476 lines
11 KiB
C
476 lines
11 KiB
C
/* $NetBSD: wav.c,v 1.13 2015/08/05 06:54:39 mrg Exp $ */
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/*
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* Copyright (c) 2002, 2009 Matthew R. Green
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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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*
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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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* WAV support for the audio tools; thanks go to the sox utility for
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* clearing up issues with WAV files.
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*/
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#include <sys/cdefs.h>
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#ifndef lint
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__RCSID("$NetBSD: wav.c,v 1.13 2015/08/05 06:54:39 mrg Exp $");
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#endif
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#include <sys/types.h>
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#include <sys/audioio.h>
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#include <sys/ioctl.h>
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#include <sys/time.h>
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#include <ctype.h>
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#include <err.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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#include <stdint.h>
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#include <unistd.h>
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#include "libaudio.h"
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#include "auconv.h"
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static const struct {
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int wenc;
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const char *wname;
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} wavencs[] = {
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{ WAVE_FORMAT_UNKNOWN, "Microsoft Official Unknown" },
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{ WAVE_FORMAT_PCM, "Microsoft PCM" },
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{ WAVE_FORMAT_ADPCM, "Microsoft ADPCM" },
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{ WAVE_FORMAT_ALAW, "Microsoft A-law" },
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{ WAVE_FORMAT_MULAW, "Microsoft mu-law" },
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{ WAVE_FORMAT_OKI_ADPCM,"OKI ADPCM" },
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{ WAVE_FORMAT_DIGISTD, "Digistd format" },
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{ WAVE_FORMAT_DIGIFIX, "Digifix format" },
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{ -1, "?Unknown?" },
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};
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const char *
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wav_enc_from_val(int encoding)
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{
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int i;
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for (i = 0; wavencs[i].wenc != -1; i++)
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if (wavencs[i].wenc == encoding)
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break;
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return (wavencs[i].wname);
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}
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/*
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* sample header is:
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*
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* RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
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*
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*/
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/*
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* WAV format helpers
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*/
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/*
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* find a .wav header, etc. returns header length on success
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*/
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ssize_t
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audio_wav_parse_hdr(void *hdr, size_t sz, u_int *enc, u_int *prec,
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u_int *sample, u_int *channels, off_t *datasize)
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{
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char *where = hdr, *owhere;
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wav_audioheaderpart part;
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wav_audioheaderfmt fmt;
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wav_audiohdrextensible ext;
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char *end = (((char *)hdr) + sz);
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u_int newenc, newprec;
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u_int16_t fmttag;
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static const char
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strfmt[4] = "fmt ",
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strRIFF[4] = "RIFF",
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strWAVE[4] = "WAVE",
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strdata[4] = "data";
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if (sz < 32)
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return (AUDIO_ENOENT);
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if (strncmp(where, strRIFF, sizeof strRIFF))
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return (AUDIO_ENOENT);
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where += 8;
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if (strncmp(where, strWAVE, sizeof strWAVE))
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return (AUDIO_ENOENT);
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where += 4;
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do {
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memcpy(&part, where, sizeof part);
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owhere = where;
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where += getle32(part.len) + 8;
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} while (where < end && strncmp(part.name, strfmt, sizeof strfmt));
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/* too short ? */
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if (where + sizeof fmt > end)
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return (AUDIO_ESHORTHDR);
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memcpy(&fmt, (owhere + 8), sizeof fmt);
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fmttag = getle16(fmt.tag);
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if (verbose)
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printf("WAVE format tag: %x\n", fmttag);
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if (fmttag == WAVE_FORMAT_EXTENSIBLE) {
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if ((uintptr_t)(where - owhere) < sizeof(fmt) + sizeof(ext))
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return (AUDIO_ESHORTHDR);
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memcpy(&ext, owhere + sizeof fmt, sizeof ext);
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if (getle16(ext.len) < sizeof(ext) - sizeof(ext.len))
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return (AUDIO_ESHORTHDR);
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fmttag = ext.sub_tag;
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if (verbose)
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printf("WAVE extensible sub tag: %x\n", fmttag);
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}
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switch (fmttag) {
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case WAVE_FORMAT_UNKNOWN:
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case IBM_FORMAT_MULAW:
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case IBM_FORMAT_ALAW:
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case IBM_FORMAT_ADPCM:
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default:
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return (AUDIO_EWAVUNSUPP);
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case WAVE_FORMAT_PCM:
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case WAVE_FORMAT_ADPCM:
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case WAVE_FORMAT_OKI_ADPCM:
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case WAVE_FORMAT_IMA_ADPCM:
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case WAVE_FORMAT_DIGIFIX:
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case WAVE_FORMAT_DIGISTD:
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switch (getle16(fmt.bits_per_sample)) {
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case 8:
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newprec = 8;
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break;
