ref: bd7489f687e30fd95abd08fbc2f1dcefcfa9a492
dir: /src/audio-in.c/
/* Copyright 2000-2002, Michael Smith <msmith@xiph.org>
2010, Monty <monty@xiph.org>
AIFF/AIFC support from OggSquish, (c) 1994-1996 Monty <xiphmont@xiph.org>
(From GPL code in oggenc relicensed by permission from Monty and Msmith)
File: audio-in.c
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#if !defined(_LARGEFILE_SOURCE)
# define _LARGEFILE_SOURCE
#endif
#if !defined(_LARGEFILE64_SOURCE)
# define _LARGEFILE64_SOURCE
#endif
#if !defined(_FILE_OFFSET_BITS)
# define _FILE_OFFSET_BITS 64
#endif
#include <limits.h>
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <sys/types.h>
#include <math.h>
#include "stack_alloc.h"
#ifdef WIN32
# include <windows.h> /*GetFileType()*/
# include <io.h> /*_get_osfhandle()*/
#endif
#ifdef ENABLE_NLS
#include <libintl.h>
#define _(X) gettext(X)
#else
#define _(X) (X)
#define textdomain(X)
#define bindtextdomain(X, Y)
#endif
#ifdef gettext_noop
#define N_(X) gettext_noop(X)
#else
#define N_(X) (X)
#endif
#include <ogg/ogg.h>
#include "opusenc.h"
#include "speex_resampler.h"
#include "lpc.h"
#include "opus_header.h"
#include "flac.h"
/* Macros for handling potentially large file offsets */
#if defined WIN32 || defined _WIN32 || defined WIN64 || defined _WIN64
# define OFF_T __int64
# define FSEEK _fseeki64
# define FTELL _ftelli64
#elif defined HAVE_FSEEKO
# define OFF_T off_t
# define FSEEK fseeko
# define FTELL ftello
#else
# define OFF_T long
# define FSEEK fseek
# define FTELL ftell
#endif
/* Macros to read header data */
#define READ_U32_LE(buf) \
(((unsigned int)(buf)[3]<<24)|((buf)[2]<<16)|((buf)[1]<<8)|((buf)[0]))
#define READ_U16_LE(buf) \
(((buf)[1]<<8)|((buf)[0]&0xff))
#define READ_U32_BE(buf) \
(((unsigned int)(buf)[0]<<24)|((buf)[1]<<16)|((buf)[2]<<8)|((buf)[3]))
#define READ_U16_BE(buf) \
(((buf)[0]<<8)|((buf)[1]&0xff))
/* Define the supported formats here */
input_format formats[] = {
{wav_id, 12, wav_open, wav_close, "wav", N_("WAV file reader")},
{aiff_id, 12, aiff_open, wav_close, "aiff", N_("AIFF/AIFC file reader")},
{flac_id, 4, flac_open, flac_close, "flac", N_("FLAC file reader")},
{oggflac_id, 33, flac_open, flac_close, "ogg", N_("Ogg FLAC file reader")},
{NULL, 0, NULL, NULL, NULL, NULL}
};
input_format *open_audio_file(FILE *in, oe_enc_opt *opt)
{
int j=0;
unsigned char *buf=NULL;
int buf_size=0, buf_filled=0;
int size,ret;
while(formats[j].id_func)
{
size = formats[j].id_data_len;
if(size >= buf_size)
{
buf = realloc(buf, size);
buf_size = size;
}
if(size > buf_filled)
{
ret = fread(buf+buf_filled, 1, buf_size-buf_filled, in);
buf_filled += ret;
if(buf_filled < size)
{ /* File truncated */
j++;
continue;
}
}
if(formats[j].id_func(buf, buf_filled))
{
/* ok, we now have something that can handle the file */
if(formats[j].open_func(in, opt, buf, buf_filled)) {
free(buf);
return &formats[j];
}
}
j++;
}
free(buf);
return NULL;
}
static void sanitize_fourcc(unsigned char *buf)
{
int i;
for(i = 0; i < 4; ++i)
if(buf[i] < ' ' || buf[i] > '~')
buf[i] = '?';
}
static int seek_forward(FILE *in, ogg_int64_t length)
{
ogg_int64_t remaining = length;
while(remaining > 0)
{
/* When OFF_T is 64 bits, only one seek is needed and the comparison
* will always be false. When OFF_T is not large enough, seek LONG_MAX
* bytes at a time (the maximum offset that basic fseek() can handle).
