Old 21st February 2006, 21:36   #1
testa12
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Convert WAV to MP3

Hi,

I've got a bunch of .WAV files which I'd like to convert to MP3.

Is this possible to achieve in WINAMP or can anyone recommend a decent conversion program?



Many thanks
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Old 21st February 2006, 22:25   #2
gaekwad2
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It's possible in Winamp using either the Transcoder or an mp3 output plugin.

Or you can use dBPowerAMP Music Converter (free mp3 encoding is limited to 30 days (unless you set up lame.exe as 'Generic CLI encoder')), or CDex, or LamedropXPd, or...

Or just use lame.exe (from the 3.97 bundle), put it in a folder with the wavs, then paste the following line into a text editor:
for %%i in (*.wav) do lame -V2 --vbr-new "%%i" "%%~ni.mp3"
save it into the same folder as whatever.bat and double-click on it.

Whatever you like best.
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Old 9th March 2006, 21:03   #3
Hondo
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This is the way I do it.

If you want top quality MP3's from WAV's then this is what I do.

Download RazorLame from this website RazorLame. and install into the folder you want to use.

Download the latest LAME encoder from hereLAME and extract into the RazorLame installation directory.

Load up the program.

Click on the Edit menu and you will see Options and LAME Options. In Options set the LAME.EXE to the razorlame directory.

In LAME Options go to the General tab and set the Output directory to the directory you are sending the mp3s. Then go to Expert and you will see at the bottom the words 'Custom Options' and a box to use 'Only use custom options'. Click on the tick-box and copy in these command line settings.

-V0 --vbr-new -m s --lowpass 22 -b32

------------------------------------------------------------

-V0 is the quality level spanning from v0 to v9 with v2 being acceptable for most people.

--vbr-new of course is the VBR method.

-m s is the mode setting. S is stereo and J is "joint-stereo". Use which one you want.

--lowpass 22 means that the encoder will use 22khz as the maximum frequency during encoding and all other frequencies will be cut off above the limit. This helps to keep the frequency range at maximum spread.

-b is the minimum bit rate that the encoder will use from in it's encoding range from 32kbps to 320kbps. Have a look at this image.


When you are ready to encode, click on the button. Then while encoding, click on the histogram as it encodes, the histogram should look like the image above.

In the image above, the encoded file has frames where it needs 320kbps to maintain quality and 32kbps to maintain quality.

Now if you were encoding in CBR at 256kbps you would have fine quality sounds up to 256 and compromised quality above 256. Also if the song started with several seconds of silence, you would be wasting bits recording at 320 when you could be recording at 32.

Dear world, STOP being afraid to use VBR ;-). You should be using VBR because the goal is to keep the quality at a consistent level throughout the song: a quality that has no audible artifacts to the average listener using relatively good equipment, and still have good file size savings.

This is the holy grail of audio compression, and something that is simply not possible with CBR. With CBR you'll have artifacts that any untrained ear can hear at some point in almost any given song. If you encoded digital silence, you'd still be encoding it with the much used bitrate of 192kbit/s. In Ogg Vorbis you could have encoded the silence at 4kbit/s!.

Fron now on, Good Night and Good Luck.
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Old 9th March 2006, 21:29   #4
gaekwad2
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Dear world, STOP being afraid to use Joint Stereo!

And don't force the encoder to waste bits on frequencies you most likely can't hear anyway.
The standard lowpass of V 0 is 19.5kHz which already higher than necessary for almost everybody (except bats and dogs).

If other words, remove the -m s and --lowpass 22, -b32 isn't necessary anymore either, and it would only save you anything with (almost) mono tracks (and only if you use Joint Stereo).

In practice you'd get the same quality, and more safety against artifacts, at a significantly lower size by using
-V 2 --vbr-new
which is the same as
--alt-preset fast standard
since unlike your command line that setting has been tested and optimized, if those tests had found mode stereo or a higher lowpass to be beneficial the default would have certainly been changed by now.
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Old 9th March 2006, 23:35   #5
benaw
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hey gaekwad just out of intrest what is your music collection ripped into? you give a lot of advise to people here and on hydrogen, i was just wondering hwo your collection is set up do you keep archival and portable files? or just alt preset standard? which every one likes so much.
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Old 9th March 2006, 23:52   #6
gaekwad2
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I'm not even registered at Hydrogenaudio (they don't like gmail and other free mail providers and I'm too lazy/cheap to find one they accept).

