Opinion: Apple -- Here to Stay

Do you have any supporting evidence for your claim other than your personal opinion?

I refuse to educate you further on obvious computer concepts and audio compression technology.

So you can't provide any supporting evidence for your claims. Figured so much.

BTW: next time you are in court and the judge asks you to provide evidence for your claims - try your "I refuse to educate you" line

LOL

When you make a claim, the burden of proof is on you.

You are wrong. All compression formats remove data.

Don't get into semantics again. The point is whether an audio stream in a file can be reconstituted to 100% of the original stream -- in other words, retain full audio quality.

The internal format it's stored in doesn't matter, as I can easily write a program to find consecutive alike bits in a WAV file and replace them with a token of which bit it was plus how many bits were replaced. Guess what, I removed data! But I haven't damaged the audio stream since I can go back, find my tokens and replace them with the original data (I've actually somewhat described the LZW compression technique for bitmaps).

You might as well say I've lost data by Zipping a file. That's stupid and irrelevant. All that matters is that I can get my data back 100%.

When does the WMA file get converted to the PCM format?

There isn't necessarily a PCM data stream in between, although that's really the easiest way. Use the system codecs, and just pipe PCM between them for conversion.

Conversion routines work against the compressed file - that is the problem.

I already told you about that. It's easiest to decode-encode. Any algorithms that attempt a direct transcode are mainly applicable to lossy target formats, rying to eliminate some of the loss that would be caused by the recompression by directly translating one compression scheme to another. Obviously, we don't worry about losing quality on recompression if we're recompressing to lossless.

I 'wrote' a small Java app off a tutorial one time that from the outside looked like it converted formats, as you pointed to file of format A and it produced file of format B. But what it actually did under the hood was decode and pipe that bitstream to an encoder.

Interesting concept, can you show me a utility that actually does this (or something similar)?

For educational purposes only. You'd understand why no one would actually do that if you read the post.

I have no idea what perfect pitch would have to do with this but I am a semiprofessional musician and I say that is pure bunk

All I know is that he can tell the difference in quality when he converts his (IIRC 2") newly-created master tapes to digital, and that's not CD-quality digital, a bit higher. He's pretty disappointed by the time it gets to a CD.

I have a home recording studio and when I use near-field monitors I can start to tell the difference between a well ripped compressed file and the original but under most circumstances this is very difficult.

After hundreds of loud concerts and a car accident (those airbags are LOUD!), I don't think I'd even be able to tell that much anymore.

When you make a claim, the burden of proof is on you.

I gave you a link on MPEG-4 audio lossless compression in 1008.

BTW, you did make the claim that converting from WMA to another format always causes loss of audio quality, and have never backed that up. Please do so.

Cool, I got the 2^10 post replying to you!

Sadly, only a true, hopeless geek would catch that and find that milestone more interesting than that 1000th post.

Personally, other stuff aside, if I were you I'd take them to a lossless format and archive them. Which format is, I think, irrelevant, as WMA lossless and Apple ALE will probably both be around for a while. If you archive them lossless, you should be able to rip them to another format with minimal quality loss, and I feel better just knowing that the files are archived on DVD somewhere.

I lost thousands of dollars of old LPs as they aged and scratched, and of course, LPs went out and I gave them away, threw them away, or sold them prior to ripping from a digital source became practical (remember 500mb hard drives? Not much music storage space there). Now, you can throw several hundred on a DVD in a lossless format, and protect your investment. Personally, I prefer Apple's system, but the platform is irrelevant. Just get some lossless backups. Take care.

I have no idea what perfect pitch would have to do with this but I am a semiprofessional musician and I say that is pure bunk (a human can hear better than standard digital sample rates is bunk). Digital music does sound more sterile but that is because of the accuracy of digital recording. Analogue signals are very susceptible to distortion and much of the distortion makes music sound warm - that is why tube equipment is often used - because it is inaccurate but the distortion makes the music sound warmer. There are reasons to dislike digitized music but it is not because your hearing can hear changes that happen between two sample points.

I was interested in this point the other day and asked someone knowledgable about this and got more or less the following response. See what you think.

Humans can't hear above roughly 20K. So sampling at 44K should be sufficient. But here's the rub. Because of the vagaries of the hardware there are frequencies above 20K that will alias to below 20k when sampled. E.g. 30K will alias to 15K. Hardware engineers have learned to analog filter these high frequencies out before the audio is sampled, thus preventing the aliasing. Unfiltered, these aliased frequencies would severely degrade the sound.

