Module Name: src
Committed By: isaki
Date: Sun Apr 21 14:00:19 UTC 2019
Modified Files:
src/sys/arch/x68k/dev [isaki-audio2]: vs.c vsvar.h
src/sys/dev/ic [isaki-audio2]: msm6258.c msm6258var.h
Log Message:
Adapt to audio2.
To generate a diff of this commit:
cvs rdiff -u -r1.50 -r1.50.2.1 src/sys/arch/x68k/dev/vs.c
cvs rdiff -u -r1.16 -r1.16.6.1 src/sys/arch/x68k/dev/vsvar.h
cvs rdiff -u -r1.25 -r1.25.2.1 src/sys/dev/ic/msm6258.c
cvs rdiff -u -r1.10 -r1.10.8.1 src/sys/dev/ic/msm6258var.h
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/arch/x68k/dev/vs.c
diff -u src/sys/arch/x68k/dev/vs.c:1.50 src/sys/arch/x68k/dev/vs.c:1.50.2.1
--- src/sys/arch/x68k/dev/vs.c:1.50 Sat Mar 16 12:09:57 2019
+++ src/sys/arch/x68k/dev/vs.c Sun Apr 21 14:00:19 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: vs.c,v 1.50 2019/03/16 12:09:57 isaki Exp $ */
+/* $NetBSD: vs.c,v 1.50.2.1 2019/04/21 14:00:19 isaki Exp $ */
/*
* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
@@ -30,7 +30,7 @@
*/
#include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: vs.c,v 1.50 2019/03/16 12:09:57 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: vs.c,v 1.50.2.1 2019/04/21 14:00:19 isaki Exp $");
#include "audio.h"
#include "vs.h"
@@ -43,7 +43,6 @@ __KERNEL_RCSID(0, "$NetBSD: vs.c,v 1.50
#include <sys/audioio.h>
#include <dev/audio_if.h>
-#include <dev/mulaw.h>
#include <machine/bus.h>
#include <machine/cpu.h>
@@ -75,11 +74,11 @@ static int vs_dmaerrintr(void *);
/* MI audio layer interface */
static int vs_open(void *, int);
static void vs_close(void *);
-static int vs_query_encoding(void *, struct audio_encoding *);
-static int vs_set_params(void *, int, int, audio_params_t *,
- audio_params_t *, stream_filter_list_t *, stream_filter_list_t *);
-static int vs_init_output(void *, void *, int);
-static int vs_init_input(void *, void *, int);
+static int vs_query_format(void *, audio_format_query_t *);
+static int vs_set_format(void *, int,
+ const audio_params_t *, const audio_params_t *,
+ audio_filter_reg_t *, audio_filter_reg_t *);
+static int vs_commit_settings(void *);
static int vs_start_input(void *, void *, int, void (*)(void *), void *);
static int vs_start_output(void *, void *, int, void (*)(void *), void *);
static int vs_halt_output(void *);
@@ -99,8 +98,7 @@ static void vs_get_locks(void *, kmutex_
/* lower functions */
static int vs_round_sr(u_long);
-static void vs_set_sr(struct vs_softc *, int);
-static inline void vs_set_po(struct vs_softc *, u_long);
+static inline void vs_set_panout(struct vs_softc *, u_long);
extern struct cfdriver vs_cd;
@@ -112,10 +110,9 @@ static int vs_attached;
static const struct audio_hw_if vs_hw_if = {
.open = vs_open,
.close = vs_close,
- .query_encoding = vs_query_encoding,
- .set_params = vs_set_params,
- .init_output = vs_init_output,
- .init_input = vs_init_input,
+ .query_format = vs_query_format,
+ .set_format = vs_set_format,
+ .commit_settings = vs_commit_settings,
.start_output = vs_start_output,
.start_input = vs_start_input,
.halt_output = vs_halt_output,
