File: [cvs.NetBSD.org] / src / sys / dev / Attic / auconv.c (download)
Revision 1.25.36.1, Wed Aug 9 06:21:18 2017 UTC (6 years, 8 months ago) by snj
Branch: netbsd-7-1
CVS Tags: netbsd-7-1-2-RELEASE, netbsd-7-1-1-RELEASE
Changes since 1.25: +3 -3
lines
Pull up following revision(s) (requested by nat in ticket #1460):
sys/dev/auconv.c: revision 1.30 via patch
sys/dev/ic/ac97.c: revision 1.97 via patch
sys/dev/pci/azalia_codec.c: revision 1.81 via patch
Mixer device bounds checking.
Analysis by Ilja van Sprundel.
|
/* $NetBSD: auconv.c,v 1.25.36.1 2017/08/09 06:21:18 snj Exp $ */
/*
* Copyright (c) 1996 The NetBSD Foundation, Inc.
* 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. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the Computer Systems
* Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``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 REGENTS OR CONTRIBUTORS 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.
*
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: auconv.c,v 1.25.36.1 2017/08/09 06:21:18 snj Exp $");
#include <sys/types.h>
#include <sys/audioio.h>
#include <sys/device.h>
#include <sys/errno.h>
#include <sys/malloc.h>
#include <sys/null.h>
#include <sys/systm.h>
#include <dev/audio_if.h>
#include <dev/auconv.h>
#include <dev/mulaw.h>
#include <machine/limits.h>
#ifndef _KERNEL
#include <stddef.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <stdbool.h>
#endif
#include <aurateconv.h> /* generated by config(8) */
#include <mulaw.h> /* generated by config(8) */
/* #define AUCONV_DEBUG */
#ifdef AUCONV_DEBUG
# define DPRINTF(x) printf x
#else
# define DPRINTF(x)
#endif
#if NAURATECONV > 0
static int auconv_rateconv_supportable(u_int, u_int, u_int);
static int auconv_rateconv_check_channels(const struct audio_format *, int,
int, const audio_params_t *,
stream_filter_list_t *);
static int auconv_rateconv_check_rates(const struct audio_format *, int,
int, const audio_params_t *,
audio_params_t *,
stream_filter_list_t *);
#endif
#ifdef AUCONV_DEBUG
static void auconv_dump_formats(const struct audio_format *, int);
#endif
static void auconv_dump_params(const audio_params_t *);
static int auconv_exact_match(const struct audio_format *, int, int,
const struct audio_params *);
static u_int auconv_normalize_encoding(u_int, u_int);
static int auconv_is_supported_rate(const struct audio_format *, u_int);
static int auconv_add_encoding(int, int, int, struct audio_encoding_set **,
int *);
#ifdef _KERNEL
#define AUCONV_MALLOC(size) malloc(size, M_DEVBUF, M_NOWAIT)
#define AUCONV_REALLOC(p, size) realloc(p, size, M_DEVBUF, M_NOWAIT)
#define AUCONV_FREE(p) free(p, M_DEVBUF)
#else
#define AUCONV_MALLOC(size) malloc(size)
#define AUCONV_REALLOC(p, size) realloc(p, size)
#define AUCONV_FREE(p) free(p)
#endif
struct audio_encoding_set {
int size;
audio_encoding_t items[1];
};
#define ENCODING_SET_SIZE(n) (offsetof(struct audio_encoding_set, items) \
+ sizeof(audio_encoding_t) * (n))
struct conv_table {
u_int encoding;
u_int validbits;
u_int precision;
stream_filter_factory_t *play_conv;
stream_filter_factory_t *rec_conv;
};
/*
* SLINEAR-16 or SLINEAR-24 should precede in a table because
* aurateconv supports only SLINEAR.
