ref: bf645afaac246967b9590ae7463f243c11d97480
dir: /sys/src/cmd/audio/mp3enc/get_audio.c/
/*
* Get Audio routines source file
*
* Copyright (c) 1999 Albert L Faber
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/* $Id: get_audio.c,v 1.61 2001/03/19 21:26:05 markt Exp $ */
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <assert.h>
#ifdef HAVE_LIMITS_H
# include <limits.h>
#endif
#include <stdio.h>
#ifdef STDC_HEADERS
# include <stdlib.h>
# include <string.h>
#else
# ifndef HAVE_STRCHR
# define strchr index
# define strrchr rindex
# endif
char *strchr(), *strrchr();
# ifndef HAVE_MEMCPY
# define memcpy(d, s, n) bcopy ((s), (d), (n))
# define memmove(d, s, n) bcopy ((s), (d), (n))
# endif
#endif
#include <math.h>
#include <sys/types.h>
#include <sys/stat.h>
#include "lame.h"
#include "main.h"
#include "get_audio.h"
#include "portableio.h"
#include "timestatus.h"
#include "lametime.h"
#ifdef WITH_DMALLOC
#include <dmalloc.h>
#endif
/* global data for get_audio.c. */
int count_samples_carefully;
int pcmbitwidth;
mp3data_struct mp3input_data; /* used by Ogg and MP3 */
unsigned int num_samples_read;
FILE *musicin;
#ifdef AMIGA_MPEGA
int lame_decode_initfile(const char *fullname,
mp3data_struct * const mp3data);
#else
int lame_decode_initfile(FILE * const fd, mp3data_struct * const mp3data);
#endif
/* read mp3 file until mpglib returns one frame of PCM data */
int lame_decode_fromfile(FILE * fd, short int pcm_l[], short int pcm_r[],
mp3data_struct * mp3data);
/* and for Vorbis: */
int lame_decode_ogg_initfile( lame_global_flags* gfp,
FILE* fd,
mp3data_struct* mp3data );
int lame_decode_ogg_fromfile( lame_global_flags* gfc,
FILE* fd,
short int pcm_l[],
short int pcm_r[],
mp3data_struct* mp3data );
static int read_samples_pcm(FILE * musicin, short sample_buffer[2304],
int frame_size, int samples_to_read);
static int read_samples_mp3(lame_global_flags * gfp, FILE * musicin,
short int mpg123pcm[2][1152], int num_chan);
static int read_samples_ogg(lame_global_flags * gfp, FILE * musicin,
short int mpg123pcm[2][1152], int num_chan);
void CloseSndFile(sound_file_format input, FILE * musicin);
FILE *OpenSndFile(lame_global_flags * gfp, char *);
/* Replacement for forward fseek(,,SEEK_CUR), because fseek() fails on pipes */
static int
fskip(FILE * fp, long offset, int whence)
{
#ifndef PIPE_BUF
char buffer[4096];
#else
char buffer[PIPE_BUF];
#endif
int read;
if (0 == fseek(fp, offset, whence))
return 0;
if (whence != SEEK_CUR || offset < 0) {
fprintf(stderr,
"fskip problem: Mostly the return status of functions is not evaluated so it is more secure to pollute <stderr>.\n");
return -1;
}
while (offset > 0) {
read = offset > sizeof(buffer) ? sizeof(buffer) : offset;
if ((read = fread(buffer, 1, read, fp)) <= 0)
return -1;
offset -= read;
}
return 0;
}
FILE *
init_outfile(char *outPath, int decode)
{
FILE *outf;
/* open the output file */
if (0 == strcmp(outPath, "-"))
lame_set_stream_binary_mode(outf = stdout);
else
if ((outf = fopen(outPath, "wb+")) == NULL)
return NULL;
return outf;
}
void
init_infile(lame_global_flags * gfp, char *inPath)
{
/* open the input file */
count_samples_carefully = 0;
pcmbitwidth = 16;
musicin = OpenSndFile(gfp, inPath);
}
void
close_infile(void)
{
CloseSndFile(input_format, musicin);
}
void
SwapBytesInWords(short *ptr, int short_words)
{ /* Some speedy code */
unsigned long val;
unsigned long *p = (unsigned long *) ptr;
#ifndef lint
# if defined(CHAR_BIT)
# if CHAR_BIT != 8
# error CHAR_BIT != 8
# endif
# else
# error can not determine number of bits in a char
# endif
#endif /* lint */
assert(sizeof(short) == 2);
#if defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 4
for (; short_words >= 2; short_words -= 2, p++) {
val = *p;
*p = ((val << 8) & 0xFF00FF00) | ((val >> 8) & 0x00FF00FF);
}
ptr = (short *) p;
for (; short_words >= 1; short_words -= 1, ptr++) {
val = *ptr;
*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
}
#elif defined(SIZEOF_UNSIGNED_LONG) && SIZEOF_UNSIGNED_LONG == 8
for (; short_words >= 4; short_words -= 4, p++) {
val = *p;
*p =
((val << 8) & 0xFF00FF00FF00FF00) | ((val >> 8) &
0x00FF00FF00FF00FF);
}
ptr = (short *) p;
for (; short_words >= 1; short_words -= 1, ptr++) {
val = *ptr;
*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
}
#else
# ifdef SIZEOF_UNSIGNED_LONG
//# warning Using unoptimized SwapBytesInWords().
