Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
Not long ago I made an extensive search for software that would do at least decoding of tape noise reduction systems. Practically nothing found, no convolution based or VST:s. No Dolby, no DBX, not even versions of single-ended analog NR, such as Philips DNL. An earlier version of the Stereo Tool VST had some kind of Dolby B decoding. http://www.stereotool.com/ ...but it isn't there anymore. I have guesses why. :-) Anyway, it didn't do the job exactly as Dolby B should. That's why I have Dolby 361 and DBX180 in my shelf. Eero ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
Pretty much the 'Holy Grail' of audio.I can't ever imagine there being a software solution for those noise reduction systems as they were very level sensitive... but i'd be really happy if i could be proven wrong.. Not long ago I made an extensive search for software that would do at least decoding of tape noise reduction systems. Practically nothing found, no convolution based or VST:s. No Dolby, no DBX, not even versions of single-ended analog NR, such as Philips DNL. An earlier version of the Stereo Tool VST had some kind of Dolby B decoding. http://www.stereotool.com/ ...but it isn't there anymore. I have guesses why. :-) Anyway, it didn't do the job exactly as Dolby B should. That's why I have Dolby 361 and DBX180 in my shelf. Eero ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130502/e22353e8/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
Hi, Dolby A (etc.) decode. Would it not be possible to to do this with convolution ?? Find a working unit, record its impulse response, use that in one of the many convolution reverb/filter plug-ins. While doing that record the encode impulse response for those who want to use it as an effect (e.g. on backing vocals). Obviously the Dolby reference level would have to be taken into account. Apologies if this is a completely hare-brained idea. Dave Hunt Date: Mon, 29 Apr 2013 21:56:30 -0500 From: David Pickett d...@fugato.com Subject: Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X) A standalone Windows app that would decode Dolby-A encoded wavefiles and output a restored non-Dolby 24-bit wavefile would be most useful. I have several recordings that I have had transfer to hi-res files still in Dolby-A format. ... even if such a program were command line only and needed to be left overnight to cook! David Date: Tue, 30 Apr 2013 10:18:37 +0100 From: Dave Malham dave.mal...@york.ac.uk Subject: Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X) snip One possibility would be to use Spice to model the circuit and just process the sound files through the model. Slower than a directly written program, but probably usable for archival work. Dave -- Date: Tue, 30 Apr 2013 10:39:27 +0100 From: Dave Malham dave.mal...@york.ac.uk Subject: [Sursound] Dolby A (was Re: Sony PCM? (was Re: DTS Headphone:X)) Dolby A used sided chained compression of signals on record (and corresponding expansion on replay) so only low level signals where affected - over the Dolby Level there was essentially no affect on the signals and they passed straight through. A good deal of psychoacoustic work was done to ensure that artefacts such as noise pumping were well hidden, including splitting up the signal into several frequency bands so that signals in one band didn't affect others and time constants could be optimised. Telecom C4 also split into bands (for the same reasons) but didn't sidechain so compression was applied at all signal levels. This meant you got more noise reduction but also more artefacts, though not as much as DBX which used neither sidechain nor bands. Dolby A, because of its sidechains had to be aligned to within half a dB of the correct level or the process went wrong because the compression thresholds went out but neither C4 nor DBX needed anything other than aligning to optimise machine dynamic range as they had no thresholds. This would seem to make Dolby A more difficult to use but as every Dolby A unit incorporated a proper Dolby Tone generator which produced a unique warbling test tone that all Dolby A tapes should have on them, this shouldn't be a problem. Dave ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 2013-05-01, Dave Hunt wrote: Dolby A (etc.) decode. Would it not be possible to to do this with convolution ?? Find a working unit, record its impulse response, use that in one of the many convolution reverb/filter plug-ins. Unfortunately this is not an option. Convolution can be used to model any system that is linear and time invariant, that is, sums and constant multiples of inputs, even if shifted arbitrarily in time, lead to sums, multiplications and equal shifts, of the resulting outputs. This does not hold for compressors or expanders, including multiband ones like the compander architecture in the