something around 612 bits/s. This is less 0.5% for 128 kbps, even
less for higher bit rates. It becomes interesting for bitrates in the range
from 8...24 kbps.
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inpossible.
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::
::
:: That name was changed because one make system (MSDOS?) interpreted
:: the '-' in quantize-pvt.c as a compiler option.
::
MSDOS can't store a name like "quantize-pvt.c", you got at most:
"quantize.c" or "quanti~1.c".
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ould never match fully.
* Prototype functions with ellipses with:
int printf ();
* never cast mallocs/callocs().
* use identically identifier for name spaces which are separated in C
and merged in C++ (struct, unions, typedefs).
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On Wed, Oct 04, 2000 at 09:42:51PM +0100, Sigbjørn Skjæret wrote:
> Just thought I'd say my thoughts on the different parameter setting function
> proposals we've had so far...
>
> Individual functions for each parameter:
>
> Pros:
> - None. ;)
>
> Cons:
> - Litters the API with "thousands"
.24 32
18 105.33 60.13 45
Binaural, Sennheiser HD 560, diffuse field.
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fft routines are wrongly prototyped:
void fft_long ( lame_internal_flags* gfc, FLOAT x_real [BLKSIZE ], int, sample_t
** );
void fft_short ( lame_internal_flags* gfc, FLOAT x_real [3] [BLKSIZE_s], int, sample_t
** );
void init_fft ( lame_internal_flags* gfc );
Right is:
void fft_long
nt_fast128_t int_fast16_t int_fast32_t
int_fast64_t int_fast128_t uint_fast8_t uint_fast16_t uint_fast32_t
uint_fast64_t uint_fast128_t bool ...
Also note: float_t and double_t
And there are also a lot of Unix types out there (having currently fixed sizes):
pid_t, dev_t, ino_t, fsid_t, gid_t, uid_t, mode_
200E.
* unlikely in music also means that it is not (urgent) necessary
to fic this bug in WinAmp to play correctly older lame encoded
files
Does it decrease performance to make this command line configurable?
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::
:: 3. Creating a dozen or so odd types, all synonymous with 'int'
:: is a great idea. But I saw a few 'int's left. I think
:: we need to replace these with more new types.
::
When you were a physicist, you were a cgs advocate and a SI enemy, right?
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s by 5:3 = 1.500015...:1, not 3:2.
c) Both should be possible.
I only need the input: a), b) or c).
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10., 0.1*PASS_BAND_RIPPLE ) - 1.;
for ( i = 1; i <= 2; i += step (i) ) {
poly = cheychev_polynom ( (double) STOP_BAND / i, ORDER );
damp = 10. * log10 ( 1. + ripple * poly * poly )
+ ( ORDER & 1 ? 0. : -PASS_BAND_RIPPLE );
printf ( "%5u Hz%9
On Sun, Oct 01, 2000 at 09:10:19PM +0200, Robert Hegemann wrote:
>
> So for backward compatibility we should make a wrapper library
> with the old interface (as much as possible) and mark this as
> old and outdated, to give clients the possibility for smooth
> migration t
Coeff [j] /= sum;
}
with:
long double sinc ( long double x )
{
if ( x == 0. ) return 1.;
if ( x < 0. ) x = -x;
return sinpi ( x ) / ( M_PIl * x );
}
long double window ( long double x )
{
if ( fabs (x) >= 1) return 0.;
ore the transition band.
But there should be no signal with medium or high tonality in the
transition band. For f>12 kHz no problem, for f>8 kHz acceptable,
for f<1 kHz unusable.
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ers 89582 Sep 30 17:50 origin-progressive.jpg
-rw-r--r-- 1 pfk users 102270 Sep 30 17:49 origin.jpg
May be the same is true for MP3 files.
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On Sat, Sep 30, 2000 at 12:49:39PM +0200, Robert Hegemann wrote:
>
> You can add --raise-smr 1 to 1.) and compare again
>
I thought the NMR is important, not the SMR. This I do not understand.
