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[src/isaki-audio2]: src/sys/arch/macppc/dev Adapt to audio2.
details: https://anonhg.NetBSD.org/src/rev/c5894799db56
branches: isaki-audio2
changeset: 455985:c5894799db56
user: isaki <isaki%NetBSD.org@localhost>
date: Mon Apr 22 12:30:26 2019 +0000
description:
Adapt to audio2.
- New audio filter can access softc so it's simple.
- Clear rval of fixphase on halt.
diffstat:
sys/arch/macppc/dev/snapper.c | 320 ++++++++++++-----------------------------
1 files changed, 93 insertions(+), 227 deletions(-)
diffs (truncated from 478 to 300 lines):
diff -r cfb9b40b205d -r c5894799db56 sys/arch/macppc/dev/snapper.c
--- a/sys/arch/macppc/dev/snapper.c Sun Apr 21 14:00:19 2019 +0000
+++ b/sys/arch/macppc/dev/snapper.c Mon Apr 22 12:30:26 2019 +0000
@@ -1,4 +1,4 @@
-/* $NetBSD: snapper.c,v 1.49.2.1 2019/04/21 05:11:21 isaki Exp $ */
+/* $NetBSD: snapper.c,v 1.49.2.2 2019/04/22 12:30:26 isaki Exp $ */
/* Id: snapper.c,v 1.11 2002/10/31 17:42:13 tsubai Exp */
/* Id: i2s.c,v 1.12 2005/01/15 14:32:35 tsubai Exp */
@@ -35,7 +35,7 @@
*/
#include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: snapper.c,v 1.49.2.1 2019/04/21 05:11:21 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: snapper.c,v 1.49.2.2 2019/04/22 12:30:26 isaki Exp $");
#include <sys/param.h>
#include <sys/audioio.h>
@@ -43,9 +43,7 @@
#include <sys/systm.h>
#include <sys/malloc.h>
-#include <dev/auconv.h>
#include <dev/audio_if.h>
-#include <dev/mulaw.h>
#include <dev/ofw/openfirm.h>
#include <macppc/dev/dbdma.h>
@@ -74,8 +72,6 @@
int sc_node;
- struct audio_encoding_set *sc_encodings;
-
void (*sc_ointr)(void *); /* dma completion intr handler */
void *sc_oarg; /* arg for sc_ointr() */
int sc_opages; /* # of output pages */
@@ -96,13 +92,16 @@
int sc_rate; /* current sampling rate */
int sc_bitspersample;
- int sc_swvol;
+ /* for SNAPPER_SWVOL */
+ u_int sc_swvol_l;
+ u_int sc_swvol_r;
u_int sc_vol_l;
u_int sc_vol_r;
u_int sc_treble;
u_int sc_bass;
u_int mixer[6]; /* s1_l, s2_l, an_l, s1_r, s2_r, an_r */
+ uint16_t sc_rval;
bus_space_handle_t sc_odmah;
bus_space_handle_t sc_idmah;
@@ -120,9 +119,10 @@
static void snapper_attach(device_t, device_t, void *);
static void snapper_defer(device_t);
static int snapper_intr(void *);
-static int snapper_query_encoding(void *, struct audio_encoding *);
-static int snapper_set_params(void *, int, int, audio_params_t *,
- audio_params_t *, stream_filter_list_t *, stream_filter_list_t *);
+static int snapper_query_format(void *, audio_format_query_t *);
+static int snapper_set_format(void *, int,
+ const audio_params_t *, const audio_params_t *,
+ audio_filter_reg_t *, audio_filter_reg_t *);
static int snapper_round_blocksize(void *, int, int, const audio_params_t *);
static int snapper_halt_output(void *);
static int snapper_halt_input(void *);
@@ -131,7 +131,6 @@
static int snapper_get_port(void *, mixer_ctrl_t *);
static int snapper_query_devinfo(void *, mixer_devinfo_t *);
static size_t snapper_round_buffersize(void *, int, size_t);
-static paddr_t snapper_mappage(void *, void *, off_t, int);
static int snapper_get_props(void *);
static int snapper_trigger_output(void *, void *, void *, int, void (*)(void *),
void *, const audio_params_t *);
@@ -153,61 +152,24 @@
static int tas3004_init(struct snapper_softc *);
static void snapper_init(struct snapper_softc *, int);
-struct snapper_codecvar {
- stream_filter_t base;
-
-#ifdef DIAGNOSTIC
-# define SNAPPER_CODECVAR_MAGIC 0xC0DEC
- uint32_t magic;
-#endif // DIAGNOSTIC
-
- int16_t rval; // for snapper_fixphase
-};
-
-static stream_filter_t *snapper_filter_factory
- (int (*)(struct audio_softc *sc, stream_fetcher_t *, audio_stream_t *, int));
-static void snapper_filter_dtor(stream_filter_t *);
-
-/* XXX We can't access the hw device softc from our audio
- * filter -- lame...
