bug80

Cygnus X1 didn't give a full explanation of his method. For instance, I showed that the signal path can be as much of influence on the quality as the codec.

Yes I know, but when you present your method & results you should mention such things

At home, the content of my MD's changes weekly, so I don't use labels, untill whiteboard-like labels are on the market that can be erased In the studio, we write our own labels. We don't have a printer there. By the way, I do have the "MD Label Projector" installed on my computer, just never used it.

Of course, there are two options: 1) The encoder in SS 2.3 has improved considerably compared to 2.2 2) The encoder in your unit is better than in mine Note that I didn't rip directly from CD. I ripped the track to a .wav file using CDeX. Sorry, I should have mentioned that. The .wav file was then encoded to ATRAC using SonicStage. I'd love to do it, but I don't own a HiMD unit, unfortunately.

Ok, my mistake. They were gone however, the songs disappeared from the MD and out of my library.

I've read a couple of times on this forum that the ATRAC encoders in Sony's hardware units perform better in terms of sound quality than the encoder used by SonicStage. I tought it would be a good idea to do a listening test to find out if this is true. I'll present the results here. Used test method I used an ABX test. For the people who don't know what that is: you can find more info here. For pval, I used a tolerance of 5% in this test. Used equipment The following hardware was used: * Sony MZ-N510 type S portable MD recorder * Sony DVP-S735D CD/DVD player * Alecto PRO-147 mixing desk * AKAI AM-A302 amplifier * Creative Live! 5.1 soundcard * Sennheiser EH 1430 headphone (attached to mixing desk) The following software was used: * SonicStage 2.3 * ABX Comparator 1.7.4 Collection of samples As a test sample, I used a .WAV with a length of 30 seconds of the song Prayer for the gun from the Dutch band Solo. The sample contains a drumcomputer with a sharp attack, a short hi-hat in the left channel and an open hi-hat in the right. It also contains a piece with piano, vocals and acoustic guitar. I recorded the sample in SP, LP2 and LP4 format using the optical out of the CD player and the optical in on the MD unit. From now on these are called hardware samples. I recorded these hardware samples to my computer in an analogue way using a connection on the mixing desk, which is attached to my Live! soundcard via the amplifier. Furthermore, I encoded the original WAV file to LP2 and LP4 using SonicStage. Because I had to make a fair comparison, I transferred these samples (which will be called software samples from now on) to MD and recorded them back using the above analogue method. Results With these samples I performed an ABX test, with 'A' being the original WAV and 'B' being an encoded sample. I used the headphone to listen to the samples. Hardware SP: 19 out of 20 (pval < 0.05%) Hardware LP2: 20/20 (pval < 0.05%) Hardware LP4: 20/20 (pval < 0.05%) Software LP2: 20/20 (pval < 0.05%) Software LP4: 20/20 (pval < 0.05%) Because this test is all about comparing hardware with software encoding, I also did an ABX test between hardware/software LP2/4: Hardware LP2 versus Software LP2: 14/20 (pval = 7.5%) Hardware LP4 versus Software LP4: 18/20 (pval < 0.05%) Conclusions First of all, I clearly heard the difference between all encoded files and the original WAV. The difference in the case of LP files was evident, these samples sounded compressed with a lot of warbling and pre-echo in the hi-hat sound. In the case of SP the difference was more subtle, but hearable. Very high frequencies were missing and some acoustic strums that were played simultaniously with a snare drum were attenuated (most probably because of the masking algorithm). In the case of LP4, I could tell the difference between the hardware and the software encoded files. And here comes the weird part: the file encoded in SonicStage sounded better!!. The hardware sample suffered from slightly more artifacts in the hi-hat sounds. In the case of LP2, I couldn't tell the difference. The result of 14 out of 20 isn't relevant, because pval is greater than the tolerance of 5%. Remarks I'm still quite shocked that I could hear the difference between SP and WAV. I thought SP should be transparent to most people with most music. It would be very accidental if the first sample I've picked would be a problem sample. What I think, is that the very high frequenies were missing because of the analogue chain MD > cable > mixing desk > cable > amplifier > cable > soundcard. Because of this, I can't really draw the conclusion that SP isn't transparent to me. Still, there seems to be a failing "masking effect" that shouldn't have to do with the way the sample is recorded. Maybe I should do a test in the future with my MD recorder directly connected to my soundcard. Second, these results are valid for this sample only. Maybe there's another sample with which I can tell the difference between hardware and software LP2, who knows. Also, the headphone I used isn't Class A. Better headphones may yield other results. * EDIT * I recorded a new WAV sample by playing the original one back on the computer, while recording it in the same way as I did from MD. If I ABX this new sample with Hardware SP, the result is 12/20 (pval = 40%), so the difference was indeed due to the signal path instead of the ATRAC encoding. I still can manage to ABX this WAV file with the hardware LP2 file with a result of 19/20 (pval < 0.05%), so that hasn't changed.

I've lost one too, with my NetMD (I wanted to transfer something back to my PC). I've just never reported it and I'm sure there are many others.

WTF?? What kind of a strange rule is that? I mean, I can understand (considering Sony's paranoid policy) that they don't want you to upload a file more than once. But can't they just block the file after one upload? Deleting something that is not yours is as bad as, or even worse than, downloading something that is not yours! Welcome!

