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Topic: -V n (in 3.95.1) (Read 242984 times) previous topic - next topic
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-V n (in 3.95.1)

Reply #75
"(The mere fact that Joint Stereo is used in lossless compression ought to be enough to destroy - in one stroke - the myth that JS "destroys stereo separation")" (cite from http://harmsy.freeuk.com/mostync/) - I like this argument! It's quite cogent  But then I don't understand for what reason most CBR mp3s are made using S mode (even on LAME). And for what reason all other Stereo modes are supported by LAME?

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Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.

CoolEdit can subtact noise, but it's not the same. It uses the noise sample not "as is", but as a kind of "spectre", am I wrong? Or did I miss some function of CoolEdit, or Adobe Audition (for now)?

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Of course they represent real "distortions"

But LAME 3.93 and GoGo didn't produce them... Is this "feature" (new to 3.95 and the following versions) just an improvement - i.e. to give more bits to the "more useful" freqs?

...So, as I understood, the only way for me to get a better quality is to get a better soundsystem and to carry out the ABX test.

By the way, can you suggest how to choose most appropiate sample for tests? I like metal music, and oftenly there are lot of live cymbals & hihats, which sound good, and have much high harmonies...
...Or it just may be any sample where I can hear distortions (or maybe just invent them ?

-V n (in 3.95.1)

Reply #76
"(The mere fact that Joint Stereo is used in lossless compression ought to be enough to destroy - in one stroke - the myth that JS "destroys stereo separation")" (cite from http://harmsy.freeuk.com/mostync/) - I like this argument! It's quite cogent  But then I don't understand for what reason most CBR mp3s are made using S mode (even on LAME).
Because of the fallacy that S mode (or shall we say, LR encoding) is better than JS mode (or shall we say, MS encoding).

And for what reason all other Stereo modes are supported by LAME?
Because in cases where L & R channels are extremely different, MS encoding is not better than LR encoding. But because the S channel is usually allocated less bitdepth, it may cause degradation.

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Acutally CoolEdit can, but it tells you absolutely nothing about how your brain interprets what your ears are hearing.
CoolEdit can subtact noise, but it's not the same. It uses the noise sample not "as is", but as a kind of "spectre", am I wrong? Or did I miss some function of CoolEdit, or Adobe Audition (for now)?
Here's how (IIRC - I don't have Audition on my office computer):
1. Open first wave.
2. Invert.
3. Select all.
4. Copy.
5. Open second wave.
6. Paste mix.

 

-V n (in 3.95.1)

Reply #77
Or did I miss some function of CoolEdit, or Adobe Audition (for now)?
You simply use the Mix Paste function (Ctrl+Shift+V), but again, it tells you nothing about what a lossy file sounds like.  Pepoluan already mentioned this, but you also need to make sure your files are synchronized since lossy files often introduce delay.

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Of course they represent real "distortions"
But LAME 3.93 and GoGo didn't produce them... Is this "feature" (new to 3.95 and the following versions) just an improvement - i.e. to give more bits to the "more useful" freqs?
I'm not sure what you're seeing or what you're not seeing but it is utterly pointless to judge how something sounds based on what it looks like.

The more bits used trying to reproduce high frequencies which you will not be able to hear mean fewer bits that are available to reproduce what it is that you can hear.  Encode something with Blade @ 192 and the same thing with Lame @ 192, I don't care what versions you use; look at them and then listen to them.

...So, as I understood, the only way for me to get a better quality is to get a better soundsystem and to carry out the ABX test.
The ABX test in this case is supposed to help you determine whether you can actually tell the difference between the original source and a lossy version of it.  I think it's commonly accepted that a high-end sound system is not necessary to accomplish this.

By the way, can you suggest how to choose most appropiate sample for tests? I like metal music, and oftenly there are lot of live cymbals & hihats, which sound good, and have much high harmonies...
...Or it just may be any sample where I can hear distortions (or maybe just invent them ?
Just use the music that you ordinarily listen to.  Samples with lots of cymbals and hi-hats are a good way to reveal some of the more easily identifiable artifacts.

Do you have foobar2000 installed on your system with the ABX component?

-V n (in 3.95.1)

Reply #78
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Do you have foobar2000 installed on your system with the ABX component?
No, I downloaded foobar2000, but didn't find any ABX component for it (using search on its site). Is abchr I downloaded a little earlier worse for some reason?