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case 16:
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newprec = 16;
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break;
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case 24:
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newprec = 24;
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break;
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case 32:
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newprec = 32;
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break;
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default:
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return (AUDIO_EWAVBADPCM);
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}
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if (newprec == 8)
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newenc = AUDIO_ENCODING_ULINEAR_LE;
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else
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newenc = AUDIO_ENCODING_SLINEAR_LE;
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break;
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case WAVE_FORMAT_ALAW:
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newenc = AUDIO_ENCODING_ALAW;
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newprec = 8;
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break;
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case WAVE_FORMAT_MULAW:
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newenc = AUDIO_ENCODING_ULAW;
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newprec = 8;
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break;
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}
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do {
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memcpy(&part, where, sizeof part);
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owhere = where;
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where += (getle32(part.len) + 8);
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} while (where < end && strncmp(part.name, strdata, sizeof strdata));
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if ((where - getle32(part.len)) <= end) {
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if (channels)
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*channels = (u_int)getle16(fmt.channels);
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if (sample)
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*sample = getle32(fmt.sample_rate);
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if (enc)
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*enc = newenc;
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if (prec)
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*prec = newprec;
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if (datasize)
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*datasize = (off_t)getle32(part.len);
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return (owhere - (char *)hdr + 8);
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}
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return (AUDIO_EWAVNODATA);
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}
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/*
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* prepare a WAV header for writing; we fill in hdrp, lenp and leftp,
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* and expect our caller (wav_write_header()) to use them.
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*/
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int
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wav_prepare_header(struct track_info *ti, void **hdrp, size_t *lenp, int *leftp)
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{
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/*
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* WAV header we write looks like this:
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*
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* bytes purpose
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* 0-3 "RIFF"
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* 4-7 file length (minus 8)
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* 8-15 "WAVEfmt "
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* 16-19 format size
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* 20-21 format tag
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* 22-23 number of channels
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* 24-27 sample rate
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* 28-31 average bytes per second
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* 32-33 block alignment
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* 34-35 bits per sample
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*
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* then for ULAW and ALAW outputs, we have an extended chunk size
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* and a WAV "fact" to add:
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*
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* 36-37 length of extension (== 0)
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* 38-41 "fact"
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* 42-45 fact size
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* 46-49 number of samples written
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* 50-53 "data"
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* 54-57 data length
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* 58- raw audio data
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*
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* for PCM outputs we have just the data remaining:
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*
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* 36-39 "data"
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* 40-43 data length
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* 44- raw audio data
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*
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* RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
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*/
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static char wavheaderbuf[64];
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char *p = wavheaderbuf;
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const char *riff = "RIFF",
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*wavefmt = "WAVEfmt ",
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*fact = "fact",
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*data = "data";
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u_int32_t filelen, fmtsz, sps, abps, factsz = 4, nsample, datalen;
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u_int16_t fmttag, nchan, align, extln = 0;
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if (ti->header_info)
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warnx("header information not supported for WAV");
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*leftp = 0;
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switch (ti->precision) {
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case 8:
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break;
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case 16:
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break;
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case 32:
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break;
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default:
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{
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static int warned = 0;
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if (warned == 0) {
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warnx("can not support precision of %d", ti->precision);
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warned = 1;
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}
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}
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return (-1);
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}
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switch (ti->encoding) {
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case AUDIO_ENCODING_ULAW:
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fmttag = WAVE_FORMAT_MULAW;
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fmtsz = 18;
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align = ti->channels;
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break;
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case AUDIO_ENCODING_ALAW:
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fmttag = WAVE_FORMAT_ALAW;
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fmtsz = 18;
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align = ti->channels;
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break;
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/*
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* we could try to support RIFX but it seems to be more portable
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* to output little-endian data for WAV files.
