dex Otaku

greenmachine: sorry I ended up just restating what you'd already said. It looked a bit [as it was after ---] like it was part of a .sig. That's what I'd expect. Most "average" headphone outputs on devices like TVs, mini stereos, &c. will be in the voltage range of line-level when at about 70%. When doing something like this, I'd set the recorder to unity gain [no gain, no cut - about 18/30 on manual levels] and then adjust the volume of the device you're recording from. Other thoughts: * Set the TV's tone controls to flat * Turn off any kind of spatialiser, virtual surround, stereo enhancement, or similar Basically, disable any form of processing to the audio [other than MTS stereo if it's from standard cable or over-the-air]. See also the Recording from any line source FAQ.

This thread is now closed.

The manual, as well as promo material, clearly states the recording length in various modes with different disc types. A reformatted MD80 holds about 28 minutes of PCM audio. There's also a table in the current HiMD FAQ that lists all of these.

It's possible that the line-out of your TV is affected by the TV's volume control. There might be an option to disable that in the TV's preferences, or you might have to turn up the volume so it's within the normal range of line-level for consumer devices like your recorder.

I hate to perpetuate the totally off-topic thing here, but this absolute dolt is beginning to irk me. You really don't have a clue, do you? MP3 works in stereo, at any bitrate, much as any other lossy, or in fact, lossless coding method does: it encodes the information in two channels. The efficiency of various encoding methods varies, yes. Most stereo encoding loans bandwidth that isn't as effectively used by one channel to the other when it needs it, on a priority basis, based on which channel is more complex. ATRAC SP works this way. Stereo encoding [as opposed to joint stereo or dual mono] with virtually every lossy format works this way. This has its pros and cons, the main con being phase distortion. This is where joint stereo [m/s] coding comes in. Joint stereo takes the sum and difference of the left and right channels and encodes them instead, using basically the same method. The fact that there's a sum channel [basically, all the monaural content] means you don't get the same kind of phase distortion, differences in levels between channels, &c. HiSP uses joint [m/s] stereo encoding, as do most current MP3 encoders with bitrates above 96kbps. [Note: technically, joint stereo as used with MP3 can dynamically switch between "stereo" and m/s encoding, and when using VBR modes, can even fall back to intensity-stereo coding for passages of extremely low complexity, like the fadeout at the end of a song]. Part of lossy encoding is getting rid of redundancy. Another part of most formats is dynamic bandwidth allocation, as I mentioned above. And yes, 64kbps mono MP3 is crap; it's a lossy encodnig format whose basis is in algorithms developed around 1990! What you hear is neither mono nor stereo. Your hearing itself is binaural [two-eared]; this gives your brain the needed information to process location and distance to a sound source. Mono and stereo in audio terms are recording, storage, and playback formats, which have nothing inherent to do with hearing whatsoever. They are [regardless of recording medium] an analogue of the original, that is, a representation of it. Sound itself comes from a variety of source types, including point sources [similar in essence to mono, though by no means the same], diffuse sources, reflections, reverberations, echoes, and all. A single point source could for all intents and purposes by considered mono, but what we -hear- of it is by no means mono, because we have two ears, and each one gets a different signal. That is, unless you happen to be standing in a perfect anechoic chamber, in which case we're back to the point source. [bug80s explanation here is better than mine. Read up on wave propagation if you really want to learn about it.] As far as your recordings are concerned, most would have been mastered in stereo [which is actually a one-dimensional medium]. The fact that you're downmixing to mono means that you're keeping only the sum of the signals [the redundant parts, as you say], and throwing away a great deal of the difference [the stereo and phased parts]. Chances are, what you're listening to is actually lacking a great deal of what was in the original recording, from phase-cancellation alone. Sorry, but you're totally wrong on this. If it was mastered in stereo, and you've downmixed it, you've thrown away anywhere from 10-50% of the information, irretrievably. As for level differences between channels, there are many reasons for it - usually it would come from using pan controls on a mixing board. Other reasons include using channel-following compression [the common method of compressing stereo signals], where the loudness of one channel determines the compression applied to both. [The only truly accurate way to compress or for that matter process most stereo signals is to use m/s processing, by the way]. Otherwise it could just be as you suggest - shoddy equipment or shoddy handling of equipment. If you're altering the mix from how it was sent out from the engineering and/or mastering studios you aren't listening to it as it was. Converting to mono falls into this category. Need I point out that WAV is a container format, not an encoding format? That WAV files can contain virtually any audio encoding you like, whether it be linear or nonlinear PCM, PWM, MP3, OGG, ADPCM, or a hundred other encoding formats? WAV is by no means perfect nor secure. It's just a way of storing information, and telling the system using the information -how- it's stored, i.e. what the encoding is. And for the person who put the bit in about FLAC vs. WAV and tagging: you can tag WAV files. There are standards for it that have been around since the container format came about. You can even, if you like, use ID3 type tags on WAV files; they end up as fact chunks that get thrown away by most software. Not that I think that HiSP is better than SP, but you're completely full of it here. Any given encoding algorithm has its own rate of efficency. It's already well-known in the audio world that newer codecs can encode far more accurately and efficiently, with less audible artifacting, than older codecs that use higher bitrates. Bitrate when encoded does not scale equally from codec to codec. They are, after all, perceptual encoding systems, and as our understanding of perception increases, accuracy and efficiency also increase. While mono recording is used extensively by the broadcast industry, it's rarely been used by consumers. The advantages of having twice the disc length are pretty nonexistent, if you ask me, when you take into account the fact that you're missing 10-50% of the audible content because you downmixed it. Of course, if your recording was originally in mono, it's another story: HiSP among others that use [m/s] joint-stereo