various kinds of Dolby NR: even if in very short term they try to behave roughly linearly so as not to add audible nonlinear distortion and they don't have time variance like e.g. a tremolo effect, most decidedly even their short term frequency response varies and addition and multiplication shift the signals over the different knees of the compander which is a visible nonlinearity. Obviously the Dolby reference level would have to be taken into account. Every time there is a reference level in a system that actually impacts how it operates, the system is guaranteed to be nonlinear, because otherwise you could freely multiply the signal by some number a before the system and by 1/a after the system without changing how the system sounds. LTI systems are a broad and useful class of systems, with a nice theory and beautiful computational properties, but they don't cover all of audio signal processing by a long shot. The systems you can simulate via convolution include constant gain, filters/equalisers with constant settings, echo, delay, reverb and all of their combinations. Linear but not time invariant systems include chorus, phasing, flanging, wah, tremolo, vibrato and like things, so they are also out for convolution, unless you continuously change the impulse response you're convolving with (usually a bad idea because convolver topology is heavily optimized for constant coefficients). Then stuff that is decidedly nonlinear is doubly out: dynamics processing, fuzz/distortion, anything of that sort. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On Wed, May 01, 2013 at 08:02:20PM +0300, Sampo Syreeni wrote: On 2013-05-01, Dave Hunt wrote: Dolby A (etc.) decode. Would it not be possible to to do this with convolution ?? Find a working unit, record its impulse response, use that in one of the many convolution reverb/filter plug-ins. Unfortunately this is not an option. Convolution can be used to model any system that is linear and time invariant While I agree with the 'not an option', the motivation is not entirely to the point. Convolution processors *can* be used to model non-linear systems, e.g. guitar amps producing lots of distortion, or an analog telephone channel. Angelo Farina wrote a very interesting paper about that some years ago. The point is that companders such as Dolby-A and Telcom are linear at least over a short time span - they do not introduce distortion. But they are certainly not time-invariant - they would be useless if they were. The compander gains (and hence the IR) depend on the current input signal and its short term history. Now even time-variant systems can be emulated using convolution, but the two issues are orthogonal: using convolution processing of the audio signals does not help in any way to simplify the implementation of the dynamics. The difficult part in writing any software emulation of the Dolby-A or similar systems is modelling the dynamic behaviour of the compander, not the actual audio processing. Such systems will have 'designed' and documented attack/release times, but analog electronics being what they are, the dynamic behaviour of the compressors and expanders will very probably not fit to any simple equations. Nor does it have to: the way Dolby-A and Telcom switch between encoding and decoding (by making the decoding algorithm the mirror image of the encoding) guarantees that the two will cancel each other, whatever they are. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
At 16:02 01-05-13, Fons Adriaensen wrote: The difficult part in writing any software emulation of the Dolby-A or similar systems is modelling the dynamic behaviour of the compander, not the actual audio processing. Such systems will have 'designed' and documented attack/release times, but analog electronics being what they are, the dynamic behaviour of the compressors and expanders will very probably not fit to any simple equations. Nor does it have to: the way Dolby-A and Telcom switch between encoding and decoding (by making the decoding algorithm the mirror image of the encoding) guarantees that the two will cancel each other, whatever they are. So, analog still rules! David ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 2013-05-01, Fons Adriaensen wrote: The point is that companders such as Dolby-A and Telcom are linear at least over a short time span - they do not introduce distortion. I started thinking about that as well. Yeah, if you put in all possible spectra, in all combinations, at all speeds of change, you can characterize a system like this rather perfectly and prolly emulate it. It's just that the amount of data is stupendous and the problem you're trying to solve is probably harder than implementing the thing by hand to begin with. But they are certainly not time-invariant - they would be useless if they were. As a technical nitpick, they are too: if