dex Otaku

Good dyanmics processing can be difficult to achieve. Keeping things sounding natural, especially when you're amplifying the quiet parts, is probably among the most difficult to attempt. As I don't use Audition, I can't give you advice on how to use its plugins. In generalised terms though, what you want is sometimes called reverse-compression and is basically the exact opposite of a gate. If you were to look at compression as on a graph [sound Forge's dynamics method uses this] what you'd be wanting to do is set a threshold, but rather than compressing everything above it [lowering, or attenuating, the levels above the threshold], you want to compress everything -below- it by amplifying it. As I said, in my experience at least it can be very difficult to do this without things sounding very unnatural. Also, the more dynamic the signal is [the bigger the difference between the loud parts you want to leave alone and the quiet parts you want to amplify] the more obvious the compression will be. Hopefully there are some Audition users who can contribute to this for you.

Different people seem to have different experiences with this, for some reason, but here's how it works on my NH700: When using the mic-in: * there is no auto-trackmarking [following silences] * auto time marking [which inserts a track mark at an interval you set] works normally When using the line-in: * auto trackmarking [following silences] is normally on, and can't be turned off * auto time marking defeats the above * sync record is used with the optical in only and is supposed to pick up trackmarks from the source you're recording from There are several threads about this on the board already, and it seems that different people [even with the same models] are experiencing different behaviour with their recorders. In any case, you can always erase the track marks on the recorder or by combining the tracks after uploading. I never use most of the editing features on the unit itself, so I'd suggest checking your manual to see what it says about removing track marks. [it's not difficult to do, I just don't know the procedure off the top of my head.] For info on combining uploaded tracks within SS, check the uploading FAQ, which can be found here: http://forums.minidisc.org/index.php?showtopic=7436 Cheers.

Edit -> Combine This has been available since at least SS 2.1, though I don't actually know whether it will allow combining of any tracks in your library. I do know that it works with tracks that have been uploaded to SS after being recorded on the HiMD itself [i.e. with a mic or using line-in]. This is probably in SS's help.

I think the most important thing to note is that the Marantz units [all of them] are designed specifically for professional broadcast use. Their size, design, functionality, durability, &c. all reflect this. They would not be suited for many of the things that MD / Hi-MD users look to do [i.e. stealth recording, location recording using a single AA battery] as they are intended for a different purpose in the end. That said, using the equipment that best suits your purposes is always a good idea, as long as you can afford it. Keep in mind that professional units will allow uploading in the clear [a definite plus] but also won't directly support the use of the smaller stealth mics that so many of us use without adapters and possibly even external preamps [as the mics we use usually require 'plug-in power' which is not the same as 48V phantom on XLR jacks], bringing the end price up even further.

To my knowledge Sony has no plans to release firmware updates for the 1st-gen units. I would like to ask, though: if you're very satisfied with its sound quality as is, then why would the inclusion of mp3 support on 2nd gen suddenly make your purchase a waste?

This is a close analogy, yes, but not exactly spot-on. You have to keep in mind that image resolution is totally independent of colour depth, whereas audio bitrate is totally dependent upon bit depth. In terms of raw audio, bitrate = (bit depth) * (sampling rate) * (number of channels) and in terms of raw video, bitrate = (colour bit depth * number of colour channels) * (framerate) * (image resolution) Since audio is a stream [time-based] it is not directly analogous to still images, which are not.

