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/*
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* Get Audio routines source file
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*
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* Copyright (c) 1999 Albert L Faber
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* $Id: get_audio.c,v 1.61 2001/03/19 21:26:05 markt Exp $ */
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include <assert.h>
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#ifdef HAVE_LIMITS_H
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# include <limits.h>
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#endif
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#include <stdio.h>
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#ifdef STDC_HEADERS
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# include <stdlib.h>
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# include <string.h>
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#else
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# ifndef HAVE_STRCHR
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# define strchr index
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# define strrchr rindex
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# endif
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char *strchr(), *strrchr();
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# ifndef HAVE_MEMCPY
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# define memcpy(d, s, n) bcopy ((s), (d), (n))
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# define memmove(d, s, n) bcopy ((s), (d), (n))
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# endif
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#endif
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#include <math.h>
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#include <sys/types.h>
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#include <sys/stat.h>
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#include "lame.h"
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#include "main.h"
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#include "get_audio.h"
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#include "portableio.h"
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#include "timestatus.h"
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#include "lametime.h"
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#ifdef WITH_DMALLOC
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#include <dmalloc.h>
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#endif
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/* global data for get_audio.c. */
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int count_samples_carefully;
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int pcmbitwidth;
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mp3data_struct mp3input_data; /* used by Ogg and MP3 */
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unsigned int num_samples_read;
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FILE *musicin;
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#ifdef AMIGA_MPEGA
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int lame_decode_initfile(const char *fullname,
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mp3data_struct * const mp3data);
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#else
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int lame_decode_initfile(FILE * const fd, mp3data_struct * const mp3data);
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#endif
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/* read mp3 file until mpglib returns one frame of PCM data */
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int lame_decode_fromfile(FILE * fd, short int pcm_l[], short int pcm_r[],
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mp3data_struct * mp3data);
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/* and for Vorbis: */
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int lame_decode_ogg_initfile( lame_global_flags* gfp,
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FILE* fd,
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mp3data_struct* mp3data );
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int lame_decode_ogg_fromfile( lame_global_flags* gfc,
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FILE* fd,
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short int pcm_l[],
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short int pcm_r[],
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mp3data_struct* mp3data );
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static int read_samples_pcm(FILE * musicin, short sample_buffer[2304],
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int frame_size, int samples_to_read);
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static int read_samples_mp3(lame_global_flags * gfp, FILE * musicin,
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short int mpg123pcm[2][1152], int num_chan);
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static int read_samples_ogg(lame_global_flags * gfp, FILE * musicin,
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short int mpg123pcm[2][1152], int num_chan);
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void CloseSndFile(sound_file_format input, FILE * musicin);
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FILE *OpenSndFile(lame_global_flags * gfp, char *);
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/* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */
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static int
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fskip(FILE * fp, long offset, int whence)
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{
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#ifndef PIPE_BUF
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char buffer[4096];
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#else
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char buffer[PIPE_BUF];
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#endif
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int read;
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if (0 == fseek(fp, offset, whence))
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return 0;
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if (whence != SEEK_CUR || offset < 0) {
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fprintf(stderr,
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"fskip problem: Mostly the return status of functions is not evaluated so it is more secure to pollute <stderr>.\n");
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return -1;
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}
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while (offset > 0) {
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read = offset > sizeof(buffer) ? sizeof(buffer) : offset;
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if ((read = fread(buffer, 1, read, fp)) <= 0)
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return -1;
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offset -= read;
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}
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return 0;
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}
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FILE *
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init_outfile(char *outPath, int decode)
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{
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FILE *outf;
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/* open the output file */
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if (0 == strcmp(outPath, "-"))
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lame_set_stream_binary_mode(outf = stdout);
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else
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if ((outf = fopen(outPath, "wb+")) == NULL)
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return NULL;
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return outf;
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}
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void
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init_infile(lame_global_flags * gfp, char *inPath)
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{
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/* open the input file */
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count_samples_carefully = 0;
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pcmbitwidth = 16;
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musicin = OpenSndFile(gfp, inPath);
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}
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void
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close_infile(void)
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{
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CloseSndFile(input_format, musicin);
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}
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void
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SwapBytesInWords(short *ptr, int short_words)
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{ /* Some speedy code */
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unsigned long val;
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unsigned long *p = (unsigned long *) ptr;
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#ifndef lint
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# if defined(CHAR_BIT)
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# if CHAR_BIT != 8
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# error CHAR_BIT != 8
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# endif
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# else
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# error can not determine number of bits in a char
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# endif
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#endif /* lint */
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assert(sizeof(short) == 2);
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#if defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 4
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for (; short_words >= 2; short_words -= 2, p++) {
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val = *p;
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*p = ((val << 8) & 0xFF00FF00) | ((val >> 8) & 0x00FF00FF);
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}
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ptr = (short *) p;
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for (; short_words >= 1; short_words -= 1, ptr++) {
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val = *ptr;
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*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
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}
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#elif defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 8
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for (; short_words >= 4; short_words -= 4, p++) {
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val = *p;
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*p =
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((val << 8) & 0xFF00FF00FF00FF00) | ((val >> 8) &
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0x00FF00FF00FF00FF);
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}
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ptr = (short *) p;
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for (; short_words >= 1; short_words -= 1, ptr++) {
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val = *ptr;
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*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
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}
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#else
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# ifdef SIZEOF_UNSIGNED_LONG
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//# warning Using unoptimized SwapBytesInWords().
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# endif
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for (; short_words >= 1; short_words -= 1, ptr++) {
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val = *ptr;
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*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
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}
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#endif
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assert(short_words == 0);
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}
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/************************************************************************
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*
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* get_audio()
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*
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* PURPOSE: reads a frame of audio data from a file to the buffer,
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* aligns the data for future processing, and separates the
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* left and right channels
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*
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************************************************************************/
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int
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get_audio(lame_global_flags * const gfp, short buffer[2][1152])
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{
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int num_channels = lame_get_num_channels( gfp );
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short insamp[2 * 1152];
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int samples_read;
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int framesize;
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int samples_to_read;
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unsigned int remaining, tmp_num_samples;
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int j;
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short *p;
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/*
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* NOTE: LAME can now handle arbritray size input data packets,
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* so there is no reason to read the input data in chuncks of
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* size "gfp->framesize". EXCEPT: the LAME graphical frame analyzer
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* will get out of sync if we read more than framesize worth of data.
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*/
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samples_to_read = framesize = gfp->framesize;
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assert(framesize <= 1152);
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/* get num_samples */
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tmp_num_samples = lame_get_num_samples( gfp );
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/* if this flag has been set, then we are carefull to read
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* exactly num_samples and no more. This is useful for .wav and .aiff
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* files which have id3 or other tags at the end. Note that if you
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* are using LIBSNDFILE, this is not necessary
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*/
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if (count_samples_carefully) {
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remaining = tmp_num_samples - Min(tmp_num_samples, num_samples_read);
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if (remaining < framesize)
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samples_to_read = remaining;
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}
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switch (input_format) {
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case sf_mp1:
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case sf_mp2:
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case sf_mp3:
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samples_read = read_samples_mp3(gfp, musicin, buffer, num_channels);
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break;
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case sf_ogg:
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samples_read = read_samples_ogg(gfp, musicin, buffer, num_channels);
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break;
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default:
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samples_read =
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read_samples_pcm(musicin, insamp, num_channels * framesize,
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num_channels * samples_to_read);
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samples_read /= num_channels;
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p = insamp;
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switch (num_channels) {
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case 1:
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for (j = 0; j < framesize; j++) {
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buffer[0][j] = *p++;
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buffer[1][j] = 0;
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}
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break;
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case 2:
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for (j = 0; j < framesize; j++) {
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buffer[0][j] = *p++;
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buffer[1][j] = *p++;
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}
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break;
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default:
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assert(0);
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break;
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}
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}
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/* if num_samples = MAX_U_32_NUM, then it is considered infinitely long.
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Don't count the samples */
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if ( tmp_num_samples != MAX_U_32_NUM )
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num_samples_read += samples_read;
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return samples_read;
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}
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313 |
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314 |
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315 |
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316 |
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317 |
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318 |
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int
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320 |
read_samples_ogg(lame_global_flags * const gfp,
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FILE * const musicin,
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short int oggpcm[2][1152], const int stereo)
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{
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int out = 0;
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325 |
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326 |
#ifdef HAVE_VORBIS
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static const char type_name[] = "Ogg Vorbis file";
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328 |
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out =
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lame_decode_ogg_fromfile( gfp,
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musicin,
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oggpcm[0],
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oggpcm[1],
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&mp3input_data );
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/*
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* out < 0: error, probably EOF
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* out = 0: not possible with lame_decode_fromfile() ???
