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NetBSD/lib/libossaudio/ossaudio.c
nia f7f198978e ossaudio(3): Clone some useful behaviour from the Solaris kernel 
Both SNDCTL_DSP_SPEED and SNDCTL_DSP_CHANNELS support a special value
0 to "query the configured value without changing it". In our case,
this actually means setting and querying the hardware rate.

I don't know if OSSv3 or FreeBSD or other implementations also do this.
But it seems safe and sensible to support.
2020-11-13 09:02:39 +00:00

1791 lines
46 KiB
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/* $NetBSD: ossaudio.c,v 1.64 2020/11/13 09:02:39 nia Exp $ */
/*-
* Copyright (c) 1997, 2020 The NetBSD Foundation, Inc.
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
* POSSIBILITY OF SUCH DAMAGE.
*/
#include <sys/cdefs.h>
__RCSID("$NetBSD: ossaudio.c,v 1.64 2020/11/13 09:02:39 nia Exp $");
/*
* This is an Open Sound System compatibility layer, which provides
* fairly complete ioctl emulation for OSSv3 and some of OSSv4.
*
* The canonical OSS specification is available at
* http://manuals.opensound.com/developer/
*
* This file is similar to sys/compat/ossaudio.c with additional OSSv4
* compatibility.
*/
#include <string.h>
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#include <sys/stat.h>
#include <errno.h>
#include <fcntl.h>
#include <stdio.h>
#include <unistd.h>
#include <limits.h>
#include <stdarg.h>
#include <stdbool.h>
#include "soundcard.h"
#undef ioctl
#define GET_DEV(com) ((com) & 0xff)
#define TO_OSSVOL(x) (((x) * 100 + 127) / 255)
#define FROM_OSSVOL(x) ((((x) > 100 ? 100 : (x)) * 255 + 50) / 100)
#define GETPRINFO(info, name) \
(((info)->mode == AUMODE_RECORD) \
? (info)->record.name : (info)->play.name)
static struct audiodevinfo *getdevinfo(int);
static int getaudiocount(void);
static int getmixercount(void);
static int getmixercontrolcount(int);
static int getcaps(int, int *);
static int getvol(u_int, u_char);
static void setvol(int, int, bool);
static void setchannels(int, int, int);
static void setblocksize(int, struct audio_info *);
static int audio_ioctl(int, unsigned long, void *);
static int mixer_oss3_ioctl(int, unsigned long, void *);
static int mixer_oss4_ioctl(int, unsigned long, void *);
static int global_oss4_ioctl(int, unsigned long, void *);
static int opaque_to_enum(struct audiodevinfo *, audio_mixer_name_t *, int);
static int enum_to_ord(struct audiodevinfo *, int);
static int enum_to_mask(struct audiodevinfo *, int);
#define INTARG (*(int*)argp)
int
_oss_ioctl(int fd, unsigned long com, ...)
{
va_list ap;
void *argp;
va_start(ap, com);
argp = va_arg(ap, void *);
va_end(ap);
if (IOCGROUP(com) == 'P')
return audio_ioctl(fd, com, argp);
else if (IOCGROUP(com) == 'M')
return mixer_oss3_ioctl(fd, com, argp);
else if (IOCGROUP(com) == 'X')
return mixer_oss4_ioctl(fd, com, argp);
else if (IOCGROUP(com) == 'Y')
return global_oss4_ioctl(fd, com, argp);
else
return ioctl(fd, com, argp);
}
static int
audio_ioctl(int fd, unsigned long com, void *argp)
{
struct audio_info tmpinfo, hwfmt;
struct audio_offset tmpoffs;
struct audio_buf_info bufinfo;
struct audio_errinfo *tmperrinfo;
struct count_info cntinfo;
struct audio_encoding tmpenc;
u_int u;
u_int encoding;
u_int precision;
int perrors, rerrors;
static int totalperrors = 0;
static int totalrerrors = 0;
oss_count_t osscount;
int idat;
int retval;
idat = 0;
switch (com) {
case SNDCTL_DSP_RESET:
retval = ioctl(fd, AUDIO_FLUSH, 0);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_SYNC:
retval = ioctl(fd, AUDIO_DRAIN, 0);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_GETERROR:
tmperrinfo = (struct audio_errinfo *)argp;
if (tmperrinfo == NULL) {
errno = EINVAL;
return -1;
}
memset(tmperrinfo, 0, sizeof(struct audio_errinfo));
if ((retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo)) < 0)
return retval;
/*
* OSS requires that we return counters that are relative to
* the last call. We must maintain state here...
*/
if (ioctl(fd, AUDIO_PERROR, &perrors) != -1) {
perrors /= ((tmpinfo.play.precision / NBBY) *
tmpinfo.play.channels);
tmperrinfo->play_underruns =
(perrors / tmpinfo.blocksize) - totalperrors;
totalperrors += tmperrinfo->play_underruns;
}
if (ioctl(fd, AUDIO_RERROR, &rerrors) != -1) {
rerrors /= ((tmpinfo.record.precision / NBBY) *
tmpinfo.record.channels);
tmperrinfo->rec_overruns =
(rerrors / tmpinfo.blocksize) - totalrerrors;
totalrerrors += tmperrinfo->rec_overruns;
}
break;
case SNDCTL_DSP_COOKEDMODE:
/*
* NetBSD is always running in "cooked mode" - the kernel
* always performs format conversions.
*/
INTARG = 1;
break;
case SNDCTL_DSP_POST:
/* This call is merely advisory, and may be a nop. */
break;
case SNDCTL_DSP_SPEED:
AUDIO_INITINFO(&tmpinfo);
/*
* In Solaris, 0 is used a special value to query the
* current rate. This seems useful to support.
*/
if (INTARG == 0) {
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
retval = ioctl(fd, AUDIO_GETFORMAT, &hwfmt);
if (retval < 0)
return retval;
INTARG = (tmpinfo.mode == AUMODE_RECORD) ?
hwfmt.record.sample_rate :
hwfmt.play.sample_rate;
}
/*
* Conform to kernel limits.
* NetBSD will reject unsupported sample rates, but OSS
* applications need to be able to negotiate a supported one.
