Hi Naoki.
changes to --nspsytune that improves CBR quality
Would that include Safe-VBR mode as well?
Shawn
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I thought I'd start a thread so we can all find out what sort of use Lame
gets put through. Okay, here's a few options I use-
-b112 --abr 150 --lowpass 10.8 --resample 32 -h -c -d -ms
for CD ripping, recording from TV
(I don't care about really high bandwidth... 24 22k sound strange
with this
Jack wrote-
Someone told me, that using -mj will destroy 3d-suround information.
Are you talking about Dolby Surround signals, or are you talking about a normal
stereo signal that just has a lot of separation between L R? As far as I'm aware,
3d-surround information isn't destroyed, but it
Frank wrote-
What about an option "adjust-level-for-psycho-model", which increases the level for
the threshold computation, so low level music is coded with more bits.
Perhaps the ATH used for masking should be dependent upon the AC volume for each
frame, down to a point (for example, -50dB),
Can I just clarify this with the guys who know the encoder inside-out-?
Should fast mode only be used when normal mode doesn't work at real-time »and« when
lower sample rates (or mono encoding) are not desirable, nor possible?
Shawn
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It's closer to a device driver than a library.
Point being that an ACM component behaves nothing like an executable?
Shawn
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I'm using LAME on Linux, so no lame.exe for me... :) I don't really know anything
about LAME internals... what is an ACM? Or is that some kind of Windows thing?
Yeah, it's Windblowz alright... ACM means Audio Compression Manager. So how do you
run Lame on Linux if there's no executable?
Benjamin Reed wrote-
so I'm forking a single lame --decode and then piping that to muliple encode
processes (to save decoding for each stream).
It's possible to call lame.exe multiple times without having the thing crash.
Therefore it seems strange to me that one requirement for being part of
John HW wrote -
found better quality on complex choral music at 32kHz (with the -k flag added so
filters were turned off) compared to 44.1kHz. Fewer artefacts were audible.
I wonder if that's because it's classical or choral music you're encoding, not
heavy metal or techno. Your music may
I think my headphones have a better frequency response than my parents' hi-fi system
which cost A$2500, incidentally the headphones came free with the system. The hi-fi
system seems to have a very heavy hump around 60Hz (estimated, not measured) in
comparison to the headphones.
I don't think
If mpg123 needs all frames to have CRC, can the VBR header contain a CRC?
Or would the CRC cause problems if it were to be embedded in a VBR header?
Shawn
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Robert wrote-
Now mpg123 works with protected VBR files.
Mark wrote-
This is probabaly the right thing to do, but:
Now, the Xing VBR header is actually a corrupt mp3 frame. Before this
fix, the Xing header was a valid mp3frame which would decode to silence.
So mpg123 now works, but there
I wrote-
I suspected that long blocks would also have better frequency resolution
than short blocks (by a factor of 3).
Monty wrote-
Yep, that's exactly it; better frequency resolution means better energy
compaction of strong tones so that they're generally represented in finer
resolution.
Hi. I have a couple of my own questions about this.
It sounds like what's being referred to as "ABR" would be not only faster,
but more reliable (in quality terms) than the traditional VBR. So what's
the use of traditional VBR now?
Could Lame be changed to set "ABR" quality using the standard
Hi everyone.
Just a quick question...
I know that short blocks require more bits to encode than long blocks, but
is that the only drawback with using short blocks? I suspected that long
blocks would also have better frequency resolution than short blocks (by a
factor of 3).
Thx
Shawn
--
MP3
I'm not quite sure actually, but I think that mp3enc3.1 lets you control more than
that ACM
I'll bet it does, I was actually referring to the command line interface with that
one... AFAIK, MP3Enc only has -no-is -dm. -no-is doesn't tell it what to use,
it only tells it what »not« to
Also, please use a recent version of LAME!
The FDD controller on my computer has burned out or something... I don't have a
connection at home either, so my computer's pretty much isolated until I can get it
fixed.
I'm going into a computer store today or tomorrow to get everything I've
Hi everyone. I've made an observation with the versions of the Fraunhofer ACM codec
that I installed in about January some time Lame 3.62. Yes, they're old versions,
maybe a lot has changed since then, but... this is what I did-
1- Encoded a sample with moderately low stereo separation
"Ampex" wrote-
Is there a program that can do this externally of an encoder? Ie: take an mp3 and
wrap it into a wav?
CDex can do it. It also does ripping/encoding, but I found that adding the WAV
header only works correctly with CDex if you do it separate to the encoding. The
latest version
Steve wrote:
What major problems did you discover? I've been using it and so
far...seemed like it was working pretty wellthough I don't see any clear
way to verify that the command line options are being done correctly..
Try typing this into a batch file-
/¯
echo %1
echo %2
echo %3
Charlton...
Windoze, eh? *grin*
I've tried LameBatch, but it refused to load 24kHz WAVs, it doesn't have the
144kBit/sec option for MPEG-2 encoding. I haven't been able to try the latest version
at home. It *should* be able to load WAVs of *any* sample rate, because, as we all
know, Lame is
Mark wrote-
I get this message trying to play back 320kbs CBR streams too, with or without
"--strictly-enforce-ISO".
So I think it just chokes at 320kbs in general. It had no problem
handling 310kbs free format.
What about 256kbit/sec with 32kHz. That should be over the ISO limitation too.