@@ -137,6 +134,18 @@ static struct audio_device vs_device = {
"vs"
};
+static const struct audio_format vs_formats = {
+ .mode = AUMODE_PLAY | AUMODE_RECORD,
+ .encoding = AUDIO_ENCODING_ADPCM,
+ .validbits = 4,
+ .precision = 4,
+ .channels = 1,
+ .channel_mask = AUFMT_MONAURAL,
+ .frequency_type = 5,
+ .frequency = { VS_RATE_3K, VS_RATE_5K, VS_RATE_7K,
+ VS_RATE_10K, VS_RATE_15K },
+};
+
struct {
u_long rate;
u_char clk;
@@ -213,7 +222,7 @@ vs_attach(device_t parent, device_t self
sc->sc_prev_vd = NULL;
sc->sc_active = 0;
mutex_init(&sc->sc_lock, MUTEX_DEFAULT, IPL_NONE);
- mutex_init(&sc->sc_intr_lock, MUTEX_DEFAULT, IPL_SCHED);
+ mutex_init(&sc->sc_intr_lock, MUTEX_DEFAULT, IPL_VM);
/* XXX */
bus_space_map(iot, PPI_ADDR, PPI_MAPSIZE, BUS_SPACE_MAP_SHIFTED,
@@ -298,172 +307,87 @@ vs_close(void *hdl)
}
static int
-vs_query_encoding(void *hdl, struct audio_encoding *fp)
+vs_query_format(void *hdl, audio_format_query_t *afp)
{
- DPRINTF(1, ("vs_query_encoding\n"));
-
- if (fp->index == 0) {
- strcpy(fp->name, AudioEslinear);
- fp->encoding = AUDIO_ENCODING_SLINEAR;
- fp->precision = 8;
- fp->flags = 0;
- return 0;
- }
- if (fp->index == 1) {
- strcpy(fp->name, AudioEslinear_be);
- fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
- fp->precision = 16;
- fp->flags = 0;
- return 0;
- }
- return EINVAL;
+ return audio_query_format(&vs_formats, 1, afp);
}
static int
vs_round_sr(u_long rate)
{
int i;
- int diff;
- int nearest;
- diff = rate;
- nearest = 0;
for (i = 0; i < NUM_RATE; i++) {
- if (rate >= vs_l2r[i].rate) {
- if (rate - vs_l2r[i].rate < diff) {
- diff = rate - vs_l2r[i].rate;
- nearest = i;
- }
- } else {
- if (vs_l2r[i].rate - rate < diff) {
- diff = vs_l2r[i].rate - rate;
- nearest = i;
- }
- }
+ if (rate == vs_l2r[i].rate)
+ return i;
}
- if (diff * 100 / rate > 15)
- return -1;
- else
- return nearest;
+ return -1;
}
static int
-vs_set_params(void *hdl, int setmode, int usemode,
- audio_params_t *play, audio_params_t *rec,
- stream_filter_list_t *pfil, stream_filter_list_t *rfil)
+vs_set_format(void *hdl, int setmode,
+ const audio_params_t *play, const audio_params_t *rec,
+ audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
{
struct vs_softc *sc;
- audio_params_t hw;
- stream_filter_factory_t *pconv;
- stream_filter_factory_t *rconv;
int rate;
sc = hdl;
- DPRINTF(1, ("vs_set_params: mode=%d enc=%d rate=%d prec=%d ch=%d: ",
- setmode, play->encoding, play->sample_rate,
- play->precision, play->channels));
+ DPRINTF(1, ("%s: mode=%d %s/%dbit/%dch/%dHz: ", __func__,
+ setmode, audio_encoding_name(play->encoding),
+ play->precision, play->channels, play->sample_rate));
/* *play and *rec are identical because !AUDIO_PROP_INDEPENDENT */
- if (play->channels != 1) {
- DPRINTF(1, ("channels not matched\n"));
- return EINVAL;
- }
-
rate = vs_round_sr(play->sample_rate);
- if (rate < 0) {
- DPRINTF(1, ("rate not matched\n"));
- return EINVAL;
- }
-
- if (play->precision == 8 && play->encoding == AUDIO_ENCODING_SLINEAR) {
- pconv = msm6258_linear8_to_adpcm;
- rconv = msm6258_adpcm_to_linear8;