My collection is mostly ripped into Musepack quality 5 (standard) which produces about the same quality as preset standard (or Vorbis q6) at an average of 180kbps.

But I'm just in the process of making lossless images and burning them to DVD-R (not enough HD space to keep them on there), then I can simply convert those into any format I want.

And then of course there are those files that... um... I didn't encode myself...
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Old 10th March 2006, 14:11   #7
benaw
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Quote:
Originally posted by gaekwad2
I'm not even registered at Hydrogenaudio
sorry mate, must of confused yours with another username who was also passinoate with simmalar topics. it was an assumption.

i'm really intrested to hear that you are looking into images i very much like the idea of that i was playing around with alchol 120% but havent kept it up i dont know which image format to use with audio i gess it probably comes down to the application you want to use but i havent figured that out yet.

i rip my music into mp3's 320 cbr lame full stero and 128 cbr j-stero for portable

i've had a look a flac a few times but never kept it up gonna have another look with that new winamp plugin that you told us about. very cool just need toput a cd drive into my current set up again.

i havent had any experence with Musepack but it does sound good it doesn't seem as supported as flac and i cant rember if flac is open source as well but i am a little pissed well a bit just stoped off to get some shoes :P

what does "at an average of 180kbps" coz it reads to me that muspack is using abr which seems very strange if it's lossless well not very but still strange.

any way gtg :P
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Old 10th March 2006, 14:22   #8
gaekwad2
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Musepack isn't lossless (maybe you're confusing it with WavPack). But it isn't abr either, the vbr setting at which it's considered transparent produces about 180kbps on average (as opposed to about 200 for lame alt-preset standard).

The lossless images are simply whole CDs ripped to single compressed files with an accompanying cue sheet, both EAC and CDex (1.60 only) can make those. They can be played as separate tracks with in_cue (or foobar).
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Old 10th March 2006, 19:05   #9
benaw
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thankyou gaekwad you have given me some ideas. what you have said was very interesting.

Quote:
Originally posted by gaekwad2
Musepack isn't lossless (maybe you're confusing it with WavPack)
hmm...http://en.wikipedia.org/wiki/Musepack mabe
still i havent used either.

i know what images are i just havent found information on what to save the images as or the cue sheets.

what image file types does winamp support? i havent used images much but am still considering it.

i really like the idea of perfictally copying the whole cd and maintaining quallity.

i have to go to bed it's ten to 7, AM and the night is over.

Last edited by benaw; 10th March 2006 at 19:21.
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Old 11th March 2006, 10:35   #10
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Audacity works great and you can encode MP3 indefinitely, or so it seems. It's what I use anyway. dBpowerAMP seems to cut off the beginning and especially the ends of my songs, even though I've cropped them to get rid of the silence. All you need to do is find the lame_enc.dll file, copy it somewhere safe, then point audacity to it and encode away (check the preferences though to set your bit rate properly as the default is 128. I use 16, but that's all a matter of preference).

void BlueWater() {water.color=blue; while(GameRunning) {if (fox.pos == InBlueWater) {fox.air--; FoxDrown(fox.air);} else {fox.air=1800; fox.flags = WantsToGetWet; } WaitFrames(1); }} // My top favorite thing in 2D Sonic (as C)
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Old 11th March 2006, 12:22   #11
gaekwad2
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Quote:
Originally posted by benaw
i know what images are i just havent found information on what to save the images as or the cue sheets.

what image file types does winamp support? i havent used images much but am still considering it.
They aren't like CD-ROM images, they're simply the whole CD (including the gap before track 1 that always gets cut off otherwise) ripped as one track and compressed to FLAC, WavPack, Monkey's Audio or whatever (you can even use mp3 if you want but then it won't be lossless anymore of course).

