LAME: -V 2 vs. -V 0

The -V 2 and -V 0 presets are popular options when encoding MP3s with LAME 3.97. They replaced '--alt-preset standard' (APS) and '--alt-preset extreme' (APX) respectively, which were used with the earlier versions of LAME.

A few years ago, many people where using APS, notably including #gamemp3s. When v.3.97 was introduced, it seems most of the 'standard' users switched to 'extreme'; some of them instantly, some of them gradually.

A reason why 'standard' was being used instead of the what seemed to be better 'extreme' is because test results show that it is transparent, which means that the majority of people can't discern quality differences between the MP3 and the uncompressed source. Being that the resulting files are smaller than those created with 'extreme', this would then be an advantage because the same sound quality can be stored with less disk space requirement, so less wasted bytes.

Personally, I'm still using -V 2. I can't notice any audible differences between it and -V 0. Plus, I see MP3 as an handy format, meaning that it should be at the best quality while being at the smallest filesize. If I want the best quality only (including the data I can't hear), then I'd use a better-suited codec, like FLAC for example. However, seeing that practically everyone around are now using -V 0, I'm wondering if I missed something that would change my mind about it.

For those who use -V 0, why are you choosing it over -V 2? Do you actually hear any improvement?

For those sticking with -V 2, why aren't you following this new trend?

Also, if anyone have test results which would prove that people can actually discern the audible quality of -V 2 and -V 0, this would be interesting to see.

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VBR V0 maxes out what MP3 can give you. Also V0 is a very tuned compression option, so I use V0 whenever I have to encode something with LAME.
However most of the time I choose FLAC for lossless compression and Ogg Vorbis or AAC for lossy compression. Vorbis and AAC are just superior, because they're based on newer and better coding techniques.

The thing I don't quite understand is why so many people keep encoding with CBR 320. Doesn't make any sense to me when you get the same with fewer bits using VBR V0.

There's nowhere I can't reach.

-V 0 sure has more potential to max out the result, yet 320 CBR would be even better in that case as it's the absolute best possible result for MP3. Going with the logic that -V 0 would be a better choice since you get the same audible quality while reducing the filesize, why wouldn't -V 2 be an even better choice if the huge majority of people couldn't hear any differences compared to -V 0? For the moment, I see the 320 vs. -V 0 usage to be quite similar to -V 0 vs. -V 2 and I'm under the impression many people chose an overkill setting just to feel more secure while not actually noticing any differences.

I'd go all the way with Vorbis if it wasn't for the fact that it is still less widely supported than MP3.

This thing is sticky, and I don't like it. I don't appreciate it.

There is some music that, for whatever reason, does not sound right when ripped to -V 2. I've found it gives some songs a hollow tone. A lot of -V 2 rips sound just fine, but -V 0 rips always sound great. There's barely any increase in file size, so I'm not sure why anyone would prefer -V 2.

I am a dolphin, do you want me on your body?

Do you have any examples?

Well, there's around a 25% increase in filesize. If someone has only 2 or 3 songs on his computer or portable MP3 player, this is indeed not quite noticeable, yet when someone has 100 GB of music (or more!), then it starts to get quite noticeable. Sure, storage devices cost a lot less today, yet is this really a reason for wasting space by using settings which don't produce any audible improvements?

As written on the poster in Mulder's office in the X-Files: "I want to believe," yet I need some proofs that -V 0 is really worth the 25% increase in filesize.

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Here's a track ripped in -V 2.

click

You can hear the static around the edges of the bass.

What kind of toxic man-thing is happening now?

Originally Posted by LiquidAcid

The thing I don't quite understand is why so many people keep encoding with CBR 320. Doesn't make any sense to me when you get the same with fewer bits using VBR V0.

It's because people on Soulseek tend to rip all of their film score CDs in 320, as they think it's the 'best' quality. Just ask anybody about VBR over at the soundtrack blog sites. Nobody would have a damn clue what VBR is and how it compares to constant bitrate encodes.

Anyhow, I used to rip CDs in -V 2 but I made the switch to -V 0 a few months ago. I wish I could say that I use -V 0 because it produces 'better' sound than -V 2, but I can't, because if I wanted 'better' sound I'd just go for lossless. I'd have to say that whichever preset one uses is based on personal preference. I use -V 0 because I want to listen to the best quality possible, while not wasting bits, in terms of mp3 encoding. Like knkwzrd said, -V 2 may sound different from -V 0 but it happens very rarely. I can discern the difference in quality depending on the preset of a hard rock/heavy metal song, but for music that's more softer-sounding than that, I can't hear anything different from the two presets.