*/
OFF_T seekstep = (OFF_T)remaining;
if (seekstep != remaining)
seekstep = LONG_MAX;
if(FSEEK(in, seekstep, SEEK_CUR))
{
/* Failed to seek; do it by reading. */
unsigned char buf[1024];
do {
size_t readstep = remaining > 1024 ? 1024 : (size_t)remaining;
readstep = fread(buf, 1, readstep, in);
if(!readstep)
return 0; /* Couldn't read more, can't read file */
remaining -= readstep;
} while (remaining);
break;
}
remaining -= seekstep;
}
return 1;
}
static int find_wav_chunk(FILE *in, char *type, unsigned int *len)
{
unsigned char buf[8];
unsigned int chunklen;
while(1)
{
if(fread(buf,1,8,in) < 8) /* Suck down a chunk specifier */
return 0; /* EOF before reaching the appropriate chunk */
chunklen = READ_U32_LE(buf+4);
if(memcmp(buf, type, 4))
{
sanitize_fourcc(buf);
fprintf(stderr, _("Skipping chunk of type \"%.4s\", length %u\n"),
buf, chunklen);
if(!seek_forward(in, (ogg_int64_t)chunklen + (chunklen & 1)))
return 0;
}
else
{
*len = chunklen;
return 1;
}
}
}
static int find_aiff_chunk(FILE *in, char *type, unsigned int *len)
{
unsigned char buf[8];
unsigned int chunklen;
int restarted = 0;
while(1)
{
if(fread(buf,1,8,in) < 8)
{
if(!restarted) {
/* Handle out of order chunks by seeking back to the start
* to retry */
restarted = 1;
if(!FSEEK(in, 12, SEEK_SET))
continue;
}
return 0;
}
chunklen = READ_U32_BE(buf+4);
if(memcmp(buf,type,4))
{
if(!seek_forward(in, (ogg_int64_t)chunklen + (chunklen & 1)))
return 0;
}
else
{
*len = chunklen;
return 1;
}
}
}
/* Read chunk of size *len and advance the file position to the next chunk.
* Returns 0 on EOF or read error. Otherwise *len is updated with the number
* of bytes placed in the buffer (the lesser of the chunk size and buffer
* size) and 1 is returned.
*/
static int read_chunk(FILE *in, unsigned char *buf, unsigned int bufsize,
unsigned int *len)
{
unsigned int chunklen = *len;
unsigned int readlen = chunklen > bufsize ? bufsize : chunklen;
if(fread(buf, 1, readlen, in) != readlen)
return 0;
if(!seek_forward(in, (ogg_int64_t)(chunklen - readlen) + (chunklen & 1)))
return 0;
*len = readlen;
return 1;
}
double read_IEEE80(unsigned char *buf)
{
int s=buf[0]&0xff;
int e=((buf[0]&0x7f)<<8)|(buf[1]&0xff);
double f=((unsigned long)(buf[2]&0xff)<<24)|
((buf[3]&0xff)<<16)|
((buf[4]&0xff)<<8) |
(buf[5]&0xff);
if(e==32767)
{
if(buf[2]&0x80)
return HUGE_VAL; /* Really NaN, but this won't happen in reality */
else
{
if(s)
return -HUGE_VAL;
else
return HUGE_VAL;
}
}
f=ldexp(f,32);
f+= ((buf[6]&0xff)<<24)|
((buf[7]&0xff)<<16)|
((buf[8]&0xff)<<8) |
(buf[9]&0xff);
return ldexp(f, e-16446);
}
/* AIFF/AIFC support adapted from the old OggSQUISH application */
int aiff_id(unsigned char *buf, int len)
{
if(len<12) return 0; /* Truncated file, probably */
if(memcmp(buf, "FORM", 4))
return 0;
if(memcmp(buf+8, "AIF",3))
return 0;
if(buf[11]!='C' && buf[11]!='F')
return 0;
return 1;
}
static int aiff_permute_matrix[6][6] =
{
{0}, /* 1.0 mono */
{0,1}, /* 2.0 stereo */
{0,2,1}, /* 3.0 channel ('wide') stereo */
{0,1,2,3}, /* 4.0 discrete quadraphonic (WARN) */
{0,2,1,3,4}, /* 5.0 surround (WARN) */
{0,1,2,3,4,5}, /* 5.1 surround (WARN)*/
};
int aiff_open(FILE *in, oe_enc_opt *opt, unsigned char *buf, int buflen)
{
int aifc; /* AIFC or AIFF? */
unsigned int len;
unsigned char buffer[22];
unsigned char buf2[8];
int bigendian = 1;
aiff_fmt format;
aifffile *aiff;
int i;
(void)buflen;/*unused*/
if(buf[11]=='C')
aifc=1;
else
aifc=0;
if(!find_aiff_chunk(in, "COMM", &len))
{
fprintf(stderr, _("Warning: No common chunk found in AIFF file\n"));
return 0; /* EOF before COMM chunk */
}
if(len < 18 || !read_chunk(in, buffer, sizeof(buffer), &len))
{
fprintf(stderr, _("ERROR: Incomplete common chunk in AIFF header\n"));
