bug80

Ok, there are fields within physics that are less exact than others, psycho acoustics for example. But since compression algorithms are all about psycho acoustics, it is not very strange that their quality is tested using blind tests. In fact, the models on which these algorithms are based (treshold of hearing, temporal/spectral masking, cochlea modeling etc) are all based on actual listening tests. I still consider physics to be an exact science, however. I just hope that the aliens stay away for a while..

And that is why blind tests are so important. Compression schemes aim at the "best" results given a certain (average) bitrate. "Best" must be defined for the "avarage" listener. So, if our world gets invaded by aliens (man, where is this discussion going to? ), they must be involved in the blind tests too. Again, I'm into acoustics and blind testing is a valid scientific method and for most problems related to audio it's even the only way to go. I've encountered some examples of this during my last research. If you find this interesting you should take a look at hydrogenaudio.org. It is a place with lots of information on audio compression and blind testing. Possible, but it is a general rule that decoders only decode, nothing more, except for things like dithering and resampling, if necessary. Everything else should be done by the encoder.

Oh, let me add a contribution to the topic. I would advice to re-encode MP3's to Hi-SP. From my experience transcoding MP3's to LP2 leads to big quality losses, especially if the MP3 was at a bitrate < 192 kbs.

Or Stella Artois, Palm... Oh yeah!

Hm. If the eternal future is at an infinite time, it will take an infinite number of time steps to reach eternal future and thus we will never reach it. My $.02 Edit: interesting last comment about the inifite past. Thinking about the big bang, the past may not be infinite. The future may not be either.

I study physics, currently working for my master degree on acoustics/sound control and I can assure you ABX testing is considered to be scientific. In fact, I used an ABX test with different subjects as a part of my research. Why? Because I had to test different algorithms that were almost identical in a mathematical sense and guess what: it turns out that people hear a difference!

I believe musicmatch uses its own encoder (not the best one around). psyhya, I've taken a look at your graphs and besides the fact that they don't say much about perceived quality, I see a lot of differences there. Some frequencies differ with a factor of 10 in amplitude, or even higher!

Not entirely true. The most obvious artifacts in the samples presented here are things like pre-echo and ringing, thus temporal artifacts. It is very likely that these don't show up in a FFT analysis. Compression involves a lot more than just throwing away less important frequencies, by the way. In the field of audio compression (and in the field of scientific acoustics in general), it is general concensus to perform listening tests to test the perceived quality of a codec or any other algorithm. A spectrum analysis (or, closely related, a correlation analysis) may only be used to couple the results of the listening tests to the mathematical properties of a signal.

I voted c < b < a, altough I had some trouble testing the samples, because it was not a blind test. It is very hard to make a fair comarison if you know which one is which. The atrac samples both had severe problems with the drums, more than the mp3 file. Does anyone have a link to the page where Sony makes the claim (including scientific evidence, hehe). I'm very curious what the test conditions were, which encoders they used (probably not LAME), background of the subjects, etc. A company making such strange claims is not new of course, WMA @ 64 kbs is also advertised as being comparable to MP3 @ 128.

Now I'm hungry again. Thanks

Oh, sorry Yes that it is also possible. It is a matter of adjusting the frequency domain coefficients in the MP3 file, transcoding is not necessary. But I don't think it is an easy operation, so it is not very likely that this is happening. Still a chance, though.

Sony logics. Don't ask why

That's not true, because of the roll-off the effective bandwidth is lower than as stated in the specs, when playing back a MP3. Sure, frequency response is not quality, but as soon as people hear a significant difference between ATRAC and MP3 playback, which is the case, the frequency repsonse can be used as an objective explanation of that difference. EDIT: forgot to anwer your question. Yes, that is possible. During playback, the unit decodes the MP3 data to PCM. After that, any filter can be applied to that raw signal.

