>
> :: >
> :: > Question: When fs_in/fs_out is not representable by a:b with little a and b,
> :: > what do you like best:
> :: >
> :: > [_] Lame rounds fs_in in a way, so fs_in/fs_out is representable by little
>a:b
> :: > [_] Lame have a function to resample exactly any rati
> ::
> 1. Chebychev filters (LTI system) have no leakage at all
> 2. Chebychev highpass filters (like all IIR filters) have a worse
>frequency resolution for high frequencies ( -> oo) and a extremly
>good for low frequencies ( -> 0).
> 3. You can program a FFT like filterbank with simpl
:: >
:: > Question: When fs_in/fs_out is not representable by a:b with little a and b,
:: > what do you like best:
:: >
:: > [_] Lame rounds fs_in in a way, so fs_in/fs_out is representable by little a:b
:: > [_] Lame have a function to resample exactly any ratio
:: > [_] bot
::
::
:: > > I think this should be a seperate utility outside of lame? Most people
:: > > encode from CDs, which usually are already correctly filtered for stuff
:: > > below 20 Hz.
:: > >
:: > For pop music this is (mostly) true. I will test several CDs. Next week.
:: > The psycho par
> > I think this should be a seperate utility outside of lame? Most people
> > encode from CDs, which usually are already correctly filtered for stuff
> > below 20 Hz.
> >
> For pop music this is (mostly) true. I will test several CDs. Next week.
> The psycho part I would nevertheless filter wi
> > Is there any application for this?
> >
> May be applications where an exact synchronization is necessary (video?
> duplex MP3 encoding/decoding?). I don't know. But if it is necessary it is
> hardly to fix. Otherwise we need additional 4 or 8 bytes of RAM to store
> a double or long double in
On Sat, Sep 30, 2000 at 10:08:09AM -0600, Mark Taylor wrote:
>
> I dont like code like that either! Good thing stuff like that
> is not in LAME.
>
> MP3 encoding, and WAV input files only support integer sampling
> frequencies.
>
AIFF supports IEEE-854 80 bit floats. AU only support 8 kHz. MP3
On Sat, Sep 30, 2000 at 10:25:42AM -0600, Mark Taylor wrote:
>
> Which filters have artifacts?
>
> low pass code is implemented with a very high quality polyphase
> filterbank. It is a near-lossless filterbank with a 256 point
> window and is claimed to have very good frequency resolution.
>
>
>
> :: As a value of 200 for BLACKSIZE showed an improvement in resampling, why
> :: does is still got a value as low as 25?
> ::
> Low pass, high pass and resampling code should be replaced by artefact-less
> program code.
>
Which filters have artifacts?
low pass code is implemented with
> Frequencies are real numbers, not integral numbers.
> And I don't like code like:
>
> if ( Frequency == 44055 || Frequency == 44056 ) Frequency = 2863636/65.L; //
>NTSC PCM
> if ( Frequency == 31468 || Frequency == 31469 ) Frequency = 2863636/91.L; //
>NTSC Hi8 Digital
> if ( F
> From: "Mark Taylor" <[EMAIL PROTECTED]>
>
> I hope to add something soon which has it precompute the exact amount
> needed. Does anyone have code which computes the lcd (largest
> common denominator) of two ints? I think the number of windows needed
> is given by: out_samplerate/(lcd(in_sampl
::
::
:: >
:: > As a value of 200 for BLACKSIZE showed an improvement in resampling, why
:: > does is still got a value as low as 25?
:: >
:: >
:: > Regards,
:: >
:: > --
:: >
:: > Gabriel Bouvigne - France
::
:: Hi Gabriel,
::
:: Increasing BLACKSIZE only improves the sha
:: As a value of 200 for BLACKSIZE showed an improvement in resampling, why
:: does is still got a value as low as 25?
::
Low pass, high pass and resampling code should be replaced by artefact-less
program code.
All three are currently done by code not being a LTI system, which results
in unn
Mark Taylor schrieb am Don, 28 Sep 2000:
> I hope to add something soon which has it precompute the exact amount
> needed. Does anyone have code which computes the lcd (largest
> common denominator) of two ints? I think the number of windows needed
> is given by: out_samplerate/(lcd(in_samplera
>
> As a value of 200 for BLACKSIZE showed an improvement in resampling, why
> does is still got a value as low as 25?
>
>
> Regards,
>
> --
>
> Gabriel Bouvigne - France
Hi Gabriel,
Increasing BLACKSIZE only improves the sharpness of the lowpass
cutoff. For resampling, I dont think we n
As a value of 200 for BLACKSIZE showed an improvement in resampling, why
does is still got a value as low as 25?
Regards,
--
Gabriel Bouvigne - France
[EMAIL PROTECTED]
mobile phone: [EMAIL PROTECTED]
icq: 12138873
MP3' Tech: www.mp3-tech.org
--
MP3 ENCODER mailing list ( http://geek.rcc.se
Bad idea to use analogue recording through a soundcard. You get added distortion from
the digital to analogue (DA) conversion from the CD player and then more distortion
going back to digital (AD) through the soundcard. If you want to EQ a song then do it
digitally. Rip the song using a CD r
Oh, okay... Looks like I got in over my head... Hmm, recording a CD thru the line-in
seems to work fine for me... Oh well :-\
About the EQing, I think Greg misunderstood me... I use the EQ in the CD ripping
process, not in the MP3 decoding process. I use Winamp to play MP3s, & I leave its EQ
f
Hi Mark,
you wrote on Wed, Oct 27 1999:
>> lame -b 128 -X5 -v -V 4 -h -k -d --resample 48 in.wav out48.mp3
>The bug is that the error message "Error: resample code not yet
>written!" was not being printed :-)
LOL! Thanks for the clarification.
>I think the upsample to 48kHz at 320kbs because
> Hi all,
>
> just gave it a try:
>
> lame -b 128 -X5 -v -V 4 -h -k -d --resample 48 in.wav out48.mp3
>
> The .wav file was grabbed from an audio-CD (44.1 kHz). mp3 sounds
> horrible, way too fast. Is this a feature or should the resampled file
> sound like the original? The FhG resamples to 4
Hi all,
just gave it a try:
lame -b 128 -X5 -v -V 4 -h -k -d --resample 48 in.wav out48.mp3
The .wav file was grabbed from an audio-CD (44.1 kHz). mp3 sounds
horrible, way too fast. Is this a feature or should the resampled file
sound like the original? The FhG resamples to 48 kHz without any
n
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