FELIPE NO

Originally Posted by Rimo

-V 0 sure has more potential to max out the result, yet 320 CBR would be even better in that case as it's the absolute best possible result for MP3.

No, you're wrong here.
"Best" for lossy encodings is determined by perceived audio quality divided by bitstream size.

Letting the encoding engine work with constant bitrate disables all fine-tuned smart algorithms that are used to allocate bits. You're wasting bits in the stream filled with zero information.

The audio data has certain "flaws" when coming out at the end of the transform coding step. The "quantize" process now allocates needed bits for the results from the transform step. This is done the smart way when VBR is enabled, brute-force when constant bitrate is used.

That's like a factory producing objects of different size but only using only package format to ship the objects and filling the rest with padding material. That's not very efficient.

MP3 is not: The more bits you throw at it, the better it gets. Most people think so, but MP3 is already limited by design. And with frame bitrates produced by V0 encodings you're pushing the technology behind MP3 to it's limits.

I think most people here don't know. But MP3 isn't limited to 320kbit/s CBR. You can e.g. tell LAME to produced freeformat bitstreams. There you can push bitrate up to 640kbit/s.
The problem is that the perceived audio quality won't increase. Again because the certain flaws I was speaking of won't go away just by throwing more and more bits at them.
There are some special test signals that are encoded better when the bitrate is that high, but nothing that appears in regular music.

Problem with freeformat streams is that the MP3 standard doesn't say that hardware devices have to be able to play it. The existance is covered by the standard but you can call a device MP3-capable even if freeformat streams are not supported. A reason why they are so rare.
However the very accurate libMAD decoding engine can playback these streams.

Originally Posted by Rimo

Going with the logic that -V 0 would be a better choice since you get the same audible quality while reducing the filesize, why wouldn't -V 2 be an even better choice if the huge majority of people couldn't hear any differences compared to -V 0?

I don't want to answer this question in that form.
You see, I rip all of my discs in FLAC. In case I want to have something on my portable player I can always re-encode the file to a lossy encoding. As the hardware decoder of portables isn't very accurate, the DAC often is crappy and the standard headphones don't reproduce the sound very well - I can even go below V2, e.g. V4 or even lower. I probably won't notice the degraded audio quality at all.
Encoding quality is just good enough to drive this low-end playback chain.

On the other hand when at home and listening to music through my "good" equipment (AKG k701 dyn. headphones, DIY headphone amp and DIY USB-DAC) I'm not that limited and certain flaws (like ringing artifacts when audience is applauding) are now detectable.

Furthermore I get tired when listening to highly compressed (encoded, not the compression as in loudness war - I also get tired of this one) audio. I can listen much longer when playing from the original disc (or a lossless encoding), also it's more relaxing for me.
There is a lot that's destroyed when doing lossy encodings. Stereo imaging, dynamic range, all sorts of artifacts.

I'm not saying that I can always distinguish between a lossy and a lossless encoding. But there are differences, which are annoying when at perceivable level.

Originally Posted by Rimo

For the moment, I see the 320 vs. -V 0 usage to be quite similar to -V 0 vs. -V 2 and I'm under the impression many people chose an overkill setting just to feel more secure while not actually noticing any differences.

See above. I consider CBR deprecated. It should be removed. The only thing why it's still there: old hardware players that have problems with variable bitrates.

Originally Posted by Rimo

I'd go all the way with Vorbis if it wasn't for the fact that it is still less widely supported than MP3.

The fascinating thing is that it's quite easy to implement it on embedded hardware (there is the integer-only imlementation vorbis-tremor, which should work on most ARM processors).
I think it's more a political thing...

What, you don't want my bikini-clad body?

Originally Posted by knkwzrd

You can hear the static around the edges of the bass.

I'd be extremely surprised if the fact -V 2 was used is really the reason why there's static in this tune. To me, it sounds more like the track was badly mastered/equalized and it clipped, and then the volume was brought back down, but still showing the damage. I thought the effect of heavy codec compression usually produce a less punchy sound, kind of muted, yet not static noise.

Do you have the same track encoded with -V 0 to compare?