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* out > 0: number of output samples
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*/
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340 |
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341 |
if (out < 0) {
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memset(oggpcm, 0, sizeof(**oggpcm) * 2 * 1152);
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return 0;
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344 |
}
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345 |
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346 |
if (lame_get_num_channels( gfp ) != mp3input_data.stereo)
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347 |
fprintf(stderr,
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"Error: number of channels has changed in %s - not supported\n",
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type_name);
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350 |
if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
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351 |
fprintf(stderr,
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"Error: sample frequency has changed in %s - not supported\n",
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type_name);
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354 |
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355 |
#else
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356 |
out = -1; /* wanna read ogg without vorbis support? */
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357 |
#endif
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358 |
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359 |
return out;
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360 |
}
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361 |
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362 |
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363 |
int
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364 |
read_samples_mp3(lame_global_flags * const gfp,
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|
365 |
FILE * const musicin, short int mpg123pcm[2][1152], int stereo)
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366 |
{
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367 |
int out;
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368 |
#if defined(AMIGA_MPEGA) || defined(HAVE_MPGLIB)
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369 |
static const char type_name[] = "MP3 file";
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370 |
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371 |
out =
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372 |
lame_decode_fromfile(musicin, mpg123pcm[0], mpg123pcm[1],
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|
373 |
&mp3input_data);
|
|
|
374 |
/*
|
|
|
375 |
* out < 0: error, probably EOF
|
|
|
376 |
* out = 0: not possible with lame_decode_fromfile() ???
|
|
|
377 |
* out > 0: number of output samples
|
|
|
378 |
*/
|
|
|
379 |
|
|
|
380 |
if (out < 0) {
|
|
|
381 |
memset(mpg123pcm, 0, sizeof(**mpg123pcm) * 2 * 1152);
|
|
|
382 |
return 0;
|
|
|
383 |
}
|
|
|
384 |
|
|
|
385 |
if ( lame_get_num_channels( gfp ) != mp3input_data.stereo )
|
|
|
386 |
fprintf(stderr,
|
|
|
387 |
"Error: number of channels has changed in %s - not supported\n",
|
|
|
388 |
type_name);
|
|
|
389 |
if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
|
|
|
390 |
fprintf(stderr,
|
|
|
391 |
"Error: sample frequency has changed in %s - not supported\n",
|
|
|
392 |
type_name);
|
|
|
393 |
|
|
|
394 |
#else
|
|
|
395 |
out = -1;
|
|
|
396 |
#endif
|
|
|
397 |
return out;
|
|
|
398 |
}
|
|
|
399 |
|
|
|
400 |
|
|
|
401 |
static int
|
|
|
402 |
WriteWaveHeader(FILE * const fp, const int pcmbytes,
|
|
|
403 |
const int freq, const int channels, const int bits)
|
|
|
404 |
{
|
|
|
405 |
int bytes = (bits + 7) / 8;
|
|
|
406 |
|
|
|
407 |
/* quick and dirty, but documented */
|
|
|
408 |
fwrite("RIFF", 1, 4, fp); // label
|
|
|
409 |
Write32BitsLowHigh(fp, pcmbytes + 44 - 8); // length in bytes without header
|
|
|
410 |
fwrite("WAVEfmt ", 2, 4, fp); // 2 labels
|
|
|
411 |
Write32BitsLowHigh(fp, 2 + 2 + 4 + 4 + 2 + 2); // length of PCM format declaration area
|
|
|
412 |
Write16BitsLowHigh(fp, 1); // is PCM?
|
|
|
413 |
Write16BitsLowHigh(fp, channels); // number of channels
|
|
|
414 |
Write32BitsLowHigh(fp, freq); // sample frequency in [Hz]
|
|
|
415 |
Write32BitsLowHigh(fp, freq * channels * bytes); // bytes per second
|
|
|
416 |
Write16BitsLowHigh(fp, channels * bytes); // bytes per sample time
|
|
|
417 |
Write16BitsLowHigh(fp, bits); // bits per sample
|
|
|
418 |
fwrite("data", 1, 4, fp); // label
|
|
|
419 |
Write32BitsLowHigh(fp, pcmbytes); // length in bytes of raw PCM data
|
|
|
420 |
|
|
|
421 |
return ferror(fp) ? -1 : 0;
|
|
|
422 |
}
|
|
|
423 |
|
|
|
424 |
/* the simple lame decoder */
|
|
|
425 |
/* After calling lame_init(), lame_init_params() and
|
|
|
426 |
* init_infile(), call this routine to read the input MP3 file
|
|
|
427 |
* and output .wav data to the specified file pointer*/
|
|
|
428 |
/* lame_decoder will ignore the first 528 samples, since these samples
|
|
|
429 |
* represent the mpglib delay (and are all 0). skip = number of additional
|
|
|
430 |
* samples to skip, to (for example) compensate for the encoder delay */
|
|
|
431 |
|
|
|
432 |
int
|
|
|
433 |
lame_decoder(lame_global_flags * gfp, FILE * outf, int skip, char *inPath,
|
|
|
434 |
char *outPath)
|
|
|
435 |
{
|
|
|
436 |
short int Buffer[2][1152];
|
|
|
437 |
int iread;
|
|
|
438 |
double wavsize;
|
|
|
439 |
int i;
|
|
|
440 |
void (*WriteFunction) (FILE * fp, char *p, int n);
|
|
|
441 |
int tmp_num_channels = lame_get_num_channels( gfp );
|
|
|
442 |
|
|
|
443 |
|
|
|
444 |
|
|
|
445 |
fprintf(stderr, "\rinput: %s%s(%g kHz, %i channel%s, ",
|
|
|
446 |
strcmp(inPath, "-") ? inPath : "<stdin>",
|
|
|
447 |
strlen(inPath) > 26 ? "\n\t" : " ",
|
|
|
448 |
lame_get_in_samplerate( gfp ) / 1.e3,
|
|
|
449 |
tmp_num_channels, tmp_num_channels != 1 ? "s" : "");
|
|
|
450 |
|
|
|
451 |
switch (input_format) {
|
|
|
452 |
case sf_mp3:
|
|
|
453 |
skip += 528 + 1; /* mp3 decoder has a 528 sample delay, plus user supplied "skip" */
|
|
|
454 |
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
|
|
|
455 |
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "III");
|
|
|
456 |
break;
|
|
|
457 |
case sf_mp2:
|
|
|
458 |
skip += 240 + 1;
|
|
|
459 |
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
|
|
|
460 |
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "II");
|
|
|
461 |
break;
|
|
|
462 |
case sf_mp1:
|
|
|
463 |
skip += 240 + 1;
|
|
|
464 |
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
|
|
|
465 |
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "I");
|
|
|
466 |
break;
|
|
|
467 |
case sf_ogg:
|
|
|
468 |
fprintf(stderr, "Ogg Vorbis");
|
|
|
469 |
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
|
|
|
470 |
break;
|
|
|
471 |
case sf_raw:
|
|
|
472 |
fprintf(stderr, "raw PCM data");
|
|
|
473 |
mp3input_data.nsamp = lame_get_num_samples( gfp );
|
|
|
474 |
mp3input_data.framesize = 1152;
|
|
|
475 |
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
|
|
|
476 |
break;
|
|
|
477 |
case sf_wave:
|
|
|
478 |
fprintf(stderr, "Microsoft WAVE");
|
|
|
479 |
mp3input_data.nsamp = lame_get_num_samples( gfp );
|
|
|
480 |
mp3input_data.framesize = 1152;
|
|
|
481 |
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
|
|
|
482 |
break;
|
|
|
483 |
case sf_aiff:
|
|
|
484 |
fprintf(stderr, "SGI/Apple AIFF");
|
|
|
485 |
mp3input_data.nsamp = lame_get_num_samples( gfp );
|
|
|
486 |
mp3input_data.framesize = 1152;
|
|
|
487 |
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
|
|
|
488 |
break;
|
|
|
489 |
default:
|
|
|
490 |
fprintf(stderr, "unknown");
|
|
|
491 |
mp3input_data.nsamp = lame_get_num_samples( gfp );