# endif
for (; short_words >= 1; short_words -= 1, ptr++) {
val = *ptr;
*ptr = ((val << 8) & 0xFF00) | ((val >> 8) & 0x00FF);
}
#endif
assert(short_words == 0);
}
/************************************************************************
*
* get_audio()
*
* PURPOSE: reads a frame of audio data from a file to the buffer,
* aligns the data for future processing, and separates the
* left and right channels
*
************************************************************************/
int
get_audio(lame_global_flags * const gfp, short buffer[2][1152])
{
int num_channels = lame_get_num_channels( gfp );
short insamp[2 * 1152];
int samples_read;
int framesize;
int samples_to_read;
unsigned int remaining, tmp_num_samples;
int j;
short *p;
/*
* NOTE: LAME can now handle arbritray size input data packets,
* so there is no reason to read the input data in chuncks of
* size "gfp->framesize". EXCEPT: the LAME graphical frame analyzer
* will get out of sync if we read more than framesize worth of data.
*/
samples_to_read = framesize = gfp->framesize;
assert(framesize <= 1152);
/* get num_samples */
tmp_num_samples = lame_get_num_samples( gfp );
/* if this flag has been set, then we are carefull to read
* exactly num_samples and no more. This is useful for .wav and .aiff
* files which have id3 or other tags at the end. Note that if you
* are using LIBSNDFILE, this is not necessary
*/
if (count_samples_carefully) {
remaining = tmp_num_samples - Min(tmp_num_samples, num_samples_read);
if (remaining < framesize)
samples_to_read = remaining;
}
switch (input_format) {
case sf_mp1:
case sf_mp2:
case sf_mp3:
samples_read = read_samples_mp3(gfp, musicin, buffer, num_channels);
break;
case sf_ogg:
samples_read = read_samples_ogg(gfp, musicin, buffer, num_channels);
break;
default:
samples_read =
read_samples_pcm(musicin, insamp, num_channels * framesize,
num_channels * samples_to_read);
samples_read /= num_channels;
p = insamp;
switch (num_channels) {
case 1:
for (j = 0; j < framesize; j++) {
buffer[0][j] = *p++;
buffer[1][j] = 0;
}
break;
case 2:
for (j = 0; j < framesize; j++) {
buffer[0][j] = *p++;
buffer[1][j] = *p++;
}
break;
default:
assert(0);
break;
}
}
/* if num_samples = MAX_U_32_NUM, then it is considered infinitely long.
Don't count the samples */
if ( tmp_num_samples != MAX_U_32_NUM )
num_samples_read += samples_read;
return samples_read;
}
int
read_samples_ogg(lame_global_flags * const gfp,
FILE * const musicin,
short int oggpcm[2][1152], const int stereo)
{
int out = 0;
#ifdef HAVE_VORBIS
static const char type_name[] = "Ogg Vorbis file";
out =
lame_decode_ogg_fromfile( gfp,
musicin,
oggpcm[0],
oggpcm[1],
&mp3input_data );
/*
* out < 0: error, probably EOF
* out = 0: not possible with lame_decode_fromfile() ???
* out > 0: number of output samples
*/
if (out < 0) {
memset(oggpcm, 0, sizeof(**oggpcm) * 2 * 1152);
return 0;
}
if (lame_get_num_channels( gfp ) != mp3input_data.stereo)
fprintf(stderr,
"Error: number of channels has changed in %s - not supported\n",
type_name);
if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
fprintf(stderr,
"Error: sample frequency has changed in %s - not supported\n",
type_name);
#else
out = -1; /* wanna read ogg without vorbis support? */
#endif
return out;
}
int
read_samples_mp3(lame_global_flags * const gfp,
FILE * const musicin, short int mpg123pcm[2][1152], int stereo)
{
int out;
#if defined(AMIGA_MPEGA) || defined(HAVE_MPGLIB)
static const char type_name[] = "MP3 file";
out =
lame_decode_fromfile(musicin, mpg123pcm[0], mpg123pcm[1],
&mp3input_data);
/*
* out < 0: error, probably EOF
* out = 0: not possible with lame_decode_fromfile() ???
* out > 0: number of output samples
*/
if (out < 0) {
memset(mpg123pcm, 0, sizeof(**mpg123pcm) * 2 * 1152);
return 0;
}
if ( lame_get_num_channels( gfp ) != mp3input_data.stereo )
fprintf(stderr,
"Error: number of channels has changed in %s - not supported\n",
type_name);
if ( lame_get_in_samplerate( gfp ) != mp3input_data.samplerate )
fprintf(stderr,
"Error: sample frequency has changed in %s - not supported\n",
type_name);
#else
out = -1;
#endif
return out;
}
static int
WriteWaveHeader(FILE * const fp, const int pcmbytes,
const int freq, const int channels, const int bits)
{
int bytes = (bits + 7) / 8;
/* quick and dirty, but documented */
fwrite("RIFF", 1, 4, fp); // label
Write32BitsLowHigh(fp, pcmbytes + 44 - 8); // length in bytes without header
fwrite("WAVEfmt ", 2, 4, fp); // 2 labels
Write32BitsLowHigh(fp, 2 + 2 + 4 + 4 + 2 + 2); // length of PCM format declaration area
Write16BitsLowHigh(fp, 1); // is PCM?