you just shift a signal in time, the output will be a shifted version of the original. I.e. they do have memory and if you deside to linearize them and then talk about their instantaneous system function, yes, that one changes in a regular, weakly nonlinear way based on what they've seen in the past. Still, all in all if you only do time shifts, they don't mind. The same is not true even of tremolo as a still-obviously-linear system, because it has an independent oscillator in it; by definition time-variance means the system is sensitive to absolute delay, not only relative,, which is not the case for any dynamics processor I know of. (It is for certain reverbs though, which use internal LFO's to break up resonances.) But yes, there's usually little point in treating homogeneity, additivity and time/shift invariance as separate things within the audio field. They interact strongly enough so that usually you either take the whole LTI package or go back to the basics. The difficult part in writing any software emulation of the Dolby-A or similar systems is modelling the dynamic behaviour of the compander, not the actual audio processing. Such systems will have 'designed' and documented attack/release times, but analog electronics being what they are, the dynamic behaviour of the compressors and expanders will very probably not fit to any simple equations. Yes. Or let's say, the sidechain where they do all of their psychoacoustical analysis and control functions. That stuff is pretty complex too, at least when we go to systems like SR. (A is somewhat easier, because it's apparently just a multichannel compander. Though, I don't seem to have access to the fundamental papers anymore. Care to slip me a copy of at least the AES ones?) Nor does it have to: the way Dolby-A and Telcom switch between encoding and decoding (by making the decoding algorithm the mirror image of the encoding) guarantees that the two will cancel each other, whatever they are. Yeah, Dolby apparently has been doing that stuff for ages: they derive the inverse of their designs very much like you'd turn an IIR filter to its FIR inverse, and vice versa. They essentially sandwich the encoder within a negative servo loop in the decoder, be the result nonlinear or not. That approach is pretty general and simple too, if you know what you're doing: even if you want to formally show that it converges, you don't need much besides asymptotically limited memory in the encoder and a bunch of monotonicity assumptions. (I don't think they've ever shown this formally, though.) To me the funkiest thing is that for the longest time Pro Logic was one of their bigger sources of income, yet that one broke the precedent: it was an open loop design. Only in Pro Logic II did they go back to their old feedback balance network ways. I'm guessing that's because there the problem is underdetermined, which makes stability and self-synchronizing tracking considerably more difficult to guarantee. Be as it may, that architecture makes the precise nonlinear dynamics difficult to characterize and emulate, except by direct translation of the circuit. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On Wed, May 01, 2013 at 04:29:46PM -0500, David Pickett wrote: At 16:02 01-05-13, Fons Adriaensen wrote: The difficult part in writing any software emulation of the Dolby-A or similar systems is modelling the dynamic behaviour of the compander, not the actual audio processing. Such systems will have 'designed' and documented attack/release times, but analog electronics being what they are, the dynamic behaviour of the compressors and expanders will very probably not fit to any simple equations. Nor does it have to: the way Dolby-A and Telcom switch between encoding and decoding (by making the decoding algorithm the mirror image of the encoding) guarantees that the two will cancel each other, whatever they are. So, analog still rules! For some value of 'rules' :-) Fact is that some simple analog circuits are quite difficult to emulate exactly in software, for the simple reason that they are 'imperfect'. Take a simple attack time circuit in a compressor. Could be a condensor being charged by a diode and resistance in series. The diode is not 'perfect', it will have some voltage drop depending on the current flowing in it. The result may well be 'desirable', i.e. in practice better than what would be obtained by using a 'perfect' diode. But it means that code such as if (x vc) vc += a * (vc - x) which assumes a 'perfect' diode, will not be an accurate model of the analog circuit for low values of x. And making it more accurate requires the code to be *much* more complicated. And that's only a simple case. Many real-life analog circuits have several of such 'imperfections' interacting with each other. None of this means that it's impossible to emulate analog audio electronics in software - just that it could be some orders of magnitude more complicated that what one would expect. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On Thu, May 02, 2013 at 12:46:36AM +0300, Sampo Syreeni wrote: As a technical nitpick, they are too: if you just shift a signal in time, the output will be a shifted version of the original. That corresponds to a very weak definition of 'time-invariant'... Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 2013-05-01, Fons Adriaensen wrote: The diode is not 'perfect', it will have some voltage drop depending on the current flowing in it. The result may well be 'desirable', i.e. in practice better than what would be obtained by using a 'perfect' diode. Perhaps the nastiest commonly used example of this I know of is biasing diodes (often using negative feedback to keep them there) right into the middle of the knee region where they're starting conduct. There the voltage-current-characteristic is locally exponential, so that you can derive logarithms and exponentials in the analog domain when you linearly amplify the local behavior. Then you can do analog computer things like multiplication via accumulation on the resulting voltages, and convert back. There's about a million things which can go wrong with this kind of an unstable setup. And it does. And then some people like it and build entire circuit topologies to amplify the defect into a unique, sellable product. If you want to emulate something like that, you can't just look at the circuit board, because there you'd be like o_O, WTF?!? Seriously, analog engineers do the weirdest things, some of which might not even be described in the diagrams e.g. for trade secret reasons. (I for example know of an analog synth whose secret still appears to be safe, because it has to do with intentional parasitics from a nearby coil, never put in the patent diagram.) In this vein the starting point prolly would be the Moog ladder, in its nonlinear region. It's hideously complex for such a charming little thing. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 2013-05-01, Fons Adriaensen wrote: As a technical nitpick, they are too: if you just shift a signal in time, the output will be a shifted version of the original. That corresponds to a very weak definition of 'time-invariant'... I didn't think there could be but one: either it varies in time or it does not. Of course you can always talk about time-invariance in intuitive terms, and that's fine for most kinds of technical work. But when you get down to it, eventually you have to operationalize your concept and know which formal, mathematical properties it has. Preferably those properties will also be somehow orthogonal to the rest of the concepts you use, such as linearity (or, actually, homogeneity+additivity, which linearity means). My definition simply says an operator F from some suitable closed class of operators over functions over an additive group is shift invariant, if for all f belonging to the domain of F, f(x+a)==F(f)(x+a). That is a useful definition which lets you do general, often useful math with the property, and it's nicely several from most other properties you might then use at the same time. In most cases, where you can somehow formally measure the relative sizes of the sets of operators with the property against those without, there are vanishingly few operators with the property, so that the property is pretty restrictive. In the fully discrete world, I seem to remember there are many more additive homomorphisms around, for example, than shift invariant ones in this sense. So it's rather restrictive at least in that sense. Still, do give me your definition of time-invariant. Perhaps there are stronger definitions I haven't heard of yet, and which can be useful in e.g. more fine grained analysis. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 30 Apr 2013, at 04:56, David Pickett wrote: A standalone Windows app that would decode Dolby-A encoded wavefiles and output a restored non-Dolby 24-bit wavefile would be most useful. I have several recordings that I have had transfer to hi-res files still in Dolby-A format. ... even if such a program were command line only and needed to be left overnight to cook! Being a fan of doing things the easy way, I'd recommend just buying a Dolby Model 363 NR unit which does both A type and SR. At any point in time there are typically a dozen or so available on Ebay for prices in the range of $150 to $300. It's difficult to model something like Dolby NR in DSP because the algorithm is defined by a circuit. You would need to very carefully benchmark a working decoder in any case because neither the patent or the JAES article really tell you how to do it. ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