Quote:

I'd have to say that whichever preset one uses is based on personal preference. I use -V 0 because I want to listen to the best quality possible, while not wasting bits, in terms of mp3 encoding.

Indeed. If it was just for my personal listening, I'd stick with -V 2. However, I actually share files I encode myself over the net and am concerned with others' satisfaction. If almost everybody now uses -V 0, I guess there's a good reason behind it and I'd be ready to make the switch as well, yet the reason still seems nebulous. The sound system I use might not be good enough to pick up the quality difference, although it's not a cheap $20 set of PC speakers. The day I'll upgrade it, this might enable me to hear what some claim exists, so I might as well start using -V 0 today. But, what if all of them are wrong and it's only a psychological illusion and an influence of the mass? I wouldn't want to encourage this.

How ya doing, buddy?

I should add that there is some tool around (python based I think, search hydrogen audio forums for it) that can transcode VBR to CBR and vice versa.

VBR to CBR is easy, just use the biggest package format ever used in the bitstream (see the factory example above). The other way is a bit more complicated, I think the author has some information about it in the thread. Sourcecode is also open AFAIK.

There's nowhere I can't reach.

I don't have the same track encoded with -V 0, but I have listened to the same album on CD, and the sound is flawless. It's not the mastering.

This thing is sticky, and I don't like it. I don't appreciate it.

Originally Posted by LiquidAcid

"Best" for lossy encodings is determined by perceived audio quality divided by bitstream size.

...

MP3 is already limited by design. And with frame bitrates produced by V0 encodings you're pushing the technology behind MP3 to it's limits.

I agree with this definition of "best". It is also a fact that MP3 is not the best codec around. Yet, is -V 0 really achieving the "best" result for this codec or wouldn't it be a little overkill like 320 CBR is to a greater extent, actually going over the limits of MP3 without any significant improvement?

Originally Posted by knkwzrd

I don't have the same track encoded with -V 0, but I have listened to the same album on CD, and the sound is flawless. It's not the mastering.

Hmm, well it's hard to prove -V 2 is really the cause then. There could have been several types of manipulations of the file (gain adjustement, transcoding, etc.) which could have generated this quality loss. The best way to compare the settings is to start from a clean lossless source, then encode it with -V 2 and -V 0 and then listen to them (ideally while doing a blind test).

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It's a proper -V 2 rip with log/cue files. I don't see how it can be anything but the encode.

It's not like this is the only track this has ever happened to, either.

I was speaking idiomatically.

Originally Posted by Rimo

I agree with this definition of "best". It is also a fact that MP3 is not the best codec around. Yet, is -V 0 really achieving the "best" result for this codec or wouldn't it be a little overkill like 320 CBR is to a greater extent, actually going over the limits of MP3 without any significant improvement?

If you want to increase efficiency you have to lift the freeformat limitation. Means the standard had to demand freeformat compatibility for all playback devices. That's not going to happen.
This way you're limited to a maximum frame bitrate of 320kbit/s when VBR V0-encoding, IF that amount of bits is really needed to encode the informaton.

You see, VBR V0 is the end of MP3. You can't get more quality without rewriting the standard. At that's not going to happen.

What kind of toxic man-thing is happening now?

Originally Posted by knkwzrd

It's a proper -V 2 rip with log/cue files. I don't see how it can be anything but the encode.

It's not like this is the only track this has ever happened to, either.

Not that I don't want to believe you, but to be certain, I'd need to start from the same source and then compare the two encodes produced. If I'd hear static in one and not the other, then this would be a sufficient proof.

Originally Posted by LiquidAcid

If you want to increase efficiency you have to lift the freeformat limitation. Means the standard had to demand freeformat compatibility for all playback devices. That's not going to happen.
This way you're limited to a maximum frame bitrate of 320kbit/s when VBR V0-encoding, IF that amount of bits is really needed to encode the informaton.

You see, VBR V0 is the end of MP3. You can't get more quality without rewriting the standard. At that's not going to happen.

I'm not implying the >320kbps bitrates since they aren't officially part of the standard MP3 format as you mentioned. What I meant is that if 320 CBR is overkill (adding padding material) to achieve the "best" result, why wouldn't -V 0 also do the same compared to -V 2, if both would be transparent to the listener?

I guess that if I could see test results confirming that -V 0 is "better" (perceived quality vs. filesize), then I would be convinced of its true "betterness".

FELIPE NO

I use 3.98 beta 5, which is considered the best current version of LAME (3.98 b6 was not so good, introduced many bugs).