- */
-static u_int snapper_vol_l = 128, snapper_vol_r = 128;
+static void
+snapper_volume(audio_filter_arg_t *arg)
+{
+ struct snapper_softc *sc;
+ const aint_t *src;
+ aint_t *dst;
+ u_int sample_count;
+ u_int i;
-/* XXX why doesn't auconv define this? */
-#define DEFINE_FILTER(name) \
-static int \
-name##_fetch_to(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int); \
-stream_filter_t * name(struct audio_softc *, \
- const audio_params_t *, const audio_params_t *); \
-stream_filter_t * \
-name(struct audio_softc *sc, const audio_params_t *from, \
- const audio_params_t *to) \
-{ \
- return snapper_filter_factory(name##_fetch_to); \
-} \
-static int \
-name##_fetch_to(struct audio_softc *sc, stream_fetcher_t *self, audio_stream_t *dst, int max_used)
-
-DEFINE_FILTER(snapper_volume)
-{
- stream_filter_t *this;
- int16_t j;
- int16_t *wp;
- int m, err;
-
- this = (stream_filter_t *)self;
- max_used = (max_used + 1) & ~1;
- if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
- return err;
- m = (dst->end - dst->start) & ~1;
- m = uimin(m, max_used);
- FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
- j = (s[0] << 8 | s[1]);
- wp = (int16_t *)d;
- *wp = ((j * snapper_vol_l) / 255);
- } FILTER_LOOP_EPILOGUE(this->src, dst);
-
- return 0;
+ sc = arg->context;
+ src = arg->src;
+ dst = arg->dst;
+ sample_count = arg->count * arg->srcfmt->channels;
+ for (i = 0; i < sample_count; i++) {
+ aint2_t l = (aint2_t)(*src++);
+ l = l * sc->sc_swvol_l / 255;
+ *dst++ = (aint_t)l;
+ }
}
/*
@@ -217,67 +179,30 @@
* Fix the phase difference by delaying the right channel
* by one sample.
*/
-DEFINE_FILTER(snapper_fixphase)
+static void
+snapper_fixphase(audio_filter_arg_t *arg)
{
- struct snapper_codecvar *cv = (struct snapper_codecvar *) self;
- stream_filter_t *this = &cv->base;
- int err, m;
- const int16_t *rp;
- int16_t *wp, rval = cv->rval;
-
-#ifdef DIAGNOSTIC
- if (cv->magic != SNAPPER_CODECVAR_MAGIC)
- panic("snapper_fixphase");
-#endif
- max_used = (max_used + 3) & ~2;
- if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
- return err;
+ struct snapper_softc *sc;
+ const aint_t *src;
+ aint_t *dst;
+ u_int i;
- /* work in stereo frames (4 bytes) */
- m = (dst->end - dst->start) & ~2;
- m = uimin(m, max_used);
- FILTER_LOOP_PROLOGUE(this->src, 4, dst, 4, m) {
- rp = (const int16_t *) s;
- wp = (int16_t *) d;
- wp[0] = rp[0];
- wp[1] = rval;
- rval = rp[1];
- } FILTER_LOOP_EPILOGUE(this->src, dst);
- cv->rval = rval;
-
- return 0;
-}
-
-static stream_filter_t *
-snapper_filter_factory(int (*fetch_to)(struct audio_softc *sc, stream_fetcher_t *, audio_stream_t *, int))
-{
- struct snapper_codecvar *this;
-
- this = malloc(sizeof(*this), M_DEVBUF, M_WAITOK | M_ZERO);
- this->base.base.fetch_to = fetch_to;
