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Topic: wav to mp3 lecture files (Read 5178 times) previous topic - next topic
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wav to mp3 lecture files

Hi all!

Okay I have a lot of WAV files that I have taken off my digital audio recorder that I want to covert to mp3.  I first tried downloading lame and typing in lame, "lame -V 2 --vbr-new class.wav class.mp3"  right?  and it shows up like its gonna encode then when i try to listen to the newly created mp3 file its all satic.  what am I doing wrong?  also is there a way that i can have all the wavs in a directory and have it automatically convert them all into mp3s using the vbr - v2 standard preset?  any help would be great!

thank you!

andrew

wav to mp3 lecture files

Reply #1
try the lamedropXP2 from the mp3 section in rarewares

Just drag em and drop...

http://www.rarewares.org/
Do your ever wonder about your soul?
Can it be saved...

wav to mp3 lecture files

Reply #2
~+Oct 17 2005, 01:04 PM--][div class='quotetop']QUOTE (Skylined ~ @ Oct 17 2005, 01:04 PM)[div class='quotemain'][!--QuoteEBegin--]try the lamedropXP2 from the mp3 section in rarewares

Just drag em and drop...

http://www.rarewares.org/
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okay thanks bud but any idea why im getting static?  this is weird.

thanks!

andrew

wav to mp3 lecture files

Reply #3
Quote
okay thanks bud but any idea why im getting static?  this is weird.


I would guess that Lame is for some reason confused about the format of the input wav files. Are they really wav or raw pcm? strereo or mono? 16bit or 8bit or whatnot? sample rate?

maybe you need to tell lame what format of the wav file.
there are switches for lame.exe

run "lame.exe --longhelp"  from command prompt and see the switches.

Edit: ok, here they are:
Code: [Select]
  Input options:
   -r              input is raw pcm
   -x              force byte-swapping of input
   -s sfreq        sampling frequency of input file (kHz) - default 44.1 kHz
   --bitwidth w    input bit width is w (default 16)

wav to mp3 lecture files

Reply #4
If it's a stand-alone digital audio recorder with a "voice recording" mode, it is very likely using some kind of compression in the WAV files, either good ol' ADPCM or one of the speech codecs (ACELP, GSM, etc.).  If that's the case, you'll need to decode the WAVs to standard 16-bit uncompressed WAVs using something like dBPowerAmp or foobar2000, or better yet, use either of the aforementioned programs to go straight to LAME in one step.
"Not sure what the question is, but the answer is probably no."

wav to mp3 lecture files

Reply #5
Quote
If it's a stand-alone digital audio recorder with a "voice recording" mode, it is very likely using some kind of compression in the WAV files, either good ol' ADPCM or one of the speech codecs (ACELP, GSM, etc.).  If that's the case, you'll need to decode the WAVs to standard 16-bit uncompressed WAVs using something like dBPowerAmp or foobar2000, or better yet, use either of the aforementioned programs to go straight to LAME in one step.
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hey yeah its using adpcm, mono, sample size 4 22khz 88brate but im confused about what exactly i type into lame to do it all in one swoop.  lame -r -V 2 --vbr-new class.wav class.mp3 doesn't work.  thanks for your help!

andrew

wav to mp3 lecture files

Reply #6
Quote
hey yeah its using adpcm, mono, sample size 4 22khz 88brate but im confused about what exactly i type into lame to do it all in one swoop.  lame -r -V 2 --vbr-new class.wav class.mp3 doesn't work.  thanks for your help!


As mixminus1 just said, to decode adpcm to wav you must use one of those programs, see his post. (Probably you will also need the appropriate adpcm plugin for decoding.)

If you correctly identified the bit depth as 4 bit and sample rate 22kHz, then the lame switches should be: --bitwidth 4  -s 22.05


/Edit: now that I'm thinking, bit depth 4 bit seems too low, it is probably bit depth of the compressed adpcm and uncompressed wav would be 8 bit?
Anyway, forget about what i said, try what mixminus1 suggests.

wav to mp3 lecture files

Reply #7
Try this:

1) Download and install dBpowerAMP Music Converter.
2) Download and install the mp3 LAME CLI plugin.
3) Select your compressed WAV files in Windows Explorer, right click and select "Convert To".
4) At the top of the dialog box, select "mp3 (lame.exe)" from the drop-down menu.
5) When it asks where the encoder is, browse to the folder where you currently have LAME installed and select lame.exe.
6) Type -V2 --vbr-new into the "command line" box.*
7) Select your output directory option.
8) Click "Convert".

*You can leave it at the sample rate of your WAVs (22.05 kHz) if you want (it will reduce the filesize), but I just did a quick test and the distortion caused by the ADPCM compression caused some noticeable "twinkling" artifacts in the resulting 22.05 kHz MP3.  Adding --resample 44.1 to the command line completely eliminated them.  The ADPCM distortion may or may not be noticeable on your particular WAVs, so may want to try it both with and without the --resample switch - again, not using it and keeping the MP3s at 22.05 kHz will result in smaller files.
"Not sure what the question is, but the answer is probably no."

wav to mp3 lecture files

Reply #8
Here's another option:  instead of --resample 44.1, you could try --lowpass 10.  That did the trick on my test file.
"Not sure what the question is, but the answer is probably no."

wav to mp3 lecture files

Reply #9
okay so i am still really confused.  I tried using db to convert to mp3 and it actually made the mp3 file bigger than the original wav file, how is that possible?  im just looking for a way in LAME to convert it all in one step.  after right clicking on the original wav file that i took straight off the audio recorder the proporties show...

bitrate 88kbs
sample size 4 bit
mono
sample rate 22k
IMA ADPCM

Thanks for your help!