*/
static const struct conv_table s8_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
change_sign8, change_sign8},
{0, 0, 0, NULL, NULL}};
static const struct conv_table u8_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
change_sign8, change_sign8},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
linear8_to_linear16, linear16_to_linear8},
{0, 0, 0, NULL, NULL}};
static const struct conv_table s16le_table[] = {
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
swap_bytes, swap_bytes},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
change_sign16, change_sign16},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
swap_bytes_change_sign16, swap_bytes_change_sign16},
{0, 0, 0, NULL, NULL}};
static const struct conv_table s16be_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
swap_bytes, swap_bytes},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
change_sign16, change_sign16},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
swap_bytes_change_sign16, swap_bytes_change_sign16},
{0, 0, 0, NULL, NULL}};
static const struct conv_table u16le_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
change_sign16, change_sign16},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
swap_bytes, swap_bytes},
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
swap_bytes_change_sign16, swap_bytes_change_sign16},
{0, 0, 0, NULL, NULL}};
static const struct conv_table u16be_table[] = {
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
change_sign16, change_sign16},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
swap_bytes, swap_bytes},
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
swap_bytes_change_sign16, swap_bytes_change_sign16},
{0, 0, 0, NULL, NULL}};
#if NMULAW > 0
static const struct conv_table mulaw_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
mulaw_to_linear16, linear16_to_mulaw},
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
mulaw_to_linear16, linear16_to_mulaw},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
mulaw_to_linear16, linear16_to_mulaw},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
mulaw_to_linear16, linear16_to_mulaw},
{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
mulaw_to_linear8, linear8_to_mulaw},
{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
mulaw_to_linear8, linear8_to_mulaw},
{0, 0, 0, NULL, NULL}};
static const struct conv_table alaw_table[] = {
{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
alaw_to_linear16, linear16_to_alaw},
{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
alaw_to_linear16, linear16_to_alaw},
{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
alaw_to_linear16, linear16_to_alaw},
{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
alaw_to_linear16, linear16_to_alaw},
{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
alaw_to_linear8, linear8_to_alaw},
{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
alaw_to_linear8, linear8_to_alaw},
{0, 0, 0, NULL, NULL}};
#endif
#ifdef AUCONV_DEBUG
static const char *encoding_dbg_names[] = {
"none", AudioEmulaw, AudioEalaw, "pcm16",
"pcm8", AudioEadpcm, AudioEslinear_le, AudioEslinear_be,
AudioEulinear_le, AudioEulinear_be,
AudioEslinear, AudioEulinear,
AudioEmpeg_l1_stream, AudioEmpeg_l1_packets,
AudioEmpeg_l1_system, AudioEmpeg_l2_stream,
AudioEmpeg_l2_packets, AudioEmpeg_l2_system,
AudioEac3
};
#endif
void
stream_filter_set_fetcher(stream_filter_t *this, stream_fetcher_t *p)
{
this->prev = p;
}
void
stream_filter_set_inputbuffer(stream_filter_t *this, audio_stream_t *stream)
{
this->src = stream;
}
stream_filter_t *
auconv_nocontext_filter_factory(
int (*fetch_to)(struct audio_softc *, stream_fetcher_t *,
audio_stream_t *, int))
{
stream_filter_t *this;
this = AUCONV_MALLOC(sizeof(stream_filter_t));
if (this == NULL)
return NULL;
this->base.fetch_to = fetch_to;
this->dtor = auconv_nocontext_filter_dtor;
this->set_fetcher = stream_filter_set_fetcher;
this->set_inputbuffer = stream_filter_set_inputbuffer;
this->prev = NULL;
this->src = NULL;
return this;
}
void
auconv_nocontext_filter_dtor(struct stream_filter *this)
{
if (this != NULL)
AUCONV_FREE(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 *sc, const audio_params_t *from, \
const audio_params_t *to) \
{ \
return auconv_nocontext_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(change_sign8)
{
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = *s ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
return 0;
}
DEFINE_FILTER(change_sign16)
{