*/
if (INTARG < 1000)
INTARG = 1000;
if (INTARG > 192000)
INTARG = 192000;
tmpinfo.play.sample_rate =
tmpinfo.record.sample_rate = INTARG;
retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo);
if (retval < 0)
return retval;
/* FALLTHRU */
case SOUND_PCM_READ_RATE:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = GETPRINFO(&tmpinfo, sample_rate);
break;
case SNDCTL_DSP_STEREO:
AUDIO_INITINFO(&tmpinfo);
tmpinfo.play.channels =
tmpinfo.record.channels = INTARG ? 2 : 1;
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = GETPRINFO(&tmpinfo, channels) - 1;
break;
case SNDCTL_DSP_GETBLKSIZE:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
setblocksize(fd, &tmpinfo);
INTARG = tmpinfo.blocksize;
break;
case SNDCTL_DSP_SETFMT:
AUDIO_INITINFO(&tmpinfo);
switch (INTARG) {
case AFMT_MU_LAW:
tmpinfo.play.precision =
tmpinfo.record.precision = 8;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_ULAW;
break;
case AFMT_A_LAW:
tmpinfo.play.precision =
tmpinfo.record.precision = 8;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_ALAW;
break;
case AFMT_U8:
tmpinfo.play.precision =
tmpinfo.record.precision = 8;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR;
break;
case AFMT_S8:
tmpinfo.play.precision =
tmpinfo.record.precision = 8;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR;
break;
case AFMT_S16_LE:
tmpinfo.play.precision =
tmpinfo.record.precision = 16;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE;
break;
case AFMT_S16_BE:
tmpinfo.play.precision =
tmpinfo.record.precision = 16;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
case AFMT_U16_LE:
tmpinfo.play.precision =
tmpinfo.record.precision = 16;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_LE;
break;
case AFMT_U16_BE:
tmpinfo.play.precision =
tmpinfo.record.precision = 16;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_BE;
break;
/*
* XXX: When the kernel supports 24-bit LPCM by default,
* the 24-bit formats should be handled properly instead
* of falling back to 32 bits.
*/
case AFMT_S24_LE:
case AFMT_S32_LE:
tmpinfo.play.precision =
tmpinfo.record.precision = 32;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE;
break;
case AFMT_S24_BE:
case AFMT_S32_BE:
tmpinfo.play.precision =
tmpinfo.record.precision = 32;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
case AFMT_AC3:
tmpinfo.play.precision =
tmpinfo.record.precision = 16;
tmpinfo.play.encoding =
tmpinfo.record.encoding = AUDIO_ENCODING_AC3;
break;
default:
/*
* OSSv4 specifies that if an invalid format is chosen
* by an application then a sensible format supported
* by the hardware is returned.
*
* In this case, we pick the current hardware format.
*/
retval = ioctl(fd, AUDIO_GETFORMAT, &hwfmt);
if (retval < 0)
return retval;
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
tmpinfo.play.encoding =
tmpinfo.record.encoding =
(tmpinfo.mode == AUMODE_RECORD) ?
hwfmt.record.encoding : hwfmt.play.encoding;
tmpinfo.play.precision =
tmpinfo.record.precision =
(tmpinfo.mode == AUMODE_RECORD) ?
hwfmt.record.precision : hwfmt.play.precision ;
break;
}
/*
* In the post-kernel-mixer world, assume that any error means
* it's fatal rather than an unsupported format being selected.
*/
retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo);
if (retval < 0)
return retval;
/* FALLTHRU */
case SOUND_PCM_READ_BITS:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
encoding = GETPRINFO(&tmpinfo, encoding);
precision = GETPRINFO(&tmpinfo, precision);
switch (encoding) {
case AUDIO_ENCODING_ULAW:
idat = AFMT_MU_LAW;
break;
case AUDIO_ENCODING_ALAW:
idat = AFMT_A_LAW;
break;
case AUDIO_ENCODING_SLINEAR_LE:
if (precision == 32)
idat = AFMT_S32_LE;
else if (precision == 24)
idat = AFMT_S24_LE;
else if (precision == 16)
idat = AFMT_S16_LE;
else
idat = AFMT_S8;
break;
case AUDIO_ENCODING_SLINEAR_BE:
if (precision == 32)
idat = AFMT_S32_BE;
else if (precision == 24)
idat = AFMT_S24_BE;
else if (precision == 16)
idat = AFMT_S16_BE;
else
idat = AFMT_S8;
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (precision == 16)
idat = AFMT_U16_LE;
else
idat = AFMT_U8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (precision == 16)
idat = AFMT_U16_BE;
else
idat = AFMT_U8;
break;
case AUDIO_ENCODING_ADPCM:
idat = AFMT_IMA_ADPCM;
break;
case AUDIO_ENCODING_AC3:
idat = AFMT_AC3;
break;
}
INTARG = idat;
break;
case SNDCTL_DSP_CHANNELS:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
setchannels(fd, tmpinfo.mode, INTARG);
/* FALLTHRU */
case SOUND_PCM_READ_CHANNELS:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = GETPRINFO(&tmpinfo, channels);
break;
case SOUND_PCM_WRITE_FILTER:
case SOUND_PCM_READ_FILTER:
errno = EINVAL;
return -1; /* XXX unimplemented */
case SNDCTL_DSP_SUBDIVIDE:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
setblocksize(fd, &tmpinfo);
idat = INTARG;
if (idat == 0)
idat = tmpinfo.play.buffer_size / tmpinfo.blocksize;
idat = (tmpinfo.play.buffer_size / idat) & -4;
AUDIO_INITINFO(&tmpinfo);
tmpinfo.blocksize = idat;
retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = tmpinfo.play.buffer_size / tmpinfo.blocksize;
break;
case SNDCTL_DSP_SETFRAGMENT:
AUDIO_INITINFO(&tmpinfo);
idat = INTARG;
if ((idat & 0xffff) < 4 || (idat & 0xffff) > 17) {
errno = EINVAL;
return -1;
}
tmpinfo.blocksize = 1 << (idat & 0xffff);
tmpinfo.hiwat = ((unsigned)idat >> 16) & 0x7fff;
if (tmpinfo.hiwat == 0) /* 0 means set to max */
tmpinfo.hiwat = 65536;
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
u = tmpinfo.blocksize;
for(idat = 0; u > 1; idat++, u >>= 1)
;
idat |= (tmpinfo.hiwat & 0x7fff) << 16;
INTARG = idat;
break;
case SNDCTL_DSP_GETFMTS:
for(idat = 0, tmpenc.index = 0;
ioctl(fd, AUDIO_GETENC, &tmpenc) == 0;
tmpenc.index++) {
switch(tmpenc.encoding) {
case AUDIO_ENCODING_ULAW:
idat |= AFMT_MU_LAW;
break;
case AUDIO_ENCODING_ALAW:
idat |= AFMT_A_LAW;
break;
case AUDIO_ENCODING_SLINEAR:
idat |= AFMT_S8;
break;
case AUDIO_ENCODING_SLINEAR_LE:
if (tmpenc.precision == 32)
idat |= AFMT_S32_LE;
else if (tmpenc.precision == 24)
idat |= AFMT_S24_LE;
else if (tmpenc.precision == 16)
idat |= AFMT_S16_LE;
else
idat |= AFMT_S8;
break;
