From d367768476fa515e65839eaa7427ac490c9316ed Mon Sep 17 00:00:00 2001 From: jsg Date: Fri, 31 Dec 2021 23:19:50 +0000 Subject: [PATCH] spelling ok ratchov@ --- sys/dev/usb/uaudio.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sys/dev/usb/uaudio.c b/sys/dev/usb/uaudio.c index 910071ac936..6e93d6e9acf 100644 --- a/sys/dev/usb/uaudio.c +++ b/sys/dev/usb/uaudio.c @@ -1,4 +1,4 @@ -/* $OpenBSD: uaudio.c,v 1.162 2021/11/22 10:17:14 mglocker Exp $ */ +/* $OpenBSD: uaudio.c,v 1.163 2021/12/31 23:19:50 jsg Exp $ */ /* * Copyright (c) 2018 Alexandre Ratchov * @@ -164,7 +164,7 @@ /* * Samples-per-frame are fractions. UAC v2.0 requires the denominator to - * be multiple of 2^16, as used in the sync pipe. On the othe hand, to + * be multiple of 2^16, as used in the sync pipe. On the other hand, to * represent sample-per-frame of all rates we support, we need the * denominator to be such that (rate / 1000) can be represented exactly, * 80 works. So we use the least common multiplier of both. @@ -178,7 +178,7 @@ #define UAUDIO_NAME_REC "record" /* - * read/write pointers for secure sequencial access of binary data, + * read/write pointers for secure sequential access of binary data, * ex. usb descriptors, tables and alike. Bytes are read using the * read pointer up to the write pointer. */ @@ -628,7 +628,7 @@ uaudio_tname(struct uaudio_softc *sc, unsigned int type, int isout) if (hi == 1) return isout ? UAUDIO_NAME_REC : UAUDIO_NAME_PLAY; - /* if theres only one input (output) use "input" ("output") */ + /* if there is only one input (output) use "input" ("output") */ if (isout) { if (sc->nout == 1) return "output"; @@ -789,7 +789,7 @@ uaudio_ranges_add(struct uaudio_ranges *r, int min, int max, int res) for (pe = &r->el; (e = *pe) != NULL; pe = &e->next) { if (min <= e->max && max >= e->min) { - DPRINTF("%s: overlaping ranges\n", __func__); + DPRINTF("%s: overlapping ranges\n", __func__); return; } if (min < e->max) @@ -1115,7 +1115,7 @@ uaudio_feature_addent(struct uaudio_softc *sc, {"gain", UAUDIO_MIX_NUM, UAUDIO_REQSEL_GAIN}, {"gainpad", UAUDIO_MIX_SW, UAUDIO_REQSEL_GAINPAD}, {"phase", UAUDIO_MIX_SW, UAUDIO_REQSEL_PHASEINV}, - {NULL, -1, -1}, /* undeflow */ + {NULL, -1, -1}, /* underflow */ {NULL, -1, -1} /* overflow */ }; struct uaudio_mixent *m, *i, **pi; @@ -1415,7 +1415,7 @@ uaudio_process_unit(struct uaudio_softc *sc, break; case UAUDIO_AC_SELECTOR: /* - * Selectors are extreamly rare, so not supported yet. + * Selectors are extremely rare, so not supported yet. */ if (!uaudio_process_srcs(sc, u, units, &p)) return 0; @@ -1591,7 +1591,7 @@ uaudio_setname_srcs(struct uaudio_softc *sc, struct uaudio_unit *u, char *name) /* * Set the name of the given unit by using both its source and * destination units. This is naming scheme is only useful to units - * that would have ambigous names if only sources or only destination + * that would have ambiguous names if only sources or only destination * were used. */ void @@ -2081,7 +2081,7 @@ uaudio_process_ac(struct uaudio_softc *sc, struct uaudio_blob *p, int ifnum) unsigned int type, subtype, id; char *name, val; - DPRINTF("%s: ifnum = %d, %zd bytes to processs\n", __func__, + DPRINTF("%s: ifnum = %d, %zd bytes to process\n", __func__, ifnum, p->wptr - p->rptr); sc->ctl_ifnum = ifnum; @@ -2291,7 +2291,7 @@ uaudio_process_ac(struct uaudio_softc *sc, struct uaudio_blob *p, int ifnum) } /* - * Parse endpoint descriptor with the following fromat: + * Parse endpoint descriptor with the following format: * * For playback there's a output data endpoint, of the * following types: @@ -2370,7 +2370,7 @@ uaudio_process_as_ep(struct uaudio_softc *sc, /* * For each AS interface setting, there's a single data * endpoint and an optional feedback endpoint. The - * synchonization type is non-zero and must be set in the data + * synchronization type is non-zero and must be set in the data * endpoints. * * However, the isoc sync type field of the attribute can't be @@ -2447,7 +2447,7 @@ uaudio_process_as_general(struct uaudio_softc *sc, /* * Parse AS format descriptor: we support only "Type 1" formats, aka * PCM. Other formats are not really audio, they are data-only - * interfaces that we don't wan't to support: ethernet is much better + * interfaces that we don't want to support: ethernet is much better * for raw data transfers. * * XXX: handle ieee 754 32-bit floating point formats. @@ -2928,7 +2928,7 @@ uaudio_stream_open(struct uaudio_softc *sc, int dir, * block boundary, which is propagated to upper layers. In the * worst case, we schedule only frames of spf_max samples, but * the device returns only frames of spf_min samples; in this - * case the amount actually transfered is at least: + * case the amount actually transferred is at least: * * min_blksz = blksz / spf_max * spf_min * @@ -3282,7 +3282,7 @@ uaudio_pdata_xfer(struct uaudio_softc *sc) /* * We accept short transfers because in case of babble/stale frames - * the tranfer will be short + * the transfer will be short */ usbd_setup_isoc_xfer(xfer->usb_xfer, s->data_pipe, sc, xfer->sizes, xfer->nframes, @@ -4021,7 +4021,7 @@ uaudio_set_params(void *self, int setmode, int usemode, } /* - * Recalculate rate index, because the choosen parameters + * Recalculate rate index, because the chosen parameters * may not support the requested one */ rateindex = uaudio_rates_indexof(uaudio_getrates(sc, p), rate); -- 2.20.1