return 0;
}
format.channels = (short)READ_U16_BE(buffer);
format.totalframes = READ_U32_BE(buffer+2);
format.samplesize = (short)READ_U16_BE(buffer+6);
format.rate = (int)read_IEEE80(buffer+8);
if(format.channels <= 0)
{
fprintf(stderr, _("ERROR: Invalid channel count in AIFF header\n"));
return 0;
}
if(aifc)
{
if(len < 22)
{
fprintf(stderr, _("Warning: AIFF-C header truncated.\n"));
return 0;
}
if(!memcmp(buffer+18, "NONE", 4))
{
bigendian = 1;
}
else if(!memcmp(buffer+18, "sowt", 4))
{
bigendian = 0;
}
else
{
sanitize_fourcc(buffer+18);
fprintf(stderr, _("ERROR: Can't handle compressed AIFF-C \"%.4s\"\n"),
buffer+18);
return 0; /* Compressed. Can't handle */
}
}
if(!find_aiff_chunk(in, "SSND", &len))
{
fprintf(stderr, _("Warning: No SSND chunk found in AIFF file\n"));
return 0; /* No SSND chunk -> no actual audio */
}
if(len < 8)
{
fprintf(stderr, _("Warning: Corrupted SSND chunk in AIFF header\n"));
return 0;
}
if(fread(buf2,1,8, in) < 8)
{
fprintf(stderr, _("Warning: Unexpected EOF reading AIFF header\n"));
return 0;
}
format.offset = READ_U32_BE(buf2);
format.blocksize = READ_U32_BE(buf2+4);
if( format.blocksize == 0 &&
(format.samplesize == 16 || format.samplesize == 8))
{
/* From here on, this is very similar to the wav code. Oh well. */
opt->rate = format.rate;
opt->channels = format.channels;
opt->samplesize = format.samplesize;
opt->read_samples = wav_read; /* Similar enough, so we use the same */
opt->total_samples_per_channel = format.totalframes;
aiff = malloc(sizeof(aifffile));
aiff->f = in;
aiff->samplesread = 0;
aiff->channels = format.channels;
aiff->samplesize = format.samplesize;
aiff->totalsamples = format.totalframes;
aiff->bigendian = bigendian;
aiff->unsigned8bit = 0;
if(aiff->channels>3)
fprintf(stderr, _("WARNING: AIFF[-C] files with more than three channels use\n"
"speaker locations incompatible with Vorbis surround definitions.\n"
"Not performing channel location mapping.\n"));
opt->readdata = (void *)aiff;
aiff->channel_permute = malloc(aiff->channels * sizeof(int));
if (aiff->channels <= 6)
/* Where we know the mappings, use them. */
memcpy(aiff->channel_permute, aiff_permute_matrix[aiff->channels-1],
sizeof(int) * aiff->channels);
else
/* Use a default 1-1 mapping */
for (i=0; i < aiff->channels; i++)
aiff->channel_permute[i] = i;
seek_forward(in, format.offset); /* Swallow some data */
return 1;
}
else
{
fprintf(stderr, _("ERROR: Unsupported AIFF/AIFC file.\n"
"Must be 8 or 16 bit PCM.\n"));
return 0;
}
}
int wav_id(unsigned char *buf, int len)
{
if(len<12) return 0; /* Something screwed up */
if(memcmp(buf, "RIFF", 4))
return 0; /* Not wave */
/*flen = READ_U32_LE(buf+4);*/ /* We don't use this */
if(memcmp(buf+8, "WAVE",4))
return 0; /* RIFF, but not wave */
return 1;
}
int wav_open(FILE *in, oe_enc_opt *opt, unsigned char *oldbuf, int buflen)
{
unsigned char buf[40];
unsigned int len;
int samplesize;
int validbits;
wav_fmt format;
wavfile *wav;
int i;
(void)buflen;/*unused*/
(void)oldbuf;/*unused*/
/* Ok. At this point, we know we have a WAV file. Now we have to detect
* whether we support the subtype, and we have to find the actual data
* We don't (for the wav reader) need to use the buffer we used to id this
* as a wav file (oldbuf)
*/
if(!find_wav_chunk(in, "fmt ", &len))
{
fprintf(stderr, _("ERROR: No format chunk found in WAV file\n"));
return 0;
}
if(len < 16)
{
fprintf(stderr, _("Warning: Unrecognised format chunk in WAV header\n"));
return 0; /* Weird format chunk */
}
/* A common error is to have a format chunk that is not 16, 18 or
* 40 bytes in size. This is incorrect, but not fatal, so we only
* warn about it instead of refusing to work with the file.
