[JahJahBinks]

Just curious, what softwares do you guys use to encode CD tracks into MP3s? How much is the difference from one to another?

 

[halcyon]

The following make a difference when encoding mp3s:

1) The software used (LAME appears to be the best for some time now)

2) The version used (LAME v. 3.92 or 3.90.2 is the best as of now, 3.93 is horribly broken and 3.94 is still being tested)

3) The settings you use to encode with the software (In case of LAME the --alt-preset settings are considered the best at the time of writing for most music. --alt-preset standard will give you an average bit rate of c. 190 kbps. Try --alt-preset extreme if you want more quality and can tolerate the higher bitrate at c. 256 kbps)

4) The extraction or 'ripping' program might also come into play, if you have a bad cd-rom drive or have very scratched disks. The general consensus is that Exact Audio Copy is the best in most situations, if you know how to configure it properly for your drive.

Best regards,
Halcyon

Refs:

LAME Complies for Win32 (do NOT use 3.93 it is broken)
http://mitiok.free.fr/

LAME recommended settings
http://www.hydrogenaudio.org/index.p...ST&f=15&t=203&

Exact Audio Copy home page
http://www.exactaudiocopy.de/

EAC configuration (from Coaster Factory)
http://www.ping.be/satcp/eac03.htm#-

 

[XzeroR3]

halcyon hit this thread dead on. Everything above is the best possible setup for encoding mp3s.

 

[JahJahBinks]

cool, thanks.

 

[JahJahBinks]

One more related question, for decoding MP3, I use WMP. What softwares do you normally use to listen to MP3? and Do they make a difference?

I like to hear the original sound so I don't use any DSP plug-ins that add colors to the sound.

 

[halcyon]

For decoding I use:

Winamp 2.x (not 3.x series as I can't get all plug-ins easily for that without using a converter)

MAD mp3 playback plugin from:
http://www.mars.org/home/rob/proj/mpeg/mad-plugin/

Resampling Wave Output plugin from:
http://www.blorp.com/~peter/zips/out_wave.zip

You have to set these plugins in the Winamp Preferences / Plugins / Input and Output sections respectively for them to be active.

Unless you have a sound card that offers freely selectable sampling rates and does NOT resample everyrything to 48 kHz, I suggest you use the resampling plugin.

It does better job than most sound cards in terms of resampling and you can tailor it to your cards internal working frequency.

However, be sure to test with and without the resampling plugin, because some cards will even resample their internal native sample rates (i.e. you upsample 44.1kHz to 48kHz with the upsampling plugin and your sound card then again resamples without clock synch at 48 kHz causing additional problems).

regards,
Halcyon

 

[Ricky Monk]

The AC'97 standard requires internal resampling to 48kHz if I'm not mistaken... meaning most modern soundcards have this wonderful 'feature'.

 

[halcyon]

Ricky,

you are right.

However, not all cards with AC97 spec resample their 48 kHz input to 48 kHz again.

Those that do, will not produce good enough sound EVEN if the input sample rate is at 48 kHz.

regards,
Halcyon

 

[Ricky Monk]

okay...
On the second point.... I'm not sure that makes sense.. you mean they actually take it and do a D-A-D on it from 48 to 48? Otherwise I would expect it to spit out the same results as there isn't really any interpolation to be odne.

 

[JahJahBinks]

In the Winamp window, just above the volume setting is a little box that displays kHz, the highest it can go is 44 for mine, that does mean it cannot upsample to 48kHz? What should I do? I already downloaded the plug-ins.

Btw, I need to have DirectSound output plug-in (out_ds.dll) in order to play right?

Also what other input plug-ins do I need to keep if I already have the MAD one (just making sure no other is overriding it)

 

[RunsWithScissors]

Listen to Halcyon - he is right on the money on all counts...