stream_filter_t *this;
int m, err, enc;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
enc = dst->param.encoding;
if (enc == AUDIO_ENCODING_SLINEAR_LE
|| enc == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
d[0] = s[0];
d[1] = s[1] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
d[0] = s[0] ^ 0x80;
d[1] = s[1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
}
return 0;
}
DEFINE_FILTER(swap_bytes)
{
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
d[0] = s[1];
d[1] = s[0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
return 0;
}
DEFINE_FILTER(swap_bytes_change_sign16)
{
stream_filter_t *this;
int m, err, enc;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
enc = dst->param.encoding;
if (enc == AUDIO_ENCODING_SLINEAR_LE
|| enc == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
d[0] = s[1];
d[1] = s[0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 2, m) {
d[0] = s[1] ^ 0x80;
d[1] = s[0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
}
return 0;
}
DEFINE_FILTER(linear8_to_linear16)
{
stream_filter_t *this;
int m, err, enc_dst, enc_src;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used / 2)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
enc_dst = dst->param.encoding;
enc_src = this->src->param.encoding;
if ((enc_src == AUDIO_ENCODING_SLINEAR_LE
&& enc_dst == AUDIO_ENCODING_SLINEAR_LE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_LE
&& enc_dst == AUDIO_ENCODING_ULINEAR_LE)) {
/*
* slinear8 -> slinear16_le
* ulinear8 -> ulinear16_le
*/
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = 0;
d[1] = s[0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE
&& enc_dst == AUDIO_ENCODING_SLINEAR_BE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_LE
&& enc_dst == AUDIO_ENCODING_ULINEAR_BE)) {
/*
* slinear8 -> slinear16_be
* ulinear8 -> ulinear16_be
*/
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = s[0];
d[1] = 0;
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE
&& enc_dst == AUDIO_ENCODING_ULINEAR_LE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_LE
&& enc_dst == AUDIO_ENCODING_SLINEAR_LE)) {
/*
* slinear8 -> ulinear16_le
* ulinear8 -> slinear16_le
*/
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = 0;
d[1] = s[0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
/*
* slinear8 -> ulinear16_be
* ulinear8 -> slinear16_be
*/
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = s[0] ^ 0x80;
d[1] = 0;
} FILTER_LOOP_EPILOGUE(this->src, dst);
}
return 0;
}
DEFINE_FILTER(linear16_to_linear8)
{
stream_filter_t *this;
int m, err, enc_src, enc_dst;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(sc, this->prev, this->src, max_used * 2)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
enc_dst = dst->param.encoding;
enc_src = this->src->param.encoding;
if ((enc_src == AUDIO_ENCODING_SLINEAR_LE
&& enc_dst == AUDIO_ENCODING_SLINEAR_LE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_LE
&& enc_dst == AUDIO_ENCODING_ULINEAR_LE)) {
/*
* slinear16_le -> slinear8
* ulinear16_le -> ulinear8
*/
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = s[1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else if ((enc_src == AUDIO_ENCODING_SLINEAR_LE
&& enc_dst == AUDIO_ENCODING_ULINEAR_LE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_LE
&& enc_dst == AUDIO_ENCODING_SLINEAR_LE)) {
/*
* slinear16_le -> ulinear8
* ulinear16_le -> slinear8
*/
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = s[1] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else if ((enc_src == AUDIO_ENCODING_SLINEAR_BE
&& enc_dst == AUDIO_ENCODING_SLINEAR_LE)
|| (enc_src == AUDIO_ENCODING_ULINEAR_BE
&& enc_dst == AUDIO_ENCODING_ULINEAR_LE)) {
/*
* slinear16_be -> slinear8
* ulinear16_be -> ulinear8
*/
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = s[0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
/*
* slinear16_be -> ulinear8
* ulinear16_be -> slinear8
*/
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = s[0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
}
return 0;
}
/**
* Set appropriate parameters in `param,' and return the index in
* the hardware capability array `formats.'
*
* @param formats [IN] An array of formats which a hardware can support.
* @param nformats [IN] The number of elements of the array.
* @param mode [IN] Either AUMODE_PLAY or AUMODE_RECORD.
* @param param [IN] Requested format. param->sw_code may be set.
* @param rateconv [IN] true if aurateconv may be used.
* @param list [OUT] stream_filters required for param.