- } else if (play->precision == 16 &&
- play->encoding == AUDIO_ENCODING_SLINEAR_BE) {
- pconv = msm6258_slinear16_to_adpcm;
- rconv = msm6258_adpcm_to_slinear16;
- } else {
- DPRINTF(1, ("prec/enc not matched\n"));
- return EINVAL;
- }
-
+ KASSERT(rate >= 0);
sc->sc_current.rate = rate;
- /* pfil and rfil are independent even if !AUDIO_PROP_INDEPENDENT */
-
if ((setmode & AUMODE_PLAY) != 0) {
- hw = *play;
- hw.encoding = AUDIO_ENCODING_ADPCM;
- hw.precision = 4;
- hw.validbits = 4;
- pfil->prepend(pfil, pconv, &hw);
+ pfil->codec = msm6258_internal_to_adpcm;
+ pfil->context = &sc->sc_codecvar;
}
if ((setmode & AUMODE_RECORD) != 0) {
- hw = *rec;
- hw.encoding = AUDIO_ENCODING_ADPCM;
- hw.precision = 4;
- hw.validbits = 4;
- rfil->prepend(rfil, rconv, &hw);
+ rfil->codec = msm6258_adpcm_to_internal;
+ rfil->context = &sc->sc_codecvar;
}
DPRINTF(1, ("accepted\n"));
return 0;
}
-static void
-vs_set_sr(struct vs_softc *sc, int rate)
+static int
+vs_commit_settings(void *hdl)
{
+ struct vs_softc *sc;
+ int rate;
+
+ sc = hdl;
+ rate = sc->sc_current.rate;
- DPRINTF(1, ("setting sample rate to %d, %d\n",
+ DPRINTF(1, ("commit_settings: sample rate to %d, %d\n",
rate, (int)vs_l2r[rate].rate));
bus_space_write_1(sc->sc_iot, sc->sc_ppi, PPI_PORTC,
(bus_space_read_1 (sc->sc_iot, sc->sc_ppi,
PPI_PORTC) & 0xf0)
| vs_l2r[rate].den);
adpcm_chgclk(vs_l2r[rate].clk);
+
+ return 0;
}
static inline void
-vs_set_po(struct vs_softc *sc, u_long po)
+vs_set_panout(struct vs_softc *sc, u_long po)
{
+
bus_space_write_1(sc->sc_iot, sc->sc_ppi, PPI_PORTC,
(bus_space_read_1(sc->sc_iot, sc->sc_ppi, PPI_PORTC)
& 0xfc) | po);
}
static int
-vs_init_output(void *hdl, void *buffer, int size)
-{
- struct vs_softc *sc;
-
- DPRINTF(1, ("%s\n", __func__));
- sc = hdl;
-
- /* Set rate and pan */
- vs_set_sr(sc, sc->sc_current.rate);
- vs_set_po(sc, VS_PANOUT_LR);
-
- return 0;
-}
-
-static int
-vs_init_input(void *hdl, void *buffer, int size)
-{
- struct vs_softc *sc;
-
- DPRINTF(1, ("%s\n", __func__));
- sc = hdl;
-
- /* Set rate */
- vs_set_sr(sc, sc->sc_current.rate);
-
- return 0;
-}
-
-static int
vs_start_output(void *hdl, void *block, int blksize, void (*intr)(void *),
void *arg)
{
@@ -501,6 +425,7 @@ vs_start_output(void *hdl, void *block,
(int)block - (int)KVADDR(vd), blksize);
if (sc->sc_active == 0) {
+ vs_set_panout(sc, VS_PANOUT_LR);
bus_space_write_1(sc->sc_iot, sc->sc_ioh,
MSM6258_CMD, MSM6258_CMD_PLAY_START);
sc->sc_active = 1;
@@ -734,30 +659,6 @@ vs_round_buffersize(void *hdl, int direc
return bufsize;
}
-#if 0
-paddr_t
-vs_mappage(void *addr, void *mem, off_t off, int prot)
-{
- struct vs_softc *sc;
- struct vs_dma *p;
-
- if (off < 0)
- return -1;
- sc = addr;
- for (p = sc->sc_dmas; p != NULL && KVADDR(p) != mem;
- p = p->vd_next)
- continue;
- if (p == NULL) {
- printf("%s: mappage: bad addr %p\n",
- device_xname(sc->sc_dev), start);
- return -1;
- }
-
- return bus_dmamem_mmap(sc->sc_dmat, p->vd_segs, p->vd_nsegs,
- off, prot, BUS_DMA_WAITOK);
-}
-#endif
-
static int
vs_get_props(void *hdl)
{
