NetBSD-5.0.2/sys/dev/ic/msm6258.c
/* $NetBSD: msm6258.c,v 1.15 2005/12/24 20:27:30 perry Exp $ */
/*
* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
/*
* OKI MSM6258 ADPCM voice synthesizer codec.
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: msm6258.c,v 1.15 2005/12/24 20:27:30 perry Exp $");
#include <sys/systm.h>
#include <sys/device.h>
#include <sys/malloc.h>
#include <sys/select.h>
#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
(int (*)(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);
static const int adpcm_estimindex[16] = {
2, 6, 10, 14, 18, 22, 26, 30,
-2, -6, -10, -14, -18, -22, -26, -30
};
static const int adpcm_estim[49] = {
16, 17, 19, 21, 23, 25, 28, 31, 34, 37,
41, 45, 50, 55, 60, 66, 73, 80, 88, 97,
107, 118, 130, 143, 157, 173, 190, 209, 230, 253,
279, 307, 337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552
};
static const int adpcm_estimstep[16] = {
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8
};
static stream_filter_t *
msm6258_factory(int (*fetch_to)(stream_fetcher_t *, audio_stream_t *, int))
{
struct msm6258_codecvar *this;
this = malloc(sizeof(*this), M_DEVBUF, M_WAITOK | M_ZERO);
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)
free(this, M_DEVBUF);
}
/*
* signed 16bit linear PCM -> OkiADPCM
*/
static inline uint8_t
pcm2adpcm_step(struct msm6258_codecvar *mc, int16_t a)
{
int estim = (int)mc->mc_estim;
int df;
short dl, c;
uint8_t b;
uint8_t s;
df = a - mc->mc_amp;
dl = adpcm_estim[estim];
c = (df / 16) * 8 / dl;
if (df < 0) {
b = (unsigned char)(-c) / 2;
s = 0x08;
} else {
b = (unsigned char)(c) / 2;
s = 0;
}
if (b > 7)
b = 7;
s |= b;
mc->mc_amp += (short)(adpcm_estimindex[(int)s] * dl);
estim += adpcm_estimstep[b];
if (estim < 0)
estim = 0;
else if (estim > 48)
estim = 48;
mc->mc_estim = estim;
return s;
}
#define DEFINE_FILTER(name) \
static int \
name##_fetch_to(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(name##_fetch_to); \
} \
static int \
name##_fetch_to(stream_fetcher_t *self, audio_stream_t *dst, int max_used)
DEFINE_FILTER(msm6258_slinear16_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(this->prev, this->src, max_used * 4)))
return err;
m = dst->end - dst->start;
m = min(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;
#if BYTE_ORDER == LITTLE_ENDIAN
ss = *(const int16_t*)s;
s = audio_stream_add_outp(this->src, s, 2);
f = pcm2adpcm_step(mc, ss);
ss = *(const int16_t*)s;
#else
ss = (s[1] << 8) | s[0];
s = audio_stream_add_outp(this->src, s, 2);
f = pcm2adpcm_step(mc, ss);
ss = (s[1] << 8) | s[0];
#endif
f |= pcm2adpcm_step(mc, ss) << 4;
*d = f;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 2);
}
} else {
while (dst->used < m && this->src->used >= 4) {
uint8_t f;
int16_t ss;
#if BYTE_ORDER == BIG_ENDIAN
ss = *(const int16_t*)s;
s = audio_stream_add_outp(this->src, s, 2);
f = pcm2adpcm_step(mc, ss);
ss = *(const int16_t*)s;
#else
ss = (s[0] << 8) | s[1];
s = audio_stream_add_outp(this->src, s, 2);
f = pcm2adpcm_step(mc, ss);
ss = (s[0] << 8) | s[1];
#endif
f |= pcm2adpcm_step(mc, ss) << 4;
*d = f;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 2);
}
}
dst->inp = d;
this->src->outp = s;
return 0;
}
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(this->prev, this->src, max_used * 2)))
return err;
m = dst->end - dst->start;
m = min(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 = ((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;
f |= pcm2adpcm_step(mc, ss) << 4;
*d = f;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 1);
}
} else {
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;
f |= pcm2adpcm_step(mc, ss) << 4;
*d = f;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 1);
}
}
dst->inp = d;
this->src->outp = s;
return 0;
}
/*
* OkiADPCM -> signed 16bit linear PCM
*/
static inline int16_t
adpcm2pcm_step(struct msm6258_codecvar *mc, uint8_t b)
{
int estim = (int)mc->mc_estim;
mc->mc_amp += adpcm_estim[estim] * adpcm_estimindex[b];
estim += adpcm_estimstep[b];
if (estim < 0)
estim = 0;
else if (estim > 48)
estim = 48;
mc->mc_estim = estim;
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(this->prev, this->src, max_used / 4)))
return err;
m = (dst->end - dst->start) & ~3;
m = min(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);
#if BYTE_ORDER == LITTLE_ENDIAN
*(int16_t*)d = s1;
d = audio_stream_add_inp(dst, d, 2);
*(int16_t*)d = s2;
#else
d[0] = s1;
d[1] = s1 >> 8;
d = audio_stream_add_inp(dst, d, 2);
d[0] = s2;
d[1] = s2 >> 8;
#endif
d = audio_stream_add_inp(dst, d, 2);
s = audio_stream_add_outp(this->src, s, 1);
}
} else {
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);
#if BYTE_ORDER == BIG_ENDIAN
*(int16_t*)d = s1;
d = audio_stream_add_inp(dst, d, 2);
*(int16_t*)d = s2;
#else
d[1] = s1;
d[0] = s1 >> 8;
d = audio_stream_add_inp(dst, d, 2);
d[1] = s2;
d[0] = s2 >> 8;
#endif
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)
{
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(this->prev, this->src, max_used / 2)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(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);
d[0] = s1 / 266;
d = audio_stream_add_inp(dst, d, 1);
d[0] = s2 / 266;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 1);
}
} else {
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);
d[0] = (s1 / 266) ^ 0x80;
d = audio_stream_add_inp(dst, d, 1);
d[0] = (s2 / 266) ^ 0x80;
d = audio_stream_add_inp(dst, d, 1);
s = audio_stream_add_outp(this->src, s, 1);
}
}
dst->inp = d;
this->src->outp = s;
return 0;
}