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ng code like:
for ( ...; i < ...; i++ ) {
k = imuldiv (i, iOld, iNew) - WINDOW_SIZE;
ScalarWindow ( Dest.fData [i], Coeff [i%iNew], sData + k );
}
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coding.
In this case compatibility can be achieved via a simple trick.
[_] Frontends also accessing this structure during the coding process.
This makes things much harder.
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unnecessary distortions. You can see this in the spectral view of
CoolEdit Pro.
LTI stands for Linear Time Invariant. Non LTI systems are generating
additional frequencies instead of only emphasing and deemphasing
frequencies.
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quot;
"pushfl \n"
"popl %eax \n"
"movl %eax,%ecx \n"
"xorl $0x0020,%eax \n"
"pushl %eax \n"
"popfl \n"
"pushfl \n"
"popl %eax \n&qu
ting the dirties possible program?
main(_){float x,sin();for(_=0;_<_*_>>8;_+=_) ...
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or)
lame_perror ( "The following error occured", error );
fwrite ( fp, 1, bb.len, bb.ptr );
if ( bb.size > 16384 )
printf ( "BitBuffer was increased by a subroutine, instead of crashing!\n" );
free_bytebuffer ( &bb );
error = lame_close ( gfp );
if (error)
)
Note: POSIX unbuffered I/O works in this way:
open/read/write/close/lseek/dup/dup2
Good operating Systems also hiding the internal structure by
memory protection (Kernel/User space).
Hope this helps.
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, reason for BUG?
:: 2880*8 = 23040 bits, more than 32767/2, reason for BUG?
::
:: I haven't looked deeper at this, I had no time yet :-(.
::
test this also with --noshort.
--decoder crashs on larger data rates and --noshort.
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So you can see: MA = P_A(z), AR = 1/Q_B(z) and ARMA = P_A(z)/Q_B(z).
Example: The easiest AR filter, a integrator (1st order) can only
be programmed by a infinite long MA filter.
Are polyphase filters LTI systems? FFT filters aren't.
And they are comparable with the
oid problems instead of patching the results of this problems.
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_max_bitrate;
...
} LAME;
#endif
LAME* lame_open ( void );
intlame_set_crc( LAME*, enable_t );
intlame_set_genuine( LAME*, bool );
intlame_set_lowpass( LAME*, double );
intlame_close ( LAME* );
double lame_report_lowpass
eopt='-b112 -V1 -q1 -mj --lowpass 15.0'
lameopt='-b128 -V0 -q1 -mj --lowpass 15.0'
lameopt='-b128 -V1 -q1 -mj --lowpass 15.0'
lameopt='-b160 -V0 -q1 -mj --lowpass 15.0'
If you are recording for CDs change fs to 44100.
May be someone can add thi
much as to cause clipping.
>
> This utility will do exactly that:
>
> http://www.chat.ru/~lrsp/English/index.html
>
I can't download any file. Download starts at 5 KB/s and decreases slowly to
50...100 B/s. Time-Out after 50...100 KB.
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for all fs.
Long Blocks Short Blocks
32 kHz: ...15.25 kHz...14.92 kHz
44.1 kHz: ...15.96 kHz...15.50 kHz
48 kHz: ...15.96 kHz...15.62 kHz
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PS: °) minimum requirements of studio equipment frequency respons
:
* white noise from 1...18 kHz (+/- 3 dB)
* attack time: ca. 0.5 ms
* release time: ca. 20...30 ms
* but: no silence between the attacks
How to capture Win95 Screen Shots? What utility would be the best?
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range from 1...10 MByte.
Some of them are really nasty for MP3. Should we collect such programs?
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move block end 0.1s =>
r/R Review
f/F Forward/Cue
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r diffusion in the frequency domain to synthese a different but
(nearly) in the same way colored noise after decoding.
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f masked signal
* For noisy signals MP3 must code nearly every spectral signal
resulting in nearly no savings
* the human ear can't distinguish noises with nearly the same
spectral power density, so it is sufficient to code
a noise signal with a similar but different SPD.
