 @@ -1,1790 +1,1795 @@
-/*	$NetBSD: ossaudio.c,v 1.65 2020/12/19 12:55:28 nia Exp $	*/
+/*	$NetBSD: ossaudio.c,v 1.66 2021/03/15 10:58:05 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 1.65 2020/12/19 12:55:28 nia Exp $");
+__RCSID("$NetBSD: ossaudio.c,v 1.66 2021/03/15 10:58:05 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:
 		/*
 		 * 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;
 		AUDIO_INITINFO(&tmpinfo);
 		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;
+		}
 		mc.type = mdi.type;
 		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:
 			mc.un.value.num_channels = mdi.un.v.num_channels;
 			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;
 		mc.type = mdi.type;
 		if (mdi.type == AUDIO_MIXER_VALUE)
 			mc.un.value.num_channels = mdi.un.v.num_channels;
 		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);
+	}
+}