bug80

Very strange. If I play a file in Sonicstage, it has the same dynamics as if I play it in Foobar, for example, so I assume that there's no compression. What kind of soundcard do you have? Maybe SS's output is routed to a channel on your soundcard that has some sort of dynamic reduction plug-in attached to it.

Does your MD deck have an optical out? I understand that your soundcard doesn't have an optical in? In that case I would search for a soundcard with optical in and use your deck to copy the MD's digitally in real time, instead of using your portable. In that case you're sure you don't get quality loss.

Most probably this is the main reason for Sony for not supporting SP downloads. To me, it's just like selling a car without the highest gear, because people might drive too fast with it. By now, it should be clear to Sony that MD isn't the first choice for Britney Spears downloading kids who just want a playback device anayway.

I still don't get why Sony doesn't offer SP downloads while most units have an optical in. The result is basically the same, but the method only takes more time.

The last thing I've heard is, that SP doesn't support DRM, and that's the reason why. But, true SP... how great would that be? Sony....????

I don't know. But I'm leaving now. I must reboot, because I want to play minesweep.

Yes you´re right. Nowadays it is no longer about the music but about making the record as loud as possible in the mastering process (grandpa speaks, haha). It´s a shame really. By the way, I´ve heard that even different masters are made for different continents (for example: a master with a small low/high boost for the VS and a master with a small mid boost for Europe), because opinions about what sounds good differs between different countries. If you look at it that way, then mastering for a certain medium isn´t such a bad idea (if we don´t take the question into account if there exists a method to overcome compression artifacts in a clever way).

Haha, seems like a typical mp3 player era question. To give you an answer: there are basically three modes on a NetMD player, with varying quality. The highest quality (SP) gives you 74 minutes on a disk, the medium quality (LP2) 2 * 74 = 148 min and the lowest quality (LP4) 4 * 74 = 296 minutes (of course, with 80 minute disks different numbers apply). If we consider a song length of about 3,5 minutes (typical for hits that you hear on the radio), then you will be able to fit 84 songs on one 74 min disk, in LP4 mode that is. However LP4 is very low quality, so if you want your songs to sound reasonable I´d suggest you use LP2, which gives you about 42 songs on one disk. I usually move whole albums to MD and than the mean song length is about 4 minutes, which gives you 37 songs on one disk in LP2 mode. But remember, I think it´s better to speak of the number of minutes you get on a disk than the number of songs, that´s just marketing talk.

I had no problems here in the Netherlands. Connect (US) shows up also. But if I'm correct The Netherlands doesn't have its own Connect site, so maybe that explains why it's working?

ok I don't really know what a mixing/master engineer should do to make a piece of music sound optimal on a MD. I do a lot of mixing (and mastering also) myself and if I'd would mix a piece of music for low-bandwidth purposes (for instance MP3@64), I would attenuate things like the hi-hat, the snare, and sounds with a sharp attack in general to avoid ringing and pre-echo. But, I would do that because I'd know that the resulting audio will sound bad anyway, and it would be just a matter of letting it sound as less bad as possible. With MD, however, we're speaking of high bitrates (at least I assume that you're aiming at SP or higher) and in that case the music should sound like the artists/producers/engineers have meant it to sound. In my opinion, if a medium isn't capable of reproducing that sound, it's a problem of the medium and not of the content. Fortunately, to some extent SP is perfectly capable of reproducing PCM quality. A final remark: I'm speaking about media that use compression here. In other fields, re-mastering is very common. For instance, I believe that audiotracks of movies get a re-mastering treatment for usage on a DVD, because in a typical home theater less dynamics are required than in a cinema.

I did a quick test also (last post in the thread) and LP2@132 did improve indeed. I didn't have any problems with hearing the remaining artifacts, though (ABX result of 16 out of 16). To me, LP2 is just far from transparent, even when listening in a noisy environment.

