MP3 uses 576 and 192. When 576 is too low for tonal music and 192 too
long for
percussions, then this is right. But a 1:8 ratio can create other
problems.
Note that MD uses 128, 256, 512 and 1024 sample blocks.
Useful are block sizes from 1 ms ... 35 ms.
Minidisc also uses mixed windows.
On Sun, Sep 24, 2000 at 10:57:39AM +0200, Gabriel Bouvigne wrote:
I've got another question about window sizes: are the short ones really
essential in VBR? Would it be possible to only use long ones, and then
allocating a lot more bits in the case of transcients? After all, Xing uses
only
On Sun, Sep 24, 2000 at 10:57:39AM +0200, Gabriel Bouvigne wrote:
Minidisc also uses mixed windows. Perhaps mixed windows would help in our
case.
I've got another question about window sizes: are the short ones really
essential in VBR? Would it be possible to only use long ones, and then
Hi Gaby
Why can we read in the litterature that humans got 25 CB but mp3 uses only
22?
let us try to get it in order:
bark scale is used by the spreading function
Bark 0 : 0-100 Hz, Bark 24: 15.5 - 20.4 kHz
masking is calculated for convolution bands
Hello Frank,
Sunday, September 24, 2000, 7:43:06 PM, you wrote:
FK How to capture Win95 Screen Shots? What utility would be the best?
press 'print screen' button (copy)
and paste into paintbrush...
8)
Best regards,
Dmitrymail to: [EMAIL PROTECTED]
Why can we read in the litterature that humans got 25 CB but mp3 uses
only
22?
let us try to get it in order:
bark scale is used by the spreading function
Bark 0 : 0-100 Hz, Bark 24: 15.5 - 20.4 kHz
masking is calculated for convolution bands
Lame uses 64 equidistant convolution
"Gabriel Bouvigne" [EMAIL PROTECTED] wrote:
Is there a scalefactor for 16kHz in AAC? (Meno, are you listening?)
AAC has scalefactorbands that fill the whole frequency range, scalefactors
are calculated for all scalefactorbands.
Bye, Menno
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MP3 ENCODER mailing list (
Gabriel Bouvigne schrieb am Son, 24 Sep 2000:
So the highest subbands don't have any scalefactor? I know that Brandebourg
said that there is no proof that 16kHz really contribute to the hearing of
the music, and then it could be intentionnal, but could it be a "bug" or
mistake in the mp3
::
:: So the highest subbands don't have any scalefactor? I know that
:: Brandebourg said that there is no proof that 16kHz really contribute to
:: the hearing of the music, and then it could be intentionnal, but could
:: it be a "bug" or mistake in the mp3 specs?
::
40 Hz...16 kHz
Hi Gaby
Why can we read in the litterature that humans got 25 CB but mp3 uses only
22?
let us try to get it in order:
bark scale is used by the spreading function
Bark 0 : 0-100 Hz, Bark 24: 15.5 - 20.4 kHz
masking is calculated for convolution bands
I have a few questions ideas - potentially stupid, but they've been
bugging me. I'd try all the ideas myself except I can't get Lame to compile
I don't have a clue how to implement them anyway.
1- Is it possible to change the sample rate by encoding frames using other
than 1152 samples? As an
From: Shawn Riley [mailto:[EMAIL PROTECTED]]
6- What's the difference between normal stereo dual channel
In terms of bitstream format, nothing, apart from the frame header. Dual
channel is simply a hint to the decoder that the two channels are intended
to be played separately, rather than
Mathew Hendry a écrit :
From: Shawn Riley [mailto:[EMAIL PROTECTED]]
6- What's the difference between normal stereo dual channel
In terms of bitstream format, nothing, apart from the frame header. Dual
channel is simply a hint to the decoder that the two channels are intended
to be
Shawn Riley a écrit :
2- Are some people saying Layer2 is actually better than Layer3 at the same
bitrates for some types of music? I wonder if quality could be improved by
switching layers midstream... Do MPEG standards support that?
I think that it's forbidden by iso
Regards,
--
Gabriel
From: Gabriel Bouvigne [mailto:[EMAIL PROTECTED]]
In dual channel, each channel has to got exactly half of the bits.
Do you have a reference for that in the ISO/IEC docs? Throughout 11172-3
stereo and dual_channel seem to be treated as entirely equivalent.
-- Mat.
--
MP3 ENCODER mailing
Mathew Hendry wrote:
normal stereo allowing a more "free" allocation of bandwidth between the
channels?
AFAIK it doesn't. I'm not sure where that idea originated.
I have been under the impression for several years that Stereo (mode 0) shares
bits between the channels. If one channel was
From: Ross Levis [mailto:[EMAIL PROTECTED]]
I have been under the impression for several years that
Stereo (mode 0) shares
bits between the channels. If one channel was more complex
than the other then
it would allocated more to the channel that required it. I
presume this is
what
Yes it is. The question is whether dual_channel is more restricted than
that.
Dual-channel is just what the name suggests. Each channel is completely
independant. I don't see any advantage of using dual-channel.
Ross.
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