Hello mr Mark Taylor,
Saturday, May 20, 2000, 7:02:00 PM, you wrote:
MT With our without the "-t", the file sizes of the decoded .wav
MT should be the same. ARe you getting the message:
MT "Oops: first frame of mpglib output will be lost" ?
Sorry, I don't get that message. Also the when
Hello Robert,
I was just going to note that now with the new 3.83 I'm getting albums
averaging 220kbit/s (-V1 -mj -h -b128), and I saw you fixed some in
the CVS already:
quantize-pvt.c 1.106 11 hours robert minor tweaks
quantize.c 1.101 11 hours robert minor tweaks
I downloaded 3.84a1_b
Hello,
To be clear, I don't mind, but am just curious:
Lame 370 gave me 1.7 speed on my Cel400@450 (dkutsanov)
Lame 383 gives me 1.0 speed on my Cel400@450 (dkutsanov)
[Lame 384 -Z gave me 1.7 speed on my Cel400@450 (dkutsanov)]
What was implemented in lame 3.8 so that the speed dropped so
RV To be clear, I don't mind, but am just curious:
RV Lame 370 gave me 1.7 speed on my Cel400@450 (dkutsanov)
RV Lame 383 gives me 1.0 speed on my Cel400@450 (dkutsanov)
RV [Lame 384 -Z gave me 1.7 speed on my Cel400@450 (dkutsanov)]
sorry should have been -Z : 0.7 speed :)
--
Best regards,
Hello Takehiro,
Sunday, May 21, 2000, 6:41:25 PM, you wrote:
TT I wrote;
I added a new scalefac_scale algorithm to CVS version of LAME,
which enables with -Y option.
TT Sorry, -Y option is used for Mark's new VBR routine
TT so I changed "new scalefactor_scale algorithm" from -Y to -Z
Hello David,
Monday, May 22, 2000, 9:50:14 AM, you wrote:
DB How about prepending some "dummy" data to the input ( zero value samples
DB ), note its length and then encode it together with the normal input.
DB When decoding , just discard the noted length of samples ( + plus the
DB ones added
Hello David,
Monday, May 22, 2000, 2:05:47 PM, you wrote:
DB Are you saying that lame ( or mp3 in general ) can not handle sound
DB coming after a period of silence ? I find it hard to believe.
If the original album contained silence in front of a song, it comes
out ok.
If you add some
Hello archrival,
appropriate alias ...
Monday, May 22, 2000, 8:43:28 PM, you wrote:
ann I don't notice any "skip" between tracks after I have burned
ann to a CD using DAO. I know there is possibly silence at the beginning and
ann end, but it is not noticeable to me whatsoever. I think some
Hello Leonardo,
Monday, May 22, 2000, 11:07:07 PM, you wrote:
LS How do you count frames of each bitrate ?
LS I want to do some stats myself (with diferentes settings and lame versions)
LS :o)
win32: musicutter: http://macik.homepage.com
click on mp3 file, and then "statistics - simple"
the
Hello,
I did some test with "-Z", and must say I'm really impressed. Maybe a
little bit more bits to hard parts, but I've found no things that
sound really off.
The size benefit is great!
Just my opinion, many may differ :)
--
Best regards,
Roelmailto:[EMAIL
Hello,
After a few dozen of those frequency analysis graphs, I noticed
something that made me curious: The 16+kHz region of a VBR encoded
file vs the 256S cbr.
http://users.belgacom.net/gc247244/extra/why_oh_why.png [8KB]
is a _very_ striking illustration, and I started thinking about this.
Hello Mark,
Tuesday, May 23, 2000, 8:14:18 PM, you wrote:
MT Actually, there really are 22 "critical bands" or "scale factor bands"
MT used by MP3. I guess we should stick to the C convention, and call the
MT last band the 21'st band. Here are the frequency ranges,
MT along with the ATH:
MT
Hello,
Is there use for 320kbit/s frames in JS mode? Lame -V1 -mj -h -b128
gives them regulary, and I remember someone told me that the max for
each channel is 160kbit/s anyways, so there would be no quality
improvement?
or
- are the saved bits used for bit reservoir?
