ode calculates first
the correlation of the two channels and forces mono on r = 0.98.
For r 0.98 the joint mode is selected (using MS + AB frames).
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How to enable MMX/3DNow/ISSE support?
MMX:remove the two '#' in the Makefile
3D-Now: ???
ISSE: ???
No documentation found.
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mentioned above, but apart from that?
::
At least the quantization process from float/double to signed short
at the end can be done differently. Differences are auditable.
Code is nearly ready (and used in my HQ resampling program).
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:: On Tue, 25 Jul 2000, Frank Klemm wrote:
::
:: How to enable MMX/3DNow/ISSE support?
:: MMX:remove the two '#' in the Makefile
::
:: Wow, I didn't know this feature. I've just tested it but it
:: doesn't seem to improve performance :-?
::
Performance improvement seems
='-mj -d -q1 --cwlimit 11.5 -X6'
for i in $cbr; do
lame_old $lopt -b$[i] ${src}.wav${dst}_lame385_cbr_${i}.mp3
lame $lopt -b$[i] ${src}.wav${dst}_lame386_cbr_${i}.mp3
done
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);
}
} while ( iread );
if ( wavsize 0 )
wavsize = 0;
wavsize *= 2 * gfp - num_channels;
decoder_progress_finish ( gfp );
/* if outf is seekable, rewind and adjust length */
if ( ! fseek ( outf, 0, SEEK_SET ) )
WriteWav ( outf, wavsize, gfp - i
Is there any possibility to
* feed lame with 24 (or may be 32 bit) PCM data?
* to decode a MP3 to 24 (or may be 32 bit) PCM data?
* To use an own float - short quantizer?
How to add code to the CVS compository?
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)?
* is IS for lame planned?
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213288 13.mp3
212497 14.mp3
212236 15.mp3
212132 16.mp3
212180 17.mp3
212181 18.mp3 and higher
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requirements are only twice
the requirements of the not sufficient 16 bit integer format.
No alignment problems at all like with 24 bit integers.
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back without resorting
:: to winamp.
::
lame --decode Source.mp3 Destination.wav
lame --decode Source.mp3 - | Program_to_read_wavs_from_stdin ...
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. movies
with english/german audio track , r=-0.20...+0.20).
There are a lot of MP3s out there with mono recordings coded with -mj and
also -ms.
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PS: What's the difference between '-mm' and '-mm -a' ?
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u are listening at 100 dB full scale SPL.
Options like '-b128' don't solve this problem. For medium level music there
are still too less bits. Only the low level parts get enough bits, may be
also too less.
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::
:: Thanx
:: Francois
mp3_check. It's ANSI C.
-r--r--r-- 1 root root16726 Jun 30 22:46 mp3_check-1.95.tar.bz
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intermediante roundings, larger range needed, because of value
shifting).
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korr.c
ible noise.
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:: Frank Klemm schrieb am Sam, 05 Aug 2000:
:: We should support an option (-ma for Mode Auto) which switches between -a -mm
:: for highly correlated channels (r 0.98 = mono), -mj for a normal
:: correlated signals (r = -1.00...-0.20, 0.20...0.91 = stereo) and -ms for nearly
::
:: 2nd problem
:: ~~~
:: Some of my CDs have a lot of DC offset. Other are well DC free. Examples:
::
::
:: What is DC?
::
Direct Current. Non wriggling part of the audio signal ;-)
n-1
DC := 1/n Sum x_i
i=0
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assignments, so the bit assignment
depends on this assignment.
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0 Aug 10 00:08 C_lame386wcbr_024.mp3
this files were okay with 385 and 386 alpha 1:
-rw-r--r-- 1 pfk users 13270 Jul 29 11:46 C_lame386wcbr_016.mp3
-rw-r--r-- 1 pfk users 19872 Jul 29 11:46 C_lame386wcbr_024.mp3
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eMail
is that it does not decode L1 and L2? Why this restriction?
::
mpg123 decodes Layer 2. Layer 1 I can't test (no file).
lame --decode tries to kill the tweeter.
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is 1% that normal people will use it.
If you need a FAQ then 90% of the album MP3 files out there are gappy.
DECODING:
A small change in every decoder and the user have nothing to do at all.
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[01:38.23] 229787 [51:03.62]no no 2
TOTAL 237160 [52:42.10](audio only)
Ripping from sector 0 (track 1 [0:00.00])
to sector 237159 (track 17 [1:38.22])
outputting to track01.cdda.wav
...
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%]**
average: 125.7 kbps
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rently bit rates below 8 kbps are a
"Mission Impossible".
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:: Has anybody tried optimizing filter banks?
::
Speed, Quality, Code size?
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so fixes
have an immediate effect. Decoder output is seldom stored for a long time.
Encoder output IS usually stored for a long time, is transmitted via
WAN networks.
That's the "little" difference.
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°) I use the term "broken" if t
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__
| 1 | 3 | 5 | 7| 9 MP3 frames
| 2 | 4 | 6| 8 |
For that lame must encode a collection of files at once.
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p
:: I think for Layer I it's 384.
::
MPEG-1 Layer I is 384 samples. MPEG-1 Layer II and III is 1152 samples.
Values are halved for LSF.
See original mpg123/common.c source file.
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Is there a ABX tool out there? A simple one is enough.
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Comments on the Web Page:[A
goldc.wav: Also low correlation between L and R makes coding
difficult.
testsignal4.wav:It *is* Enya. One of the songs of the Albums
"Watermark" and "Shepherd Moon". But it is not the
:: Frank Klemm [EMAIL PROTECTED] wrote:
:: I've been doing some tests on the accuracy of decoder implementations.
::
:: A full quality-optimized decoder will always fail the test.
