0de9e471b6
extern int auconv_set_converter(const struct audio_format *, int, int, struct audio_params *, int); extern int auconv_create_encodings(const struct audio_format *, int, struct audio_encoding_set **); extern int auconv_delete_encodings(struct audio_encoding_set *); extern int auconv_query_encoding(const struct audio_encoding_set *, audio_encoding_t *); These are helper functions for implementing audio_hw_if::set_params() and audio_hw_if::query_encodings().
1033 lines
27 KiB
C
1033 lines
27 KiB
C
/* $NetBSD: auconv.c,v 1.11 2004/11/13 08:08:22 kent Exp $ */
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/*
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* Copyright (c) 1996 The NetBSD Foundation, Inc.
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* 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. All advertising materials mentioning features or use of this software
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* must display the following acknowledgement:
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* This product includes software developed by the Computer Systems
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* Engineering Group at Lawrence Berkeley Laboratory.
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* 4. Neither the name of the University nor of the Laboratory may be used
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* to endorse or promote products derived from this software without
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* specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
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* LIABILITY, 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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#include <sys/cdefs.h>
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__KERNEL_RCSID(0, "$NetBSD: auconv.c,v 1.11 2004/11/13 08:08:22 kent Exp $");
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#include <sys/types.h>
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#include <sys/audioio.h>
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#include <sys/errno.h>
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#include <sys/malloc.h>
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#include <sys/null.h>
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#include <sys/systm.h>
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#include <dev/audio_if.h>
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#include <dev/auconv.h>
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#include <dev/mulaw.h>
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#include <machine/limits.h>
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#ifndef _KERNEL
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#include <stddef.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#endif
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#include <aurateconv.h>
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#include <mulaw.h>
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/* #define AUCONV_DEBUG */
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#if NAURATECONV > 0
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static int auconv_rateconv_supportable(u_int, u_int, u_int);
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static int auconv_rateconv_check_channels(const struct audio_format *, int,
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int, struct audio_params *);
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static int auconv_rateconv_check_rates(const struct audio_format *, int,
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int, struct audio_params *);
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#endif
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#ifdef AUCONV_DEBUG
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static void auconv_dump_formats(const struct audio_format *, int);
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#endif
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static int auconv_exact_match(const struct audio_format *, int, int,
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const struct audio_params *);
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static u_int auconv_normalize_encoding(u_int, u_int);
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static int auconv_is_supported_rate(const struct audio_format *, u_long);
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static int auconv_add_encoding(int, int, int, struct audio_encoding_set **,
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int *);
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#ifdef _KERNEL
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#define AUCONV_MALLOC(size) malloc(size, M_DEVBUF, M_NOWAIT)
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#define AUCONV_REALLOC(p, size) realloc(p, size, M_DEVBUF, M_NOWAIT)
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#define AUCONV_FREE(p) free(p, M_DEVBUF)
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#else
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#define AUCONV_MALLOC(size) malloc(size)
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#define AUCONV_REALLOC(p, size) realloc(p, size)
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#define AUCONV_FREE(p) free(p)
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#define FALSE 0
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#define TRUE 1
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#endif
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struct audio_encoding_set {
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int size;
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audio_encoding_t items[1];
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};
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#define ENCODING_SET_SIZE(n) (offsetof(struct audio_encoding_set, items) \
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+ sizeof(audio_encoding_t) * (n))
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struct conv_table {
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u_int hw_encoding;
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u_int hw_precision;
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u_int hw_subframe;
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void (*play_conv)(void *, u_char *, int);
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void (*rec_conv)(void *, u_char *, int);
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int factor;
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};
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/*
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* SLINEAR-16 or SLINEAR-24 should precede in a table because
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* aurateconv supports only SLINEAR.