- to avoid switching too
Hello Gabriel,
Saturday, May 27, 2000, 2:44:36 PM, you wrote:
GB Good point here. If I'm not too wrong, a kind of normalization can be done
GB by just changing the scalefactor value. In this case, such an option would not
change the
GB encoding process. Lame could encode each frame as usual,
Hello Gabriel,
Sunday, May 28, 2000, 12:09:47 PM, you wrote:
This "scalefactor", would this touch the "non uniform quantization" as
described in the Davis Yen Pan doc?
free quote: LIII raises input to 3/4th power to get more consistent SN
GB ratio
over the range of quant values.
GB It would
Hello Takehiro,
Sunday, May 28, 2000, 7:05:46 PM, you wrote:
TT it seems FhG and Thomson changed their patent policy.
TT see http://www.mp3licensing.com/index.html
thanks :)
quotes:
Hello Roel,
Sunday, May 28, 2000, 8:26:01 PM, you wrote:
RV Hello Takehiro,
RV Sunday, May 28, 2000,
Hello Dmitry,
I was wondering: the compiled versions on your site, are they using
MMX? I tried lame384a1_n.zip vs lame384a1_p.zip on my cel450, and I
notice no difference. If it's not the case, could you please
compile a mmx version?
thank you
--
Best regards,
Roel
Hello Charlton,
CH No, it DOESN'T work.
CH I THINK THAT THE LAMEBATCH LINK SHOULD BE TAKEN OFF OF THE LAME WEBPAGE.
My mothertongue is not english, so no big statement here.
Simply: rtfm. I, and with me 100s of users seem to manage to get it
working in vbr mode perfectly so please don't come
Hello All,
I tested to ogg encoder, and must say I'm somewhat disappointed.
Hardly not to after all those "mp3 killer" messages at slashdot etc...
I did a quick listening test, and if this is what lame sounded like
last year, as Mark said, then it has come a _long_ way.
It totally devastates:
Hello Greg,
Monday, June 26, 2000, 7:45:22 AM, you wrote:
GM Us lesser engineers have to show off more.. :P
dilbert joke
GM Anyways, I'm tired of seeing perceptual codecs blasted because of stupid
GM non-perceptual tests.. (which this wasn't but it was close) so it was fun
GM to do something
Hello All,
I've been out for a month (exams), and now the last few days I get
mails like:
GOD Hi man,
GOD Did you receive a lot of complaints from pissed off users already?
GOD It seems that the LAME vers released for the past 3 days make files
GOD about 15% bigger than those until 384a1_2L. I
Hello Takehiro,
Tuesday, June 27, 2000, 2:35:57 AM, you wrote:
"R" == Roel VdB [EMAIL PROTECTED] writes:
TT R did the standard Takehiro VBR mode change? (you know the
TT R "Y"-"-Z" one) What was wrong with the older vbr mode, the size
TT R was notic
Hello,
[MP3 ENCODER] Marks new VBR now the default
I know I'm not a developer myself, but I'll try to make some points.
If I understand correctly, there was beta up to 3.83, and then both
Mark Taylor and Takehiro Tominaga put forward a new VBR algo
(respectively -Y and -Z).
I tested -Z, was
Hello Robert,
Tuesday, June 27, 2000, 5:49:38 PM, you wrote:
RH Hi Roel!
I tested -Z, was very happy with it, and it became default for some
builds.
RH Takehiro's scalefac_scale feature (-Z, it was not another new VBR mode)
RH had to be turned off for VBR, because the VBR modes already
Hello Shawn,
Monday, July 03, 2000, 6:44:12 PM, you wrote:
SR If mpg123 needs all frames to have CRC, can the VBR header contain a CRC?
SR Or would the CRC cause problems if it were to be embedded in a VBR header?
That seems to be the world upside down. Adapting an encoder in order
to fit a
Hello,
In order to give that idea I had last month about an "album-header",
in order to complete (imho) mp3 as a platform to encode all kinds of
cds seamlessly, as opposed to only non-live/mix cds given current
implementation, a chance, I'd like to get some information :) [yes, it
is english]
I
Hello Robert,
Tuesday, July 04, 2000, 6:18:27 PM, you wrote:
I will fix it in a few hours, when I'm home.
Robert
RH I checked in a first fix, more tuning may be necessary.
cool.