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a MP3 to a fix-point PCM file. These decoders have problems
with high level clipping and with low level quanization noise.
MP3 can handle a dynamic of up to 400 dB.
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.
::
top is displaying a lot of nonsense for RT tasks.
You can write RT programs allocating 50% of the CPU power and top
still says the program generates a load below 1%.
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e source code base?
::
How about making an exact plan what part of code moving to what file?
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uintint
> >
> > float float gives a double.
>
>
> Silly, you are saying above that it is fully wrong and then
> telling the same at the bottom.
>
When it is the same it should be gave the same result. But it gave different
results.
Also tables are much nicer. Table lookup is faster and more error proof than
text parsing ;-)
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printf ( "%30.12f\n", (float)x * y_float ); // give a different result on gcc
printf ( "%30.12f\n", (double)x * (double)y );
printf ( "a*a is %g, sizeof(a*a) is %u\n", a*a, sizeof(a*a) ); // ;-)
return 0;
}
Maybe Richie made an arrangement with the FOR
On Mon, Sep 18, 2000 at 06:40:23PM +0200, Robert Hegemann wrote:
> Frank Klemm schrieb am Mon, 18 Sep 2000:
> > ::
> > :: 1) one type is long double, the other will be casted to long double
> > :: 2) one type is double, the other will be casted to double
> &
frequency response of the audio equipment and listening room, especially
interferences and hall. See remarks in AAC listening test.
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claration. (-paramuse will suppress message)
hello.c:3:29: Parameter argv not used
Finished LCLint checking --- 3 code errors found
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s on the hardware, so
also 13 bit CPUs with 1's complement and really strange arithmetic
(a+b-b!=a on overrun) can be supported).
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s
Emacs Settings
~~
So, you can either get rid of GNU emacs, or change it to use saner values.
To do the latter, you can stick the following in your .emacs file:
(defun linux-c-mode ()
"C mode with adjusted defaults for use with the Linux kernel."
(interactive)
(c-mode)
(c-set
.00 bit
Listening X X is B? 20/20 524288.00 1.00 bit
Listening X
Is this a bug in toolame or a bug in the MP2 decoder?
-v15 (toolame uses a small letter and large numbers are
high quality, large size) should be sufficient for MP2.
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::
:: Albert you are right, but this shows that it is necessary to be
:: resolved, not casted.
::
Compile programs with gnatmake, not with gcc ;-)
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PS: Ada programs are compiled with gnatmake. Make functionality is part of
Ada itself.
eMail
|
\___||
x 55 dB(A) \___/ \
_ \___
\_ \___
\_
\_
\_
\_
x\ 40 dB(A)\_
\ \_
\ \_
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::
:: 1) one type is long double, the other will be casted to long double
:: 2) one type is double, the other will be casted to double
:: 3) one type is float, the other will be casted to float
Fully wrong.
The rest I haven't checked.
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put signal, aharmonic distortions and also preecho
problems, with can be reduced by variable analyzer window size.
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point interpolation
instead of the sinc interpolator?
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r. Here it
is a very fast piece of code looking for uncritical to code parts in the PCM
file due to the problem of an empty bit pool at coding start (fixed size
pools are also possible, but more difficult).
The first great success where to search very silent places.
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that.
An functions become shorter. Programmers may like large functions,
at least maintainers not.
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play shows it in CPU time, i.e. in real CPU clocks.
The REAL time display shows the REAL world time which normal humans really
interest.
The last is the remaining real world time until the program is finished.
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worth the risks and effort
:: to put this into lame.
::
Currently Life Music or Classic you must encode as one big file and then
split. Some players are able to play such generated files without a gap.
MP3 files generated file by file are not usable for HQ archives.
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[Charset iso-8859-2 unsupported, filtering to ASCII...]
:: Frank Klemm wrote:
:: >
:: > The following outputs have the following meanings:
:: >
:: > [ ] p = 0.00%, never used
:: > [%..] 0.00% < p < 0.01%
:: > [%.0] 0.01%
:: ETA - Estimated time of arrival.
::
Why this time changes continuously?
Is the estimation so bad?