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*/
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static const struct conv_table s8_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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linear8_to_linear16_le, linear16_to_linear8_le, 2},
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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linear8_to_linear16_be, linear16_to_linear8_be, 2},
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{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
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change_sign8, change_sign8, 1},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table u8_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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ulinear8_to_slinear16_le, slinear16_to_ulinear8_le, 2},
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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ulinear8_to_slinear16_be, slinear16_to_ulinear8_be, 2},
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{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
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change_sign8, change_sign8, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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linear8_to_linear16_le, linear16_to_linear8_le, 2},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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linear8_to_linear16_be, linear16_to_linear8_be, 2},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table s16le_table[] = {
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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swap_bytes, swap_bytes, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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change_sign16_le, change_sign16_le, 1},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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swap_bytes_change_sign16_be, change_sign16_swap_bytes_be, 1},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table s16be_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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swap_bytes, swap_bytes, 1},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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change_sign16_be, change_sign16_be, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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swap_bytes_change_sign16_le, change_sign16_swap_bytes_le, 1},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table u16le_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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change_sign16_le, change_sign16_le, 1},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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swap_bytes, swap_bytes, 1},
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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swap_bytes_change_sign16_be, change_sign16_swap_bytes_be, 1},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table u16be_table[] = {
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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change_sign16_be, change_sign16_be, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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swap_bytes, swap_bytes, 1},
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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swap_bytes_change_sign16_le, change_sign16_swap_bytes_le, 1},
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{0, 0, 0, NULL, NULL, 0}};
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#if NMULAW > 0
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static const struct conv_table mulaw_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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mulaw_to_slinear16_le, slinear16_to_mulaw_le, 2},
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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mulaw_to_slinear16_be, slinear16_to_mulaw_be, 2},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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mulaw_to_ulinear16_le, ulinear16_to_mulaw_le, 2},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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mulaw_to_ulinear16_be, ulinear16_to_mulaw_be, 2},
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{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
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mulaw_to_slinear8, slinear8_to_mulaw, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
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mulaw_to_ulinear8, ulinear8_to_mulaw, 1},
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{0, 0, 0, NULL, NULL, 0}};
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static const struct conv_table alaw_table[] = {
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{AUDIO_ENCODING_SLINEAR_LE, 16, 16,
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alaw_to_slinear16_le, slinear16_to_alaw_le, 2},
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{AUDIO_ENCODING_SLINEAR_BE, 16, 16,
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alaw_to_slinear16_be, slinear16_to_alaw_be, 2},
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{AUDIO_ENCODING_ULINEAR_LE, 16, 16,
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alaw_to_ulinear16_le, ulinear16_to_alaw_le, 2},
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{AUDIO_ENCODING_ULINEAR_BE, 16, 16,
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alaw_to_ulinear16_be, ulinear16_to_alaw_be, 2},
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{AUDIO_ENCODING_SLINEAR_LE, 8, 8,
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alaw_to_slinear8, slinear8_to_alaw, 1},
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{AUDIO_ENCODING_ULINEAR_LE, 8, 8,
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alaw_to_ulinear8, ulinear8_to_alaw, 1},
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{0, 0, 0, NULL, NULL, 0}};
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#endif
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void
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change_sign8(void *v, u_char *p, int cc)
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{
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while (--cc >= 0) {
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*p ^= 0x80;
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++p;
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}
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}
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void
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change_sign16_le(void *v, u_char *p, int cc)
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{
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while ((cc -= 2) >= 0) {
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p[1] ^= 0x80;
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p += 2;
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}
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}
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void
