17ac2f03ee
Accept mixer values above max (100). Handle the info from AUDIO_MIXER_DEVINFO properly, parts of it is opaque.
773 lines
20 KiB
C
773 lines
20 KiB
C
/* $NetBSD: ossaudio.c,v 1.11 2000/07/04 09:56:14 augustss Exp $ */
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/*-
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* Copyright (c) 1997 The NetBSD Foundation, Inc.
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* All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. All advertising materials mentioning features or use of this software
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* must display the following acknowledgement:
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* This product includes software developed by the NetBSD
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* Foundation, Inc. and its contributors.
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* 4. Neither the name of The NetBSD Foundation nor the names of its
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* contributors may be used to endorse or promote products derived
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* from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
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* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
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* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
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* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
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* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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* POSSIBILITY OF SUCH DAMAGE.
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*/
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/*
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* This is an OSS (Linux) sound API emulator.
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* It provides the essentials of the API.
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*/
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/* XXX This file is essentially the same as sys/compat/ossaudio.c.
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* With some preprocessor magic it could be the same file.
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*/
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#include <string.h>
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#include <sys/types.h>
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#include <sys/ioctl.h>
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#include <sys/audioio.h>
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#include <sys/stat.h>
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#include <errno.h>
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#include "soundcard.h"
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#undef ioctl
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#define GET_DEV(com) ((com) & 0xff)
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#define TO_OSSVOL(x) ((x) * 100 / 255)
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#define FROM_OSSVOL(x) (((x) > 100 ? 100 : (x)) * 255 / 100)
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static struct audiodevinfo *getdevinfo(int);
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static void setblocksize(int, struct audio_info *);
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static int audio_ioctl(int, unsigned long, void *);
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static int mixer_ioctl(int, unsigned long, void *);
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static int opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq);
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static int enum_to_ord(struct audiodevinfo *di, int enm);
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static int enum_to_mask(struct audiodevinfo *di, int enm);
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#define INTARG (*(int*)argp)
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int
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_oss_ioctl(int fd, unsigned long com, void *argp)
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{
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if (IOCGROUP(com) == 'P')
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return audio_ioctl(fd, com, argp);
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else if (IOCGROUP(com) == 'M')
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return mixer_ioctl(fd, com, argp);
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else
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return ioctl(fd, com, argp);
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}
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static int
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audio_ioctl(int fd, unsigned long com, void *argp)
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{
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struct audio_info tmpinfo;
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struct audio_offset tmpoffs;
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struct audio_buf_info bufinfo;
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struct count_info cntinfo;
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struct audio_encoding tmpenc;
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u_int u;
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int idat, idata;
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int retval;
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switch (com) {
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case SNDCTL_DSP_RESET:
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retval = ioctl(fd, AUDIO_FLUSH, 0);
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if (retval < 0)
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return retval;
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break;
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case SNDCTL_DSP_SYNC:
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case SNDCTL_DSP_POST:
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retval = ioctl(fd, AUDIO_DRAIN, 0);
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if (retval < 0)
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return retval;
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break;
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case SNDCTL_DSP_SPEED:
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AUDIO_INITINFO(&tmpinfo);
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tmpinfo.play.sample_rate =
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tmpinfo.record.sample_rate = INTARG;
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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/* FALLTHRU */
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case SOUND_PCM_READ_RATE:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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INTARG = tmpinfo.play.sample_rate;
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break;
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case SNDCTL_DSP_STEREO:
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AUDIO_INITINFO(&tmpinfo);
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tmpinfo.play.channels =
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tmpinfo.record.channels = INTARG ? 2 : 1;
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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INTARG = tmpinfo.play.channels - 1;
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break;
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case SNDCTL_DSP_GETBLKSIZE:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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setblocksize(fd, &tmpinfo);
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INTARG = tmpinfo.blocksize;
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break;