case AUDIO_ENCODING_SLINEAR_BE:
if (tmpenc.precision == 32)
idat |= AFMT_S32_BE;
else if (tmpenc.precision == 24)
idat |= AFMT_S24_BE;
else if (tmpenc.precision == 16)
idat |= AFMT_S16_BE;
else
idat |= AFMT_S8;
break;
case AUDIO_ENCODING_ULINEAR:
idat |= AFMT_U8;
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (tmpenc.precision == 16)
idat |= AFMT_U16_LE;
else
idat |= AFMT_U8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (tmpenc.precision == 16)
idat |= AFMT_U16_BE;
else
idat |= AFMT_U8;
break;
case AUDIO_ENCODING_ADPCM:
idat |= AFMT_IMA_ADPCM;
break;
case AUDIO_ENCODING_AC3:
idat |= AFMT_AC3;
break;
default:
break;
}
}
INTARG = idat;
break;
case SNDCTL_DSP_GETOSPACE:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
setblocksize(fd, &tmpinfo);
bufinfo.fragsize = tmpinfo.blocksize;
bufinfo.fragments = tmpinfo.hiwat - (tmpinfo.play.seek
+ tmpinfo.blocksize - 1) / tmpinfo.blocksize;
bufinfo.fragstotal = tmpinfo.hiwat;
bufinfo.bytes = tmpinfo.hiwat * tmpinfo.blocksize
- tmpinfo.play.seek;
*(struct audio_buf_info *)argp = bufinfo;
break;
case SNDCTL_DSP_GETISPACE:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
setblocksize(fd, &tmpinfo);
bufinfo.fragsize = tmpinfo.blocksize;
bufinfo.fragments = tmpinfo.record.seek / tmpinfo.blocksize;
bufinfo.fragstotal =
tmpinfo.record.buffer_size / tmpinfo.blocksize;
bufinfo.bytes = tmpinfo.record.seek;
*(struct audio_buf_info *)argp = bufinfo;
break;
case SNDCTL_DSP_NONBLOCK:
idat = 1;
retval = ioctl(fd, FIONBIO, &idat);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_GETCAPS:
retval = getcaps(fd, (int *)argp);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_SETTRIGGER:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
AUDIO_INITINFO(&tmpinfo);
if (tmpinfo.mode & AUMODE_PLAY)
tmpinfo.play.pause = (INTARG & PCM_ENABLE_OUTPUT) == 0;
if (tmpinfo.mode & AUMODE_RECORD)
tmpinfo.record.pause = (INTARG & PCM_ENABLE_INPUT) == 0;
(void)ioctl(fd, AUDIO_SETINFO, &tmpinfo);
/* FALLTHRU */
case SNDCTL_DSP_GETTRIGGER:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
idat = 0;
if ((tmpinfo.mode & AUMODE_PLAY) && !tmpinfo.play.pause)
idat |= PCM_ENABLE_OUTPUT;
if ((tmpinfo.mode & AUMODE_RECORD) && !tmpinfo.record.pause)
idat |= PCM_ENABLE_INPUT;
INTARG = idat;
break;
case SNDCTL_DSP_GETIPTR:
retval = ioctl(fd, AUDIO_GETIOFFS, &tmpoffs);
if (retval < 0)
return retval;
cntinfo.bytes = tmpoffs.samples;
cntinfo.blocks = tmpoffs.deltablks;
cntinfo.ptr = tmpoffs.offset;
*(struct count_info *)argp = cntinfo;
break;
case SNDCTL_DSP_CURRENT_IPTR:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
/* XXX: 'samples' may wrap */
memset(osscount.filler, 0, sizeof(osscount.filler));
osscount.samples = tmpinfo.record.samples /
((tmpinfo.record.precision / NBBY) *
tmpinfo.record.channels);
osscount.fifo_samples = tmpinfo.record.seek /
((tmpinfo.record.precision / NBBY) *
tmpinfo.record.channels);
*(oss_count_t *)argp = osscount;
break;
case SNDCTL_DSP_GETOPTR:
retval = ioctl(fd, AUDIO_GETOOFFS, &tmpoffs);
if (retval < 0)
return retval;
cntinfo.bytes = tmpoffs.samples;
cntinfo.blocks = tmpoffs.deltablks;
cntinfo.ptr = tmpoffs.offset;
*(struct count_info *)argp = cntinfo;
break;
case SNDCTL_DSP_CURRENT_OPTR:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
/* XXX: 'samples' may wrap */
memset(osscount.filler, 0, sizeof(osscount.filler));
osscount.samples = tmpinfo.play.samples /
((tmpinfo.play.precision / NBBY) *
tmpinfo.play.channels);
osscount.fifo_samples = tmpinfo.play.seek /
((tmpinfo.play.precision / NBBY) *
tmpinfo.play.channels);
*(oss_count_t *)argp = osscount;
break;
case SNDCTL_DSP_SETPLAYVOL:
setvol(fd, INTARG, false);
/* FALLTHRU */
case SNDCTL_DSP_GETPLAYVOL:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = getvol(tmpinfo.play.gain, tmpinfo.play.balance);
break;
case SNDCTL_DSP_SETRECVOL:
setvol(fd, INTARG, true);
/* FALLTHRU */
case SNDCTL_DSP_GETRECVOL:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = getvol(tmpinfo.record.gain, tmpinfo.record.balance);
break;
case SNDCTL_DSP_SKIP:
case SNDCTL_DSP_SILENCE:
errno = EINVAL;
return -1;
case SNDCTL_DSP_SETDUPLEX:
idat = 1;
retval = ioctl(fd, AUDIO_SETFD, &idat);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_GETODELAY:
retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo);
if (retval < 0)
return retval;
idat = tmpinfo.play.seek + tmpinfo.blocksize / 2;
INTARG = idat;
break;
case SNDCTL_DSP_PROFILE:
/* This gives just a hint to the driver,
* implementing it as a NOP is ok
*/
break;
case SNDCTL_DSP_MAPINBUF:
case SNDCTL_DSP_MAPOUTBUF:
case SNDCTL_DSP_SETSYNCRO:
errno = EINVAL;
return -1; /* XXX unimplemented */
default:
errno = EINVAL;
return -1;
}
return 0;
}
/* If the NetBSD mixer device should have more than NETBSD_MAXDEVS devices
* some will not be available to OSS applications */
#define NETBSD_MAXDEVS 64
struct audiodevinfo {
int done;
dev_t dev;
int16_t devmap[SOUND_MIXER_NRDEVICES],
rdevmap[NETBSD_MAXDEVS];
char names[NETBSD_MAXDEVS][MAX_AUDIO_DEV_LEN];
int enum2opaque[NETBSD_MAXDEVS];
u_long devmask, recmask, stereomask;
u_long caps;
int source;
};
static int
opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq)
{
int i, o;
for (i = 0; i < NETBSD_MAXDEVS; i++) {
o = di->enum2opaque[i];
if (o == opq)
break;
if (o == -1 && label != NULL &&
!strncmp(di->names[i], label->name, sizeof di->names[i])) {
di->enum2opaque[i] = opq;
break;
}
}
if (i >= NETBSD_MAXDEVS)
i = -1;
/*printf("opq_to_enum %s %d -> %d\n", label->name, opq, i);*/
return (i);
}
static int
enum_to_ord(struct audiodevinfo *di, int enm)
{
if (enm >= NETBSD_MAXDEVS)
return (-1);
/*printf("enum_to_ord %d -> %d\n", enm, di->enum2opaque[enm]);*/
return (di->enum2opaque[enm]);
}
static int
enum_to_mask(struct audiodevinfo *di, int enm)
{
int m;
if (enm >= NETBSD_MAXDEVS)
return (0);
m = di->enum2opaque[enm];
if (m == -1)
m = 0;
/*printf("enum_to_mask %d -> %d\n", enm, di->enum2opaque[enm]);*/
return (m);
}
/*
* Collect the audio device information to allow faster
* emulation of the OSSv3 mixer ioctls. Cache the information
* to eliminate the overhead of repeating all the ioctls needed
* to collect the information.
