I wrote before about the Accuphase T-1000 and about my interest about it. Since I record FM it is supers beneficial to me that Accuphase has digitalized multiplex decoder with digital out before the output DA converters. This would allow sucking out of tuner a row digital stream bypassing the habitual output stage and a separate A/D converter. The benefits are unquestionable and the only concern that I had if the Accuphase T-1000 might be the tuner to do the job.

The Accuphase T-1000 is 4 varactors design that is very much not the same as a number of well implemented LC section but for the urban environments where I can fry a chicken on antenna it might do. Let parented that whatever they did in the Accuphase is fine and then the subject of my “dilemma” become clear. Here is the key.

The Accuphase T-1000 runs after a detector A/D processing into 24/192. Then the multiplexing done at digital domain with understandable bits lost during DSP processing. The T-1000 outputs 16/48 to it’s SPIDIF.  The 48kHz is 1X for 192 kHz which is fine. I presume that 48kHz sampling is sufficient to care the 16khz restricted signal. My primary concern is about the 16 bit rate. A played a lot with A/D conversion and I know that the 16 bit is the last bit rate that I would use. I have a number of good A/D processors (Lavry AD122 and Pacific) and with both of then I get way better results at 20 bit then at 16 (Pacific works much better at 16 bit then Lavry). So, to me in my recording endeavor the 16 bit is a big no-no…

So, lately I had an opportunity to have access to Accuphase T-1000 and to try it. However, learning about the fact that it sends out just 16 bit I made the decisions do not even exercise this opportunity. Come on who the hell need a devised with output stream of 16 bit?

However a couple of days back I have a conversation with people I know who advised me that my judgments about the inferiority of 16bit of Accuphase’s out was idiotic. According to them that 96dB of 16bit’s dymick range is more then enough to cover the limited by nature FM dymick range. Furthermore FM own noise always cover the younger bits with noise (or the natural dithering), so if the youngest 16 bit are already in nose then what the purpose to add more bits? To have more dynamic range of noise?

It sound very reasonable and from the argument above it very much might be that 16bit of raw and not processed after recording stream very much might be sufficient for FM. Honestly, I do not have an opinion at this point.

I posted this request at the Yahoo FM group, the place reportedly owned by the Reichsführer-SS Bob Fitzgerald, I have mention that Moron-wanna-be-in-charge before. Thankfully that cretin does not show up there and the air in that bulletin blog is free from his stench. Other folks in there looks like fine. Anyhow, here are some of the  responses that I got there on subject.

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The way I understand it, the number of bits mainly correlates to the maximum signal to noise ratio...or noise floor in a linear digital system.  The sampling rate defines the absolute frequency response limit.

16 bits should be capable of a noise floor that's at least 90db down from 100% modulation (75khz of deviation).

I think it's unlikely that you can get a noise floor in the real world of FM reception that would exceed 90db.  Most stations don't have their noise floor that far down anyway....especially not in stereo...although some (but very few) are close.

So, (IMHO), I'd say 16 bits should be enough resolution.  If more bits are easily available, that's fine, but 16 should be at or beyond the rest of the signal paths noise floor.

A 44khz sampling rate would limit you to a maximum frequency response of about 20khz.  Since FM is limited to about 15khz, a 44khz sampling rate would be enough.  Moving the sampling rate higher does allow for less abrupt filtering however, and some folks think that sounds better.  But all the frequencies transmitted will be reproduced easily with a 44khz sampling rate.

Hope this helps a little,

Dave O.

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A few years ago I needed a way to record FM broadcast in order to do A/B comparisons of small adjustments I was making to the processor of an FM public radio station. In order to do this accurately, I needed a recorder whose reproduced sound was very close (if not identical) to the original being recorded. Of course, I was at the "mercy" of the A/D and D/A converters in the recorder, as well as its limitations of bits and sample rate.

So there were several independent contributing variables. Suffice it to say that, in my case, I was unable to come up with a 16bit 48kHz recording that exactly preserved the definition and sound to my ears. For example I tried a Panasonic SV-3800 DAT recorder at 16bits, 48kHz. (Close, but not quite exact.)

I did hit the "jack pot" with an old Alesis ADAT Type II XT20 20-bit Digital Audio Recorder. With it running at 20-bits, 96kHz, I don't think I could tell the difference between the FM source and the digital recording. It was extremely close, and good enough to call "identical" for my purposes. So I made my tests and comparisons that way, using the old Alesis ADAT 20-bit recorder.

The above experience is as close as I can come to answering your question.

In your case, the advantage of the Accuphase T-1000's direct DSP MPX processing and AES/EBU (or SPDIF?) output might be worth it despite its 16-bit 44.1kHz rate? It would be an intriguing experiment I'd like to run myself.

-Greg

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Yes, Greg, what you say is direct reflection of my experiences. David Obergoenner is correct there is no justification to have more than 16 bit and 44K for recording dynamically and bandwidth restricted FM broadcasts. I intellectually agree but practically I am not sure.