|
|
|
492 |
mp3input_data.framesize = 1152;
|
|
|
493 |
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
|
|
|
494 |
assert(0);
|
|
|
495 |
break;
|
|
|
496 |
}
|
|
|
497 |
|
|
|
498 |
fprintf(stderr, ")\noutput: %s%s(16 bit, Microsoft WAVE)\n",
|
|
|
499 |
strcmp(outPath, "-") ? outPath : "<stdout>",
|
|
|
500 |
strlen(outPath) > 45 ? "\n\t" : " ");
|
|
|
501 |
|
|
|
502 |
if (skip > 0)
|
|
|
503 |
fprintf(stderr, "skipping initial %i samples (encoder+decoder delay)\n",
|
|
|
504 |
skip);
|
|
|
505 |
|
|
|
506 |
if ( 0 == lame_get_disable_waveheader( gfp ) )
|
|
|
507 |
WriteWaveHeader(outf, 0x7FFFFFFF, lame_get_in_samplerate( gfp ),
|
|
|
508 |
tmp_num_channels,
|
|
|
509 |
16);
|
|
|
510 |
/* unknown size, so write maximum 32 bit signed value */
|
|
|
511 |
|
|
|
512 |
wavsize = -skip;
|
|
|
513 |
WriteFunction = swapbytes ? WriteBytesSwapped : WriteBytes;
|
|
|
514 |
mp3input_data.totalframes = mp3input_data.nsamp / mp3input_data.framesize;
|
|
|
515 |
|
|
|
516 |
assert(tmp_num_channels >= 1 && tmp_num_channels <= 2);
|
|
|
517 |
|
|
|
518 |
do {
|
|
|
519 |
iread = get_audio(gfp, Buffer); /* read in 'iread' samples */
|
|
|
520 |
mp3input_data.framenum += iread / mp3input_data.framesize;
|
|
|
521 |
wavsize += iread;
|
|
|
522 |
|
|
|
523 |
if (!silent)
|
|
|
524 |
decoder_progress(gfp, &mp3input_data);
|
|
|
525 |
|
|
|
526 |
skip -= (i = skip < iread ? skip : iread); /* 'i' samples are to skip in this frame */
|
|
|
527 |
|
|
|
528 |
for (; i < iread; i++) {
|
|
|
529 |
if ( lame_get_disable_waveheader( gfp ) ) {
|
|
|
530 |
WriteFunction(outf, (char *) Buffer[0] + i, sizeof(short));
|
|
|
531 |
if (tmp_num_channels == 2)
|
|
|
532 |
WriteFunction(outf, (char *) Buffer[1] + i, sizeof(short));
|
|
|
533 |
}
|
|
|
534 |
else {
|
|
|
535 |
Write16BitsLowHigh(outf, Buffer[0][i]);
|
|
|
536 |
if (tmp_num_channels == 2)
|
|
|
537 |
Write16BitsLowHigh(outf, Buffer[1][i]);
|
|
|
538 |
}
|
|
|
539 |
}
|
|
|
540 |
} while (iread);
|
|
|
541 |
|
|
|
542 |
i = (16 / 8) * tmp_num_channels;
|
|
|
543 |
assert(i > 0);
|
|
|
544 |
if (wavsize <= 0) {
|
|
|
545 |
fprintf(stderr, "WAVE file contains 0 PCM samples\n");
|
|
|
546 |
wavsize = 0;
|
|
|
547 |
}
|
|
|
548 |
else if (wavsize > 0xFFFFFFD0 / i) {
|
|
|
549 |
fprintf(stderr,
|
|
|
550 |
"Very huge WAVE file, can't set filesize accordingly\n");
|
|
|
551 |
wavsize = 0xFFFFFFD0;
|
|
|
552 |
}
|
|
|
553 |
else {
|
|
|
554 |
wavsize *= i;
|
|
|
555 |
}
|
|
|
556 |
|
|
|
557 |
if ( 0 == lame_get_disable_waveheader( gfp ) )
|
|
|
558 |
if (!fseek(outf, 0l, SEEK_SET)) /* if outf is seekable, rewind and adjust length */
|
|
|
559 |
WriteWaveHeader(outf, wavsize, lame_get_in_samplerate( gfp ),
|
|
|
560 |
tmp_num_channels, 16);
|
|
|
561 |
fclose(outf);
|
|
|
562 |
|
|
|
563 |
decoder_progress_finish(gfp);
|
|
|
564 |
return 0;
|
|
|
565 |
}
|
|
|
566 |
|
|
|
567 |
|
|
|
568 |
|
|
|
569 |
|
|
|
570 |
|
|
|
571 |
|
|
|
572 |
#if defined(LIBSNDFILE)
|
|
|
573 |
|
|
|
574 |
#if 0 /* currently disabled */
|
|
|
575 |
# include "sndfile.h" // prototype for sf_get_lib_version()
|
|
|
576 |
void
|
|
|
577 |
print_sndlib_version(FILE * fp)
|
|
|
578 |
{
|
|
|
579 |
char tmp[80];
|
|
|
580 |
sf_get_lib_version(tmp, sizeof(tmp));
|
|
|
581 |
fprintf(fp,
|
|
|
582 |
"Input handled by %s (http://www.zip.com.au/~erikd/libsndfile/)\n",
|
|
|
583 |
tmp);
|
|
|
584 |
}
|
|
|
585 |
#endif
|
|
|
586 |
|
|
|
587 |
/*
|
|
|
588 |
** Copyright (C) 1999 Albert Faber
|
|
|
589 |
**
|
|
|
590 |
* This library is free software; you can redistribute it and/or
|
|
|
591 |
* modify it under the terms of the GNU Library General Public
|
|
|
592 |
* License as published by the Free Software Foundation; either
|
|
|
593 |
* version 2 of the License, or (at your option) any later version.
|
|
|
594 |
*
|
|
|
595 |
* This library is distributed in the hope that it will be useful,
|
|
|
596 |
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
|
597 |
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
|
598 |
* Library General Public License for more details.
|
|
|
599 |
*
|
|
|
600 |
* You should have received a copy of the GNU Library General Public
|
|
|
601 |
* License along with this library; if not, write to the
|
|
|
602 |
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
|
|
603 |
* Boston, MA 02111-1307, USA.
|
|
|
604 |
*/
|
|
|
605 |
|
|
|
606 |
|
|
|
607 |
|
|
|
608 |
|
|
|
609 |
|
|
|
610 |
|
|
|
611 |
void
|
|
|
612 |
CloseSndFile(sound_file_format input, FILE * musicin)
|
|
|
613 |
{
|
|
|
614 |
SNDFILE *gs_pSndFileIn = (SNDFILE *) musicin;
|
|
|
615 |
if (input == sf_mp1 || input == sf_mp2 || input == sf_mp3) {
|
|
|
616 |
#ifndef AMIGA_MPEGA
|
|
|
617 |
if (fclose(musicin) != 0) {
|
|
|
618 |
fprintf(stderr, "Could not close audio input file\n");
|
|
|
619 |
exit(2);
|
|
|
620 |
}
|
|
|
621 |
#endif
|
|
|
622 |
}
|
|
|
623 |
else {
|
|
|
624 |
if (gs_pSndFileIn) {
|
|
|
625 |
if (sf_close(gs_pSndFileIn) != 0) {
|
|
|
626 |
fprintf(stderr, "Could not close sound file \n");
|
|
|
627 |
exit(2);
|
|
|
628 |
}
|
|
|
629 |
}
|
|
|
630 |
}
|
|
|
631 |
}
|
|
|
632 |
|
|
|
633 |
|
|
|
634 |
|
|
|
635 |
FILE *
|
|
|
636 |
OpenSndFile(lame_global_flags * gfp, char *inPath)
|
|
|
637 |
{
|
|
|
638 |
char *lpszFileName = inPath;
|
|
|
639 |
FILE *musicin;
|
|
|
640 |
SNDFILE *gs_pSndFileIn;
|
|
|
641 |
SF_INFO gs_wfInfo;
|
|
|
642 |
|
|
|
643 |
if (input_format == sf_mp1 ||
|
|
|
644 |
input_format == sf_mp2 || input_format == sf_mp3) {
|
|
|
645 |
#ifdef AMIGA_MPEGA
|
|
|
646 |
if (-1 == lame_decode_initfile(lpszFileName, &mp3input_data)) {
|
|
|
647 |
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
|
|
|
648 |
lpszFileName);
|
|
|
649 |
exit(1);
|
|
|
650 |
}
|
|
|
651 |
#endif
|
|
|
652 |
#ifdef HAVE_MPGLIB
|
|
|
653 |
if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
|
|
|
654 |
fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
|
|
|
655 |
exit(1);
|
|
|
656 |
}
|
|
|
657 |
if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
|
|
|
658 |
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
|
|
|
659 |
lpszFileName);
|
|
|
660 |
exit(1);
|
|
|
661 |
}
|
|
|
662 |
#endif
|
|
|
663 |
|
|
|
664 |
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
|
|
|
665 |
fprintf( stderr,
|
|
|
666 |
"Unsupported number of channels: %ud\n",
|
|
|
667 |
mp3input_data.stereo );
|
|
|
668 |
exit( 1 );
|
|
|
669 |
}
|
|
|
670 |
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
|
|
|
671 |
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
|
|
|
672 |
}
|
|
|
673 |
else if (input_format == sf_ogg) {
|
|
|
674 |
#ifdef HAVE_VORBIS
|
|
|
675 |
if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
|
|
|
676 |
fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
|
|
|
677 |
exit(1);
|
|
|
678 |
}
|
|
|
679 |
if ( -1 == lame_decode_ogg_initfile( gfp,
|
|
|
680 |
musicin,
|
|
|
681 |
&mp3input_data ) ) {
|
|
|
682 |
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
|
|
|
683 |
lpszFileName);
|
|
|
684 |
exit(1);
|
|
|
685 |
}
|
|
|
686 |
#else
|
|
|
687 |
fprintf(stderr, "mp3enc not compiled with libvorbis support.\n");
|
|
|
688 |
exit(1);
|
|
|
689 |
#endif
|
|
|
690 |
|
|
|
691 |
|
|
|
692 |
}
|
|
|
693 |
else {
|
|
|
694 |
|
|
|
695 |
/* Try to open the sound file */
|
|
|
696 |
/* set some defaults incase input is raw PCM */
|
|
|
697 |
gs_wfInfo.seekable = (input_format != sf_raw); /* if user specified -r, set to not seekable */
|
|
|
698 |
gs_wfInfo.samplerate = lame_get_in_samplerate( gfp );
|
|
|
699 |
gs_wfInfo.pcmbitwidth = 16;
|
|
|
700 |
gs_wfInfo.channels = lame_get_num_channels( gfp );
|
|
|
701 |
#ifndef WORDS_BIGENDIAN
|
|
|
702 |
/* little endian */
|
|
|
703 |
if (swapbytes)
|
|
|
704 |
gs_wfInfo.format = SF_FORMAT_RAW_BE;
|
|
|
705 |
else
|
|
|
706 |
gs_wfInfo.format = SF_FORMAT_RAW_LE;
|
|
|
707 |
#else
|
|
|
708 |
if (swapbytes)
|
|
|
709 |
gs_wfInfo.format = SF_FORMAT_RAW_LE;
|
|
|
710 |
else
|
|
|
711 |