If the original format was MP3 (or another lossy format), then this happens because these formats are not gapless, i.e. there's a short silence between tracks. Some of these formats are gapless (for instance OGG), but SonicStage can't convert them anyway. If you've ripped the original CD directly onto MD, then there's something wrong with your player/settings.

You know what would be great? An OpenMG plugin for iTunes, just like there used to be for Realplayer. That would be best of both worlds. You'll be able to manage your music in an intuitive way using iTunes and transfer them to MD if you'd like. Furthermore, it would be a good move from Sony to release a proper (stand-alone) ATRAC encoder where you can adjust settings. For example: fast/lower quality encoding versus slow/higher quality encoding. ATRAC is bashed in audio comminities for its quality, and I don't think that would be necessary if only the software encoders were better. Of course the best scenario would be Sony making the ATRAC algorithm open source, but I guess that would never happen

Like I said in another topic:

dex Otaku is right. By the way, recent listening tests point out that latest versions of the OGG, MP3, AAC, MPC and AAC algorithms all beat ATRAC in terms of sound quality at 128 kb/s. However, they haven't tested the built-in encoders of MD units. Too bad, because I heard those encoders perform better than the encoder that comes with SonicStage. Why don't they use that algorithm in SonicStage anyway? Is it because of speed? I don't mind to wait a little longer for encoding when the resulting ATRAC files will sound way better!

I guess you can also make a CD image with Nero and download it to MD using SimpleBurner?

But we can download to MD in LP2 or 4, which is ATRAC. I personally think that this limitation has to do with copy protection. Because we're not able to copy CD's digitally in a high quality format (SP) to MD, it makes it less atractive to copy them. Of cource, we can still keep high quality copies on our computer if we want to, so it doesn't really make sense. I would love to be able to download in the SP format by the way. Just to listen to my CD's in high quality on the road (yes I buy CD's Sony!).

Hm. Maybe my system just isn't fast enough It's going to be hard to find out where the bottleneck is. There are a lot of options.

I downloaded ffdshow 2005-02-01 and installed it. I tried converting a "problem" mp3 within SonicStage 2.3 with the following ffdshow settings: * mp3 codec disabled (worked) * mp3 codec "mp3lib" (didn't work) * mp3 codec "mp3lib" and bit-depth to 16 bits only (didn't work) * mp3 codec "mp3lib" with Fraunhofer system codec disabled (didn't work) * mp3 codec "libmad" and all possible bit-depths (worked!) So, the problem lies in the mp3lib codec. FYI, here are my specs: Athlon Thunderbird 800 MHz, 384 MB RAM, Windows XP SP2 System codec: Fraunhofer IIS MPEG Layer 3 codec (advanced) Disabling the Fraunhofer codec didn't prevent SS from crashing, but it started decoding (this problem mp3 crashes at 19%). I guess SS doesn't use the system codec, but it's own. * EDIT * dex Otaku, what kind processor do you have, an Intel? Maybe the problem is processor specific. As far as I know, ffdshow uses instructions like SSE and MMX, if available.

Sorry I will test that version as soon as I can.

Which one? I'm using the newest version I could find (oct 12 2004). If I have some time left this week, I will try to reproduce your tests on my PC. Maybe I can find out where things go wrong exactly.

This is very interesting. Personally, I don't think SonicStage uses the codecs of ffdhow, if they're enabled. It's maybe just that the filters of ffdshow may have a problem with some versions of Fraunhofer's codec, for instance, and that will be the reason SonicStage crashes on some systems with ffdshow installed. However it's strange that SS crashes, while other programs work fine.

Just a note, 0 dB is pretty loud. In fact, it's the loudest you can get on a digital medium. I don't know the exact treshold of Sony's MD recorders, but I guess the recorder counts something like -30 dB as "silent" or maybe even -60 dB. Ontopic: I own a NetMD, so digital uploading is not an issue for me, but if I'd own a Hi-MD I think I'll always make an analog backup copy before uploading. I don't trust SonicStage in anything

What version do you use? I'm not at my own PC at the moment, but I think I have the newest version (older versions showed bad video performance on my PC).

That's the problem. It does include audiocodecs. But, as I mentioned above, I don't understand why, because by default they aren't doing anything, except intercepting your audiostreams and confusing SonicStage.

Have you checked your USB cable? I've read something like this before and in that case it was a bad cable.

Hey, I'm glad I could help you out! ffdshow is a so-called codec pack. Most people install codec packs on their PC's when they want to watch movies in "exotic" formats like DivX or XviD. The programmers of ffdshow thought it was a nice idea to add audio codecs to the package. The reason behind that decision is unclear. ffdshow has some "extra" audio features, like EQ, reverb and convolution, but those are switched off by default (so, why don't they disable the complete audiocodecs by default?). Anyway, SonicStage has a problem with those codecs. A complete uninstall of ffdshow is not necessary, it's enough to disable the audio codecs and leave the video codecs as is. I'll try the Kazaa codecs mentioned by LowMD. Once again, I'm glad I could help! I'm sure more people have the "ffdshow issue", so there should be a sticky topic about this, I guess.

The reason that we're not able to tranfer in SP via USB is probably some sort of copy protection? I guess Sony tought, that not being able to copy cd's digitally in high quality makes it less attractive to copy them. Now that Hi-MD owners are able to do such transfers makes it out of date IMO. Same for USB uploads from MD to PC. If Sony releases some sort of firmware update to overcome this limitations, I will forgive them for all the frustrations I had in the past with recording my own music analog to PC, crashing SS, etc