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*/
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case AUDIO_ENCODING_ULINEAR_BE:
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case AUDIO_ENCODING_SLINEAR_BE:
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case AUDIO_ENCODING_ULINEAR_LE:
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case AUDIO_ENCODING_SLINEAR_LE:
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case AUDIO_ENCODING_PCM16:
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#if BYTE_ORDER == LITTLE_ENDIAN
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case AUDIO_ENCODING_ULINEAR:
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case AUDIO_ENCODING_SLINEAR:
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#endif
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fmttag = WAVE_FORMAT_PCM;
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fmtsz = 16;
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align = ti->channels * (ti->precision / 8);
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break;
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default:
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#if 0 // move into record.c, and maybe merge.c
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{
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static int warned = 0;
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if (warned == 0) {
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const char *s = wav_enc_from_val(ti->encoding);
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if (s == NULL)
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warnx("can not support encoding of %s", s);
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else
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warnx("can not support encoding of %d", ti->encoding);
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warned = 1;
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}
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}
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#endif
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ti->format = AUDIO_FORMAT_NONE;
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return (-1);
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}
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nchan = ti->channels;
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sps = ti->sample_rate;
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/* data length */
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if (ti->outfd == STDOUT_FILENO)
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datalen = 0;
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else if (ti->total_size != -1)
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datalen = ti->total_size;
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else
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datalen = 0;
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/* file length */
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filelen = 4 + (8 + fmtsz) + (8 + datalen);
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if (fmttag != WAVE_FORMAT_PCM)
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filelen += 8 + factsz;
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abps = (double)align*ti->sample_rate / (double)1 + 0.5;
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nsample = (datalen / ti->precision) / ti->sample_rate;
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/*
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* now we've calculated the info, write it out!
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*/
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#define put32(x) do { \
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u_int32_t _f; \
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putle32(_f, (x)); \
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memcpy(p, &_f, 4); \
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} while (0)
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#define put16(x) do { \
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u_int16_t _f; \
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putle16(_f, (x)); \
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memcpy(p, &_f, 2); \
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} while (0)
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memcpy(p, riff, 4);
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p += 4; /* 4 */
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put32(filelen);
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p += 4; /* 8 */
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memcpy(p, wavefmt, 8);
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p += 8; /* 16 */
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put32(fmtsz);
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p += 4; /* 20 */
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put16(fmttag);
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p += 2; /* 22 */
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put16(nchan);
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p += 2; /* 24 */
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put32(sps);
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p += 4; /* 28 */
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put32(abps);
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p += 4; /* 32 */
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put16(align);
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p += 2; /* 34 */
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put16(ti->precision);
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p += 2; /* 36 */
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/* NON PCM formats have an extended chunk; write it */
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if (fmttag != WAVE_FORMAT_PCM) {
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put16(extln);
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p += 2; /* 38 */
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memcpy(p, fact, 4);
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p += 4; /* 42 */
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put32(factsz);
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p += 4; /* 46 */
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put32(nsample);
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p += 4; /* 50 */
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}
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memcpy(p, data, 4);
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p += 4; /* 40/54 */
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put32(datalen);
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p += 4; /* 44/58 */
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#undef put32
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#undef put16
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*hdrp = wavheaderbuf;
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*lenp = (p - wavheaderbuf);
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return 0;
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}
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write_conv_func
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wav_write_get_conv_func(struct track_info *ti)
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{
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write_conv_func conv_func = NULL;
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switch (ti->encoding) {
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/*
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* we could try to support RIFX but it seems to be more portable
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* to output little-endian data for WAV files.
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*/
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case AUDIO_ENCODING_ULINEAR_BE:
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#if BYTE_ORDER == BIG_ENDIAN
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case AUDIO_ENCODING_ULINEAR:
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#endif
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if (ti->precision == 16)
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conv_func = change_sign16_swap_bytes_be;
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else if (ti->precision == 32)
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conv_func = change_sign32_swap_bytes_be;
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break;
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case AUDIO_ENCODING_SLINEAR_BE:
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#if BYTE_ORDER == BIG_ENDIAN
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case AUDIO_ENCODING_SLINEAR:
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#endif
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if (ti->precision == 8)
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conv_func = change_sign8;
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else if (ti->precision == 16)
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conv_func = swap_bytes;
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else if (ti->precision == 32)
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conv_func = swap_bytes32;
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break;
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case AUDIO_ENCODING_ULINEAR_LE:
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#if BYTE_ORDER == LITTLE_ENDIAN
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case AUDIO_ENCODING_ULINEAR:
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#endif
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if (ti->precision == 16)
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conv_func = change_sign16_le;
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else if (ti->precision == 32)
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conv_func = change_sign32_le;
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break;
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case AUDIO_ENCODING_SLINEAR_LE:
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case AUDIO_ENCODING_PCM16:
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#if BYTE_ORDER == LITTLE_ENDIAN
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case AUDIO_ENCODING_SLINEAR:
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#endif
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if (ti->precision == 8)
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conv_func = change_sign8;
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break;
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default:
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ti->format = AUDIO_FORMAT_NONE;
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}
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return conv_func;
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}
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