Now, if you can sample with roughly double the bit rate (I think 96K is where the industry is now) that aliasing problem goes away therefore the need to pre-filter goes away. So the sound is subjectively better, not because we need to hear frequencies above 20K but because we did not have to filter those frequencies to prevent aliasing. Thus it is the filter that did the damage but we needed the filter because the sampling rate was so low.

Make sense?

Like so many have been trying to tell you people, drinking the Apple Kool-aid effects your ability to reason.

I own four businesses, and am starting a fifth.

I don't know how you live, but I travel between cruises, and my four homes, spread across the US. In October, we will spend a week in a villa, north of Rome, and sail back to the US on a two week trans-Atlantic.

Earlier, in the summer, we will wander in the Southwest, and go to Mexico aboard a Celebrity ship. My businesses all have good people to care for them, so I don't need even sign a check.

With my lowly PowerBook, we can conduct business anywhere in the world. You say that's no problem for a PC guy? I say I can't even tell you how to open the back, and don't really care!

Go pedal your snakeoil and bravo sierra to someone that doesn't know any better, boy! Gwennie is my girl, Georgie probably suits you better...

When you make a claim, the burden of proof is on you.

I gave you a link on MPEG-4 audio lossless compression in 1008.

I was not asking for a link to a description of a lossless format - you claim there is a such thing as a lossless conversion between two different audio compression formats - I am asking you to point me to a lossless format conversion process. I am curious if such a thing actually exists.

Humans can't hear above roughly 20K. So sampling at 44K should be sufficient. But here's the rub. Because of the vagaries of the hardware there are frequencies above 20K that will alias to below 20k when sampled.

The frequency I was speaking of was not audio frequency but the frequency of samples. Digital recording takes a "picture" (sample) of the sound at a certain frequency and then represents this "picture" numerically. There are people that claim they can hear the missing sound that happens in the music in-between the samples.

Now, if you can sample with roughly double the bit rate (I think 96K is where the industry is now) that aliasing problem goes away therefore the need to pre-filter goes away. So the sound is subjectively better, not because we need to hear frequencies above 20K but because we did not have to filter those frequencies to prevent aliasing. Thus it is the filter that did the damage but we needed the filter because the sampling rate was so low. Make sense?

Yes, I think I understand what you talking about.

Personally, other stuff aside, if I were you I'd take them to a lossless format and archive them. Which format is, I think, irrelevant, as WMA lossless and Apple ALE will probably both be around for a while. If you archive them lossless, you should be able to rip them to another format with minimal quality loss, and I feel better just knowing that the files are archived on DVD somewhere.

I was just trying to determine if there is a clean way to convert one compressed format to another. If you recorded the output of a WMA into some raw data format like WAV and then ripped that to a lossless format conceptually there would be no loss of quality. But from what I understand (and I could be wrong) conversions routines run algorithms against the compressed file to convert formats and this degrades the file.

The issue is really conceptual. As it stands right now I use my WMA files on my computer and in my Creative Nomad Jukebox Zen Xtra (Creative just can't give products short names) and everything is fine.

I lost thousands of dollars of old LPs as they aged and scratched, and of course,

I have a fairly large record collection - when you say you lost your records to age, do you mean usage? Or something else?

Now, you can throw several hundred on a DVD in a lossless format, and protect your investment.

Up till now I have found ripping records to be a tedious task although I have been using my digital recording software to do it (Sonar). What utilities have you used to rip records? I have CD version of a lot of the mainstream records but I have a fair amount of material that is only available on record.

you claim there is a such thing as a lossless conversion between two different audio compression formats

To make it simpler, I'll describe conversion between two lossless compression formats. I compress in FLAC, which doesn't degrade the quality, and convert that (decode/encode) into WavPack lossless, please tell me where any possible loss of data can occur, unless you're purposely clipping sampling rate or bit depth? Please back that with sources.

onversions routines run algorithms against the compressed file to convert formats and this degrades the file.

I just realized what you might be thinking of. You will lose data if you decide to mess with bit depth and sampling rates while you convert even between lossless formats. So if I go from lossless FormatA at 24-bit 96 KHz to lossless FormatB at 16-bit 44.1 KHz, yes I will lose a LOT of audio data and quality with it.

Okay, I just did a test using iTunes using these steps:

Converted a short music clip to WAV; this will be the reference file for loss of data, and it's 286,526 bytes long
Converted the WAV to AIFF (lossless)
Converted the AIFF to Apple Lossless (where loss would occur, during conversion between two compressed formats)
Converted the Apple Lossless to WAV

Resulting WAV file size: 286,526 bytes.