*/
static struct audiodevinfo *
getdevinfo(int fd)
{
mixer_devinfo_t mi;
int i, j, e;
static struct {
const char *name;
int code;
} *dp, devs[] = {
{ AudioNmicrophone, SOUND_MIXER_MIC },
{ AudioNline, SOUND_MIXER_LINE },
{ AudioNcd, SOUND_MIXER_CD },
{ AudioNdac, SOUND_MIXER_PCM },
{ AudioNaux, SOUND_MIXER_LINE1 },
{ AudioNrecord, SOUND_MIXER_IMIX },
{ AudioNmaster, SOUND_MIXER_VOLUME },
{ AudioNtreble, SOUND_MIXER_TREBLE },
{ AudioNbass, SOUND_MIXER_BASS },
{ AudioNspeaker, SOUND_MIXER_SPEAKER },
/* { AudioNheadphone, ?? },*/
{ AudioNoutput, SOUND_MIXER_OGAIN },
{ AudioNinput, SOUND_MIXER_IGAIN },
/* { AudioNmaster, SOUND_MIXER_SPEAKER },*/
/* { AudioNstereo, ?? },*/
/* { AudioNmono, ?? },*/
{ AudioNfmsynth, SOUND_MIXER_SYNTH },
/* { AudioNwave, SOUND_MIXER_PCM },*/
{ AudioNmidi, SOUND_MIXER_SYNTH },
/* { AudioNmixerout, ?? },*/
{ 0, -1 }
};
static struct audiodevinfo devcache = { .done = 0 };
struct audiodevinfo *di = &devcache;
struct stat sb;
size_t mlen, dlen;
/* Figure out what device it is so we can check if the
* cached data is valid.
*/
if (fstat(fd, &sb) < 0)
return 0;
if (di->done && di->dev == sb.st_dev)
return di;
di->done = 1;
di->dev = sb.st_dev;
di->devmask = 0;
di->recmask = 0;
di->stereomask = 0;
di->source = ~0;
di->caps = 0;
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
di->devmap[i] = -1;
for(i = 0; i < NETBSD_MAXDEVS; i++) {
di->rdevmap[i] = -1;
di->names[i][0] = '\0';
di->enum2opaque[i] = -1;
}
for(i = 0; i < NETBSD_MAXDEVS; i++) {
mi.index = i;
if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
break;
switch(mi.type) {
case AUDIO_MIXER_VALUE:
for(dp = devs; dp->name; dp++) {
if (strcmp(dp->name, mi.label.name) == 0)
break;
dlen = strlen(dp->name);
mlen = strlen(mi.label.name);
if (dlen < mlen
&& mi.label.name[mlen-dlen-1] == '.'
&& strcmp(dp->name,
mi.label.name + mlen - dlen) == 0)
break;
}
if (dp->code >= 0) {
di->devmap[dp->code] = i;
di->rdevmap[i] = dp->code;
di->devmask |= 1 << dp->code;
if (mi.un.v.num_channels == 2)
di->stereomask |= 1 << dp->code;
strlcpy(di->names[i], mi.label.name,
sizeof di->names[i]);
}
break;
}
}
for(i = 0; i < NETBSD_MAXDEVS; i++) {
mi.index = i;
if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
break;
if (strcmp(mi.label.name, AudioNsource) != 0)
continue;
di->source = i;
switch(mi.type) {
case AUDIO_MIXER_ENUM:
for(j = 0; j < mi.un.e.num_mem; j++) {
e = opaque_to_enum(di,
&mi.un.e.member[j].label,
mi.un.e.member[j].ord);
if (e >= 0)
di->recmask |= 1 << di->rdevmap[e];
}
di->caps = SOUND_CAP_EXCL_INPUT;
break;
case AUDIO_MIXER_SET:
for(j = 0; j < mi.un.s.num_mem; j++) {
e = opaque_to_enum(di,
&mi.un.s.member[j].label,
mi.un.s.member[j].mask);
if (e >= 0)
di->recmask |= 1 << di->rdevmap[e];
}
break;
}
}
return di;
}
static int
mixer_oss3_ioctl(int fd, unsigned long com, void *argp)
{
struct audiodevinfo *di;
struct mixer_info *omi;
struct audio_device adev;
mixer_ctrl_t mc;
u_long idat, n;
int i;
int retval;
int l, r, error, e;
idat = 0;
di = getdevinfo(fd);
if (di == 0)
return -1;
switch (com) {
case OSS_GETVERSION:
idat = SOUND_VERSION;
break;
case SOUND_MIXER_INFO:
case SOUND_OLD_MIXER_INFO:
error = ioctl(fd, AUDIO_GETDEV, &adev);
if (error)
return (error);
omi = argp;
if (com == SOUND_MIXER_INFO)
omi->modify_counter = 1;
strlcpy(omi->id, adev.name, sizeof omi->id);
strlcpy(omi->name, adev.name, sizeof omi->name);
return 0;
case SOUND_MIXER_READ_RECSRC:
if (di->source == -1) {
errno = EINVAL;
return -1;
}
mc.dev = di->source;
if (di->caps & SOUND_CAP_EXCL_INPUT) {
mc.type = AUDIO_MIXER_ENUM;
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
if (retval < 0)
return retval;
e = opaque_to_enum(di, NULL, mc.un.ord);
if (e >= 0)
idat = 1 << di->rdevmap[e];
} else {
mc.type = AUDIO_MIXER_SET;
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
if (retval < 0)
return retval;
e = opaque_to_enum(di, NULL, mc.un.mask);
if (e >= 0)
idat = 1 << di->rdevmap[e];
}
break;
case SOUND_MIXER_READ_DEVMASK:
idat = di->devmask;
break;
case SOUND_MIXER_READ_RECMASK:
idat = di->recmask;
break;
case SOUND_MIXER_READ_STEREODEVS:
idat = di->stereomask;
break;
case SOUND_MIXER_READ_CAPS:
idat = di->caps;
break;
case SOUND_MIXER_WRITE_RECSRC:
case SOUND_MIXER_WRITE_R_RECSRC:
if (di->source == -1) {
errno = EINVAL;
return -1;
}
mc.dev = di->source;
idat = INTARG;
if (di->caps & SOUND_CAP_EXCL_INPUT) {
mc.type = AUDIO_MIXER_ENUM;
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
if (idat & (1 << i))
break;
if (i >= SOUND_MIXER_NRDEVICES ||
di->devmap[i] == -1) {
errno = EINVAL;
return -1;
}
mc.un.ord = enum_to_ord(di, di->devmap[i]);
} else {
mc.type = AUDIO_MIXER_SET;
mc.un.mask = 0;
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