By the way, using abchr, I couldn't hear those artifacts I see... It was only one test as now, but maybe the sample I used wasn't suitable. I'll try another samples and/or another sound system (and/or someone else's ears ).

I also tried to mix paste (thanx Pepoluan and Greynol for that idea ) samples using Adobe Audition, but it was too difficult to guess how great is the "lossy offset", mentioned by Greynol. Simple comparison of files' lenght did not help me. Is there any bright idea how to find out that lossy offset?

As I see, EAC can determine the offset generated by specific codec, but this operation returns an error for some reason (I tried to determine offset for GoGo and two different LAMEs). What can be a reason? I didn't find anything about it at EAC website's FAQ.

And some more questions about notorious spectral analysis. )

1) for two same-sounding (for me ) samples, is the one with "nicer" spectre (i.e. most bordering upon original) a better one (concerning quality)? Or it is no judge of quality even for same-sounding samples?

2) For two samples encoded using the same original, do same-looking graphs signify the same quality? (of course, accurate within my eyes )

And more questions... (It would be great grateful (edited ) if the LAME developer will answer )
1) Why qval=3 is now default (at least, when using -V2 --vbr-new)? As I remember it was qval=2 in older LAME versions...
2) What is --vbr-new?
3) Are there any differences between two spellings: -V 2 (readme file provided with LAME 3.97) and -V2 (found here, on hydrogenaudio.org)?
4) Please tell me, does -k really "destroy the quality of files" (like Firon said)? Almost any LAMEs mp3 graph remains same-looking when using this key, and the only difference is lowpass-filter-collapse dissapperance. Or that means nothing again?..

Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME).
But isn't JS (when using at LAME, on high bitrates) mostly M/S stereo? Or your mean forced M/S, without using L/R Stereo when it's needed?

-V n (in 3.95.1)

Reply #79
I downloaded foobar2000, but didn't find any ABX component for it (using search on its site).
It is included as part of the installation.  I don't remember if it gets selected by default.

Is abchr I downloaded a little earlier worse for some reason?
From what I understand, this test is designed so that people could rank different files rather than determine whether they could tell the difference between two files.

By the way, using abchr, I couldn't hear those artifacts I see... It was only one test as now, but maybe the sample I used wasn't suitable.
It sounds like abchr was able to accomplish the same thing as abx from the point of view that you realized that you couldn't tell the difference.

I also tried to mix paste (thanx Pepoluan and Greynol for that idea ) samples using Adobe Audition, but it was too difficult to guess how great is the "lossy offset", mentioned by Greynol. Simple comparison of files' lenght did not help me. Is there any bright idea how to find out that lossy offset?
You need to switch to Multitrack View and load the two files one under the other, zoom in to a portion of the audio that you can easily distinguish.  Drag one of the files so they line up to the precise sample and then lock them.  Right click on one file and choose edit.  Then select the entire track.  Switch back to multitrack view again.  Right click on the other file and choose edit.  Copy what is currently selected.  Switch back to multitrack view.  Right click on the first file again.  Use the mix-paste function and check that you want to invert both channels.

As I see, EAC can determine the offset generated by specific codec, but this operation returns an error for some reason (I tried to determine offset for GoGo and two different LAMEs). What can be a reason?
It's a pretty outdated and IMO useless feature.  Anyway, I wouldn't bother with this at the moment.

1) for two same-sounding (for me ) samples, is the one with "nicer" spectre (i.e. most bordering upon original) a better one (concerning quality)? Or it is no judge of quality even for same-sounding samples?
What you hear is all that matters.  It is nice to see what lossy codecs throw out but please give up on this method of determining quality.

2) For two samples encoded using the same original, do same-looking graphs signify the same quality? (of course, accurate within my eyes )
Please give up on this method of determing quality.

-V n (in 3.95.1)

Reply #80
Thanx a lot, you all just broke down my hasty and wrong conclusions.

But I still would be gratefull if someone answers my remaining questions.

Quote
1) Why qval=3 is now default (at least, when using -V2 --vbr-new)? As I remember it was qval=2 in older LAME versions...
2) What is --vbr-new?
3) Are there any differences between two spellings: -V 2 (readme file provided with LAME 3.97) and -V2 (found here, on hydrogenaudio.org)?
4) Please tell me, does -k really "destroy the quality of files" (like Firon said)? Almost any LAMEs mp3 graph remains same-looking when using this key, and the only difference is lowpass-filter-collapse dissapperance. Or that means nothing again?..