A few years back when I was experimented with idea of FM recording I was playing good FM broadcasts into my amps and compared it to a chain with tuner-A/D-PC-D/A-amps. I was shooting to the situation where the sound from tuner direct and the sound from digitalizing chain would be absolutely not distinguishable, and I am taking in context of quite serious audio installation the can handle some complex aspects of audio reproductions. So, despite my using some very offensive AD and DA possessor I was not able to get good result at 16 bit. I had much better sound at 20 bit and 44K and I ended up with 24/88K. The Accuphase T-1000 is SPDIF 16/48K but on another side it has one less A/D processor and one less output stage – go figure how it might be. It would be interesting to know if they have more than 16 bits after multiplexing. If tay do then whatever the Accuphase T-1000 sound not that it might sound much better (I am sure my D/A is better then what is in the stock Accuphase). So, perhaps someone shall start to raise this issue and in a year or so Accuphase would do the Makr II with 20 bit resolution?  THAT would be very interesting… Unfortunately the US Accuphase representative is virtually dead and looks like he is a clueless Moron. As a result the user's interests are not well presented with Accuphase new tuner… Anyhow, I still contemplate if I need to drift toward the digital multiplexing and digital out. The idea is very lucrative and very elegant, I juts would like to have 20 bit for sake of intellectual comfort. I like different multiplex decoders, I have 4 very high quietly sounding decoders and I would have more if I know anything better is available.  Since I record FM to digital the idea of having digital after UHF, after detector, I think is very promising. Just give me 20 bits out….

Romy

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Hello,

it is a very interesting question.

In my opinion 16 bits should be sufficient,in fact I think this is a lower boundary for FM digitalization. I merely agree with your opinion that FM noise will works as a "natural dithering" in other words it will acts as the "noise smearing" technique "modulating" the rough nature of the 16 bits. A similar technique plus noise shapping and versampling ones was used by PHILIPS in their low cost 14 bits DACs to increase the resolution to a 16 bits equivalent one.

A very similar technique is Super Bit Mapping in CD leading to 18-20 bits equivalent resolution. If I remember it well the optimal "smearing" noise amplitude should be approx. 3 times higher than LS bit amplitude. If we get as a maximal FM tuner reachable S/N ratio of 87-90dB(A-weighted)and as a CD maximal S/N ratio of 100dB(A-weighted)we can see that the difference between them is nearly the same these 3times(10dB). That's why I think the 16 bits are sufficient, of course this 16 bits
stream must be then processed by using of adequate oversampling and noise shapping techiques to get a good result.

Best regards,
Cvetan

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I would not use any less than 24bits/88Ks/s for quality stereo recording, but you already know this

My take on this is that EVERY A/D conversion introduces a loss of audio quality for various reasons.

The first question is what is the actual effective number of bits in the A/D convertor?  I get around 18bits out of my cheap Crystal delta-sigma chip, you get close to 23bits with your best pro convertors. The T1000 is around 19 or 20bits.

The next question is what is the bit width of the data path used in the T1000? I doubt its better than 24bits. Any Accuphase engineers around to counter? You need at least a 48bit fixed point path to maintain sound quality (Rane used to have a app note about data path width on its website)

DSP is funny in that round off errors can be exacerbated by different math operations. So you need a very large data path all the way thru processing until the last step or D/A conversion. Then you choose from various dithers,filters ,etc. All of those leave a distinct sonic fingerprint when downsampling to 16bit. On top of this, all processing must be real time which limits smple rate and resolution. No thank you!

What type of input filtering is used, slow roll off or sharp? This is fixed by the AD chip

How is the AD chip clocked? Dedicated low jitter clock or slaved from DSP PLL clock (YUM!)?

Ideally you could bypass all these internal "compromises" by recording the composite signal directly from the Pulse Count Detector. This would require 1 channel of quality external 24bit/192ks/s A/D conversion. You would then "post-process" the data into stereo using the massive DSP power of modern GPUs (nvidia CUDA for example). I know of one DIYer that is using CUDA code for multichannel 4 way crossovers in REAL TIME so post processing to only 2 channel MPX would be much faster, maybe even 5-10X real time. GPUs typically have 128 bit data paths so data integrity and avoidance of overflows is much better than a 48bit DSP

If you code it, then you can choose resolution, sample rate,dither functions, and filters that work best with YOUR D/A convertor.

Of course if you have managed to get this far you don't need a T1000, could use the detector out of a TU-X1

Please keep in mind that I am limiting this "extreme overkill is no vice" type discussion to the very best programming and station quality. There is no point in doing this for commercial radio! Theoretically, 14bits/38ks/s is what generally considered "FM quality". So much for theory..if you want that then get a Sony hd radio.

-J

Yes, it is what it is. I spoke with Dima last night and he passed his view on the subjects. He does not agree that 24 bit really exists in out 24 bit converts. He feels that we get 20-22 bit in the very best cases. He also think that all our 16-bit converted are in fact 14 bits. Interesting…

BTW, here I think a fascinating chart about the depth of bits and sampling rates. I would live the semi-stupid satisfaction column aside…

http://www.newformresearch.com/fidelity-potential-index.htm

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This all brings us back to fm_login's original question. (By the way, fm_login, what's your first name?) Speaking from first-hand experience, I've found that the best broadcast analog stereo generators (e.g., the Orban 8100) easily pass greater definition than redbook (44.1/16) when your source is better than redbook. For this reason, if I could wave my magic wand, I'd have all FM radio stations use source material better than redbook (e.g., 24/96) and I'd have them convert their entire chains (including STLs, if digital) to that higher definition. But, unfortunately for me, it isn't going to happen. (I know, Dave O., you were probably holding your breath regarding this pie-in-the sky wish of mine!) We're lucky these days to have stations running uncompressed 44.1/16 all the way through their chains, with no sample-rate conversions nor down-conversions.

