syko

I'm assuming you're using a remote to scroll through your songs? If so I had the same problem with my remote, where everytime I would change songs, it would either: Do what I tell it to door do one or more of the following Go to previous track instead of nextTurn up the volumeGo to next SOUND setting (eg. from EQ to DN)Skip a whole groupStop or pause playIt turned out to be a faulty remote that just kept getting worse and worse (it was an MC33EL). I had 2 of them and both developed the same problem. If you weren't using the remote to scroll songs and you are talking about the backlight of the RH1 itself... then sorry I can't help you If it is the remote, I'd consider using another remote for now, or at least check the contacts are clean and proper. It's cheaper to replace the remote than the whole unit itself. Side note: I'm using the MC40ELK, and it's the best remote I've ever had in terms of utility!

I'm an audiophile, and this is what I think... Atrac3 132k is much much better than Atrac3+ 64k. Try recording music in atrac3+ 64k with very high stereo separation, compare it to 132k atrac3 and tell me what you think. The amount of artifacts heard especially at 10kHz+... I'd rather eat glass, to put it simply. I would also say that I hear more artifacts in 132k atrac3 than in a well-encoded 128k mp3 file, so atrac3 is definitely not my favourite audio format. Atrac3+ actually steps up this quality though, with atrac3+ 64k sounding much better than atrac3 66k, but it definitely doesn't compare to 128k mp3 as Sony claimed in a test, nor 132k atrac3. It sounds more like mp3 at 92k, maybe less. I'll admit atrac3+ 64k as good stereo replication comparable to mp3 92k, but the artifacts it still generates, unavoidable due to low bit rate is still awful. It's only 192k atrac3+ that I find audio quality acceptible.

My unit did that at the in the first days and I couldn't figure out why. But it stopped a doing it a few days later and it remembers where I turned off the unit. At that time I was mucking around with charging, so it could've been that you need to fully charge the battery for it to work properly. Who knows why you need to do this ~_~

Double-sided Minidiscs will never happen because the current recording system requires both sides of the Minidisc. Information is recorded by using a magnetic head that actually is in contact with the disc. Information is not burned onto the disc by a laser but bits are augmented by the magnetic head touching the top of the disc. The is the reason why recording drains your battery so fast compared to reading. More torque from the motor is required because of friction between disc and head. However the advantage of this magneto-optical system is that nothing is burnt. The disc will last much much longer. Information is read by using a laser that reads off the bottom of the disc.

My RH1 is considerably louder than my NH1 when the laser head is moving. However spinning noises don't seem to be a problem. The instructions manual did say mechanical noise is normal, and is part of the "power saving mechanism". I wouldn't worry about it. My N910 makes a shrilling screeching noise sometimes when the laser head tracks, and it's done that for the past year without it failing. Its still doing well right now. Bear in mind, a 1gb HiMD disc is quieter because the bits a smaller than a 74min disc, so it doesn't have to spin as fast to get the same data rate (okay, it's just a theory but it sounds good ).

Double-pressing the scroll wheel also works As far as I am aware, all new remotes will work on older model units. Some buttons on the newer remotes could be reduntant or may have different functions, such as the scroll wheel not working on the MC-40ELK if plugged into a non-HiMD unit. Old remotes will work on new model units, but may not support or have buttons of all the features that a new unit will support. For example if an old remote doesn't have an A-B repeat, it won't do A-B repeat even if the unit has the ability to it on a newer remote. Surprisingly, old remotes that have screens (like RM MC-11) actually support the equaliser even if the units at that time only had MegaBass settings!