I think it's not the estimated time of arrival, but the remaining
time until arrival. RTUA?
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ne who can translate this into *good* english?
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question would be, what is best:
* Showing the percentage relative to the number of already coded frames
* Showing the percentage relative to the number of expected frames
Both have advantages and disadvantages.
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el(s), 16 bit/sample
out: 48000 Hz, 2 channel(s), 32 bit/s
full huffman search ON
73.8 seconds running time
encoding finished.
$ _
This feature is enabled for -qual=7...9 and without any quality selector.
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There are two possiblities for display update frequency:
1st)
#define I_HAVE_NEVER_SEEN_LAME_ON_A_486_100_OR_A_ATHLON_1000
and display updates every 50 frames (MPEG-1) or 100 frames (MPEG-2)
(Why this differences?)
2nd)
Do not define I_HAVE_NEVER_SEEN_LAME_ON_A_486
, but the outer layers of lame.
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poem and now I can ricite the poem without any problem ;-)
The same can be done with: phone, voice, radio, tape, cd and studio.
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brhist function to display percentages between 0.05 and 1
more accurate.
[%.5]
Can be read as 5 %. (%. = parts per thousand) or as
.5 % = 0.5 %
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>48 320 7680 0
> --
>kHz kilo Hertz
>kbps kilo bits per second
>bpf bits per frame
>
That's why FhG switches to 48 kHz on 320 kbps.
But:
A lot of players and soundcards have problems with 48 kHz
Upsampling of lame is worse (
LUDED
4 name_extention_INCLUDE
5 name_DOT_extention
6 somethingelse_H
Select one and apply it to ALL files. 3) seems to be a good joice.
Someone against this?
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lso
runs, but the output have nothing to do with MP3.
Okay, C is a language "guarantee for nothing, compile everything", so it's
very difficult to write something which is "complient C code".
Yesterday I compiled "The old man and the sea", got only 3 warnings, but
* only two frames are 320 kbps, so also use -B256 to avoid problems
with some players
Have someone measured the coding delay for FhG and lame ???
How many PCM samples must be feeded to achieve the first MP3 frame?
May be FhG have a longer delay to see more of it's "future"?
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have a lot of shaping noise in this
range, so the Encoder is confused by this noise.
:: And if it's not the case, why only lowpassing and not bandpassing?
::
Bandpassing is better, indeed. If available.
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becomes more and more unlikely for high
bitrates.
This is better than a sharp break.
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Is it possible to extend the report at the end?
128.0 kbps frames: 1 total, 571 short, 8917 Mid/Side
"%5.1f kbps\tframes: %5lu total, %4lu short, %5lu Mid/Side\n"
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610 KB
voice (0...12 kHz) fs=24 kHz fs=32 kHz
1.8 MB 2.0 MB
So I suggest to use block 1, which uses the fs with the lowest bitrate.
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ency high level
hum) you can achieve entropy differences of 350 kbps for LR vs. MS:
L ca. 80 kbps
R ca. 80 kbps
M ca. 250 kbps
S ca. 250 kbps
Due to the low frequency hum correlation is about 70% . But MS stereo needs
nearly (unavailable) 500 kbps.
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32
-110 dB 2 kHz0.01 32
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) 36864 36864 36864
What would be done in VBR/ABR if the needed frame size is between two
allowed sizes? Emitting frames with wobbling effective frame size or also
using a (small) bit pool and a more constant effective frame size?
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100, calculate:
floor ( 4. * ld (32767/19100) ) = floor ( 4. * 0.77867 ) = floor ( 3.1147 ) = 3
Enlarge scale factors by 3.
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.
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*R
Right = W - 10*R
oder
Left = -W + 10*R
Right = +W + 10*R
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ger?
> | > To do this, change:
> | >
> | > BLACKSIZE = 200change in util.h
> | > filter_l = 191 change in util.c
> | >
200 - 19 == 191 ???
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) And since I wanted to make something
:: that could normalize, then had the idea of trimming, and then had the one of
:: DC adjust. I think I'll go this way, since I have to scan the whole file for
:: the other processings...