if (idat & (1 << i)) {
if (di->devmap[i] == -1) {
errno = EINVAL;
return -1;
}
mc.un.mask |=
enum_to_mask(di, di->devmap[i]);
}
}
}
return ioctl(fd, AUDIO_MIXER_WRITE, &mc);
default:
if (MIXER_READ(SOUND_MIXER_FIRST) <= com &&
com < MIXER_READ(SOUND_MIXER_NRDEVICES)) {
n = GET_DEV(com);
if (di->devmap[n] == -1) {
errno = EINVAL;
return -1;
}
mc.dev = di->devmap[n];
mc.type = AUDIO_MIXER_VALUE;
doread:
mc.un.value.num_channels =
di->stereomask & (1 << (u_int)n) ? 2 : 1;
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
if (retval < 0)
return retval;
if (mc.type != AUDIO_MIXER_VALUE) {
errno = EINVAL;
return -1;
}
if (mc.un.value.num_channels != 2) {
l = r =
mc.un.value.level[AUDIO_MIXER_LEVEL_MONO];
} else {
l = mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT];
r = mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
}
idat = TO_OSSVOL(l) | (TO_OSSVOL(r) << 8);
break;
} else if ((MIXER_WRITE_R(SOUND_MIXER_FIRST) <= com &&
com < MIXER_WRITE_R(SOUND_MIXER_NRDEVICES)) ||
(MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))) {
n = GET_DEV(com);
if (di->devmap[n] == -1) {
errno = EINVAL;
return -1;
}
idat = INTARG;
l = FROM_OSSVOL((u_int)idat & 0xff);
r = FROM_OSSVOL(((u_int)idat >> 8) & 0xff);
mc.dev = di->devmap[n];
mc.type = AUDIO_MIXER_VALUE;
if (di->stereomask & (1 << (u_int)n)) {
mc.un.value.num_channels = 2;
mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
} else {
mc.un.value.num_channels = 1;
mc.un.value.level[AUDIO_MIXER_LEVEL_MONO] =
(l + r) / 2;
}
retval = ioctl(fd, AUDIO_MIXER_WRITE, &mc);
if (retval < 0)
return retval;
if (MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))
return 0;
goto doread;
} else {
errno = EINVAL;
return -1;
}
}
INTARG = (int)idat;
return 0;
}
static int
mixer_oss4_ioctl(int fd, unsigned long com, void *argp)
{
oss_audioinfo *tmpai;
oss_card_info *cardinfo;
oss_mixext *ext;
oss_mixext_root root;
oss_mixer_enuminfo *ei;
oss_mixer_value *mv;
oss_mixerinfo *mi;
oss_sysinfo sysinfo;
dev_t devno;
struct stat tmpstat;
struct audio_device dev;
struct audio_format_query fmtq;
struct mixer_devinfo mdi;
struct mixer_ctrl mc;
char devname[32];
size_t len;
int newfd = -1, tmperrno;
int i, noffs;
int retval;
/*
* Note: it is difficult to translate the NetBSD concept of a "set"
* mixer control type to the OSSv4 API, as far as I can tell.
*
* This means they are treated like enums, i.e. only one entry in the
* set can be selected at a time.
*/
switch (com) {
case SNDCTL_AUDIOINFO:
/*
* SNDCTL_AUDIOINFO_EX is intended for underlying hardware devices
* that are to be opened in "exclusive mode" (bypassing the normal
* kernel mixer for exclusive control). NetBSD does not support
* bypassing the kernel mixer, so it's an alias of SNDCTL_AUDIOINFO.
*/
case SNDCTL_AUDIOINFO_EX:
case SNDCTL_ENGINEINFO:
devno = 0;
tmpai = (struct oss_audioinfo*)argp;
if (tmpai == NULL) {
errno = EINVAL;
return -1;
}
/*
* If the input device is -1, guess the device related to
* the open mixer device.
*/
if (tmpai->dev < 0) {
fstat(fd, &tmpstat);
if ((tmpstat.st_rdev & 0xff00) == 0x2a00)
devno = tmpstat.st_rdev & 0xff;
if (devno >= 0x80)
tmpai->dev = devno & 0x7f;
}
if (tmpai->dev < 0)
tmpai->dev = 0;
snprintf(tmpai->devnode, sizeof(tmpai->devnode),
"/dev/audio%d", tmpai->dev);
if ((newfd = open(tmpai->devnode, O_WRONLY)) < 0) {
if ((newfd = open(tmpai->devnode, O_RDONLY)) < 0) {
return newfd;
}
}
retval = ioctl(newfd, AUDIO_GETDEV, &dev);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
if (getcaps(newfd, &tmpai->caps) < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
snprintf(tmpai->name, sizeof(tmpai->name),
"%s %s", dev.name, dev.version);
tmpai->busy = 0;
tmpai->pid = -1;
ioctl(newfd, SNDCTL_DSP_GETFMTS, &tmpai->iformats);
tmpai->oformats = tmpai->iformats;
tmpai->magic = -1; /* reserved for "internal use" */
memset(tmpai->cmd, 0, sizeof(tmpai->cmd));
tmpai->card_number = -1;
memset(tmpai->song_name, 0,
sizeof(tmpai->song_name));
memset(tmpai->label, 0, sizeof(tmpai->label));
tmpai->port_number = 0;
tmpai->mixer_dev = tmpai->dev;
tmpai->legacy_device = tmpai->dev;
tmpai->enabled = 1;
tmpai->flags = -1; /* reserved for "future versions" */
tmpai->min_rate = 1000;
tmpai->max_rate = 192000;
tmpai->nrates = 0;
tmpai->min_channels = 1;
tmpai->max_channels = 2;
for (fmtq.index = 0;
ioctl(newfd, AUDIO_QUERYFORMAT, &fmtq) != -1; ++fmtq.index) {
if (fmtq.fmt.channels > (unsigned)tmpai->max_channels)
tmpai->max_channels = fmtq.fmt.channels;
}
tmpai->binding = -1; /* reserved for "future versions" */
tmpai->rate_source = -1;
/*
* 'handle' is supposed to be globally unique. The closest
* we have to that is probably device nodes.