True, the patent and JAES article are as usual not the full story. The only problem with the second user stuff is whether the thing itself is working properly. Access to a test rig would be useful - never used the 363 but I've had a lot of experience with the earlier 361 unit which uses the CAT 22 Dolby A card. We had the full test rig that you interposed between the card and the main body of the 361. I found it essential to run each card through test at least a couple of times a year - not that there were a lot of faults, but there were some and with the devices being thirty years old. Dolby ran a wonderful service where you phoned up, told them you had a faulty card and they would ship you a replacement without waiting for your one to arrive back with them and at no cost to you apart from post. Even in York this meant that the most you were without a channel was 24 hours - I can remember cycling to York Railway Station to pick up units direct off the train! In London the turn round was a couple of hours and the courier would take the faulty unit back so not even the cost of return postage (or rail freight). Amazing trust and service, wonder if anyone does that for anything these days! One possibility would be to use Spice to model the circuit and just process the sound files through the model. Slower than a directly written program, but probably usable for archival work. Dave On 30 April 2013 08:55, Eric Benjamin eb...@pacbell.net wrote: On 30 Apr 2013, at 04:56, David Pickett wrote: A standalone Windows app that would decode Dolby-A encoded wavefiles and output a restored non-Dolby 24-bit wavefile would be most useful. I have several recordings that I have had transfer to hi-res files still in Dolby-A format. ... even if such a program were command line only and needed to be left overnight to cook! Being a fan of doing things the easy way, I'd recommend just buying a Dolby Model 363 NR unit which does both A type and SR. At any point in time there are typically a dozen or so available on Ebay for prices in the range of $150 to $300. It's difficult to model something like Dolby NR in DSP because the algorithm is defined by a circuit. You would need to very carefully benchmark a working decoder in any case because neither the patent or the JAES article really tell you how to do it. ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- As of 1st October 2012, I have retired from the University, so this disclaimer is redundant These are my own views and may or may not be shared by my employer Dave Malham Ex-Music Research Centre Department of Music The University of York Heslington York YO10 5DD UK 'Ambisonics - Component Imaging for Audio' -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130430/bd677040/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
i work in an ethnomusicology archives here in india, and we have about 5000 hours of pcm f1 recordings dating back to 1983 (when I even did digital field recordings using the F1). we have two F1s and one 501. None have digital outputs, but cricklewood electronics (?) produced a card which produced a sdif signal out of the F1 that we are using to convert our pcm recordings to wav files without redigitising. umashankar Date: Mon, 29 Apr 2013 15:02:32 +0100 From: richarddob...@blueyonder.co.uk To: sursound@music.vt.edu Subject: [Sursound] Sony PCM? (was Re: DTS Headphone:X) On 29/04/2013 11:53, Jon Honeyball wrote: I have a pcm-f1 tape of the Zuccherelli stuff from 30 years ago, for those with long memories. Must pull that into a wav file, but my f1 has no digital output. Hmmm As it happens, I have a Sony PCM-701ES (incl SPDIF) with the CDP digital port added (designed, along with the SoundSTreamer it connected to, by Dave Malham), sitting around doing nothing. It cost several arms and legs when bought new, back in 1987. Last time I used it, several years ago, it recorded 16bit audio nicely to a bog-standard (and cheap) VHS recorder. My question is, simply, are these things still in use/demand anywhere (e.g. for recovering vintage F1 recordings, which I merely assume it can do)? I also used the CDP port to connect (via a tiny bit of DIY buffer electronics) to a now utterly obsolete but cute IDE-based 56001 dsp development card. The port gives you direct access to the otherwise internal serial data and clock lines. There is currently one on Ebay Buy It Now for £150 plus shipping. I guess shipping by UK courier would be around £25. I would only take the plunge on Ebay if I could be sure of a price good enough to justify letting it go. Richard Dobson ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130429/6c6a7dc8/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
