395 lines
10 KiB
C
395 lines
10 KiB
C
/* $NetBSD: msm6258.c,v 1.15 2005/12/24 20:27:30 perry Exp $ */
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/*
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* Copyright (c) 2001 Tetsuya Isaki. All rights reserved.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions
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* are met:
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* 1. Redistributions of source code must retain the above copyright
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* notice, this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright
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* notice, this list of conditions and the following disclaimer in the
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* documentation and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
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* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
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* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
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* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
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* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
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* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
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* SUCH DAMAGE.
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*/
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/*
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* OKI MSM6258 ADPCM voice synthesizer codec.
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*/
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#include <sys/cdefs.h>
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__KERNEL_RCSID(0, "$NetBSD: msm6258.c,v 1.15 2005/12/24 20:27:30 perry Exp $");
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#include <sys/systm.h>
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#include <sys/device.h>
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#include <sys/malloc.h>
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#include <sys/select.h>
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#include <sys/audioio.h>
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#include <dev/audio_if.h>
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#include <dev/auconv.h>
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#include <dev/audiovar.h>
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#include <dev/mulaw.h>
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#include <dev/ic/msm6258var.h>
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struct msm6258_codecvar {
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stream_filter_t base;
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short mc_amp;
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char mc_estim;
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};
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static stream_filter_t *msm6258_factory
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(int (*)(stream_fetcher_t *, audio_stream_t *, int));
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static void msm6258_dtor(struct stream_filter *);
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static inline uint8_t pcm2adpcm_step(struct msm6258_codecvar *, int16_t);
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static inline int16_t adpcm2pcm_step(struct msm6258_codecvar *, uint8_t);
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static const int adpcm_estimindex[16] = {
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2, 6, 10, 14, 18, 22, 26, 30,
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-2, -6, -10, -14, -18, -22, -26, -30
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};
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static const int adpcm_estim[49] = {
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16, 17, 19, 21, 23, 25, 28, 31, 34, 37,
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41, 45, 50, 55, 60, 66, 73, 80, 88, 97,
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107, 118, 130, 143, 157, 173, 190, 209, 230, 253,
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279, 307, 337, 371, 408, 449, 494, 544, 598, 658,
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724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552
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};
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static const int adpcm_estimstep[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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static stream_filter_t *
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msm6258_factory(int (*fetch_to)(stream_fetcher_t *, audio_stream_t *, int))
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{
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struct msm6258_codecvar *this;
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this = malloc(sizeof(*this), M_DEVBUF, M_WAITOK | M_ZERO);
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this->base.base.fetch_to = fetch_to;
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this->base.dtor = msm6258_dtor;
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this->base.set_fetcher = stream_filter_set_fetcher;
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this->base.set_inputbuffer = stream_filter_set_inputbuffer;
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return &this->base;
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}
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static void
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msm6258_dtor(struct stream_filter *this)
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{
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if (this != NULL)
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free(this, M_DEVBUF);
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}
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/*
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* signed 16bit linear PCM -> OkiADPCM
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*/
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static inline uint8_t
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pcm2adpcm_step(struct msm6258_codecvar *mc, int16_t a)
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{
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int estim = (int)mc->mc_estim;
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int df;
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short dl, c;
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uint8_t b;
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uint8_t s;
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df = a - mc->mc_amp;
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dl = adpcm_estim[estim];
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c = (df / 16) * 8 / dl;
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if (df < 0) {
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b = (unsigned char)(-c) / 2;
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s = 0x08;
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} else {
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b = (unsigned char)(c) / 2;
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s = 0;
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}
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if (b > 7)
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b = 7;
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s |= b;
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mc->mc_amp += (short)(adpcm_estimindex[(int)s] * dl);
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estim += adpcm_estimstep[b];
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if (estim < 0)
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estim = 0;
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else if (estim > 48)
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estim = 48;
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mc->mc_estim = estim;
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return s;
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}
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#define DEFINE_FILTER(name) \
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static int \
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name##_fetch_to(stream_fetcher_t *, audio_stream_t *, int); \
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stream_filter_t * \
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name(struct audio_softc *sc, const audio_params_t *from, \