Scott wrote:
I dont' see why it shouldnt work - you don't need to decode, just unpack the frame
and then repack it.
then Mark wrote:
Easiest solution (but it wont shrink your files): go into each frame and replace the
12 bit "part2_3_length" field for channel 1 (both granules) with 0.
Okay,
Could we add an option to Lame so it fades in/fades out over a transition time of X
seconds if it starts with significantly more volume than analog silence?
If it's just applause that spans the tracks, they could probably be stitched together
using a crossfade after decoding/before burning.
Is there a way to mix a JS MP3 to mono VBR by discarding all side information in M/S
frames, downmixing all L/R frames, re-writing as mono VBR?
I have some MP3s at 44/128/JS 22/56/JS that sound bad, but when I decode them
downmix them in a wave editor, they sound okay, I'd prefer to be
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
Mark wrote:
If no problems show up with 3.80, then in a few weeks I think
we should make another official release, and finally drop all
the dist10 patching stuff!
I think it's great that this could be done, but there must be something I've missed.
My understanding is that a licence would
If you remember my question a while ago about crippling wavs so they sound bad at all
but the highest bitrates, this is a follow-up.
My band is recording its album in the near future. I know MP3 has its limitations, I
thought it would be useful for the band from a business perspective if we
David wrote-
On the site you recommend WinAMP for the listening,
while just recently a winamp bug was discussed on this
list, which add corruptions during playback of mp3s !
The bug is present in winamp 2.62 ( latest as of this writing ),
as reported on this list.
Can anyone say whether it
Shawn Riley wrote:
Can anyone say whether it would be wise to use an earlier version of Nitrane's
decoder (distributed w/ Winamp version 2.1-something ) with Winamp ver2.6-something?
David wrote:
Note that some winamp versions ( early 2.x , I think ) didn't use
Nitrane , but Faunhofer !
I'll
Hey, excellent. Does it resample as well? If so, what sort of job does it do? Not that
it matters... Lame does that anyway. The trouble with EQing to taste is that you need
excellent pitch recognition a lot of guesswork to make it worthwhile if you can't
actually hear what the EQing is going
Thanx, I'll try it out some time I'm out of town at the moment. The only thing I
have to lose is disk space a lot of bad MP3ing habits... :-D So which EQ does it
use? And how does it compare to, for example, standalone CD rippers such as CDex, EAC?
Shawn
At 05:17 PM 5/1/00 +0600, Zia
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
20800Hz to 22050Hz is 101.034 cents (in pitch terms :-) That contributes to about .84%
of the audible spectrum. Whether those frequencies can be heard, or whether they're
even put on CDs is a completely different matter. If they're not going to be heard,
then there's not much point in having
Is 1 kilobit per second supposed to be 1024 bits per second, or 1000 bits per second?
Shawn
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Mark, I've almost done that patch.
Shawn
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This is the patch I wrote to allow the user to set a target compression ratio (rather
than a bitrate) for determining the bitrate stereo mode (regular Stereo vs Joint
Stereo). I thought this would be useful for batch encoding when the source files are a
mixture of mono stereo, /or of varying
Oops, change this-
/* Now, having calculated the new bitrate, we should recalculate the compression
ratio. */
compression_ratio = gfp-out_samplerate*16*gfp-stereo/(1000.0*gfp-brate);
to this-
/* Now, having calculated the new bitrate, we should recalculate the compression
ratio. */
Sony have made a lower bitrate version called ATRAC3 that is available as
software.
http://www.world.sony.com/Electronics/ATRAC3/
It's used by the Sony Vaio portable music player.
According to reviews the encoder uses some encryption so that you can only
play back the files on the pc in which
Yamaha MD multitrack recorders uses slightly modified ATRAC. When you use
MD from SONY, they sound much worse than MD recorded on multitrack from
Yamaha. But standard stereo mode sounds same as SONY.
Since you normally only record one instrument per track on the MD recorders, I'd
expect it to
I just had an idea about command line args.
Would it be feasible to have a command line such as -
x:\xx\lame.exe --compression 6.0 in.wav out.mp3
- to automatically select a stereo mode, bitrate, filter option, etc. based on how
much the file should be compressed? You could just make Lame choose
Oops...
At 09:13 AM 4/10/00 -0400, Greg wrote:
On some samples with mp3, 320k is not transparent to such listeners on any
sound equipment
What about a mono MP3 sample at 320kBit/sec (vs a mono PCM sample at 705.6kBit/sec)? I
guess that defeats the purpose of MP3, but it would be interesting to
This kind of hack shouldn't be encouraged because
...
Okay. It works fine for me... *sigh* I don't think Steve's going to be streaming his
Playstation songs over the internet though... May I ask why you're using/developing
MP3 format if you're so particular about sound quality?
I should
At 04:33 PM 3/31/00 -0700, Mark wrote:
Compressing by a factor of 14.7 is pushing the limits of
mp3. It would sound pretty bad without the lowpass filter.
If the 96kbs/44.1khz really sounds better, I would think the
96kbs/32khz, with the same amount of filtering (--lowpass 12.5) would
sound
Three from me today (!)
I was wondering if it's possible to incorporate Lame into the Windows 95 ACM.
Shawn
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At 11:40 PM 3/29/00 -0700, Mark wrote:
and this will disable mid/side encoding and force
regular stereo encoding. For regular stereo, the L and R
So the encoder would have a cow if it was using forced JS that happened?
Is there a way to optimise the Joint Stereo encoding mode for surround
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
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