8a962f23f2
the original ad-audiomp branch notes: Add MP locking to the audio drivers. Making the audio drivers MP safe is necessary before efforts can be made to make the VM system MP safe. The are two locks per device instance, an ISR lock and a character device lock. The ISR lock replaces calls to splaudio()/splx(), and will be held across calls to device methods which were called at splaudio() before (e.g. trigger_output). The character device lock is held across calls to nearly all of the methods, excluding some only used for initialization, e.g. get_locks. Welcome to 5.99.57.
395 lines
10 KiB
C
395 lines
10 KiB
C
/* $NetBSD: msm6258.c,v 1.17 2011/11/23 23:07:32 jmcneill 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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*
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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 1.17 2011/11/23 23:07:32 jmcneill Exp $");
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#include <sys/systm.h>
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#include <sys/device.h>
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#include <sys/kmem.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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(struct audio_softc *,
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int (*)(struct audio_softc *, 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(struct audio_softc *asc,
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int (*fetch_to)(struct audio_softc *, stream_fetcher_t *, audio_stream_t *, int))
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{
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struct msm6258_codecvar *this;
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this = kmem_alloc(sizeof(struct msm6258_codecvar), KM_SLEEP);
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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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kmem_free(this, sizeof(struct msm6258_codecvar));
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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(struct audio_softc *, 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(sc, name##_fetch_to); \
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} \
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static int \
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name##_fetch_to(struct audio_softc *asc, 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(asc, 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(asc, 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(asc, 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(asc, 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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