encoding, when given a monaural signal, basically give the single signal the entire available encoding bandwidth. If you insist on downmixing your music, or if you're recording live from a monaural source like a single microphone - what you're losing in playback time, you're gaining in encoding quality. Mono is perfection! Mono is perfection! Blah blah blah. Hearing the same in both your ears is not perfection. We have two ears for a reason: to perceive the difference between them. Mono also does not mean more power. Downmixing from 2-channel stereo usually involves adding the left and right channels and then dividing the resulting signal by half, or, alternately, dividing each by half and then adding them. The reason you feel "more power" from your downmixed recordings is likely only because you've convinced yourself of it. Look up: placebo effect. Whatever medium a recording is mastered in will be the reference [as close to perfection as possible] for that recording. Altering that in any way [such as mixing 5.1 to stereo, or stereo to mono] means you're destroying part of the signal, and you're destroying the original intent of the authors and engineers. Further: dual mono amplifiers means that each amplifier has its own dedicated circuitry for everything, including power supplies. It doesn't have anything at all to do with what signals go into the amps, or what signals come out of the amps. Again - that has nothing whatsoever to do with whether the signal that is amplified is monaural, stereo, quadraphonic, 5.1, ambisonic, supersonic, subsonic, hydrophonic, or anything. It just means that EACH AMP IS ITS OWN BLOODY AMP. [stereo amplifiers usually contain discrete amplification circuitry for each channel, but share a single power supply.] Volta is pretty cool, actually. E knows eir stuff. You obviously don't know your ass from your goddamned elbow. Monaural: In the simplest terms, is a one-channel recording and/or playback system. In recording terms, is a recording and storage format that uses a single channel. In playback terms, is the exact opposite of the recording or mixing/mastering chain, where the same one signal [monaural recording] is played through any configuration of loudspeakers; any system other than a single point source for playback will be inherently inaccurate. [Especially headphones.] Wikipedia's version? "Monaural sound reproduction is single channel. Typically there is only one microphone, one loudspeaker, or in the case of headphones or multiple loudspeakers they are fed from a common signal path, and in the case of multiple microphones, mixed into a single signal path at some stage." In lossy audio-encoding terms - mono refers to an encoding system where a single channel of audio is encoded into a given bandwidth, either variable or fixed. Stereo: In the simplest terms, is a recording and/or playback system that uses more than one channel. Stereo is commonly used to refer to two-channel systems, though it in fact refers to any number of channels greater than one. In recording terms, is a recording and storage format that uses two or more channels, usually two representing left and right and giving a one-dimensional [one axis of change is represented] recording. In playback terms, is the exact opposite of the recording chain [for acoustic stereo recordings], or the same as the mixing/mastering chain [for mixed recordings]. For two-channel systems, it is where two signals representing a one-dimensional recording are played through any configuration of loudspeakers that represent a configuration similar to that used when mastering the recording [usually left and right, set in front of and at between 30-45 degrees from straight ahead of the listener]. Any other configuration will be inherently inaccurate compared to the original. Note that the vast majority of two-channel stereo recordings are not actually intended for headphone listening. Excerpt from Wikipedia: "Stereo or stereophony generally refers to dual-channel sound recording and sound reproduction – sound that contains data for more than one speaker simultaneously. Compact disc audio and some radio broadcasts are stereo. The purpose of stereo recording is to recreate a more natural listening experience where the spatial location of the source of a sound is, at least in part, reproduced. Stereo comes from the Greek word for solid, and the term can be applied to any system using more than one channel, such as the multichannel audio 5.1- and 6.1-channel systems used on high-end film and television productions. However it is more commonly used to refer exclusively to two-channel systems." In lossy audio-encoding terms - stereo usually refers to a system where multiple channels share the same given bandwidth, either variable or fixed. The bandwidth can be divided equally between channels [as with "dual mono" encoding], or shared by priority [determined by complexity] between channels [as with most stereo encoding and m/s joint-stereo encoding methods]. Binaural: In recording terms, is a two-channel recording [technically also stereo] system using of microphones placed at the same distance apart and angles as an average human's middle ears, with similar occlusion and reflection characteristics caused by the average human's head, pinnae, and sometimes torso, imbedded in the recording by the presence of an actual or artificial head as part of the microphone itself. This differs from stereo in that phase angle, occlusion, and reflection are inherent parts of the recording that are necessary for actual binaural playback. In playback terms, is the exact opposite of the recording chain, requiring the use of headphones. Playing true binaural recordings over stereo loudspeakers is inherently inaccurate, as the speakers are commonly at the wrong angle to the listener, and are equalised in a completely different manner from headphones. From the Wikipedia entry: "Binaural recording is a method of recording audio which uses a special microphone arrangement. The term "binaural" has often been confused as a synonym for the word "stereo", and this is partially due to a large amount of misuse in the mid-1950s by the recording industry, as a marketing buzzword. In truth, binaural recordings are the best way to reproduce stereo with headphones. Typical stereo recordings are mixed for loudspeaker arrangements, and do not factor in natural crossfeed or sonic shaping of the head and ear, since these things happen naturally as a person listens." Lastly, as bug80 already alluded to one point of, decoding of PCM does take power in specific cases. Most consumer sound cards in Windows boxes use CPU power for everything from system EQ to the volume control to multiple-stream mixing, not to mention bit-depth requantization and resampling. To take it a step further than that, most consumer A/D and D/A converters are not in fact PCM converters at all, but variations on PWM converters, meaning the PCM bitstream has to be converted by the hardware - just like ATRAC or MP3 or AC3 or DTS or FLAC - before being converted to analogue.