We do still have a few, smaller public radio stations around the country who are still running excellent sounding all-analog chains capable of conveying considerably more definition than redbook. (I think somebody in this forum posted a while back about such a station in San Francisco who have deliberately kept their chain all-analog, and who play vinyl, for that reason?) I personally like the sound of "old-fashioned" FM stations running very high-quality analog chains - and especially where they don't need STLs because their transmitters are co-located with their transmitters. (Not a very practical, realistic situation, but there are still a few such stations around the country.)

fm_login, my personal experience recording the best FM broadcast is as follows: If, to do it, you'll have additional A/D and D/A conversions in the path, I find that 20/96 (or 24/96) sounds a little closer (than 44.1/16) to the analog outputs of your best tuners that you're recording. However, per Dave O.'s point, if you're recording directly from a 44.1/16 (or 48/16?) digital output of the Accuphase, then I'd expect your best result by recording at exactly that rate (with no sample rate or bit conversions, nor any unnecessary additional A/D nor D/A conversions). This is my expectation, although I don't have an Accuphase T-1000 to try it first-hand.

Greg

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An FM signal contains the band limited and compressed analog conversion of an
analog input from whatever source. The FM channel resolution is less than a
16/44 digital equivalent. Can you explain what is the effect of having extra
bits and sampling rates in the digital representation of the source before
conversion, filtering and limiting? Where and how can the additional resolution
be hidden in the signal and then revealed in the output (this would be a
necessary condition for claims of the ability to hear the additional resolution
of the source)? The answer could have significant implications in information
theory (of course we are not talking about digital FM).

Al

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This whole thread makes me feel a bit like Alice. In any case I
can't resist adding my own prejudices and preconceptions to the
wisdom accumulating in this thread.

1. I agree that changing the sample rate of the source is unlikely to
have a positive impact on the quality. However increasing the bit
depth seems like it would be totally harmless, as long as you don't
later have to decrease it again, in which case I would think it would
have been better to have just left it alone in the first place.

2. Analog FM transmitters operate at a sample rate of 38 kHz, so the
T1000 is already making at least one sample rate conversion to output
a 44.1 or 48 ksps stream.

3. It was my perhaps misguided impression that many modern DACs
actually change the sample rate internally as part of their principle
of operation, even when feed from a source operating at their
"native" sample rate.

4. I have to wonder how many sample rate conversions are involved in
a modern "analog" FM transmission, considering the original source,
studio equipment, studio to transmitter link, audio processing at the
transmitter, and finally the digitally implemented "analog" FM
modulator in the FM transmitter? Can anyone enlighten me, and the
rest of us, about this issue? I also note that when a whole chain of
digital audio devices all operate at the same nominal sample rate,
some devices do what amounts to a sample rate conversion to convert
the input signal clock to match the phase and frequency of a fixed
internal sample clock.

John

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Al, somewhere below you made an excellent remark: The answer could have significant implications in information
theory. Sure the answer would not make any dense in information theory but the question itself bring a paradox to the table. There were a number of excellent comments made but I did not get any new “knowledge” or revelation from the thread. To summarize how I see the things it would be the following.

1)     There is not technical justification why 44/16 shall not be sufficient to handle FM signal

2)     In contrarily to #1 the reproduced FM signal is greatly benefited with extra 2 bit of resolution and 2X sampling rate (I use multibit A/D and D/A)

3)     In accordance with #2 the T1000 most likely restrict itself with 16 digital output but might have advantage by the fact that A/D output strange is not in signal path.

4)     T1000 might be at disadvantage by the fact that it needs to A/D composite signal at 192K. With higher sampling rate all problems of accuracy come to the play, particularly within the cheap implementation.

5)     It is not know how T1000 arrives to 16Bit from 24 bit if initial conversion. If they round the left over it then it is not good.

6)     It is not clear how the fast converter lives alone with RF stages in T1000.  I spend a LOT of efforts to isolate my digital form my RF. I do not see in T1000 those efforts. Would the T1000 give 75dB noise?

Anyhow, I think that market for FM tuner with digital output would grow, at least among the freaks who do FM. I use external A/D but if to do the MPX decoding on digital domain then the design begs to out digital.  I would like to hear more about digital decoders, this architecture and the prospect to outs 20 bit out of tuners.  I think the right way to answer it is to commission a digital multiplex decoder and feed it right from detector.  I very mildly consider it but… my RF playback, even as it is now, is way much capable then my Boston stations can furnish, so it does not make since to me. Still, the T1000 is available within my rich and I would like to decide for myself if it is intellectually “cleanly designed” unit.