*/
strlcpy(tmpai->handle, tmpai->devnode,
sizeof(tmpai->handle));
tmpai->next_play_engine = 0;
tmpai->next_rec_engine = 0;
argp = tmpai;
close(newfd);
break;
case SNDCTL_CARDINFO:
cardinfo = (oss_card_info *)argp;
if (cardinfo == NULL) {
errno = EINVAL;
return -1;
}
if (cardinfo->card != -1) {
snprintf(devname, sizeof(devname),
"/dev/audio%d", cardinfo->card);
newfd = open(devname, O_RDONLY);
if (newfd < 0)
return newfd;
} else {
newfd = fd;
}
retval = ioctl(newfd, AUDIO_GETDEV, &dev);
tmperrno = errno;
if (newfd != fd)
close(newfd);
if (retval < 0) {
errno = tmperrno;
return retval;
}
strlcpy(cardinfo->shortname, dev.name,
sizeof(cardinfo->shortname));
snprintf(cardinfo->longname, sizeof(cardinfo->longname),
"%s %s %s", dev.name, dev.version, dev.config);
memset(cardinfo->hw_info, 0, sizeof(cardinfo->hw_info));
/*
* OSSv4 does not document this ioctl, and claims it should
* not be used by applications and is provided for "utiltiy
* programs included in OSS". We follow the Solaris
* implementation (which is documented) and leave these fields
* unset.
*/
cardinfo->flags = 0;
cardinfo->intr_count = 0;
cardinfo->ack_count = 0;
break;
case SNDCTL_SYSINFO:
memset(&sysinfo, 0, sizeof(sysinfo));
strlcpy(sysinfo.product,
"OSS/NetBSD", sizeof(sysinfo.product));
strlcpy(sysinfo.version,
"4.01", sizeof(sysinfo.version));
strlcpy(sysinfo.license,
"BSD", sizeof(sysinfo.license));
sysinfo.versionnum = SOUND_VERSION;
sysinfo.numaudios =
sysinfo.numcards =
getaudiocount();
sysinfo.numaudioengines = 1;
sysinfo.numsynths = 1;
sysinfo.nummidis = -1;
sysinfo.numtimers = -1;
sysinfo.nummixers = getmixercount();
*(struct oss_sysinfo *)argp = sysinfo;
break;
case SNDCTL_MIXERINFO:
mi = (oss_mixerinfo *)argp;
if (mi == NULL) {
errno = EINVAL;
return -1;
}
snprintf(devname, sizeof(devname), "/dev/mixer%d", mi->dev);
if ((newfd = open(devname, O_RDONLY)) < 0)
return newfd;
retval = ioctl(newfd, AUDIO_GETDEV, &dev);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
strlcpy(mi->id, devname, sizeof(mi->id));
snprintf(mi->name, sizeof(mi->name),
"%s %s", dev.name, dev.version);
mi->card_number = mi->dev;
mi->port_number = 0;
mi->magic = 0;
mi->enabled = 1;
mi->caps = 0;
mi->flags = 0;
mi->nrext = getmixercontrolcount(newfd) + 1;
mi->priority = UCHAR_MAX - mi->dev;
strlcpy(mi->devnode, devname, sizeof(mi->devnode));
mi->legacy_device = mi->dev;
break;
case SNDCTL_MIX_DESCRIPTION:
/* No description available. */
errno = ENOSYS;
return -1;
case SNDCTL_MIX_NRMIX:
INTARG = getmixercount();
break;
case SNDCTL_MIX_NREXT:
snprintf(devname, sizeof(devname), "/dev/mixer%d", INTARG);
if ((newfd = open(devname, O_RDONLY)) < 0)
return newfd;
INTARG = getmixercontrolcount(newfd) + 1;
close(newfd);
break;
case SNDCTL_MIX_EXTINFO:
ext = (oss_mixext *)argp;
snprintf(devname, sizeof(devname), "/dev/mixer%d", ext->dev);
if ((newfd = open(devname, O_RDONLY)) < 0)
return newfd;
if (ext->ctrl == 0) {
/*
* NetBSD has no concept of a "root mixer control", but
* OSSv4 requires one to work. We fake one at 0 and
* simply add 1 to all real control indexes.
*/
retval = ioctl(newfd, AUDIO_GETDEV, &dev);
tmperrno = errno;
close(newfd);
if (retval < 0) {
errno = tmperrno;
return -1;
}
memset(&root, 0, sizeof(root));
strlcpy(root.id, devname, sizeof(root.id));
snprintf(root.name, sizeof(root.name),
"%s %s", dev.name, dev.version);
strlcpy(ext->id, devname, sizeof(ext->id));
snprintf(ext->extname, sizeof(ext->extname),
"%s %s", dev.name, dev.version);
strlcpy(ext->extname, "root", sizeof(ext->extname));
ext->type = MIXT_DEVROOT;
ext->minvalue = 0;
ext->maxvalue = 0;
ext->flags = 0;
ext->parent = -1;
ext->control_no = -1;
ext->update_counter = 0;
ext->rgbcolor = 0;
memcpy(&ext->data, &root,
sizeof(root) > sizeof(ext->data) ?