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case SNDCTL_DSP_SETFMT:
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AUDIO_INITINFO(&tmpinfo);
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switch (INTARG) {
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case AFMT_MU_LAW:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 8;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_ULAW;
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break;
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case AFMT_A_LAW:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 8;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_ALAW;
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break;
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case AFMT_U8:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 8;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR;
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break;
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case AFMT_S8:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 8;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR;
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break;
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case AFMT_S16_LE:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 16;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE;
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break;
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case AFMT_S16_BE:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 16;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE;
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break;
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case AFMT_U16_LE:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 16;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_LE;
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break;
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case AFMT_U16_BE:
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tmpinfo.play.precision =
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tmpinfo.record.precision = 16;
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tmpinfo.play.encoding =
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tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_BE;
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break;
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default:
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return EINVAL;
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}
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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/* FALLTHRU */
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case SOUND_PCM_READ_BITS:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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switch (tmpinfo.play.encoding) {
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case AUDIO_ENCODING_ULAW:
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idat = AFMT_MU_LAW;
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break;
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case AUDIO_ENCODING_ALAW:
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idat = AFMT_A_LAW;
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break;
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case AUDIO_ENCODING_SLINEAR_LE:
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if (tmpinfo.play.precision == 16)
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idat = AFMT_S16_LE;
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else
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idat = AFMT_S8;
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break;
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case AUDIO_ENCODING_SLINEAR_BE:
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if (tmpinfo.play.precision == 16)
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idat = AFMT_S16_BE;
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else
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idat = AFMT_S8;
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break;
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case AUDIO_ENCODING_ULINEAR_LE:
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if (tmpinfo.play.precision == 16)
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idat = AFMT_U16_LE;
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else
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idat = AFMT_U8;
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break;
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case AUDIO_ENCODING_ULINEAR_BE:
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if (tmpinfo.play.precision == 16)
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idat = AFMT_U16_BE;
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else
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idat = AFMT_U8;
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break;
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case AUDIO_ENCODING_ADPCM:
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idat = AFMT_IMA_ADPCM;
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break;
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}
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INTARG = idat;
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break;
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case SNDCTL_DSP_CHANNELS:
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AUDIO_INITINFO(&tmpinfo);
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tmpinfo.play.channels =
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tmpinfo.record.channels = INTARG;
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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/* FALLTHRU */
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case SOUND_PCM_READ_CHANNELS:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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INTARG = tmpinfo.play.channels;
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break;
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case SOUND_PCM_WRITE_FILTER:
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case SOUND_PCM_READ_FILTER:
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errno = EINVAL;
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return -1; /* XXX unimplemented */
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case SNDCTL_DSP_SUBDIVIDE:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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setblocksize(fd, &tmpinfo);
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idat = INTARG;
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if (idat == 0)
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idat = tmpinfo.play.buffer_size / tmpinfo.blocksize;
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idat = (tmpinfo.play.buffer_size / idat) & -4;
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AUDIO_INITINFO(&tmpinfo);
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tmpinfo.blocksize = idat;
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retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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INTARG = tmpinfo.play.buffer_size / tmpinfo.blocksize;
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break;
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case SNDCTL_DSP_SETFRAGMENT:
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AUDIO_INITINFO(&tmpinfo);
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idat = INTARG;
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if ((idat & 0xffff) < 4 || (idat & 0xffff) > 17)
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return EINVAL;
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tmpinfo.blocksize = 1 << (idat & 0xffff);
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tmpinfo.hiwat = ((unsigned)idat >> 16) & 0x7fff;
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if (tmpinfo.hiwat == 0) /* 0 means set to max */