An even easier method to decode MP3 to WAV using LAME as the decoder (yes, I mean decoder even though LAME is mostly thought of as only an encoder) using a drag-n-drop app called LameDropXPd which can be found here:

http://www.inf.ufpr.br/%7Erja00/mp3.html

Get the version 1.21 cited here:

LameDropXPd: OggDrop-like mp3 encoder/decoder with lame 3.92 - V.1.21 - 2002-07-08 - ICL6 compile - 448Kb - by John33 - with Ogg-MP3 transcoding

no installation necessary - just drop it on your HD and launch it, then drag MP3 files onto it and it decodes them into WAV and also has support for OGG.

I did notice that the option to select an output directory (other than the same dir where your source MP3 files are located) did not work for me under Win98. Minor problem IMO.

Hope this helps

 

[JahJahBinks]

How do I use LAME to convert a track on CD to mp3 file?

 

[RunsWithScissors]

Follow the steps outlined in this guide:

http://www.chrismyden.com/nuke/modul...&file=painless

It basically walks you through all the SW you will need and how to configure and use it.

Try using the LAME settings "-alt-preset standard" and maybe even "-alt-preset extreme" and see if you can hear any difference.

If you can't hear any difference, just use "-alt-preset standard" which encodes to MP3 using variable bit rate (VBR) and gives you a very good balance of quality to file size. The extreme setting is slightly higher quality and also larger file size.

enjoy!

 

[JahJahBinks]

Quote:

Originally posted by JahJahBinks In the Winamp window, just above the volume setting is a little box that displays kHz, the highest it can go is 44 for mine, that does mean it cannot upsample to 48kHz? What should I do? I already downloaded the plug-ins. Btw, I need to have DirectSound output plug-in (out_ds.dll) in order to play right? Also what other input plug-ins do I need to keep if I already have the MAD one (just making sure no other is overriding it)

Could anyone answer my questions? Thanks

 

[halcyon]

JahJahBinks,

1. Download the resampling plugin I have linked to in my second post

2. Unzip it and put the contents of that zip folder (not the folder itself!) inside the Plugins folder inside your Winamp folder. For example, if you have Winamp installed in c:\Program Files\Winamp, then you should put the file out_wave_ssrc.dll inside the folder c:\Program Files\Winamp\Plugins

3. Restart Winamp

4. Inside Winamp go to: Options / Preferences / Plug-Ins / Output
Double click on WaveOut output xxxx SSRC

5. At the bottom of the new window you can see "Resampling". Select Target Sample Rate: 48000 Hz and Target bits-per sample: 16

6. Press OK. Quit Winamp. STart it again.

7. Play back songs and Winamp will resample them to 48 kHz. NB! Your Winamp screen will STILL say 44.1 kHz or whatever happens to be the sample rate of the files you play.

8. Listen to the sound and make sure no anomalies (clicking) can be heard in the sound. If not, you are upsampling succesfully and *hopefully* bypassing the internal resampling of your sound card


Ricky Monk,

Again, it's not quite as simple as that.

The card will take in 48 kHz sample rate data so it has two ways to deal with it. Either it can try to lock in (say, using VCO or PLL loops) to the incoming data synch and read it as it is.

Or it can resample the incoming data at the interla rate of the sound card (48 kHz). Now, in theory, if the stream sent to the sound card and the sound card itself both had perfectly stable clocks, this resampling to 48 kHz again would result in 1:1 bit perfect data.

However, no clock on earth is perfect and resampling the data to the same sampling rate will result in increased jitter. After this jittery bitstream is fed to the DA converter on the card, the result is poorer sound.

So, to your questions: no, it doesn't do an extra digital to analog conversion in between and YES, resampling to the same sample rate can cause further detorieration of the digital signal AT the DAC unit, if the clocks do not match (and they rarely do).

best regards,
Halcyon

PS One must remember that digital 'bits' are encoded as analog waveforms (electrical signals) that in theory should have no phase modulation in the waveform. However in reality they always have this phase modulation and it is called jitter. If this jitter is not removed before doing the DA conversion, it will cause extra noise in the output (analog) signal.