Sure. That's why I suggested to wait for Sony to come up with an official statement. Although in my opinion it is very likely that it was intentional. By the way, if Sony gives a reaction, they will probably deny that it was intentional anyway. Sony will try to keep the damage to a minimum. * EDIT * I just thought of the following: there is a (small) chance that the roll-off is not applied inside the unit, but by Sonicstage, before transfering the MP3's over to the player.

Maybe "Linux" should be added to the topic title. As long as Wine doesn't support USB drivers, Sonicstage will never work through Wine, unfortunately. I wonder if Sony will sell more units if they make Sonicstage work natively on Mac OS & Linux. I think so and it will probably result in more cash than they have to spend on extra software engineers to port Sonicstage to that platforms.

The problem is, there will be a lot of people that buy a 2nd gen unit, because of the MP3 playback feature. For me, it would be the main reason to finally make the switch to Hi-MD. I could fit a lot of music on one Hi-MD disc when I rip my CD´s using LAME --alt-preset standard, which is mostly transparant to me and has a much lower bitrate than Hi-SP, on avarage. Right now, I´m happy I waited for a while before buying one. Somehow I knew there would be some hidden drawbacks. If I would´ve bought one, Sony´s trick (let´s assume it´s a trick) would´ve forced me to use ATRAC. So, I would be unable to use the feature I bought the player for initially. I call that misleading marketing, something Sony is famous for. I love music and I love it even more when it doesn´t sound bad.

Guys & girls, how about this: we give Sony untill the end of this month to come up with an official statement, in which they state that a nasty bug has made its way into the latest generation of minidisc players (or something), that makes MP3 playback sound bad compared to ATRAC. Sony should offer users a firmware upgrade (for free, of course) that fixes this issue. If they don´t do this, we can send a message to all major technics & audio news sites (CNet and the like) in which we make clear that Sony most probably scr*wed its customers by degrading the MP3 decoder, to advertise their own ATRAC format. If I was working for Sony, I would not let it come to this point. I think it´s time we strike back! If you think about it, we´ve always been very patient with Sony here at MDCF. We pointed out things that could be improved on the hardware and software side, helped new users who had problems with either their units or Sonicstage, etc. Despite all the ignorance of Sony, most people here at MDCF sticked to MD, because they believe in the format (at least I do). Maybe it´s time we show that Sony can´t do things like they did now and simply get away with it.

Girl power!

You´re right that lower bitrates need more filtering of high frequencies, but this should be done by the encoder, not the decoder. Even at very low bitrates, it is not up to the player to do any filtering. This is also not done with ATRAC, so why would a player do it with MP3 files? See the comment above

Of course, but that is not the issue here. The graphs show that the RH10 cuts off high frequencies that are present in the original MP3 file. A decoder should not do that. Switching off the low pass in Lame is not recommended, by the way. The presets are optimized for giving the best overall results, giving a certain bitrate.

Speakiig of bugs, I´d say the ¨issues¨ 2nd generation units have with MP3 playback demand for a firmware upgrade.

That's not the point. Look at the difference between the original MP3 and the hardware decoded MP3.

Peel, thanks for doing the test! I think it is now proven something is wrong with the MP3 decoder in 2nd gen units (or the RH10 at least), whether it is by accident or on purpose (although I can't imagine Sony introducing an approx. 3 dB/oct lo-pass filter by accident).

Yes, you can speak of it as some sort of audio zip, but actually it is possible to play the files without unpacking in advance. There are plugins available for Winamp, Foobar2000, Nero, etc. It even supports tagging. The typical reduction in file size with FLAC is 40-60%. Player support is still lacking indeed, but for the moment FLAC is mostly used by people who want to backup their CD's in a lossless format, so that they can re-encode to whatever lossy format they like (MP3,OGG..).

As long as there's no *real* alternative for studios, radio stations, journalists, musicians, etc for a high quality recording system with swappable media, I don't see MD disappearing. That said, it could be possible that Sony has a future format in mind, for example decks and portable recorders with 100 Gb flash cards, or something.