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tmpinfo.hiwat = 65536;
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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u = tmpinfo.blocksize;
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for(idat = 0; u > 1; idat++, u >>= 1)
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;
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idat |= (tmpinfo.hiwat & 0x7fff) << 16;
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INTARG = idat;
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break;
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case SNDCTL_DSP_GETFMTS:
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for(idat = 0, tmpenc.index = 0;
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ioctl(fd, AUDIO_GETENC, &tmpenc) == 0;
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tmpenc.index++) {
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switch(tmpenc.encoding) {
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case AUDIO_ENCODING_ULAW:
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idat |= AFMT_MU_LAW;
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break;
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case AUDIO_ENCODING_ALAW:
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idat |= AFMT_A_LAW;
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break;
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case AUDIO_ENCODING_SLINEAR:
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idat |= AFMT_S8;
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break;
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case AUDIO_ENCODING_SLINEAR_LE:
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if (tmpenc.precision == 16)
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idat |= AFMT_S16_LE;
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else
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idat |= AFMT_S8;
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break;
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case AUDIO_ENCODING_SLINEAR_BE:
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if (tmpenc.precision == 16)
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idat |= AFMT_S16_BE;
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else
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idat |= AFMT_S8;
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break;
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case AUDIO_ENCODING_ULINEAR:
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idat |= AFMT_U8;
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break;
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case AUDIO_ENCODING_ULINEAR_LE:
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if (tmpenc.precision == 16)
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idat |= AFMT_U16_LE;
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else
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idat |= AFMT_U8;
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break;
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case AUDIO_ENCODING_ULINEAR_BE:
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if (tmpenc.precision == 16)
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idat |= AFMT_U16_BE;
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else
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idat |= AFMT_U8;
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break;
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case AUDIO_ENCODING_ADPCM:
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idat |= AFMT_IMA_ADPCM;
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break;
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default:
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break;
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}
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}
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INTARG = idat;
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break;
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case SNDCTL_DSP_GETOSPACE:
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case SNDCTL_DSP_GETISPACE:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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setblocksize(fd, &tmpinfo);
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bufinfo.fragsize = tmpinfo.blocksize;
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bufinfo.fragments = /* XXX */
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bufinfo.fragstotal = tmpinfo.play.buffer_size / bufinfo.fragsize;
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bufinfo.bytes = tmpinfo.play.buffer_size;
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*(struct audio_buf_info *)argp = bufinfo;
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break;
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case SNDCTL_DSP_NONBLOCK:
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idat = 1;
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retval = ioctl(fd, FIONBIO, &idat);
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if (retval < 0)
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return retval;
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break;
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case SNDCTL_DSP_GETCAPS:
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retval = ioctl(fd, AUDIO_GETPROPS, &idata);
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if (retval < 0)
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return retval;
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idat = DSP_CAP_TRIGGER; /* pretend we have trigger */
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if (idata & AUDIO_PROP_FULLDUPLEX)
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idat |= DSP_CAP_DUPLEX;
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if (idata & AUDIO_PROP_MMAP)
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idat |= DSP_CAP_MMAP;
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INTARG = idat;
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break;
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#if 0
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case SNDCTL_DSP_GETTRIGGER:
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retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo);
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if (retval < 0)
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return retval;
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idat = (tmpinfo.play.pause ? 0 : PCM_ENABLE_OUTPUT) |
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(tmpinfo.record.pause ? 0 : PCM_ENABLE_INPUT);
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retval = copyout(&idat, SCARG(uap, data), sizeof idat);
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if (retval < 0)
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return retval;
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break;
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case SNDCTL_DSP_SETTRIGGER:
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AUDIO_INITINFO(&tmpinfo);
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retval = copyin(SCARG(uap, data), &idat, sizeof idat);
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if (retval < 0)
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return retval;
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tmpinfo.play.pause = (idat & PCM_ENABLE_OUTPUT) == 0;
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tmpinfo.record.pause = (idat & PCM_ENABLE_INPUT) == 0;
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(void) ioctl(fd, AUDIO_SETINFO, &tmpinfo);
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retval = copyout(&idat, SCARG(uap, data), sizeof idat);
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if (retval < 0)
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return retval;
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break;
