[PCWorld]

Post your comments for Better Music, From Your PC, for Free here

[ColoradoUser]

I was not able to find the 2000 ASIO DLL at foobar website. Also, (may be related), not able to find Off-line settings in the Start Menu. Output was not an option in the Preferences menu.

What's up?

[mjd420nova]

I've done a lot of testing of "lossless" compression schemes and not found a one that is truly "lossless" FLAC is one of the better formats, but is not as lossless as some may think. I'll explain the testing criteria I use. Using a dual signal generator, two different signals, both sine waves, one at 1,000 cycles and the second at 600 cycles. I use a frequency counter to calibrate the input frequency and the output from an amplifier to insure that what I get out is exactly the same as I put in. I not only look at the 1,000 and 600 cycle tones but also at the 1,600 and 400 beat note signals. I then record them into the PC using line inputs to a Creative Audigy 2 card with 16 bit PCM, 44KC stereo, .WAV format and save to the hard drive. I then test two ways, one playing back direct from the drive and one by saving to a CD and playing back from there. Then I do the same, but save on the PC using the FLAC format, both to the drive and to a CD. The results were better than most compression formats, signals came out at 1,008 and 591 cycles respectively. The beat notes were closer, at 402 and 1,596 but as you can see, not exact. Now with just two tones this is close but not exact and we all know that music is a very large and complex waveform, as many as a thousand different tones and beat notes all mixed together. Any compression scheme has to discard some of those notes in order to save space and once discarded these notes cannot be replicated when decompressed. The results are very disconcerting if I use a triangle or square wave signal, as these two are ripe with harmonics and beat notes become very, very complex. I also tried a 24 bit sample rate without any improvement in results and the format is not playable on a regular CD player of PC unless equipted with a card with a 24 bit sample rate. To retain the original sound I always use the .WAV format to insure against any loss of content. This will take up more hard drive space but without any loss in content I feel it is worth the added usage. One minute of content takes a little over 10 MB of space using the 16bit sample rate, 24 bit takes 50% more space but maybe desirable for more content but is not playable on a standard CD player.

[noresh07]

a few weeks back i tried the reco jet audio. i read your article and could not resist another try. kept your article open and followed all the steps .quick, clear and clean. thanks. loaded the new foobar and thus far bass sounds and piano solos are great. crisp and clear.
so far i am pleased with the sound. thank you
may peace and harmony surround you

[fastbullet]

A very well written article.
I would dearly love to see more written as well.
Too often, when articles include downloads or suppliments the them they fail to address where or how to place them. You have covered this very well.
As to the subject matter, I'm afraid it is useless to me now because I have dumped Windows and moved to Linux only.
I hope that I am fortunate enough to find Linux technical pieces similar to this one written as thoroughly. So far, I have not.

[wharvey48]

Ditto on all counts! This isn't worth the trouble.

[Paul1957]

I converted a WAV file to FLAC then back to WAV. The original WAV file was bit-for-bit exactly the same as the converted WAV file. This tells me that FLAC is lossless but your FLAC player is not.

[exodus]

Use Winamp.

[irogerk]

New to this, Getting the audio in, are these pgm's required to not have XP degrade?.

Have a creative video editor box, use that? or purchase a new audio card?

[jag32266]

foobar 2000 has a components page-
http://www.foobar200...ents/index.html

that has a list for asio, look for the file half way down that's called "asio support" from the link above
hth

[mikkelbreiler]

[In reply to mjd420nova|~5183]

I read your post with a lot of skeptiscism regarding your testing.

If I was to test lossless formats I'd make a wave file with complex contents, then make a copy of that file in FLAC format, and unpack that again, and compare the unpacked used-to-be-FLAC file with the original file. If no bytes are changed then FLAC is lossless, if on the other hands one bye is changed then FLAC is not lossless. Testing your analog converter on your cd player through your sound card is another excersize and I cannot understand what role it plays in testing two digital formats, WAV and FLAC - other than making the resuts fuzzy and unreliable.

By your account you are comparing digital formats by sampling it over your soundcard. And also you are looking at an analog signal in the digital stream of data, digital 1's and 0's are approximations of the actual analog sound, so I'd expect the results to be near identical, but not identical.

I cannot understand why you evaluate FLAC this way.

[wwsras]

I've used foobar with ASIO since this article was written. This combo has always beaten any other audio player I've tried, hands down. Can't thank you enough for giving me the best audio experience from my pc.

[Nuke61]

Even though it's old, nice article and good tips. I had no idea that XP mucks with audio that much

This post has been edited by Nuke61: 04 March 2012 - 09:07 AM

[mjd420nova]

mikkelbreiler, on 09 September 2008 - 07:57 AM, said:

[In reply to mjd420nova|~5183]

I read your post with a lot of skeptiscism regarding your testing.

If I was to test lossless formats I'd make a wave file with complex contents, then make a copy of that file in FLAC format, and unpack that again, and compare the unpacked used-to-be-FLAC file with the original file. If no bytes are changed then FLAC is lossless, if on the other hands one bye is changed then FLAC is not lossless. Testing your analog converter on your cd player through your sound card is another excersize and I cannot understand what role it plays in testing two digital formats, WAV and FLAC - other than making the resuts fuzzy and unreliable.

By your account you are comparing digital formats by sampling it over your soundcard. And also you are looking at an analog signal in the digital stream of data, digital 1's and 0's are approximations of the actual analog sound, so I'd expect the results to be near identical, but not identical.

I cannot understand why you evaluate FLAC this way.

I can only explain these testing methods as more of a lab experiment. Losses usually occur in the initial sampling, caused by the bandwidth and sampling rates. Anytime limits are placed, content is lost. This was evident when choosing sample rates and comparing to "real life" sample rates. Few people can hear much audio above 14,000 cycles or below 40 cycles. This really makes sample bandwidths a limit that discards much content. Limits are placed by the sound card too, their actual bandwidths are a product of connectors and internal hardware. I used the Creative card as it is a mid-range unit and one of the most prolific. I liken the process to trying to pour a gallon of water into a funnel that only holds a quart at a time. Some water will get spilled and can't be recovered or "synthesized" to make it look like water when the water is sent back through the funnel the other way, you'll still only get a quart back. Analog music is very complex and reproducing it in digital form places limits on how much can be sampled and converted to digital. That first conversion is the beginning and it's downhill from there. A bit like making a copy of a copy of a copy. Each conversion, either from analog to digital or digital to analog involves limits that further narrows the the sampling rates and bandwidth.

[Nuke61]

mjd420nova, on 04 March 2012 - 11:04 AM, said:

mikkelbreiler, on 09 September 2008 - 07:57 AM, said:

I cannot understand why you evaluate FLAC this way.

I can only explain these testing methods as more of a lab experiment. Losses usually occur in the initial sampling, caused by the bandwidth and sampling rates.

What seems to me to be a far easier and more accurate method of determining whether a claimed lossless music format is truly lossless is to use Foobar2000. Create a lossy/lossless version of the original .WAV file, open both in Foobar2000, and select the digital comparison. Foobar2000 will report whether the two versions are bit for bit identical.

Another way is to use Audacity. Open the original in Audacity, open the 2nd in Audacity but create an inverse version, then sum the two versions. If they are identical, the summation will be zero. Flac is indeed lossless, and so is AAC.