Lame 3.99.5z, a functional extension
Lame 3.99.5z, a functional extension
Sep 18 2012, 23:06
Joined: 9-October 05
From: Dormagen, Germany
Member No.: 25015
You can download it from here.
Whatís the functional extension?
It offers VBR quality settings -V3+ to -V0+ and -V0+eco (economic version of -V0+).
What are -Vn+ and -V0+eco good for?
They improve pre-echo behavior.
Beyond that, they combine the quality advantages of VBR (regarding pre-echo) with the quality advantages of CBR/ABR (with respect to ringing and other tonal issues).
Lame users can be classified into three categories:
a) Users who donít care about rare quality issues and/or care much about small file size.
The common way for these users to work with Lame is to use -V5, -V4, or similar.
b) Users who donít like to have obvious and especially ugly issues in their music even when theyíre rare but who care about file size as well.
The common way for these users to work with Lame is to use -V2, -V3, or similar.
CBR 192 or similar, or ABR in this bitrate range, is an alternative (but seldom used).
c) Users who want overall transparency or at least a quality which comes close to it, and who donít care much about file size.
The common way for these users to work with Lame is to use -V0, -V1, or similar as a VBR method, or to use CBR 320 or 256. Using very high bitrate ABR is an alternative (seldom used).
For users of group b) and c) -Vn+/-V0+eco offers significant quality advantages:
We have two major issue classes with most of the lossy codecs:
- temporal smearing (pre-echo) issues
- ringing (tremolo) and other tonal issues.
Letís look at the worst samples I know for these classes:
- eig (extremely strong temporal smearing)
- lead-voice (extreme ringing, for instance at sec. 0..2)
With samples like these users of group b) canít be very happy when using -V2 or -V3, because the ringing issues are very obvious and ugly. The temporal smearing of eig is pretty obvious as well, especially around sec. 3. Using CBR/ABR 192 or similar is a good procedure to fight the ringing, but temporal smearing is much worse than with VBR, itís real ugly.
Things donít really change when using slightly increased quality settings.
For users of group c) itís exactly the same thing, with quality requirements and quality received just both on a higher level.
So the traditional way of doing things isnít totally satisfactory.
Users of group b) can use -V3+ or -V2+ (recommended) and get much better results in the overall view.
Users of group c) can use -V1+ or -V0+eco (recommended) or -V0+ (recommended for the paranoid like me) and get transparency or close-to-transparency. Sure itís impossible to prove transparency for the universe of music, but itís true for the samples mentioned. And as these are very outstanding samples within their problem classes and because of the technical details of -Vn+ described below itís plausible that the approach works rather universally.
How is it done?
-Vn+ uses -Vn internally (-V0+eco uses -V0), but the accuracy demands for short blocks are increased. Short blocks are used when the encoder takes care of good temporal resolution. Audio data bitrate is kept rather high also with long blocks which are normally used.
These audio data requirements are helpful for any kind of problem, they are not restricted to ringing or pre-echo issues.
Moreover a strategy is used which is targeting at providing close to maximum possible audio data space for short blocks.
Whatís the price to pay?
Compared to -Vn the increased accuracy demands of -Vn+ raise average bitrate. As -Vn+ is targeting at significant quality improvements compared to -V2 for real bad samples, we need an average bitrate around 200 kbps at least.
-V3+ and -V2+ are designed for users of group b) above, and as such take care of average bitrate not to be much higher than 200 kbps. For my test set of various pop music average bitrate is 205 kbps for -V3+, and 217 kbps for -V2+.
For users of group c) I allow for the full quality resp. average bitrate range mp3 can offer.
-V1+ takes an average bitrate of 257 kbps for my test set, -V0+ takes 317 kbps.
-V0+eco (economic version of -V0+) takes 266 kbps. So -V0+eco comes nearly for free as -V0 takes 260 kbps for my test set.
Unlike versions I published before, mp3packer isnít really needed any more to squeeze the unused bits out of the mp3 file (with the exception of fractional settings like -V0.5+ between -V1+ and -V0+).
mp3packer brings average bitrate down by only 1 kbps maximum for -Vn+ between -V3+ and -V2+, by 1 to 2 kbps for -Vn+ between -V2+ and -V1+, and by 2 kbps for -V0+ and -V0+eco. So I think we can forget about mp3packer with these settings.
sets the minimum audio data bitrate for short blocks to x [kbps] when using -Vn+ or -V0+eco, with x in the range 150..450.
Defaults are 360,370,420,440,440 kbps for -V3+,-V2+,-V1+,-V0+eco,-V0+ resp.
sets the minimum audio data bitrate for long blocks to x [kbps] when using -Vn+ or -V0+eco, with x in the range 110..310.
Defaults are 160,170,215,220,290 kbps for -V3+,-V2+,-V1+,-V0+eco,-V0+ resp.
prints detailed information for each frame (L/R or M/S representation, blocktype of both granules, available audio data bits, audio data bits used, etc.). Works for both -Vn and -Vn+.
lame3100m -V1 --insane-factor 0.75
Nov 1 2012, 16:02
Joined: 23-May 12
Member No.: 100087
I have enjoyed reading this very technical thread even though I am not in any way a developer of codecs. I have a very tangential question that I pose purely as a matter of curiosity.
While reading about these efforts to more effectively resolve "pre-echo", which reduces the transparency of LAME encoded lossy files, by focusing on how Short Blocks are used by the encoder, I found myself thinking through the explanations for why this problem occurs.
Several posters indicated that the encoder (all MP3 encoders?) use Short Blocks for encoding areas of dynamic change in music - sharp attacks, instant volume surges, percussive sounds, etc., and that the LAME encoder does not necessarily do this as efficiently or cleanly as it could, which thereby causes the "pre-echo" and reduces the perceived transparency of the resultant lossy file. Please correct me if my understanding is not correct.
My question is whether the way a musical selection is mastered affects the encoder's performance in this regard. Would a rock musical selection mastered in the "modern" way, e.g. more compressed, possibly brickwalled, with dynamic range of about 6dbs or so, be "easier" for LAME to encode because of the fact that the dynamic changes within that selection have been truncated relative to a more dynamic selection, such as a classical symphony recording (which may have a dynamic range in the 16-18db area)?
If so, is this the reason why most pop, rock, and country music is mastered in this way nowadays, e.g. because it will sound "better" when ripped to a lossy format? (Don't know if AAC has similar issues)
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