| You should apply a 16 Hz lowpass filter for DC removal. Note that lowest
| organ note has 16.3Hz.
|
| Did you hear tones under 16Hz? Did you have speakerboxes that you will
give
| these low frequencies? I want to made sub-woofer with 16-30Hz range for
my
| home stereo, but no lower.
|
| Well.
Heribert Maier wrote:
Hello folks,
for some reason I have to convert something like 200 CDs to mp3 and
they should stay on 4CDs. An average bitrate of around 32kbps should
be o.k..
As lots of these CDs (old recordings) are actually mono, although it is
not specified on the cover, and as
Hi all,
I'd like to present the MP3 data framewise (9 frames at once) to the
decoder I'm implementing. Does anyone know of a piece of C code for
splitting a MP3 file into single frames ?
Thanks in advance.
Regards.
Patrick
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
David Bridson wrote:
Just read Mr. Maier's email. I would also like to know
when stereo recordings started being used. Is there an
online source that gives a summary of recording technology history?
I know that the first film ever released in Stereo was in 1940.
Space is not super
| fubox volume
| 10 Hz 6250 l
|
| --
| Mit freundlichen Grüßen
| Frank Klemm
I will use double bandpass double chamber box (6-th order, "BOSE" type),
size (nice piece of furniture :-) approx 300 liters (270lt bassreflex at
20Hz + 30lt bassreflex at 56Hz) with single speaker at Qts
| Odeslatel: hudson kingery [EMAIL PROTECTED]
| I would like to encode a set of Billie Holiday CDs that cover 1933 thru
1938. I would like advice on the "best" options for LAME.
|
| When I checked the WAVs with CoolEdit Pro they show up as stereo. But the
difference between LR "appears" very
| Odesílatel: Stephan Ebertshäuser [EMAIL PROTECTED]
| I just want to ask which layer2 encoder do you prefer or recommend??
| (SOLOH or toolame?)
I like hardware DAB codecs :(expensive)
But if I will time, I try toolame and soloh to test it.
Regards
Jaroslav Lukesh
for some reason I have to convert something like 200 CDs to mp3 and
they should stay on 4CDs. An average bitrate of around 32kbps should
be o.k..
What lame-options do you recommend?
I did some very unprofessional testing and came to this:
-V4 -mm -h -b24 -q1 --resample 22.05
I would
I just want to ask which layer2 encoder do you prefer or recommend??
(SOLOH or toolame?)
I'd recommend using Qdesign.
If you want a free one, I'd use toolame rather than Soloh, as toolame is
soloh+some improvements.
Regards,
--
Gabriel Bouvigne - France
[EMAIL PROTECTED]
icq: 12138873
// pfk
// For 44.1 kHz
// 1... 96 kbps: Mono better than ugly stereo °)
-- // 97...159 kbps: Joint Stereo
-- // 160...192 kbps: Force Joint Stereo bandwidth not enough for LR
stereo, but reducing switching artefacts
// 193...kbps: Stereo enough
lame -m m -h -b32 --voice --lowpass 8 --lowpass-width 1
--highpass 0.054 --highpass-width 0.04 --cwlimit 7
--resample 22.05 --noshort a.wav a.mp3
NOTES:
cwlimit=lowpass minus lowpass-width (kHz)
Did you played a lot with cwlimit? Could you please share your results about
it?
Regards,
--
If you look at the sources of mp3_check you'll find something that scan MP3
files. So it should be easy to only keep the first frames...
- Original Message -
From: "Patrick Ndjiki-Nya" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 07, 2000 9:27 AM
Subject: [MP3
Hello all,
this is a forward of a thread on
http://bboard.mp3.com/mp3/ubb/Forum1/HTML/003127.html :
OK, I gave in and did a quick resampling test.
It shows why you shouldn't use lame to do your resampling.
Here's three plots. 1) Original signal. 2) Resampled via Cool Edit, 3)
Resampled via
Bug is not the right word. Unless someone has been really screwing
with the code. It's just not the most accurate way of doing it but
it is reasonably fast.
Trust me There are worse resampling routines in theis world.
--
Scott Manley (AKA Szyzyg)
Streaming Media Hacker
www.myplay.com
This could very will be a bug, but it my also just be the correct
resonce of the medium quality lowpass filter used by LAME.
For resampling, LAME is using a 19 point lowpass filter with
a Blackman window. The position of the window is adjusted
to do the resampling at the same time.
For a
Mark Taylor wrote:
This could very will be a bug, but it my also just be the correct
resonce of the medium quality lowpass filter used by LAME.
For resampling, LAME is using a 19 point lowpass filter with
a Blackman window. The position of the window is adjusted
to do the resampling at
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:: | | Anyway I think that the very low frequencies are used in music like
:: | | drumbass with very good sound systems. The infra bass is something I
:: | really
:: | | like in clubs ;)
:: | | Since I want to encode files in good quality (maybe playable in a club)
:: | I'd
:: | | prefer
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:: // pfk
:: // For 44.1 kHz
:: // 1... 96 kbps: Mono better than ugly stereo °)
:: -- // 97...159 kbps: Joint Stereo
:: -- // 160...192 kbps: Force Joint Stereo bandwidth not enough for LR stereo,
:but reducing switching artefacts
:: // 193...kbps:
Some source code:
// nasm -f elf wav_korr_asm.asm; gcc -O2 -s -o wav_korr wav_korr.c wav_korr_asm.o -lm
// or:
// gcc -DNONASM -O2 -s -o wav_korr wav_korr.c -lm
#include stdio.h
#include unistd.h
#include math.h
#include sys/types.h
#include sys/stat.h
#include fcntl.h
#include memory.h
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::
:: As you one remarked - yet another option nobody knows the ideal value
:: for ;-)
::
:: Upsampling uses a different algorithm - has anyone looked at that?
::
Yes. Sounds bad. Especially CDs with some distortions above 15 kHz
(dithering noise, 15.75 kHz or 16 kHz tones).
--
Mit
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Now a list of "a" values. Sorry about using a little bit of math.
So see on the list below.
There are some stereo pieces of music with a "a" 1.667 or 0.600
1.25/1.92 Giora Feidman/Rabbi Chaim's Dance -- [15] Tishrei Saba.wav
1.02/1.75 technik/Fraunhofer_Beispiele/track7.wav
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