gs_wfInfo.format = SF_FORMAT_RAW_BE;
|
|
|
712 |
#endif
|
|
|
713 |
|
|
|
714 |
gs_pSndFileIn = sf_open_read(lpszFileName, &gs_wfInfo);
|
|
|
715 |
musicin = (SNDFILE *) gs_pSndFileIn;
|
|
|
716 |
|
|
|
717 |
/* Check result */
|
|
|
718 |
if (gs_pSndFileIn == NULL) {
|
|
|
719 |
sf_perror(gs_pSndFileIn);
|
|
|
720 |
fprintf(stderr, "Could not open sound file \"%s\".\n",
|
|
|
721 |
lpszFileName);
|
|
|
722 |
exit(1);
|
|
|
723 |
}
|
|
|
724 |
|
|
|
725 |
if ((gs_wfInfo.format == SF_FORMAT_RAW_LE) ||
|
|
|
726 |
(gs_wfInfo.format == SF_FORMAT_RAW_BE)) input_format = sf_raw;
|
|
|
727 |
|
|
|
728 |
#ifdef _DEBUG_SND_FILE
|
|
|
729 |
DEBUGF("\n\nSF_INFO structure\n");
|
|
|
730 |
DEBUGF("samplerate :%d\n", gs_wfInfo.samplerate);
|
|
|
731 |
DEBUGF("samples :%d\n", gs_wfInfo.samples);
|
|
|
732 |
DEBUGF("channels :%d\n", gs_wfInfo.channels);
|
|
|
733 |
DEBUGF("pcmbitwidth :%d\n", gs_wfInfo.pcmbitwidth);
|
|
|
734 |
DEBUGF("format :");
|
|
|
735 |
|
|
|
736 |
/* new formats from sbellon@sbellon.de 1/2000 */
|
|
|
737 |
|
|
|
738 |
switch (gs_wfInfo.format & SF_FORMAT_TYPEMASK) {
|
|
|
739 |
case SF_FORMAT_WAV:
|
|
|
740 |
DEBUGF("Microsoft WAV format (big endian). ");
|
|
|
741 |
break;
|
|
|
742 |
case SF_FORMAT_AIFF:
|
|
|
743 |
DEBUGF("Apple/SGI AIFF format (little endian). ");
|
|
|
744 |
break;
|
|
|
745 |
case SF_FORMAT_AU:
|
|
|
746 |
DEBUGF("Sun/NeXT AU format (big endian). ");
|
|
|
747 |
break;
|
|
|
748 |
case SF_FORMAT_AULE:
|
|
|
749 |
DEBUGF("DEC AU format (little endian). ");
|
|
|
750 |
break;
|
|
|
751 |
case SF_FORMAT_RAW:
|
|
|
752 |
DEBUGF("RAW PCM data. ");
|
|
|
753 |
break;
|
|
|
754 |
case SF_FORMAT_PAF:
|
|
|
755 |
DEBUGF("Ensoniq PARIS file format. ");
|
|
|
756 |
break;
|
|
|
757 |
case SF_FORMAT_SVX:
|
|
|
758 |
DEBUGF("Amiga IFF / SVX8 / SV16 format. ");
|
|
|
759 |
break;
|
|
|
760 |
case SF_FORMAT_NIST:
|
|
|
761 |
DEBUGF("Sphere NIST format. ");
|
|
|
762 |
break;
|
|
|
763 |
default:
|
|
|
764 |
assert(0);
|
|
|
765 |
break;
|
|
|
766 |
}
|
|
|
767 |
|
|
|
768 |
switch (gs_wfInfo.format & SF_FORMAT_SUBMASK) {
|
|
|
769 |
case SF_FORMAT_PCM:
|
|
|
770 |
DEBUGF("PCM data in 8, 16, 24 or 32 bits.");
|
|
|
771 |
break;
|
|
|
772 |
case SF_FORMAT_FLOAT:
|
|
|
773 |
DEBUGF("32 bit Intel x86 floats.");
|
|
|
774 |
break;
|
|
|
775 |
case SF_FORMAT_ULAW:
|
|
|
776 |
DEBUGF("U-Law encoded.");
|
|
|
777 |
break;
|
|
|
778 |
case SF_FORMAT_ALAW:
|
|
|
779 |
DEBUGF("A-Law encoded.");
|
|
|
780 |
break;
|
|
|
781 |
case SF_FORMAT_IMA_ADPCM:
|
|
|
782 |
DEBUGF("IMA ADPCM.");
|
|
|
783 |
break;
|
|
|
784 |
case SF_FORMAT_MS_ADPCM:
|
|
|
785 |
DEBUGF("Microsoft ADPCM.");
|
|
|
786 |
break;
|
|
|
787 |
case SF_FORMAT_PCM_BE:
|
|
|
788 |
DEBUGF("Big endian PCM data.");
|
|
|
789 |
break;
|
|
|
790 |
case SF_FORMAT_PCM_LE:
|
|
|
791 |
DEBUGF("Little endian PCM data.");
|
|
|
792 |
break;
|
|
|
793 |
case SF_FORMAT_PCM_S8:
|
|
|
794 |
DEBUGF("Signed 8 bit PCM.");
|
|
|
795 |
break;
|
|
|
796 |
case SF_FORMAT_PCM_U8:
|
|
|
797 |
DEBUGF("Unsigned 8 bit PCM.");
|
|
|
798 |
break;
|
|
|
799 |
case SF_FORMAT_SVX_FIB:
|
|
|
800 |
DEBUGF("SVX Fibonacci Delta encoding.");
|
|
|
801 |
break;
|
|
|
802 |
case SF_FORMAT_SVX_EXP:
|
|
|
803 |
DEBUGF("SVX Exponential Delta encoding.");
|
|
|
804 |
break;
|
|
|
805 |
default:
|
|
|
806 |
assert(0);
|
|
|
807 |
break;
|
|
|
808 |
}
|
|
|
809 |
|
|
|
810 |
DEBUGF("\n");
|
|
|
811 |
DEBUGF("pcmbitwidth :%d\n", gs_wfInfo.pcmbitwidth);
|
|
|
812 |
DEBUGF("sections :%d\n", gs_wfInfo.sections);
|
|
|
813 |
DEBUGF("seekable :\n", gs_wfInfo.seekable);
|
|
|
814 |
#endif
|
|
|
815 |
|
|
|
816 |
(void) lame_set_num_samples( gfp, gs_wfInfo.samples );
|
|
|
817 |
if( -1 == lame_set_num_channels( gfp, gs_wfInfo.channels ) ) {
|
|
|
818 |
fprintf( stderr,
|
|
|
819 |
"Unsupported number of channels: %ud\n",
|
|
|
820 |
gs_wfInfo.channels );
|
|
|
821 |
exit( 1 );
|
|
|
822 |
}
|
|
|
823 |
(void) lame_set_in_samplerate( gfp, gs_wfInfo.samplerate );
|
|
|
824 |
pcmbitwidth = gs_wfInfo.pcmbitwidth;
|
|
|
825 |
}
|
|
|
826 |
|
|
|
827 |
if (lame_get_num_samples( gfp ) == MAX_U_32_NUM) {
|
|
|
828 |
/* try to figure out num_samples */
|
|
|
829 |
double flen = lame_get_file_size( lpszFileName );
|
|
|
830 |
|
|
|
831 |
if (flen >= 0) {
|
|
|
832 |
/* try file size, assume 2 bytes per sample */
|
|
|
833 |
if (input_format == sf_mp1 ||
|
|
|
834 |
input_format == sf_mp2 || input_format == sf_mp3) {
|
|
|
835 |
double totalseconds =
|
|
|
836 |
(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
|
|
|
837 |
unsigned long tmp_num_samples =
|
|
|
838 |
totalseconds * lame_get_in_samplerate( gfp );
|
|
|
839 |
|
|
|
840 |
(void) lame_set_num_samples( gfp, tmp_num_samples );
|
|
|
841 |
mp3input_data.nsamp = tmp_num_samples;
|
|
|
842 |
}
|
|
|
843 |
else {
|
|
|
844 |
lame_set_num_samples( gfp,
|
|
|
845 |
flen / (2 * lame_get_num_channels( gfp )) );
|
|
|
846 |
}
|
|
|
847 |
}
|
|
|
848 |
}
|
|
|
849 |
|
|
|
850 |
|
|
|
851 |
return musicin;
|
|
|
852 |
}
|
|
|
853 |
|
|
|
854 |
|
|
|
855 |
/************************************************************************
|
|
|
856 |
*
|
|
|
857 |
* read_samples()
|
|
|
858 |
*
|
|
|
859 |
* PURPOSE: reads the PCM samples from a file to the buffer
|
|
|
860 |
*
|
|
|
861 |
* SEMANTICS:
|
|
|
862 |
* Reads #samples_read# number of shorts from #musicin# filepointer
|
|
|
863 |
* into #sample_buffer[]#. Returns the number of samples read.
|
|
|
864 |
*
|
|
|
865 |
************************************************************************/
|
|
|
866 |
|
|
|
867 |
static int
|
|
|
868 |
read_samples_pcm(FILE * const musicin, short sample_buffer[2304],
|
|
|
869 |
int frame_size /* unused */ , int samples_to_read)
|
|
|
870 |
{
|
|
|
871 |
int i;
|
|
|
872 |
int samples_read;
|
|
|
873 |
|
|
|
874 |
samples_read =
|
|
|
875 |
sf_read_short((SNDFILE *) musicin, sample_buffer, samples_to_read);
|
|
|
876 |
|
|
|
877 |
switch (pcmbitwidth) {
|
|
|
878 |
case 8:
|
|
|
879 |
for (i = 0; i < samples_read; i++)
|
|
|
880 |
sample_buffer[i] <<= 8;
|
|
|
881 |
break;
|
|
|
882 |
case 16:
|
|
|
883 |
break;
|
|
|
884 |
default:
|
|
|
885 |
fprintf(stderr, "Only 8 and 16 bit input files supported \n");
|
|
|
886 |
exit(1);
|
|
|
887 |
}
|
|
|
888 |
|
|
|
889 |
return samples_read;
|
|
|
890 |
}
|
|
|
891 |
|
|
|
892 |
|
|
|
893 |
#else /* defined(LIBSNDFILE) */
|
|
|
894 |
|
|
|
895 |
/************************************************************************
|
|
|
896 |
************************************************************************
|
|
|
897 |
************************************************************************
|
|
|
898 |
************************************************************************
|
|
|
899 |
************************************************************************
|
|
|
900 |
************************************************************************
|
|
|
901 |
*
|
|
|
902 |
* OLD ISO/LAME routines follow. Used if you dont have LIBSNDFILE
|
|
|
903 |
* or for stdin/stdout support
|
|
|
904 |
*
|
|
|
905 |
************************************************************************
|
|
|
906 |
************************************************************************
|
|
|
907 |
************************************************************************
|
|
|
908 |
************************************************************************
|
|
|
909 |
************************************************************************
|
|
|
910 |
************************************************************************/
|
|
|
911 |
|
|
|
912 |
|
|
|
913 |
|
|
|
914 |
/************************************************************************
|
|
|
915 |
*
|
|
|
916 |
* read_samples()
|
|
|
917 |
*
|
|
|
918 |
* PURPOSE: reads the PCM samples from a file to the buffer
|
|
|
919 |
*
|
|
|
920 |
* SEMANTICS:
|
|
|
921 |
* Reads #samples_read# number of shorts from #musicin# filepointer
|
|
|
922 |
* into #sample_buffer[]#. Returns the number of samples read.