* Please, if you have a program that's creating format chunks of
* sizes other than 16 or 18 bytes in size, report a bug to the
* author.
*/
if(len!=16 && len!=18 && len!=40)
fprintf(stderr,
_("Warning: INVALID format chunk in wav header.\n"
" Trying to read anyway (may not work)...\n"));
if(!read_chunk(in, buf, sizeof(buf), &len))
{
fprintf(stderr, _("ERROR: Incomplete format chunk in WAV header\n"));
return 0;
}
format.format = READ_U16_LE(buf);
format.channels = READ_U16_LE(buf+2);
format.samplerate = READ_U32_LE(buf+4);
format.bytespersec = READ_U32_LE(buf+8);
format.align = READ_U16_LE(buf+12);
format.samplesize = READ_U16_LE(buf+14);
if(format.channels == 0)
{
fprintf(stderr, _("ERROR: Zero channels in WAV header\n"));
return 0;
}
if(format.format == 0xfffe) /* WAVE_FORMAT_EXTENSIBLE */
{
if(len<40)
{
fprintf(stderr, _("ERROR: Extended WAV format header invalid (too small)\n"));
return 0;
}
validbits = READ_U16_LE(buf+18);
if(validbits < 1 || validbits > format.samplesize)
validbits = format.samplesize;
format.mask = READ_U32_LE(buf+20);
/* warn the user if the format mask is not a supported/expected type */
switch(format.mask){
case 1539: /* 4.0 using side surround instead of back */
fprintf(stderr, _("WARNING: WAV file uses side surround instead of rear for quadraphonic;\n"
"remapping side speakers to rear in encoding.\n"));
break;
case 1551: /* 5.1 using side instead of rear */
fprintf(stderr, _("WARNING: WAV file uses side surround instead of rear for 5.1;\n"
"remapping side speakers to rear in encoding.\n"));
break;
case 319: /* 6.1 using rear instead of side */
fprintf(stderr, _("WARNING: WAV file uses rear surround instead of side for 6.1;\n"
"remapping rear speakers to side in encoding.\n"));
break;
case 255: /* 7.1 'Widescreen' */
fprintf(stderr, _("WARNING: WAV file is a 7.1 'Widescreen' channel mapping;\n"
"remapping speakers to Vorbis 7.1 format.\n"));
break;
case 0: /* default/undeclared */
case 1: /* mono */
case 3: /* stereo */
case 51: /* quad */
case 55: /* 5.0 */
case 63: /* 5.1 */
case 1807: /* 6.1 */
case 1599: /* 7.1 */
break;
default:
fprintf(stderr, _("WARNING: Unknown WAV surround channel mask: %u\n"
"Blindly mapping speakers using default SMPTE/ITU ordering.\n"),
format.mask);
break;
}
format.format = READ_U16_LE(buf+24);
}
else
{
validbits = format.samplesize;
}
if(format.format == 1)
{
samplesize = format.samplesize/8;
opt->read_samples = wav_read;
}
else if(format.format == 3)
{
validbits = 24;
samplesize = 4;
opt->read_samples = wav_ieee_read;
}
else
{
fprintf(stderr, _("ERROR: Unsupported WAV file type.\n"
"Must be standard PCM or type 3 floating point PCM.\n"));
return 0;
}
if(!find_wav_chunk(in, "data", &len))
{
fprintf(stderr, _("ERROR: No data chunk found in WAV file\n"));
return 0;
}
if(format.align != format.channels * samplesize) {
/* This is incorrect according to the spec. Warn loudly, then ignore
* this value.