Don't panic, as soon as you connect a MD player you'll be able to select "SP mode" before transferring. At least here it works. It's still no true SP though, but ATRAC 132 kbs packed as a SP file.

First impressions: The new version looks better and seems to "respond" a little faster. Deleting tracks via NetMD seems to take much more time, however.

I don't think so, because in that case they have two options: 1) Mix and master as they've always done (going for the best sound quality for their music) and then live with any artifacts from the ATRAC encoding 2) Make consessions to the quality by attenuating parts of the music/spectrum that will be a problem for the ATRAC encoder. In case 2), it is no longer about making the music sound best, but it is about making the music sound best on a certain device/medium, which is not really the best philisophy behind mixing/mastering.

Installed without problems here. By the way, it seems that the built-in ATRAC encoder has improved a little, see my last post in this thread.

UPDATE I installed the new SS version (3.0) and did a quick test to see if the encoder has improved, and it seems it did (a little)! I converted the same sample I used before with SS 3.0 to LP2 and did an ABX test between this one and the SS 2.3 version: SS 2.3 (LP2) versus SS 3.0 (LP2): 15/16 Original WAV versus SS 3.0 (LP2): 16/16 So, I could tell the difference between the two SS versions and SS 3.0 sounded better in fact, with slightly less artifacts in the hi-hat and even the vocals were more smooth (some distortion is present in the vocals with SS 2.3). As expected, the "new" LP2 still differs a lot from WAV. Sorry HiMD people, I still don't own a HiMD unit, so I can't test higher bitrates.

True, but it´s more time consuming compared to downloading through USB. Being able to download true SP to MD via USB would be very nice

A ) I think this should be posted in the Software Support forum B ) Sounds like a FFDSHOW related problem. read this. Hope this helps!

I've always thought that LP2 was good enough for me untill I got myself a better pair of headphones. Man are those artifacts annoying It's a shame LP2 is the maximum bitrate for downloads to NetMD (I don't own HiMD).

Well, go ahead! I'm curious too. Yes, I took that into account. I didn't complare 2.2 and 2.3. Furthermore, it was a blind test, so no user-bias was present in the tests.

I'm afraid so, or you can search for a MD deck with optical out and connect that to a soundcard with optical in. In that way, you can copy your files without loss of quality.

But you don't think ATRAC3 audio at a bitrate > 132 kbs would be supported by NetMD units? Even it has the same apparent bitrate as SP? EDIT: In other words, do you mean that 132 kbs is the maximum bitrate supported by the ATRAC3 format?

I think so too. But then, it maybe should be possible to send high bitrate ATRAC files to a NetMD with the same size as SP files (at least, I guess the files should have a certain "apparent" bitrate because of sectors etc). And if that's possible, it would be very nice if Sony would implement that feauture in next version of Sonicstage. In that case, we would have the benefit of ATRAC files up to 256 kbs on our NetMD, and Sony still can use DRM on the files (it seems that DRM doesn't work on SP files). Or am I wrong?

I've never managed to get gapless conversion with SS either. It's a shame, because ATRAC is gapless by nature. For true gapless encoding, an encoder must "glue" two files together. This is not possible if two files are encoded completely seperate (as opposed to batch processing). SS simply can't perform batch processing, because it transfers a file to MD while converting the next one.

I hope I'm correct: CD-Audio: Frequency range 0 - 22.05 kHz, dynamics 96 dB (16 bits per sample) DVD-A: Freq range 0 - 96 kHz, dynamics 144 dB (24 bit) SACD: Freq range 0 - 100 kHz, dynamics 120 dB (equivalent to 20 bit) Remarks: * I believe the standard frequency range for CD-audio is 5 - 22050 kHz * The sampling frequency is twice the nyquist frequency for CD and DVD-A, and 2.8 MHz for SACD * Because SACD uses 2.8 MHz / 1 bit sampling, I use the term equivalent to 20 bit. * The values for DVD-A are the maximum possible values more info on DVD-A whitepaper on SACD