::
:: The "round to the nearest integer" is the worst thing you can do.
::
:: I don't
oreseeable jumps in the code. Floating point becomes fast and
unforeseeable jumps slow on modern computers.
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t ...
Worse: s shp s shp s shp ...
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s in kHz
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:: Frank Klemm [EMAIL PROTECTED] writes:
:: If the Output is a AIFF floating point PCM file, there is no rounding to the
:: nearest integer (in the first approach). Otherwise you have to quantize
:: the result of the MP3 decoding process. The most often used (and worst)
:: operator
nd_of_line
);
}
...
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R and VBR have nearly the same data rate.
For fm, radio, tape, hifi, cd and studio I've used ca. 30 minutes of music
(rock, classic, jazz). CBR and VBR have also nearly the same data rate.
May be I adjust this again in the beta phase.
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t'. Please check for errors. Search for 'short'
and 'sample_t' and check for the right data type.
Another hint: short is a slow data type on Pentium/32 bit Code and DEC
Alpha. It makes no sense to write code like:
short i;
for (i=0; i1024; i++) ...
Please use an int.
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ng between user input and internal representation should be done
near the user interface, not in the most inner loop of an utility
function.
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Why a quadratic interpolation is used for upsampling instead of a sinc
interpolator?
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:: Frank Klemm schrieb am Fre, 01 Sep 2000:
:: lame/CODING_STYLE, version 0.001 ;-)
:: -
:: This is the first try of a Coding Style:
::
:: notes on some points
::
:: * Don't use tabulators (the character
How to compile lame on a system where only a C++ compiler is available (the
C compiler costs extra money)? Currently lame generates nearly uncountable
errors with a C++ compiler.
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isturb the remaining thousands of warnings. C++ have problems
more problems to compile every collection of ASCII characters ;-)
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turn value of the final fwrite().
There are runtime libs out there which reports an error for the request of
writing 1 item of 0 bytes.
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:: How do you get mlame to batch encode MP3-MP3 without adding a ".mp3" to
:: every file; resulting in a file with ".mp3.mp3" as extentions?
::
:: Brent
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::
Fiexed.
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I have heavy problems replacing the definition of the element:
internal_flags
in the structure lame_global_flags.
lame_internal_flags is defined in lame.h with:
//
/* more internal variables, which will not exist
hen Grüßen
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as little endian, otherwise as big endian.
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el Sol -- [5] Fragilidad.wav
5.396%6.870% 56.256% Sting/Nada como el Sol -- [5] Fragilidad.wav
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with 20 CD albums):
-mm 100%
-mj -mm * 1.62 ...1.88
-mf -mj * 0.999...1.001
-ms -mm * 1.985...2.001
-md is not documented.
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ou write a C program with 1 serious error/1500 LOC, you are a top
programmer.
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.
::
Waiting for more than 2 or 3 days makes it nearly impossible to check in
changes. That are the results of the usage of SourceSafe at work.
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:: would it be possible to encode an mp3 at ABR 320 using frames larger
:: than 320 kbps?
:: e.g. 440kbps , 560kbps
::
560 kbps result in distortions.
551 kbps is the last without heavy artefacts.
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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.
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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.
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1.412 : 1
:: +3.5 99.9 9.5 1.496 : 1
::
Search also for "masking_lower" in "lame.c".
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-|
MS/LR ^^^
gpsycho ^^
coder ^
coder (out of bit scenario)
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°) circular overlapping buffer
Two successive buffers are filled with the same data.
| 1 2 3 4 5
[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 possible between the channels
:: (maybe also
- bit pool fuel
- (the future of the audio file using one mechanism I mailed before)
- user options?
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°) SPL = Sound preasure level
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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.
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error bug removed
*/
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/`
if 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
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a remark on this piece of code.
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Decoding of Layer 3 doesn't work anymore.
Last checkout/commit/checkout in the night So/Mo worked, now it is broken.
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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.
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new option:
--nice: Changes priority depending on system load
(uses clock, gettimeofday and sleep: nice is not
portable and useless to it bad design)
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phone
::
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
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:: 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.
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)
*
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this problem. For -ms I doubt that this is
the case. See ^^^
Also note that "a" tends to slip from big positive
values directly to opposite negative values.
See also next e-Mail.
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eMai
intfd;
report_init ();
if (argc 2)
readfile ( "stdin", 0 );
else
while ( (name = *++argv) != NULL ) {
if ( (fd = open ( name, O_RDONLY )) = 0 ) {
readfile ( name, fd );
close ( fd );
} else {
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ik/Fraunhofer_Beispiele/extended-PCM/BlackBird.wav
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nally, 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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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 scanning the whole
file.
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change in util.h
| filter_l = 191change in util.c
|
200 - 19 == 191 ???
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.
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. * ld (32767/19100) ) = floor ( 4. * 0.77867 ) = floor ( 3.1147 ) = 3
Enlarge scale factors by 3.
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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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) 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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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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e.
:: And if it's not the case, why only lowpassing and not bandpassing?
::
Bandpassing is better, indeed. If available.
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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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hing 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 the
program does not run. Strange ;-)
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any problem ;-)
The same can be done with: phone, voice, radio, tape, cd and studio.
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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 (downsampling is good, but not very good)
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, but the outer layers of lame.
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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
.
$ _
This feature is enabled for -qual=7...9 and without any quality selector.
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of already coded frames
* Showing the percentage relative to the number of expected frames
Both have advantages and disadvantages.
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:: 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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that.
An functions become shorter. Programmers may like large functions,
at least maintainers not.
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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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[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% p 0.05%
::[%.1
and also preecho
problems, with can be reduced by variable analyzer window size.
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