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#else
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case SNDCTL_DSP_GETTRIGGER:
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case SNDCTL_DSP_SETTRIGGER:
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/* XXX Do nothing for now. */
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INTARG = PCM_ENABLE_OUTPUT;
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break;
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#endif
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case SNDCTL_DSP_GETIPTR:
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retval = ioctl(fd, AUDIO_GETIOFFS, &tmpoffs);
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if (retval < 0)
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return retval;
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cntinfo.bytes = tmpoffs.samples;
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cntinfo.blocks = tmpoffs.deltablks;
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cntinfo.ptr = tmpoffs.offset;
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*(struct count_info *)argp = cntinfo;
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break;
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case SNDCTL_DSP_GETOPTR:
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retval = ioctl(fd, AUDIO_GETOOFFS, &tmpoffs);
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if (retval < 0)
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return retval;
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cntinfo.bytes = tmpoffs.samples;
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cntinfo.blocks = tmpoffs.deltablks;
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cntinfo.ptr = tmpoffs.offset;
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*(struct count_info *)argp = cntinfo;
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break;
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case SNDCTL_DSP_MAPINBUF:
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case SNDCTL_DSP_MAPOUTBUF:
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case SNDCTL_DSP_SETSYNCRO:
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case SNDCTL_DSP_SETDUPLEX:
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case SNDCTL_DSP_PROFILE:
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errno = EINVAL;
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return -1; /* XXX unimplemented */
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default:
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errno = EINVAL;
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return -1;
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}
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return 0;
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}
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/* If the NetBSD mixer device should have more than NETBSD_MAXDEVS devices
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* some will not be available to Linux */
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#define NETBSD_MAXDEVS 64
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struct audiodevinfo {
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int done;
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dev_t dev;
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int16_t devmap[SOUND_MIXER_NRDEVICES],
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rdevmap[NETBSD_MAXDEVS];
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char names[NETBSD_MAXDEVS][MAX_AUDIO_DEV_LEN];
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int enum2opaque[NETBSD_MAXDEVS];
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u_long devmask, recmask, stereomask;
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u_long caps, source;
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};
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static int
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opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq)
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{
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int i, o;
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for (i = 0; i < NETBSD_MAXDEVS; i++) {
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o = di->enum2opaque[i];
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if (o == opq)
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break;
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if (o == -1 && label != NULL &&
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!strncmp(di->names[i], label->name, sizeof di->names[i])) {
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di->enum2opaque[i] = opq;
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break;
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}
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}
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if (i >= NETBSD_MAXDEVS)
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i = -1;
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/*printf("opq_to_enum %s %d -> %d\n", label->name, opq, i);*/
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return (i);
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}
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static int
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enum_to_ord(struct audiodevinfo *di, int enm)
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{
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if (enm >= NETBSD_MAXDEVS)
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return (-1);
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/*printf("enum_to_ord %d -> %d\n", enm, di->enum2opaque[enm]);*/
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return (di->enum2opaque[enm]);
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}
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static int
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enum_to_mask(struct audiodevinfo *di, int enm)
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{
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int m;
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if (enm >= NETBSD_MAXDEVS)
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return (0);
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m = di->enum2opaque[enm];
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if (m == -1)
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m = 0;
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/*printf("enum_to_mask %d -> %d\n", enm, di->enum2opaque[enm]);*/
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return (m);
|
|
}
|
|
|
|
/*
|
|
* Collect the audio device information to allow faster
|
|
* emulation of the Linux mixer ioctls. Cache the information
|
|
* to eliminate the overhead of repeating all the ioctls needed
|
|
* to collect the information.
|
|
*/
|
|
static struct audiodevinfo *
|
|
getdevinfo(int fd)
|
|
{
|
|
mixer_devinfo_t mi;
|
|
int i, j, e;
|
|
static struct {
|
|
char *name;
|
|
int code;
|
|
} *dp, devs[] = {
|
|
{ AudioNmicrophone, SOUND_MIXER_MIC },
|
|
{ AudioNline, SOUND_MIXER_LINE },
|
|
{ AudioNcd, SOUND_MIXER_CD },
|
|
{ AudioNdac, SOUND_MIXER_PCM },
|
|
{ AudioNrecord, SOUND_MIXER_IMIX },
|
|
{ AudioNmaster, SOUND_MIXER_VOLUME },
|
|
{ AudioNtreble, SOUND_MIXER_TREBLE },
|
|
{ AudioNbass, SOUND_MIXER_BASS },
|
|
{ AudioNspeaker, SOUND_MIXER_SPEAKER },
|
|
/* { AudioNheadphone, ?? },*/
|
|
{ AudioNoutput, SOUND_MIXER_OGAIN },
|
|
{ AudioNinput, SOUND_MIXER_IGAIN },
|
|
/* { AudioNmaster, SOUND_MIXER_SPEAKER },*/
|
|
/* { AudioNstereo, ?? },*/
|
|
/* { AudioNmono, ?? },*/
|
|
{ AudioNfmsynth, SOUND_MIXER_SYNTH },
|
|
/* { AudioNwave, SOUND_MIXER_PCM },*/
|
|
{ AudioNmidi, SOUND_MIXER_SYNTH },
|
|
/* { AudioNmixerout, ?? },*/
|
|
{ 0, -1 }
|
|
};
|
|
static struct audiodevinfo devcache = { 0 };
|
|
struct audiodevinfo *di = &devcache;
|
|
struct stat sb;
|
|
|
|
/* Figure out what device it is so we can check if the
|
|
* cached data is valid.