Sad thing Dmitry Kutsanov has no more alpha's on his page. :(
Traffic problems? Maybe host the download page on
Hello,
I've been experimenting all day with the "vbr_mt" mode, because of the
great speed advantage over "vbr_rh" in it's current form.
No need to tell anyone the vbr_mt generates very large files, and I
found -V3 (mt) to be somewhat the filesize-equivalent of -V1 (rh).
What bothered me was the
Hello Mark,
Monday, July 10, 2000, 6:21:20 PM, you wrote:
MT The thing I worry about with VBR is the following:
MT A VBR with an average bitrate of 180kbs may sound as
MT good as a 200kbs CBR 99% of the time. But 1% of the time
MT the psycho acoustics could screw up and use 128kbs
MT when it
Hello Mark,
Monday, July 10, 2000, 7:27:20 AM, you wrote:
MT yes, please send me the .wav files with a description of the
MT problem. I have two samples already with artificts, but I
MT haven't had a chance to figure out what is wrong.
MT Mark
I put it (temp) on my website: (1.900kbyte)
Hello,
I was just listening to that 'obvious artifact-creator' that makes vbr_mt do
strange things. Sad thing is that I cannot really easily detect it, on
my stereo that is. I'm just guessing more people will find it hard to
detect. It is however, very clearly present when I listen with my
Hello Steve,
Monday, July 10, 2000, 11:27:36 PM, you wrote:
As noted in the other post, I, and many with me have very little to
complain about in with the =3.85 vbr_rh mode... Cannot find any
glitches since 3.83, encoded a few hundreth albums and counting...
SS So wait, I'm not quite
Hello,
I know I keep going on about this one, but I am reasonably convinced
something is wrong beside that "analog silence". 386 keeps giving me
_consistently_ frames below 128k with "-b128" specified. I've been
told this has to do with analog silence, but there is simply _no
silence_ in next
Hello Robert,
Tuesday, July 11, 2000, 4:44:49 PM, you wrote:
RH OK, once again, there is no -b128 bug!!
My apologies. i don't/didn't understand. I'll read everything over
again and try to comprehend why 3.85 and below functioned as I
expected and 3.86 doesn't.
thanks and sorry for the
Hello David,
Tuesday, July 11, 2000, 6:18:06 PM, you wrote:
D What is this -q1 parameter i've seen here on the list ?
D is it for VBR,ABR, or CBR ?
D is there -qx or only q1, tell me about it
-q1 should be an optimalisation by Takehiro Tominaga running a more
extensive search to find smallest
Hello Chris,
Tuesday, July 11, 2000, 11:37:35 PM, you wrote:
CH We know that LAME now has roughly the same quality of MP3Enc 3.1. But,
CH as far as I am concerned, full huffman search hasn't been implemented
CH on LAME yet. I've noticed LAME 3.8x produces better quality than 3.70,
CH and I
Hello,
Tuesday, July 11, 2000, 6:25:46 AM, you wrote:
MT If you want to try further adjusting the tunings
MT the real thing you want to play with is the dbQ[] array
MT in vbrquantize.c:
MT static const FLOAT8 dbQ[10]=
MT{-5.5,-4.25,-3.0,-2.50, -1.75, -.75, -.5, -.25, .25, .75};
I've
Hello Naoki,
Thursday, July 13, 2000, 9:10:08 AM, you wrote:
Jaroslav Sure. But not only metal. All music needs bandwidtw up to 100kHz. Ears
Jaroslav cannot hear stable sinus frequency, but music is not sinus; music is
Jaroslav impulses, that have more energy at band 20kHz, and you can hear
Hello,
after a lot of testing, I came to something like this in
vbrquantize.c
dbQ[10]={-6.06,-4.4,-2.9,-1.57, -0.4, 0.61, 1.45, 2.13, 2.65, 3.0};
_combined_ with "-q1" this should (for average files)
get -V9,-V4,-V1 back to the size it was with vbr_rh, all the other
values interpolated
Hello Mark,
I put it (temp) on my website: (1.900kbyte)
http://users.belgacom.net/gc247244/extra/velvet.zip
Ok, I found some _visual_ confirmation of the noise component I keep
hearing in the R channel. This is there with vbr_mt, but not with
vbr_rh.