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const audio_params_t *to) \
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{ \
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return msm6258_factory(name##_fetch_to); \
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} \
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static int \
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name##_fetch_to(stream_fetcher_t *self, audio_stream_t *dst, int max_used)
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DEFINE_FILTER(msm6258_slinear16_to_adpcm)
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{
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stream_filter_t *this;
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struct msm6258_codecvar *mc;
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uint8_t *d;
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const uint8_t *s;
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int m, err, enc_src;
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this = (stream_filter_t *)self;
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mc = (struct msm6258_codecvar *)self;
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if ((err = this->prev->fetch_to(this->prev, this->src, max_used * 4)))
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return err;
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m = dst->end - dst->start;
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m = min(m, max_used);
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d = dst->inp;
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s = this->src->outp;
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enc_src = this->src->param.encoding;
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if (enc_src == AUDIO_ENCODING_SLINEAR_LE) {
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while (dst->used < m && this->src->used >= 4) {
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uint8_t f;
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int16_t ss;
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#if BYTE_ORDER == LITTLE_ENDIAN
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ss = *(const int16_t*)s;
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s = audio_stream_add_outp(this->src, s, 2);
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f = pcm2adpcm_step(mc, ss);
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ss = *(const int16_t*)s;
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#else
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ss = (s[1] << 8) | s[0];
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s = audio_stream_add_outp(this->src, s, 2);
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f = pcm2adpcm_step(mc, ss);
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ss = (s[1] << 8) | s[0];
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#endif
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f |= pcm2adpcm_step(mc, ss) << 4;
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*d = f;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 2);
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}
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} else {
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while (dst->used < m && this->src->used >= 4) {
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uint8_t f;
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int16_t ss;
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#if BYTE_ORDER == BIG_ENDIAN
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ss = *(const int16_t*)s;
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s = audio_stream_add_outp(this->src, s, 2);
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f = pcm2adpcm_step(mc, ss);
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ss = *(const int16_t*)s;
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#else
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ss = (s[0] << 8) | s[1];
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s = audio_stream_add_outp(this->src, s, 2);
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f = pcm2adpcm_step(mc, ss);
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ss = (s[0] << 8) | s[1];
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#endif
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f |= pcm2adpcm_step(mc, ss) << 4;
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*d = f;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 2);
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}
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}
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dst->inp = d;
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this->src->outp = s;
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return 0;
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}
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DEFINE_FILTER(msm6258_linear8_to_adpcm)
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{
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stream_filter_t *this;
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struct msm6258_codecvar *mc;
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uint8_t *d;
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const uint8_t *s;
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int m, err, enc_src;
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this = (stream_filter_t *)self;
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mc = (struct msm6258_codecvar *)self;
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if ((err = this->prev->fetch_to(this->prev, this->src, max_used * 2)))
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return err;
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m = dst->end - dst->start;
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m = min(m, max_used);
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d = dst->inp;
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s = this->src->outp;
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enc_src = this->src->param.encoding;
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if (enc_src == AUDIO_ENCODING_SLINEAR_LE) {
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while (dst->used < m && this->src->used >= 4) {
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uint8_t f;
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int16_t ss;
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ss = ((int16_t)s[0]) * 256;
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s = audio_stream_add_outp(this->src, s, 1);
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f = pcm2adpcm_step(mc, ss);
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ss = ((int16_t)s[0]) * 256;
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f |= pcm2adpcm_step(mc, ss) << 4;
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*d = f;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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} else {
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while (dst->used < m && this->src->used >= 4) {
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uint8_t f;
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int16_t ss;
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ss = ((int16_t)(s[0] ^ 0x80)) * 256;
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s = audio_stream_add_outp(this->src, s, 1);
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f = pcm2adpcm_step(mc, ss);
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ss = ((int16_t)(s[0] ^ 0x80)) * 256;
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f |= pcm2adpcm_step(mc, ss) << 4;
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*d = f;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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}
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dst->inp = d;
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this->src->outp = s;
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return 0;
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}
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/*
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* OkiADPCM -> signed 16bit linear PCM
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*/
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static inline int16_t
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adpcm2pcm_step(struct msm6258_codecvar *mc, uint8_t b)
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{
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int estim = (int)mc->mc_estim;
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mc->mc_amp += adpcm_estim[estim] * adpcm_estimindex[b];