* @return The index of selected audio_format entry. -1 if the device
* can not support the specified param.
*/
int
auconv_set_converter(const struct audio_format *formats, int nformats,
int mode, const audio_params_t *param, int rateconv,
stream_filter_list_t *list)
{
audio_params_t work;
const struct conv_table *table;
stream_filter_factory_t *conv;
int enc;
int i, j;
#ifdef AUCONV_DEBUG
DPRINTF(("%s: ENTER rateconv=%d\n", __func__, rateconv));
auconv_dump_formats(formats, nformats);
#endif
enc = auconv_normalize_encoding(param->encoding, param->precision);
/* check support by native format */
i = auconv_exact_match(formats, nformats, mode, param);
if (i >= 0) {
DPRINTF(("%s: LEAVE with %d (exact)\n", __func__, i));
return i;
}
#if NAURATECONV > 0
/* native format with aurateconv */
DPRINTF(("%s: native with aurateconv\n", __func__));
if (rateconv
&& auconv_rateconv_supportable(enc, param->precision,
param->validbits)) {
i = auconv_rateconv_check_channels(formats, nformats,
mode, param, list);
if (i >= 0) {
DPRINTF(("%s: LEAVE with %d (aurateconv1)\n", __func__, i));
return i;
}
}
#endif
/* check for emulation */
DPRINTF(("%s: encoding emulation\n", __func__));
table = NULL;
switch (enc) {
case AUDIO_ENCODING_SLINEAR_LE:
if (param->precision == 8)
table = s8_table;
else if (param->precision == 16)
table = s16le_table;
break;
case AUDIO_ENCODING_SLINEAR_BE:
if (param->precision == 8)
table = s8_table;
else if (param->precision == 16)
table = s16be_table;
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (param->precision == 8)
table = u8_table;
else if (param->precision == 16)
table = u16le_table;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (param->precision == 8)
table = u8_table;
else if (param->precision == 16)
table = u16be_table;
break;
#if NMULAW > 0
case AUDIO_ENCODING_ULAW:
table = mulaw_table;
break;
case AUDIO_ENCODING_ALAW:
table = alaw_table;
break;
#endif
}
if (table == NULL) {
DPRINTF(("%s: LEAVE with -1 (no-emultable)\n", __func__));
return -1;
}
work = *param;
for (j = 0; table[j].precision != 0; j++) {
work.encoding = table[j].encoding;
work.precision = table[j].precision;
work.validbits = table[j].validbits;
i = auconv_exact_match(formats, nformats, mode, &work);
if (i >= 0) {
conv = mode == AUMODE_PLAY
? table[j].play_conv : table[j].rec_conv;
list->append(list, conv, &work);
DPRINTF(("%s: LEAVE with %d (emultable)\n", __func__, i));
return i;
}
}
/* not found */
#if NAURATECONV > 0
/* emulation with aurateconv */
DPRINTF(("%s: encoding emulation with aurateconv\n", __func__));
if (!rateconv) {
DPRINTF(("%s: LEAVE with -1 (no-rateconv)\n", __func__));
return -1;
}
work = *param;
for (j = 0; table[j].precision != 0; j++) {
if (!auconv_rateconv_supportable(table[j].encoding,
table[j].precision,
table[j].validbits))
continue;
work.encoding = table[j].encoding;
work.precision = table[j].precision;
work.validbits = table[j].validbits;
i = auconv_rateconv_check_channels(formats, nformats,
mode, &work, list);
if (i >= 0) {
/* work<=>hw conversion is already registered */
conv = mode == AUMODE_PLAY
? table[j].play_conv : table[j].rec_conv;
/* register userland<=>work conversion */
list->append(list, conv, &work);
DPRINTF(("%s: LEAVE with %d (rateconv2)\n", __func__, i));
return i;
}
}
#endif
DPRINTF(("%s: LEAVE with -1 (bottom)\n", __func__));
return -1;
}
#if NAURATECONV > 0
static int
auconv_rateconv_supportable(u_int encoding, u_int precision, u_int validbits)
{
if (encoding != AUDIO_ENCODING_SLINEAR_LE