sizeof(ext->data) : sizeof(root));
return 0;
}
mdi.index = ext->ctrl - 1;
retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
ext->flags = MIXF_READABLE | MIXF_WRITEABLE | MIXF_POLL;
ext->parent = mdi.mixer_class + 1;
strlcpy(ext->id, mdi.label.name, sizeof(ext->id));
strlcpy(ext->extname, mdi.label.name, sizeof(ext->extname));
len = strlen(ext->extname);
memset(ext->data, 0, sizeof(ext->data));
ext->control_no = -1;
ext->update_counter = 0;
ext->rgbcolor = 0;
switch (mdi.type) {
case AUDIO_MIXER_CLASS:
ext->type = MIXT_GROUP;
ext->parent = 0;
ext->minvalue = 0;
ext->maxvalue = 0;
break;
case AUDIO_MIXER_ENUM:
ext->maxvalue = mdi.un.e.num_mem;
ext->minvalue = 0;
for (i = 0; i < mdi.un.e.num_mem; ++i) {
ext->enum_present[i / 8] |= (1 << (i % 8));
}
if (mdi.un.e.num_mem == 2) {
if (!strcmp(mdi.un.e.member[0].label.name, AudioNoff) &&
!strcmp(mdi.un.e.member[1].label.name, AudioNon)) {
ext->type = MIXT_MUTE;
} else {
ext->type = MIXT_ENUM;
}
} else {
ext->type = MIXT_ENUM;
}
break;
case AUDIO_MIXER_SET:
ext->maxvalue = mdi.un.s.num_mem;
ext->minvalue = 0;
#ifdef notyet
/*
* XXX: This is actually the correct type for "set"
* controls, but it seems no real world software
* supports it. The only documentation exists in
* the OSSv4 headers and describes it as "reserved
* for Sun's implementation".
*/
ext->type = MIXT_ENUM_MULTI;
#else
ext->type = MIXT_ENUM;
#endif
for (i = 0; i < mdi.un.s.num_mem; ++i) {
ext->enum_present[i / 8] |= (1 << (i % 8));
}
break;
case AUDIO_MIXER_VALUE:
ext->maxvalue = UCHAR_MAX + 1;
ext->minvalue = 0;
if (mdi.un.v.num_channels == 2) {
ext->type = MIXT_STEREOSLIDER;
} else {
ext->type = MIXT_MONOSLIDER;
}
break;
}
close(newfd);
break;
case SNDCTL_MIX_ENUMINFO:
ei = (oss_mixer_enuminfo *)argp;
if (ei == NULL) {
errno = EINVAL;
return -1;
}
if (ei->ctrl == 0) {
errno = EINVAL;
return -1;
}
snprintf(devname, sizeof(devname), "/dev/mixer%d", ei->dev);
if ((newfd = open(devname, O_RDONLY)) < 0)
return newfd;
mdi.index = ei->ctrl - 1;
retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi);
tmperrno = errno;
close(newfd);
if (retval < 0) {
errno = tmperrno;
return retval;
}
ei->version = 0;
switch (mdi.type) {
case AUDIO_MIXER_ENUM:
ei->nvalues = mdi.un.e.num_mem;
noffs = 0;
for (i = 0; i < ei->nvalues; ++i) {
ei->strindex[i] = noffs;
len = strlen(mdi.un.e.member[i].label.name) + 1;
if ((noffs + len) >= sizeof(ei->strings)) {
errno = ENOMEM;
return -1;
}
memcpy(ei->strings + noffs,
mdi.un.e.member[i].label.name, len);
noffs += len;
}
break;
case AUDIO_MIXER_SET:
ei->nvalues = mdi.un.s.num_mem;
noffs = 0;
for (i = 0; i < ei->nvalues; ++i) {
ei->strindex[i] = noffs;
len = strlen(mdi.un.s.member[i].label.name) + 1;
if ((noffs + len) >= sizeof(ei->strings)) {
errno = ENOMEM;
return -1;
}
memcpy(ei->strings + noffs,
mdi.un.s.member[i].label.name, len);
noffs += len;
}
break;
default:
errno = EINVAL;
return -1;
}
break;
case SNDCTL_MIX_WRITE:
mv = (oss_mixer_value *)argp;
if (mv == NULL) {
errno = EINVAL;
return -1;
}
if (mv->ctrl == 0) {
errno = EINVAL;
return -1;
}
snprintf(devname, sizeof(devname), "/dev/mixer%d", mv->dev);
if ((newfd = open(devname, O_RDWR)) < 0)
return newfd;
mdi.index = mc.dev = mv->ctrl - 1;
retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
switch (mdi.type) {
case AUDIO_MIXER_ENUM:
if (mv->value >= mdi.un.e.num_mem) {
close(newfd);
errno = EINVAL;
return -1;
}
mc.un.ord = mdi.un.e.member[mv->value].ord;
break;
case AUDIO_MIXER_SET:
if (mv->value >= mdi.un.s.num_mem) {
close(newfd);
errno = EINVAL;
return -1;
}
#ifdef notyet
mc.un.mask = 0;
for (i = 0; i < mdi.un.s.num_mem; ++i) {
if (mv->value & (1 << i)) {
mc.un.mask |= mdi.un.s.member[mv->value].mask;
}
}
#else
mc.un.mask = mdi.un.s.member[mv->value].mask;
#endif
break;
case AUDIO_MIXER_VALUE:
if (mdi.un.v.num_channels != 2) {
for (i = 0; i < mdi.un.v.num_channels; ++i) {
mc.un.value.level[i] = mv->value;
}
} else {
mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] =
(mv->value >> 0) & 0xFF;
mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] =
(mv->value >> 8) & 0xFF;
}
break;
}
retval = ioctl(newfd, AUDIO_MIXER_WRITE, &mc);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
close(newfd);
break;
case SNDCTL_MIX_READ:
mv = (oss_mixer_value *)argp;
if (mv == NULL) {
errno = EINVAL;
return -1;
}
if (mv->ctrl == 0) {
errno = EINVAL;
return -1;
}
snprintf(devname, sizeof(devname), "/dev/mixer%d", mv->dev);
if ((newfd = open(devname, O_RDWR)) < 0)
return newfd;
mdi.index = mc.dev = (mv->ctrl - 1);
retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
mc.dev = mdi.index;
retval = ioctl(newfd, AUDIO_MIXER_READ, &mc);
if (retval < 0) {
tmperrno = errno;
close(newfd);
errno = tmperrno;
return retval;
}
close(newfd);
mv->value = 0;
switch (mdi.type) {
case AUDIO_MIXER_ENUM:
for (i = 0; i < mdi.un.e.num_mem; ++i) {
if (mc.un.ord == mdi.un.e.member[i].ord) {
mv->value = i;
break;
}
}
break;
case AUDIO_MIXER_SET:
for (i = 0; i < mdi.un.s.num_mem; ++i) {
#ifdef notyet
if (mc.un.mask & mdi.un.s.member[i].mask)
mv->value |= (1 << i);
#else
if (mc.un.mask == mdi.un.s.member[i].mask) {
mv->value = i;
break;
}
#endif
}
break;
case AUDIO_MIXER_VALUE:
if (mdi.un.v.num_channels != 2) {
mv->value = mc.un.value.level[0];
} else {
mv->value = \
((mc.un.value.level[1] & 0xFF) << 8) |
((mc.un.value.level[0] & 0xFF) << 0);
}
break;
default:
errno = EINVAL;
return -1;
}
break;
default:
errno = EINVAL;
return -1;
}
return 0;
}
static int
global_oss4_ioctl(int fd, unsigned long com, void *argp)
{
int retval = 0;
switch (com) {
/*
* These ioctls were added in OSSv4 with the idea that
* applications could apply strings to audio devices to
* display what they are using them for (e.g. with song
* names) in mixer applications. In practice, the popular
* implementations of the API in FreeBSD and Solaris treat
* these as a no-op and return EINVAL, and no software in the
* wild seems to use them.