I used 385: "lame -V1 -mj -h -q1"
I used
Hello Robert,
Wednesday, July 19, 2000, 2:16:28 PM, you wrote:
Please download zip#2 from http://r3mix.50g.com [100mb fast webspace,
but no direct links allowed :(]
RH Sorry, but I can't reach that site!
help, murphy or just typical. never had any problems, but maybe your
"lynx" and the
Hello Mark,
Wednesday, July 19, 2000, 5:14:31 PM, you wrote:
MS The site is dead for me also ... 768K DSL from Ohio USA. Murphy's law has
MS been strong this year.
MS mark stephens
yes it seems. thanks for letting me know. So, this afternoon both
sites are down. (1st time in 4 months)
Hello Robert,
Thursday, July 20, 2000, 12:59:20 AM, you wrote:
RH I just fixed that problem with that *pseudo endless loop* :-)
Yes, just tested, and works like a charm...
RH It may have solved the problems with these clips too.
RH If you like to confirm this...
I think about 50% of all
Hello Johan,
Sunday, July 23, 2000, 12:55:48 PM, you wrote:
JA When i try to encode the track "Kort Introduktion till Nada Yoga" from
JA the "Jag gör vad som helst för lite solsken" by Fläskkvartetten (swedish
JA band) lame hangs, it begins to encode the track but it does'nt get
JA anywhere it
Hello,
A reporter I know is planning on writing a book about mp3. (benelux)
We all know the downsides of SDMI and consorts, so I was wondering, is
there a site or so that has a nice presentation and arguments why to
choose MP3 above other formats? (that lists benefits to artists, consumers,
Hello David,
Tuesday, July 25, 2000, 4:25:35 PM, you wrote:
DB Don't forget about Vorbis ( www.vorbis.com and www.vorbis.org )
DB David Balazic
:), thanks for that shameless plug :)
If there's a section on possible successors, I'll mention ogg vorbis.
You know I tested the codec and found it
Hello Mark,
MT I would guess it was introduced in 3.85: it requires a combination of
MT scalefac_scale (defaulted in 3.85, before that enabled with -Y ), and
MT not using enough low pass filtering for the amount of compression.
If I believe the history log and my own tests, scalefac_scale was
Hello,
I believe Robert said that mp3 frames overlap 50%, then would it be
sufficient to init some values using _only_ the previous frame in
order to play the next ok?
So: I want good playback, starting with frame N, would it suffice to
load up frame N-1, and then start playing @ frame N? (or
Hi,
The Album-iD specs are somewhat finished, and I wrote a program to
generate AiD's and stick them in front of mp3's. (all+sources at
http://albumheader.cjb.net)
All seems to work as planned, even LAME decodes the AiD-mp3 without
problems at first sight:
RV "Club System - Volume 16.mp3"
Hello Alberto,
Monday, July 31, 2000, 8:56:53 PM, you wrote:
Now I thought I'd seen it all, it comes to my attention cue files, so
I assume also CD's, use a 'frame' as base unit, being 1/75 s. Anyone
heard of this, or did everyone mistook the xx:yy:zz for
min:sec:sec/100?
AG
Hello Stephan,
Wednesday, August 02, 2000, 1:06:26 AM, you wrote:
In 3.86a it seems, that this feature is always switched on.
I'am using:
-mj --vbr-old -V1 -b128 -F
Adding -h has no influence.
With the older 3.85,
-mj -V1 -b128 -F
gives better results as
-mj -V1 -b128 -F -q1.
SE
Hello Stephan,
Wednesday, August 02, 2000, 12:23:01 PM, you wrote:
SE No! I realy mean 3.85 -q1. And it´s very audible.
SE I will give some samples, if anyone tells me where I can upload them. (never
SE done before)
thanks! would be nice. if you cannot find a place to upload
them/email to,
Hello,
normally I got 'm from:
ftp://cedric.vabo.cz/pub/linux/apps/lame/
but last few days the site was down for me, and today I find a 46 byte
file.
I also know of that overnight CVS archive, but I cannot do anything
with that because it's filled with ",v" files.
any other alternative
Hello Stephan,
SE Yes! This is also my experience. Especially when the Wav-File contains a
SE lot of noise (such as copys from cassetes)
SE No! I realy mean 3.85 -q1. And it´s very audible.