A few months ago I had to sort out a PCM701 with one of my spdif cards in (the ones I used to do for Audio Design). We went through three PCM units before we found one that worked fully in replay mode (the original, one from the Uni and the final one, off Ebay). There seems to be something in the electronics that becomes increasingly unreliable with time in, I think, the clocking circuits which I find very worrying especially. Given the fact that archiving houses (in the UK in particular, the British Library) very sensibly bought up a lot of machines when HHB finally stopped sponsoring production, there aren't likely to be many working boxes around any more, so guard any you have that work very carefully! Dave PS I don't have any more of the spdif and the chip it was based on is no longer available so I can't make any more! On 29 April 2013 15:23, umashankar manthravadi umasha...@hotmail.comwrote: i work in an ethnomusicology archives here in india, and we have about 5000 hours of pcm f1 recordings dating back to 1983 (when I even did digital field recordings using the F1). we have two F1s and one 501. None have digital outputs, but cricklewood electronics (?) produced a card which produced a sdif signal out of the F1 that we are using to convert our pcm recordings to wav files without redigitising. umashankar Date: Mon, 29 Apr 2013 15:02:32 +0100 From: richarddob...@blueyonder.co.uk To: sursound@music.vt.edu Subject: [Sursound] Sony PCM? (was Re: DTS Headphone:X) On 29/04/2013 11:53, Jon Honeyball wrote: I have a pcm-f1 tape of the Zuccherelli stuff from 30 years ago, for those with long memories. Must pull that into a wav file, but my f1 has no digital output. Hmmm As it happens, I have a Sony PCM-701ES (incl SPDIF) with the CDP digital port added (designed, along with the SoundSTreamer it connected to, by Dave Malham), sitting around doing nothing. It cost several arms and legs when bought new, back in 1987. Last time I used it, several years ago, it recorded 16bit audio nicely to a bog-standard (and cheap) VHS recorder. My question is, simply, are these things still in use/demand anywhere (e.g. for recovering vintage F1 recordings, which I merely assume it can do)? I also used the CDP port to connect (via a tiny bit of DIY buffer electronics) to a now utterly obsolete but cute IDE-based 56001 dsp development card. The port gives you direct access to the otherwise internal serial data and clock lines. There is currently one on Ebay Buy It Now for £150 plus shipping. I guess shipping by UK courier would be around £25. I would only take the plunge on Ebay if I could be sure of a price good enough to justify letting it go. Richard Dobson ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130429/6c6a7dc8/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- As of 1st October 2012, I have retired from the University, so this disclaimer is redundant These are my own views and may or may not be shared by my employer Dave Malham Ex-Music Research Centre Department of Music The University of York Heslington York YO10 5DD UK 'Ambisonics - Component Imaging for Audio' -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130429/92752c61/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
At 10:59 29-04-13, Dave Malham wrote: A few months ago I had to sort out a PCM701 with one of my spdif cards in (the ones I used to do for Audio Design). We went through three PCM units before we found one that worked fully in replay mode (the original, one from the Uni and the final one, off Ebay). There seems to be something in the electronics that becomes increasingly unreliable with time in, I think, the clocking circuits which I find very worrying especially. Is this perhaps (hopefully) just old Cs? David (Luckily I transferred all my F1 tapes to DAT, and recently to HD) ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
It is possible but I wasn't at all sure that it wasn't the ceramic resonator used in the oscillator. Sooner or later we may have to resort to writing software to do the job - assuming we can find working Betamax machines. Fortunately the encoding is very well documented in the various manuals, unlike some of the more modern systems. Dave On 29 April 2013 17:21, David Pickett d...@fugato.com wrote: At 10:59 29-04-13, Dave Malham wrote: A few months ago I had to sort out a PCM701 with one of my spdif cards in (the ones I used to do for Audio Design). We went through three PCM units before we found one that worked fully in replay mode (the original, one from the Uni and the final one, off Ebay). There seems to be something in the electronics that becomes increasingly unreliable with time in, I think, the clocking circuits which I find very worrying especially. Is this perhaps (hopefully) just old Cs? David (Luckily I transferred all my F1 tapes to DAT, and recently to HD) __**_ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/**mailman/listinfo/sursoundhttps://mail.music.vt.edu/mailman/listinfo/sursound -- As of 1st October 2012, I have retired from the University, so this disclaimer is redundant These are my own views and may or may not be shared