*/
case SNDCTL_SETSONG:
case SNDCTL_GETSONG:
case SNDCTL_SETNAME:
case SNDCTL_SETLABEL:
case SNDCTL_GETLABEL:
errno = EINVAL;
retval = -1;
break;
default:
errno = EINVAL;
retval = -1;
break;
}
return retval;
}
static int
getcaps(int fd, int *out)
{
int props, caps;
if (ioctl(fd, AUDIO_GETPROPS, &props) < 0)
return -1;
caps = DSP_CAP_TRIGGER;
if (props & AUDIO_PROP_FULLDUPLEX)
caps |= DSP_CAP_DUPLEX;
if (props & AUDIO_PROP_MMAP)
caps |= DSP_CAP_MMAP;
if (props & AUDIO_PROP_CAPTURE)
caps |= PCM_CAP_INPUT;
if (props & AUDIO_PROP_PLAYBACK)
caps |= PCM_CAP_OUTPUT;
*out = caps;
return 0;
}
static int
getaudiocount(void)
{
char devname[32];
int ndevs = 0;
int tmpfd;
int tmperrno = errno;
do {
snprintf(devname, sizeof(devname),
"/dev/audio%d", ndevs);
if ((tmpfd = open(devname, O_RDONLY)) != -1 ||
(tmpfd = open(devname, O_WRONLY)) != -1) {
ndevs++;
close(tmpfd);
}
} while (tmpfd != -1);
errno = tmperrno;
return ndevs;
}
static int
getmixercount(void)
{
char devname[32];
int ndevs = 0;
int tmpfd;
int tmperrno = errno;
do {
snprintf(devname, sizeof(devname),
"/dev/mixer%d", ndevs);
if ((tmpfd = open(devname, O_RDONLY)) != -1) {
ndevs++;
close(tmpfd);
}
} while (tmpfd != -1);
errno = tmperrno;
return ndevs;
}
static int
getmixercontrolcount(int fd)
{
struct mixer_devinfo mdi;
int ndevs = 0;
do {
mdi.index = ndevs++;
} while (ioctl(fd, AUDIO_MIXER_DEVINFO, &mdi) != -1);
return ndevs > 0 ? ndevs - 1 : 0;
}
static int
getvol(u_int gain, u_char balance)
{
u_int l, r;
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;
}
return TO_OSSVOL(l) | (TO_OSSVOL(r) << 8);
}
static void
setvol(int fd, int volume, bool record)
{
u_int lgain, rgain;
struct audio_info tmpinfo;
struct audio_prinfo *prinfo;
AUDIO_INITINFO(&tmpinfo);
prinfo = record ? &tmpinfo.record : &tmpinfo.play;
lgain = FROM_OSSVOL((volume >> 0) & 0xff);
rgain = FROM_OSSVOL((volume >> 8) & 0xff);
if (lgain == rgain) {
prinfo->gain = lgain;
prinfo->balance = AUDIO_MID_BALANCE;
} else if (lgain < rgain) {
prinfo->gain = rgain;
prinfo->balance = AUDIO_RIGHT_BALANCE -
(AUDIO_MID_BALANCE * lgain) / rgain;
} else {
prinfo->gain = lgain;
prinfo->balance = (AUDIO_MID_BALANCE * rgain) / lgain;
}
(void)ioctl(fd, AUDIO_SETINFO, &tmpinfo);
}
/*
* When AUDIO_SETINFO fails to set a channel count, the application's chosen
* number is out of range of what the kernel allows.
*
* When this happens, we use the current hardware settings. This is just in
* case an application is abusing SNDCTL_DSP_CHANNELS - OSSv4 always sets and
* returns a reasonable value, even if it wasn't what the user requested.
*
* Solaris guarantees this behaviour if nchannels = 0.
*
* XXX: If a device is opened for both playback and recording, and supports
* fewer channels for recording than playback, applications that do both will
* behave very strangely. OSS doesn't allow for reporting separate channel
* counts for recording and playback. This could be worked around by always
* mixing recorded data up to the same number of channels as is being used
* for playback.
*/
static void
setchannels(int fd, int mode, int nchannels)
{
struct audio_info tmpinfo, hwfmt;
if (ioctl(fd, AUDIO_GETFORMAT, &hwfmt) < 0) {
errno = 0;
hwfmt.record.channels = hwfmt.play.channels = 2;
}
if (mode & AUMODE_PLAY) {
AUDIO_INITINFO(&tmpinfo);
tmpinfo.play.channels = nchannels;
if (ioctl(fd, AUDIO_SETINFO, &tmpinfo) < 0) {
errno = 0;
AUDIO_INITINFO(&tmpinfo);
tmpinfo.play.channels = hwfmt.play.channels;
(void)ioctl(fd, AUDIO_SETINFO, &tmpinfo);
}
}
if (mode & AUMODE_RECORD) {
AUDIO_INITINFO(&tmpinfo);
tmpinfo.record.channels = nchannels;
if (ioctl(fd, AUDIO_SETINFO, &tmpinfo) < 0) {
errno = 0;
AUDIO_INITINFO(&tmpinfo);
tmpinfo.record.channels = hwfmt.record.channels;
(void)ioctl(fd, AUDIO_SETINFO, &tmpinfo);
}
}
}
/*
* Check that the blocksize is a power of 2 as OSS wants.
* If not, set it to be.
*/
static void
setblocksize(int fd, struct audio_info *info)
{
struct audio_info set;
size_t s;
if (info->blocksize & (info->blocksize-1)) {
for(s = 32; s < info->blocksize; s <<= 1)
;
AUDIO_INITINFO(&set);
set.blocksize = s;
ioctl(fd, AUDIO_SETINFO, &set);
ioctl(fd, AUDIO_GETBUFINFO, info);
}
}
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