I apologise for seeming offensive. I do tend to get somewhat indignant when I point users to the precise answers to their issues, and they not only promptly ignore the answers, but criticise what they either didn't even read, or didn't understand, rather than asking for clarification. [Note: "I need help!" is not asking for clarification. "What do you mean by blah blah blah in step 4 of blah?" is asking for clarification.] Indeed, it isn't my intent that you go risking your recordings. It is my desire that you read the directions on how to transfer your recordings to your computer, using either analogue or digital means, which are in the FAQ. Your original query basically asked me to rewrite the FAQ, which I'm not going to do. If you don't understand some part of it, please, just ask for clarification. Doing so would also improve the FAQ, since any clarification would be incorporated in it. None of these methods will put your recordings at risk, and none of them depend on your using any specific version of SonicStage [other than one which is recent enough to support control of your Hi-MD unit]. First, how does my being in Canada make me irrelevant to your query? To the best of my understanding, SonicStage is the same program and Windows [in its various incarnations] is the same operating system whether they are run on a computer in Buenos Aires, Frankfurt, Belfast, Toronto, Tokyo, or even, for that matter, on the moon. Changing the language does not alter the underlying programs, only their interfaces. Second, "dex's options for paranoid users" is there because some users are paranoid that SS will trash tracks during upload. This appears, from where I'm standing, to apply 100% to your situation. And no, it doesn't make it clear to me why you're asking for help, since that exact section of the FAQ addresses the questions you've asked. If my language is unclear or too technical, why not ask instead for clarification of what I said? Instead you appear to have ignored the answers I gave altogether, which seems odd to me, considering the fact that they are precisely what you were asking for. Third, though I'm not sure from the context you use "I am sure you have read the postings" in, I'll point out that the original uploading FAQ was written by myself, and the updated version was partly written and subsequently fully edited by myself. I'm pretty sure I know its contents, if that's what you were inferring. Again, if there's anything in there that is totally unclear, please point it out, and it will be changed. I obviously don't understand what the problem is [it doesn't help that you haven't stated what it is]. Do you require Connect support for your region, which, as I also already said, is the only actual difference I've ever seen between the regional versions of SS, other than language? Or is it that when you attempt to run the SS installer, it directs you automatically to the older version once it detects where you are connected from? If this is the case, I don't have a quick solution for you, but I would suggest watching this thread as it will eventually resolve this particular problem: http://forums.minidisc.org/index.php?showtopic=8183 If this is not the case, and you can go to http://sonicstage.connect.com/SonicStageInstaller.exe and it begins the install for SS 3.0 - then I fail to understand how being outside of the US or Canada limits your computer's ability to run it. With the exception of Sony's detection of certain ISPs and the country they're in, and the subsequent forcing the installation of the older version of SS - all you have to do is download and install it, as countless others around the world have already done. As for what Sony say [uS and Canada support only] - Sony also say that they won't support SS users on other than factory-installed operating systems. This would leave something like 75% of users out of luck if they called for support and told the truth. By your logic, Sony's stating that SS won't work in your country somehow erects a magical shield around it which repeals the laws of computing for only their software. To my knowledge, this is not the case [though it would certainly be an interesting world if it were possible]. Again, the countless users around the world who are already using SS 3.0 and have downloaded it from US servers would tend to back me up on this. Regardless of what the case is, here are my suggestions to you: * ignore what Sony said about support for the US and Canada * download [hopefully] SS 3.0 from here: http://sonicstage.connect.com/SonicStageInstaller.exe * install it and use it * if you're still worried about losing your recordings, or if Sony/Connect limit you to downloading SS 2.1, then back them up first as detailed in the FAQ * if something in the FAQ isn't clear, ask for clarification Lastly, if what you were actually asking for in the first place was a tutorial on how to use Audacity, then I'd suggest looking here: http://audacity.sourceforge.net/help/

PCM stands for Pulse Code Modulation, and in easy terms uses binary numbers to represent the amplitude of a signal with one number, called a sample, for every pulse of a fixed clock, called the sampling rate. i.e. an analogue to digital converter measures the ampltude of a signal and converts that measurement to a binary number once for every pulse of the sampling clock. Amplitude is stored at a set bit-depth, and often also uses a signing bit to signify positive or negative amplitude. With CD, the bit depth is 16 bits, meaning that the amplitude is recorded as a value between 0 and 2^15 along with the sign, giving a range from -32,767 to +32,768, for a total of 2^16 values, or 65,536. Another way of saying this is that the amplitude is sliced into 65,536 segments, and the system records which one it's currently at. On a common graph of the analogue signal this would represent the vertical axis - amplitude, or voltage in the case of the electrical signal being converted to digital. The sampling rate is set by a fixed clock. With CD, this pulses once every 44,100 times per second. On a common graph of the analogue signal this would represent the horizontal axis, or time. CD standard uses 16 bits signed, 44,100Hz sampling rate, and 2 channels, resulting in a bitrate of 1,411,200 bits per second. A simple analogy to this would be for you to write down the temperature on a thermometre once every 5 minutes; you and the thermometre are the converter in this example, your "depth" is the number of decimal places you record to on the temperature scale you use, and your sampling rate is 5 minutes.