Write16BitsLowHigh(fp, channels); // number of channels
Write32BitsLowHigh(fp, freq); // sample frequency in [Hz]
Write32BitsLowHigh(fp, freq * channels * bytes); // bytes per second
Write16BitsLowHigh(fp, channels * bytes); // bytes per sample time
Write16BitsLowHigh(fp, bits); // bits per sample
fwrite("data", 1, 4, fp); // label
Write32BitsLowHigh(fp, pcmbytes); // length in bytes of raw PCM data
return ferror(fp) ? -1 : 0;
}
/* the simple lame decoder */
/* After calling lame_init(), lame_init_params() and
* init_infile(), call this routine to read the input MP3 file
* and output .wav data to the specified file pointer*/
/* lame_decoder will ignore the first 528 samples, since these samples
* represent the mpglib delay (and are all 0). skip = number of additional
* samples to skip, to (for example) compensate for the encoder delay */
int
lame_decoder(lame_global_flags * gfp, FILE * outf, int skip, char *inPath,
char *outPath)
{
short int Buffer[2][1152];
int iread;
double wavsize;
int i;
void (*WriteFunction) (FILE * fp, char *p, int n);
int tmp_num_channels = lame_get_num_channels( gfp );
fprintf(stderr, "\rinput: %s%s(%g kHz, %i channel%s, ",
strcmp(inPath, "-") ? inPath : "<stdin>",
strlen(inPath) > 26 ? "\n\t" : " ",
lame_get_in_samplerate( gfp ) / 1.e3,
tmp_num_channels, tmp_num_channels != 1 ? "s" : "");
switch (input_format) {
case sf_mp3:
skip += 528 + 1; /* mp3 decoder has a 528 sample delay, plus user supplied "skip" */
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "III");
break;
case sf_mp2:
skip += 240 + 1;
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "II");
break;
case sf_mp1:
skip += 240 + 1;
fprintf(stderr, "MPEG-%u%s Layer %s", 2 - gfp->version,
lame_get_out_samplerate( gfp ) < 16000 ? ".5" : "", "I");
break;
case sf_ogg:
fprintf(stderr, "Ogg Vorbis");
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
break;
case sf_raw:
fprintf(stderr, "raw PCM data");
mp3input_data.nsamp = lame_get_num_samples( gfp );
mp3input_data.framesize = 1152;
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
break;
case sf_wave:
fprintf(stderr, "Microsoft WAVE");
mp3input_data.nsamp = lame_get_num_samples( gfp );
mp3input_data.framesize = 1152;
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
break;
case sf_aiff:
fprintf(stderr, "SGI/Apple AIFF");
mp3input_data.nsamp = lame_get_num_samples( gfp );
mp3input_data.framesize = 1152;
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
break;
default:
fprintf(stderr, "unknown");
mp3input_data.nsamp = lame_get_num_samples( gfp );
mp3input_data.framesize = 1152;
skip = 0; /* other formats have no delay *//* is += 0 not better ??? */
assert(0);
break;
}
fprintf(stderr, ")\noutput: %s%s(16 bit, Microsoft WAVE)\n",
strcmp(outPath, "-") ? outPath : "<stdout>",
strlen(outPath) > 45 ? "\n\t" : " ");
if (skip > 0)
fprintf(stderr, "skipping initial %i samples (encoder+decoder delay)\n",
skip);
if ( 0 == lame_get_disable_waveheader( gfp ) )
WriteWaveHeader(outf, 0x7FFFFFFF, lame_get_in_samplerate( gfp ),
tmp_num_channels,
16);
/* unknown size, so write maximum 32 bit signed value */
wavsize = -skip;
WriteFunction = swapbytes ? WriteBytesSwapped : WriteBytes;
mp3input_data.totalframes = mp3input_data.nsamp / mp3input_data.framesize;
assert(tmp_num_channels >= 1 && tmp_num_channels <= 2);
do {
iread = get_audio(gfp, Buffer); /* read in 'iread' samples */
mp3input_data.framenum += iread / mp3input_data.framesize;
wavsize += iread;
if (!silent)
decoder_progress(gfp, &mp3input_data);
skip -= (i = skip < iread ? skip : iread); /* 'i' samples are to skip in this frame */
for (; i < iread; i++) {
if ( lame_get_disable_waveheader( gfp ) ) {
WriteFunction(outf, (char *) Buffer[0] + i, sizeof(short));
if (tmp_num_channels == 2)
WriteFunction(outf, (char *) Buffer[1] + i, sizeof(short));
}
else {
Write16BitsLowHigh(outf, Buffer[0][i]);
if (tmp_num_channels == 2)
Write16BitsLowHigh(outf, Buffer[1][i]);
}
}
} while (iread);
i = (16 / 8) * tmp_num_channels;
assert(i > 0);
if (wavsize <= 0) {
fprintf(stderr, "WAVE file contains 0 PCM samples\n");
wavsize = 0;
}
else if (wavsize > 0xFFFFFFD0 / i) {
fprintf(stderr,
"Very huge WAVE file, can't set filesize accordingly\n");
wavsize = 0xFFFFFFD0;
}
else {
wavsize *= i;
}
if ( 0 == lame_get_disable_waveheader( gfp ) )
if (!fseek(outf, 0l, SEEK_SET)) /* if outf is seekable, rewind and adjust length */
WriteWaveHeader(outf, wavsize, lame_get_in_samplerate( gfp ),
tmp_num_channels, 16);
fclose(outf);
decoder_progress_finish(gfp);
return 0;
}
#if defined(LIBSNDFILE)
#if 0 /* currently disabled */
# include "sndfile.h" // prototype for sf_get_lib_version()
void
print_sndlib_version(FILE * fp)
{
char tmp[80];
sf_get_lib_version(tmp, sizeof(tmp));
fprintf(fp,
"Input handled by %s (http://www.zip.com.au/~erikd/libsndfile/)\n",
tmp);
}
#endif
/*
** Copyright (C) 1999 Albert Faber
**
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
void
CloseSndFile(sound_file_format input, FILE * musicin)
{
SNDFILE *gs_pSndFileIn = (SNDFILE *) musicin;
if (input == sf_mp1 || input == sf_mp2 || input == sf_mp3) {
#ifndef AMIGA_MPEGA
if (fclose(musicin) != 0) {
fprintf(stderr, "Could not close audio input file\n");
exit(2);
}
#endif
}
else {
if (gs_pSndFileIn) {
if (sf_close(gs_pSndFileIn) != 0) {
