/* $NetBSD: ossaudio.c,v 1.64 2020/11/13 09:02:39 nia Exp $ */ /*- * Copyright (c) 1997, 2020 The NetBSD Foundation, Inc. * All rights reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * 1. Redistributions of source code must retain the above copyright * notice, this list of conditions and the following disclaimer. * 2. Redistributions in binary form must reproduce the above copyright * notice, this list of conditions and the following disclaimer in the * documentation and/or other materials provided with the distribution. * * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE * POSSIBILITY OF SUCH DAMAGE. */ #include __RCSID("$NetBSD: ossaudio.c,v 1.64 2020/11/13 09:02:39 nia Exp $"); /* * This is an Open Sound System compatibility layer, which provides * fairly complete ioctl emulation for OSSv3 and some of OSSv4. * * The canonical OSS specification is available at * http://manuals.opensound.com/developer/ * * This file is similar to sys/compat/ossaudio.c with additional OSSv4 * compatibility. */ #include #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 getaudiocount(void); static int getmixercount(void); static int getmixercontrolcount(int); static int getcaps(int, int *); static int getvol(u_int, u_char); static void setvol(int, int, bool); static void setchannels(int, int, int); static void setblocksize(int, struct audio_info *); static int audio_ioctl(int, unsigned long, void *); static int mixer_oss3_ioctl(int, unsigned long, void *); static int mixer_oss4_ioctl(int, unsigned long, void *); static int global_oss4_ioctl(int, unsigned long, void *); static int opaque_to_enum(struct audiodevinfo *, audio_mixer_name_t *, int); static int enum_to_ord(struct audiodevinfo *, int); static int enum_to_mask(struct audiodevinfo *, int); #define INTARG (*(int*)argp) int _oss_ioctl(int fd, unsigned long com, ...) { va_list ap; void *argp; va_start(ap, com); argp = va_arg(ap, void *); va_end(ap); if (IOCGROUP(com) == 'P') return audio_ioctl(fd, com, argp); else if (IOCGROUP(com) == 'M') return mixer_oss3_ioctl(fd, com, argp); else if (IOCGROUP(com) == 'X') return mixer_oss4_ioctl(fd, com, argp); else if (IOCGROUP(com) == 'Y') return global_oss4_ioctl(fd, com, argp); else return ioctl(fd, com, argp); } static int audio_ioctl(int fd, unsigned long com, void *argp) { struct audio_info tmpinfo, hwfmt; struct audio_offset tmpoffs; struct audio_buf_info bufinfo; struct audio_errinfo *tmperrinfo; struct count_info cntinfo; struct audio_encoding tmpenc; u_int u; u_int encoding; u_int precision; int perrors, rerrors; static int totalperrors = 0; static int totalrerrors = 0; oss_count_t osscount; int idat; int retval; idat = 0; switch (com) { case SNDCTL_DSP_RESET: retval = ioctl(fd, AUDIO_FLUSH, 0); if (retval < 0) return retval; break; case SNDCTL_DSP_SYNC: retval = ioctl(fd, AUDIO_DRAIN, 0); if (retval < 0) return retval; break; case SNDCTL_DSP_GETERROR: tmperrinfo = (struct audio_errinfo *)argp; if (tmperrinfo == NULL) { errno = EINVAL; return -1; } memset(tmperrinfo, 0, sizeof(struct audio_errinfo)); if ((retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo)) < 0) return retval; /* * OSS requires that we return counters that are relative to * the last call. We must maintain state here... */ if (ioctl(fd, AUDIO_PERROR, &perrors) != -1) { perrors /= ((tmpinfo.play.precision / NBBY) * tmpinfo.play.channels); tmperrinfo->play_underruns = (perrors / tmpinfo.blocksize) - totalperrors; totalperrors += tmperrinfo->play_underruns; } if (ioctl(fd, AUDIO_RERROR, &rerrors) != -1) { rerrors /= ((tmpinfo.record.precision / NBBY) * tmpinfo.record.channels); tmperrinfo->rec_overruns = (rerrors / tmpinfo.blocksize) - totalrerrors; totalrerrors += tmperrinfo->rec_overruns; } break; case SNDCTL_DSP_COOKEDMODE: /* * NetBSD is always running in "cooked mode" - the kernel * always performs format conversions. */ INTARG = 1; break; case SNDCTL_DSP_POST: /* This call is merely advisory, and may be a nop. */ break; case SNDCTL_DSP_SPEED: AUDIO_INITINFO(&tmpinfo); /* * In Solaris, 0 is used a special value to query the * current rate. This seems useful to support. */ if (INTARG == 0) { retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; retval = ioctl(fd, AUDIO_GETFORMAT, &hwfmt); if (retval < 0) return retval; INTARG = (tmpinfo.mode == AUMODE_RECORD) ? hwfmt.record.sample_rate : hwfmt.play.sample_rate; } /* * Conform to kernel limits. * NetBSD will reject unsupported sample rates, but OSS * applications need to be able to negotiate a supported one. */ if (INTARG < 1000) INTARG = 1000; if (INTARG > 192000) INTARG = 192000; tmpinfo.play.sample_rate = tmpinfo.record.sample_rate = INTARG; retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo); if (retval < 0) return retval; /* FALLTHRU */ case SOUND_PCM_READ_RATE: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; INTARG = GETPRINFO(&tmpinfo, sample_rate); break; case SNDCTL_DSP_STEREO: AUDIO_INITINFO(&tmpinfo); tmpinfo.play.channels = tmpinfo.record.channels = INTARG ? 