by my employer Dave Malham Ex-Music Research Centre Department of Music The University of York Heslington York YO10 5DD UK 'Ambisonics - Component Imaging for Audio' -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130429/c67c59da/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 29/04/2013 16:59, Dave Malham wrote: A few months ago I had to sort out a PCM701 with one of my spdif cards in (the ones I used to do for Audio Design). We went through three PCM units before we found one that worked fully in replay mode (the original, one from the Uni and the final one, off Ebay). There seems to be something in the electronics that becomes increasingly unreliable with time in, I think, the clocking circuits which I find very worrying especially. Given the fact that archiving houses (in the UK in particular, the British Library) very sensibly bought up a lot of machines when HHB finally stopped sponsoring production, there aren't likely to be many working boxes around any more, so guard any you have that work very carefully! Dave PS I don't have any more of the spdif and the chip it was based on is no longer available so I can't make any more! Interesting - time for some testing. Last time I turned it on, it all worked, but that was at least 5 years ago, or approx when said cheap video recorder started chewing tapes. I have forwarded this to Archer Endrich; he may still have a few kits lying around. We are still scratching our heads about what to do with all this old CDP kit! Clarification: I just checked it, my model is the PCM 601-ESD (not the 701), it has its own spdif i/o as well as the CDP port. Richard Dobson ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
FWIW, I've got a few working systems, F1, 601, 701 and two Betamax VCRs (PAL only). The 601 and 701 have a SPDIF output. They are still used in our studio for clients who require transfers. Kees de Visser Galaxy Classics On 29 Apr 2013, at 17:59, Dave Malham wrote: A few months ago I had to sort out a PCM701 with one of my spdif cards in (the ones I used to do for Audio Design). We went through three PCM units before we found one that worked fully in replay mode (the original, one from the Uni and the final one, off Ebay). There seems to be something in the electronics that becomes increasingly unreliable with time in, I think, the clocking circuits which I find very worrying especially. Given the fact that archiving houses (in the UK in particular, the British Library) very sensibly bought up a lot of machines when HHB finally stopped sponsoring production, there aren't likely to be many working boxes around any more, so guard any you have that work very carefully! Dave PS I don't have any more of the spdif and the chip it was based on is no longer available so I can't make any more! ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 29 Apr 2013, at 18:42, umashankar manthravadi wrote: many years ago, I tried to convince people it is worth producing a software PCM F-1 decoder, using a low cost video card and a a VHS player (all our PCM F1 recordings are on VHS). I thought it would be simple, but nobody showed any interest. Umashankar A good friend of mine is a gifted DSP programmer and I remember having asked him years ago if he could make what you describe. He probably could, but it's not easy, will take many hours to develop and the potential user base is very small. Have you checked if there are any (Sony) patents that could pose problems ? Great idea though (same for a software Dolby A/SR decoder, which isn't avaialble AFAIK). Kees de Visser Galaxy Classics ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
Hi, It would have been difficult in t'owld days using dsp's with their (then) very small memories and the requirement to use assembler because of the general lack of high performance high level language compilers. Pretty certain it could be done with an ARM with C or C++ these days. However, it would still take quite a bit of work for a pretty small market. If it ever gets done it'll be by someone who is either (a) a masochist or (b) in desperate need with a pile of vital tapes and no pcm units available. There wouldn't be any problem with patents for the PCM units as they were first marketed 30 years ago which would put patent dates at 21 years or more so they would have expired - Dolby A (which was introduced in 1966), Dolby B (1968) and Dolby SR (1986) all fall into the same category. Dave On 29 April 2013 18:37, Kees de Visser k...