SE I will give some samples, if anyone tells me where I can upload them. (never
SE done before)
for
Hello Francois,
Friday, August 04, 2000, 1:22:51 AM, you wrote:
FdT I am looking for a program to check the integrity of mp3's. I know that Nero
Burning Rom does some sort of check on mp3's before burning them, but I haven't been
able to find a similar utility
FdT to check my MP3's without
Hello alex,
abcc I feel guilty using a list mainly devoted to an open source codec (LAME) to
abcc further the development of ClearBand's 'proprietary' codec. (Is a standards
abcc based codec implementation proprietary? We don't sell the codec - we sell a
abcc multicast system, mostly to ISPs
Hello Frank,
Saturday, August 05, 2000, 1:11:36 PM, you wrote:
FK I have a set of WAV files without silence gaps. After converting to MP3 and
reconverting to
FK WAV there are gaps (20...40 ms) between the files. How can I prevent this?
http://albumid.cjb.net has exactly that in mind.
--
Hello Mark,
Sunday, August 06, 2000, 11:34:04 PM, you wrote:
MT I tend to agree with this, and I think we should disable
MT scalefac_scale for now (it can still be enabled with -q1
MT for testing)
after some re-consideration this seems wisest imo too. after some
reports of -q1 producing poorer
Hello Mark,
MT Naoki's latest work makes a significant improvement to the psycho
MT acoustics, so you might want to try it with --nspsytune.
MT (scalefac_scale can still be enabled with -q1). Some of the stuff in
MT --nspsytune will make it into the default settings soon.
MT Mark
[ALL using
Hello,
RV finding: "--nspsytune" sounds _a lot_ worse than the normal psymodel.
RV The graphs show a lower overall distortion amplitude, but there is
RV this noise that I can even clearly hear upto V1 (didn't test V0).
I triple-checked this. Remember those noise graphs I made
(original-decoded
Hello Robert,
Monday, August 07, 2000, 6:47:16 PM, you wrote:
In order to get the (much) higher --nspsytune filesizes down, I used
"--athlower -21" (or -20--23) to compensate. [seems to go negative
:)]
RH I'm sorry to say, but in my opinion it is a really bad idea to lower the file
RH size
Hi,
The author of CoolPlayer (win32) is considering to put in mpg123
instead of the xaudio decoder. http://www.daansystems.com/coolplayer/
Downside is that it does not decode L1 and L2? Why this restriction?
thanks
--
Best regards,
Roelmailto:[EMAIL PROTECTED]
Hi,
The author of CoolPlayer (free,win32) is considering to put in mpg123
instead of the xaudio decoder. http://www.daansystems.com/coolplayer/
The only reason that holds him back is that the GPL.txt says he should
include his source code with the program.
Is there anyone who actually cares
Hello Monty,
Thursday, August 10, 2000, 9:48:27 AM, you wrote:
The only reason that holds him back is that the GPL.txt says he should
include his source code with the program.
Is there anyone who actually cares about this?
M Yes, a *very big* yes. He will incur serious wrath from the
Hello Cavallo,
Thursday, August 10, 2000, 8:40:19 PM, you wrote:
CdC i found this on the net
CdC http://www.emagic.de/english/products/software/zap.html
CdC what do u think about?
go for the bestfree lossless audio compressor:
http://www.monkeysaudio.com/
--
Best regards,
Roel
Hi,
I don't know how hard it is to get lame converted to a win32 codec,
but I think it would have it's uses.
Any win32 people wanting to give it a shot?
I first was recalcitrant to the idea, but this mail has shown me a
different perspective (it's like the 10th I get of these):
ZM Hi,
ZM
ZM
Hello Rob,
Wednesday, August 16, 2000, 6:03:28 AM, you wrote:
RL I'd also appreciate feedback. Are the results easy to understand? Is there any
RL information that could be added to supplement the results? Are there any other
RL relevant links to related information?
Xaudio 1.3.1 [x86] seems
Hello Rob,
Thursday, August 17, 2000, 12:54:08 AM, you wrote:
RL If you can point me to a specific implementation I can try to test it
RL directly. The only requirement I have is that the implementation support some
RL way of saving the decoded output to a file (e.g. WAV).
Would be great:
Hello Gabriel,
Thursday, August 17, 2000, 5:59:46 PM, you wrote:
RL If you can point me to a specific implementation I can try to test it
RL directly. The only requirement I have is that the implementation
GB support some
RL way of saving the decoded output to a file (e.g. WAV).
