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.
This commit is contained in:
isaki
parent
b3c09b3fa6
commit
93d792beb2
171
sys/dev/pci/sv.c
171
sys/dev/pci/sv.c
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@ -1,4 +1,4 @@
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/* $NetBSD: sv.c,v 1.54.2.1 2019/04/21 05:11:22 isaki Exp $ */
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/* $NetBSD: sv.c,v 1.54.2.2 2019/04/29 09:30:18 isaki Exp $ */
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/* $OpenBSD: sv.c,v 1.2 1998/07/13 01:50:15 csapuntz Exp $ */
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/*
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@ -67,7 +67,7 @@
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*/
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#include <sys/cdefs.h>
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__KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.2.1 2019/04/21 05:11:22 isaki Exp $");
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__KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.2.2 2019/04/29 09:30:18 isaki Exp $");
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#include <sys/param.h>
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#include <sys/systm.h>
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@ -81,8 +81,6 @@ __KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.54.2.1 2019/04/21 05:11:22 isaki Exp $");
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#include <sys/audioio.h>
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#include <dev/audio_if.h>
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#include <dev/mulaw.h>
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#include <dev/auconv.h>
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#include <dev/ic/i8237reg.h>
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#include <dev/pci/svreg.h>
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@ -141,10 +139,10 @@ static int sv_freemem(struct sv_softc *, struct sv_dma *);
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static void sv_init_mixer(struct sv_softc *);
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static int sv_open(void *, int);
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static int sv_query_encoding(void *, struct audio_encoding *);
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static int sv_set_params(void *, int, int, audio_params_t *,
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audio_params_t *, stream_filter_list_t *,
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stream_filter_list_t *);
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static int sv_query_format(void *, audio_format_query_t *);
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static int sv_set_format(void *, int,
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const audio_params_t *, const audio_params_t *,
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audio_filter_reg_t *, audio_filter_reg_t *);
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static int sv_round_blocksize(void *, int, int, const audio_params_t *);
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static int sv_trigger_output(void *, void *, void *, int, void (*)(void *),
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void *, const audio_params_t *);
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@ -158,8 +156,6 @@ static int sv_mixer_get_port(void *, mixer_ctrl_t *);
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static int sv_query_devinfo(void *, mixer_devinfo_t *);
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static void * sv_malloc(void *, int, size_t);
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static void sv_free(void *, void *, size_t);
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static size_t sv_round_buffersize(void *, int, size_t);
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static paddr_t sv_mappage(void *, void *, off_t, int);
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static int sv_get_props(void *);
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static void sv_get_locks(void *, kmutex_t **, kmutex_t **);
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@ -169,8 +165,8 @@ void sv_dumpregs(struct sv_softc *sc);
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static const struct audio_hw_if sv_hw_if = {
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.open = sv_open,
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.query_encoding = sv_query_encoding,
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.set_params = sv_set_params,
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.query_format = sv_query_format,
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.set_format = sv_set_format,
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.round_blocksize = sv_round_blocksize,
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.halt_output = sv_halt_output,
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.halt_input = sv_halt_input,
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@ -180,32 +176,25 @@ static const struct audio_hw_if sv_hw_if = {
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.query_devinfo = sv_query_devinfo,
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.allocm = sv_malloc,
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.freem = sv_free,
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.round_buffersize = sv_round_buffersize,
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.mappage = sv_mappage,
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.get_props = sv_get_props,
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.trigger_output = sv_trigger_output,
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.trigger_input = sv_trigger_input,
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.get_locks = sv_get_locks,
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};
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#define SV_NFORMATS 4
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#define SV_FORMAT(enc, prec, ch, chmask) \
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{ \
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.mode = AUMODE_PLAY | AUMODE_RECORD, \
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.encoding = (enc), \
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.validbits = (prec), \
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.precision = (prec), \
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.channels = (ch), \
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.channel_mask = (chmask), \
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.frequency_type = 0, \
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.frequency = { 2000, 48000 }, \
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}
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static const struct audio_format sv_formats[SV_NFORMATS] = {
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SV_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 2, AUFMT_STEREO),
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SV_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 1, AUFMT_MONAURAL),
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SV_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 2, AUFMT_STEREO),
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SV_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 1, AUFMT_MONAURAL),
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static const struct audio_format sv_formats[] = {
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{
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.mode = AUMODE_PLAY | AUMODE_RECORD,
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.encoding = AUDIO_ENCODING_SLINEAR_LE,
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.validbits = 16,
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.precision = 16,
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.channels = 2,
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.channel_mask = AUFMT_STEREO,
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.frequency_type = 0,
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.frequency = { 2000, 48000 },
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},
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};
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#define SV_NFORMATS __arraycount(sv_formats)
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static void
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@ -582,105 +571,25 @@ sv_open(void *addr, int flags)
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}
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static int
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sv_query_encoding(void *addr, struct audio_encoding *fp)
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sv_query_format(void *addr, audio_format_query_t *afp)
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{
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switch (fp->index) {
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case 0:
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strcpy(fp->name, AudioEulinear);
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fp->encoding = AUDIO_ENCODING_ULINEAR;