fprintf(stderr, "Could not close sound file \n");
exit(2);
}
}
}
}
FILE *
OpenSndFile(lame_global_flags * gfp, char *inPath)
{
char *lpszFileName = inPath;
FILE *musicin;
SNDFILE *gs_pSndFileIn;
SF_INFO gs_wfInfo;
if (input_format == sf_mp1 ||
input_format == sf_mp2 || input_format == sf_mp3) {
#ifdef AMIGA_MPEGA
if (-1 == lame_decode_initfile(lpszFileName, &mp3input_data)) {
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
lpszFileName);
exit(1);
}
#endif
#ifdef HAVE_MPGLIB
if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
exit(1);
}
if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
lpszFileName);
exit(1);
}
#endif
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
mp3input_data.stereo );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
}
else if (input_format == sf_ogg) {
#ifdef HAVE_VORBIS
if ((musicin = fopen(lpszFileName, "rb")) == NULL) {
fprintf(stderr, "Could not find \"%s\".\n", lpszFileName);
exit(1);
}
if ( -1 == lame_decode_ogg_initfile( gfp,
musicin,
&mp3input_data ) ) {
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
lpszFileName);
exit(1);
}
#else
fprintf(stderr, "mp3enc not compiled with libvorbis support.\n");
exit(1);
#endif
}
else {
/* Try to open the sound file */
/* set some defaults incase input is raw PCM */
gs_wfInfo.seekable = (input_format != sf_raw); /* if user specified -r, set to not seekable */
gs_wfInfo.samplerate = lame_get_in_samplerate( gfp );
gs_wfInfo.pcmbitwidth = 16;
gs_wfInfo.channels = lame_get_num_channels( gfp );
#ifndef WORDS_BIGENDIAN
/* little endian */
if (swapbytes)
gs_wfInfo.format = SF_FORMAT_RAW_BE;
else
gs_wfInfo.format = SF_FORMAT_RAW_LE;
#else
if (swapbytes)
gs_wfInfo.format = SF_FORMAT_RAW_LE;
else
gs_wfInfo.format = SF_FORMAT_RAW_BE;
#endif
gs_pSndFileIn = sf_open_read(lpszFileName, &gs_wfInfo);
musicin = (SNDFILE *) gs_pSndFileIn;
/* Check result */
if (gs_pSndFileIn == NULL) {
sf_perror(gs_pSndFileIn);
fprintf(stderr, "Could not open sound file \"%s\".\n",
lpszFileName);
exit(1);
}
if ((gs_wfInfo.format == SF_FORMAT_RAW_LE) ||
(gs_wfInfo.format == SF_FORMAT_RAW_BE)) input_format = sf_raw;
#ifdef _DEBUG_SND_FILE
DEBUGF("\n\nSF_INFO structure\n");
DEBUGF("samplerate :%d\n", gs_wfInfo.samplerate);
DEBUGF("samples :%d\n", gs_wfInfo.samples);
DEBUGF("channels :%d\n", gs_wfInfo.channels);
DEBUGF("pcmbitwidth :%d\n", gs_wfInfo.pcmbitwidth);
DEBUGF("format :");
/* new formats from sbellon@sbellon.de 1/2000 */
switch (gs_wfInfo.format & SF_FORMAT_TYPEMASK) {
case SF_FORMAT_WAV:
DEBUGF("Microsoft WAV format (big endian). ");
break;
case SF_FORMAT_AIFF:
DEBUGF("Apple/SGI AIFF format (little endian). ");
break;
case SF_FORMAT_AU:
DEBUGF("Sun/NeXT AU format (big endian). ");
break;
case SF_FORMAT_AULE:
DEBUGF("DEC AU format (little endian). ");
break;
case SF_FORMAT_RAW:
DEBUGF("RAW PCM data. ");
break;
case SF_FORMAT_PAF:
DEBUGF("Ensoniq PARIS file format. ");
break;
case SF_FORMAT_SVX:
DEBUGF("Amiga IFF / SVX8 / SV16 format. ");
break;
case SF_FORMAT_NIST:
DEBUGF("Sphere NIST format. ");
break;
default:
assert(0);
break;
}
switch (gs_wfInfo.format & SF_FORMAT_SUBMASK) {
case SF_FORMAT_PCM:
DEBUGF("PCM data in 8, 16, 24 or 32 bits.");
break;
case SF_FORMAT_FLOAT:
DEBUGF("32 bit Intel x86 floats.");
break;
case SF_FORMAT_ULAW:
DEBUGF("U-Law encoded.");
break;
case SF_FORMAT_ALAW:
DEBUGF("A-Law encoded.");
break;
case SF_FORMAT_IMA_ADPCM:
DEBUGF("IMA ADPCM.");
break;
case SF_FORMAT_MS_ADPCM:
DEBUGF("Microsoft ADPCM.");
break;
case SF_FORMAT_PCM_BE:
DEBUGF("Big endian PCM data.");
break;
case SF_FORMAT_PCM_LE:
DEBUGF("Little endian PCM data.");
break;
case SF_FORMAT_PCM_S8:
DEBUGF("Signed 8 bit PCM.");
break;
case SF_FORMAT_PCM_U8:
DEBUGF("Unsigned 8 bit PCM.");
break;
case SF_FORMAT_SVX_FIB:
DEBUGF("SVX Fibonacci Delta encoding.");
break;
case SF_FORMAT_SVX_EXP:
DEBUGF("SVX Exponential Delta encoding.");
break;
default:
assert(0);
break;
}
DEBUGF("\n");
DEBUGF("pcmbitwidth :%d\n", gs_wfInfo.pcmbitwidth);
DEBUGF("sections :%d\n", gs_wfInfo.sections);
DEBUGF("seekable :\n", gs_wfInfo.seekable);
#endif
(void) lame_set_num_samples( gfp, gs_wfInfo.samples );
if( -1 == lame_set_num_channels( gfp, gs_wfInfo.channels ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
gs_wfInfo.channels );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, gs_wfInfo.samplerate );
pcmbitwidth = gs_wfInfo.pcmbitwidth;
}
if (lame_get_num_samples( gfp ) == MAX_U_32_NUM) {
/* try to figure out num_samples */
double flen = lame_get_file_size( lpszFileName );
if (flen >= 0) {
/* try file size, assume 2 bytes per sample */
if (input_format == sf_mp1 ||
input_format == sf_mp2 || input_format == sf_mp3) {
double totalseconds =
(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
unsigned long tmp_num_samples =
totalseconds * lame_get_in_samplerate( gfp );
(void) lame_set_num_samples( gfp, tmp_num_samples );
mp3input_data.nsamp = tmp_num_samples;
}
else {
lame_set_num_samples( gfp,
flen / (2 * lame_get_num_channels( gfp )) );
}
}
}
return musicin;
}
/************************************************************************
*
* read_samples()
*
* PURPOSE: reads the PCM samples from a file to the buffer
*
* SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#. Returns the number of samples read.