I use the optimised LAMEDropXpd build. It should be recommended to newbies, it's easy to configure and use (and easy to tell people how to configure and use). I don't see any reason to keep the old dos command line programs (or at least, to recommend them).

I have no idea what "V" quality I use, but I use VBR with an average bitrate of 224 kbps, which is "quality 90" in LAMEDrop.


Using VBR of lower than 192 increases too many artifacts in too many cases, so I don't use it.
On the other hand, the vast majority of people can't hear a difference between 192-320, so it's not worth it to go the whole hog. if you're complaining about artifacts at those bitrates, you shouldn't be using MP3 as a format, you should be using lossless.

Ogg Vorbis: I used to be a major fan, but since it's not supported and very similar to MP3 in terms of quality (although I like the filesize), I only include Ogg downloads in a ZIP at the bottom of soundtrack pages on my game music website.

If I'm going to use lossy, I'll use high quality MP3, otherwise I'll listen to original WAV's on CD or PC.
I strongly feel people talking about differences between 192-320, or 320 CBR, etc, are msising the point.

320 CBR: Waste of filespace. Good VBR settings with a good encoder (nothing before LAME 3.97), with an 192 average bitrate or better, will produce a quicker encoded, audibly the same quality, MP3 file. There's simply no reason to use 320 CBR- if you like good quality audio and are pretty set on it, don't use MP3, use lossless codecs.

MP3 at high bitrate VBR (192-320) will produce very acceptable quality files for common listening.

- Spike

What, you don't want my bikini-clad body?

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Jam it back in, in the dark.

Rimo, you bring up a very good question. And I agree that you would probably be well-served sticking to -V2. It is my view that "upgrading" to -V0 does little more than offer the comfort that your audio is in higher bitrate, consuming that additional space without sounding significantly better.

I want to know, honestly, how many people discussing this have tried a scientific ABX listening test to evaluate what settings work for them? (ABX test software is easily available, via a component in the Foobar2000 audio player.) Because when differences in sensory input become small, such as when comparing high-quality encoding modes, subconscious psychological biases can easily distort one's perception. Only double-blind tests eliminate this strong suggestive bias. They do this by removing the listener's ability to know which sample is which, such as knowing that a certain sample is encoded at a higher bitrate, knowledge that often distorts the listener's perception such that they falsely "hear" it as being better because they expect it to be better.

To illustrate my point, recently I posted on OverClocked ReMix about how the size-restricted encode of Valse Aeris on the site had very noticeably jarring artifacts because it was encoded with an average bitrate of 103Kbps. This was while listening on my Sennheiser HD 280 Pro headcans. After people disagreed with my assessment of it sounding so horrible, I tried listening to it on my Klipsch ProMedia 2.1 speakers. The difference I heard clearly on my headphones I actually couldn't hear AT ALL on my speakers, even at a bitrate as low as 103Kbps!

So while I could easily ABX the difference between Valse Aeris encodes 12 out of 12 times on my headphones, I gave up trying to ABX the difference on my speakers after a few trials when the program basically said I could toss a coin and get the same results.

Before going on, I would like to note that I was ABX comparing the 103Kbps ABR encode against a -V2 --vbr-new encode of about 198Kbps.

After noticing that the bitrate could actually be bumped up from 103Kbps and still fit OverClocked ReMix's size limit, I fooled around and was able to get it to an average bitrate of 114Kbps staying within the limit. And the most surprising thing for me was that with only this 11Kbps average increase, the audio artifacts that were formerly so very audible and evident on my headphones became surprisingly subtle! And yet -V2, the command line I live by, encoded it an entire 95Kbps higher. With that drastic a decrease in flaws with only an 11Kbps increase, I somehow doubt that I would need a mode as high as -V2, with its much more drastic increase, to get transparent sound.

I also fooled around with -V6 while trying to fit Valse Aeris within the filesize limit. While it actually went over the limit (with a bitrate of 122Kbps), I noticed that it seemed to sound even better than the 114Kbps ABR (not surprising), but while I thought that I could still hear some subtle audio artifacts, I knew it would start to be a challenge to actually ABX test them at this point. And remember that was using just -V6, on my hi-fi headphones! And I couldn't hear the difference at all on my mid-level speakers!

