/* $NetBSD: ossaudio.c,v 1.36.2.2 2020/04/27 14:32:34 martin 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 __RCSID("$NetBSD: ossaudio.c,v 1.36.2.2 2020/04/27 14:32:34 martin 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 #include #include #include #include #include #include #include #include #include #include #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 */ 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); } }