*
************************************************************************/
static int
read_samples_pcm(FILE * const musicin, short sample_buffer[2304],
int frame_size /* unused */ , int samples_to_read)
{
int i;
int samples_read;
samples_read =
sf_read_short((SNDFILE *) musicin, sample_buffer, samples_to_read);
switch (pcmbitwidth) {
case 8:
for (i = 0; i < samples_read; i++)
sample_buffer[i] <<= 8;
break;
case 16:
break;
default:
fprintf(stderr, "Only 8 and 16 bit input files supported \n");
exit(1);
}
return samples_read;
}
#else /* defined(LIBSNDFILE) */
/************************************************************************
************************************************************************
************************************************************************
************************************************************************
************************************************************************
************************************************************************
*
* OLD ISO/LAME routines follow. Used if you dont have LIBSNDFILE
* or for stdin/stdout support
*
************************************************************************
************************************************************************
************************************************************************
************************************************************************
************************************************************************
************************************************************************/
/************************************************************************
*
* read_samples()
*
* PURPOSE: reads the PCM samples from a file to the buffer
*
* SEMANTICS:
* Reads #samples_read# number of shorts from #musicin# filepointer
* into #sample_buffer[]#. Returns the number of samples read.
*
************************************************************************/
int
read_samples_pcm(FILE * musicin, short sample_buffer[2304], int frame_size,
int samples_to_read)
{
int samples_read;
int iswav = (input_format == sf_wave);
if (16 == pcmbitwidth) {
samples_read = fread(sample_buffer, 2, samples_to_read, musicin);
}
else if (8 == pcmbitwidth) {
char temp[2304];
int i;
samples_read = fread(temp, 1, samples_to_read, musicin);
for (i = 0; i < samples_read; ++i) {
/* note: 8bit .wav samples are unsigned */
/* map [0,255] -> [-32768,32767] */
sample_buffer[i] = ((short int)temp[i] - 128)*256 + 127;
}
}
else {
fprintf(stderr, "Only 8 and 16 bit input files supported \n");
exit(1);
}
if (ferror(musicin)) {
fprintf(stderr, "Error reading input file\n");
exit(1);
}
if (16 == pcmbitwidth) {
/* intel=littleEndian. wav files are always little endian */
#ifndef WORDS_BIGENDIAN
/* little endian */
if (!iswav)
SwapBytesInWords(sample_buffer, samples_read);
#else
/* big endian */
if (iswav)
SwapBytesInWords(sample_buffer, samples_read);
#endif
if (swapbytes)
SwapBytesInWords(sample_buffer, samples_read);
}
return samples_read;
}
/* AIFF Definitions */
#define IFF_ID_FORM 0x464f524d /* "FORM" */
#define IFF_ID_AIFF 0x41494646 /* "AIFF" */
#define IFF_ID_COMM 0x434f4d4d /* "COMM" */
#define IFF_ID_SSND 0x53534e44 /* "SSND" */
#define IFF_ID_MPEG 0x4d504547 /* "MPEG" */
#define WAV_ID_RIFF 0x52494646 /* "RIFF" */
#define WAV_ID_WAVE 0x57415645 /* "WAVE" */
#define WAV_ID_FMT 0x666d7420 /* "fmt " */
#define WAV_ID_DATA 0x64617461 /* "data" */
/*****************************************************************************
*
* Read Microsoft Wave headers
*
* By the time we get here the first 32-bits of the file have already been
* read, and we're pretty sure that we're looking at a WAV file.
