Module Name: src
Committed By: isaki
Date: Mon Apr 29 09:30:18 UTC 2019
Modified Files:
src/sys/dev/pci [isaki-audio2]: sv.c
Log Message:
Adapt to audio2.
- Drop INDEPENDENT property. Both play and rec seems to share the
sample rate (however, they require different calculations?).
- XXX It's better to modify frequency list more strictly.
- Remove obsoleted and empty methods.
To generate a diff of this commit:
cvs rdiff -u -r1.54.2.1 -r1.54.2.2 src/sys/dev/pci/sv.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/pci/sv.c
diff -u src/sys/dev/pci/sv.c:1.54.2.1 src/sys/dev/pci/sv.c:1.54.2.2
--- src/sys/dev/pci/sv.c:1.54.2.1 Sun Apr 21 05:11:22 2019
+++ src/sys/dev/pci/sv.c Mon Apr 29 09:30:18 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: sv.c,v 1.54.2.1 2019/04/21 05:11:22 isaki Exp $ */
+/* $NetBSD: sv.c,v 1.54.2.2 2019/04/29 09:30:18 isaki Exp $ */
/* $OpenBSD: sv.c,v 1.2 1998/07/13 01:50:15 csapuntz Exp $ */
/*
@@ -67,7 +67,7 @@
*/
#include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.2.1 2019/04/21 05:11:22 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.2.2 2019/04/29 09:30:18 isaki Exp $");
#include <sys/param.h>
#include <sys/systm.h>
@@ -81,8 +81,6 @@ __KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.
#include <sys/audioio.h>
#include <dev/audio_if.h>
-#include <dev/mulaw.h>
-#include <dev/auconv.h>
#include <dev/ic/i8237reg.h>
#include <dev/pci/svreg.h>
@@ -141,10 +139,10 @@ static int sv_freemem(struct sv_softc *,
static void sv_init_mixer(struct sv_softc *);
static int sv_open(void *, int);
-static int sv_query_encoding(void *, struct audio_encoding *);
-static int sv_set_params(void *, int, int, audio_params_t *,
- audio_params_t *, stream_filter_list_t *,
- stream_filter_list_t *);
+static int sv_query_format(void *, audio_format_query_t *);
+static int sv_set_format(void *, int,
+ const audio_params_t *, const audio_params_t *,
+ audio_filter_reg_t *, audio_filter_reg_t *);
static int sv_round_blocksize(void *, int, int, const audio_params_t *);
static int sv_trigger_output(void *, void *, void *, int, void (*)(void *),
void *, const audio_params_t *);
@@ -158,8 +156,6 @@ static int sv_mixer_get_port(void *, mix
static int sv_query_devinfo(void *, mixer_devinfo_t *);
static void * sv_malloc(void *, int, size_t);
static void sv_free(void *, void *, size_t);
-static size_t sv_round_buffersize(void *, int, size_t);
-static paddr_t sv_mappage(void *, void *, off_t, int);
static int sv_get_props(void *);
static void sv_get_locks(void *, kmutex_t **, kmutex_t **);
@@ -169,8 +165,8 @@ void sv_dumpregs(struct sv_softc *sc)
static const struct audio_hw_if sv_hw_if = {
.open = sv_open,
- .query_encoding = sv_query_encoding,
- .set_params = sv_set_params,
+ .query_format = sv_query_format,
+ .set_format = sv_set_format,
.round_blocksize = sv_round_blocksize,
.halt_output = sv_halt_output,
.halt_input = sv_halt_input,
@@ -180,32 +176,25 @@ static const struct audio_hw_if sv_hw_if
.query_devinfo = sv_query_devinfo,
.allocm = sv_malloc,
.freem = sv_free,
- .round_buffersize = sv_round_buffersize,
- .mappage = sv_mappage,
.get_props = sv_get_props,
.trigger_output = sv_trigger_output,
.trigger_input = sv_trigger_input,
.get_locks = sv_get_locks,
};
-#define SV_NFORMATS 4
-#define SV_FORMAT(enc, prec, ch, chmask) \
- { \
- .mode = AUMODE_PLAY | AUMODE_RECORD, \
- .encoding = (enc), \
- .validbits = (prec), \
- .precision = (prec), \
- .channels = (ch), \
- .channel_mask = (chmask), \
- .frequency_type = 0, \
- .frequency = { 2000, 48000 }, \
- }
-static const struct audio_format sv_formats[SV_NFORMATS] = {
- SV_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 2, AUFMT_STEREO),
- SV_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 1, AUFMT_MONAURAL),
- SV_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 2, AUFMT_STEREO),
- SV_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 1, AUFMT_MONAURAL),
+static const struct audio_format sv_formats[] = {
+ {
+ .mode = AUMODE_PLAY | AUMODE_RECORD,