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change_sign16_be(void *v, u_char *p, int cc)
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{
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while ((cc -= 2) >= 0) {
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p[0] ^= 0x80;
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p += 2;
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}
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}
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void
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swap_bytes(void *v, u_char *p, int cc)
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{
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u_char t;
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while ((cc -= 2) >= 0) {
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t = p[0];
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p[0] = p[1];
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p[1] = t;
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p += 2;
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}
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}
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void
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swap_bytes_change_sign16_le(void *v, u_char *p, int cc)
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{
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u_char t;
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while ((cc -= 2) >= 0) {
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t = p[1];
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p[1] = p[0] ^ 0x80;
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p[0] = t;
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p += 2;
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}
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}
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void
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swap_bytes_change_sign16_be(void *v, u_char *p, int cc)
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{
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u_char t;
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while ((cc -= 2) >= 0) {
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t = p[0];
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p[0] = p[1] ^ 0x80;
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p[1] = t;
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p += 2;
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}
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}
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void
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change_sign16_swap_bytes_le(void *v, u_char *p, int cc)
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{
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swap_bytes_change_sign16_be(v, p, cc);
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}
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void
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change_sign16_swap_bytes_be(void *v, u_char *p, int cc)
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{
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swap_bytes_change_sign16_le(v, p, cc);
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}
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void
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linear8_to_linear16_le(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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p += cc;
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q += cc * 2;
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while (--cc >= 0) {
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q -= 2;
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q[1] = *--p;
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q[0] = 0;
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}
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}
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void
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linear8_to_linear16_be(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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p += cc;
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q += cc * 2;
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while (--cc >= 0) {
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q -= 2;
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q[0] = *--p;
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q[1] = 0;
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}
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}
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void
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linear16_to_linear8_le(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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while (--cc >= 0) {
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*q++ = p[1];
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p += 2;
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}
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}
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void
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linear16_to_linear8_be(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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while (--cc >= 0) {
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*q++ = p[0];
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p += 2;
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}
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}
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void
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ulinear8_to_slinear16_le(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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p += cc;
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q += cc * 2;
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while (--cc >= 0) {
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q -= 2;
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q[1] = *--p ^ 0x80;
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q[0] = 0;
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}
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}
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void
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ulinear8_to_slinear16_be(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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p += cc;
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q += cc * 2;
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while (--cc >= 0) {
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q -= 2;
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q[0] = *--p ^ 0x80;
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q[1] = 0;
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}
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}
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void
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slinear16_to_ulinear8_le(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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while (--cc >= 0) {