*
*****************************************************************************/
static int
parse_wave_header(lame_global_flags * gfp, FILE * sf)
{
int format_tag = 0;
int channels = 0;
int block_align = 0;
int bits_per_sample = 0;
int samples_per_sec = 0;
int avg_bytes_per_sec = 0;
int is_wav = 0;
long data_length = 0, file_length, subSize = 0;
int loop_sanity = 0;
file_length = Read32BitsHighLow(sf);
if (Read32BitsHighLow(sf) != WAV_ID_WAVE)
return 0;
for (loop_sanity = 0; loop_sanity < 20; ++loop_sanity) {
int type = Read32BitsHighLow(sf);
if (type == WAV_ID_FMT) {
subSize = Read32BitsLowHigh(sf);
if (subSize < 16) {
/*DEBUGF(
"'fmt' chunk too short (only %ld bytes)!", subSize); */
return 0;
}
format_tag = Read16BitsLowHigh(sf);
subSize -= 2;
channels = Read16BitsLowHigh(sf);
subSize -= 2;
samples_per_sec = Read32BitsLowHigh(sf);
subSize -= 4;
avg_bytes_per_sec = Read32BitsLowHigh(sf);
subSize -= 4;
block_align = Read16BitsLowHigh(sf);
subSize -= 2;
bits_per_sample = Read16BitsLowHigh(sf);
subSize -= 2;
/* DEBUGF(" skipping %d bytes\n", subSize); */
if (subSize > 0) {
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
return 0;
};
}
else if (type == WAV_ID_DATA) {
subSize = Read32BitsLowHigh(sf);
data_length = subSize;
is_wav = 1;
/* We've found the audio data. Read no further! */
break;
}
else {
subSize = Read32BitsLowHigh(sf);
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
return 0;
}
}
if (format_tag != 1) {
return 0; /* oh no! non-supported format */
}
if (is_wav) {
/* make sure the header is sane */
if( -1 == lame_set_num_channels( gfp, channels ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
channels );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, samples_per_sec );
pcmbitwidth = bits_per_sample;
(void) lame_set_num_samples( gfp,
data_length / (channels * ((bits_per_sample+7) / 8)) );
}
return is_wav;
}
/************************************************************************
* aiff_check2
*
* PURPOSE: Checks AIFF header information to make sure it is valid.
* returns 0 on success, 1 on errors
************************************************************************/
int
aiff_check2(const char *file_name, IFF_AIFF * const pcm_aiff_data)
{
if (pcm_aiff_data->sampleType != IFF_ID_SSND) {
fprintf(stderr, "Sound data is not PCM in '%s'\n", file_name);
return 1;
}
if (pcm_aiff_data->sampleSize != sizeof(short) * CHAR_BIT) {
fprintf(stderr, "Sound data is not %i bits in '%s'\n",
sizeof(short) * CHAR_BIT, file_name);
return 1;
}
if (pcm_aiff_data->numChannels != 1 && pcm_aiff_data->numChannels != 2) {
fprintf(stderr, "Sound data is not mono or stereo in '%s'\n",
file_name);
return 1;
}
if (pcm_aiff_data->blkAlgn.blockSize != 0) {
fprintf(stderr, "Block size is not 0 bytes in '%s'\n", file_name);
return 1;
}
if (pcm_aiff_data->blkAlgn.offset != 0) {
fprintf(stderr, "Block offset is not 0 bytes in '%s'\n", file_name);
return 1;
}
return 0;
}
/*****************************************************************************
*
* Read Audio Interchange File Format (AIFF) headers.
*
* By the time we get here the first 32 bits of the file have already been
* read, and we're pretty sure that we're looking at an AIFF file.
*
*****************************************************************************/
static int
parse_aiff_header(lame_global_flags * gfp, FILE * sf)
{
int is_aiff = 0;
long chunkSize = 0, subSize = 0;
IFF_AIFF aiff_info;
memset(&aiff_info, 0, sizeof(aiff_info));
chunkSize = Read32BitsHighLow(sf);
if (Read32BitsHighLow(sf) != IFF_ID_AIFF)
return 0;
while (chunkSize > 0) {
int type = Read32BitsHighLow(sf);
chunkSize -= 4;
/* DEBUGF(
"found chunk type %08x '%4.4s'\n", type, (char*)&type); */
/* don't use a switch here to make it easier to use 'break' for SSND */
if (type == IFF_ID_COMM) {
subSize = Read32BitsHighLow(sf);
chunkSize -= subSize;
aiff_info.numChannels = Read16BitsHighLow(sf);
subSize -= 2;
aiff_info.numSampleFrames = Read32BitsHighLow(sf);
subSize -= 4;
aiff_info.sampleSize = Read16BitsHighLow(sf);
subSize -= 2;
aiff_info.sampleRate = ReadIeeeExtendedHighLow(sf);
subSize -= 10;
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
return 0;
}
else if (type == IFF_ID_SSND) {
subSize = Read32BitsHighLow(sf);
chunkSize -= subSize;
aiff_info.blkAlgn.offset = Read32BitsHighLow(sf);
subSize -= 4;
aiff_info.blkAlgn.blockSize = Read32BitsHighLow(sf);
subSize -= 4;
if (fskip(sf, (long) aiff_info.blkAlgn.offset, SEEK_CUR) != 0)
return 0;
aiff_info.sampleType = IFF_ID_SSND;
is_aiff = 1;
/* We've found the audio data. Read no further! */
break;
}
else {
subSize = Read32BitsHighLow(sf);
chunkSize -= subSize;
if (fskip(sf, (long) subSize, SEEK_CUR) != 0)
return 0;
}
}
/* DEBUGF("Parsed AIFF %d\n", is_aiff); */
if (is_aiff) {
/* make sure the header is sane */
if (0 != aiff_check2("name" /*???????????? */ , &aiff_info))
return 0;
if( -1 == lame_set_num_channels( gfp, aiff_info.numChannels ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
aiff_info.numChannels );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, aiff_info.sampleRate );
pcmbitwidth = aiff_info.sampleSize;
(void) lame_set_num_samples( gfp, aiff_info.numSampleFrames );
}
return is_aiff;
}
/************************************************************************
*
* parse_file_header
*
* PURPOSE: Read the header from a bytestream. Try to determine whether
* it's a WAV file or AIFF without rewinding, since rewind
* doesn't work on pipes and there's a good chance we're reading
* from stdin (otherwise we'd probably be using libsndfile).