I found this hack a GOD SEND. I previously owned an NH1 that had an output of 5mW+5mW, which is the standard in Australian terms. I found the volume control on that unit perfect with my $100 earphones, which do unfortunately asks a bit more from the unit's amplifier than the original earphones supplied (probably vol 20 on my expensive earphones is equivalent to vol 17 on the supplied ones). I found I listened to my music around volume 18, and experienced no "clipping" until around vol 28 with heavy bass (it's insanely loud there anyway so I don't go there, if at all). You must realise though that with these HD amplifiers, they do not distort as such, but it cuts the volume down for that split second where it would otherwise distort. It's terribly abrasive to your ears if you have loud bass as the volume fluctuates as much as the bass booms - like someone playing with the volume knob turning it up and down. Now with my new RH1, I was extremely happy with this unit (bought a week ago). I listened, I liked, until I reached volume 20... I noticed my music was a little quieter as usual, so I turned up to around 23 to suffice. To my utter disappointment, I experienced horrible volume fluctuations when bass was driven through my earphones! I consulted the manual and made a horrible discovery... the unfortunate European units still had their power levels gimped (at 4.5mW). Now I thought, "Oh! I'm in Australia so I shouldn't have this problem!" Wrong! It seems that Australia too is now shipped with the European model which had a 4.5mW output. I actually found this fact extremely hard to swallow and I was extremely disappointed, nearly depressed. It was an excellent unit, but not even living near Europe, I (and other Aussies) were forced to suffer a drop in power output. Depressed for the fact that I thought I had a perfect unit, but I found the drop in output power a flaw I cannot overlook. I spent a lot of money on the unit!! It may not sound like much, but here's an experiment to prove my point. Find the largest headphones you can find and plug them into your unit (RH1 of course!). - If you have a 5mW output, you shouldn't experience any volume fluctuations until around volume 27. Also it still should be at a only medium-loud volume since the unit's amplifier cannot really supply enough power to drive your large headphones. - If you have a 4.5mW output however, you will suffer volume fluctuations around volume 23, and turning up the bass will exacerbate the problem (if your bass is +10dB @100Hz, you can't turn up past 21). If you have relatively sensitive earphones, you may consider me crazy for having volume up around 20. If I were using stock earphones I'd think the same, but I've got really expensive earphones I'd really rather use instead (Sony NX1 if you're wondering). They are less sensitive so therefore I must turn up my volume to suffice. I said 5mW offered the best volume range for my earphones, but 4.5mW doesn't cut it at all. I'm extremely happy now that my unit is finally on par with all my other MiniDisc units (5mW). Out of all my units (5 of them), I considered the RH1 the worst, simply because of this 4.5mW output - that was how much it affected me. I'm sure you'd hate it too if you bought a portable audio device at a very high price (for your wage), only to realise it cannot provide the power to drive your favourite head/earphones. So in concluding: - European models all have a capped 4.5mW output, not 4.8mW. - The volume difference between 4.5 and 5mW is not really discernable to the untrained ear between Vol 1-15 (Vol 15 @ 4.5mW ~~ Vol 13 @ 5mW). - In the 4.5mW unit, the volume never really rises above 25, so 25 is nearly just as loud as 30. It is even quieter if bass is involved. (Vol 25 ~~ Vol 30 @ 4.5mW ~~ Vol 22 @ 5mW, if you're lucky). ** Most importantly: - The effects of this power cap is most felt when using relatively large ear/headphones that require a lot of power to drive at respectable volumes. - As a sound recording engineer, I can tell you this volume increase is not a placebo effect. (If you can tell the difference between 128k and 160k mp3 audio then you're into music as much as I am!) (Oh alright, I'll write up a simple table)** 4.5mW and 5mW equiv. Vol 10 = Vol 11 Vol 15 = Vol 13 Vol 20 = Vol 17 Vol 22 = Vol 19 Vol 25 = Vol 19-20 Vol 30 = Vol 19-21 - The range (eg 19-21) denotes that the volume fluctuates that much when driven at that volume. Imagine how annoying it is. ** This is assuming you're playing a properly normalised track. I love this site. I have no objections to these hacks being readily available on these forums. Keep up the great work!

There is just about no difference between 256kbps and 192kbps. Perhaps the only difference you will hear is the stereo accuracy and very slight artifacts in 192k with very high frequencies compared to 256k. High frequencies require much more data produce accurately than low frequency sounds, so frequency response drops dramatically as bitrates drop. This is because each packet of data is too small and there aren't enough bits for the decoder to replicate the original high frequency sound. This frequency response is not encoder dependent - it's bitrate dependent, so it's not the encoder's fault that it can't produce frequencies that high; it's the file structure that is limiting its ability to reproduce them. The decoder won't be selective in what it processes. It will always try to process the same sound, and depending on what bitrate is used, will try to stuff as much information into each packet as possible. If it does find a frequency that will not fit into the smaller data packet accurately, most likely it will drop that part (frequency cutoff!). Each encoder will have its own algorithm in figuring out which parts to cut or compress whilst trying to maintain its highest accuracy to the original. Note also that most mp3s' frequency cut off at 192k is around 16-18kHz, which further justifies that this is not an encoder problem. The higher bit rate will always have the higher quality sound and in no way it will produce more artifacts. ------------------------------------- Here's something I can compare it to that might help understanding this issue. "0 1" means one cycle. The definition of Hertz (Hz) is the number of "0 1"s per second - cycles per second. ---------- Here is a data packet representing a sound - 16 bits in its original form. 0 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 - This string is fed into the encoder once per second. Assume the sound is at 8Hz because there are 8 "0 1" sequences fed into the encoder per second. This is its theoretical frequency cut-off because there can only be a maximum of 8 "0 1" sequences in one data packet. ---------- Now here's a data packet representing a sound - 10 bits. 0 1 0 1 0 1 0 1 0 1 - Note now that there can only be a maximum of 5 "0 1" sequences. This is the new theoretical frequency cut off of 5Hz for this type of file. ---------- Of course, with something say like 192kbit/s, there are 192000 bits to play with fed into the decoder per second, so things are less crude than my examples. This does not mean the frequency cutoff is 96kHz - sound is never so pure like a square wave as represented by my "0 1" sequence. Music is a combination of 0 and 1s with many waves at once. This is the very basic idea of an audio file.

