1st problem
~~~~~~~~~~~
I have a set of WAV files without silence gaps. After converting to MP3 and
reconverting to
WAV there are gaps (20...40 ms) between the files. How can I prevent this?
2nd problem
~~~~~~~~~~~
Some of my CDs have a lot of DC offset. Other are well DC free. Examples:
x[AC] y[AC] r type x[DC] y[DC] File
26.110% 30.231% 37.382% MS-Stereo -8.418% -6.779% Giora
Feidman/track01.cdda.wav
24.437% 26.815% 62.902% MS-Stereo -7.890% -6.384% Giora
Feidman/track02.cdda.wav
28.233% 17.670% 88.636% MS-Stereo -7.832% -5.380% Giora
Feidman/track03.cdda.wav
44.400% 42.320% 77.321% MS-Stereo -10.005% -7.835% Giora
Feidman/track04.cdda.wav
27.438% 29.094% 74.308% MS-Stereo -7.997% -6.360% Giora
Feidman/track05.cdda.wav
38.190% 38.762% 73.798% MS-Stereo -8.416% -6.575% Giora
Feidman/track06.cdda.wav
33.937% 35.373% 52.249% MS-Stereo -9.008% -7.106% Giora
Feidman/track07.cdda.wav
25.220% 24.489% 83.892% MS-Stereo -7.981% -6.185% Giora
Feidman/track08.cdda.wav
23.899% 26.594% 76.254% MS-Stereo -8.135% -6.559% Giora
Feidman/track09.cdda.wav
27.257% 29.149% 73.527% MS-Stereo -8.739% -7.057% Giora
Feidman/track10.cdda.wav
44.419% 36.258% 79.641% MS-Stereo -9.816% -7.550% Giora
Feidman/track11.cdda.wav
22.294% 31.883% 89.198% MS-Stereo -8.411% -7.360% Giora
Feidman/track12.cdda.wav
37.214% 35.780% 81.549% MS-Stereo -9.638% -7.372% Giora
Feidman/track13.cdda.wav
33.170% 41.641% 87.566% MS-Stereo -9.481% -7.850% Giora
Feidman/track14.cdda.wav
33.711% 42.198% 88.124% MS-Stereo -9.345% -7.669% Giora
Feidman/track15.cdda.wav
43.713% 45.161% 73.291% MS-Stereo -9.884% -7.836% Giora
Feidman/track16.cdda.wav
36.152% 34.970% 73.292% MS-Stereo -9.071% -7.062% Giora
Feidman/track17.cdda.wav
37.865% 34.517% 71.336% MS-Stereo -9.279% -7.062% Giora
Feidman/track18.cdda.wav
34.980% 39.203% 49.169% MS-Stereo -8.949% -7.188% Giora
Feidman/track19.cdda.wav
29.621% 32.464% 81.196% MS-Stereo -8.881% -7.007% Giora
Feidman/track20.cdda.wav
70.782% 68.893% 53.126% MS-Stereo -0.006% -0.008% Jazz Lyrik
Prosa/track01.cdda.wav
39.614% 45.291% 81.413% MS-Stereo -0.006% -0.007% Jazz Lyrik
Prosa/track02.cdda.wav
71.708% 73.715% 99.879% Mono -0.010% -0.010% Jazz Lyrik
Prosa/track03.cdda.wav
31.473% 32.412% 99.844% Mono -0.007% -0.007% Jazz Lyrik
Prosa/track04.cdda.wav
71.731% 73.783% 99.925% Mono -0.005% -0.005% Jazz Lyrik
Prosa/track05.cdda.wav
36.925% 37.863% 99.791% Mono -0.007% -0.007% Jazz Lyrik
Prosa/track06.cdda.wav
28.091% 28.811% 99.835% Mono -0.006% -0.006% Jazz Lyrik
Prosa/track07.cdda.wav
51.676% 52.912% 99.507% Mono -0.008% -0.008% Jazz Lyrik
Prosa/track08.cdda.wav
48.875% 49.597% 81.123% MS-Stereo -0.007% -0.007% Jazz Lyrik
Prosa/track09.cdda.wav
55.150% 56.083% 99.873% Mono -0.001% -0.001% Jazz Lyrik
Prosa/track10.cdda.wav
47.875% 49.324% 99.749% Mono -0.007% -0.007% Jazz Lyrik
Prosa/track11.cdda.wav