Andrew

wav to mp3 lecture files

Reply #10
Quote
okay so i am still really confused.  I tried using db to convert to mp3 and it actually made the mp3 file bigger than the original wav file, how is that possible?  im just looking for a way in LAME to convert it all in one step.  after right clicking on the original wav file that i took straight off the audio recorder the proporties show...

bitrate 88kbs
sample size 4 bit
mono
sample rate 22k
IMA ADPCM

Thanks for your help!

Andrew
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It is indeed possible that the resulting MP3 may be larger, since you're using VBR @ -V2, which attempts to maintain a target quality setting (in the case of -V2, transparency to the original source) with no regard to bitrate, and the source is a compressed WAV file, which still has to be decompressed to standard 16-bit PCM (dBpowerAMP is doing it internally) in order to be encoded by LAME.  Compared to that *decompressed* file, the -V2 encoded MP3 is quite a bit smaller.

Since mono speech is the source, -V2 is really overkill.  If you're trying to get as "clean" a sound as possible (although with ADPCM, you've got that fuzzy distortion that's always going to be part of the original sound), you might try

-V5 (or even -V8) --vbr-new

(again, I would recommend using the --lowpass switch, at 10 or maybe even lower, say 7).

If just maintaining intelligibility is enough, you can push things down into the 32 kb/sec range, with either

--abr 32 --lowpass 7

or for a "should play back on any MP3 device" setting

-b 32 --resample 22.05 --lowpass 5.5

(To any of the regulars reading this:  Yes, I know this has been covered many times before by others, myself included...it's just a slow day at work.) 
"Not sure what the question is, but the answer is probably no."

wav to mp3 lecture files

Reply #11
okay thank you very much for your replys but db costs money and i want to figure out how to covert it to just .WAV then go to mp3.  the compressed wav is 22mb and the uncompressed is 88mb.  i mean that makes sense then as to why the mp3 file is 32mb i guess with -V2.  So i guess my question is how do I use lame to decode it to Wav then to MP3 in one step, or is that even possible?  what could i use instead of DB thats free if its not possible?

thank you so much!

Andrew

wav to mp3 lecture files

Reply #12
Quote
okay thank you very much for your replys but db costs money
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I was under the impression that only the *integrated* MP3 encoding and the PowerPack DSP features have a registration fee - someone please correct me if I'm wrong on this, but I have never seen the word "shareware" associated with dBpowerAMP Music Converter anywhere...some of the other dBpowerAMP programs, yes, but never MC.

In any event, another option would be [a href="http://www.foobar2000.org]foobar2000[/url].  It is completely freeware, but requires some configuration, specifically, the "Diskwriter" option in the Preferences menu.  It can decompress to standard WAV, as well as going straight to LAME using command-line parameters.

You could also use Winamp's Diskwriter feature to decompress to WAV, and if you install the FileWriter plugin, you can also go straight to LAME with command-line parameters.
"Not sure what the question is, but the answer is probably no."

wav to mp3 lecture files

Reply #13
alright cool thanks i figured out how to get foobar to decompress the wav file to 88mb which is what it should be, but im not exactly sure how you would get it lame to then kick and and compress it.  im looking to get a file size about 1/6 or so of the 88mb file, so maybe around 10-15 mb mp3.  you were saying using -b 32 --resample 22.05 --lowpass 5.5 or something similar but im not exactly sure where to put that so that lame will do it automatically.

thanks for all your help!

andrew

wav to mp3 lecture files

Reply #14
Quote
alright cool thanks i figured out how to get foobar to decompress the wav file to 88mb which is what it should be, but im not exactly sure how you would get it lame to then kick and and compress it.  im looking to get a file size about 1/6 or so of the 88mb file, so maybe around 10-15 mb mp3.  you were saying using -b 32 --resample 22.05 --lowpass 5.5 or something similar but im not exactly sure where to put that so that lame will do it automatically.

thanks for all your help!

andrew
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1) In foobar2000's Preferences, click on "Diskwriter".
2) In the middle of the page, you'll see "Output presets" and a drop-down menu.  Click on the "Add new" button underneath it.
3) Select "Command line encoder" and click OK.  This will open a settings dialog box.
4) In the Encoder field, browse to your lame.exe and select it.  In the Extension field type "mp3", no quotes, of course. In Parameters, type the LAME command line.  For example:

-b 32 --resample 22.05 --lowpass 5.5 - %d

Note that this particular one is for "strict compatibility," i.e. CBR (some hardware MP3 players have less-than-perfect ABR/VBR support), and you'll get better results using one of the --abr or -V command lines from earlier in the thread (or even just using a higher lowpass setting in the above command line, as you'll be able to evaluate its effect on each file).  Just replace the above with the correct parameters, and then always end the line with

- %d

This tells LAME to take its input from stdin (-) and output to a file (%d).
Type a name to describe the profile in the Display Name field.

5) Now, when you right-click on the files in the playlist and select "Convert" -> "Run conversion", you should be able to select your new profile from the drop-down menu.
"Not sure what the question is, but the answer is probably no."