bluestax

Thanks for the links. I followed them and kept on going and finally came across: http://www.minidisc.org/mj_ja3es.html Not directly for thr RH1 but probably close enough. In this, Sony give their reasons for the resampling. All I can say is that it's pure techno-bullshit! They claim that resampling a 44.1KHz signal reduces jitter and corrects for any inacuracy in the crystal oscillator of the sending device. Now,resampling has absolutely nothing to do with jitter. It's just a case of reclocking at the receiving end. Anyway, if there is really bad jitter then it will cause bit errors that will be (hopefully) corrected by the error corection at the receiveing end. Let's look at correcting the signal due to the transmitting oscillator being at slightly the wrong frequency. Why do it? The digital output is clocked by the transmitters oscillator but the information in the signal indicates that it was originally sampled at 44.1KHz and this is independent of the frequncy at which it is transmitted. Transmitting at slightly off the correct frequency will only cause problems if you want to listen to the result in real-time. If you're just trying to make a digital copy then the transmission frequency does not matter. You could transmit at double (or half) the frequency and (provided the receiver can lock on to it) get a perfect copy at the other end. That's what digital is all about! Think about it. If the transmitter is slightly off and the 'effective' transmission rate is, say, 44.0KHz but the RH1 resamples this and converts it to 44.1KHz then every time it is played on the RH1 it will have the error that was introduced by the 'faulty' tramsmitter. Who wants that? The other claim in the document is that the distortion introduced by resampling is better than -120dB. So what? If you don't resample then the distortion is ZERO (or minus infinity in dBs). How much are these guys paid to come up with such utter twaddle? Del-Boy

Coming to MiniDisc from DAT is a traumatic event. I find that my RH1 insists on resampling a digital input down (or up) to 44.1KHz. OK, I can understand down-sampling may be needed to allow the write system to keep up. But up-sampling??? What a waste of space! And then I find that the volume levels are also digitally adjusted. Why? Surely a digital output is optimised to the correct level. Why mess with it? Just adjust the volume at playback and leave the raw data alone. Does anybody know if Sony publish distortion figures for the resampling and volume controls? They're obviously embedded in the DSP chips in the RH1 so I'm sure that the quality is not great. I've spent a long time studying resampling techniques and other digital signal modifiers to know there's a hell of a difference between the end products. Perhaps the only way I'm going to find out is to record digitally from a CD in PCM and then transfer it to my PC and compare the result with a 'ripped' copy. I'm a bit scared of what I may find. Del-Boy

I've just tried recording to my RH1 from my DAB radio through the optical link and two things surprise me. Firstly, I can control the record volume. I'm used to using a DAT with optical input and there is no way to control the level. It comes through verbatim from the source with no changes to the digital stream whatsoever. Secondly, DAB comes out at 48KHz sample rate but the RH1 translates it to 44.1KHz for PCM recording. Some processing is going on with the digital input and I'm NOT happy since I have no control over it. Why can't they leave it alone? Is the RH1 not capable of recording at a 48KHz sample rate? Any comments? Del-Boy

I saw that and was tempted but got mine from HomeCinemaTV.co.uk for £207 (including postage). It looks like the shortages are over. Up to now I've been a DAT-head but I'v often hankered for something smaller. The features of the RH1 finally swayed me. The ability to record in PCM and transfer quickly in digital mode is essential IMHO. I cracked the method of reading DATs directly into a PC some time ago but it's a slow process since it's almost real-time. So, I tried mine out for the first time by recording a gig by my son's band on Wednesday night. With the same microphone that I use with my (Sony) DAT, the sound seems somewhat 'warmer'. Not so much top-end. I also found that I'd got the recording level a bit high (I never use AGC) so the waveforms clip a bit in places. I don't find the recording level indicators as good as those on my DAT. The big problem came next day when I tried to transfer to my PC. My Windows 2000 PC doesn't like SonicStage at all. It crashes every time I try to transfer - but I think that's a USB problem. My laptop's OK but when I tried to transfer the recording, it died halfway with disk error. Listening through the recording, there are three, very short silences which I guess are all disk errors. I split the track to isolate them and managed to transfer all the rest. I'm going to have to resort to using the line-out socket and going through analogue if i want to get the complete set to my PC. Not good! Has anybody else had problems like this? How do I prevent it in future or do I have to live with it? Del-Boy