&& encoding != AUDIO_ENCODING_SLINEAR_BE)
return false;
if (precision != 16 && precision != 24 && precision != 32)
return false;
if (precision < validbits)
return false;
return true;
}
static int
auconv_rateconv_check_channels(const struct audio_format *formats, int nformats,
int mode, const audio_params_t *param,
stream_filter_list_t *list)
{
audio_params_t hw_param;
int ind, n;
hw_param = *param;
/* check for the specified number of channels */
ind = auconv_rateconv_check_rates(formats, nformats, mode, param,
&hw_param, list);
if (ind >= 0)
return ind;
/* check for larger numbers */
for (n = param->channels + 1; n <= AUDIO_MAX_CHANNELS; n++) {
hw_param.channels = n;
ind = auconv_rateconv_check_rates(formats, nformats, mode,
param, &hw_param, list);
if (ind >= 0)
return ind;
}
/* check for stereo:monaural conversion */
if (param->channels == 2) {
hw_param.channels = 1;
ind = auconv_rateconv_check_rates(formats, nformats, mode,
param, &hw_param, list);
if (ind >= 0)
return ind;
}
return -1;
}
static int
auconv_rateconv_check_rates(const struct audio_format *formats, int nformats,
int mode, const audio_params_t *param,
audio_params_t *hw_param, stream_filter_list_t *list)
{
int ind, i, j, enc, f_enc;
u_int rate, minrate, maxrate, orig_rate;
/* exact match */
ind = auconv_exact_match(formats, nformats, mode, hw_param);
if (ind >= 0)
goto found;
/* determine min/max of specified encoding/precision/channels */
minrate = UINT_MAX;
maxrate = 0;
enc = auconv_normalize_encoding(param->encoding,
param->precision);
for (i = 0; i < nformats; i++) {
if (!AUFMT_IS_VALID(&formats[i]))
continue;
if ((formats[i].mode & mode) == 0)
continue;
f_enc = auconv_normalize_encoding(formats[i].encoding,
formats[i].precision);
if (f_enc != enc)
continue;
if (formats[i].validbits != hw_param->validbits)
continue;
if (formats[i].precision != hw_param->precision)
continue;
if (formats[i].channels != hw_param->channels)
continue;
if (formats[i].frequency_type == 0) {
if (formats[i].frequency[0] < minrate)
minrate = formats[i].frequency[0];
if (formats[i].frequency[1] > maxrate)
maxrate = formats[i].frequency[1];
} else {
for (j = 0; j < formats[i].frequency_type; j++) {
if (formats[i].frequency[j] < minrate)
minrate = formats[i].frequency[j];
if (formats[i].frequency[j] > maxrate)
maxrate = formats[i].frequency[j];
}
}
}
if (maxrate == 0)
return -1;
/* try multiples of sample_rate */
orig_rate = hw_param->sample_rate;
for (i = 2; (rate = param->sample_rate * i) <= maxrate; i++) {
hw_param->sample_rate = rate;
ind = auconv_exact_match(formats, nformats, mode, hw_param);
if (ind >= 0)
goto found;
}
hw_param->sample_rate = param->sample_rate >= minrate
? maxrate : minrate;
ind = auconv_exact_match(formats, nformats, mode, hw_param);
if (ind >= 0)
goto found;
hw_param->sample_rate = orig_rate;
return -1;
found:
list->append(list, aurateconv, hw_param);
return ind;
}
#endif /* NAURATECONV */
#ifdef AUCONV_DEBUG
static void
auconv_dump_formats(const struct audio_format *formats, int nformats)
{
const struct audio_format *f;
int i, j;
for (i = 0; i < nformats; i++) {
f = &formats[i];
printf("[%2d]: mode=", i);
if (!AUFMT_IS_VALID(f)) {
printf("INVALID");
} else if (f->mode == AUMODE_PLAY) {
printf("PLAY");
} else if (f->mode == AUMODE_RECORD) {
printf("RECORD");
} else if (f->mode == (AUMODE_PLAY | AUMODE_RECORD)) {
printf("PLAY|RECORD");
} else {
printf("0x%x", f->mode);
}
printf(" enc=%s", encoding_dbg_names[f->encoding]);
printf(" %u/%ubit", f->validbits, f->precision);