SP has evolved over the years, so trying to say that SP now = SP in 1992 is patently false. Just thought I'd point that out.

Are you running a software firewall that is blocking SS and SB's requests to CDDB?

This is true, but archival usage dictates writing once and not messing with the disc again.

To my knowledge, the only significant difference [other than language support, possibly] between the different Regionalised versions of SS is that they change where the Connect page directs you to. I'm in Canada, and have -never- downloaded a Canadian version of SS. Why? Why bother! I'd have to stick with an older version just for a feature I will never use [Connect]. As to how to import/export without using SS, try looking in the HiMD upload FAQ. It describes three methods that don't use SS's upload features, down in that section titled "dex's options for paranoid users." No offense, but if you had actually spent the 5 minutes reading those "piles and piles of often irrelevant advice" you would have found more than one variation of exactly what you're looking for. The uploading FAQ is here: http://forums.minidisc.org/index.php?showtopic=7436 In any case, you don't actually have to stick with SS 2.1, unless you can't read English, which you obviously can [and can write it quite well obviously, as well]. If you have troubles installing SS 2.3 or 3.0 from the Connect.com site because of your region, there are links in software forum here to packaged installs of both. Cheers.

How the iPod ran circles around the Walkman, commentary on the job Sir Howard has ahead of him.

SS sorts things using each track's tag info. If you want to move tracks within the library, simply change the tags to reflect where you want things moved to.

Refer to the uploading FAQ in the "HiMD essential info" subforum, in the HiMD forum. Welcome to the forum, btw.

Memory, unless it's flash, requires power and processing in order to maintain its contents [i.e. refresh]. What you're suggesting is either putting 1GB of RAM in there, which would suck battery power at an incredibly alarming rate, or putting 1GB of flash in there, which would call to question - since you now have a flash player, why are you using discs at all? There's also the slight issue of it taking about 30 minutes to copy the entire [1GB] disc to the memory. ROMBUSTERS is correct, actually. With a truly random noise signal, the compression would be 1:1 or worse, due to overhead thanks to subcode et al. The likelihood of this ever actually occuring is small enough to consider zero, however. How often do you find sound sources that are actual white noise [i.e. even distribution over the entire recording bandwidth]? It's not a ridiculous claim, but it -is- the difference between theory and practise. In terms of computation, PCM requires no processing at all for raw stream playback, and FLAC requires probably about as much as decoding an MP3 does. Lossless-packed formats require processing; PCM doesn't, unless you're oversampling, EQ'ing, or the like. WAV [a container format], incidentally, is usually used to store audio in PCM [a data format that goes in the container]. The CPU overhead you'd see with it is likely either from I/O or directsound mutilatiing the signal, not the audio being decoded - since it doesn't need to be decoded.

Note also that due to there apparently being no lowpass filtration on the input to the ADC on many MD and HiMD units, a very easy way to trip up ATRAC/3/plus compression [or cause audible harmonics in PCM recordings] is to record sources that include high-intensity ultrasonics. This is likely to blame for at least a couple of the examples on the page linked to above.