Would be
Hello Rob,
Friday, August 18, 2000, 4:07:52 AM, you wrote:
RL If you can point me to a specific implementation I can try to test it
RL directly. The only requirement I have is that the implementation support
RL some way of saving the decoded output to a file (e.g. WAV).
Would be great:
Hello,
I don't know if any, and if, how much JS (something like the choice
between MS or S) can be tweaked.
Take this for example:
http://r3mix.50g.com/velvet.zip (1900kB)
I was doing testing for Qdesign, and this was the conclusion:
1) FHG hq 192S : terrible, high squeeks in R channel
2)
Hello,
I don't know if any, and if, how much JS (something like the choice
between MS or S) can be tweaked.
Take this for example:
http://r3mix.50g.com/velvet.zip (1900kB)
I was doing testing for Qdesign, and this was the conclusion:
1) FHG hq 192S : terrible, high squeeks in R channel
2)
Hello Mathew,
Monday, August 21, 2000, 11:38:55 AM, you wrote:
MH Which version of the QDesign coder is this? With the last version of MVP I
MH tried, "fast" and "high quality" modes produced identical output for both
MH MP2 and MP3.
MVP 2.5*. All verified by David McIntosh (employee of QD),
Hi,
After a whole lot of testing and listening it came to me: "-mj nor
-ms" are optimal quality-wise.
* -ms unnecesarely wastes bits most of the time
* -mj has M/S making too much unnecesary mistakes:
If I understand correctly, the "-mj" is evaluating if a frame
qualifies for M/S coding
Hello Gabriel,
Tuesday, August 22, 2000, 12:43:07 PM, you wrote:
GB First, please note that it has been a long time I didn't really looked
GB inside of the Lame code, so I'll perhaps tell a few wrong statements. (btw,
GB please could anyone explain me when to use the word "tell" and when
Hello Gabriel,
Tuesday, August 22, 2000, 3:59:45 PM, you wrote:
GB I'm not equiped for listening tests here (only an awe64)
GB , but is the velvet
GB problem the thing I'm hearing in the right channel? (or am I thinking I'm
GB hearing something?)
I have $25 sb128 pci and $38 Sennheiser HD-490.
Hello Mark,
Tuesday, August 22, 2000, 9:18:22 PM, you wrote:
MT Problem is, this is a lot of work and it is not clear that it would
MT really improve things.
does it mean anthing if I say it will? :)
MT The hard part is how do you tell if M/S gives
MT better results than S? The only way is by
Hello alexram,
amr I have huge Different between Orginal Wav and MP3.
what do you mean by that? I'm listening to the 4 clips pasted
together, and I cannot really keep them apart. (been looping for 15
minutes now, nice groove) All four filled with noise.
what's the "huge difference"? in
Hello Holger,
Thursday, August 31, 2000, 11:32:13 AM, you wrote:
HD just a quick note that I've release final version 1.1.0 of RazorLame,
HD available from http://www.dors.de/razorlame/
I tested the 1.1.0 yesterday, so I think it's still the beta, but it
works like a charm for me.
If I had to
Hello Holger,
Thursday, August 31, 2000, 2:31:48 PM, you wrote:
HD It's hard to come up with a good option dialog for RL, given the vast
HD amount of options LAME has to offer. I think it's a good idea to put
HD those settings in categories, however, I agree, it can get a bit
HD unclearly...
I
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
Hello Frank,
Sunday, September 10, 2000, 6:39:22 PM, you wrote:
FK But there are no controls to affect the switching more sensitively.
FK So, for instance, a switch can be added to set a penalty bitrate for
FK the MS coding theme:
FK -mS 10 use MS coding if it saves 10 kbps
FK -mS 20 use MS
Hello Frank,
Sunday, September 10, 2000, 11:38:40 PM, you wrote:
FK This model is much better than the hard switch to forced LR frames.
I agree that JS has benefits. Big issue was what your criterium will
be.
FK Currently there is a lot of music out there where 160 kbps with default
FK
Hello Gabriel,
Monday, September 11, 2000, 5:43:55 PM, you wrote:
I am fundamentally agains crippling an encoder to fit the needs of an
inept decoder. If 320 is chosen by LAME on -V1, it is there for a
reason!