|
|
|
923 |
*
|
|
|
924 |
************************************************************************/
|
|
|
925 |
|
|
|
926 |
int
|
|
|
927 |
read_samples_pcm(FILE * musicin, short sample_buffer[2304], int frame_size,
|
|
|
928 |
int samples_to_read)
|
|
|
929 |
{
|
|
|
930 |
int samples_read;
|
|
|
931 |
int iswav = (input_format == sf_wave);
|
|
|
932 |
|
|
|
933 |
if (16 == pcmbitwidth) {
|
|
|
934 |
samples_read = fread(sample_buffer, 2, samples_to_read, musicin);
|
|
|
935 |
}
|
|
|
936 |
else if (8 == pcmbitwidth) {
|
|
|
937 |
char temp[2304];
|
|
|
938 |
int i;
|
|
|
939 |
samples_read = fread(temp, 1, samples_to_read, musicin);
|
|
|
940 |
for (i = 0; i < samples_read; ++i) {
|
|
|
941 |
/* note: 8bit .wav samples are unsigned */
|
|
|
942 |
/* map [0,255] -> [-32768,32767] */
|
|
|
943 |
sample_buffer[i] = ((short int)temp[i] - 128)*256 + 127;
|
|
|
944 |
}
|
|
|
945 |
}
|
|
|
946 |
else {
|
|
|
947 |
fprintf(stderr, "Only 8 and 16 bit input files supported \n");
|
|
|
948 |
exit(1);
|
|
|
949 |
}
|
|
|
950 |
if (ferror(musicin)) {
|
|
|
951 |
fprintf(stderr, "Error reading input file\n");
|
|
|
952 |
exit(1);
|
|
|
953 |
}
|
|
|
954 |
|
|
|
955 |
|
|
|
956 |
|
|
|
957 |
if (16 == pcmbitwidth) {
|
|
|
958 |
/* intel=littleEndian. wav files are always little endian */
|
|
|
959 |
#ifndef WORDS_BIGENDIAN
|
|
|
960 |
/* little endian */
|
|
|
961 |
if (!iswav)
|
|
|
962 |
SwapBytesInWords(sample_buffer, samples_read);
|
|
|
963 |
#else
|
|
|
964 |
/* big endian */
|
|
|
965 |
if (iswav)
|
|
|
966 |
SwapBytesInWords(sample_buffer, samples_read);
|
|
|
967 |
#endif
|
|
|
968 |
|
|
|
969 |
if (swapbytes)
|
|
|
970 |
SwapBytesInWords(sample_buffer, samples_read);
|
|
|
971 |
}
|
|
|
972 |
|
|
|
973 |
return samples_read;
|
|
|
974 |
}
|
|
|
975 |
|
|
|
976 |
|
|
|
977 |
|
|
|
978 |
/* AIFF Definitions */
|
|
|
979 |
|
|
|
980 |
#define IFF_ID_FORM 0x464f524d /* "FORM" */
|
|
|
981 |
#define IFF_ID_AIFF 0x41494646 /* "AIFF" */
|
|
|
982 |
#define IFF_ID_COMM 0x434f4d4d /* "COMM" */
|
|
|
983 |
#define IFF_ID_SSND 0x53534e44 /* "SSND" */
|
|
|
984 |
#define IFF_ID_MPEG 0x4d504547 /* "MPEG" */
|
|
|
985 |
|
|
|
986 |
|
|
|
987 |
#define WAV_ID_RIFF 0x52494646 /* "RIFF" */
|
|
|
988 |
#define WAV_ID_WAVE 0x57415645 /* "WAVE" */
|
|
|
989 |
#define WAV_ID_FMT 0x666d7420 /* "fmt " */
|
|
|
990 |
#define WAV_ID_DATA 0x64617461 /* "data" */
|
|
|
991 |
|
|
|
992 |
|
|
|
993 |
|
|
|
994 |
|
|
|
995 |
/*****************************************************************************
|
|
|
996 |
*
|
|
|
997 |
* Read Microsoft Wave headers
|
|
|
998 |
*
|
|
|
999 |
* By the time we get here the first 32-bits of the file have already been
|
|
|
1000 |
* read, and we're pretty sure that we're looking at a WAV file.
|
|
|
1001 |
*
|
|
|
1002 |
*****************************************************************************/
|
|
|
1003 |
|
|
|
1004 |
static int
|
|
|
1005 |
parse_wave_header(lame_global_flags * gfp, FILE * sf)
|
|
|
1006 |
{
|
|
|
1007 |
int format_tag = 0;
|
|
|
1008 |
int channels = 0;
|
|
|
1009 |
int block_align = 0;
|
|
|
1010 |
int bits_per_sample = 0;
|
|
|
1011 |
int samples_per_sec = 0;
|
|
|
1012 |
int avg_bytes_per_sec = 0;
|
|
|
1013 |
|
|
|
1014 |
|
|
|
1015 |
int is_wav = 0;
|
|
|
1016 |
long data_length = 0, file_length, subSize = 0;
|
|
|
1017 |
int loop_sanity = 0;
|
|
|
1018 |
|
|
|
1019 |
file_length = Read32BitsHighLow(sf);
|
|
|
1020 |
|
|
|
1021 |
if (Read32BitsHighLow(sf) != WAV_ID_WAVE)
|
|
|
1022 |
return 0;
|
|
|
1023 |
|
|
|
1024 |
for (loop_sanity = 0; loop_sanity < 20; ++loop_sanity) {
|
|
|
1025 |
int type = Read32BitsHighLow(sf);
|
|
|
1026 |
|
|
|
1027 |
if (type == WAV_ID_FMT) {
|
|
|
1028 |
subSize = Read32BitsLowHigh(sf);
|
|
|
1029 |
if (subSize < 16) {
|
|
|
1030 |
/*DEBUGF(
|
|
|
1031 |
"'fmt' chunk too short (only %ld bytes)!", subSize); */
|
|
|
1032 |
return 0;
|
|
|
1033 |
}
|
|
|
1034 |
|
|
|
1035 |
format_tag = Read16BitsLowHigh(sf);
|
|
|
1036 |
subSize -= 2;
|
|
|
1037 |
channels = Read16BitsLowHigh(sf);
|
|
|
1038 |
subSize -= 2;
|
|
|
1039 |
samples_per_sec = Read32BitsLowHigh(sf);
|
|
|
1040 |
subSize -= 4;
|
|
|
1041 |
avg_bytes_per_sec = Read32BitsLowHigh(sf);
|
|
|
1042 |
subSize -= 4;
|
|
|
1043 |
block_align = Read16BitsLowHigh(sf);
|
|
|
1044 |
subSize -= 2;
|
|
|
1045 |
bits_per_sample = Read16BitsLowHigh(sf);
|
|
|
1046 |
subSize -= 2;
|
|
|
1047 |
|
|
|
1048 |
/* DEBUGF(" skipping %d bytes\n", subSize); */
|
|
|
1049 |
|
|
|
1050 |
if (subSize > 0) {
|
|
|
1051 |
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
|
|
|
1052 |
return 0;
|
|
|
1053 |
};
|
|
|
1054 |
|
|
|
1055 |
}
|
|
|
1056 |
else if (type == WAV_ID_DATA) {
|
|
|
1057 |
subSize = Read32BitsLowHigh(sf);
|
|
|
1058 |
data_length = subSize;
|
|
|
1059 |
is_wav = 1;
|
|
|
1060 |
/* We've found the audio data. Read no further! */
|
|
|
1061 |
break;
|
|
|
1062 |
|
|
|
1063 |
}
|
|
|
1064 |
else {
|
|
|
1065 |
subSize = Read32BitsLowHigh(sf);
|
|
|
1066 |
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
|
|
|
1067 |
return 0;
|
|
|
1068 |
}
|
|
|
1069 |
}
|
|
|
1070 |
|
|
|
1071 |
if (format_tag != 1) {
|
|
|
1072 |
return 0; /* oh no! non-supported format */
|
|
|
1073 |
}
|
|
|
1074 |
|
|
|
1075 |
|
|
|
1076 |
if (is_wav) {
|
|
|
1077 |
/* make sure the header is sane */
|
|
|
1078 |
if( -1 == lame_set_num_channels( gfp, channels ) ) {
|
|
|
1079 |
fprintf( stderr,
|
|
|
1080 |
"Unsupported number of channels: %ud\n",
|
|
|
1081 |
channels );
|
|
|
1082 |
exit( 1 );
|
|
|
1083 |
}
|
|
|
1084 |
(void) lame_set_in_samplerate( gfp, samples_per_sec );
|
|
|
1085 |
pcmbitwidth = bits_per_sample;
|
|
|
1086 |
(void) lame_set_num_samples( gfp,
|
|
|
1087 |
data_length / (channels * ((bits_per_sample+7) / 8)) );
|
|
|
1088 |
}
|
|
|
1089 |
return is_wav;
|
|
|
1090 |
}
|
|
|
1091 |
|
|
|
1092 |
|
|
|
1093 |
|
|
|
1094 |
/************************************************************************
|
|
|
1095 |
* aiff_check2
|
|
|
1096 |
*
|
|
|
1097 |
* PURPOSE: Checks AIFF header information to make sure it is valid.