Index: src/sys/arch/x68k/dev/vsvar.h
diff -u src/sys/arch/x68k/dev/vsvar.h:1.16 src/sys/arch/x68k/dev/vsvar.h:1.16.6.1
--- src/sys/arch/x68k/dev/vsvar.h:1.16 Sat Sep 30 04:07:04 2017
+++ src/sys/arch/x68k/dev/vsvar.h Sun Apr 21 14:00:19 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: vsvar.h,v 1.16 2017/09/30 04:07:04 isaki Exp $ */
+/* $NetBSD: vsvar.h,v 1.16.6.1 2019/04/21 14:00:19 isaki Exp $ */
/*
* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
@@ -106,4 +106,6 @@ struct vs_softc {
void (*sc_rintr)(void *);
void *sc_parg;
void *sc_rarg;
+
+ struct msm6258_codecvar sc_codecvar;
};
Index: src/sys/dev/ic/msm6258.c
diff -u src/sys/dev/ic/msm6258.c:1.25 src/sys/dev/ic/msm6258.c:1.25.2.1
--- src/sys/dev/ic/msm6258.c:1.25 Mon Sep 3 16:29:31 2018
+++ src/sys/dev/ic/msm6258.c Sun Apr 21 14:00:19 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: msm6258.c,v 1.25 2018/09/03 16:29:31 riastradh Exp $ */
+/* $NetBSD: msm6258.c,v 1.25.2.1 2019/04/21 14:00:19 isaki Exp $ */
/*
* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
@@ -30,7 +30,7 @@
*/
#include <sys/cdefs.h>
-__KERNEL_RCSID(0, "$NetBSD: msm6258.c,v 1.25 2018/09/03 16:29:31 riastradh Exp $");
+__KERNEL_RCSID(0, "$NetBSD: msm6258.c,v 1.25.2.1 2019/04/21 14:00:19 isaki Exp $");
#include <sys/systm.h>
#include <sys/device.h>
@@ -39,21 +39,9 @@ __KERNEL_RCSID(0, "$NetBSD: msm6258.c,v
#include <sys/audioio.h>
#include <dev/audio_if.h>
-#include <dev/auconv.h>
#include <dev/audiovar.h>
-#include <dev/mulaw.h>
#include <dev/ic/msm6258var.h>
-struct msm6258_codecvar {
- stream_filter_t base;
- short mc_amp;
- char mc_estim;
-};
-
-static stream_filter_t *msm6258_factory
- (struct audio_softc *,
- int (*)(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int));
-static void msm6258_dtor(struct stream_filter *);
static inline uint8_t pcm2adpcm_step(struct msm6258_codecvar *, int16_t);
static inline int16_t adpcm2pcm_step(struct msm6258_codecvar *, uint8_t);
@@ -75,29 +63,6 @@ static const int adpcm_estimstep[16] = {
-1, -1, -1, -1, 2, 4, 6, 8
};
-static int16_t buzzer; /* sound for debug */
-
-static stream_filter_t *
-msm6258_factory(struct audio_softc *asc,
- int (*fetch_to)(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int))
-{
- struct msm6258_codecvar *this;
-
- this = kmem_alloc(sizeof(struct msm6258_codecvar), KM_SLEEP);
- this->base.base.fetch_to = fetch_to;
- this->base.dtor = msm6258_dtor;
- this->base.set_fetcher = stream_filter_set_fetcher;
- this->base.set_inputbuffer = stream_filter_set_inputbuffer;
- return &this->base;
-}
-
-static void
-msm6258_dtor(struct stream_filter *this)
-{
- if (this != NULL)
- kmem_free(this, sizeof(struct msm6258_codecvar));
-}
-
/*
* signed 16bit linear PCM -> OkiADPCM
*/
@@ -134,140 +99,38 @@ pcm2adpcm_step(struct msm6258_codecvar *
return s;
}
-#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 *sc, const audio_params_t *from, \
- const audio_params_t *to) \
-{ \
- return msm6258_factory(sc, name##_fetch_to); \
-} \
-static int \
