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Lame setting suggestion...
MixMan
post Mar 26 2003, 00:06
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I have some tapes with voice recordings on them (ex. humour tapes, jokes). which is the best LAME setting that will produce the maximum quality for that kind of audio? And for tapes with normal music ?
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yourtallness
post Mar 26 2003, 00:22
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I think that for audio containing only speech, u could try a format called speex.
As for music, I would recommend the --alt-presets, but I'll leave it up to someone who
knows better to guide you...


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DickD
post Mar 26 2003, 14:41
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For maximum quality you can use lame --alt-preset standard (or above) just like you do for music at transparent quality.

(Aside from encoding, you obviously need to go about capturing the sound in digital format carefully to preserve quality from analogue sources, preferably using Dolby NR where it was used on the source and taking care to avoid distortion and clipping, and turning off other analogue inputs if they add noise. I've found that digital noise reduction in EAC's WAV editor is also very good).

It will encode everything the ear perceives as well as the MP3 format allows it to.

By the same token, MusePack (.MPC) encoding with mppenc --quality 5 --xlevel (or above) will also achieve maximum quality and transparency and a smaller file size, but obviously it's not an MP3 and not compatible with hardware players.

However, if the speech is relatively easy to encode, especially if it's almost mono, you will probably find it results in very few MP3 frames above 128 kbps, and only uses more for the few tricky frames (e.g. applause). This is because --alt-preset standard uses 128 kbps as a minimum (except for digital silence, which is 32 kbps) to overcome problems with a few tricky music samples. Lowering the minimum bitrate below 128 kbps saves very little in file size on stereo music (partly because spare space at 128 kbps can be used for bit reservoir I guess), so there was little downside to this precaution.

If file size isn't important, don't worry. If it is, either use MusePack, or you could try permitting low bitrates by modifying the --alt-preset command line.

You could try using:

lame --alt-preset standard -b 32

The psymodel ought to still provide enough bits to encode everything audible without wasting too many when it could dip below 128 kbps.

For music, it isn't as well tested as the unmodified --alt-preset standard and doesn't save many bits at all so it's not usually worth risking, but for speech it may be OK.

In a test I just tried on one stereo music file, it just went down from 177.6 kbps to 176.7 kbps by enabling -b 32, and most of the very few frames below 96 kbps came at the end of the file during and after fadeout. It was a tape recording (the track isn't on the CD version of the album) with Dolby B at 44.1 kHz, then 24 dB of noise reduction in EAC's WAV editor based on a noise profile taken from the tape lead-in. It sounds really good from a casual listen, but why take the risk to save just 0.9 kbps over the unmodified preset?

CODE
Encoding as 44.1 kHz VBR(q=2) j-stereo MPEG-1 Layer III (ca. 7.4x) qval=2
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
  8675/8678  (100%)|    2:47/    2:47|    2:47/    2:47|   1.3541x|    0:00
 32 [   1] *
 40 [   0]
 48 [  12] %
 56 [  23] %
 64 [  29] %
 80 [  59] %%
 96 [  78] %%
112 [ 204] %%%%*
128 [ 661] %%%%%%%%%%%%**
160 [3280] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%**
192 [3171] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
224 [ 819] %%%%%%%%%%%%%%%%%
256 [ 255] %%%%%%
320 [  86] %%
average: 176.7 kbps   LR: 8390 (96.68%)   MS: 288 (3.319%)

Writing LAME Tag...done


As an aside, for lower quality (e.g. converting to lower sampling rates) to produce speech for the web in tiny downloads, I've found Ogg Vorbis is excellent (even though it's not tuned for my switches) and produces less audible and tiring artifacts and Dalek noises than most other codecs including voice codecs like Speex that I tried. (For some uses, like VoIP, low latency of Speex is a key advantage, but not for prerecorded speeches on the web). See my signature for more details.
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jmvalin
post Mar 26 2003, 18:55
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QUOTE (MixMan @ Mar 25 2003 - 06:06 PM)
I have some tapes with voice recordings on them (ex. humour tapes, jokes). which is the best LAME setting that will produce the maximum quality for that kind of audio? And for tapes with normal music ?

Of course I'm biased because I'm the author of Speex, but I think by using Speex with VBR, you can probably get the same quality as MP3 at about 1/2 to 1/4 the bit-rate.
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/\/ephaestous
post Mar 26 2003, 19:06
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I think Speex too.

Anyway you should do some processing to the tapes after transfering them to the pc, rmoving hiss, and constant noises that are hard to encode and don't belong to the audio will drop the bitrates.

I REALLY THINK APS FOR SPEECH IS AN OVERKILL.


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mithrandir
post Mar 27 2003, 01:34
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It depends on what you mean by "maximum quality". Maximum quality, by definition, excludes MP3 all together so I guess you mean "as good as I can get".

Assuming you have LAME 3.90.2, try something like --alt-preset 64 (I think that's the lowest bitrate you can use with 3.90.2) for voice recordings. It's not the ideal choice since LAME is not optimized for low-bitrate encodings but you'll probably find it's fine and doesn't take up much space. It's also not "maximum quality" but need voice recordings be high-bitrate? I don't think so.

It's recommended that you use --alt-preset standard for normal music.
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MixMan
post Mar 28 2003, 00:29
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Ok thank you all guys! I'll try those suggestions and choose the best for me.. thanks again..


Edit:
I thought of something else.. when I capture the audio from those tapes, at what frequency value should I set the Cool Edit (or Sound Forge) at? 44kHz, 22kHz ???

This post has been edited by MixMan: Mar 28 2003, 00:34
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Lev
post Mar 28 2003, 10:20
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What frequency lowpass should be used with encodings from tapes?

Whats the actual frequency limit (theoretical and in reality) of a tape?


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Oge_user
post Mar 28 2003, 11:10
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From http://www.tpub.com/neets/book23/101.htm:

QUOTE
The frequency-response specification of a magnetic tape recorder is sometimes called the bandwidth. A typical frequency-response specification might read within + / - 3 db from 100 Hz to 100 kHz at 60 ips. This means the magnetic tape recorder is capable of recording all frequencies between 100 Hz and 100 kHz at 60 inches per second (ips) without varying the output amplitude more than 3 dB.


For the setting to use for tapes with normal music I suggest --alt-preset standard too.

And if you use a noise reduction filter on the music maybe you could also lower the lowpass to avoid artifacts: maybe I'm wrong but I've found that the sound processed with noise reduction filters produce artifacts easily when encoded to mp3.

This post has been edited by Oge_user: Mar 28 2003, 11:11


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jmvalin
post Mar 28 2003, 18:48
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QUOTE (MixMan @ Mar 27 2003 - 06:29 PM)
I thought of something else.. when I capture the audio from those tapes, at what frequency value should I set the Cool Edit (or Sound Forge) at? 44kHz, 22kHz ???

If you're going to use Speex, the recommended sampling rate would be either 16 kHz or 32 kHz.
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