85.609% 91.884% 60.015% MS-Stereo -0.005% -0.004% Jazz Lyrik
Prosa/track12.cdda.wav
31.389% 38.746% 86.307% MS-Stereo -0.007% -0.007% Jazz Lyrik
Prosa/track13.cdda.wav
76.018% 74.017% 80.323% MS-Stereo -0.008% -0.008% Jazz Lyrik
Prosa/track14.cdda.wav
41.697% 52.852% 75.873% MS-Stereo -0.006% -0.006% Jazz Lyrik
Prosa/track15.cdda.wav
68.815% 71.073% 99.863% Mono -0.009% -0.009% Jazz Lyrik
Prosa/track16.cdda.wav
42.912% 43.491% 98.428% Mono -0.006% -0.007% Jazz Lyrik
Prosa/track17.cdda.wav
101.106% 99.551% 63.043% MS-Stereo -0.014% -0.020% Jazz Lyrik
Prosa/track18.cdda.wav
57.308% 60.230% 90.311% MS-Stereo -0.007% -0.007% Jazz Lyrik
Prosa/track19.cdda.wav
87.427% 82.443% 40.229% MS-Stereo -0.007% -0.009% Jazz Lyrik
Prosa/track20.cdda.wav
46.851% 47.622% 87.358% MS-Stereo -0.007% -0.007% Jazz Lyrik
Prosa/track21.cdda.wav
69.743% 77.524% 71.732% MS-Stereo -0.007% -0.007% Jazz Lyrik
Prosa/track22.cdda.wav
Problem No.1 is that it creaks Kr#[&\/§$%&!°^z! terrible on the start and
the end of such a file.
Problem No.2 is that at the same bitrate DC decreases sound quality (is this
a feature of MP3 or a bug in Lame?)
I don't like high pass filtering in the frequency domain (don't reduces DC
at the beginning of the file, may be removes wanted frequencies), I often do
high pass filtering in the polynom domain (Legendre).
0nd order polynom filtering is DC canceling, 1st order is an additional
trend elimination, ...
A huge disadvantage is that you must read the whole file before processing,
so I use a preprocessor generating a 32-bit float AIFF file (more accuracy
without intermediante roundings, larger range needed, because of value
shifting).
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721 home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
------------ korr.c --------------------
// nasm -f elf korr_asm.asm; gcc -O2 -s -o korr korr.c korr_asm.o
// or
// gcc -O2 -s -DNONASM -o korr korr.c
#include <stdio.h>
#include <unistd.h>
#include <math.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <memory.h>
typedef signed short stereo [2];
typedef struct {
unsigned long long n;
long double x;
long double x2;
long double y;
long double y2;
long double xy;
} korr_t;
korr_t k;
#ifdef NONASM
void analyze ( const stereo* p, size_t len )
{
long double _x = 0, _x2 = 0, _y = 0, _y2 = 0, _xy = 0;
long double t1;
long double t2;
k.n += len;
for ( ; len--; p++ ) {
#if 1
_x += (t1 = (double)(*p)[0]);
_x2 += t1*t1;
_y += (t2 = (double)(*p)[1]);
_y2 += t2*t2;
_xy += t1*t2;
#else
_x += (*p)[0];
_x2 += (*p)[0] * (*p)[0];
_y += (*p)[1];
_y2 += (*p)[1] * (*p)[1];
_xy += (*p)[0] * (*p)[1];
#endif
}
k.x += _x ;
k.x2 += _x2;
k.y += _y ;
k.y2 += _y2;
k.xy += _xy;
}
#else
extern void __analyze ( const stereo* p, size_t len, korr_t* dst );
#define analyze(ptr,len) __analyze ( ptr, len, &k )
#endif
void init ( void )
{
memset ( &k, 0, sizeof(k) );