-name##_fetch_to(struct audio_softc *asc, stream_fetcher_t *self, audio_stream_t *dst, int max_used)
-
-DEFINE_FILTER(msm6258_slinear16_to_adpcm)
+void
+msm6258_internal_to_adpcm(audio_filter_arg_t *arg)
{
- stream_filter_t *this;
struct msm6258_codecvar *mc;
- uint8_t *d;
- const uint8_t *s;
- int m, err, enc_src;
-
- this = (stream_filter_t *)self;
- mc = (struct msm6258_codecvar *)self;
- if ((err = this->prev->fetch_to(asc, this->prev, this->src, max_used * 4)))
- return err;
- m = dst->end - dst->start;
- m = uimin(m, max_used);
- d = dst->inp;
- s = this->src->outp;
- enc_src = this->src->param.encoding;
- if (enc_src == AUDIO_ENCODING_SLINEAR_LE) {
- while (dst->used < m && this->src->used >= 4) {
- uint8_t f;
- int16_t ss;
- ss = le16toh(*(const int16_t*)s);
- f = pcm2adpcm_step(mc, ss);
- s = audio_stream_add_outp(this->src, s, 2);
- ss = le16toh(*(const int16_t*)s);
- f |= pcm2adpcm_step(mc, ss) << 4;
- s = audio_stream_add_outp(this->src, s, 2);
- *d = f;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else if (enc_src == AUDIO_ENCODING_SLINEAR_BE) {
- while (dst->used < m && this->src->used >= 4) {
- uint8_t f;
- int16_t ss;
- ss = be16toh(*(const int16_t*)s);
- s = audio_stream_add_outp(this->src, s, 2);
- f = pcm2adpcm_step(mc, ss);
- ss = be16toh(*(const int16_t*)s);
- s = audio_stream_add_outp(this->src, s, 2);
- f |= pcm2adpcm_step(mc, ss) << 4;
- *d = f;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else {
-#if defined(DIAGNOSTIC)
- panic("msm6258_slinear16_to_adpcm: unsupported enc_src(%d)", enc_src);
-#endif
- /* dummy run */
- while (dst->used < m && this->src->used >= 4) {
- s = audio_stream_add_outp(this->src, s, 2);
- s = audio_stream_add_outp(this->src, s, 2);
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 1);
- }
- }
- dst->inp = d;
- this->src->outp = s;
- return 0;
-}
+ const aint_t *src;
+ uint8_t *dst;
+ u_int sample_count;
+ u_int i;
+
+ KASSERT((arg->count & 1) == 0);
+
+ mc = arg->context;
+ src = arg->src;
+ dst = arg->dst;
+ sample_count = arg->count * arg->srcfmt->channels;
+ for (i = 0; i < sample_count / 2; i++) {
+ aint_t s;
+ uint8_t f;
+
+ s = *src++;
+ s >>= AUDIO_INTERNAL_BITS - 16;
+ f = pcm2adpcm_step(mc, s);
+
+ s = *src++;
+ s >>= AUDIO_INTERNAL_BITS - 16;
+ f |= pcm2adpcm_step(mc, s) << 4;
-DEFINE_FILTER(msm6258_linear8_to_adpcm)
-{
- stream_filter_t *this;
- struct msm6258_codecvar *mc;
- uint8_t *d;
- const uint8_t *s;
- int m, err, enc_src;
-
- this = (stream_filter_t *)self;
- mc = (struct msm6258_codecvar *)self;
- if ((err = this->prev->fetch_to(asc, this->prev, this->src, max_used * 2)))
- return err;
- m = dst->end - dst->start;
- m = uimin(m, max_used);
- d = dst->inp;
- s = this->src->outp;
- enc_src = this->src->param.encoding;
- if (enc_src == AUDIO_ENCODING_SLINEAR_LE
- || enc_src == AUDIO_ENCODING_SLINEAR_BE) {
- while (dst->used < m && this->src->used >= 4) {
- uint8_t f;
- int16_t ss;
- ss = ((int16_t)s[0]) * 256;
- s = audio_stream_add_outp(this->src, s, 1);
- f = pcm2adpcm_step(mc, ss);
- ss = ((int16_t)s[0]) * 256;
- s = audio_stream_add_outp(this->src, s, 1);