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estim += adpcm_estimstep[b];
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if (estim < 0)
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estim = 0;
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else if (estim > 48)
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estim = 48;
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mc->mc_estim = estim;
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return mc->mc_amp;
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}
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DEFINE_FILTER(msm6258_adpcm_to_slinear16)
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{
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stream_filter_t *this;
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struct msm6258_codecvar *mc;
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uint8_t *d;
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const uint8_t *s;
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int m, err, enc_dst;
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this = (stream_filter_t *)self;
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mc = (struct msm6258_codecvar *)self;
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max_used = (max_used + 3) & ~3; /* round up multiple of 4 */
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if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 4)))
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return err;
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m = (dst->end - dst->start) & ~3;
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m = min(m, max_used);
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d = dst->inp;
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s = this->src->outp;
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enc_dst = dst->param.encoding;
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if (enc_dst == AUDIO_ENCODING_SLINEAR_LE) {
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while (dst->used < m && this->src->used >= 1) {
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uint8_t a;
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int16_t s1, s2;
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a = s[0];
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s1 = adpcm2pcm_step(mc, a & 0x0f);
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s2 = adpcm2pcm_step(mc, a >> 4);
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#if BYTE_ORDER == LITTLE_ENDIAN
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*(int16_t*)d = s1;
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d = audio_stream_add_inp(dst, d, 2);
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*(int16_t*)d = s2;
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#else
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d[0] = s1;
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d[1] = s1 >> 8;
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d = audio_stream_add_inp(dst, d, 2);
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d[0] = s2;
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d[1] = s2 >> 8;
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#endif
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d = audio_stream_add_inp(dst, d, 2);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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} else {
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while (dst->used < m && this->src->used >= 1) {
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uint8_t a;
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int16_t s1, s2;
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a = s[0];
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s1 = adpcm2pcm_step(mc, a & 0x0f);
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s2 = adpcm2pcm_step(mc, a >> 4);
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#if BYTE_ORDER == BIG_ENDIAN
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*(int16_t*)d = s1;
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d = audio_stream_add_inp(dst, d, 2);
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*(int16_t*)d = s2;
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#else
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d[1] = s1;
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d[0] = s1 >> 8;
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d = audio_stream_add_inp(dst, d, 2);
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d[1] = s2;
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d[0] = s2 >> 8;
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#endif
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d = audio_stream_add_inp(dst, d, 2);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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}
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dst->inp = d;
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this->src->outp = s;
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return 0;
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}
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DEFINE_FILTER(msm6258_adpcm_to_linear8)
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{
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stream_filter_t *this;
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struct msm6258_codecvar *mc;
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uint8_t *d;
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const uint8_t *s;
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int m, err, enc_dst;
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this = (stream_filter_t *)self;
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mc = (struct msm6258_codecvar *)self;
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max_used = (max_used + 1) & ~1; /* round up multiple of 4 */
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if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 2)))
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return err;
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m = (dst->end - dst->start) & ~1;
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m = min(m, max_used);
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d = dst->inp;
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s = this->src->outp;
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enc_dst = dst->param.encoding;
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if (enc_dst == AUDIO_ENCODING_SLINEAR_LE) {
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while (dst->used < m && this->src->used >= 1) {
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uint8_t a;
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int16_t s1, s2;
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a = s[0];
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s1 = adpcm2pcm_step(mc, a & 0x0f);
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s2 = adpcm2pcm_step(mc, a >> 4);
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d[0] = s1 / 266;
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d = audio_stream_add_inp(dst, d, 1);
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d[0] = s2 / 266;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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} else {
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while (dst->used < m && this->src->used >= 1) {
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uint8_t a;
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int16_t s1, s2;
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a = s[0];
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s1 = adpcm2pcm_step(mc, a & 0x0f);
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s2 = adpcm2pcm_step(mc, a >> 4);
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d[0] = (s1 / 266) ^ 0x80;
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d = audio_stream_add_inp(dst, d, 1);
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d[0] = (s2 / 266) ^ 0x80;
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d = audio_stream_add_inp(dst, d, 1);
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s = audio_stream_add_outp(this->src, s, 1);
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}
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}
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dst->inp = d;
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this->src->outp = s;
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return 0;
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}
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