2 : 1; (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo); retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; INTARG = GETPRINFO(&tmpinfo, channels) - 1; break; case SNDCTL_DSP_GETBLKSIZE: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; setblocksize(fd, &tmpinfo); INTARG = tmpinfo.blocksize; break; case SNDCTL_DSP_SETFMT: AUDIO_INITINFO(&tmpinfo); switch (INTARG) { case AFMT_MU_LAW: tmpinfo.play.precision = tmpinfo.record.precision = 8; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_ULAW; break; case AFMT_A_LAW: tmpinfo.play.precision = tmpinfo.record.precision = 8; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_ALAW; break; case AFMT_U8: tmpinfo.play.precision = tmpinfo.record.precision = 8; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR; break; case AFMT_S8: tmpinfo.play.precision = tmpinfo.record.precision = 8; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR; break; case AFMT_S16_LE: tmpinfo.play.precision = tmpinfo.record.precision = 16; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE; break; case AFMT_S16_BE: tmpinfo.play.precision = tmpinfo.record.precision = 16; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE; break; case AFMT_U16_LE: tmpinfo.play.precision = tmpinfo.record.precision = 16; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_LE; break; case AFMT_U16_BE: tmpinfo.play.precision = tmpinfo.record.precision = 16; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_ULINEAR_BE; break; /* * XXX: When the kernel supports 24-bit LPCM by default, * the 24-bit formats should be handled properly instead * of falling back to 32 bits. */ case AFMT_S24_LE: case AFMT_S32_LE: tmpinfo.play.precision = tmpinfo.record.precision = 32; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_LE; break; case AFMT_S24_BE: case AFMT_S32_BE: tmpinfo.play.precision = tmpinfo.record.precision = 32; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_SLINEAR_BE; break; case AFMT_AC3: tmpinfo.play.precision = tmpinfo.record.precision = 16; tmpinfo.play.encoding = tmpinfo.record.encoding = AUDIO_ENCODING_AC3; break; default: /* * OSSv4 specifies that if an invalid format is chosen * by an application then a sensible format supported * by the hardware is returned. * * In this case, we pick the current hardware format. */ retval = ioctl(fd, AUDIO_GETFORMAT, &hwfmt); if (retval < 0) return retval; retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo); if (retval < 0) return retval; tmpinfo.play.encoding = tmpinfo.record.encoding = (tmpinfo.mode == AUMODE_RECORD) ? hwfmt.record.encoding : hwfmt.play.encoding; tmpinfo.play.precision = tmpinfo.record.precision = (tmpinfo.mode == AUMODE_RECORD) ? hwfmt.record.precision : hwfmt.play.precision ; break; } /* * In the post-kernel-mixer world, assume that any error means * it's fatal rather than an unsupported format being selected. */ retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo); if (retval < 0) return retval; /* FALLTHRU */ case SOUND_PCM_READ_BITS: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; encoding = GETPRINFO(&tmpinfo, encoding); precision = GETPRINFO(&tmpinfo, precision); switch (encoding) { case AUDIO_ENCODING_ULAW: idat = AFMT_MU_LAW; break; case AUDIO_ENCODING_ALAW: idat = AFMT_A_LAW; break; case AUDIO_ENCODING_SLINEAR_LE: if (precision == 32) idat = AFMT_S32_LE; else if (precision == 24) idat = AFMT_S24_LE; else if (precision == 16) idat = AFMT_S16_LE; else idat = AFMT_S8; break; case AUDIO_ENCODING_SLINEAR_BE: if (precision == 32) idat = AFMT_S32_BE; else if (precision == 24) idat = AFMT_S24_BE; else if (precision == 16) idat = AFMT_S16_BE; else idat = AFMT_S8; break; case AUDIO_ENCODING_ULINEAR_LE: if (precision == 16) idat = AFMT_U16_LE; else idat = AFMT_U8; break; case AUDIO_ENCODING_ULINEAR_BE: if (precision == 16) idat = AFMT_U16_BE; else idat = AFMT_U8; break; case AUDIO_ENCODING_ADPCM: idat = AFMT_IMA_ADPCM; break; case AUDIO_ENCODING_AC3: idat = AFMT_AC3; break; } INTARG = idat; break; case SNDCTL_DSP_CHANNELS: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; setchannels(fd, tmpinfo.mode, INTARG); /* FALLTHRU */ case SOUND_PCM_READ_CHANNELS: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; INTARG = GETPRINFO(&tmpinfo, channels); break; case SOUND_PCM_WRITE_FILTER: case SOUND_PCM_READ_FILTER: errno = EINVAL; return -1; /* XXX unimplemented */ case SNDCTL_DSP_SUBDIVIDE: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; setblocksize(fd, &tmpinfo); idat = INTARG; if (idat == 0) idat = tmpinfo.play.buffer_size / tmpinfo.blocksize; idat = (tmpinfo.play.buffer_size / idat) & -4; AUDIO_INITINFO(&tmpinfo); tmpinfo.blocksize = idat; retval = ioctl(fd, AUDIO_SETINFO, &tmpinfo); if (retval < 0) return retval; INTARG = tmpinfo.play.buffer_size / tmpinfo.blocksize; break; case SNDCTL_DSP_SETFRAGMENT: AUDIO_INITINFO(&tmpinfo); idat = INTARG; if ((idat & 0xffff) < 4 || (idat & 0xffff) > 17) { errno = EINVAL; return -1; } tmpinfo.blocksize = 1 << (idat & 0xffff); tmpinfo.hiwat = ((unsigned)idat >> 16) & 0x7fff; if (tmpinfo.hiwat == 0) /* 0 means set to max */ tmpinfo.hiwat = 65536; (void) ioctl(fd, AUDIO_SETINFO, &tmpinfo); retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; u = tmpinfo.blocksize; for(idat = 0; u > 1; idat++, u >>= 1) ; idat |= (tmpinfo.hiwat & 0x7fff) << 16; INTARG = idat; break; case SNDCTL_DSP_GETFMTS: for(idat = 0, tmpenc.index = 0; ioctl(fd, AUDIO_GETENC, &tmpenc) == 0; tmpenc.index++) { switch(tmpenc.encoding) { case AUDIO_ENCODING_ULAW: idat |= AFMT_MU_LAW; break; case AUDIO_ENCODING_ALAW: idat |= AFMT_A_LAW; break; case AUDIO_ENCODING_SLINEAR: idat |= AFMT_S8; break; case AUDIO_ENCODING_SLINEAR_LE: if (tmpenc.precision == 32) idat |= AFMT_S32_LE; else if (tmpenc.precision == 24) idat |= AFMT_S24_LE; else if (tmpenc.precision == 16) idat |= AFMT_S16_LE; else idat |= AFMT_S8; break; case AUDIO_ENCODING_SLINEAR_BE: if (tmpenc.precision == 32) idat |= AFMT_S32_BE; else if (tmpenc.precision == 24) idat |= AFMT_S24_BE; else if (tmpenc.precision == 16) idat |= AFMT_S16_BE; else idat |= AFMT_S8; break; case AUDIO_ENCODING_ULINEAR: idat |= AFMT_U8; break; case AUDIO_ENCODING_ULINEAR_LE: if (tmpenc.precision == 16) idat |= AFMT_U16_LE; else idat |= AFMT_U8; break; case AUDIO_ENCODING_ULINEAR_BE: if (tmpenc.precision == 16) idat |= AFMT_U16_BE; else idat |= AFMT_U8; break; case AUDIO_ENCODING_ADPCM: idat |= AFMT_IMA_ADPCM; break; case AUDIO_ENCODING_AC3: idat |= AFMT_AC3; break; default: break; } } INTARG = idat; break; case SNDCTL_DSP_GETOSPACE: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; setblocksize(fd, &tmpinfo); bufinfo.fragsize = tmpinfo.blocksize; bufinfo.fragments = tmpinfo.hiwat - (tmpinfo.play.seek + tmpinfo.blocksize - 1) / tmpinfo.blocksize; bufinfo.fragstotal = tmpinfo.hiwat; bufinfo.bytes = tmpinfo.hiwat * tmpinfo.blocksize - tmpinfo.play.seek; *(struct audio_buf_info *)argp = bufinfo; break; case SNDCTL_DSP_GETISPACE: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; setblocksize(fd, &tmpinfo); bufinfo.fragsize = tmpinfo.blocksize; bufinfo.fragments = tmpinfo.record.seek / tmpinfo.blocksize; bufinfo.fragstotal = tmpinfo.record.buffer_size / tmpinfo.blocksize; bufinfo.bytes = tmpinfo.record.seek; *(struct audio_buf_info *)argp = bufinfo; break; case SNDCTL_DSP_NONBLOCK: idat = 1; retval = ioctl(fd, FIONBIO, &idat); if (retval < 0) return retval; break; case SNDCTL_DSP_GETCAPS: retval = getcaps(fd, (int *)argp); if (retval < 0) return retval; break; case SNDCTL_DSP_SETTRIGGER: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; AUDIO_INITINFO(&tmpinfo); if (tmpinfo.mode & AUMODE_PLAY) tmpinfo.play.pause = (INTARG & PCM_ENABLE_OUTPUT) == 0; if (tmpinfo.mode & AUMODE_RECORD) tmpinfo.record.pause = (INTARG & PCM_ENABLE_INPUT) == 0; (void)ioctl(fd, AUDIO_SETINFO, &tmpinfo); /* FALLTHRU */ case SNDCTL_DSP_GETTRIGGER: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; idat = 0; if ((tmpinfo.mode & AUMODE_PLAY) && !tmpinfo.play.pause) idat |= PCM_ENABLE_OUTPUT; if ((tmpinfo.mode & AUMODE_RECORD) && !tmpinfo.record.pause) idat |= PCM_ENABLE_INPUT; INTARG = idat; break; case SNDCTL_DSP_GETIPTR: retval = ioctl(fd, AUDIO_GETIOFFS, &tmpoffs); if (retval < 0) return retval; cntinfo.bytes = tmpoffs.samples; cntinfo.blocks = tmpoffs.deltablks; cntinfo.ptr = tmpoffs.offset; *(struct count_info *)argp = cntinfo; break; case SNDCTL_DSP_CURRENT_IPTR: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; /* XXX: 'samples' may wrap */ memset(osscount.filler, 0, sizeof(osscount.filler)); osscount.samples = tmpinfo.record.samples / ((tmpinfo.record.precision / NBBY) * tmpinfo.record.channels); osscount.fifo_samples = tmpinfo.record.seek / ((tmpinfo.record.precision / NBBY) * tmpinfo.record.channels); *(oss_count_t *)argp = osscount; break; case SNDCTL_DSP_GETOPTR: retval = ioctl(fd, AUDIO_GETOOFFS, &tmpoffs); if (retval < 0) return retval; cntinfo.bytes = tmpoffs.samples; cntinfo.blocks = tmpoffs.deltablks; cntinfo.ptr = tmpoffs.offset; *(struct count_info *)argp = cntinfo; break; case SNDCTL_DSP_CURRENT_OPTR: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; /* XXX: 'samples' may wrap */ memset(osscount.filler, 0, sizeof(osscount.filler)); osscount.samples = tmpinfo.play.samples / ((tmpinfo.play.precision / NBBY) * tmpinfo.play.channels); osscount.fifo_samples = tmpinfo.play.seek / ((tmpinfo.play.precision / NBBY) * tmpinfo.play.channels); *(oss_count_t *)argp = osscount; break; case SNDCTL_DSP_SETPLAYVOL: setvol(fd, INTARG, false); /* FALLTHRU */ case SNDCTL_DSP_GETPLAYVOL: retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo); if (retval < 0) return retval; INTARG = getvol(tmpinfo.play.gain, tmpinfo.play.balance); break; case SNDCTL_DSP_SETRECVOL: setvol(fd, INTARG, true); /* FALLTHRU */ case SNDCTL_DSP_GETRECVOL: retval = ioctl(fd, AUDIO_GETINFO, &tmpinfo); if (retval < 0) return retval; INTARG = getvol(tmpinfo.record.gain, tmpinfo.record.balance); break; case SNDCTL_DSP_SKIP: case SNDCTL_DSP_SILENCE: errno = EINVAL; return -1; case SNDCTL_DSP_SETDUPLEX: idat = 1; retval = ioctl(fd, AUDIO_SETFD, &idat); if (retval < 0) return retval; break; case SNDCTL_DSP_GETODELAY: retval = ioctl(fd, AUDIO_GETBUFINFO, &tmpinfo); if (retval < 0) return retval; idat = tmpinfo.play.seek + tmpinfo.blocksize / 2; INTARG = idat; break; case SNDCTL_DSP_PROFILE: /* This gives just a hint to the driver, * implementing it as a NOP is ok */ break; case SNDCTL_DSP_MAPINBUF: case SNDCTL_DSP_MAPOUTBUF: case SNDCTL_DSP_SETSYNCRO: errno = EINVAL; return -1; /* XXX unimplemented */ default: errno = EINVAL; return -1; } return 0; } /* If the NetBSD mixer device should have more than NETBSD_MAXDEVS devices * some will not be available to OSS applications */ #define NETBSD_MAXDEVS 64 struct audiodevinfo { int done; dev_t dev; int16_t devmap[SOUND_MIXER_NRDEVICES], rdevmap[NETBSD_MAXDEVS]; char names[NETBSD_MAXDEVS][MAX_AUDIO_DEV_LEN]; int enum2opaque[NETBSD_MAXDEVS]; u_long devmask, recmask, stereomask; u_long caps; int source; }; static int opaque_to_enum(struct audiodevinfo *di, audio_mixer_name_t *label, int opq) { int i, o; for (i = 0; i < NETBSD_MAXDEVS; i++) { o = di->enum2opaque[i]; if (o == opq) break; if (o == -1 && label != NULL && !strncmp(di->names[i], label->name, sizeof di->names[i])) { di->enum2opaque[i] = opq; break; } } if (i >= NETBSD_MAXDEVS) i = -1; /*printf("opq_to_enum %s %d -> %d\n", label->name, opq, i);*/ return (i); } static int enum_to_ord(struct audiodevinfo *di, int enm) { if (enm >= NETBSD_MAXDEVS) return (-1); /*printf("enum_to_ord %d -> %d\n", enm, di->enum2opaque[enm]);*/ return (di->enum2opaque[enm]); } static int enum_to_mask(struct audiodevinfo *di, int enm) { int m; if (enm >= NETBSD_MAXDEVS) return (0); m = di->enum2opaque[enm]; if (m == -1) m = 0; /*printf("enum_to_mask %d -> %d\n", enm, di->enum2opaque[enm]);*/ return (m); } /* * Collect the audio device information to allow faster * emulation of the OSSv3 mixer ioctls. Cache the information * to eliminate the overhead of repeating all the ioctls needed * to collect the information. */ static struct audiodevinfo * getdevinfo(int fd) { mixer_devinfo_t mi; int i, j, e; static struct { const char *name; int code; } *dp, devs[] = { { AudioNmicrophone, SOUND_MIXER_MIC }, { AudioNline, SOUND_MIXER_LINE }, { AudioNcd, SOUND_MIXER_CD }, { AudioNdac, SOUND_MIXER_PCM }, { AudioNaux, SOUND_MIXER_LINE1 }, { AudioNrecord, SOUND_MIXER_IMIX }, { AudioNmaster, SOUND_MIXER_VOLUME }, { AudioNtreble, SOUND_MIXER_TREBLE }, { AudioNbass, SOUND_MIXER_BASS }, { AudioNspeaker, SOUND_MIXER_SPEAKER }, /* { AudioNheadphone, ?? },*/ { AudioNoutput, SOUND_MIXER_OGAIN }, { AudioNinput, SOUND_MIXER_IGAIN }, /* { AudioNmaster, SOUND_MIXER_SPEAKER },*/ /* { AudioNstereo, ?? },*/ /* { AudioNmono, ?? },*/ { AudioNfmsynth, SOUND_MIXER_SYNTH }, /* { AudioNwave, SOUND_MIXER_PCM },*/ { AudioNmidi, SOUND_MIXER_SYNTH }, /* { AudioNmixerout, ?? },*/ { 0, -1 } }; static struct audiodevinfo devcache = { .done = 0 }; struct audiodevinfo *di = &devcache; struct stat sb; size_t mlen, dlen; /* Figure out what device it is so we can check if the * cached data is valid. */ if (fstat(fd, &sb) < 0) return 0; if (di->done && di->dev == sb.st_dev) return di; di->done = 1; di->dev = sb.st_dev; di->devmask = 0; di->recmask = 0; di->stereomask = 0; di->source = ~0; di->caps = 0; for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) di->devmap[i] = -1; for(i = 0; i < NETBSD_MAXDEVS; i++) { di->rdevmap[i] = -1; di->names[i][0] = '\0'; di->enum2opaque[i] = -1; } for(i = 0; i < NETBSD_MAXDEVS; i++) { mi.index = i; if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0) break; switch(mi.type) { case AUDIO_MIXER_VALUE: for(dp = devs; dp->name; dp++) { if (strcmp(dp->name, mi.label.name) == 0) break; dlen = strlen(dp->name); mlen = strlen(mi.label.name); if (dlen < mlen && mi.label.name[mlen-dlen-1] == '.' && strcmp(dp->name, mi.label.name + mlen - dlen) == 0) break; } if (dp->code >= 0) { di->devmap[dp->code] = i; di->rdevmap[i] = dp->code; di->devmask |= 1 << dp->code; if (mi.un.v.num_channels == 2) di->stereomask |= 1 << dp->code; strlcpy(di->names[i], mi.label.name, sizeof di->names[i]); } break; } } for(i = 0; i < NETBSD_MAXDEVS; i++) { mi.index = i; if (ioctl(fd, AUDIO_MIXER_DEVINFO, &mi) < 0) break; if (strcmp(mi.label.name, AudioNsource) != 0) continue; di->source = i; switch(mi.type) { case AUDIO_MIXER_ENUM: for(j = 0; j < mi.un.e.num_mem; j++) { e = opaque_to_enum(di, &mi.un.e.member[j].label, mi.un.e.member[j].ord); if (e >= 0) di->recmask |= 1 << di->rdevmap[e]; } di->caps = SOUND_CAP_EXCL_INPUT; break; case AUDIO_MIXER_SET: for(j = 0; j < mi.un.s.num_mem; j++) { e = opaque_to_enum(di, &mi.un.s.member[j].label, mi.un.s.member[j].mask); if (e >= 0) di->recmask |= 1 << di->rdevmap[e]; } break; } } return di; } static int mixer_oss3_ioctl(int fd, unsigned long com, void *argp) { struct audiodevinfo *di; struct mixer_info *omi; struct audio_device adev; mixer_ctrl_t mc; u_long idat, n; int i; int retval; int l, r, error, e; idat = 0; di = getdevinfo(fd); if (di == 0) return -1; switch (com) { case OSS_GETVERSION: idat = SOUND_VERSION; break; case SOUND_MIXER_INFO: case SOUND_OLD_MIXER_INFO: error = ioctl(fd, AUDIO_GETDEV, &adev); if (error) return (error); omi = argp; if (com == SOUND_MIXER_INFO) omi->modify_counter = 1; strlcpy(omi->id, adev.name, sizeof omi->id); strlcpy(omi->name, adev.name, sizeof omi->name); return 0; case SOUND_MIXER_READ_RECSRC: if (di->source == -1) { errno = EINVAL; return -1; } mc.dev = di->source; if (di->caps & SOUND_CAP_EXCL_INPUT) { mc.type = AUDIO_MIXER_ENUM; retval = ioctl(fd, AUDIO_MIXER_READ, &mc); if (retval < 0) return retval; e = opaque_to_enum(di, NULL, mc.un.ord); if (e >= 0) idat = 1 << di->rdevmap[e]; } else { mc.type = AUDIO_MIXER_SET; retval = ioctl(fd, AUDIO_MIXER_READ, &mc); if (retval < 0) return retval; e = opaque_to_enum(di, NULL, mc.un.mask); if (e >= 0) idat = 1 << di->rdevmap[e]; } break; case SOUND_MIXER_READ_DEVMASK: idat = di->devmask; break; case SOUND_MIXER_READ_RECMASK: idat = di->recmask; break; case SOUND_MIXER_READ_STEREODEVS: idat = di->stereomask; break; case SOUND_MIXER_READ_CAPS: idat = di->caps; break; case SOUND_MIXER_WRITE_RECSRC: case SOUND_MIXER_WRITE_R_RECSRC: if (di->source == -1) { errno = EINVAL; return -1; } mc.dev = di->source; idat = INTARG; if (di->caps & SOUND_CAP_EXCL_INPUT) { mc.type = AUDIO_MIXER_ENUM; for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) if (idat & (1 << i)) break; if (i >= SOUND_MIXER_NRDEVICES || di->devmap[i] == -1) { errno = EINVAL; return -1; } mc.un.ord = enum_to_ord(di, di->devmap[i]); } else { mc.type = AUDIO_MIXER_SET; mc.un.mask = 0; for(i = 0; i < SOUND_MIXER_NRDEVICES; i++) { if (idat & (1 << i)) { if (di->devmap[i] == -1) { errno = EINVAL; return -1; } mc.un.mask |= enum_to_mask(di, di->devmap[i]); } } } return ioctl(fd, AUDIO_MIXER_WRITE, &mc); default: if (MIXER_READ(SOUND_MIXER_FIRST) <= com && com < MIXER_READ(SOUND_MIXER_NRDEVICES)) { n = GET_DEV(com); if (di->devmap[n] == -1) { errno = EINVAL; return -1; } mc.dev = di->devmap[n]; mc.type = AUDIO_MIXER_VALUE; doread: mc.un.value.num_channels = di->stereomask & (1 << (u_int)n) ? 2 : 1; retval = ioctl(fd, AUDIO_MIXER_READ, &mc); if (retval < 0) return retval; if (mc.type != AUDIO_MIXER_VALUE) { errno = EINVAL; return -1; } if (mc.un.value.num_channels != 2) { l = r = mc.un.value.level[AUDIO_MIXER_LEVEL_MONO]; } else { l = mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT]; r = mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT]; } idat = TO_OSSVOL(l) | (TO_OSSVOL(r) << 8); break; } else if ((MIXER_WRITE_R(SOUND_MIXER_FIRST) <= com && com < MIXER_WRITE_R(SOUND_MIXER_NRDEVICES)) || (MIXER_WRITE(SOUND_MIXER_FIRST) <= com && com < MIXER_WRITE(SOUND_MIXER_NRDEVICES))) { n = GET_DEV(com); if (di->devmap[n] == -1) { errno = EINVAL; return -1; } idat = INTARG; l = FROM_OSSVOL((u_int)idat & 0xff); r = FROM_OSSVOL(((u_int)idat >> 8) & 0xff); mc.dev = di->devmap[n]; mc.type = AUDIO_MIXER_VALUE; if (di->stereomask & (1 << (u_int)n)) { mc.un.value.num_channels = 2; mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l; mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r; } else { mc.un.value.num_channels = 1; mc.un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l + r) / 2; } retval = ioctl(fd, AUDIO_MIXER_WRITE, &mc); if (retval < 0) return retval; if (MIXER_WRITE(SOUND_MIXER_FIRST) <= com && com < MIXER_WRITE(SOUND_MIXER_NRDEVICES)) return 0; goto doread; } else { errno = EINVAL; return -1; } } INTARG = (int)idat; return 0; } static int mixer_oss4_ioctl(int fd, unsigned long com, void *argp) { oss_audioinfo *tmpai; oss_card_info *cardinfo; oss_mixext *ext; oss_mixext_root root; oss_mixer_enuminfo *ei; oss_mixer_value *mv; oss_mixerinfo *mi; oss_sysinfo sysinfo; dev_t devno; struct stat tmpstat; struct audio_device dev; struct audio_format_query fmtq; struct mixer_devinfo mdi; struct mixer_ctrl mc; char devname[32]; size_t len; int newfd = -1, tmperrno; int i, noffs; int retval; /* * Note: it is difficult to translate the NetBSD concept of a "set" * mixer control type to the OSSv4 API, as far as I can tell. * * This means they are treated like enums, i.e. only one entry in the * set can be selected at a time. */ switch (com) { case SNDCTL_AUDIOINFO: /* * SNDCTL_AUDIOINFO_EX is intended for underlying hardware devices * that are to be opened in "exclusive mode" (bypassing the normal * kernel mixer for exclusive control). NetBSD does not support * bypassing the kernel mixer, so it's an alias of SNDCTL_AUDIOINFO. */ case SNDCTL_AUDIOINFO_EX: case SNDCTL_ENGINEINFO: devno = 0; tmpai = (struct oss_audioinfo*)argp; if (tmpai == NULL) { errno = EINVAL; return -1; } /* * If the input device is -1, guess the device related to * the open mixer device. */ if (tmpai->dev < 0) { fstat(fd, &tmpstat); if ((tmpstat.st_rdev & 0xff00) == 0x2a00) devno = tmpstat.st_rdev & 0xff; if (devno >= 0x80) tmpai->dev = devno & 0x7f; } if (tmpai->dev < 0) tmpai->dev = 0; snprintf(tmpai->devnode, sizeof(tmpai->devnode), "/dev/audio%d", tmpai->dev); if ((newfd = open(tmpai->devnode, O_WRONLY)) < 0) { if ((newfd = open(tmpai->devnode, O_RDONLY)) < 0) { return newfd; } } retval = ioctl(newfd, AUDIO_GETDEV, &dev); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } if (getcaps(newfd, &tmpai->caps) < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } snprintf(tmpai->name, sizeof(tmpai->name), "%s %s", dev.name, dev.version); tmpai->busy = 0; tmpai->pid = -1; ioctl(newfd, SNDCTL_DSP_GETFMTS, &tmpai->iformats); tmpai->oformats = tmpai->iformats; tmpai->magic = -1; /* reserved for "internal use" */ memset(tmpai->cmd, 0, sizeof(tmpai->cmd)); tmpai->card_number = -1; memset(tmpai->song_name, 0, sizeof(tmpai->song_name)); memset(tmpai->label, 0, sizeof(tmpai->label)); tmpai->port_number = 0; tmpai->mixer_dev = tmpai->dev; tmpai->legacy_device = tmpai->dev; tmpai->enabled = 1; tmpai->flags = -1; /* reserved for "future versions" */ tmpai->min_rate = 1000; tmpai->max_rate = 192000; tmpai->nrates = 0; tmpai->min_channels = 1; tmpai->max_channels = 2; for (fmtq.index = 0; ioctl(newfd, AUDIO_QUERYFORMAT, &fmtq) != -1; ++fmtq.index) { if (fmtq.fmt.channels > (unsigned)tmpai->max_channels) tmpai->max_channels = fmtq.fmt.channels; } tmpai->binding = -1; /* reserved for "future versions" */ tmpai->rate_source = -1; /* * 'handle' is supposed to be globally unique. The closest * we have to that is probably device nodes. */ strlcpy(tmpai->handle, tmpai->devnode, sizeof(tmpai->handle)); tmpai->next_play_engine = 0; tmpai->next_rec_engine = 0; argp = tmpai; close(newfd); break; case SNDCTL_CARDINFO: cardinfo = (oss_card_info *)argp; if (cardinfo == NULL) { errno = EINVAL; return -1; } if (cardinfo->card != -1) { snprintf(devname, sizeof(devname), "/dev/audio%d", cardinfo->card); newfd = open(devname, O_RDONLY); if (newfd < 0) return newfd; } else { newfd = fd; } retval = ioctl(newfd, AUDIO_GETDEV, &dev); tmperrno = errno; if (newfd != fd) close(newfd); if (retval < 0) { errno = tmperrno; return retval; } strlcpy(cardinfo->shortname, dev.name, sizeof(cardinfo->shortname)); snprintf(cardinfo->longname, sizeof(cardinfo->longname), "%s %s %s", dev.name, dev.version, dev.config); memset(cardinfo->hw_info, 0, sizeof(cardinfo->hw_info)); /* * OSSv4 does not document this ioctl, and claims it should * not be used by applications and is provided for "utiltiy * programs included in OSS". We follow the Solaris * implementation (which is documented) and leave these fields * unset. */ cardinfo->flags = 0; cardinfo->intr_count = 0; cardinfo->ack_count = 0; break; case SNDCTL_SYSINFO: memset(&sysinfo, 0, sizeof(sysinfo)); strlcpy(sysinfo.product, "OSS/NetBSD", sizeof(sysinfo.product)); strlcpy(sysinfo.version, "4.01", sizeof(sysinfo.version)); strlcpy(sysinfo.license, "BSD", sizeof(sysinfo.license)); sysinfo.versionnum = SOUND_VERSION; sysinfo.numaudios = sysinfo.numcards = getaudiocount(); sysinfo.numaudioengines = 1; sysinfo.numsynths = 1; sysinfo.nummidis = -1; sysinfo.numtimers = -1; sysinfo.nummixers = getmixercount(); *(struct oss_sysinfo *)argp = sysinfo; break; case SNDCTL_MIXERINFO: mi = (oss_mixerinfo *)argp; if (mi == NULL) { errno = EINVAL; return -1; } snprintf(devname, sizeof(devname), "/dev/mixer%d", mi->dev); if ((newfd = open(devname, O_RDONLY)) < 0) return newfd; retval = ioctl(newfd, AUDIO_GETDEV, &dev); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } strlcpy(mi->id, devname, sizeof(mi->id)); snprintf(mi->name, sizeof(mi->name), "%s %s", dev.name, dev.version); mi->card_number = mi->dev; mi->port_number = 0; mi->magic = 0; mi->enabled = 1; mi->caps = 0; mi->flags = 0; mi->nrext = getmixercontrolcount(newfd) + 1; mi->priority = UCHAR_MAX - mi->dev; strlcpy(mi->devnode, devname, sizeof(mi->devnode)); mi->legacy_device = mi->dev; break; case SNDCTL_MIX_DESCRIPTION: /* No description available. */ errno = ENOSYS; return -1; case SNDCTL_MIX_NRMIX: INTARG = getmixercount(); break; case SNDCTL_MIX_NREXT: snprintf(devname, sizeof(devname), "/dev/mixer%d", INTARG); if ((newfd = open(devname, O_RDONLY)) < 0) return newfd; INTARG = getmixercontrolcount(newfd) + 1; close(newfd); break; case SNDCTL_MIX_EXTINFO: ext = (oss_mixext *)argp; snprintf(devname, sizeof(devname), "/dev/mixer%d", ext->dev); if ((newfd = open(devname, O_RDONLY)) < 0) return newfd; if (ext->ctrl == 0) { /* * NetBSD has no concept of a "root mixer control", but * OSSv4 requires one to work. We fake one at 0 and * simply add 1 to all real control indexes. */ retval = ioctl(newfd, AUDIO_GETDEV, &dev); tmperrno = errno; close(newfd); if (retval < 0) { errno = tmperrno; return -1; } memset(&root, 0, sizeof(root)); strlcpy(root.id, devname, sizeof(root.id)); snprintf(root.name, sizeof(root.name), "%s %s", dev.name, dev.version); strlcpy(ext->id, devname, sizeof(ext->id)); snprintf(ext->extname, sizeof(ext->extname), "%s %s", dev.name, dev.version); strlcpy(ext->extname, "root", sizeof(ext->extname)); ext->type = MIXT_DEVROOT; ext->minvalue = 0; ext->maxvalue = 0; ext->flags = 0; ext->parent = -1; ext->control_no = -1; ext->update_counter = 0; ext->rgbcolor = 0; memcpy(&ext->data, &root, sizeof(root) > sizeof(ext->data) ? sizeof(ext->data) : sizeof(root)); return 0; } mdi.index = ext->ctrl - 1; retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } ext->flags = MIXF_READABLE | MIXF_WRITEABLE | MIXF_POLL; ext->parent = mdi.mixer_class + 1; strlcpy(ext->id, mdi.label.name, sizeof(ext->id)); strlcpy(ext->extname, mdi.label.name, sizeof(ext->extname)); len = strlen(ext->extname); memset(ext->data, 0, sizeof(ext->data)); ext->control_no = -1; ext->update_counter = 0; ext->rgbcolor = 0; switch (mdi.type) { case AUDIO_MIXER_CLASS: ext->type = MIXT_GROUP; ext->parent = 0; ext->minvalue = 0; ext->maxvalue = 0; break; case AUDIO_MIXER_ENUM: ext->maxvalue = mdi.un.e.num_mem; ext->minvalue = 0; for (i = 0; i < mdi.un.e.num_mem; ++i) { ext->enum_present[i / 8] |= (1 << (i % 8)); } if (mdi.un.e.num_mem == 2) { if (!strcmp(mdi.un.e.member[0].label.name, AudioNoff) && !strcmp(mdi.un.e.member[1].label.name, AudioNon)) { ext->type = MIXT_MUTE; } else { ext->type = MIXT_ENUM; } } else { ext->type = MIXT_ENUM; } break; case AUDIO_MIXER_SET: ext->maxvalue = mdi.un.s.num_mem; ext->minvalue = 0; #ifdef notyet /* * XXX: This is actually the correct type for "set" * controls, but it seems no real world software * supports it. The only documentation exists in * the OSSv4 headers and describes it as "reserved * for Sun's implementation". */ ext->type = MIXT_ENUM_MULTI; #else ext->type = MIXT_ENUM; #endif for (i = 0; i < mdi.un.s.num_mem; ++i) { ext->enum_present[i / 8] |= (1 << (i % 8)); } break; case AUDIO_MIXER_VALUE: ext->maxvalue = UCHAR_MAX + 1; ext->minvalue = 0; if (mdi.un.v.num_channels == 2) { ext->type = MIXT_STEREOSLIDER; } else { ext->type = MIXT_MONOSLIDER; } break; } close(newfd); break; case SNDCTL_MIX_ENUMINFO: ei = (oss_mixer_enuminfo *)argp; if (ei == NULL) { errno = EINVAL; return -1; } if (ei->ctrl == 0) { errno = EINVAL; return -1; } snprintf(devname, sizeof(devname), "/dev/mixer%d", ei->dev); if ((newfd = open(devname, O_RDONLY)) < 0) return newfd; mdi.index = ei->ctrl - 1; retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi); tmperrno = errno; close(newfd); if (retval < 0) { errno = tmperrno; return retval; } ei->version = 0; switch (mdi.type) { case AUDIO_MIXER_ENUM: ei->nvalues = mdi.un.e.num_mem; noffs = 0; for (i = 0; i < ei->nvalues; ++i) { ei->strindex[i] = noffs; len = strlen(mdi.un.e.member[i].label.name) + 1; if ((noffs + len) >= sizeof(ei->strings)) { errno = ENOMEM; return -1; } memcpy(ei->strings + noffs, mdi.un.e.member[i].label.name, len); noffs += len; } break; case AUDIO_MIXER_SET: ei->nvalues = mdi.un.s.num_mem; noffs = 0; for (i = 0; i < ei->nvalues; ++i) { ei->strindex[i] = noffs; len = strlen(mdi.un.s.member[i].label.name) + 1; if ((noffs + len) >= sizeof(ei->strings)) { errno = ENOMEM; return -1; } memcpy(ei->strings + noffs, mdi.un.s.member[i].label.name, len); noffs += len; } break; default: errno = EINVAL; return -1; } break; case SNDCTL_MIX_WRITE: mv = (oss_mixer_value *)argp; if (mv == NULL) { errno = EINVAL; return -1; } if (mv->ctrl == 0) { errno = EINVAL; return -1; } snprintf(devname, sizeof(devname), "/dev/mixer%d", mv->dev); if ((newfd = open(devname, O_RDWR)) < 0) return newfd; mdi.index = mc.dev = mv->ctrl - 1; retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } switch (mdi.type) { case AUDIO_MIXER_ENUM: if (mv->value >= mdi.un.e.num_mem) { close(newfd); errno = EINVAL; return -1; } mc.un.ord = mdi.un.e.member[mv->value].ord; break; case AUDIO_MIXER_SET: if (mv->value >= mdi.un.s.num_mem) { close(newfd); errno = EINVAL; return -1; } #ifdef notyet mc.un.mask = 0; for (i = 0; i < mdi.un.s.num_mem; ++i) { if (mv->value & (1 << i)) { mc.un.mask |= mdi.un.s.member[mv->value].mask; } } #else mc.un.mask = mdi.un.s.member[mv->value].mask; #endif break; case AUDIO_MIXER_VALUE: if (mdi.un.v.num_channels != 2) { for (i = 0; i < mdi.un.v.num_channels; ++i) { mc.un.value.level[i] = mv->value; } } else { mc.un.value.level[AUDIO_MIXER_LEVEL_LEFT] = (mv->value >> 0) & 0xFF; mc.un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = (mv->value >> 8) & 0xFF; } break; } retval = ioctl(newfd, AUDIO_MIXER_WRITE, &mc); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } close(newfd); break; case SNDCTL_MIX_READ: mv = (oss_mixer_value *)argp; if (mv == NULL) { errno = EINVAL; return -1; } if (mv->ctrl == 0) { errno = EINVAL; return -1; } snprintf(devname, sizeof(devname), "/dev/mixer%d", mv->dev); if ((newfd = open(devname, O_RDWR)) < 0) return newfd; mdi.index = mc.dev = (mv->ctrl - 1); retval = ioctl(newfd, AUDIO_MIXER_DEVINFO, &mdi); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } mc.dev = mdi.index; retval = ioctl(newfd, AUDIO_MIXER_READ, &mc); if (retval < 0) { tmperrno = errno; close(newfd); errno = tmperrno; return retval; } close(newfd); mv->value = 0; switch (mdi.type) { case AUDIO_MIXER_ENUM: for (i = 0; i < mdi.un.e.num_mem; ++i) { if (mc.un.ord == mdi.un.e.member[i].ord) { mv->value = i; break; } } break; case AUDIO_MIXER_SET: for (i = 0; i < mdi.un.s.num_mem; ++i) { #ifdef notyet if (mc.un.mask & mdi.un.s.member[i].mask) mv->value |= (1 << i); #else if (mc.un.mask == mdi.un.s.member[i].mask) { mv->value = i; break; } #endif } break; case AUDIO_MIXER_VALUE: if (mdi.un.v.num_channels != 2) { mv->value = mc.un.value.level[0]; } else { mv->value = \ ((mc.un.value.level[1] & 0xFF) << 8) | ((mc.un.value.level[0] & 0xFF) << 0); } break; default: errno = EINVAL; return -1; } break; default: errno = EINVAL; return -1; } return 0; } static int global_oss4_ioctl(int fd, unsigned long com, void *argp) { int retval = 0; switch (com) { /* * These ioctls were added in OSSv4 with the idea that * applications could apply strings to audio devices to * display what they are using them for (e.g. with song * names) in mixer applications. In practice, the popular * implementations of the API in FreeBSD and Solaris treat * these as a no-op and return EINVAL, and no software in the * wild seems to use them. */ case SNDCTL_SETSONG: case SNDCTL_GETSONG: case SNDCTL_SETNAME: case SNDCTL_SETLABEL: case SNDCTL_GETLABEL: errno = EINVAL; retval = -1; break; default: errno = EINVAL; retval = -1; break; } return retval; } static int getcaps(int fd, int *out) { int props, caps; if (ioctl(fd, AUDIO_GETPROPS, &props) < 0) return -1; caps = DSP_CAP_TRIGGER; if (props & AUDIO_PROP_FULLDUPLEX) caps |= DSP_CAP_DUPLEX; if (props & AUDIO_PROP_MMAP) caps |= DSP_CAP_MMAP; if (props & AUDIO_PROP_CAPTURE) caps |= PCM_CAP_INPUT; if (props & AUDIO_PROP_PLAYBACK) caps |= PCM_CAP_OUTPUT; *out = caps; return 0; } static int getaudiocount(void) { char devname[32]; int ndevs = 0; int tmpfd; int tmperrno = errno; do { snprintf(devname, sizeof(devname), "/dev/audio%d", ndevs); if ((tmpfd = open(devname, O_RDONLY)) != -1 || (tmpfd = open(devname, O_WRONLY)) != -1) { ndevs++; close(tmpfd); } } while (tmpfd != -1); errno = tmperrno; return ndevs; } static int getmixercount(void) { char devname[32]; int ndevs = 0; int tmpfd; int tmperrno = errno; do { snprintf(devname, sizeof(devname), "/dev/mixer%d", ndevs); if ((tmpfd = open(devname, O_RDONLY)) != -1) { ndevs++; close(tmpfd); } } while (tmpfd != -1); errno = tmperrno; return ndevs; } static int getmixercontrolcount(int fd) { struct mixer_devinfo mdi; int ndevs = 0; do { mdi.index = ndevs++; } while (ioctl(fd, AUDIO_MIXER_DEVINFO, &mdi) != -1); return ndevs > 0 ? ndevs - 1 : 0; } static int getvol(u_int gain, u_char balance) { u_int l, r; if (balance == AUDIO_MID_BALANCE) { l = r = gain; } else if (balance < AUDIO_MID_BALANCE) { l = gain; r = (balance * gain) / AUDIO_MID_BALANCE; } else { r = gain; l = ((AUDIO_RIGHT_BALANCE - balance) * gain) / AUDIO_MID_BALANCE; } return TO_OSSVOL(l) | (TO_OSSVOL(r) << 8); } static void setvol(int fd, int volume, bool record) { u_int lgain, rgain; struct audio_info tmpinfo; struct audio_prinfo *prinfo; AUDIO_INITINFO(&tmpinfo); prinfo = record ? &tmpinfo.record : &tmpinfo.play; lgain = FROM_OSSVOL((volume >> 0) & 0xff); rgain = FROM_OSSVOL((volume >> 8) & 0xff); if (lgain == rgain) { prinfo->gain = lgain; prinfo->balance = AUDIO_MID_BALANCE; } else if (lgain < rgain) { prinfo->gain = rgain; prinfo->balance = AUDIO_RIGHT_BALANCE - (AUDIO_MID_BALANCE * lgain) / rgain; } else { prinfo->gain = lgain; prinfo->balance = (AUDIO_MID_BALANCE * rgain) / lgain; } (void)ioctl(fd, AUDIO_SETINFO, &tmpinfo); } /* * When AUDIO_SETINFO fails to set a channel count, the application's chosen * number is out of range of what the kernel allows. * * When this happens, we use the current hardware settings. This is just in * case an application is abusing SNDCTL_DSP_CHANNELS - OSSv4 always sets and * returns a reasonable value, even if it wasn't what the user requested. * * Solaris guarantees this behaviour if nchannels = 0. * * XXX: If a device is opened for both playback and recording, and supports * fewer channels for recording than playback, applications that do both will * behave very strangely. OSS doesn't allow for reporting separate channel * counts for recording and playback. This could be worked around by always * mixing recorded data up to the same number of channels as is being used * for playback. */ static void setchannels(int fd, int mode, int nchannels) { struct audio_info tmpinfo, hwfmt; if (ioctl(fd, AUDIO_GETFORMAT, &hwfmt) < 0) { errno = 0; hwfmt.record.channels = hwfmt.play.channels = 2; } if (mode & AUMODE_PLAY) { AUDIO_INITINFO(&tmpinfo); tmpinfo.play.channels = nchannels; if (ioctl(fd, AUDIO_SETINFO, &tmpinfo) < 0) { errno = 0; AUDIO_INITINFO(&tmpinfo); tmpinfo.play.channels = hwfmt.play.channels; (void)ioctl(fd, AUDIO_SETINFO, &tmpinfo); } } if (mode & AUMODE_RECORD) { AUDIO_INITINFO(&tmpinfo); tmpinfo.record.channels = nchannels; if (ioctl(fd, AUDIO_SETINFO, &tmpinfo) < 0) { errno = 0; AUDIO_INITINFO(&tmpinfo); tmpinfo.record.channels = hwfmt.record.channels; (void)ioctl(fd, AUDIO_SETINFO, &tmpinfo); } } } /* * Check that the blocksize is a power of 2 as OSS wants. * If not, set it to be. */ static void setblocksize(int fd, struct audio_info *info) { struct audio_info set; size_t s; if (info->blocksize & (info->blocksize-1)) { for(s = 32; s < info->blocksize; s <<= 1) ; AUDIO_INITINFO(&set); set.blocksize = s; ioctl(fd, AUDIO_SETINFO, &set); ioctl(fd, AUDIO_GETBUFINFO, info); } }