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fp->precision = 8;
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fp->flags = 0;
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return 0;
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case 1:
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strcpy(fp->name, AudioEmulaw);
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fp->encoding = AUDIO_ENCODING_ULAW;
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fp->precision = 8;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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case 2:
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strcpy(fp->name, AudioEalaw);
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fp->encoding = AUDIO_ENCODING_ALAW;
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fp->precision = 8;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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case 3:
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strcpy(fp->name, AudioEslinear);
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fp->encoding = AUDIO_ENCODING_SLINEAR;
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fp->precision = 8;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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case 4:
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strcpy(fp->name, AudioEslinear_le);
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fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
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fp->precision = 16;
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fp->flags = 0;
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return 0;
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case 5:
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strcpy(fp->name, AudioEulinear_le);
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fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
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fp->precision = 16;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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case 6:
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strcpy(fp->name, AudioEslinear_be);
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fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
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fp->precision = 16;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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case 7:
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strcpy(fp->name, AudioEulinear_be);
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fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
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fp->precision = 16;
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fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
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return 0;
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default:
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return EINVAL;
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}
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return audio_query_format(sv_formats, SV_NFORMATS, afp);
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}
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static int
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sv_set_params(void *addr, int setmode, int usemode, audio_params_t *play,
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audio_params_t *rec, stream_filter_list_t *pfil, stream_filter_list_t *rfil)
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sv_set_format(void *addr, int setmode,
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const audio_params_t *play, const audio_params_t *rec,
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audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
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{
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struct sv_softc *sc;
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audio_params_t *p;
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uint32_t val;
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sc = addr;
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p = NULL;
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/*
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* This device only has one clock, so make the sample rates match.
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*/
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if (play->sample_rate != rec->sample_rate &&
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usemode == (AUMODE_PLAY | AUMODE_RECORD)) {
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if (setmode == AUMODE_PLAY) {
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rec->sample_rate = play->sample_rate;
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setmode |= AUMODE_RECORD;
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} else if (setmode == AUMODE_RECORD) {
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play->sample_rate = rec->sample_rate;
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setmode |= AUMODE_PLAY;
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} else
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return EINVAL;
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}
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if (setmode & AUMODE_RECORD) {
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p = rec;
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if (auconv_set_converter(sv_formats, SV_NFORMATS,
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AUMODE_RECORD, rec, FALSE, rfil) < 0)
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return EINVAL;
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}
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if (setmode & AUMODE_PLAY) {
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p = play;
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if (auconv_set_converter(sv_formats, SV_NFORMATS,
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AUMODE_PLAY, play, FALSE, pfil) < 0)
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return EINVAL;
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}
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/* *play and *rec are the identical because !AUDIO_PROP_INDEPENDENT. */
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if (p == NULL)
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return 0;
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val = p->sample_rate * 65536 / 48000;
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val = play->sample_rate * 65536 / 48000;
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/*
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* If the sample rate is exactly 48 kHz, the fraction would overflow the
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* register, so we have to bias it. This causes a little clock drift.
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@ -1440,34 +1349,10 @@ sv_free(void *addr, void *ptr, size_t size)
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}
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}
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static size_t
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sv_round_buffersize(void *addr, int direction, size_t size)
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{
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return size;
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}
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static paddr_t
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sv_mappage(void *addr, void *mem, off_t off, int prot)
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{
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struct sv_softc *sc;
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struct sv_dma *p;
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sc = addr;
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if (off < 0)
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return -1;
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for (p = sc->sc_dmas; p && KERNADDR(p) != mem; p = p->next)
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continue;
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if (p == NULL)
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return -1;
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return bus_dmamem_mmap(sc->sc_dmatag, p->segs, p->nsegs,
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off, prot, BUS_DMA_WAITOK);
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}
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static int
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sv_get_props(void *addr)
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{
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return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | AUDIO_PROP_FULLDUPLEX;
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return AUDIO_PROP_MMAP | AUDIO_PROP_FULLDUPLEX;
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}
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static void
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