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NetBSD/lib/libossaudio/ossaudio.c
mycroft cbfac6596e Make SNDCTL_DSP_POST a nop. It's explicitly *not* supposed to sleep, and as 
it's merely advisory (and in fact is implemented as a nop in the OSS->ALSA
shim), it should be safe to ignore it.
2001-12-24 00:10:46 +00:00

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/* $NetBSD: ossaudio.c,v 1.16 2001/12/24 00:10:46 mycroft Exp $ */
/*-
* Copyright (c) 1997 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.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the NetBSD
* Foundation, Inc. and its contributors.
* 4. Neither the name of The NetBSD Foundation nor the names of its
* contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* 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.
*/
/*
* This is an OSS (Linux) sound API emulator.
* It provides the essentials of the API.
*/
/* XXX This file is essentially the same as sys/compat/ossaudio.c.
* With some preprocessor magic it could be the same file.
*/
#include <string.h>
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/audioio.h>
#include <sys/stat.h>
#include <errno.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)
static struct audiodevinfo *getdevinfo(int);
static void setblocksize(int, struct audio_info *);
static int audio_ioctl(int, unsigned long, void *);
static int mixer_ioctl(int, unsigned long, void *);
static int opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq);
static int enum_to_ord(struct audiodevinfo *di, int enm);
static int enum_to_mask(struct audiodevinfo *di, int enm);
#define INTARG (*(int*)argp)
int
_oss_ioctl(int fd, unsigned long com, void *argp)
{
if (IOCGROUP(com) == 'P')
return audio_ioctl(fd, com, argp);
else if (IOCGROUP(com) == 'M')
return mixer_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;
struct audio_offset tmpoffs;
struct audio_buf_info bufinfo;
struct count_info cntinfo;
struct audio_encoding tmpenc;
u_int u;
int idat, idata;
int retval;
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_POST:
/* This call is merely advisory, and may be a nop. */
break;
case SNDCTL_DSP_SPEED:
AUDIO_INITINFO(&tmpinfo);
tmpinfo.play.sample_rate =
tmpinfo.record.sample_rate = INTARG;
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
/* FALLTHRU */
case SOUND_PCM_READ_RATE:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = tmpinfo.play.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_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = tmpinfo.play.channels - 1;
break;
case SNDCTL_DSP_GETBLKSIZE:
retval = ioctl(fd, AUDIO_GETINFO, &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;
default:
return EINVAL;
}
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
/* FALLTHRU */
case SOUND_PCM_READ_BITS:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
switch (tmpinfo.play.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 (tmpinfo.play.precision == 16)
idat = AFMT_S16_LE;
else
idat = AFMT_S8;
break;
case AUDIO_ENCODING_SLINEAR_BE:
if (tmpinfo.play.precision == 16)
idat = AFMT_S16_BE;
else
idat = AFMT_S8;
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (tmpinfo.play.precision == 16)
idat = AFMT_U16_LE;
else
idat = AFMT_U8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (tmpinfo.play.precision == 16)
idat = AFMT_U16_BE;
else
idat = AFMT_U8;
break;
case AUDIO_ENCODING_ADPCM:
idat = AFMT_IMA_ADPCM;
break;
}
INTARG = idat;
break;
case SNDCTL_DSP_CHANNELS:
AUDIO_INITINFO(&tmpinfo);
tmpinfo.play.channels =
tmpinfo.record.channels = INTARG;
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
/* FALLTHRU */
case SOUND_PCM_READ_CHANNELS:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
INTARG = tmpinfo.play.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_GETINFO, &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)
return EINVAL;
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_GETINFO, &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 == 16)
idat |= AFMT_S16_LE;
else
idat |= AFMT_S8;
break;
case AUDIO_ENCODING_SLINEAR_BE:
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;
default:
break;
}
}
INTARG = idat;
break;
case SNDCTL_DSP_GETOSPACE:
retval = ioctl(fd, AUDIO_GETINFO, &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_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
setblocksize(fd, &tmpinfo);
bufinfo.fragsize = tmpinfo.blocksize;
bufinfo.fragments = tmpinfo.hiwat -
(tmpinfo.record.seek + tmpinfo.blocksize - 1)/tmpinfo.blocksize;
bufinfo.fragstotal = tmpinfo.hiwat;
bufinfo.bytes = tmpinfo.hiwat * tmpinfo.blocksize - 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 = ioctl(fd, AUDIO_GETPROPS, &idata);
if (retval < 0)
return retval;
idat = DSP_CAP_TRIGGER; /* pretend we have trigger */
if (idata & AUDIO_PROP_FULLDUPLEX)
idat |= DSP_CAP_DUPLEX;
if (idata & AUDIO_PROP_MMAP)
idat |= DSP_CAP_MMAP;
INTARG = idat;
break;
#if 0
case SNDCTL_DSP_GETTRIGGER:
retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
if (retval < 0)
return retval;
idat = (tmpinfo.play.pause ? 0 : PCM_ENABLE_OUTPUT) |
(tmpinfo.record.pause ? 0 : PCM_ENABLE_INPUT);
retval = copyout(&idat, SCARG(uap, data), sizeof idat);
if (retval < 0)
return retval;
break;
case SNDCTL_DSP_SETTRIGGER:
AUDIO_INITINFO(&tmpinfo);
retval = copyin(SCARG(uap, data), &idat, sizeof idat);
if (retval < 0)
return retval;
tmpinfo.play.pause = (idat & PCM_ENABLE_OUTPUT) == 0;
tmpinfo.record.pause = (idat & PCM_ENABLE_INPUT) == 0;
(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
retval = copyout(&idat, SCARG(uap, data), sizeof idat);
if (retval < 0)
return retval;
break;
#else
case SNDCTL_DSP_GETTRIGGER:
case SNDCTL_DSP_SETTRIGGER:
/* XXX Do nothing for now. */
INTARG = PCM_ENABLE_OUTPUT;
break;
#endif
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_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_MAPINBUF:
case SNDCTL_DSP_MAPOUTBUF:
case SNDCTL_DSP_SETSYNCRO:
case SNDCTL_DSP_SETDUPLEX:
case SNDCTL_DSP_PROFILE:
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 Linux */
#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, 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 Linux 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 {
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 = { 0 };
struct audiodevinfo *di = &devcache;
struct stat sb;
/* 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;
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;
strncpy(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;
}
int
mixer_ioctl(int fd, unsigned long com, void *argp)
{
struct audiodevinfo *di;
struct mixer_info *omi;
struct audio_device adev;
mixer_ctrl_t mc;
int idat;
int i;
int retval;
int l, r, n, error, e;
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;
strncpy(omi->id, adev.name, sizeof omi->id);
strncpy(omi->name, adev.name, sizeof omi->name);
return 0;
case SOUND_MIXER_READ_RECSRC:
if (di->source == -1)
return EINVAL;
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)
return EINVAL;
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)
return EINVAL;
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)
return EINVAL;
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)
return EINVAL;
mc.dev = di->devmap[n];
mc.type = AUDIO_MIXER_VALUE;
doread:
mc.un.value.num_channels = di->stereomask & (1<<n) ? 2 : 1;
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
if (retval < 0)
return retval;
if (mc.type != AUDIO_MIXER_VALUE)
return EINVAL;
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)
return EINVAL;
idat = INTARG;
l = FROM_OSSVOL( idat & 0xff);
r = FROM_OSSVOL((idat >> 8) & 0xff);
mc.dev = di->devmap[n];
mc.type = AUDIO_MIXER_VALUE;
if (di->stereomask & (1<<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 = idat;
return 0;
}
/*
* 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;
int 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_GETINFO, info);
}
}
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