- f |= pcm2adpcm_step(mc, ss) << 4;
- *d = f;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else if (enc_src == AUDIO_ENCODING_ULINEAR_LE
- || enc_src == AUDIO_ENCODING_ULINEAR_BE) {
- while (dst->used < m && this->src->used >= 4) {
- uint8_t f;
- int16_t ss;
- ss = ((int16_t)(s[0] ^ 0x80)) * 256;
- s = audio_stream_add_outp(this->src, s, 1);
- f = pcm2adpcm_step(mc, ss);
- ss = ((int16_t)(s[0] ^ 0x80)) * 256;
- s = audio_stream_add_outp(this->src, s, 1);
- f |= pcm2adpcm_step(mc, ss) << 4;
- *d = f;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else {
-#if defined(DIAGNOSTIC)
- panic("msm6258_linear8_to_adpcm: unsupported enc_src(%d)", enc_src);
-#endif
- /* dummy run */
- while (dst->used < m && this->src->used >= 4) {
- s = audio_stream_add_outp(this->src, s, 1);
- s = audio_stream_add_outp(this->src, s, 1);
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 1);
- }
+ *dst++ = (uint8_t)f;
}
- dst->inp = d;
- this->src->outp = s;
- return 0;
}
+
/*
* OkiADPCM -> signed 16bit linear PCM
*/
@@ -289,126 +152,31 @@ adpcm2pcm_step(struct msm6258_codecvar *
return mc->mc_amp;
}
-DEFINE_FILTER(msm6258_adpcm_to_slinear16)
-{
- stream_filter_t *this;
- struct msm6258_codecvar *mc;
- uint8_t *d;
- const uint8_t *s;
- int m, err, enc_dst;
-
- this = (stream_filter_t *)self;
- mc = (struct msm6258_codecvar *)self;
- max_used = (max_used + 3) & ~3; /* round up multiple of 4 */
- if ((err = this->prev->fetch_to(asc, this->prev, this->src, max_used / 4)))
- return err;
- m = (dst->end - dst->start) & ~3;
- m = uimin(m, max_used);
- d = dst->inp;
- s = this->src->outp;
- enc_dst = dst->param.encoding;
- if (enc_dst == AUDIO_ENCODING_SLINEAR_LE) {
- while (dst->used < m && this->src->used >= 1) {
- uint8_t a;
- int16_t s1, s2;
- a = s[0];
- s1 = adpcm2pcm_step(mc, a & 0x0f);
- s2 = adpcm2pcm_step(mc, a >> 4);
- *(int16_t*)d = htole16(s1);
- d = audio_stream_add_inp(dst, d, 2);
- *(int16_t*)d = htole16(s2);
- d = audio_stream_add_inp(dst, d, 2);
- s = audio_stream_add_outp(this->src, s, 1);
- }
- } else if (enc_dst == AUDIO_ENCODING_SLINEAR_BE) {
- while (dst->used < m && this->src->used >= 1) {
- uint8_t a;
- int16_t s1, s2;
- a = s[0];
- s1 = adpcm2pcm_step(mc, a & 0x0f);
- s2 = adpcm2pcm_step(mc, a >> 4);
- *(int16_t*)d = htobe16(s1);
- d = audio_stream_add_inp(dst, d, 2);
- *(int16_t*)d = htobe16(s2);
- d = audio_stream_add_inp(dst, d, 2);
- s = audio_stream_add_outp(this->src, s, 1);
- }
- } else {
-#if defined(DIAGNOSTIC)
- panic("msm6258_adpcm_to_slinear16: unsupported enc_dst(%d)", enc_dst);
-#endif
- /* dummy run */
- while (dst->used < m && this->src->used >= 1) {
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 2);
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 2);
- s = audio_stream_add_outp(this->src, s, 1);
- }
- }
- dst->inp = d;
- this->src->outp = s;
- return 0;
-}
-
-DEFINE_FILTER(msm6258_adpcm_to_linear8)
+void
+msm6258_adpcm_to_internal(audio_filter_arg_t *arg)
{
- stream_filter_t *this;
struct msm6258_codecvar *mc;
- uint8_t *d;
- const uint8_t *s;
- int m, err, enc_dst;
