libossaudio from NetBSD mostly by Lennart Augustsson <augustss@cs.chalmers.se>

1 files changed, 699 insertions, 0 deletions

diff --git a/lib/libossaudio/ossaudio.c b/lib/libossaudio/ossaudio.c
new file mode 100644
index 00000000000..9b9b7215847
--- /dev/null
+++ b/lib/libossaudio/ossaudio.c
@@ -0,0 +1,699 @@
+/*	$OpenBSD: ossaudio.c,v 1.1 1998/05/01 09:23:00 provos Exp $	*/
+/*	$NetBSD: ossaudio.c,v 1.5 1998/03/23 00:39:18 augustss 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 / 255)
+#define FROM_OSSVOL(x) ((x) * 255 / 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 *);
+
+#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:
+	case SNDCTL_DSP_POST:
+		retval = ioctl(fd, AUDIO_DRAIN, 0);
+		if (retval < 0)
+			return retval;
+		break;
+	case SNDCTL_DSP_SPEED:
+		AUDIO_INITINFO(&tmpinfo);
+		tmpinfo.play.sample_rate =
+		tmpinfo.record.sample_rate = INTARG;
+		(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
+		/* fall into ... */
+	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);
+		/* fall into ... */
+	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);
+		/* fall into ... */
+	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 = (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; 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++) {
+			if (tmpenc.flags & AUDIO_ENCODINGFLAG_EMULATED)
+				continue; /* Don't report emulated modes */
+			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:
+	case SNDCTL_DSP_GETISPACE:
+		retval = ioctl(fd, AUDIO_GETINFO, (caddr_t)&tmpinfo);
+		if (retval < 0)
+			return retval;
+		setblocksize(fd, &tmpinfo);
+		bufinfo.fragsize = tmpinfo.blocksize;
+		bufinfo.fragments = /* XXX */
+		bufinfo.fragstotal = tmpinfo.play.buffer_size / bufinfo.fragsize;
+		bufinfo.bytes = tmpinfo.play.buffer_size;
+		*(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, (caddr_t)&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, (caddr_t)&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, (caddr_t)&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 32 devices
+ * some will not be available to Linux */
+#define NETBSD_MAXDEVS 32
+struct audiodevinfo {
+	int done;
+	dev_t dev;
+	int16_t devmap[SOUND_MIXER_NRDEVICES], 
+	        rdevmap[NETBSD_MAXDEVS];
+        u_long devmask, recmask, stereomask;
+	u_long caps, source;
+};
+
+/* 
+ * 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;
+	static struct {
+		char *name;
+		int code;
+	} *dp, devs[] = {
+		{ AudioNmicrophone,	SOUND_MIXER_MIC },
+		{ AudioNline,		SOUND_MIXER_LINE },
+		{ AudioNcd,		SOUND_MIXER_CD },
+		{ AudioNdac,		SOUND_MIXER_PCM },
+		{ 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 = -1;
+	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;
+	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;
+			}
+			break;
+		case AUDIO_MIXER_ENUM:
+			if (strcmp(mi.label.name, AudioNsource) == 0) {
+				int j;
+				di->source = i;
+				for(j = 0; j < mi.un.e.num_mem; j++)
+					di->recmask |= 1 << di->rdevmap[mi.un.e.member[j].ord];
+				di->caps = SOUND_CAP_EXCL_INPUT;
+			}
+			break;
+		case AUDIO_MIXER_SET:
+			if (strcmp(mi.label.name, AudioNsource) == 0) {
+				int j;
+				di->source = i;
+				for(j = 0; j < mi.un.s.num_mem; j++) {
+					int k, mask = mi.un.s.member[j].mask;
+					if (mask) {
+						for(k = 0; !(mask & 1); mask >>= 1, k++)
+							;
+						di->recmask |= 1 << di->rdevmap[k];
+					}
+				}
+			}
+			break;
+		}
+	}
+	return di;
+}
+
+int
+mixer_ioctl(int fd, unsigned long com, void *argp)
+{	       
+	struct audiodevinfo *di;
+	mixer_ctrl_t mc;
+	int idat;
+	int i;
+	int retval;
+	int l, r, n;
+
+	di = getdevinfo(fd);
+	if (di == 0)
+		return -1;
+
+	switch (com) {
+	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;
+			idat = 1 << di->rdevmap[mc.un.ord];
+		} else {
+			int k;
+			unsigned int mask;
+			mc.type = AUDIO_MIXER_SET;
+			retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
+			if (retval < 0)
+				return retval;
+			idat = 0;
+			for(mask = mc.un.mask, k = 0; mask; mask >>= 1, k++)
+				if (mask & 1)
+					idat |= 1 << di->rdevmap[k];
+		}
+		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 = 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 |= 1 << 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);
+	}
+}
+