}
const char* type ( double r )
{
if ( r > +0.98 ) return "Mono";
if ( r > +0.91 ) return "?";
if ( r > +0.37 ) return "MS-Stereo";
if ( r > -0.37 ) return "Stereo";
return "???";
}
void report_init ( void )
{
printf ( " x[AC] y[AC] r type x[DC] y[DC] File\n");
}
void report ( const char* name )
{
long double scale = sqrt ( 1.e5 / (1<<29) ); // Sine Full Scale is +10 dB, 7327 =
100%
long double r;
long double sx;
long double sy;
long double x;
long double y;
// printf ("n=%Lu (7036596)\nx=%Lf (-1136345)\ny=%Lf (-783749)\nxy=%Lf
(103252784558988)\nx²=%Lf (182857029624921)\ny²=%Lf
(201045032621475)\n\n",k.n,k.x,k.y,k.xy,k.x2,k.y2);
r = (k.x2*k.n - k.x*k.x) * (k.y2*k.n - k.y*k.y);
r = r > 0. ? (k.xy*k.n - k.x*k.y) / sqrt (r) : 1.l;
sx = k.n > 1 ? sqrt ( (k.x2 - k.x*k.x/k.n) / (k.n - 1) ) : 0.l;
sy = k.n > 1 ? sqrt ( (k.y2 - k.y*k.y/k.n) / (k.n - 1) ) : 0.l;
x = k.n > 0 ? k.x/k.n : 0.l;
y = k.n > 0 ? k.y/k.n : 0.l;
printf ( "%7.3Lf%% %7.3Lf%% %7.3Lf%% %-9s %+7.3Lf%% %+7.3Lf%% %s\n",
sx * scale, sy * scale, 100. * r, type (r), x * scale, y * scale, name );
fflush ( stdout );
}
void readfile ( const char* name )
{
stereo s [2048];
size_t samples;
int fd;
init ();
if ( name ) {
if ( (fd = open ( name, O_RDONLY )) < 0 )
return;
read ( fd, s, 44 );
while ( ( samples = read (fd, s, sizeof(s)) ) > 0 )
analyze ( s, samples / sizeof (*s) );
close ( fd );
report ( name );
} else {
read ( 0, s, 44 );
while ( ( samples = read ( 0, s, sizeof(s)) ) > 0 )
analyze ( s, samples / sizeof (*s) );
report ( "<stdin>" );
}
}
int main ( int argc, char** argv )
{
report_init ();
if (argc < 2)
readfile ( NULL );
else
while ( *++argv )
readfile ( *argv );
return 0;
}
------------ korr_asm.asm ---------------------
BITS 32
SECTION .text
;
; void __analyze ( const stereo* p, size_t len, korr_t* result );
;
; esp+12 result
; esp+ 8 len
; esp+ 4 p
; esp+ 0 return address
;
GLOBAL __analyze:function
__analyze
mov eax, [esp+12]
mov ecx, [esp+ 8]
mov edx, [esp+ 4]
add dword [eax+0],ecx
adc dword [eax+4],byte 0
fldz
fldz
fldz
fldz
fldz ; Sxy Sy² Sy Sx² Sx
lbl:
fild word [edx+0] ; x Sxy Sy² Sy Sx² Sx
fld st0 ; x x Sxy Sy² Sy Sx² Sx
fmul st0,st0 ; x² x Sxy Sy² Sy Sx² Sx
faddp st5,st0 ; x Sxy Sy² Sy Sx² Sx
fadd st5,st0
fild word [edx+2] ; y x Sxy Sy² Sy Sx² Sx
fadd st4,st0
add edx,byte 4
fmul st1,st0 ; y xy Sxy Sy² Sy Sx² Sx
fmul st0,st0 ; y² xy Sxy Sy² Sy Sx² Sx
faddp st3,st0 ; xy Sxy Sy² Sy Sx² Sx
faddp st1,st0 ; Sxy Sy² Sy Sx² Sx
dec ecx
jnz lbl
fld tword [eax+56]
faddp st1,st0
fstp tword [eax+56] ; xy
fld tword [eax+44]
faddp st1,st0
fstp tword [eax+44] ; y²
fld tword [eax+32]
faddp st1,st0
fstp tword [eax+32] ; y
fld tword [eax+20]
faddp st1,st0
fstp tword [eax+20] ; x²
fld tword [eax+ 8]
faddp st1,st0
fstp tword [eax+ 8] ; x
ret
;
-----------------------------------------------------
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