Looking at the Hydrogen Audio wiki for LAME encoding, notice that it specifies -V3 through -V0 as "High quality: HiFi, home or quiet listening", and additionally states that "-V4 --vbr-new should be close to perceptual transparency". After my experiences with Valse Aeris, I suspect that this wiki entry is quite close to the truth. -V2 ought to be more than fine for encoding all your music, and if one is interested in greater space efficiency, -V3 and -V4 may be worth testing out.

(Note, that if you ABX test your own audio, you must know what to look for. Percussion that is high in the mix, especially cymbals and casanets, tends to artifact. And there are several testers and devs over at Hydrogen Audio that can tell you what other "problem" samples to look out for; just make sure you're talking to someone who actually knows their stuff.)

Also, I'd like to correct a few statements I saw above:

There is no reason to think that -V0 is the most you can get out of MP3, nor that -V0 is more tuned than any other preset. -b 320 will indeed get you the "best quality possible" because -V0 is unlikely to ever result in a 320Kbps MP3, even if there is enough audio information to be encoded to fill all those bits usefully. However, audible differences between 320Kbps and -V0 are probably about as unlikely to occur as those between -V1 and -V0, so this discussion about the "best quality of MP3" is really more academic than practical.

And LAME 3.98 beta 5 is considered the "best" version of LAME, by what authority or consensus? Hydrogen Audio still lists the recommended version as LAME 3.97 stable. Be careful of being on the bleeding edge, using betas and such. While 3.98 beta 6 seems to have some obvious flaws, without prudent testing what's to say that 3.98 beta 5 doesn't have more subtle regressions?

There's nowhere I can't reach.

@Moguta: CBR 320 is wasting bits when compared to VBR V0. That's why it's inferior.

This thing is sticky, and I don't like it. I don't appreciate it.

Glad to see this thread didn't go into oblivion yet!

To sum up the main points:
- Going with a too low bitrate will degrade the audio.
- Going with a too high bitrate will waste storage space.
- To find which bitrate level to use is a personal process and it should go with our own audition/equipment limits.
- In general, high quality VBR is the best way to go.

The problem is: which high quality VBR setting to use? This goes with the fact that MP3s are being shared among people and not everybody have an identical audition/equipment. Personally, I'm not excited about the idea of downloading files encoded at -V 8 and actually consider -V 0 to be slightly too much. Similarly, there are people who frown on anything lower than -V 0.

Limiting the possibilities to -V 2 and up, the difference between these presets are relatively small, yet there doesn't seem to be a general consensus to attest which one is the best to use. However, many people here are currently using -V 0, yet I'm far from convinced they actually hear a quality difference.

I'd be ready to switch to -V 0 if it's a reality that so many people can discern a difference. Yet if it's 1 out of a million who can hear an artifact on a 1 second cymbal sample and the rest are fooled by their mind, is it worth the filesize increase? In the same line of idea, I'd also be ready to use a lower preset, but kind of consider -V 2 to be the standard (I guess the name creates this effect).

I am a dolphin, do you want me on your body?

Originally Posted by LiquidAcid

@Moguta: CBR 320 is wasting bits when compared to VBR V0. That's why it's inferior.

I apologize, I thought when you mentioned "V0 maxes out what MP3 can give you" that you meant it gave the highest possible quality regardless of all else. So what you're really talking about then is bitrate efficiency, or in other words, how you can get the most perceptually-transparent audio with the least bits wasted to inaudible audio.

Along the lines of your own argument, I would propose that -V2 and perhaps even -V3 or -V4 are superior to -V0 because -V0 wastes bits with rarely any additional audible quality.

However, I notice you do mention hearing artifacts in -V2 encodes. Can you provably demonstrate that you hear artifacts in -V2 that you do not hear in -V0, by way of ABX testing? I have never been able to pick up any differences, but I am not arrogant enough to deny you the opportunity to prove that you do hear something that I do not.

Originally Posted by LiquidAcid

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Thanks for that. I was actually thinking of that very document when Spikey mentioned the newer betas of LAME, though I didn't remember where it was. It's nice to see such additional improvements, and although any mentioned quality changes are things that should make it better, unfortunately they may regress in certain cases, notably in the recent beta 6. However, I do see one line that looks promisingly definite -- "Known problem samples for the new VBR code: many of them are at an acceptable quality level now; with a big 'Thank You' to Francis Niechcial" -- because it appears that careful listening tests must have been performed to make this assertion. Also, it's interesting that --vbr-new (now default in 3.98 betas, evidently? does the old mode need --vbr-old or something?) is getting a new psymodel. This seems like the most volatile change, with the potential to incur much improvement or regression.