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*q++ = p[1] ^ 0x80;
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p += 2;
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}
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}
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void
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slinear16_to_ulinear8_be(void *v, u_char *p, int cc)
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{
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u_char *q = p;
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while (--cc >= 0) {
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*q++ = p[0] ^ 0x80;
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p += 2;
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}
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}
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/**
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* Set appropriate parameters in `param,' and return the index in
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* the hardware capability array `formats.'
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*
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* @param formats [IN] An array of formats which a hardware can support.
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* @param nformats [IN] The number of elements of the array.
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* @param mode [IN] Either AUMODE_PLAY or AUMODE_RECORD.
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* @param param [IN/OUT] Requested format. param->sw_code may be set.
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* @param rateconv [IN] TRUE if aurateconv may be used.
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* @return The index of selected audio_format entry. -1 if the device
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* can not support the specified param.
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*/
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int
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auconv_set_converter(const struct audio_format *formats, int nformats,
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int mode, struct audio_params *param, int rateconv)
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{
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struct audio_params work;
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const struct conv_table *table;
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int enc;
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int i, j;
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#ifdef AUCONV_DEBUG
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auconv_dump_formats(formats, nformats);
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#endif
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work = *param;
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work.sw_code = NULL;
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work.factor = 1;
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work.factor_denom = 1;
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work.hw_sample_rate = work.sample_rate;
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work.hw_encoding = work.encoding;
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work.hw_precision = work.precision;
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/* work.hw_subframe = work.precision; */
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work.hw_channels = work.channels;
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enc = auconv_normalize_encoding(work.encoding, work.precision);
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/* check support by native format */
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i = auconv_exact_match(formats, nformats, mode, &work);
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if (i >= 0) {
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*param = work;
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return i;
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}
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work = *param;
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#if NAURATECONV > 0
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/* native format with aurateconv */
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if (rateconv
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&& auconv_rateconv_supportable(enc, work.hw_precision,
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work.hw_precision)) {
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i = auconv_rateconv_check_channels(formats, nformats,
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mode, &work);
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if (i >= 0) {
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*param = work;
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return i;
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}
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}
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#endif
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|
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/* check for emulation */
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table = NULL;
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switch (enc) {
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case AUDIO_ENCODING_SLINEAR_LE:
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if (param->precision == 8)
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table = s8_table;
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else if (param->precision == 16)
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table = s16le_table;
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break;
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case AUDIO_ENCODING_SLINEAR_BE:
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if (param->precision == 8)
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table = s8_table;
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else if (param->precision == 16)
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table = s16be_table;
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break;
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case AUDIO_ENCODING_ULINEAR_LE:
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if (param->precision == 8)
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table = u8_table;
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else if (param->precision == 16)
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table = u16le_table;
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break;
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case AUDIO_ENCODING_ULINEAR_BE:
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if (param->precision == 8)
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table = u8_table;
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else if (param->precision == 16)
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table = u16be_table;
