hi

While i did search on this forum (not good enough, i guess), i didnt find a definitive answer.
I would like to get rid of AdvaLame, as Foobar can do it, it seems.

I think ive configured things properly
here

I copied/pasted my command line generated by advalame, for instance this :
-b 128 -m j --vbr-new -V 0 -B 256 -q 0
(256 vbr)
but foobar comes up with an error message :

Error writing to file (Encoder has terminated prematurely with code 1; please re-check parameters) : file://C:\Documents and Settings\Administrator\Desktop\AIRLI04XFX(1).mp3

thanks for the help

The command line is completely wrong. -b is CBR, -V is vbr, you use both in one command line. Start from scratch, i.e. put the lame.exe into the fb2k dir and use the converter preset. If you then choose "Custom" you see the corresponding command line and can make changes.

-b is minimum bitrate and -B is maximum bitrate, so they are specifying -V0 between 128 and 256kbps.

You would use:


However, I would recommend that you simply use the inbuilt foobar LAME preset with the slider set to V0 and the Fast mode checkbox checked, which essentially gives you:


Is there a reason that you are restricting -V0?

@geoslake:

HA recommended LAME settings

Great, thanks a million guys.
My command line included the %d but no "-" prior to it, its all good now.
I added the "noreplaygain" option.
About my settings, 256 max stereo and 128 max for mono (seem to be equivalent, quality wise, right ?) is all I need (its actually more than i need already ).


ehm, because i want best quality ? (not right ?)
Maybe stupid to use this along with my vbr minimal bitrate setting ?

Oops, I was talking crap, sorry.

Just trust the mighty LAME and it's VBR algorithms. If you want to target an approximate average bitrate, there's a table as a sticky thread here...

You'll actually get better quality overall by using the quality settings by themselves, without tinkering. (-V0, -V1, -V2, etc.)

At best, tinkering with the settings will give you the same quality at the expense of greatly decreased efficiency.

In short, it's usually best to stick with the presets for the best quality.

ok, so i should go for :
-V 5 --noreplaygain - %d INSTEAD OF
-b 32 -m j --vbr-new -V 0 -B 128 -q 0 --noreplaygain - %d

-V 0 --noreplaygain - %d INSTEAD OF
-b 128 -m j --vbr-new -V 0 -B 256 -q 0 --noreplaygain - %d

-V 5 - mm --noreplaygain - %d INSTEAD OF
-b 32 -m m --vbr-new -V 0 -B 128 -q 0 --noreplaygain - %d

this last setting is what i use for mono voice (i still need great quality, its for music), and im not 100% sure
-V 5 calls a preset, which is stereo i guess, but i "tell him" to use mono instead...should i rather use something else ? (no preset)

Also theres really no need for joint stereo ? (at the time, i was told it was better)

thanks again

Joint Stereo is used by default so there's no need to explicitly specify it. You also don't need to specify a switch for mono output. Just use the DSP plugin for downmixing to mono if you want mono output. Simply stick with the recommended settings as it will give you best results. Use the table from the wiki to choose a quality setting that best matches your desired average bitrate.

I actually dont wanna downmix to mono, i encode samples which are already mono, so do I still need a switch for that ? (i guess not)
So, for my mono wav to mono mp3 needs, could I simply use :
-V 5 --noreplaygain - %d
(which is about 128 stereo)
for an equivalent to 256 mono ?

thx

edit, ok, after some testings, it seems lame adapts its kbps according to source mono/stereo status.
I mean, if i select "v 5" its about 128kbps for stereo BUT about the half if source is mono...

If you're encoding mono files with joint-stereo, then you will practically have a mono file.

Exactly, because you don't specify a bitrate with the V switch but a quality level. That's why you shouldn't use -b and -B. If you limit the minimum bitrate this can only decrease the efficiency of the encoder because lower bitrates cannot be used where it would not hurt. If you limit the maximum bitrate you can only decrease the quality because the encoder cannot use higher bitrates where it would be necessary.

ok, all good, thank you very much