Result of comp (compare file) command run on both first and final WAV files in binary mode: "Files compare OK"

Now can we end this?

I'm entering at the end and have no interest in reading back further than the last 5 or 6 posts let alone commenting on them, but getting back to Apple and wireless applications, specifically AAC, it ain't here yet. On the other side, a new introduction to the 802.11 permits Squeezebox 2's improved support for WAV and AIFF uncompressed audio for all of our audiophiles who enjoy the rich highs of 26k frequencies.

I have a wireless LAN, lost packets, and eventually hardwired my station rather than continuing to hit the damn thing, so I'm not sure how well these work.

I own four businesses....I travel...cruises...four homes... villa...Rome...sail back to the US...wander in the Southwest...Mexico aboard a Celebrity ship...Gwennie is my girl

In cyberspace, you can be anybody you want to be

Result of comp (compare file) command run on both first and final WAV files in binary mode: "Files compare OK"
Now can we end this?

You are not even close. You said there is a way to convert one audio compression format to another without degrading the file. What you just did was convert uncompressed audio to a lossless format and back - of course that will work (unless they are lying about the format being lossless).

You claim there is a way to convert one compression format to another without degrading the filed - I have just been asking you to provide supporting evidence for this claim other than your personal opinion.

If you wish to create another test - this time try starting out with a compressed file.

The frequency I was speaking of was not audio frequency but the frequency of samples.

Can't really talk about one without talking about the other. They are inextricably linked. The Nyquist frequency establishes this relationship. It says that you must sample at twice the frequency of any sound (or phenomenon) you want to capture.

Digital recording takes a "picture" (sample) of the sound at a certain frequency and then represents this "picture" numerically.

Right, and it is the Nyquist relationship that talks about how fast you must sample to accurately capture a given frequency. If you want to capture at 20K you must sample at 40K (according to Nyquist) which is undoubtedly the historical reason why the CD standard decided on a 44K sampling frequency.

There are people that claim they can hear the missing sound that happens in the music in-between the samples.

Yes, and that is the reason for my post. What I was trying to point out is this. We all pretty much agree that CD sound, as good as it is, is not as good as the "real thing". One explanation is what you allude to hear - i.e. that some people claim they can hear sounds at 25K (for example) and those would clearly be absent from a CD recording.

What I was trying to provide is an alternative explanation, for the "sterileness" of CD sound. And that explanation is not the missing frequencies at, say, 25K, but rather the necessity to pre-filter the sound before sampling to eliminate aliases which would appear in a recording sampled at 44k but which would not appear in a recording sampled at 96k. By sampling at the higher frequency you can get rid of the analog filtering and then you get back that "presence" of analog recordings or put another way lose that "sterility" of digital recordings.

Can't really talk about one without talking about the other. They are inextricably linked. The Nyquist frequency establishes this relationship. It says that you must sample at twice the frequency of any sound (or phenomenon) you want to capture.

Interesting...

Yes, and that is the reason for my post. What I was trying to point out is this. We all pretty much agree that CD sound, as good as it is, is not as good as the "real thing".

If you mean live performance, ok. But if you mean some other recorded media, not sure I agree.

What I was trying to provide is an alternative explanation, for the "sterileness" of CD sound.

I now have a better understanding of what you saying (I looked up Nyguist freguency). But I have actually experienced adding tube equipment into the process can warm things up and it does work to some degree. I guess there are many factors.

By sampling at the higher frequency you can get rid of the analog filtering and then you get back that "presence" of analog recordings or put another way lose that "sterility" of digital recordings.

Interesting.

All I know is that he can tell the difference in quality when he converts his (IIRC 2") newly-created master tapes to digital, and that's not CD-quality digital, a bit higher. He's pretty disappointed by the time it gets to a CD.

It still has a lot to do with those beat frequencies. Your sampling rate can also add unwanted beat frequencies in fractional frequencies of the sampling rate. These change the perceived sound between the analog audio that the microphone picked up and the digitized and then re-converted to analog sounds a speaker outputs.

Digitizing is similar to quantum steps. The higher the sampling rate, the smaller the quanta the analog signal is chopped into for storage. The data is stored as a description of discrete square wave steps instead of smoothly rising and falling sine wave... that square wave is an average of the beginning and ending values of the sample rate bit. When reproduced, the Analog wave is not reproduced... just square waves.

Don't share this information with Last Visible Dog... he is being ignored. Besides, he wouldn't believe you anyway.