 @@ -1,1141 +1,1141 @@
-/*	$NetBSD: ossaudio.c,v 1.43 2020/04/19 13:44:50 nia Exp $	*/
+/*	$NetBSD: ossaudio.c,v 1.44 2020/04/20 12:01:44 nia 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.
+ *
  * 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 1.43 2020/04/19 13:44:50 nia Exp $");
+__RCSID("$NetBSD: ossaudio.c,v 1.44 2020/04/20 12:01:44 nia Exp $");
 /*
  * 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 <fcntl.h>
 #include <stdio.h>
 #include <unistd.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 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_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_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 count_info cntinfo;
 	struct audio_encoding tmpenc;
 	struct oss_sysinfo tmpsysinfo;
 	struct oss_audioinfo *tmpaudioinfo;
 	audio_device_t tmpaudiodev;
 	struct stat tmpstat;
 	dev_t devno;
 	char version[32] = "4.01";
 	char license[16] = "NetBSD";
 	u_int u;
 	u_int encoding;
 	u_int precision;
 	int idat, idata;
 	int retval;
 	int i;
 	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_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;
 		/*
 		 * The default NetBSD behavior if an unsupported sample rate
 		 * is set is to return an error code and keep the rate at the
 		 * default of 8000 Hz.
+		 *
 		 * However, OSS specifies that a sample rate supported by the
 		 * hardware is returned if the exact rate could not be set.
+		 *
 		 * So, if the chosen sample rate is invalid, set and return
 		 * the current hardware rate.
 		 */
 		if (ioctl(fd, AUDIO_SETINFO, &tmpinfo) < 0) {
 			/* Don't care that SETINFO failed the first time... */
 			errno = 0;
 			retval = ioctl(fd, AUDIO_GETFORMAT, &hwfmt);
 			if (retval < 0)
 				return retval;
 			retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
 			if (retval < 0)
 				return retval;
 			tmpinfo.play.sample_rate =
 			tmpinfo.record.sample_rate =
 			    (tmpinfo.mode == AUMODE_RECORD) ?
 			    hwfmt.record.sample_rate :
 			    hwfmt.play.sample_rate;
 			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)
 			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_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 = ioctl(fd, AUDIO_GETPROPS, &idata);
 		if (retval < 0)
 			return retval;
 		idat = DSP_CAP_TRIGGER;
 		if (idata & AUDIO_PROP_FULLDUPLEX)
 			idat |= DSP_CAP_DUPLEX;
 		if (idata & AUDIO_PROP_MMAP)
 			idat |= DSP_CAP_MMAP;
 		INTARG = idat;
 		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_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_SYSINFO:
 		strlcpy(tmpsysinfo.product, "OSS/NetBSD",
 		    sizeof tmpsysinfo.product);
 		strlcpy(tmpsysinfo.version, version, sizeof tmpsysinfo.version);
 		strlcpy(tmpsysinfo.license, license, sizeof tmpsysinfo.license);
 		tmpsysinfo.versionnum = SOUND_VERSION;
 		memset(tmpsysinfo.options, 0, 8);
 		tmpsysinfo.numaudios = OSS_MAX_AUDIO_DEVS;
 		tmpsysinfo.numaudioengines = 1;
 		memset(tmpsysinfo.openedaudio, 0, sizeof(tmpsysinfo.openedaudio));
 		tmpsysinfo.numsynths = 1;
 		tmpsysinfo.nummidis = -1;
 		tmpsysinfo.numtimers = -1;
 		tmpsysinfo.nummixers = 1;
 		tmpsysinfo.numcards = 1;
 		memset(tmpsysinfo.openedmidi, 0, sizeof(tmpsysinfo.openedmidi));
 		*(struct oss_sysinfo *)argp = tmpsysinfo;
 		break;
 	case SNDCTL_ENGINEINFO:
 	case SNDCTL_AUDIOINFO:
 		devno = 0;
 		tmpaudioinfo = (struct oss_audioinfo*)argp;
 		if (tmpaudioinfo == NULL)
 			return EINVAL;
 		if (tmpaudioinfo->dev < 0) {
 			fstat(fd, &tmpstat);
 			if ((tmpstat.st_rdev & 0xff00) == 0x2a00)
 				devno = tmpstat.st_rdev & 0xff;
 			if (devno >= 0x80)
 				tmpaudioinfo->dev = devno & 0x7f;
+		}
 		if (tmpaudioinfo->dev < 0)
 			tmpaudioinfo->dev = 0;
 		snprintf(tmpaudioinfo->devnode, OSS_DEVNODE_SIZE,
 		    "/dev/audio%d", tmpaudioinfo->dev);
 		retval = ioctl(fd, AUDIO_GETDEV, &tmpaudiodev);
 		if (retval < 0)
 			return retval;
 		retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
 		if (retval < 0)
 			return retval;
 		retval = ioctl(fd, AUDIO_GETPROPS, &idata);
 		if (retval < 0)
 			return retval;
-		idat = DSP_CAP_TRIGGER; /* pretend we have trigger */
+		idat = DSP_CAP_TRIGGER;
 		if (idata & AUDIO_PROP_FULLDUPLEX)
 			idat |= DSP_CAP_DUPLEX;
 		if (idata & AUDIO_PROP_MMAP)
 			idat |= DSP_CAP_MMAP;
 		idat = PCM_CAP_INPUT | PCM_CAP_OUTPUT;
 		strlcpy(tmpaudioinfo->name, tmpaudiodev.name,
 		    sizeof tmpaudioinfo->name);
 		tmpaudioinfo->busy = tmpinfo.play.open;
 		tmpaudioinfo->pid = -1;
 		tmpaudioinfo->caps = idat;
 		ioctl(fd, SNDCTL_DSP_GETFMTS, &tmpaudioinfo->iformats);
 		tmpaudioinfo->oformats = tmpaudioinfo->iformats;
 		tmpaudioinfo->magic = -1;
 		memset(tmpaudioinfo->cmd, 0, 64);
 		tmpaudioinfo->card_number = -1;
 		memset(tmpaudioinfo->song_name, 0, 64);
 		memset(tmpaudioinfo->label, 0, 16);
 		tmpaudioinfo->port_number = tmpinfo.play.port;
 		tmpaudioinfo->mixer_dev = tmpaudioinfo->dev;
 		tmpaudioinfo->legacy_device = -1;
 		tmpaudioinfo->enabled = 1;
 		tmpaudioinfo->flags = -1;
 		tmpaudioinfo->min_rate = tmpinfo.play.sample_rate;
 		tmpaudioinfo->max_rate = tmpinfo.play.sample_rate;
 		tmpaudioinfo->nrates = 2;
 		for (i = 0; i < tmpaudioinfo->nrates; i++)
 			tmpaudioinfo->rates[i] = tmpinfo.play.sample_rate;
 		tmpaudioinfo->min_channels = tmpinfo.play.channels;
 		tmpaudioinfo->max_channels = tmpinfo.play.channels;
 		tmpaudioinfo->binding = -1;
 		tmpaudioinfo->rate_source = -1;
 		memset(tmpaudioinfo->handle, 0, 16);
 		tmpaudioinfo->next_play_engine = 0;
 		tmpaudioinfo->next_rec_engine = 0;
 		argp = tmpaudioinfo;
 		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:
 		return EINVAL;
 	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 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;
 	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 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 {
 		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;
+}
 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;
 	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)
 			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 << (u_int)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((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
 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.
+ *
  * 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);
+	}
+}