Most people don't hear the conversion from 192k mp3 => 132k atrac, which is perfectly normal. However, I am sorry to say, but a quality degradation does exist, with frequencies above 8k affected and to a certain extent the quality of spatial sound in stereo. If you want virtually no loss, you will have to convert them to SP, but just about everyone will be happy with LP2. Heck, I can't stand the slight audio degradation, but I'm sticking to LP2 simply because of the extra storage on one disc.

There's a lot of things frequency response graphs don't tell, such as stereo replication and accuracy of sound to the original. Differences in the frequency response can easily be remedied through an equiliser, but nuances in accuracy, such as the dreaded "swishing" noise low bitrate files produces can't be eradicated (is it only me in this world that hears it??). Stereo replication of low bitrate audio becomes a major challenge, but surprisingly enough ATRAC3plus does a decent job at 64k - it is noticibly better than standard ATRAC. This comes from a person that can hear the difference between 128 and 160k mp3 audio . As a rule of thumb, I generally try to avoid anything below 160k, but 256k is such a massive jump it's overkill

I've done a quick and brief test on this DSEE technology, and here are my findings. DSEE will improve all forms of audio formats played back from your computer. However, the improvement will be limited to the bitrates and type of audio played. The most notable enhancement turning this on is the increased spatial sensation of stereo sound. The other is the slight tweaking of the audio's frequency response curve. It is enhancing the high-end noise of around 4kHz onwards, but also making the bass sound deeper by reducing 200-300Hz frequency and increasing 80Hz and below frequencies. The most important note is that it affects *all* audio formats including the lower bitrate ones such as ATRAC at 48kbit/s. I have tested this on 48k ATRAC and 64k ATRAC and I do hear a difference. Unfortunately though the quality of the sound of 48 and 64k limits the effect DSEE has. I can certainly hear the increased spatial sensation, but not the increased high frequency response. I would say only about 5% of the general public can hear this slight improvement in sound, but it should be just audible by sound experts and audiophiles like myself. Most likely you will only hear this difference using head or earphones. And I can hear this difference using my laptop's Integrated sound card with fairly high-quality headphones! I can give you a much deeper analysis on the changes in quality when I test this using my external soundcard and high-quality earphones. So in summary: -DSEE will improve all sound played back by SonicStage, meaning it works for ATRAC, WMA and MP3. -How much you can hear of an improvement depends on the quality of the audio file. The higher the quality, the more pronounced the improvement becomes. -Typically, very few will be able to hear improvements with 48k unless you have a very trained ear with head or earphones. ---------------------------------------------- Second test using better earphones: -Upper frequencies increased from about 12kHz, not 4kHz - (perhaps someone can provide a frequency response graph?) This is based on what I hear. -Turning this on makes bass frequencies sound less grungy - it's more crisp. -Vocals are affected most in terms of increased sense of space in stereo. It utilises rear speakers more if you are using surround sound.

After reading all of that, should I be glad I bought an MZ-RH1 instead of an RH10? It seems the mp3 playback is not worth it (and I also thought the remote was very scarce). Too bad the RH1 only uses Li-Ion

You actually can transfer Mp3->ATRAC3 songs into your NetMD at SP mode. In transfer mode, there's a button named "Transfer Mode" at the centre of the window. Click on it and a list of modes should come up. Tt should be the bottom option (Transfer in SP mode). But I have issues with that because the quality of SP from the computer still doesn't compare with what SP quality you'd get if you recorded your songs digitally. This is because you've converted your Mp3s to 132kbit/s and SP quality is about ~292kbit/s (don't remember but it was in the high 200s). So actually transferring songs in SP mode from the computer is just about the same as transferring songs in LP2 because quality is unnoticeable in the both of them. You can check how much minutes you have left in SonicStage2 by the little bar on the top right hand corner. The time left is displayed in SP mode, so if you transfer in LP2 mode, there's actually twice as much time as displayed.

I guess headphones are harder to power because they have a larger diaphragm than conventional earbud earphones. The softer the diaphragm, the more powerful the bass can be (which gives some low dynamic range) and since the diaphragm is smaller, the high frequencies are better. That's what I think anyway, but don't crucify me if I'm wrong.

The volume for N707 when playing on an amp is quite low. Unfortunately, there isn't any way to increase the max volume to play on your amp. If you hacked you N707, your line out wouldn't do anything - it only increases Vol to 30 and turns all bass/treble off.