-V n (in 3.95.1)

Reply #81
2) The new VBR method, faster and reputed several times to be of equal or better quality than the old.
3) No.
4) Well, you're choosing to keep additional frequencies which you probably cannot hear and which the MP3 format has some difficulties encoding. And yes, sight graphs mean very little in the context of audio!

-V n (in 3.95.1)

Reply #82
1 - because it makes no difference for VBR NEW, note: LAME 3.98 qval will default to 0, but it still makes no difference!
2 - it's a different approach for VBR encoding, we started development some years ago, it will be the default VBR mode for LAME 3.98
3 - no, with single letter options taking parameters you can leave the blank out
4 - -k is a shortcut for setting high-/low-pass filters to minimum/maximum. LAME 3.98 will not have this switch anymore, but you can still change lowpass as before.

-V n (in 3.95.1)

Reply #83
Wow, LAME 3.98 should be interesting!  Thanks for the information, robert!

-V n (in 3.95.1)

Reply #84
Thanks Dv1989 and thanks Robert (an answer from LAMEs developer is a nice thing)!

2 - it's a different approach for VBR encoding, we started development some years ago, it will be the default VBR mode for LAME 3.98

But why an old algorythm is default for now? Is the new one still a "flaw design"?

3 - no, with single letter options taking parameters you can leave the blank out

What's about single letter? -V 2 and -V2 is what I'm interested about.  Dv1989 said that's all the same...

-V n (in 3.95.1)

Reply #85
Before 3.97 development started --vbr-new wasn't widely used though it was available before (called 'fast' mode then).
With 3.97 (where I think has been work done with --vbr-new) it was found --vbr-new is of comparable quality to --vbr-old. Sometimes it's better, sometimes it's worse, and in the overall view there seems to be a tendency that is't slightly better. AFAIK we don't have a very solid basis to prefer one over the other qualitywise, but as --vbr-new is faster it is preferable to use --vbr-new.

Within the same version I think it's wise not to change the defaults, and you can use whatever you like by own settings.

3.98 will be a major step ahead, vbr behavior is significantly improving, and --vbr-new is default with 3.98.
lame3995o -Q1.7 --lowpass 17

-V n (in 3.95.1)

Reply #86
But why an old algorythm is default for now? Is the new one still a "flaw design"?

The developers were obviously not confident enough, or did feel that it made sense, to have LAME default to the new mode. As we have just discovered, this has changed by now! Ooh, it's exciting stuff.

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3 - no, with single letter options taking parameters you can leave the blank out

What's about single letter? -V 2 and -V2 is what I'm interested about.  Dv1989 said that's all the same...

Robert meant the same as I: you can leave out the space between the switch (in this case -V) and the parameter (i.e. 2) - and the end result will be the same.

-V n (in 3.95.1)

Reply #87
Thanx for all! I'll try using -V2 --vbr-new, and I think it will be good for me.

-V n (in 3.95.1)

Reply #88
Setting min/max bitrate is not needed in 3.97 when using VBR.
Spectral analysis is virtually useless for measuring sound quality.

Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128 (on some songs - oftenly)... is it good?

-V n (in 3.95.1)

Reply #89
Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128... is it good?

Between the reservoir and complexity of the frames to be encoded, less than 128 kbits can still provide transparency.

-V n (in 3.95.1)

Reply #90
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You are partially wrong.

In low-bitrate, Joint Stereo is really Intensity Stereo. *That* will collapse stereo separation.

Not in LAME though. LAME doesn't have Intensity Stereo (unless it got it recently...), but FhG and probably others do.


Also, M/S stereo will make things sound WORSE. Leave it as joint stereo (it is NOT lossy in LAME).

But isn't JS (when using at LAME, on high bitrates) mostly M/S stereo? Or your mean forced M/S, without using L/R Stereo when it's needed?


I'm sorry. I meant forcing L/R stereo (-m s) for all frames.

Hmm... But when not setting minimum frame bitrate, LAME uses bitrate lower than 128 (on some songs - oftenly)... is it good?

Yes, because it determined that it didn't need 128 for that particular frame to reach the quality level you specified.

-V n (in 3.95.1)

Reply #91
Thanks again  Now I see, LAME is even more clever than I thought a few days ago.
I like it, and I like this forum and its members who helped me so much.