Titles and starting times for each track get stored in a cue sheet which is just a text file saved with the extension .cue

If you have in_cue you can then add the .cue file to your playlist and when you start playing it it'll show you the titles,
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Old 10th April 2006, 09:56   #12
sonic-x
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hello, outta curiousity i was experimenting with 2 diff lame_enc.dll encoders, the newest one <july 6 2004 size 163kb>
and an old one from cdex 1.50 <july 19th 2002 size 97.5kb>
i am kinda curious about this conendrum, i encoded a song with both encode dll files using cdex 1.50 encoding mpeg 1L3(mp3)<vbr new, bit min 32 bit max 320 j-stereo vbr 0 (highest)> on a 4 min song and noticed that the older encoder dll had a 2 meg diff (smaller) from the newest lame_enc.dll, i also noticed that in the fluctuation of the kbps from both songs the older one had what seemed a wider range (i could swear it went as low as 32kbps on silence parts and raced to 320 at loud parts) where as the new encoders product song only went as low as 192 kbps and as hig as 320, the purpose of this experiment was to try and find the absolute perfect balance between compress and quality from both encoders, i wanted to try and encode an mp3 with a full range of vbr <32-320> instead of having it as low as 192, high as 320, does anyone have a suggestion with finding an encoder or setting that will use a full range of vbr in mp3? i also hear from some people that vorbis is superior to mp3, i experimented with encoding vbr on that one as well using highest setting, and noticed that the fluctuated was slower or wider between kbps, ^^ if anyone has a suggestion for a lossy encoder that has perfect vbr pitch for every part of an mp3 with a range of 32-320 please let me know ^^
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Old 10th April 2006, 10:49   #13
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You can't directly compare vbr between old and new lame versions.

Old ones use a different vbr algorithm (and psychoacoustic model) and quality scale.

With the new one you could probably use vbr 2, 3 or 4 and get better quality at the same size (or smaller files with the same sound quality), then it would probably use a wider bitrate range as well.
(But due to a particularity of the mp3 design the displayed bitrates aren't the 'really used bitrates' anyway. The encoder may actually use a wider range than displayed.)

Btw, the latest recommended lame version is 3.97 beta 2 from late 2005 which despite being called beta is more tested and safer to use than 3.96.1.

As for Vorbis, it is generally better than mp3 (even more so at low bitrates). You should also make sure you have the latest encoder though (currently aoTuV beta 4.51).
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Old 10th April 2006, 14:23   #14
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so use the latest lame codec around 2-3-4 till i get my size, sounds good, do you know if that's a yes on the fact that vbr is capable utilizing 32-320 evenly on the bit rate scale?

also on the topic of vorbis do i need to use the win32 encoder app aoTuV or can i keep usin cdex and swap vorbis_enc.dll to the newest one? btw thanks for answering so fast i looked all over the net it's hard to find the nominal best ratio for size vs quality for vorbis and mp3 ^^ i'd like to get it meg for min but i also want to elmiminate the chance of artifacts ^^ so many choices... lol
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Old 10th April 2006, 14:47   #15
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Quote:
Originally posted by sonic-x
so use the latest lame codec around 2-3-4 till i get my size, sounds good, do you know if that's a yes on the fact that vbr is capable utilizing 32-320 evenly on the bit rate scale?
Not sure what you mean by evenly, it can use the whole range (unless you limit it) but naturally some bitrates will be used more often than others.
Quote:
Originally posted by sonic-x
also on the topic of vorbis do i need to use the win32 encoder app aoTuV or can i keep usin cdex and swap vorbis_enc.dll to the newest one? btw thanks for answering so fast i looked all over the net it's hard to find the nominal best ratio for size vs quality for vorbis and mp3 ^^ i'd like to get it meg for min but i also want to elmiminate the chance of artifacts ^^ so many choices... lol
You can replace the dll(s), eg. at Rarewares get the 'Ogg Vorbis dlls using aoTuVb4.51 for CDex and WinLame' and unzip them into the CDex directory.
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Old 10th April 2006, 18:01   #16
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ahh lol by evenly i was thinking that the lame encoder was giving undue bit rate according to it's algorithm, as i was comparing the 2 algorithms from the old and new lame_enc.dll, chances are it was using all of the range i just wasn't seeing it displayed on my winamp lol, thanks a bunch with the vorbis, that gives me a head start ^^ hmm if you were to encode the quality on a song lower then highest with mp3 vbr and you liked that song but you wanted to make it average on space what would you suggest?
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