Originally Posted by Rimo

The problem is: which high quality VBR setting to use? This goes with the fact that MP3s are being shared among people and not everybody have an identical audition/equipment. Personally, I'm not excited about the idea of downloading files encoded at -V 8 and actually consider -V 0 to be slightly too much. Similarly, there are people who frown on anything lower than -V 0.

Indeed. However, I am naturally suspicious of the claims of some people with hi-fi equipment that they can hear the benefit of -V0 with regularity. If anything, the differences should be subtle and only exist with rare problem samples. And problem samples tend to artifact with -V0 as well as lower qualities such as -V2 since such "problem samples" tend to give trouble for the entire encoder or psymodel, rather than only being troublesome for certain quality steps. In such cases, the difference is usually how they artifact, not whether they artifact.

Also, I hope no one would be excited about downloading -V8 encoded music, considering they'd be getting 32KHz-resampled audio at bitrates around 85Kbps. I imagine it's enough to hurt anyone's ears.

Originally Posted by Rimo

Limiting the possibilities to -V 2 and up, the difference between these presets are relatively small, yet there doesn't seem to be a general consensus to attest which one is the best to use. However, many people here are currently using -V 0, yet I'm far from convinced they actually hear a quality difference.

Limiting the possibilities to -V2 and up, I would definitely choose -V2. I submit that extremely few people are actually able to hear the rare, small differences between those ~245Kbps -V0 files and the more space-and-bandwidth-efficient ~190Kbps -V2 files.

For this reason, I really wish that #gamemp3s had not increased their encoding mode to -V0. Not only I am quite hesitant to transcode the files lower to save my own space, at risk of additional degradation, but I think that #gamemp3s' distribution would benefit from smaller torrents that take up less space on users' computers. If I remember correctly, the decision to increase the encoding standard was simply done by putting a poll on the website. It's no surprise that voters wanted the "upgrade", with the common mentality that higher bitrates and higher quality switches must be better than anything lower.

Originally Posted by Rimo

In the same line of idea, I'd also be ready to use a lower preset, but kind of consider -V 2 to be the standard (I guess the name creates this effect).

I plan on personally messing around with -V3 and -V4 to check out how well they hold up with my ears and headphones. But, also, it's prudent to remember that the majority of people (and we're not exactly talking about the demographic that would visit Hydrogen Audio or my ripping guide) don't mind 128Kbps CBR for casual listening. So even if you can begin to hear what's different, perhaps one should question whether those differences will even be enough to be annoying or jarring... especially when considering quality modes just bordering audible transparency.

How ya doing, buddy?

@Moguta: My definition of "wasting bits" isn't the same as yours. You define wasted bits by "not so much" improvement on the perceived audio.

That's not my "wasted bits" definition, which is much more concrete: unused bits in the MPEG bitstream, unused because the framesize is too big for the bits delivered from the transform coding step.

Additional Spam:

Originally Posted by LiquidAcid

@Moguta: My definition of "wasting bits" isn't the same as yours. You define wasted bits by "not so much" improvement on the perceived audio.

That's not my "wasted bits" definition, which is much more concrete: unused bits in the MPEG bitstream, unused because the framesize is too big for the bits delivered from the transform coding step.

EDIT: Oh yes, also it wasn't me stating that I could hear artifacts in V2 which wasn't there with V0.

What kind of toxic man-thing is happening now?

Originally Posted by LiquidAcid

@Moguta: My definition of "wasting bits" isn't the same as yours. You define wasted bits by "not so much" improvement on the perceived audio.

That's not my "wasted bits" definition, which is much more concrete: unused bits in the MPEG bitstream, unused because the framesize is too big for the bits delivered from the transform coding step.

Ah, I see. So you're talking about the fact that it's CBR, and the disadvantages inherent in such an encoding mode.

I agree that CBR is certainly less than optimal. But I don't see how that means that "V0 maxes out what MP3 can give you". There may be some wasted bits, yes, but it would seem that 320Kbps CBR "maxes out" because it can still end up encoding more audio information than a -V0 encode. And not only that, but seeing that most lossless audio tends to have bitrates somewhere around 900Kbps, I can't help but doubt that there are many cases besides digital silence where 320Kbps worth of audio is not present. Of course, I could be wrong, since MP3 does store different information than lossless formats.