|
|
|
1098 |
* returns 0 on success, 1 on errors
|
|
|
1099 |
************************************************************************/
|
|
|
1100 |
|
|
|
1101 |
int
|
|
|
1102 |
aiff_check2(const char *file_name, IFF_AIFF * const pcm_aiff_data)
|
|
|
1103 |
{
|
|
|
1104 |
if (pcm_aiff_data->sampleType != IFF_ID_SSND) {
|
|
|
1105 |
fprintf(stderr, "Sound data is not PCM in '%s'\n", file_name);
|
|
|
1106 |
return 1;
|
|
|
1107 |
}
|
|
|
1108 |
if (pcm_aiff_data->sampleSize != sizeof(short) * CHAR_BIT) {
|
|
|
1109 |
fprintf(stderr, "Sound data is not %i bits in '%s'\n",
|
|
|
1110 |
sizeof(short) * CHAR_BIT, file_name);
|
|
|
1111 |
return 1;
|
|
|
1112 |
}
|
|
|
1113 |
if (pcm_aiff_data->numChannels != 1 && pcm_aiff_data->numChannels != 2) {
|
|
|
1114 |
fprintf(stderr, "Sound data is not mono or stereo in '%s'\n",
|
|
|
1115 |
file_name);
|
|
|
1116 |
return 1;
|
|
|
1117 |
}
|
|
|
1118 |
if (pcm_aiff_data->blkAlgn.blockSize != 0) {
|
|
|
1119 |
fprintf(stderr, "Block size is not 0 bytes in '%s'\n", file_name);
|
|
|
1120 |
return 1;
|
|
|
1121 |
}
|
|
|
1122 |
if (pcm_aiff_data->blkAlgn.offset != 0) {
|
|
|
1123 |
fprintf(stderr, "Block offset is not 0 bytes in '%s'\n", file_name);
|
|
|
1124 |
return 1;
|
|
|
1125 |
}
|
|
|
1126 |
|
|
|
1127 |
return 0;
|
|
|
1128 |
}
|
|
|
1129 |
|
|
|
1130 |
/*****************************************************************************
|
|
|
1131 |
*
|
|
|
1132 |
* Read Audio Interchange File Format (AIFF) headers.
|
|
|
1133 |
*
|
|
|
1134 |
* By the time we get here the first 32 bits of the file have already been
|
|
|
1135 |
* read, and we're pretty sure that we're looking at an AIFF file.
|
|
|
1136 |
*
|
|
|
1137 |
*****************************************************************************/
|
|
|
1138 |
|
|
|
1139 |
static int
|
|
|
1140 |
parse_aiff_header(lame_global_flags * gfp, FILE * sf)
|
|
|
1141 |
{
|
|
|
1142 |
int is_aiff = 0;
|
|
|
1143 |
long chunkSize = 0, subSize = 0;
|
|
|
1144 |
IFF_AIFF aiff_info;
|
|
|
1145 |
|
|
|
1146 |
memset(&aiff_info, 0, sizeof(aiff_info));
|
|
|
1147 |
chunkSize = Read32BitsHighLow(sf);
|
|
|
1148 |
|
|
|
1149 |
if (Read32BitsHighLow(sf) != IFF_ID_AIFF)
|
|
|
1150 |
return 0;
|
|
|
1151 |
|
|
|
1152 |
while (chunkSize > 0) {
|
|
|
1153 |
int type = Read32BitsHighLow(sf);
|
|
|
1154 |
chunkSize -= 4;
|
|
|
1155 |
|
|
|
1156 |
/* DEBUGF(
|
|
|
1157 |
"found chunk type %08x '%4.4s'\n", type, (char*)&type); */
|
|
|
1158 |
|
|
|
1159 |
/* don't use a switch here to make it easier to use 'break' for SSND */
|
|
|
1160 |
if (type == IFF_ID_COMM) {
|
|
|
1161 |
subSize = Read32BitsHighLow(sf);
|
|
|
1162 |
chunkSize -= subSize;
|
|
|
1163 |
|
|
|
1164 |
aiff_info.numChannels = Read16BitsHighLow(sf);
|
|
|
1165 |
subSize -= 2;
|
|
|
1166 |
aiff_info.numSampleFrames = Read32BitsHighLow(sf);
|
|
|
1167 |
subSize -= 4;
|
|
|
1168 |
aiff_info.sampleSize = Read16BitsHighLow(sf);
|
|
|
1169 |
subSize -= 2;
|
|
|
1170 |
aiff_info.sampleRate = ReadIeeeExtendedHighLow(sf);
|
|
|
1171 |
subSize -= 10;
|
|
|
1172 |
|
|
|
1173 |
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
|
|
|
1174 |
return 0;
|
|
|
1175 |
|
|
|
1176 |
}
|
|
|
1177 |
else if (type == IFF_ID_SSND) {
|
|
|
1178 |
subSize = Read32BitsHighLow(sf);
|
|
|
1179 |
chunkSize -= subSize;
|
|
|
1180 |
|
|
|
1181 |
aiff_info.blkAlgn.offset = Read32BitsHighLow(sf);
|
|
|
1182 |
subSize -= 4;
|
|
|
1183 |
aiff_info.blkAlgn.blockSize = Read32BitsHighLow(sf);
|
|
|
1184 |
subSize -= 4;
|
|
|
1185 |
|
|
|
1186 |
if (fskip(sf, (long) aiff_info.blkAlgn.offset, SEEK_CUR) != 0)
|
|
|
1187 |
return 0;
|
|
|
1188 |
|
|
|
1189 |
aiff_info.sampleType = IFF_ID_SSND;
|
|
|
1190 |
is_aiff = 1;
|
|
|
1191 |
|
|
|
1192 |
/* We've found the audio data. Read no further! */
|
|
|
1193 |
break;
|
|
|
1194 |
|
|
|
1195 |
}
|
|
|
1196 |
else {
|
|
|
1197 |
subSize = Read32BitsHighLow(sf);
|
|
|
1198 |
chunkSize -= subSize;
|
|
|
1199 |
|
|
|
1200 |
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
|
|
|
1201 |
return 0;
|
|
|
1202 |
}
|
|
|
1203 |
}
|
|
|
1204 |
|
|
|
1205 |
/* DEBUGF("Parsed AIFF %d\n", is_aiff); */
|
|
|
1206 |
if (is_aiff) {
|
|
|
1207 |
/* make sure the header is sane */
|
|
|
1208 |
if (0 != aiff_check2("name" /*???????????? */ , &aiff_info))
|
|
|
1209 |
return 0;
|
|
|
1210 |
if( -1 == lame_set_num_channels( gfp, aiff_info.numChannels ) ) {
|
|
|
1211 |
fprintf( stderr,
|
|
|
1212 |
"Unsupported number of channels: %ud\n",
|
|
|
1213 |
aiff_info.numChannels );
|
|
|
1214 |
exit( 1 );
|
|
|
1215 |
}
|
|
|
1216 |
(void) lame_set_in_samplerate( gfp, aiff_info.sampleRate );
|
|
|
1217 |
pcmbitwidth = aiff_info.sampleSize;
|
|
|
1218 |
(void) lame_set_num_samples( gfp, aiff_info.numSampleFrames );
|
|
|
1219 |
}
|
|
|
1220 |
return is_aiff;
|
|
|
1221 |
}
|
|
|
1222 |
|
|
|
1223 |
|
|
|
1224 |
|
|
|
1225 |
/************************************************************************
|
|
|
1226 |
*
|
|
|
1227 |
* parse_file_header
|
|
|
1228 |
*
|
|
|
1229 |
* PURPOSE: Read the header from a bytestream. Try to determine whether
|
|
|
1230 |
* it's a WAV file or AIFF without rewinding, since rewind
|
|
|
1231 |
* doesn't work on pipes and there's a good chance we're reading
|
|
|
1232 |
* from stdin (otherwise we'd probably be using libsndfile).
|
|
|
1233 |
*
|
|
|
1234 |
* When this function returns, the file offset will be positioned at the
|
|
|
1235 |
* beginning of the sound data.
|
|
|
1236 |
*
|
|
|
1237 |
************************************************************************/
|
|
|
1238 |
|
|
|
1239 |
void
|
|
|
1240 |
parse_file_header(lame_global_flags * gfp, FILE * sf)
|
|
|
1241 |
{
|
|
|
1242 |
|
|
|
1243 |
int type = Read32BitsHighLow(sf);
|
|
|
1244 |
/*
|
|
|
1245 |
DEBUGF(
|
|
|
1246 |
"First word of input stream: %08x '%4.4s'\n", type, (char*) &type);
|
|
|
1247 |
*/
|
|
|
1248 |
count_samples_carefully = 0;
|
|
|
1249 |
input_format = sf_raw;
|
|
|
1250 |
|
|
|
1251 |
if (type == WAV_ID_RIFF) {
|
|
|
1252 |
/* It's probably a WAV file */
|
|
|
1253 |
if (parse_wave_header(gfp, sf)) {
|
|
|
1254 |
input_format = sf_wave;
|
|
|
1255 |
count_samples_carefully = 1;
|
|
|
1256 |
} else {
|
|
|
1257 |
fprintf( stderr, "Warning: corrupt or unsupported WAVE format\n");
|
|
|
1258 |
}
|
|
|
1259 |
}
|
|
|
1260 |
else if (type == IFF_ID_FORM) {
|
|
|
1261 |
/* It's probably an AIFF file */
|
|
|
1262 |
if (parse_aiff_header(gfp, sf)) {
|
|
|
1263 |
input_format = sf_aiff;
|
|
|
1264 |
count_samples_carefully = 1;