The bit about getting shocks certainly isn't normal, but the bit about hum when using a plug-in powered mic at the same time as AC power -is- normal. All MDs I've tried this with [using stock AC adapters] have done this. The solution to that particular problem is usually to buy or make a regulated, filtered power adapter, as it appears that Sony are too cheap to include $0.02 capacitors or $0.05 voltage regulators in theirs. [Yes, I've also tried the same with regulated power, and as with battery power, there is no hum. Incidentally, their video cameras with plug-in power mic jacks also have the same problem, depending on what kind of power supply they come with.]

What you are doing wrong: using Windows Media Player [for anything]. Kidding. Or. Well. No, I'm not kidding. It's partly an opinion thing, and partly the experience that WiMP is simply crap, from basically every standpoint.

I'm glad I didn't read this thread before. I'm too tired right now to rip DJ_THE_CROW's methods and opinions into shreds. My god, where do people get this crap from?

You call that definitive? It raises God, which basically discounts it completely.

Woo. If the discs are older than 1992, the chances are basically nil that this is in fact the issue. The issue is more likely to be that the discs themselves are dying. They might work in a standalone player [which only reads at 1x speed] but any attempt at ripping will exhibit the full effects of bit rot. I have 400+ CDs, with maybe 1/2 of them made before 1995. Only two exhibit bit rot; it first showed up as swishing sounds [obvious distortion] on most players. Now they are completely unplayable [sound much like a data CDR put in a player that doesn't recognise such discs for what they are and auto-mute or refuse to play].

It's passed through the headphone jack, as the popup that comes up the first time you attempt it cheerfully and clearly informs you.

I'm not sure how long, actually. I have never had it split anything I exported. The old standalone wave converter split things at the length of a data CD-R. It seemed silly to me, but again, I never ran into it. I'm pretty sure I've exported recordings in excess of 100 minutes without them being split, though.

Counting on a chicken having to come from somewhere, and assuming it would be genetic mutation, maybe some random neutrino flinging through DNA at the instant of first cell division or something, the egg would have to be first; thing is, the egg wouldn't come from a chicken, but the chicken would come from the egg.

My vote, simply a != b != c, isn't there. HiLP sounds like crap to me, but then, so do most 128kbps MP3s. It is possible to make acceptable-sounding 128kbps MP3s [by using ATH filtering and joint stereo, or by simply cutting off all bandwidth above 12kHz, for example]. HiLP .. while it generally sounds like crap to me [keeping in mind that SS-encoded LP2 does as well], the artifacting is less annoying than with most poorly-encoded 128kbps MP3s. For uncomplicated live recording [i.e. non-crucial voice recordings] HiLP is quite acceptable in my books. Given the choice, I'd rather listen to nothing than any of the three [HiLP, 128kbps MP3, or SS-encoded LP2], to be honest.