*/
fprintf(stderr, _("Warning: WAV 'block alignment' value is incorrect, "
"ignoring.\n"
"The software that created this file is incorrect.\n"));
}
if(format.samplesize == samplesize*8 &&
(format.samplesize == 24 || format.samplesize == 16 ||
format.samplesize == 8 ||
(format.samplesize == 32 && format.format == 3)))
{
/* OK, good - we have the one supported format,
now we want to find the size of the file */
opt->rate = format.samplerate;
opt->channels = format.channels;
opt->samplesize = validbits;
opt->total_samples_per_channel = 0;
wav = malloc(sizeof(wavfile));
wav->f = in;
wav->samplesread = 0;
wav->bigendian = 0;
wav->unsigned8bit = format.samplesize == 8;
wav->channels = format.channels; /* This is in several places. The price
of trying to abstract stuff. */
wav->samplesize = format.samplesize;
wav->totalsamples = 0;
if(len>(format.channels*samplesize*4U) && len<((1U<<31)-65536) && opt->ignorelength!=1)
{
/* Chunk length is plausible. Limit the audio data read to
this length so that we do not misinterpret any additional
chunks after this as audio. Also use this length to report
percent progress. */
wav->totalsamples = opt->total_samples_per_channel =
len/(format.channels*samplesize);
}
#ifdef WIN32
/*On Mingw/Win32 fseek() returns zero on pipes.*/
else if (opt->ignorelength==1 || ((GetFileType((HANDLE)_get_osfhandle(fileno(in)))&~FILE_TYPE_REMOTE)!=FILE_TYPE_DISK))
#else
else if (opt->ignorelength==1)
#endif
{
/* Assume audio data continues until EOF.
No percent progress will be reported. */
}
else
{
/* Assume audio data continues until EOF.
If the stream is seekable and the current file size can
be determined, use that to estimate percent progress
(opt->total_samples_per_channel), but not to limit the
number of samples read (wav->totalsamples). Continue to
read until EOF even if the file grows while reading. */
OFF_T pos[2];
pos[0] = FTELL(in);
if(pos[0] >= 0 && !FSEEK(in, 0, SEEK_END))
{
pos[1] = FTELL(in);
FSEEK(in, pos[0], SEEK_SET);
if(pos[1] > pos[0])
opt->total_samples_per_channel = (pos[1]-pos[0])/(format.channels*samplesize);
}
}
opt->readdata = (void *)wav;
wav->channel_permute = malloc(wav->channels * sizeof(int));
if (wav->channels <= 8)
/* Where we know the mappings, use them. */
memcpy(wav->channel_permute, wav_permute_matrix[wav->channels-1],
sizeof(int) * wav->channels);
else
/* Use a default 1-1 mapping */
for (i=0; i < wav->channels; i++)
wav->channel_permute[i] = i;
return 1;
}
else
{
fprintf(stderr, _("ERROR: Unsupported WAV sample size.\n"
"Must be 8, 16, or 24 bit PCM or 32 bit floating point PCM.\n"));
return 0;
}
}
long wav_read(void *in, float *buffer, int samples)
{
wavfile *f = (wavfile *)in;
int sampbyte = f->samplesize / 8;
signed char *buf = alloca(samples*sampbyte*f->channels);
long bytes_read = fread(buf, 1, samples*sampbyte*f->channels, f->f);
int i,j;
opus_int64 realsamples;
int *ch_permute = f->channel_permute;
if(f->totalsamples > 0 && f->samplesread +
bytes_read/(sampbyte*f->channels) > f->totalsamples) {
bytes_read = sampbyte*f->channels*(f->totalsamples - f->samplesread);
}
realsamples = bytes_read/(sampbyte*f->channels);
f->samplesread += realsamples;
if(f->samplesize==8)
{
if(f->unsigned8bit)
{
unsigned char *bufu = (unsigned char *)buf;
for(i = 0; i < realsamples; i++)
{
for(j=0; j < f->channels; j++)
{
buffer[i*f->channels+j]=((int)(bufu[i*f->channels + ch_permute[j]])-128)/128.0f;
}
}
}
else
{
for(i = 0; i < realsamples; i++)
{
for(j=0; j < f->channels; j++)
{