*
* When this function returns, the file offset will be positioned at the
* beginning of the sound data.
*
************************************************************************/
void
parse_file_header(lame_global_flags * gfp, FILE * sf)
{
int type = Read32BitsHighLow(sf);
/*
DEBUGF(
"First word of input stream: %08x '%4.4s'\n", type, (char*) &type);
*/
count_samples_carefully = 0;
input_format = sf_raw;
if (type == WAV_ID_RIFF) {
/* It's probably a WAV file */
if (parse_wave_header(gfp, sf)) {
input_format = sf_wave;
count_samples_carefully = 1;
} else {
fprintf( stderr, "Warning: corrupt or unsupported WAVE format\n");
}
}
else if (type == IFF_ID_FORM) {
/* It's probably an AIFF file */
if (parse_aiff_header(gfp, sf)) {
input_format = sf_aiff;
count_samples_carefully = 1;
}
}
if (input_format == sf_raw) {
/*
** Assume it's raw PCM. Since the audio data is assumed to begin
** at byte zero, this will unfortunately require seeking.
*/
if (fseek(sf, 0L, SEEK_SET) != 0) {
/* ignore errors */
}
input_format = sf_raw;
}
}
void
CloseSndFile(sound_file_format input, FILE * musicin)
{
if (fclose(musicin) != 0) {
fprintf(stderr, "Could not close audio input file\n");
exit(2);
}
}
FILE *
OpenSndFile(lame_global_flags * gfp, char *inPath)
{
FILE *musicin;
/* set the defaults from info incase we cannot determine them from file */
lame_set_num_samples( gfp, MAX_U_32_NUM );
if (!strcmp(inPath, "-")) {
lame_set_stream_binary_mode(musicin = stdin); /* Read from standard input. */
}
else {
if ((musicin = fopen(inPath, "rb")) == NULL) {
fprintf(stderr, "Could not find \"%s\".\n", inPath);
exit(1);
}
}
if (input_format == sf_mp1 ||
input_format == sf_mp2 || input_format == sf_mp3) {
#ifdef AMIGA_MPEGA
if (-1 == lame_decode_initfile(inPath, &mp3input_data)) {
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
inPath);
exit(1);
}
#endif
#ifdef HAVE_MPGLIB
if (-1 == lame_decode_initfile(musicin, &mp3input_data)) {
fprintf(stderr, "Error reading headers in mp3 input file %s.\n",
inPath);
exit(1);
}
#endif
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
mp3input_data.stereo );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
}
else if (input_format == sf_ogg) {
#ifdef HAVE_VORBIS
if ( -1 == lame_decode_ogg_initfile( gfp,
musicin,
&mp3input_data ) ) {
fprintf(stderr, "Error reading headers in ogg input file %s.\n",
inPath);
exit(1);
}
if( -1 == lame_set_num_channels( gfp, mp3input_data.stereo ) ) {
fprintf( stderr,
"Unsupported number of channels: %ud\n",
mp3input_data.stereo );
exit( 1 );
}
(void) lame_set_in_samplerate( gfp, mp3input_data.samplerate );
(void) lame_set_num_samples( gfp, mp3input_data.nsamp );
#else
fprintf(stderr, "mp3enc not compiled with libvorbis support.\n");
exit(1);
#endif
}
else {
if (input_format != sf_raw) {
parse_file_header(gfp, musicin);
}
if (0 && input_format == sf_raw) {
fprintf(stderr, "Assuming raw pcm input file");
if (swapbytes)
fprintf(stderr, " : Forcing byte-swapping\n");
else
fprintf(stderr, "\n");
}
}
if (lame_get_num_samples( gfp ) == MAX_U_32_NUM && musicin != stdin) {
double flen = lame_get_file_size(inPath); /* try to figure out num_samples */
if (flen >= 0) {
/* try file size, assume 2 bytes per sample */
if (input_format == sf_mp1 ||
input_format == sf_mp2 || input_format == sf_mp3) {
if (mp3input_data.bitrate > 0) {
double totalseconds =
(flen * 8.0 / (1000.0 * mp3input_data.bitrate));
unsigned long tmp_num_samples =
totalseconds * lame_get_in_samplerate( gfp );
(void) lame_set_num_samples( gfp, tmp_num_samples );
mp3input_data.nsamp = tmp_num_samples;
}
}
else {
(void) lame_set_num_samples( gfp,
flen / (2 * lame_get_num_channels( gfp )) );
}
}
}
return musicin;
}
#endif /* defined(LIBSNDFILE) */
#if defined(HAVE_MPGLIB)
static int
check_aid(const unsigned char *header)
{
return 0 == strncmp(header, "AiD\1", 4);
}
/*
* Please check this and don't kill me if there's a bug
* This is a (nearly?) complete header analysis for a MPEG-1/2/2.5 Layer I, II or III
* data stream
*/
static int
is_syncword_mp123(const void *const headerptr)
{
const unsigned char *const p = headerptr;
static const char abl2[16] =
{ 0, 7, 7, 7, 0, 7, 0, 0, 0, 0, 0, 8, 8, 8, 8, 8 };
if ((p[0] & 0xFF) != 0xFF)
return 0; // first 8 bits must be '1'
if ((p[1] & 0xE0) != 0xE0)
return 0; // next 3 bits are also
if ((p[1] & 0x18) == 0x08)
return 0; // no MPEG-1, -2 or -2.5
if ((p[1] & 0x06) == 0x00)
return 0; // no Layer I, II and III
if ((p[2] & 0xF0) == 0xF0)
return 0; // bad bitrate
if ((p[2] & 0x0C) == 0x0C)
return 0; // no sample frequency with (32,44.1,48)/(1,2,4)
