>At these types of average bitrates, I think you might be better off
>with CBR instead of VBR. This is because with an average bitrate
>230kbs, you only need an extra 90kbs to go up to 320kbs. 90kbs
>is only 40% of the average frame size - these types of fluctuations
>are easily handled by the
::
::
:: That name was changed because one make system (MSDOS?) interpreted
:: the '-' in quantize-pvt.c as a compiler option.
::
MSDOS can't store a name like "quantize-pvt.c", you got at most:
"quantize.c" or "quanti~1.c".
--
Mit freundlichen Grüßen
Frank Klemm
eMail | [EMAIL PROTEC
On Thu, 5 Oct 2000, Robert Hegemann wrote:
> > >From your recent postings I'm detecting that you think -q1 can only rarely
> > give a sonic improvement. In fact it is more likely to degrade the sound
> > over -q2? If so, the Roel recommendation of -q1, seems a little dangerous?
> > You think the
Hello Robert,
Thursday, October 05, 2000, 12:08:21 AM, you wrote:
RH> I don't know any track where the use of -q1 improves sound quality
RH> compared to a same sized -q2. That's why I'm asking you all.
The reason I use it on -V1 is: I don't get poorer quality (still
waiting for m
--
On Thu, 5 Oct 2000 11:05:14
Roel VdB wrote:
>Hello Robert,
>
>Thursday, October 05, 2000, 12:08:21 AM, you wrote:
>RH> I don't know any track where the use of -q1 improves sound quality
>RH> compared to a same sized -q2. That's why I'm asking you all.
>
>The reason I use
Gargos Chode wrote:
> -V1 -mj -b128 -q2 -d -p -k -F --nspsytune --athlower -35 -X3.
Some thoughts:
-p & -F will have no effect on sound quality. I have had mixed results with
nspsytune. -X2 & X3 both produce massively larger average bitrates than all the
others. I've never played with -d.
Ross> Gargos Chode wrote:
Ross>
Ross> > -V1 -mj -b128 -q2 -d -p -k -F --nspsytune --athlower -35 -X3.
Ross>
Ross> Some thoughts:
Ross>
Ross> -p & -F will have no effect on sound quality. I have had mixed results with
nspsytune. -X2 & X3 both produce massively larger average bitrates than al
--
On Thu, 05 Oct 2000 21:30:23
Ross Levis wrote:
>Gargos Chode wrote:
>
>> -V1 -mj -b128 -q2 -d -p -k -F --nspsytune --athlower -35 -X3.
>
>Some thoughts:
>
>-p & -F will have no effect on sound quality. I have had mixed results with
>nspsytune. -X2 & X3 both produce massively larger ave
Hello,
On Thu, 05 Oct 2000 20:02:11
Naoki Shibata wrote:
> First, one should not specify "--athlower -35". This may significantly
>degrade sound quality.
>
> I always used -q1 while tuning --nspsytune. I think -q1 doesn't
>degrade sound quality so much with --nspsytune.
>
> Theoretically
> > I'd vote for limiting it to 8205. The quality loss would be very
minimal. It
> > won't solve winamp's problem, but it's easy to fix in the decoder, now
that
> > we know exactly what the problem is.
> > The advantage is that 8205 is 0x1FFE, which is not a valid syncword for
> > mpeg1-2. With t
if lame writes a 16 bit crc for every frame (using -p switch),
doesn't that mean there are 16 less bits for sound data for each
frame? couldn't that affect sound quality? is this getting
carried away a little too much? :)
On Thu, Oct 05, 2000 at 09:30:23PM +1300, Ross Levis wrote:
> -p & -F
This is a forwarded message
Date: Thursday, October 05, 2000, 11:23:08 AM
Subject: Problems with Lame 3.88alphas + A question
===8<==Original message text===
Hi,
Also the lame 3.88alphas don't run propley on my machine.
When I encode a file the screen keep rolling. This i
RAW sizes differ between the original and the encoded-decoded files, headers
appear to be same (44 bytes) size.
- original wav - raw = header
t4 14,276,68414,276,640 44
- encoded then decoded back (cbr and vbr)
t4_b256_ms_h14,275,8
>> - If the parameter's type changes, the API has to change.
> WOW, IF YOU CHANGE THE MEANING OF A PARAMETER,
> THIS WOULD BE A CHANGE IN THE API ALREADY AND
> WOULD BREAK BACKWARD COMPATIBILITY !!!
The 3-function tag-pair API would handle this fluidly, with no problems
what-s
Hi, I seem to have hit a wall here... I want to do an mp3 ripper for Mac
OSX, which I thought would be a simple enough project, but it's getting
more complicated. The OS automatically mounts cd's in a /Audio CD
directory as aiff files. I thought this would make things easy as I could
just write a
Robert Hegemann schrieb:
> Mark Powell schrieb am Mon, 02 Okt 2000:
> > On Fri, 29 Sep 2000, Robert Hegemann wrote:
> >
> > > does someone know any sample where a VBR encoded MP3 with -q1
> > > gives a better sounding MP3 compared to a same sized VBR with -q2 ?
> >
> > >From your recent posting
Hello LAME Developers,
I was just wondering if anyone has been to able build the GTK version of
LAME 3.87. I have tried with the makefile for MSVC and by using the project
files. I have gotten the same error in both cases:
c:\lame-beta\src\lame3.87\main.c(154) : error C4013: 'lame_decoder'
und
[...]
>So I'm back to square 1, how can I go from AIFC to MP3? I have no real
>experience with sound file formats and I really just want to write a front
>end which uses tools written by people who know a lot more than me such as
>lame :) Any tips are appreciated.
There's one really simple soluti
Howdy,
Unless they have changed it significantly since I last looked at it (quite a
while ago) AIFC is just AIFF with the added possibility of using compressed
audio instead of raw PCM.
Either format is 'chunk' based, like RIFF-WAVE. That is, an AIFF/C file
consists of a number of chunks, most
Hello,
Hrmm... that is an interesting idea. I completely hadn't thought of this. Does this
actually take away bits from being used to encode the audio frame? If so then what is
the real use of this switch? I had thought this switch would help to prevent the mp3
from being possibly corrupte
> You're right Mark, compared to Lame 387 MMX --abr 128 Xing is only two
> times faster Bo)
>
> Regards,
> Wim Speekenbrink
Using 160kbps for both LAME and Xing, encoding "Dire straits - telegraph road"
LAME takes about 1.5 times longer than Xing.
I thought the difference was greater, but I had
Add the following proto-type just above the main() function
int lame_decoder(lame_global_flags *gfp,FILE *outf,int skip);
and you should be set
Albert
http://www.cdex.n3.net/
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
- Original Message -
From: "Nathan D. Blomquist" <[EMAIL PR
Hello Gargos,
Thursday, October 05, 2000, 12:08:31 PM, you wrote:
GC> Have you tried using -q1 on fatboy.wav? It sounds significantly
GC> worse than -h or -q2. If you dont have this file let me know and
GC> I will send it to you.
I agree that -q1 sounds worse on this one using "-V1 -mj -b128 -
Hello,
Roel, maybe you should give these settings a try on that track:
-V1 -mj -b128 -q2 -d -k --nspsytune --athlower -35 -X3
The bitrate stays pretty low (~224kbps) and it sounds very good... almost identical to
the original. These are the only settings I could find that produce a smaller f
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