+ .encoding = AUDIO_ENCODING_SLINEAR_LE,
+ .validbits = 16,
+ .precision = 16,
+ .channels = 2,
+ .channel_mask = AUFMT_STEREO,
+ .frequency_type = 0,
+ .frequency = { 2000, 48000 },
+ },
};
+#define SV_NFORMATS __arraycount(sv_formats)
static void
@@ -582,105 +571,25 @@ sv_open(void *addr, int flags)
}
static int
-sv_query_encoding(void *addr, struct audio_encoding *fp)
+sv_query_format(void *addr, audio_format_query_t *afp)
{
- switch (fp->index) {
- case 0:
- strcpy(fp->name, AudioEulinear);
- fp->encoding = AUDIO_ENCODING_ULINEAR;
- fp->precision = 8;
- fp->flags = 0;
- return 0;
- case 1:
- strcpy(fp->name, AudioEmulaw);
- fp->encoding = AUDIO_ENCODING_ULAW;
- fp->precision = 8;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- case 2:
- strcpy(fp->name, AudioEalaw);
- fp->encoding = AUDIO_ENCODING_ALAW;
- fp->precision = 8;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- case 3:
- strcpy(fp->name, AudioEslinear);
- fp->encoding = AUDIO_ENCODING_SLINEAR;
- fp->precision = 8;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- case 4:
- strcpy(fp->name, AudioEslinear_le);
- fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
- fp->precision = 16;
- fp->flags = 0;
- return 0;
- case 5:
- strcpy(fp->name, AudioEulinear_le);
- fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
- fp->precision = 16;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- case 6:
- strcpy(fp->name, AudioEslinear_be);
- fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
- fp->precision = 16;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- case 7:
- strcpy(fp->name, AudioEulinear_be);
- fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
- fp->precision = 16;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- return 0;
- default:
- return EINVAL;
- }
+ return audio_query_format(sv_formats, SV_NFORMATS, afp);
}
static int
-sv_set_params(void *addr, int setmode, int usemode, audio_params_t *play,
- audio_params_t *rec, stream_filter_list_t *pfil, stream_filter_list_t *rfil)
+sv_set_format(void *addr, int setmode,
+ const audio_params_t *play, const audio_params_t *rec,
+ audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
{
struct sv_softc *sc;
- audio_params_t *p;
uint32_t val;
sc = addr;
- p = NULL;
- /*
- * This device only has one clock, so make the sample rates match.
- */
- if (play->sample_rate != rec->sample_rate &&
- usemode == (AUMODE_PLAY | AUMODE_RECORD)) {
- if (setmode == AUMODE_PLAY) {
- rec->sample_rate = play->sample_rate;
- setmode |= AUMODE_RECORD;
- } else if (setmode == AUMODE_RECORD) {
- play->sample_rate = rec->sample_rate;
- setmode |= AUMODE_PLAY;
- } else
- return EINVAL;
- }
- if (setmode & AUMODE_RECORD) {
- p = rec;
- if (auconv_set_converter(sv_formats, SV_NFORMATS,
- AUMODE_RECORD, rec, FALSE, rfil) < 0)
- return EINVAL;
- }
- if (setmode & AUMODE_PLAY) {
- p = play;
- if (auconv_set_converter(sv_formats, SV_NFORMATS,
- AUMODE_PLAY, play, FALSE, pfil) < 0)
- return EINVAL;
- }
-
- if (p == NULL)
- return 0;
+ /* *play and *rec are the identical because !AUDIO_PROP_INDEPENDENT. */
- val = p->sample_rate * 65536 / 48000;
+ val = play->sample_rate * 65536 / 48000;
/*
* If the sample rate is exactly 48 kHz, the fraction would overflow the
* register, so we have to bias it. This causes a little clock drift.
@@ -1440,34 +1349,10 @@ sv_free(void *addr, void *ptr, size_t si
}
}
-static size_t
-sv_round_buffersize(void *addr, int direction, size_t size)
-{
-
- return size;
-}
-
-static paddr_t
-sv_mappage(void *addr, void *mem, off_t off, int prot)
-{
- struct sv_softc *sc;
- struct sv_dma *p;
-
- sc = addr;
- if (off < 0)
- return -1;
- for (p = sc->sc_dmas; p && KERNADDR(p) != mem; p = p->next)
- continue;
- if (p == NULL)
- return -1;
- return bus_dmamem_mmap(sc->sc_dmatag, p->segs, p->nsegs,
- off, prot, BUS_DMA_WAITOK);
-}
-
static int
sv_get_props(void *addr)
{
- return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | AUDIO_PROP_FULLDUPLEX;
+ return AUDIO_PROP_MMAP | AUDIO_PROP_FULLDUPLEX;
}
static void