printf(" %uch", f->channels);
printf(" channel_mask=");
if (f->channel_mask == AUFMT_MONAURAL) {
printf("MONAURAL");
} else if (f->channel_mask == AUFMT_STEREO) {
printf("STEREO");
} else if (f->channel_mask == AUFMT_SURROUND4) {
printf("SURROUND4");
} else if (f->channel_mask == AUFMT_DOLBY_5_1) {
printf("DOLBY5.1");
} else {
printf("0x%x", f->channel_mask);
}
if (f->frequency_type == 0) {
printf(" %uHz-%uHz", f->frequency[0],
f->frequency[1]);
} else {
printf(" %uHz", f->frequency[0]);
for (j = 1; j < f->frequency_type; j++)
printf(",%uHz", f->frequency[j]);
}
printf("\n");
}
}
static void
auconv_dump_params(const audio_params_t *p)
{
printf("enc=%s", encoding_dbg_names[p->encoding]);
printf(" %u/%ubit", p->validbits, p->precision);
printf(" %uch", p->channels);
printf(" %uHz", p->sample_rate);
printf("\n");
}
#else
static void
auconv_dump_params(const audio_params_t *p)
{
}
#endif /* AUCONV_DEBUG */
/**
* a sub-routine for auconv_set_converter()
*/
static int
auconv_exact_match(const struct audio_format *formats, int nformats,
int mode, const audio_params_t *param)
{
int i, enc, f_enc;
DPRINTF(("%s: ENTER: mode=0x%x target:", __func__, mode));
auconv_dump_params(param);
enc = auconv_normalize_encoding(param->encoding,
param->precision);
DPRINTF(("%s: target normalized: %s\n", __func__,
encoding_dbg_names[enc]));
for (i = 0; i < nformats; i++) {
if (!AUFMT_IS_VALID(&formats[i]))
continue;
if ((formats[i].mode & mode) == 0)
continue;
f_enc = auconv_normalize_encoding(formats[i].encoding,
formats[i].precision);
DPRINTF(("%s: format[%d] normalized: %s\n",
__func__, i, encoding_dbg_names[f_enc]));
if (f_enc != enc)
continue;
/**
* XXX we need encoding-dependent check.
* XXX Is to check precision/channels meaningful for
* MPEG encodings?
*/
if (enc != AUDIO_ENCODING_AC3) {
if (formats[i].validbits != param->validbits)
continue;
if (formats[i].precision != param->precision)
continue;
if (formats[i].channels != param->channels)
continue;
}
if (!auconv_is_supported_rate(&formats[i],
param->sample_rate))
continue;
return i;
}
return -1;
}
/**
* a sub-routine for auconv_set_converter()
* SLINEAR ==> SLINEAR_<host-endian>
* ULINEAR ==> ULINEAR_<host-endian>
* SLINEAR_BE 8bit ==> SLINEAR_LE 8bit
* ULINEAR_BE 8bit ==> ULINEAR_LE 8bit
* This should be the same rule as audio_check_params()
*/
static u_int
auconv_normalize_encoding(u_int encoding, u_int precision)
{
int enc;
enc = encoding;
if (enc == AUDIO_ENCODING_SLINEAR_LE)
return enc;
if (enc == AUDIO_ENCODING_ULINEAR_LE)
return enc;
#if BYTE_ORDER == LITTLE_ENDIAN
if (enc == AUDIO_ENCODING_SLINEAR)
return AUDIO_ENCODING_SLINEAR_LE;
else if (enc == AUDIO_ENCODING_ULINEAR)
return AUDIO_ENCODING_ULINEAR_LE;
#else
if (enc == AUDIO_ENCODING_SLINEAR)
enc = AUDIO_ENCODING_SLINEAR_BE;
else if (enc == AUDIO_ENCODING_ULINEAR)
enc = AUDIO_ENCODING_ULINEAR_BE;
#endif
if (precision == 8 && enc == AUDIO_ENCODING_SLINEAR_BE)
return AUDIO_ENCODING_SLINEAR_LE;
if (precision == 8 && enc == AUDIO_ENCODING_ULINEAR_BE)
return AUDIO_ENCODING_ULINEAR_LE;
return enc;
}
/**
* a sub-routine for auconv_set_converter()
*/
static int
auconv_is_supported_rate(const struct audio_format *format, u_int rate)
{
u_int i;
if (format->frequency_type == 0) {
return format->frequency[0] <= rate
&& rate <= format->frequency[1];
}
for (i = 0; i < format->frequency_type; i++) {
if (format->frequency[i] == rate)
return true;
}
return false;
}
/**
* Create an audio_encoding_set besed on hardware capability represented
* by audio_format.