GB This point is debatable.
GB I am in the clan of the people using -B 256, and
The best thing so far about Ogg Vorbis has been the marketing.
I tested one file with the new b2, and even with -m6, the best possible
Vorbis setting, resulting in a 350kbit/s file it sounds poor.
There are obvious low-frequency bass distortions, which mp3 at
256kbit/s bitrate doesn't show.
M Anyway, he apologized, I apologize, we were on the bad crack today. We can
M finish this up over some beer or good single malt.
I feel this 'urge' to let everyone on this list know that I think beer
is a poor man's drink. ;))
Roel - the best barfight in the northern hemisphere
[curious
Hello,
first some data: (-V1 -mj -h -b128 -q1)
3.86 (nommx)
Encoding c.wav to c-386-nommx.mp3
Encoding as 44.1 kHz VBR(q=1) j-stereo MPEG1 LayerIII ( 6.0x estimated) qval=1
Frame | CPU/estimated | time/estimated | play/CPU | ETA
12498/ 12498(100%)| 0:04:07/ 0:04:07|
Hello Mark,
Thursday, September 28, 2000, 12:27:02 PM, you wrote:
MP On Thu, 28 Sep 2000, Gabriel Bouvigne wrote:
# of S frames/ total # of M/S frames]. Room enough on the lines :)
4) Why does the MMX mode and non-MMX mode give different output on my
Cel450/Win95OSR2? Isn't MMX
Hello Dmitry,
Thursday, September 28, 2000, 1:34:50 PM, you wrote:
D nommx version was compiled with ic 4.5 (with project files)
D mmx version was compiled with makefile.msvc (ic4.5)
D may be here is the problem
since the nommx ic version seems to output abberant data, could you
please try and
Hello Robert,
Thursday, September 28, 2000, 8:12:59 PM, you wrote:
RH Dmitry schrieb am Don, 28 Sep 2000:
but what version i have to upload???
from project file or from makefile???
with 'Robert's alternate code' enable or disable???
RH Well, officially the one with 'Robert's
Hello David,
Saturday, September 30, 2000, 6:25:31 PM, you wrote:
DB I have a couple of MP3s which I encoded with LAME a long time ago. I made
DB the mistake of putting them on a Zip disk. Now they've got errors in them
DB which they didn't have in the first place, but Nero 4 is refusing to put
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 my
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 -q1
Hello Gargos,
Friday, October 06, 2000, 2:13:24 AM, you wrote:
GC Hello,
GC Roel, maybe you should give these settings a try on that track:
GC -V1 -mj -b128 -q2 -d -k --nspsytune --athlower -35 -X3
GC The bitrate stays pretty low (~224kbps) and it sounds very good...
GC almost identical to
Hello Gargos,
Saturday, October 07, 2000, 1:40:57 AM, you wrote:
GC Im not sure which part exactly you mean sounds very poor.
The graphs you provided show a lower noise, this because --nspsytune
probably. It simply sounds poor, really poor. It sounds nothing like
the original on my
Hello Naoki,
Saturday, October 07, 2000, 1:00:29 PM, you wrote:
NS --nspsytune doesn't work correctly if RH extensions are enabled.
I just tried on the version without extensions. I don't understand
the extra benefit of the nspsytune. Please explain to me what flaw needed
fixing in the
NS Perhaps you are hoping joint stereo to be improved, but --nspsytune
NS doesn't improve anything with joint stereo.
NS --nspsytune is intended to improve result of vbrtest.wav and square
NS wave with VBR. You can obtain vbrtest.wav from website of lame. And,
NS --nspsytune improves CBR
Hello Naoki,
Sunday, October 08, 2000, 4:55:50 PM, you wrote:
Roel addendum: nonetheless JS influence or not, "--nspsytune" only has
Roel negative implications on the -V1 (-q1) I use. Perfectly possible 128k
Roel sounds better, just wanted to express that imo -V1 does not.
Roel
Roel I don't
Hi,
with "--mp3input" is the mp3 first completely decoded to raw and then
re-encoded, or is there some frame-reshaping going on?
a pub in the neigbourhood wants to encode 500 albums into both VBR -V1
and 192S. question of course is: is there a difference between:
A)
lame -V1 X.wav
lame
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