buffer[i*f->channels+j]=buf[i*f->channels + ch_permute[j]]/128.0f;
}
}
}
}
else if(f->samplesize==16)
{
if(!f->bigendian)
{
for(i = 0; i < realsamples; i++)
{
for(j=0; j < f->channels; j++)
{
buffer[i*f->channels+j] = ((buf[i*2*f->channels + 2*ch_permute[j] + 1]<<8) |
(buf[i*2*f->channels + 2*ch_permute[j]] & 0xff))/32768.0f;
}
}
}
else
{
for(i = 0; i < realsamples; i++)
{
for(j=0; j < f->channels; j++)
{
buffer[i*f->channels+j]=((buf[i*2*f->channels + 2*ch_permute[j]]<<8) |
(buf[i*2*f->channels + 2*ch_permute[j] + 1] & 0xff))/32768.0f;
}
}
}
}
else if(f->samplesize==24)
{
if(!f->bigendian) {
for(i = 0; i < realsamples; i++)
{
for(j=0; j < f->channels; j++)
{
buffer[i*f->channels+j] = ((buf[i*3*f->channels + 3*ch_permute[j] + 2] << 16) |
(((unsigned char *)buf)[i*3*f->channels + 3*ch_permute[j] + 1] << 8) |
(((unsigned char *)buf)[i*3*f->channels + 3*ch_permute[j]] & 0xff))
/ 8388608.0f;
}
}
}
else {
fprintf(stderr, _("Big endian 24 bit PCM data is not currently "
"supported, aborting.\n"));
return 0;
}
}
else {
fprintf(stderr, _("Internal error: attempt to read unsupported "
"bitdepth %d\n"), f->samplesize);
return 0;
}
return realsamples;
}
long wav_ieee_read(void *in, float *buffer, int samples)
{
wavfile *f = (wavfile *)in;
float *buf = alloca(samples*4*f->channels); /* de-interleave buffer */
long bytes_read = fread(buf,1,samples*4*f->channels, f->f);
int i,j;
opus_int64 realsamples;
if(f->totalsamples > 0 && f->samplesread +
bytes_read/(4*f->channels) > f->totalsamples)
bytes_read = 4*f->channels*(f->totalsamples - f->samplesread);
realsamples = bytes_read/(4*f->channels);
f->samplesread += realsamples;
for(i=0; i < realsamples; i++)
for(j=0; j < f->channels; j++)
buffer[i*f->channels+j] = buf[i*f->channels + f->channel_permute[j]];
return realsamples;
}
void wav_close(void *info)
{
wavfile *f = (wavfile *)info;
free(f->channel_permute);
free(f);
}
int raw_open(FILE *in, oe_enc_opt *opt, unsigned char *buf, int buflen)
{
wavfile *wav = malloc(sizeof(wavfile));
int i;
(void)buf;/*unused*/
(void)buflen;/*unused*/
wav->f = in;
wav->samplesread = 0;
wav->bigendian = opt->endianness;
wav->unsigned8bit = opt->samplesize == 8;
wav->channels = opt->channels;
wav->samplesize = opt->samplesize;
wav->totalsamples = 0;
wav->channel_permute = malloc(wav->channels * sizeof(int));
for (i=0; i < wav->channels; i++)
wav->channel_permute[i] = i;
opt->read_samples = wav_read;
opt->readdata = (void *)wav;
opt->total_samples_per_channel = 0; /* raw mode, don't bother */
return 1;
}
typedef struct {
audio_read_func real_reader;
void *real_readdata;
int channels;
float scale_factor;
} scaler;
static long read_scaler(void *data, float *buffer, int samples) {
scaler *d = data;
long in_samples = d->real_reader(d->real_readdata, buffer, samples);
int i;
for(i=0; i < d->channels*in_samples; i++) {
buffer[i] *= d->scale_factor;
}
return in_samples;
}
void setup_scaler(oe_enc_opt *opt, float scale) {
scaler *d = calloc(1, sizeof(scaler));
d->real_reader = opt->read_samples;
d->real_readdata = opt->readdata;
opt->read_samples = read_scaler;
opt->readdata = d;
d->channels = opt->channels;
d->scale_factor = scale;
}
typedef struct {
audio_read_func real_reader;
void *real_readdata;
ogg_int64_t *original_samples;
int channels;
int lpc_ptr;
int *extra_samples;
float *lpc_out;
} padder;
/* Read audio data, appending padding to make up any gap
* between the available and requested number of samples
* with LPC-predicted data to minimize the pertubation of
* the valid data that falls in the same frame.