- this->base.dtor = snapper_filter_dtor;
- this->base.set_fetcher = stream_filter_set_fetcher;
- this->base.set_inputbuffer = stream_filter_set_inputbuffer;
-
-#ifdef DIAGNOSTIC
- this->magic = SNAPPER_CODECVAR_MAGIC;
-#endif
- return (stream_filter_t *) this;
-}
-
-static void
-snapper_filter_dtor(stream_filter_t *this)
-{
- if (this != NULL)
- free(this, M_DEVBUF);
+ sc = arg->context;
+ src = arg->src;
+ dst = arg->dst;
+ for (i = 0; i < arg->count; i++) {
+ *dst++ = *src++;
+ *dst++ = sc->sc_rval;
+ sc->sc_rval = *src++;
+ }
}
CFATTACH_DECL_NEW(snapper, sizeof(struct snapper_softc), snapper_match,
snapper_attach, NULL, NULL);
const struct audio_hw_if snapper_hw_if = {
- .query_encoding = snapper_query_encoding,
- .set_params = snapper_set_params,
+ .query_format = snapper_query_format,
+ .set_format = snapper_set_format,
.round_blocksize = snapper_round_blocksize,
.halt_output = snapper_halt_output,
.halt_input = snapper_halt_input,
@@ -286,7 +211,6 @@
.get_port = snapper_get_port,
.query_devinfo = snapper_query_devinfo,
.round_buffersize = snapper_round_buffersize,
- .mappage = snapper_mappage,
.get_props = snapper_get_props,
.trigger_output = snapper_trigger_output,
.trigger_input = snapper_trigger_input,
@@ -522,25 +446,22 @@
{ 0x00, 0x00, 0x00 } /* Mute */
};
-#define SNAPPER_NFORMATS 2
-#define SNAPPER_FORMAT(prec) \
- { \
- .mode = AUMODE_PLAY | AUMODE_RECORD, \
- .encoding = AUDIO_ENCODING_SLINEAR_BE, \
- .validbits = (prec), \
- .precision = (prec), \
- .channels = 2, \
- .channel_mask = AUFMT_STEREO, \
- .frequency_type = 3, \
- .frequency = { 32000, 44100, 48000 }, \
+/* The HW actually supports precisions more than 16bit, but 16bit is enough. */
+static const struct audio_format snapper_formats[] = {
+ {
+ .mode = AUMODE_PLAY | AUMODE_RECORD,
+ .encoding = AUDIO_ENCODING_SLINEAR_BE,
+ .validbits = 16,
+ .precision = 16,
+ .channels = 2,
+ .channel_mask = AUFMT_STEREO,
+ .frequency_type = 3,
+ .frequency = { 32000, 44100, 48000 },
}
-static const struct audio_format snapper_formats[SNAPPER_NFORMATS] = {
- SNAPPER_FORMAT(16),
- SNAPPER_FORMAT(24),
};
+#define SNAPPER_NFORMATS __arraycount(snapper_formats)
-#define TUMBLER_NFORMATS 1
-static const struct audio_format tumbler_formats[TUMBLER_NFORMATS] = {
+static const struct audio_format tumbler_formats[] = {
{
.mode = AUMODE_PLAY | AUMODE_RECORD,
.encoding = AUDIO_ENCODING_SLINEAR_BE,
@@ -552,6 +473,7 @@
.frequency = { 32000, 44100, 48000, 96000 },
},
};
+#define TUMBLER_NFORMATS __arraycount(tumbler_formats)
static bus_size_t amp_mute;
static bus_size_t headphone_mute;
@@ -744,20 +666,11 @@
if (strcmp(compat, "tumbler") == 0)
sc->sc_mode = SNAPPER_IS_TAS3001;
-
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