|
|
|
1265 |
}
|
|
|
1266 |
}
|
|
|
1267 |
if (input_format == sf_raw) {
|
|
|
1268 |
/*
|
|
|
1269 |
** Assume it's raw PCM. Since the audio data is assumed to begin
|
|
|
1270 |
** at byte zero, this will unfortunately require seeking.
|
|
|
1271 |
*/
|
|
|
1272 |
if (fseek(sf, 0L, SEEK_SET) != 0) {
|
|
|
1273 |
/* ignore errors */
|
|
|
1274 |
}
|
|
|
1275 |
input_format = sf_raw;
|
|
|
1276 |
}
|
|
|
1277 |
}
|
|
|
1278 |
|
|
|
1279 |
|
|
|
1280 |
|
|
|
1281 |
void
|
|
|
1282 |
CloseSndFile(sound_file_format input, FILE * musicin)
|
|
|
1283 |
{
|
|
|
1284 |
if (fclose(musicin) != 0) {
|
|
|
1285 |
fprintf(stderr, "Could not close audio input file\n");
|
|
|
1286 |
exit(2);
|
|
|
1287 |
}
|
|
|
1288 |
}
|
|
|
1289 |
|
|
|
1290 |
|
|
|
1291 |
|
|
|
1292 |
|
|
|
1293 |
|
|
|
1294 |
FILE *
|
|
|
1295 |
OpenSndFile(lame_global_flags * gfp, char *inPath)
|
|
|
1296 |
{
|
|
|
1297 |
FILE *musicin;
|
|
|
1298 |
|
|
|
1299 |
/* set the defaults from info incase we cannot determine them from file */
|
|
|
1300 |
lame_set_num_samples( gfp, MAX_U_32_NUM );
|
|
|
1301 |
|
|
|
1302 |
|
|
|
1303 |
if (!strcmp(inPath, "-")) {
|
|
|
1304 |
lame_set_stream_binary_mode(musicin = stdin); /* Read from standard input. */
|
|
|
1305 |
}
|
|
|
1306 |
else {
|
|
|
1307 |
if ((musicin = fopen(inPath, "rb")) == NULL) {
|
|
|
1308 |
fprintf(stderr, "Could not find \"%s\".\n", inPath);
|
|
|
1309 |
exit(1);
|
|
|
1310 |
}
|
|
|
1311 |
}
|
|
|
1312 |
|
|
|
1313 |
if (input_format == sf_mp1 ||
|
|
|
1314 |
input_format == sf_mp2 || input_format == sf_mp3) {
|
|
|
1315 |
#ifdef AMIGA_MPEGA
|
|
|
1316 |
if (-1 == lame_decode_initfile(inPath, &mp3input_data)) {
|
|
|
1317 |
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
|
|
|
1318 |
inPath);
|
|
|
1319 |
exit(1);
|
|
|
1320 |
}
|
|
|
1321 |
#endif
|
|
|
1322 |
#ifdef HAVE_MPGLIB
|
|
|
1323 |
if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
|
|
|
1324 |
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
|
|
|
1325 |
inPath);
|
|
|
1326 |
exit(1);
|
|
|
1327 |
}
|
|
|
1328 |
#endif
|
|
|
1329 |
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
|
|
|
1330 |
fprintf( stderr,
|
|
|
1331 |
"Unsupported number of channels: %ud\n",
|
|
|
1332 |
mp3input_data.stereo );
|
|
|
1333 |
exit( 1 );
|
|
|
1334 |
}
|
|
|
1335 |
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
|
|
|
1336 |
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
|
|
|
1337 |
}
|
|
|
1338 |
else if (input_format == sf_ogg) {
|
|
|
1339 |
#ifdef HAVE_VORBIS
|
|
|
1340 |
if ( -1 == lame_decode_ogg_initfile( gfp,
|
|
|
1341 |
musicin,
|
|
|
1342 |
&mp3input_data ) ) {
|
|
|
1343 |
fprintf(stderr, "Error reading headers in ogg input file %s.\n",
|
|
|
1344 |
inPath);
|
|
|
1345 |
exit(1);
|
|
|
1346 |
}
|
|
|
1347 |
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
|
|
|
1348 |
fprintf( stderr,
|
|
|
1349 |
"Unsupported number of channels: %ud\n",
|
|
|
1350 |
mp3input_data.stereo );
|
|
|
1351 |
exit( 1 );
|
|
|
1352 |
}
|
|
|
1353 |
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
|
|
|
1354 |
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
|
|
|
1355 |
#else
|
|
|
1356 |
fprintf(stderr, "mp3enc not compiled with libvorbis support.\n");
|
|
|
1357 |
exit(1);
|
|
|
1358 |
#endif
|
|
|
1359 |
}
|
|
|
1360 |
else {
|
|
|
1361 |
if (input_format != sf_raw) {
|
|
|
1362 |
parse_file_header(gfp, musicin);
|
|
|
1363 |
}
|
|
|
1364 |
|
|
|
1365 |
if (0 && input_format == sf_raw) {
|
|
|
1366 |
fprintf(stderr, "Assuming raw pcm input file");
|
|
|
1367 |
if (swapbytes)
|
|
|
1368 |
fprintf(stderr, " : Forcing byte-swapping\n");
|
|
|
1369 |
else
|
|
|
1370 |
fprintf(stderr, "\n");
|
|
|
1371 |
}
|
|
|
1372 |
}
|
|
|
1373 |
|
|
|
1374 |
|
|
|
1375 |
if (lame_get_num_samples( gfp ) == MAX_U_32_NUM && musicin != stdin) {
|
|
|
1376 |
double flen = lame_get_file_size(inPath); /* try to figure out num_samples */
|
|
|
1377 |
|
|
|
1378 |
if (flen >= 0) {
|
|
|
1379 |
|
|
|
1380 |
/* try file size, assume 2 bytes per sample */
|
|
|
1381 |
if (input_format == sf_mp1 ||
|
|
|
1382 |
input_format == sf_mp2 || input_format == sf_mp3) {
|
|
|
1383 |
if (mp3input_data.bitrate > 0) {
|
|
|
1384 |
double totalseconds =
|
|
|
1385 |
(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
|
|
|
1386 |
unsigned long tmp_num_samples =
|
|
|
1387 |
totalseconds * lame_get_in_samplerate( gfp );
|
|
|
1388 |
|
|
|
1389 |
(void) lame_set_num_samples( gfp, tmp_num_samples );
|
|
|
1390 |
mp3input_data.nsamp = tmp_num_samples;
|
|
|
1391 |
}
|
|
|
1392 |
}
|
|
|
1393 |
else {
|
|
|
1394 |
(void) lame_set_num_samples( gfp,
|
|
|
1395 |
flen / (2 * lame_get_num_channels( gfp )) );
|
|
|
1396 |
}
|
|
|
1397 |
}
|
|
|
1398 |
}
|
|
|
1399 |
return musicin;
|
|
|
1400 |
}
|
|
|
1401 |
#endif /* defined(LIBSNDFILE) */
|
|
|
1402 |
|
|
|
1403 |
|
|
|
1404 |
|
|
|
1405 |
|
|
|
1406 |
|
|
|
1407 |
#if defined(HAVE_MPGLIB)
|
|
|
1408 |
static int
|
|
|
1409 |
check_aid(const unsigned char *header)
|
|
|
1410 |
{
|
|
|
1411 |
return 0 == strncmp(header, "AiD\1", 4);
|
|
|
1412 |
}
|
|
|
1413 |
|
|
|
1414 |
/*
|
|
|
1415 |
* Please check this and don't kill me if there's a bug
|
|
|
1416 |
* This is a (nearly?) complete header analysis for a MPEG-1/2/2.5 Layer I, II or III
|
|
|
1417 |
* data stream
|
|
|
1418 |
*/
|
|
|
1419 |
|
|
|
1420 |
static int
|
|
|
1421 |
is_syncword_mp123(const void *const headerptr)
|
|
|
1422 |
{
|
|
|
1423 |
const unsigned char *const p = headerptr;
|
|
|
1424 |
static const char abl2[16] =
|
|
|
1425 |
{ 0, 7, 7, 7, 0, 7, 0, 0, 0, 0, 0, 8, 8, 8, 8, 8 };
|
|
|
1426 |
|
|
|
1427 |
if ((p[0] & 0xFF) != 0xFF)
|
|
|
1428 |
return 0; // first 8 bits must be '1'
|
|
|
1429 |
if ((p[1] & 0xE0) != 0xE0)
|
|
|
1430 |
return 0; // next 3 bits are also
|
|
|
1431 |
if ((p[1] & 0x18) == 0x08)
|
|
|
1432 |
return 0; // no MPEG-1, -2 or -2.5
|
|
|
1433 |
if ((p[1] & 0x06) == 0x00)
|
|
|
1434 |
return 0; // no Layer I, II and III
|
|
|
1435 |
if ((p[2] & 0xF0) == 0xF0)
|
|
|
1436 |
return 0; // bad bitrate
|
|
|
1437 |
if ((p[2] & 0x0C) == 0x0C)
|
|
|
1438 |
return 0; // no sample frequency with (32,44.1,48)/(1,2,4)
|
|
|
1439 |
if ((p[1] & 0x06) == 0x04) // illegal Layer II bitrate/Channel Mode comb
|
|
|
1440 |
if (abl2[p[2] >> 4] & (1 << (p[3] >> 6)))
|
|
|
1441 |
return 0;
|
|
|
1442 |
return 1;
|
|
|
1443 |
}
|
|
|
1444 |
|
|
|
1445 |
static int
|
|
|
1446 |
is_syncword_mp3(const void *const headerptr)
|
|
|
1447 |
{
|
|
|
1448 |