*/
static long read_padder(void *data, float *buffer, int samples) {
padder *d = data;
long in_samples = d->real_reader(d->real_readdata, buffer, samples);
int i, extra=0;
const int lpc_order=32;
if(d->original_samples)*d->original_samples+=in_samples;
if(in_samples<samples){
if(d->lpc_ptr<0){
d->lpc_out=calloc(d->channels * *d->extra_samples, sizeof(*d->lpc_out));
if(in_samples>lpc_order*2){
float *lpc=alloca(lpc_order*sizeof(*lpc));
for(i=0;i<d->channels;i++){
vorbis_lpc_from_data(buffer+i,lpc,in_samples,lpc_order,d->channels);
vorbis_lpc_predict(lpc,buffer+i+(in_samples-lpc_order)*d->channels,
lpc_order,d->lpc_out+i,*d->extra_samples,d->channels);
}
}
d->lpc_ptr=0;
}
extra=samples-in_samples;
if(extra>*d->extra_samples)extra=*d->extra_samples;
*d->extra_samples-=extra;
}
memcpy(buffer+in_samples*d->channels,d->lpc_out+d->lpc_ptr*d->channels,extra*d->channels*sizeof(*buffer));
d->lpc_ptr+=extra;
return in_samples+extra;
}
void setup_padder(oe_enc_opt *opt,ogg_int64_t *original_samples) {
padder *d = calloc(1, sizeof(padder));
d->real_reader = opt->read_samples;
d->real_readdata = opt->readdata;
opt->read_samples = read_padder;
opt->readdata = d;
d->channels = opt->channels;
d->extra_samples = &opt->extraout;
d->original_samples=original_samples;
d->lpc_ptr = -1;
d->lpc_out = NULL;
}
void clear_padder(oe_enc_opt *opt) {
padder *d = opt->readdata;
opt->read_samples = d->real_reader;
opt->readdata = d->real_readdata;
if(d->lpc_out)free(d->lpc_out);
free(d);
}
typedef struct {
SpeexResamplerState *resampler;
audio_read_func real_reader;
void *real_readdata;
float *bufs;
int channels;
int bufpos;
int bufsize;
int done;
} resampler;
static long read_resampled(void *d, float *buffer, int samples)
{
resampler *rs = d;
int out_samples=0;
float *pcmbuf;
int *inbuf;
pcmbuf=rs->bufs;
inbuf=&rs->bufpos;
while(out_samples<samples){
int i;
int reading, ret;
unsigned in_len, out_len;
out_len=samples-out_samples;
reading=rs->bufsize-*inbuf;
if(reading>1024)reading=1024;
ret=rs->real_reader(rs->real_readdata, pcmbuf+*inbuf*rs->channels, reading);
*inbuf+=ret;
in_len=*inbuf;
speex_resampler_process_interleaved_float(rs->resampler, pcmbuf, &in_len, buffer+out_samples*rs->channels, &out_len);
out_samples+=out_len;
if(ret==0&&in_len==0){
for(i=out_samples*rs->channels;i<samples*rs->channels;i++)buffer[i]=0;
return out_samples;
}
for(i=0;i<rs->channels*(*inbuf-(long int)in_len);i++)pcmbuf[i]=pcmbuf[i+rs->channels*in_len];
*inbuf-=in_len;
}
return out_samples;
}
int setup_resample(oe_enc_opt *opt, int complexity, long outfreq) {
resampler *rs = calloc(1, sizeof(resampler));
int err;
rs->bufsize = 5760*2; /* Have at least two output frames worth, just in case of ugly ratios */
rs->bufpos = 0;
rs->real_reader = opt->read_samples;
rs->real_readdata = opt->readdata;
rs->channels = opt->channels;
rs->done = 0;
rs->resampler = speex_resampler_init(rs->channels, opt->rate, outfreq, complexity, &err);
if(err!=0)fprintf(stderr, _("resampler error: %s\n"), speex_resampler_strerror(err));
opt->skip+=speex_resampler_get_output_latency(rs->resampler);
rs->bufs = malloc(sizeof(float) * rs->bufsize * opt->channels);
