */
|
|
int
|
|
mixer_close(struct audio_softc *sc, int flags __unused, int ifmt __unused,
|
|
struct lwp *l)
|
|
{
|
|
|
|
DPRINTF(("mixer_close: sc %p\n", sc));
|
|
mixer_remove(sc, l);
|
|
return 0;
|
|
}
|
|
|
|
int
|
|
mixer_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag,
|
|
struct lwp *l)
|
|
{
|
|
const struct audio_hw_if *hw;
|
|
int error;
|
|
|
|
DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n",
|
|
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
|
|
hw = sc->hw_if;
|
|
error = EINVAL;
|
|
switch (cmd) {
|
|
case FIOASYNC:
|
|
mixer_remove(sc, l); /* remove old entry */
|
|
if (*(int *)addr) {
|
|
struct mixer_asyncs *ma;
|
|
ma = malloc(sizeof (struct mixer_asyncs),
|
|
M_DEVBUF, M_WAITOK);
|
|
ma->next = sc->sc_async_mixer;
|
|
ma->proc = l->l_proc;
|
|
sc->sc_async_mixer = ma;
|
|
}
|
|
error = 0;
|
|
break;
|
|
|
|
case AUDIO_GETDEV:
|
|
DPRINTF(("AUDIO_GETDEV\n"));
|
|
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_DEVINFO:
|
|
DPRINTF(("AUDIO_MIXER_DEVINFO\n"));
|
|
((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */
|
|
error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_READ:
|
|
DPRINTF(("AUDIO_MIXER_READ\n"));
|
|
error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
|
|
break;
|
|
|
|
case AUDIO_MIXER_WRITE:
|
|
DPRINTF(("AUDIO_MIXER_WRITE\n"));
|
|
error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
|
|
if (!error && hw->commit_settings)
|
|
error = hw->commit_settings(sc->hw_hdl);
|
|
if (!error)
|
|
mixer_signal(sc);
|
|
break;
|
|
|
|
default:
|
|
if (hw->dev_ioctl)
|
|
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, l);
|
|
else
|
|
error = EINVAL;
|
|
break;
|
|
}
|
|
DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n",
|
|
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
|
|
return error;
|
|
}
|
|
#endif /* NAUDIO > 0 */
|
|
|
|
#include "midi.h"
|
|
|
|
#if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0)
|
|
#include <sys/param.h>
|
|
#include <sys/systm.h>
|
|
#include <sys/device.h>
|
|
#include <sys/audioio.h>
|
|
#include <dev/audio_if.h>
|
|
#endif
|
|
|
|
#if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0)
|
|
int
|
|
audioprint(void *aux, const char *pnp)
|
|
{
|
|
struct audio_attach_args *arg;
|
|
const char *type;
|
|
|
|
if (pnp != NULL) {
|
|
arg = aux;
|
|
switch (arg->type) {
|
|
case AUDIODEV_TYPE_AUDIO:
|
|
type = "audio";
|
|
break;
|
|
case AUDIODEV_TYPE_MIDI:
|
|
type = "midi";
|
|
break;
|
|
case AUDIODEV_TYPE_OPL:
|
|
type = "opl";
|
|
break;
|
|
case AUDIODEV_TYPE_MPU:
|
|
type = "mpu";
|
|
break;
|
|
default:
|
|
panic("audioprint: unknown type %d", arg->type);
|
|
}
|
|
aprint_normal("%s at %s", type, pnp);
|
|
}
|
|
return UNCONF;
|
|
}
|
|
|
|
#endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */
|
|
|
|
#if NAUDIO > 0
|
|
void
|
|
audio_powerhook(int why, void *aux)
|
|
{
|
|
struct audio_softc *sc;
|
|
const struct audio_hw_if *hwp;
|
|
|
|
sc = (struct audio_softc *)aux;
|
|
hwp = sc->hw_if;
|
|
|
|
switch (why) {
|
|
case PWR_SOFTSUSPEND:
|
|
if (sc->sc_pbus == TRUE)
|
|
hwp->halt_output(sc->hw_hdl);
|
|
if (sc->sc_rbus == TRUE)
|
|
hwp->halt_input(sc->hw_hdl);
|
|
break;
|
|
case PWR_SOFTRESUME:
|
|
if (sc->sc_pbus == TRUE)
|
|
audiostartp(sc);
|
|
if (sc->sc_rbus == TRUE)
|
|
audiostartr(sc);
|
|
break;
|
|
}
|
|
|
|
return;
|
|
}
|
|
#endif /* NAUDIO > 0 */
|