const unsigned char *const p = headerptr;
|
|
|
1449 |
|
|
|
1450 |
if ((p[0] & 0xFF) != 0xFF)
|
|
|
1451 |
return 0; // first 8 bits must be '1'
|
|
|
1452 |
if ((p[1] & 0xE0) != 0xE0)
|
|
|
1453 |
return 0; // next 3 bits are also
|
|
|
1454 |
if ((p[1] & 0x18) == 0x08)
|
|
|
1455 |
return 0; // no MPEG-1, -2 or -2.5
|
|
|
1456 |
if ((p[1] & 0x06) != 0x02)
|
|
|
1457 |
return 0; // no Layer III (can be merged with 'next 3 bits are also' test, but don't do this, this decreases readability)
|
|
|
1458 |
if ((p[2] & 0xF0) == 0xF0)
|
|
|
1459 |
return 0; // bad bitrate
|
|
|
1460 |
if ((p[2] & 0x0C) == 0x0C)
|
|
|
1461 |
return 0; // no sample frequency with (32,44.1,48)/(1,2,4)
|
|
|
1462 |
return 1;
|
|
|
1463 |
}
|
|
|
1464 |
|
|
|
1465 |
|
|
|
1466 |
int
|
|
|
1467 |
lame_decode_initfile(FILE * fd, mp3data_struct * mp3data)
|
|
|
1468 |
{
|
|
|
1469 |
// VBRTAGDATA pTagData;
|
|
|
1470 |
// int xing_header,len2,num_frames;
|
|
|
1471 |
unsigned char buf[100];
|
|
|
1472 |
int ret;
|
|
|
1473 |
int len, aid_header;
|
|
|
1474 |
short int pcm_l[1152], pcm_r[1152];
|
|
|
1475 |
|
|
|
1476 |
memset(mp3data, 0, sizeof(mp3data_struct));
|
|
|
1477 |
lame_decode_init();
|
|
|
1478 |
|
|
|
1479 |
len = 4;
|
|
|
1480 |
if (fread(&buf, 1, len, fd) != len)
|
|
|
1481 |
return -1; /* failed */
|
|
|
1482 |
aid_header = check_aid(buf);
|
|
|
1483 |
if (aid_header) {
|
|
|
1484 |
if (fread(&buf, 1, 2, fd) != 2)
|
|
|
1485 |
return -1; /* failed */
|
|
|
1486 |
aid_header = (unsigned char) buf[0] + 256 * (unsigned char) buf[1];
|
|
|
1487 |
fprintf(stderr, "Album ID found. length=%i \n", aid_header);
|
|
|
1488 |
/* skip rest of AID, except for 6 bytes we have already read */
|
|
|
1489 |
fskip(fd, aid_header - 6, SEEK_CUR);
|
|
|
1490 |
|
|
|
1491 |
/* read 4 more bytes to set up buffer for MP3 header check */
|
|
|
1492 |
len = fread(&buf, 1, 4, fd);
|
|
|
1493 |
}
|
|
|
1494 |
|
|
|
1495 |
|
|
|
1496 |
/* look for valid 4 byte MPEG header */
|
|
|
1497 |
if (len < 4)
|
|
|
1498 |
return -1;
|
|
|
1499 |
while (!is_syncword_mp123(buf)) {
|
|
|
1500 |
int i;
|
|
|
1501 |
for (i = 0; i < len - 1; i++)
|
|
|
1502 |
buf[i] = buf[i + 1];
|
|
|
1503 |
if (fread(buf + len - 1, 1, 1, fd) != 1)
|
|
|
1504 |
return -1; /* failed */
|
|
|
1505 |
}
|
|
|
1506 |
|
|
|
1507 |
|
|
|
1508 |
#if 0
|
|
|
1509 |
/* buffer 48 bytes so we can check for Xing header */
|
|
|
1510 |
len2 = fread(&buf[len], 1, 48 - len, fd);
|
|
|
1511 |
if (len2 != 48 - len)
|
|
|
1512 |
return -1;
|
|
|
1513 |
len = 48;
|
|
|
1514 |
|
|
|
1515 |
/* check first 48 bytes for Xing header */
|
|
|
1516 |
xing_header = GetVbrTag(&pTagData, (unsigned char *) buf);
|
|
|
1517 |
|
|
|
1518 |
if (xing_header && pTagData.headersize >= 48) {
|
|
|
1519 |
num_frames = pTagData.frames;
|
|
|
1520 |
fprintf(stderr,
|
|
|
1521 |
"\rXing VBR header dectected. MP3 file has %i frames\n",
|
|
|
1522 |
num_frames);
|
|
|
1523 |
|
|
|
1524 |
// skip the rest of the Xing header. LAME decoder ignores TOC data
|
|
|
1525 |
fskip(fd, pTagData.headersize - 48, SEEK_CUR);
|
|
|
1526 |
// buffer a few more bytes for next header check:
|
|
|
1527 |
len = fread(buf, 1, 4, fd);
|
|
|
1528 |
|
|
|
1529 |
}
|
|
|
1530 |
else {
|
|
|
1531 |
/* we have read 48 bytes, but did not find a Xing header */
|
|
|
1532 |
/* lets try and rewind the stream: */
|
|
|
1533 |
if (fseek(fd, -44, SEEK_CUR) != 0) {
|
|
|
1534 |
/* backwards fseek failed. input is probably a pipe */
|
|
|
1535 |
/* keep 'len' unchanged */
|
|
|
1536 |
}
|
|
|
1537 |
else {
|
|
|
1538 |
len -= 44;
|
|
|
1539 |
}
|
|
|
1540 |
}
|
|
|
1541 |
#endif
|
|
|
1542 |
|
|
|
1543 |
// now parse the current buffer looking for MP3 headers
|
|
|
1544 |
// we dont want to feed too much data to lame_decode1_headers -
|
|
|
1545 |
// we dont want it to actually decode the first frame
|
|
|
1546 |
// (as of 11/00: mpglib modified so that for the first frame where
|
|
|
1547 |
// headers are parsed, no data will be decoded. So the above is
|
|
|
1548 |
// now a moot point.
|
|
|
1549 |
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
|
|
|
1550 |
if (-1 == ret)
|
|
|
1551 |
return -1;
|
|
|
1552 |
|
|
|
1553 |
/* repeat until we decode a valid mp3 header */
|
|
|
1554 |
while (!mp3data->header_parsed) {
|
|
|
1555 |
len = fread(buf, 1, sizeof(buf), fd);
|
|
|
1556 |
if (len != sizeof(buf))
|
|
|
1557 |
return -1;
|
|
|
1558 |
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
|
|
|
1559 |
if (-1 == ret)
|
|
|
1560 |
return -1;
|
|
|
1561 |
}
|
|
|
1562 |
|
|
|
1563 |
|
|
|
1564 |
#if 1
|
|
|
1565 |
if (mp3data->totalframes > 0) {
|
|
|
1566 |
/* mpglib found a Xing VBR header and computed nsamp & totalframes */
|
|
|
1567 |
}
|
|
|
1568 |
else {
|
|
|
1569 |
mp3data->nsamp = MAX_U_32_NUM;
|
|
|
1570 |
}
|
|
|
1571 |
#else
|
|
|
1572 |
mp3data->nsamp = MAX_U_32_NUM;
|
|
|
1573 |
if (xing_header && num_frames) {
|
|
|
1574 |
mp3data->nsamp = mp3data->framesize * num_frames;
|
|
|
1575 |
}
|
|
|
1576 |
#endif
|
|
|
1577 |
|
|
|
1578 |
|
|
|
1579 |
/*
|
|
|
1580 |
fprintf(stderr,"ret = %i NEED_MORE=%i \n",ret,MP3_NEED_MORE);
|
|
|
1581 |
fprintf(stderr,"stereo = %i \n",mp.fr.stereo);
|
|
|
1582 |
fprintf(stderr,"samp = %i \n",freqs[mp.fr.sampling_frequency]);
|
|
|
1583 |
fprintf(stderr,"framesize = %i \n",framesize);
|
|
|
1584 |
fprintf(stderr,"bitrate = %i \n",mp3data->bitrate);
|
|
|
1585 |
fprintf(stderr,"num frames = %ui \n",num_frames);
|
|
|
1586 |
fprintf(stderr,"num samp = %ui \n",mp3data->nsamp);
|
|
|
1587 |
fprintf(stderr,"mode = %i \n",mp.fr.mode);
|
|
|
1588 |
*/
|
|
|
1589 |
|
|
|
1590 |
return 0;
|
|
|
1591 |
}
|
|
|
1592 |
|
|
|
1593 |
/*
|
|
|
1594 |
For lame_decode_fromfile: return code
|
|
|
1595 |
-1 error
|
|
|
1596 |
|
|
|
1597 |
n number of samples output. either 576 or 1152 depending on MP3 file.
|
|
|
1598 |
*/
|
|
|
1599 |
int
|
|
|
1600 |
lame_decode_fromfile(FILE * fd, short pcm_l[], short pcm_r[],
|
|
|
1601 |
mp3data_struct * mp3data)
|
|
|
1602 |
{
|
|
|
1603 |
int ret = 0, len;
|
|
|
1604 |
unsigned char buf[100];
|
|
|
1605 |
/* read until we get a valid output frame */
|
|
|
1606 |
while (0 == ret) {
|
|
|
1607 |
len = fread(buf, 1, 100, fd);
|
|
|
1608 |
if (len != 100)
|
|
|
1609 |
return -1;
|
|
|
1610 |
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
|
|
|
1611 |
if (ret == -1)
|
|
|
1612 |
return -1;
|
|
|
1613 |
}
|
|
|
1614 |
return ret;
|
|
|
1615 |
}
|
|
|
1616 |
#endif /* defined(HAVE_MPGLIB) */
|
|
|
1617 |
|
|
|
1618 |
/* end of get_audio.c */
|