@galaxyclassics.com wrote: On 29 Apr 2013, at 18:42, umashankar manthravadi wrote: many years ago, I tried to convince people it is worth producing a software PCM F-1 decoder, using a low cost video card and a a VHS player (all our PCM F1 recordings are on VHS). I thought it would be simple, but nobody showed any interest. Umashankar A good friend of mine is a gifted DSP programmer and I remember having asked him years ago if he could make what you describe. He probably could, but it's not easy, will take many hours to develop and the potential user base is very small. Have you checked if there are any (Sony) patents that could pose problems ? Great idea though (same for a software Dolby A/SR decoder, which isn't avaialble AFAIK). Kees de Visser Galaxy Classics ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound -- As of 1st October 2012, I have retired from the University, so this disclaimer is redundant These are my own views and may or may not be shared by my employer Dave Malham Ex-Music Research Centre Department of Music The University of York Heslington York YO10 5DD UK 'Ambisonics - Component Imaging for Audio' -- next part -- An HTML attachment was scrubbed... URL: https://mail.music.vt.edu/mailman/private/sursound/attachments/20130429/54666ae8/attachment.html ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 2013-04-29, Dave Malham wrote: Pretty certain it could be done with an ARM with C or C++ these days. Yes, though the data rate of the incoming video stream is a bit steep (20-30MB/s), so that you need a rather muscular DSP to keep up, and you probably won't want to go the easiest way which would be to use some existing software to capture the uncompressed video and then write an offline program to decode it. I'm not too sure there are standard formats for uncompressed video which let you do reliable separation of specific scanlines either, which is what is needed here. And you definitely don't want the pain of working directly with the baseband video signal -- though the necessary SDR code might be included in GNU Radio or some similar toolkit, and theoretically you can ignore chroma. However, it would still take quite a bit of work for a pretty small market. If it ever gets done it'll be by someone who is either (a) a masochist or (b) in desperate need with a pile of vital tapes and no pcm units available. To me it seems getting the tapes accepted into some library's preservation program would be the easiest way to get the funding for the initial development. Or, perhaps Sony would be interested, given that there have to be a number of interesting and important masters out there which they could benefit from if (re)mastered. There wouldn't be any problem with patents for the PCM units as they were first marketed 30 years ago which would put patent dates at 21 years or more so they would have expired - Dolby A (which was introduced in 1966), Dolby B (1968) and Dolby SR (1986) all fall into the same category. That's distinctly easier because the formats aren't such a kluge, and if you just want to do one-time migration, your code wouldn't have to be ultra optimized either. (If you can just throw cycles at it, analog filters are easy to emulate with an evil enough oversampling ratio.) Level calibration could prove a bit of a challenge there, and there might be nastier restorative actions you might want to take at the same time, like locking onto bias phase, but otherwise it doesn't seem like rocket surgery to me. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
At 11:34 29-04-13, Dave Malham wrote: It is possible but I wasn't at all sure that it wasn't the ceramic resonator used in the oscillator. What mechanism causes deterioration in one of these? David ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
At 12:37 29-04-13, Kees de Visser wrote: Great idea though (same for a software Dolby A/SR decoder, which isn't avaialble AFAIK). A standalone Windows app that would decode Dolby-A encoded wavefiles and output a restored non-Dolby 24-bit wavefile would be most useful. I have several recordings that I have had transfer to hi-res files still in Dolby-A format. ... even if such a program were command line only and needed to be left overnight to cook! David ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
Re: [Sursound] Sony PCM? (was Re: DTS Headphone:X)
On 30 Apr 2013, at 04:56, David Pickett wrote: A standalone Windows app that would decode Dolby-A encoded wavefiles and output a restored non-Dolby 24-bit wavefile would be most useful. I have several recordings that I have had transfer to hi-res files still in Dolby-A format. ... even if such a program were command line only and needed to be left overnight to cook! The DSP friend I mentioned before has written a software Telcom C4 decoder for a client (custom made, not for sale). Telcom was a German (Telefunken) tape noise reduction system, equivalent (claimed superior) to Dolby. Just to say that it can be done if there's (financial) interest. IIRC Telcom was less critical than Dolby regarding playback calibration, so decoding Dolby wav files might be a bit more complex. Kees de Visser Galaxy Classics ___ Sursound mailing list Sursound@music.vt.edu https://mail.music.vt.edu/mailman/listinfo/sursound