-
- this = (stream_filter_t *)self;
- mc = (struct msm6258_codecvar *)self;
- max_used = (max_used + 1) & ~1; /* round up multiple of 4 */
- if ((err = this->prev->fetch_to(asc, this->prev, this->src, max_used / 2)))
- return err;
- m = (dst->end - dst->start) & ~1;
- m = uimin(m, max_used);
- d = dst->inp;
- s = this->src->outp;
- enc_dst = dst->param.encoding;
- if (enc_dst == AUDIO_ENCODING_SLINEAR_LE) {
- while (dst->used < m && this->src->used >= 1) {
- uint8_t a;
- int16_t s1, s2;
- a = s[0];
- s = audio_stream_add_outp(this->src, s, 1);
- s1 = adpcm2pcm_step(mc, a & 0x0f);
- s2 = adpcm2pcm_step(mc, a >> 4);
- d[0] = s1 / 256;
- d = audio_stream_add_inp(dst, d, 1);
- d[0] = s2 / 256;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else if (enc_dst == AUDIO_ENCODING_ULINEAR_LE) {
- while (dst->used < m && this->src->used >= 1) {
- uint8_t a;
- int16_t s1, s2;
- a = s[0];
- s = audio_stream_add_outp(this->src, s, 1);
- s1 = adpcm2pcm_step(mc, a & 0x0f);
- s2 = adpcm2pcm_step(mc, a >> 4);
- d[0] = (s1 / 256) ^ 0x80;
- d = audio_stream_add_inp(dst, d, 1);
- d[0] = (s2 / 256) ^ 0x80;
- d = audio_stream_add_inp(dst, d, 1);
- }
- } else {
-#if defined(DIAGNOSTIC)
- panic("msm6258_adpcm_to_linear8: unsupported enc_dst(%d)", enc_dst);
-#endif
- /* dummy run */
- while (dst->used < m && this->src->used >= 1) {
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 1);
- *d = buzzer++;
- d = audio_stream_add_inp(dst, d, 1);
- s = audio_stream_add_outp(this->src, s, 1);
- }
+ const uint8_t *src;
+ aint_t *dst;
+ u_int sample_count;
+ u_int i;
+
+ KASSERT((arg->count & 1) == 0);
+
+ mc = arg->context;
+ src = arg->src;
+ dst = arg->dst;
+ sample_count = arg->count * arg->srcfmt->channels;
+ for (i = 0; i < sample_count / 2; i++) {
+ uint8_t a = *src++;
+ aint_t s;
+
+ s = adpcm2pcm_step(mc, a & 0x0f);
+ s <<= AUDIO_INTERNAL_BITS - 16;
+ *dst++ = s;
+
+ s = adpcm2pcm_step(mc, a >> 4);
+ s <<= AUDIO_INTERNAL_BITS - 16;
+ *dst++ = s;
}
- dst->inp = d;
- this->src->outp = s;
- return 0;
}
Index: src/sys/dev/ic/msm6258var.h
diff -u src/sys/dev/ic/msm6258var.h:1.10 src/sys/dev/ic/msm6258var.h:1.10.8.1
--- src/sys/dev/ic/msm6258var.h:1.10 Sat Sep 2 12:57:35 2017
+++ src/sys/dev/ic/msm6258var.h Sun Apr 21 14:00:19 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: msm6258var.h,v 1.10 2017/09/02 12:57:35 isaki Exp $ */
+/* $NetBSD: msm6258var.h,v 1.10.8.1 2019/04/21 14:00:19 isaki Exp $ */
/*
* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
@@ -28,7 +28,13 @@
/*
* OKI MSM6258 ADPCM voice synthesizer codec.
*/
-extern stream_filter_factory_t msm6258_slinear16_to_adpcm;
-extern stream_filter_factory_t msm6258_linear8_to_adpcm;
-extern stream_filter_factory_t msm6258_adpcm_to_slinear16;
-extern stream_filter_factory_t msm6258_adpcm_to_linear8;
+
+struct msm6258_codecvar {
+ int16_t mc_amp;
+ int8_t mc_estim;
+
+ aint_t pending_sample; /* hold one pending sample. */
+};
+
+extern void msm6258_adpcm_to_internal(audio_filter_arg_t *);
+extern void msm6258_internal_to_adpcm(audio_filter_arg_t *);