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break;
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#if NMULAW > 0
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case AUDIO_ENCODING_ULAW:
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table = mulaw_table;
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break;
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case AUDIO_ENCODING_ALAW:
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table = alaw_table;
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break;
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#endif
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}
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if (table == NULL)
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return -1;
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work = *param;
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for (j = 0; table[j].hw_precision != 0; j++) {
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work.hw_encoding = table[j].hw_encoding;
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work.hw_precision = table[j].hw_precision;
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/* work.hw_subframe = table[j].hw_subframe; */
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i = auconv_exact_match(formats, nformats, mode, &work);
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if (i >= 0) {
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*param = work;
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param->sw_code = mode == AUMODE_PLAY
|
|
? table[j].play_conv : table[j].rec_conv;
|
|
param->factor = table[j].factor;
|
|
param->factor_denom = 1;
|
|
return i;
|
|
}
|
|
}
|
|
/* not found */
|
|
|
|
#if NAURATECONV > 0
|
|
/* emulation with aurateconv */
|
|
if (!rateconv)
|
|
return -1;
|
|
work = *param;
|
|
for (j = 0; table[j].hw_precision != 0; j++) {
|
|
if (!auconv_rateconv_supportable(table[j].hw_encoding,
|
|
table[j].hw_precision,
|
|
table[j].hw_subframe))
|
|
continue;
|
|
work.hw_encoding = table[j].hw_encoding;
|
|
work.hw_precision = table[j].hw_precision;
|
|
/* work.hw_subframe = table[j].hw_subframe; */
|
|
i = auconv_rateconv_check_channels(formats, nformats,
|
|
mode, &work);
|
|
if (i >= 0) {
|
|
*param = work;
|
|
param->sw_code = mode == AUMODE_PLAY
|
|
? table[j].play_conv : table[j].rec_conv;
|
|
param->factor = table[j].factor;
|
|
param->factor_denom = 1;
|
|
return i;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
return -1;
|
|
}
|
|
|
|
#if NAURATECONV > 0
|
|
static int
|
|
auconv_rateconv_supportable(u_int encoding, u_int precision, u_int subframe)
|
|
{
|
|
if (encoding != AUDIO_ENCODING_SLINEAR_LE
|
|
&& encoding != AUDIO_ENCODING_SLINEAR_BE)
|
|
return FALSE;
|
|
if (precision != 16 && precision != 24)
|
|
return FALSE;
|
|
if (precision != subframe)
|
|
return FALSE;
|
|
return TRUE;
|
|
}
|
|
|
|
static int
|
|
auconv_rateconv_check_channels(const struct audio_format *formats, int nformats,
|
|
int mode, struct audio_params *param)
|
|
{
|
|
int ind, n;
|
|
|
|
/* check for the specified number of channels */
|
|
ind = auconv_rateconv_check_rates(formats ,nformats, mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
|
|
/* check for larger numbers */
|
|
for (n = param->channels + 1; n <= AUDIO_MAX_CHANNELS; n++) {
|
|
param->hw_channels = n;
|
|
ind = auconv_rateconv_check_rates(formats, nformats,
|
|
mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
}
|
|
|
|
/* check for stereo:monaural conversion */
|
|
if (param->channels == 2) {
|
|
param->hw_channels = 1;
|
|
ind = auconv_rateconv_check_rates(formats, nformats,
|
|
mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
}
|
|
param->hw_channels = param->channels;
|
|
return -1;
|
|
}
|
|
|
|
static int
|
|
auconv_rateconv_check_rates(const struct audio_format *formats, int nformats,
|
|
int mode, struct audio_params *param)
|
|
{
|
|
int ind, i, j, enc, f_enc;
|
|
u_long rate, minrate, maxrate;
|
|
|
|
/* exact match */
|
|
ind = auconv_exact_match(formats, nformats, mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
|
|
/* determine min/max of specified encoding/precision/channels */
|
|
minrate = ULONG_MAX;
|
|
maxrate = 0;
|
|
enc = auconv_normalize_encoding(param->hw_encoding,
|
|
param->hw_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].precision != param->hw_precision)
|
|
continue;
|
|
if (formats[i].subframe_size != param->hw_precision /*hw_subframe*/)
|
|
continue;
|
|
if (formats[i].channels != param->hw_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 */
|
|
for (i = 2; (rate = param->sample_rate * i) <= maxrate; i++) {
|
|
param->hw_sample_rate = rate;
|
|
ind = auconv_exact_match(formats, nformats, mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
}
|
|
|
|
param->hw_sample_rate = param->sample_rate >= minrate
|
|
? maxrate : minrate;
|
|
ind = auconv_exact_match(formats, nformats, mode, param);
|
|
if (ind >= 0)
|
|
return ind;
|
|
param->hw_sample_rate = param->sample_rate;
|
|
return -1;
|
|
}
|
|
#endif /* NAURATECONV */
|
|
|
|
#ifdef AUCONV_DEBUG
|
|
static void
|
|
auconv_dump_formats(const struct audio_format *formats, int nformats)
|
|
{
|
|
static const char *encoding_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
|
|
};
|
|
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_names[f->encoding]);
|
|
printf(" %u/%ubit", f->precision, f->subframe_size);
|
|
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(" %luHz-%luHz", f->frequency[0],
|
|
f->frequency[1]);
|
|
} else {
|
|
printf(" %luHz", f->frequency[0]);
|
|
for (j = 1; j < f->frequency_type; j++)
|
|
printf(",%luHz", f->frequency[j]);
|
|
}
|
|
printf("\n");
|
|
}
|
|
}
|
|
#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 struct audio_params *param)
|
|
{
|
|
int i, enc, f_enc;
|
|
|
|
enc = auconv_normalize_encoding(param->hw_encoding,
|
|
param->hw_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;
|
|
/**
|
|
* XXX we need encoding-dependent check.
|
|
* XXX Is to check precision/channels meaningful for
|
|
* MPEG encodings?
|
|
*/
|
|
if (formats[i].precision != param->hw_precision)
|
|
continue;
|
|
if (formats[i].subframe_size != param->hw_precision /*hw_subframe*/)
|
|
continue;
|
|
if (formats[i].channels != param->hw_channels)
|
|
continue;
|
|
if (!auconv_is_supported_rate(&formats[i],
|
|
param->hw_sample_rate))
|
|
continue;
|
|
return i;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
/**
|
|
* a sub-routine for auconv_set_converter()
|
|
*/
|
|
static u_int
|
|
auconv_normalize_encoding(u_int encoding, u_int precision)
|
|
{
|
|
int enc;
|
|
|
|
enc = encoding;
|
|
#if BYTE_ORDER == LITTLE_ENDIAN
|
|
if (enc == AUDIO_ENCODING_SLINEAR)
|
|
enc = AUDIO_ENCODING_SLINEAR_LE;
|
|
else if (enc == AUDIO_ENCODING_ULINEAR)
|
|
enc = 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)
|
|
enc = AUDIO_ENCODING_SLINEAR_LE;
|
|
if (precision == 8 && enc == AUDIO_ENCODING_ULINEAR_BE)
|
|
enc = 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_long 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));
|
|
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:
|
|
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
|
|
};
|
|
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 >= 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;
|
|
}
|