Well, no. Sensitivity has to do with the efficiency of the microphone at a given SPL. A high-sensitivity mic will put out a greater signal than a low-sensitivity one. It makes sense that if your sensitivity increases, your dynamic range will as well. If the mic is more efficient but has the same self-noise, the SNR will increase accordingly, and so will the dynamic range. With my train example - no, the mics never clipped [and neither did the preamp]. Generally speaking, most mics and solid-state preamps don't suffer from problems with transients [when it all of a sudden appears on the meters]. What I was implying when I said the thing about the train was that the train [or at least, the engine] was likely louder than 105dB, in which case the high-sens mics would have distorted. Side note: if you're recording with the AGC on, -that- is when you'll have problems with those sudden attacks. This is a side-effect of AGC, not of the mics. I never record important sources, natural sources, music &c. using AGC. I almost always leave AGC turned on for recording speech, though, as it so happens that the low-sensitivity version of the SP-TFB-2s has exactly the right sensitivity for everyday sounds to fall well below the threshold of the AGC's compression, meaning the AGC only kicks in when something really loud happens. Going by the rated sensitivity of the high-sens version, everyday sounds would constantly be riding at or just below said threshold, making the AGC far more obvious. Perhaps it would help to try and explain what these mean: "Open circuit sensitivity" is a measurement of how much level a mic puts out for a given sound level. The current international standard usually uses a reference level of 94dB [1Pa] compared with 1V, i.e. a sensitivity of 0dB would have the microphone putting out 1V when transducing 94dB SPL; if we had mics like that, we wouldn't even need preamps! See here for a good quick reference to SNR: http://en.wikipedia.org/wiki/Signal-to-noise_ratio Likewise for dynamic range: http://en.wikipedia.org/wiki/Dynamic_range General audio measurement terminology: http://en.wikipedia.org/wiki/Audio_system_measurements While you're there, also check out: http://en.wikipedia.org/wiki/Microphone and http://en.wikipedia.org/wiki/Binaural_recording And, for that matter, the category itself of: http://en.wikipedia.org/wiki/Category:Audio_engineering Note that what most companies refer to as "binaural microphones" in fact use variants of the A-B stereo mic'ing technique, and have nothing whatsoever to do with binaural recording; this is part of the reason why I don't actually like listening to many if not most recordings made this way over speakers, and most specifically the reason why I keep insisting on pointing out that most binaural mics are not even pseudo-binaural, let alone binaural. Quoted directly from the wikipedia entry on microphones: "The A-B technique uses two omnidirectional microphones at an especial distance to each other (20 centimeters up to some meters). Stereo information consists in large time-of-arrival distances and some sound level differences. On playback, with too large A-B the stereo image can be perceived as somewhat unnatural, as if the left and right channel are independent sound sources, without an even spread from left to right. A-B recordings are not so good for mono playback because the time-of-arrival differences can lead to certain frequency components being canceled out and other being amplified, the so-called comb-filtering effect, but the stereo sound can be really convincing. If you use wide A-B for big orchestras, you can fill the center with another microphone. Then you get the famous "Decca tree", which has brought us many good sounding recordings.

The difference between them is actually very small, which is the other part of the reason why I chose the low-sens version. Going by SP's info: Signal To Noise Ratio Low Sensitivity 58dB/High Sensitivity 62dB Open Circuit Sensitivity Low Sensitivity-42dB/High Sensitivity-35dB Maximum Input Sound Level 105dB/120dB Dynamic Range 81dB/96dB I find it interesting that while sensitivity is 7DB higher for the high sens version, and the dynamic range is 15dB higher, the SNR is only 4dB higher. In the end, I based my decision on the 105dB vs. 120dB max SPL alone. 105dB is enough for small club concerts, for instance, but not enough to record a train engine driving by at a distance of only 1.5 meters. I think this was probably the right way to go, because ultimately, the only way I'd be able to exploit a 96dB dynamic range would be to lock myself in a soundproof room with a 4000W amplifier and a large speaker stack. I have never encountered a situation where the actual dynamic range of the recording, from the ambient noise floor of the environment to the loudest sound encountered, was greater than between 60-70dB [and those situations are pretty exceptional, actually, like being 1.5m away from a train driving by at 80km/h in the middle of the country where there's no one and no other traffic around]. I'm probably just confusing this, though. Let me round it down to this: If you think you're going to be recording sounds louder than 105dB most often, go with low sens. If you think you'll more often be recording quiet sounds like birdsong, go with high sens.

Yes, all encoding/transcoding done by SonicStage is in stereo. It's possible that 2nd-gen units might play back mono mp3s correctly.

Type-R is only 7 years old [released in 1998].

I knew I'd be using the mics for a variety of applications, and the few extra dB of undistorted SPL [allowed by the low-sens mic] was more important to me.

Note: for reference, the center '-' above the record meters on HiMD recorders is -12dBfs.

Thank you to Michael1980 for passing on this link in another thread: http://www.minidisct.com/forum/showthread....gacy#post298121 It appears that 2nd-gen may be able to write MDLP discs from SS, but not record them in realtime, contrary to what I thought before [no MDLP writing at all].

Hopefully this is correct; it would be nice to at least have some backward-compatibility. Thanks for the link.