*
* Usage:
* foo_attach(...) {
* :
* if (auconv_create_encodings(formats, nformats,
* &sc->sc_encodings) != 0) {
* // attach failure
* }
*
* @param formats [IN] An array of formats which a hardware can support.
* @param nformats [IN] The number of elements of the array.
* @param encodings [OUT] receives an address of an audio_encoding_set.
* @return errno; 0 for success.
*/
int
auconv_create_encodings(const struct audio_format *formats, int nformats,
struct audio_encoding_set **encodings)
{
struct audio_encoding_set *buf;
int capacity;
int i;
int err;
#define ADD_ENCODING(enc, prec, flags) do { \
err = auconv_add_encoding(enc, prec, flags, &buf, &capacity); \
if (err != 0) goto err_exit; \
} while (/*CONSTCOND*/0)
capacity = 10;
buf = AUCONV_MALLOC(ENCODING_SET_SIZE(capacity));
if (buf == NULL) {
err = ENOMEM;
goto err_exit;
}
buf->size = 0;
for (i = 0; i < nformats; i++) {
if (!AUFMT_IS_VALID(&formats[i]))
continue;
switch (formats[i].encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
ADD_ENCODING(formats[i].encoding,
formats[i].precision, 0);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
#if NMULAW > 0
if (formats[i].precision == 8
|| formats[i].precision == 16) {
ADD_ENCODING(AUDIO_ENCODING_ULAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ALAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
}
#endif
break;
case AUDIO_ENCODING_SLINEAR_BE:
ADD_ENCODING(formats[i].encoding,
formats[i].precision, 0);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
#if NMULAW > 0
if (formats[i].precision == 8
|| formats[i].precision == 16) {
ADD_ENCODING(AUDIO_ENCODING_ULAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ALAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
}
#endif
break;
case AUDIO_ENCODING_ULINEAR_LE:
ADD_ENCODING(formats[i].encoding,
formats[i].precision, 0);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
#if NMULAW > 0
if (formats[i].precision == 8
|| formats[i].precision == 16) {
ADD_ENCODING(AUDIO_ENCODING_ULAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ALAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
}
#endif
break;
case AUDIO_ENCODING_ULINEAR_BE:
ADD_ENCODING(formats[i].encoding,
formats[i].precision, 0);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_BE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ULINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_SLINEAR_LE,
formats[i].precision,
AUDIO_ENCODINGFLAG_EMULATED);
#if NMULAW > 0
if (formats[i].precision == 8
|| formats[i].precision == 16) {
ADD_ENCODING(AUDIO_ENCODING_ULAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
ADD_ENCODING(AUDIO_ENCODING_ALAW, 8,
AUDIO_ENCODINGFLAG_EMULATED);
}
#endif
break;
case AUDIO_ENCODING_ULAW:
case AUDIO_ENCODING_ALAW:
case AUDIO_ENCODING_ADPCM:
case AUDIO_ENCODING_MPEG_L1_STREAM:
case AUDIO_ENCODING_MPEG_L1_PACKETS:
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
case AUDIO_ENCODING_MPEG_L2_STREAM:
case AUDIO_ENCODING_MPEG_L2_PACKETS:
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