Point of interest: I use SS now -only- as a library for uploaded tracks; I use it to upload tracks, title them, combine them, and export them. In this regard v3.x actually works very well, though attempting to use it as a complete music library is next to useless, I've found. Revision: I've started using SS as a music library in a limited way once again. This is mainly due to the mp3 compatibility of the RH10; it is flawed, but it's good enough for portable listening when I go for walks, and it gives me fast access to the very large existing library of MP3s I have from my own CD collection [still existing because I had an MP3-CD player before]. For the record, anything that isn't already encoded I use HiSP for [usually through Simple Burner].

Any clues? Sure. It's called DRM, and it's made to work that way. Had you read the relevant sections in your manual or in SS's help you would know this. The songs in any particular library are only good for that one installation of SS, unless you use the SS backup tool. They can not be moved or copied to another machine without using the backup tool. Your only solution at this point is to rebuild the library by re-ripping all of the content. This is part of why I don't use SS as a library for anything other than uploaded recordings. Simple Burner is far more effective for making MDs or HiMDs from your own music collection. On my machine [Athlon 2500+ with reliable C2-capable CD and DVD drives] it takes 4-7 minutes to rip and encode an entire 80-minute CD to HiSP.

Note that only .1A at pretty much any voltage is enough to kill someone.

"Only that which is provisional endures." [blaise Pascal] Hence.. Anything permanent is bound not to last, and anything temporary is bound to last. Hence.. God is dead, and I am immortal [and so are you]. [Hooray for shoddy logic.]

Many lossy compression schemes have difficulty converting signals that ride the 0dBfs margin. Most albums produced since the late 1990s are "bit-pushed", i.e. compressed and limited to make the overall volume of the content louder during playback. On normal equipment [i.e. a CD player] this doesn't cause a problem. Note that I'm using two meanings of "compression" here. First, that of dynamic range compression [the traditional audio term], which at its extreme is known as limiting; this is how AGC works when recording. The second type of compression is data compression, which is the blanket term for everything from lossless packing [such as MLP used on DVD-Audio and FLAC] to lossy encoding like Atrac and MP3. Highly-compressed content, rather than consisting of sounds that have an actual dynamic range, is basically made up of a series of extreme transients. Virtually all lossy encoding formats have difficulty with extreme transients. During encoding, these can cause many problems, usually showing up as pre-echo or ringing - things that all ATRAC formats are known for having audible difficulty with, as with the infamous "castanets" test. During decoding, the same highly compressed material will often show up as straight-out clipping distortion [which it almost, but not quite, is to begin with]. This is likely what you're hearing. MP3 is also known to have the same problem, hence the reasoning behind peak-normalising recordings to 98% of 0dBfs before encoding. LAME doesn't often have this problem, by my experience. This is probably due to its tweaked filtering and encoding algorithms. If the music you're having problems with does this on a consistent basis, I'd suggest ripping the entire album to WAV with a program such as Exact Audio Copy, which can do the peak-normalisation on the fly as per settings that are easily accessible in its preferences. If you rip an entire album as a disc image with cuesheet while normalising, you can then make a disc image with a program like Nero Burning ROM that will load with its Imagedrive utility, giving you access to the image as if it were still a CD, meaning that SonicStage or Simple Burner can find it in the CDDB and still give you all the tags correctly.

If the MP3-CD or DVD player converts output to PCM on its digital out, and does not flag the stream as uncopiable [look up SCMS on wikipedia] it should be copiable.

BTW - measuring the noisefloor is pretty useless without also measuring exactly how much gain the preamp is exercising, as well as its ceiling [headroom]. The noisefloor alone doesn't actually give you any useful information at all. [Well, it does give you the spectra, but without having a reference of some kind and knowing the actual gain, the spectra don't actually tell you anything useful.]

NiMH cells do indeed suffer from a memory effect, though it is far less pronounced than that of NiCd cells. Lithium ion rechargeables have no memory effect, to my knowledge.

The RH10's battery charge time, according to specs listed on audiocubes, is 3.5 hours.

You can use SS to edit groups, titles &c. and track order on both legacy MDs and HiMDs. Unfortunately, they don't allow batch processing [i.e. changing the artist on multiple tracks at once] but you can still edit the information, and do so -MUCH- more quickly than on your player.

There are many threads on this already, but here's an attempt at a short answer: With legacy MD / MDLP discs you cannot upload to a PC via USB, period. With HIMD-formatted discs [1GB or reformatted standard MDs] you cannot upload tracks that were downloaded from a PC via USB, period. With HIMD-formatted discs you CAN upload tracks recorded via the analogue or optical input; this will work to any installation of SonicStage, but can only be done ONCE. During uploading, tracks are marked as uploaded, and cannot be copied again from the original disc. If you try to do so, SS will [depending on the version you're using] either simply erase the tracks as though you're during a "check-in" [NetMD terminology] or give you a warning that the tracks were already uploaded and will be erased [sS v3.x does this]. The entire purpose of HiMD's digital rights management is to prevent you from copying tracks from one PC to another. The even shorter answer: No, you can't share music using HiMD.