|
|
*/
|
|
if (fstat(fd, &sb) < 0)
|
|
return 0;
|
|
if (di->done && di->dev == sb.st_dev)
|
|
return di;
|
|
|
|
di->done = 1;
|
|
di->dev = sb.st_dev;
|
|
di->devmask = 0;
|
|
di->recmask = 0;
|
|
di->stereomask = 0;
|
|
di->source = ~0;
|
|
di->caps = 0;
|
|
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
|
|
di->devmap[i] = -1;
|
|
for(i = 0; i < NETBSD_MAXDEVS; i++) {
|
|
di->rdevmap[i] = -1;
|
|
di->names[i][0] = '\0';
|
|
di->enum2opaque[i] = -1;
|
|
}
|
|
for(i = 0; i < NETBSD_MAXDEVS; i++) {
|
|
mi.index = i;
|
|
if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
|
|
break;
|
|
switch(mi.type) {
|
|
case AUDIO_MIXER_VALUE:
|
|
for(dp = devs; dp->name; dp++)
|
|
if (strcmp(dp->name, mi.label.name) == 0)
|
|
break;
|
|
if (dp->code >= 0) {
|
|
di->devmap[dp->code] = i;
|
|
di->rdevmap[i] = dp->code;
|
|
di->devmask |= 1 << dp->code;
|
|
if (mi.un.v.num_channels == 2)
|
|
di->stereomask |= 1 << dp->code;
|
|
strncpy(di->names[i], mi.label.name,
|
|
sizeof di->names[i]);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
for(i = 0; i < NETBSD_MAXDEVS; i++) {
|
|
mi.index = i;
|
|
if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0)
|
|
break;
|
|
if (strcmp(mi.label.name, AudioNsource) != 0)
|
|
continue;
|
|
di->source = i;
|
|
switch(mi.type) {
|
|
case AUDIO_MIXER_ENUM:
|
|
for(j = 0; j < mi.un.e.num_mem; j++) {
|
|
e = opaque_to_enum(di,
|
|
&mi.un.e.member[j].label,
|
|
mi.un.e.member[j].ord);
|
|
if (e >= 0)
|
|
di->recmask |= 1 << di->rdevmap[e];
|
|
}
|
|
di->caps = SOUND_CAP_EXCL_INPUT;
|
|
break;
|
|
case AUDIO_MIXER_SET:
|
|
for(j = 0; j < mi.un.s.num_mem; j++) {
|
|
e = opaque_to_enum(di,
|
|
&mi.un.s.member[j].label,
|
|
mi.un.s.member[j].mask);
|
|
if (e >= 0)
|
|
di->recmask |= 1 << di->rdevmap[e];
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
return di;
|
|
}
|
|
|
|
int
|
|
mixer_ioctl(int fd, unsigned long com, void *argp)
|
|
{
|
|
struct audiodevinfo *di;
|
|
struct mixer_info *omi;
|
|
struct audio_device adev;
|
|
mixer_ctrl_t mc;
|
|
int idat;
|
|
int i;
|
|
int retval;
|
|
int l, r, n, error, e;
|
|
|
|
di = getdevinfo(fd);
|
|
if (di == 0)
|
|
return -1;
|
|
|
|
switch (com) {
|
|
case OSS_GETVERSION:
|
|
idat = SOUND_VERSION;
|
|
break;
|
|
case SOUND_MIXER_INFO:
|
|
case SOUND_OLD_MIXER_INFO:
|
|
error = ioctl(fd, AUDIO_GETDEV, &adev);
|
|
if (error)
|
|
return (error);
|
|
omi = argp;
|
|
if (com == SOUND_MIXER_INFO)
|
|
omi->modify_counter = 1;
|
|
strncpy(omi->id, adev.name, sizeof omi->id);
|
|
strncpy(omi->name, adev.name, sizeof omi->name);
|
|
return 0;
|
|
case SOUND_MIXER_READ_RECSRC:
|
|
if (di->source == -1)
|
|
return EINVAL;
|
|
mc.dev = di->source;
|
|
if (di->caps & SOUND_CAP_EXCL_INPUT) {
|
|
mc.type = AUDIO_MIXER_ENUM;
|
|
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
|
|
if (retval < 0)
|
|
return retval;
|
|
e = opaque_to_enum(di, NULL, mc.un.ord);
|
|
if (e >= 0)
|
|
idat = 1 << di->rdevmap[e];
|
|
} else {
|
|
mc.type = AUDIO_MIXER_SET;
|
|
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
|
|
if (retval < 0)
|
|
return retval;
|
|
e = opaque_to_enum(di, NULL, mc.un.mask);
|
|
if (e >= 0)
|
|
idat = 1 << di->rdevmap[e];
|
|
}
|
|
break;
|
|
case SOUND_MIXER_READ_DEVMASK:
|
|
idat = di->devmask;
|
|
break;
|
|
case SOUND_MIXER_READ_RECMASK:
|
|
idat = di->recmask;
|
|
break;
|
|
case SOUND_MIXER_READ_STEREODEVS:
|
|
idat = di->stereomask;
|
|
break;
|
|
case SOUND_MIXER_READ_CAPS:
|
|
idat = di->caps;
|
|
break;
|
|
case SOUND_MIXER_WRITE_RECSRC:
|
|
case SOUND_MIXER_WRITE_R_RECSRC:
|
|
if (di->source == -1)
|
|
return EINVAL;
|
|
mc.dev = di->source;
|
|
idat = INTARG;
|
|
if (di->caps & SOUND_CAP_EXCL_INPUT) {
|
|
mc.type = AUDIO_MIXER_ENUM;
|
|
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++)
|
|
if (idat & (1 << i))
|
|
break;
|
|
if (i >= SOUND_MIXER_NRDEVICES ||
|
|
di->devmap[i] == -1)
|
|
return EINVAL;
|
|
mc.un.ord = enum_to_ord(di, di->devmap[i]);
|
|
} else {
|
|
mc.type = AUDIO_MIXER_SET;
|
|
mc.un.mask = 0;
|
|
for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
|
|
if (idat & (1 << i)) {
|
|
if (di->devmap[i] == -1)
|
|
return EINVAL;
|
|
mc.un.mask |= enum_to_mask(di, di->devmap[i]);
|
|
}
|
|
}
|
|
}
|
|
return ioctl(fd, AUDIO_MIXER_WRITE, &mc);
|
|
default:
|
|
if (MIXER_READ(SOUND_MIXER_FIRST) <= com &&
|
|
com < MIXER_READ(SOUND_MIXER_NRDEVICES)) {
|
|
n = GET_DEV(com);
|
|
if (di->devmap[n] == -1)
|
|
return EINVAL;
|
|
mc.dev = di->devmap[n];
|
|
mc.type = AUDIO_MIXER_VALUE;
|
|
doread:
|
|
mc.un.value.num_channels = di->stereomask & (1<<n) ? 2 : 1;
|
|
retval = ioctl(fd, AUDIO_MIXER_READ, &mc);
|
|
if (retval < 0)
|
|
return retval;
|
|
if (mc.type != AUDIO_MIXER_VALUE)
|
|
return EINVAL;
|
|
if (mc.un.value.num_channels != 2) {
|
|
l = r = mc.un.value.level[AUDIO_MIXER_LEVEL_MONO];
|
|
} else {
|
|
l = mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT];
|
|
r = mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
|
|
}
|
|
idat = TO_OSSVOL(l) | (TO_OSSVOL(r) << 8);
|
|
break;
|
|
} else if ((MIXER_WRITE_R(SOUND_MIXER_FIRST) <= com &&
|
|
com < MIXER_WRITE_R(SOUND_MIXER_NRDEVICES)) ||
|
|
(MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
|
|
com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))) {
|
|
n = GET_DEV(com);
|
|
if (di->devmap[n] == -1)
|
|
return EINVAL;
|
|
idat = INTARG;
|
|
l = FROM_OSSVOL( idat & 0xff);
|
|
r = FROM_OSSVOL((idat >> 8) & 0xff);
|
|
mc.dev = di->devmap[n];
|
|
mc.type = AUDIO_MIXER_VALUE;
|
|
if (di->stereomask & (1<<n)) {
|
|
mc.un.value.num_channels = 2;
|
|
mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
|
|
mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
|
|
} else {
|
|
mc.un.value.num_channels = 1;
|
|
mc.un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
|
|
}
|
|
retval = ioctl(fd, AUDIO_MIXER_WRITE, &mc);
|
|
if (retval < 0)
|
|
return retval;
|
|
if (MIXER_WRITE(SOUND_MIXER_FIRST) <= com &&
|
|
com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))
|
|
return 0;
|
|
goto doread;
|
|
} else {
|
|
errno = EINVAL;
|
|
return -1;
|
|
}
|
|
}
|
|
INTARG = idat;
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Check that the blocksize is a power of 2 as OSS wants.
|
|
* If not, set it to be.
|
|
*/
|
|
static void
|
|
setblocksize(int fd, struct audio_info *info)
|
|
{
|
|
struct audio_info set;
|
|
int s;
|
|
|
|
if (info->blocksize & (info->blocksize-1)) {
|
|
for(s = 32; s < info->blocksize; s <<= 1)
|
|
;
|
|
AUDIO_INITINFO(&set);
|
|
set.blocksize = s;
|
|
ioctl(fd, AUDIO_SETINFO, &set);
|
|
ioctl(fd, AUDIO_GETINFO, info);
|
|
}
|
|
}
|
|
|