opt->read_samples = read_resampled;
opt->readdata = rs;
if(opt->total_samples_per_channel)
opt->total_samples_per_channel = (opus_int64)
((double)opt->total_samples_per_channel * ((double)outfreq/(double)opt->rate));
opt->rate = outfreq;
return 0;
}
void clear_resample(oe_enc_opt *opt) {
resampler *rs = opt->readdata;
opt->read_samples = rs->real_reader;
opt->readdata = rs->real_readdata;
speex_resampler_destroy(rs->resampler);
free(rs->bufs);
free(rs);
}
typedef struct {
audio_read_func real_reader;
void *real_readdata;
float *bufs;
float *matrix;
int in_channels;
int out_channels;
} downmix;
static long read_downmix(void *data, float *buffer, int samples)
{
downmix *d = data;
long in_samples = d->real_reader(d->real_readdata, d->bufs, samples);
int i,j,k,in_ch,out_ch;
in_ch=d->in_channels;
out_ch=d->out_channels;
for(i=0;i<in_samples;i++){
for(j=0;j<out_ch;j++){
float *samp;
samp=&buffer[i*out_ch+j];
*samp=0;
for(k=0;k<in_ch;k++){
*samp+=d->bufs[i*in_ch+k]*d->matrix[in_ch*j+k];
}
}
}
return in_samples;
}
int setup_downmix(oe_enc_opt *opt, int out_channels) {
static const float stupid_matrix[7][8][2]={
/*2*/ {{1,0},{0,1}},
/*3*/ {{1,0},{0.7071f,0.7071f},{0,1}},
/*4*/ {{1,0},{0,1},{0.866f,0.5f},{0.5f,0.866f}},
/*5*/ {{1,0},{0.7071f,0.7071f},{0,1},{0.866f,0.5f},{0.5f,0.866f}},
/*6*/ {{1,0},{0.7071f,0.7071f},{0,1},{0.866f,0.5f},{0.5f,0.866f},{0.7071f,0.7071f}},
/*7*/ {{1,0},{0.7071f,0.7071f},{0,1},{0.866f,0.5f},{0.5f,0.866f},{0.6123f,0.6123f},{0.7071f,0.7071f}},
/*8*/ {{1,0},{0.7071f,0.7071f},{0,1},{0.866f,0.5f},{0.5f,0.866f},{0.866f,0.5f},{0.5f,0.866f},{0.7071f,0.7071f}},
};
float sum;
downmix *d;
int i,j;
if(opt->channels<=out_channels || out_channels>2 || opt->channels<=0 || out_channels<=0) {
fprintf(stderr, _("Downmix must actually downmix and only knows mono/stereo out.\n"));
return 0;
}
if(out_channels==2 && opt->channels>8) {
fprintf(stderr, _("Downmix only knows how to mix >8ch to mono.\n"));
return 0;
}
d = calloc(1, sizeof(downmix));
d->bufs = malloc(sizeof(float)*opt->channels*4096);
d->matrix = malloc(sizeof(float)*opt->channels*out_channels);
d->real_reader = opt->read_samples;
d->real_readdata = opt->readdata;
d->in_channels=opt->channels;
d->out_channels=out_channels;
if(out_channels==1&&d->in_channels>8){
for(i=0;i<d->in_channels;i++)d->matrix[i]=1.0f/d->in_channels;
}else if(out_channels==2){
for(j=0;j<d->out_channels;j++)
for(i=0;i<d->in_channels;i++)d->matrix[d->in_channels*j+i]=
stupid_matrix[opt->channels-2][i][j];
}else{
for(i=0;i<d->in_channels;i++)d->matrix[i]=
(stupid_matrix[opt->channels-2][i][0])+
(stupid_matrix[opt->channels-2][i][1]);
}
sum=0;
for(i=0;i<d->in_channels*d->out_channels;i++)sum+=d->matrix[i];
sum=(float)out_channels/sum;
for(i=0;i<d->in_channels*d->out_channels;i++)d->matrix[i]*=sum;
opt->read_samples = read_downmix;
opt->readdata = d;
opt->channels = out_channels;
return out_channels;
}
void clear_downmix(oe_enc_opt *opt) {
downmix *d = opt->readdata;
opt->read_samples = d->real_reader;
opt->readdata = d->real_readdata;
opt->channels = d->in_channels; /* other things in cleanup rely on this */
free(d->bufs);
free(d->matrix);
free(d);
}