::
trimming and normalization can be done without first
ding.
:: Finally, all that is mathematically right, should not be right in life.
::
To find a mathematical solution, fill out the sheet:
aim (note: not a know solution, really the aim):
addition conditions:
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1.131
technik/tapedeck/aiwa_929.wav
38.973% 43.374% 87.996% MS-Stereo 0.002%0.009% 1.113
technik/Fraunhofer_Beispiele/extended-PCM/BlackBird.wav
6.572%6.202% 71.891% MS-Stereo 0.944
technik/Fraunhofer_Beispiele/extended-PCM/BlackBird.wav
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
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freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
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nnels not supported: %s\n", header[11], name );
break;
}
}
int main ( int argc, char** argv )
{
char* name;
intfd;
report_init ();
if (argc < 2)
readfile ( "", 0 );
else
while ( (name = *++argv) != NULL )
s
or -mf. But 160 kbps should have enough reservoir
to cover this problem. For -ms I doubt that this is
the case. See ^^^
Also note that "a" tends to slip from big positive
-mj, -mf, -ms)
*
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Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
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set of reserved/key words are
:: enough to make porting a chore, usually an unnecessary one. Then you have
:: C99 with a pile of features that C++ lacks, such as variable-length arrays.
::
So it is wise to use the subset of C89/C95/C99 and C++ that is identically
in this languages.
--
Mit freundlic
::
Yes. The question is what you hear first, the distortion or the
primary tone.
:: I want to made sub-woofer with 16-30Hz range for my
:: home stereo, but no lower.
::
Note:
fu box volume
50 Hz10 l
40 Hz24 l
30 Hz77 l
20 Hz 390 l
10 Hz 6250 l
--
Mit freundliche
new option:
--nice: Changes priority depending on system load
(uses clock, gettimeofday and sleep: nice is not
portable and useless to it bad design)
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone
Ich benötige ein neues File für mlame/mlame.bat
mlame_corr.c
oder so.
mlame soll (auf Wunsch) automatisch zwischen -ms, -mj und -mm umschalten.
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641
Decoding of Layer 3 doesn't work anymore.
Last checkout/commit/checkout in the night So/Mo worked, now it is broken.
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitschei
the rounding errors cause some problems
:: | >
:: | Rounding cases problems, especially on high quality/low noise audio.
:: |
:: | The simpliest way is to add 0.75 LSB triangle noise before rounding.
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
ph
error bug removed
*/
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
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diff -bwB --brief $file1 $file2; then
echo equal $i
else
echo -e "\033[7mDifferent: $i\033[0m"
mgdiff -title "mgdiff $i" -geometry 1536x800 -args "-bw" $file1 $file2 &
sleep 2
fi
done
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTE
ded a remark on this piece of code.
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PROTECTED]
phone | +49 (3641) 64-2721home: +49 (3641) 390545
sMail | R.-Breitscheid-Str. 43, 07747 Jena, Germany
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MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
:: Also possibly fixed a mono-decoding bug in layer1.c?!
::
Fixed. First I tried to print out the Layer I/II/III file format (download
of Staroffice was necessary). Guessing is really not my strength.
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTECTED] home: [EMAIL PR
1.189 : 1
:: > +2 97.2 3.651.259 : 1
:: > +2.5 99.0 5.431.334 : 1
:: > +3 99.8 8.4 1.412 : 1
:: > +3.5 99.9 9.5 1.496 : 1
::
Search also for &quo
e data:
|--|
MS/LR ^^^
gpsycho ^^
coder ^
coder ^^^^ (out of bit scenario)
--
Mit freundlichen Grüßen
Frank Klemm
°) circular overlapping buffer
Two successive buffers are
[Charset iso-8859-2 unsupported, filtering to ASCII...]
:: Frank Klemm wrote:
:: >
:: > :: > -md is not documented.
:: > ::
:: > :: What is it ? Does anyone made a dual channel mode?
:: > ::
:: > ::
:: > It's like -ms, but no bitrate balance is
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