if ((p[1] & 0x06) == 0x04) // illegal Layer II bitrate/Channel Mode comb
if (abl2[p[2] >> 4] & (1 << (p[3] >> 6)))
return 0;
return 1;
}
static int
is_syncword_mp3(const void *const headerptr)
{
const unsigned char *const p = headerptr;
if ((p[0] & 0xFF) != 0xFF)
return 0; // first 8 bits must be '1'
if ((p[1] & 0xE0) != 0xE0)
return 0; // next 3 bits are also
if ((p[1] & 0x18) == 0x08)
return 0; // no MPEG-1, -2 or -2.5
if ((p[1] & 0x06) != 0x02)
return 0; // no Layer III (can be merged with 'next 3 bits are also' test, but don't do this, this decreases readability)
if ((p[2] & 0xF0) == 0xF0)
return 0; // bad bitrate
if ((p[2] & 0x0C) == 0x0C)
return 0; // no sample frequency with (32,44.1,48)/(1,2,4)
return 1;
}
int
lame_decode_initfile(FILE * fd, mp3data_struct * mp3data)
{
// VBRTAGDATA pTagData;
// int xing_header,len2,num_frames;
unsigned char buf[100];
int ret;
int len, aid_header;
short int pcm_l[1152], pcm_r[1152];
memset(mp3data, 0, sizeof(mp3data_struct));
lame_decode_init();
len = 4;
if (fread(&buf, 1, len, fd) != len)
return -1; /* failed */
aid_header = check_aid(buf);
if (aid_header) {
if (fread(&buf, 1, 2, fd) != 2)
return -1; /* failed */
aid_header = (unsigned char) buf[0] + 256 * (unsigned char) buf[1];
fprintf(stderr, "Album ID found. length=%i \n", aid_header);
/* skip rest of AID, except for 6 bytes we have already read */
fskip(fd, aid_header - 6, SEEK_CUR);
/* read 4 more bytes to set up buffer for MP3 header check */
len = fread(&buf, 1, 4, fd);
}
/* look for valid 4 byte MPEG header */
if (len < 4)
return -1;
while (!is_syncword_mp123(buf)) {
int i;
for (i = 0; i < len - 1; i++)
buf[i] = buf[i + 1];
if (fread(buf + len - 1, 1, 1, fd) != 1)
return -1; /* failed */
}
#if 0
/* buffer 48 bytes so we can check for Xing header */
len2 = fread(&buf[len], 1, 48 - len, fd);
if (len2 != 48 - len)
return -1;
len = 48;
/* check first 48 bytes for Xing header */
xing_header = GetVbrTag(&pTagData, (unsigned char *) buf);
if (xing_header && pTagData.headersize >= 48) {
num_frames = pTagData.frames;
fprintf(stderr,
"\rXing VBR header dectected. MP3 file has %i frames\n",
num_frames);
// skip the rest of the Xing header. LAME decoder ignores TOC data
fskip(fd, pTagData.headersize - 48, SEEK_CUR);
// buffer a few more bytes for next header check:
len = fread(buf, 1, 4, fd);
}
else {
/* we have read 48 bytes, but did not find a Xing header */
/* lets try and rewind the stream: */
if (fseek(fd, -44, SEEK_CUR) != 0) {
/* backwards fseek failed. input is probably a pipe */
/* keep 'len' unchanged */
}
else {
len -= 44;
}
}
#endif
// now parse the current buffer looking for MP3 headers
// we dont want to feed too much data to lame_decode1_headers -
// we dont want it to actually decode the first frame
// (as of 11/00: mpglib modified so that for the first frame where
// headers are parsed, no data will be decoded. So the above is
// now a moot point.
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
if (-1 == ret)
return -1;
/* repeat until we decode a valid mp3 header */
while (!mp3data->header_parsed) {
len = fread(buf, 1, sizeof(buf), fd);
if (len != sizeof(buf))
return -1;
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
if (-1 == ret)
return -1;
}
#if 1
if (mp3data->totalframes > 0) {
/* mpglib found a Xing VBR header and computed nsamp & totalframes */
}
else {
mp3data->nsamp = MAX_U_32_NUM;
}
#else
mp3data->nsamp = MAX_U_32_NUM;
if (xing_header && num_frames) {
mp3data->nsamp = mp3data->framesize * num_frames;
}
#endif
/*
fprintf(stderr,"ret = %i NEED_MORE=%i \n",ret,MP3_NEED_MORE);
fprintf(stderr,"stereo = %i \n",mp.fr.stereo);
fprintf(stderr,"samp = %i \n",freqs[mp.fr.sampling_frequency]);
fprintf(stderr,"framesize = %i \n",framesize);
fprintf(stderr,"bitrate = %i \n",mp3data->bitrate);
fprintf(stderr,"num frames = %ui \n",num_frames);
fprintf(stderr,"num samp = %ui \n",mp3data->nsamp);
fprintf(stderr,"mode = %i \n",mp.fr.mode);
*/
return 0;
}
/*
For lame_decode_fromfile: return code
-1 error
0 ok, but need more data before outputing any samples
n number of samples output. either 576 or 1152 depending on MP3 file.
*/
int
lame_decode_fromfile(FILE * fd, short pcm_l[], short pcm_r[],
mp3data_struct * mp3data)
{
int ret = 0, len;
unsigned char buf[100];
/* read until we get a valid output frame */
while (0 == ret) {
len = fread(buf, 1, 100, fd);
if (len != 100)
return -1;
ret = lame_decode1_headers(buf, len, pcm_l, pcm_r, mp3data);
if (ret == -1)
return -1;
}
return ret;
}
#endif /* defined(HAVE_MPGLIB) */
/* end of get_audio.c */