But even if I'm wrong about that, I'm not sure that "VBR V0 is the end of MP3" or that "You can't get more quality without rewriting the standard". After all, why continue to develop LAME if no improvements can be made without a format rewrite? And additionally, one can always make a higher VBR preset. For example, a preset could encode everything at 320Kbps except if the frame did not have enough information to do so, so no bits are wasted.

Of course, once we get the idea that "Best for lossy encodings is determined by perceived audio quality divided by bitstream size", it begins to seem apparent that an encoding mode might possibly be too high, even if it doesn't waste bits by the strict definition. And that's why I think that -V0 is an unnecessary ~%25 size increase, just as I'm sure my proposed 320Kbps VBR setting would be overkill.

Originally Posted by LiquidAcid

EDIT: Oh yes, also it wasn't me stating that I could hear artifacts in V2 which wasn't there with V0.

So are you saying that you can't distinguish -V0 from -V2? I thought that you were implying the ability to hear a difference here:

Originally Posted by LiquidAcid

You see, I rip all of my discs in FLAC. In case I want to have something on my portable player I can always re-encode the file to a lossy encoding. As the hardware decoder of portables isn't very accurate, the DAC often is crappy and the standard headphones don't reproduce the sound very well - I can even go below V2, e.g. V4 or even lower. I probably won't notice the degraded audio quality at all.
Encoding quality is just good enough to drive this low-end playback chain.

On the other hand when at home and listening to music through my "good" equipment (AKG k701 dyn. headphones, DIY headphone amp and DIY USB-DAC) I'm not that limited and certain flaws (like ringing artifacts when audience is applauding) are now detectable.

Furthermore I get tired when listening to highly compressed (encoded, not the compression as in loudness war - I also get tired of this one) audio. I can listen much longer when playing from the original disc (or a lossless encoding), also it's more relaxing for me.
There is a lot that's destroyed when doing lossy encodings. Stereo imaging, dynamic range, all sorts of artifacts.

I'm not saying that I can always distinguish between a lossy and a lossless encoding. But there are differences, which are annoying when at perceivable level.

Originally Posted by LiquidAcid

The fascinating thing is that it's quite easy to implement it on embedded hardware (there is the integer-only imlementation vorbis-tremor, which should work on most ARM processors).
I think it's more a political thing...

About portable Vorbis implementation... It's not really so trivial. For every supported format, space must be allocated in a chip's limited ROM space to store the decoding procedures. Also, optimizing the decoding for high performance is critical to both the available processing power of a chip and the battery life of any portable digital player. I know that the first iRiver devices to support Ogg Vorbis could only support files in certain bitrate ranges, or the player would be forced to skip to the next song.

I agree that it would be great if Vorbis had wider hardware adoption. But the ratio of effort to implement vs. consumer demand just seems too out of kilter for it to be worthwhile for manufacturers.

FELIPE NO

Nearly every lossy codec uses transform coding, which is totally different from (linear) prediction coding used by lossless codecs like FLAC and WAVPack.

How ya doing, buddy?

The argument about -V 0 and CBR 320 seems a bit silly to me, so I'm going to stay out of it. I'll just address the original question.

Originally Posted by Rimo

For those who use -V 0, why are you choosing it over -V 2? Do you actually hear any improvement?

I think for a lot of people, or at least for me, we want the standard of "maximum" quality (barring, of course, that the LAME technology is still being developed to this day) without having to attempt to discern whether or not it is actually noticeable.

Let me put it this way: perhaps some specific pieces of audio do lose quality if encoded with -V 2. But, perhaps some don't. Who really wants to encode every MP3 over and over again until you've found the point you can hear the difference at? For one MP3, it may be -V 2, but for another MP3, it may be as low as -V 4. And, VBR tends to be so high quality anyway, that who can honestly tell the difference most of the time?

I don't think it's worth trying to find something that is "probably" just as good as the accepted "best" of today's LAME technology, or using something at the possible risk of it not being better. -V 0 is the accepted best today, so I'm going to use it.

Yet still, it would be a waste to go with the "actual best", which would be CBR 320. Again, no difference can really be heard, and it's a monster when it comes to hard drive size.

This is why I prefer -V 0 personally; it's a guarantee that you are getting today's best, and for not a gross amount more hard drive size. To use CBR 320 would just be... too paranoid.

Jam it back in, in the dark.