case AUDIO_ENCODING_AC3:
ADD_ENCODING(formats[i].encoding,
formats[i].precision, 0);
break;
case AUDIO_ENCODING_SLINEAR:
case AUDIO_ENCODING_ULINEAR:
case AUDIO_ENCODING_LINEAR:
case AUDIO_ENCODING_LINEAR8:
default:
printf("%s: invalid encoding value "
"for audio_format: %d\n",
__func__, formats[i].encoding);
break;
}
}
*encodings = buf;
return 0;
err_exit:
if (buf != NULL)
AUCONV_FREE(buf);
*encodings = NULL;
return err;
}
/**
* a sub-routine for auconv_create_encodings()
*/
static int
auconv_add_encoding(int enc, int prec, int flags,
struct audio_encoding_set **buf, int *capacity)
{
static const char *encoding_names[] = {
NULL, AudioEmulaw, AudioEalaw, NULL,
NULL, AudioEadpcm, AudioEslinear_le, AudioEslinear_be,
AudioEulinear_le, AudioEulinear_be,
AudioEslinear, AudioEulinear,
AudioEmpeg_l1_stream, AudioEmpeg_l1_packets,
AudioEmpeg_l1_system, AudioEmpeg_l2_stream,
AudioEmpeg_l2_packets, AudioEmpeg_l2_system,
AudioEac3
};
struct audio_encoding_set *set;
struct audio_encoding_set *new_buf;
audio_encoding_t *e;
int i;
set = *buf;
/* already has the same encoding? */
e = set->items;
for (i = 0; i < set->size; i++, e++) {
if (e->encoding == enc && e->precision == prec) {
/* overwrite EMULATED flag */
if ((e->flags & AUDIO_ENCODINGFLAG_EMULATED)
&& (flags & AUDIO_ENCODINGFLAG_EMULATED) == 0) {
e->flags &= ~AUDIO_ENCODINGFLAG_EMULATED;
}
return 0;
}
}
/* We don't have the specified one. */
if (set->size >= *capacity) {
new_buf = AUCONV_REALLOC(set,
ENCODING_SET_SIZE(*capacity + 10));
if (new_buf == NULL)
return ENOMEM;
*buf = new_buf;
set = new_buf;
*capacity += 10;
}
e = &set->items[set->size];
e->index = 0;
strlcpy(e->name, encoding_names[enc], MAX_AUDIO_DEV_LEN);
e->encoding = enc;
e->precision = prec;
e->flags = flags;
set->size += 1;
return 0;
}
/**
* Delete an audio_encoding_set created by auconv_create_encodings().
*
* Usage:
* foo_detach(...) {
* :
* auconv_delete_encodings(sc->sc_encodings);
* :
* }
*
* @param encodings [IN] An audio_encoding_set which was created by
* auconv_create_encodings().
* @return errno; 0 for success.
*/
int auconv_delete_encodings(struct audio_encoding_set *encodings)
{
if (encodings != NULL)
AUCONV_FREE(encodings);
return 0;
}
/**
* Copy the element specified by aep->index.
*
* Usage:
* int foo_query_encoding(void *v, audio_encoding_t *aep) {
* struct foo_softc *sc = (struct foo_softc *)v;
* return auconv_query_encoding(sc->sc_encodings, aep);
* }
*
* @param encodings [IN] An audio_encoding_set created by
* auconv_create_encodings().
* @param aep [IN/OUT] resultant audio_encoding_t.
*/
int
auconv_query_encoding(const struct audio_encoding_set *encodings,
audio_encoding_t *aep)
{
if (aep->index < 0 || aep->index >= encodings->size)
return EINVAL;
strlcpy(aep->name, encodings->items[aep->index].name,
MAX_AUDIO_DEV_LEN);
aep->encoding = encodings->items[aep->index].encoding;
aep->precision = encodings->items[aep->index].precision;
aep->flags = encodings->items[aep->index].flags;
return 0;
}
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