Segher Boessenkool wrote:
On some signals, however, highpass filters can do bad things as well.
For example, after processing this file
http://www.geocities.com/SiliconValley/Bit/5683/test1c.zip
with a 5 Hz highpass filter, the peak amplitude increased
to about 52000, resulting
Segher Boessenkool wrote:
Perhaps someone could find a wav wich produces this disturbing high frequency
echos while encoding with lame due to low frequency signals. This could be
bassNN_N.wav from SQAM (at low bitrates).
about anything from "Grotus" as well (_very_ fat bass)
On some
Does anyone know how a high pass filter is usually implemented? The
convolution approach (like the low pass filter) seems like it would be
expensive and require a lot of extra internal buffering: a 10Hz
signal takes 4410 samples to represent one period. To get good
frequency resolution (so you
On Wed, 10 May 2000, Naoki Shibata wrote:
2- Resample the input file to something ridiculously low like
40Hz, subtract this from the original WAV. (Although that would
probably require the above method anyway.)
Assume that the original sampling frequency is f.
Downsampling to
I have 3 ideas, but I'm not sure if they'd even work, let alone how to code them if
they did...
1- Try simulating the behaviour of ideal capacitors/resistors/op-amps/etc. Maybe a
highpass filter could be "built" out of computer code put just before the
resampling function. (Or is *that*
_very_ few. Every mdct sample is about 38 Hz wide, so you wouldn't even cut
away the first (or zeroeth, whatever) sample.
There is no such thing as 'zeroeth' !
Remember the "when does the millenium start" problem ? :-)
Hey, C and Perl both start arrays with index 0. I trust them more
Shawn 2- Resample the input file to something ridiculously low like 40Hz, subtract
this from the original WAV. (Although that would probably require the above method
anyway.)
Assume that the original sampling frequency is f.
Downsampling to sampling frequency f/n (n is integer) is easy,
Segher Boessenkool wrote:
_very_ few. Every mdct sample is about 38 Hz wide, so you wouldn't even cut
away the first (or zeroeth, whatever) sample.
There is no such thing as 'zeroeth' !
Remember the "when does the millenium start" problem ? :-)
The reason ISO recommends highpass filtering
Shawn Riley wrote:
Does anyone know how a high pass filter is usually implemented? The
convolution approach (like the low pass filter) seems like it would be
expensive and require a lot of extra internal buffering: a 10Hz
signal takes 4410 samples to represent one period. To get good
Hi
Has everyone tested the results of such a highpass filter ?
Has anyone time to prefilter a wave (with mathlab, scilab, or something else),
send it through lame and rate the results with and without prefiltering ?
Perhaps someone could find a wav wich produces this disturbing high frequency
Perhaps someone could find a wav wich produces this disturbing high frequency
echos while encoding with lame due to low frequency signals. This could be
bassNN_N.wav from SQAM (at low bitrates).
about anything from "Grotus" as well (_very_ fat bass)
Ciao,
Segher
--
MP3 ENCODER mailing
While I preety much agree with the rest of the post, I must comment on
this one basis.
On Tue, 9 May 2000, Gabriel Bouvigne wrote:
[snip]
why a 14Hz high pass filter and not a 20Hz one:
because the lowest tone produced by a true musical instrument is 16Hz (it's
from organ).
[snip]
And the
Hello
I would vote for a default 14Hz high pass filter, removable with the -k
option.
Why an high pass filter:
*theorical minimum audible freq for humans is 20Hz (also very discutable)
*most soundcards are unable to reproduce less than 20Hz frequencies
*most speakers are unable to
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