[sdiy] Using dual BBD chips for higher clock frequency

Mike Bryant mbryant at futurehorizons.com
Sun Jun 26 22:36:02 CEST 2022


Hello Hannes

'adaptive differential modulation' or ADPCM was the original attempt at reducing the bit size of digitised audio compared with the PCM used first in the phone system and the in CDs.   It used roughly half the memory for the same quality, or you could reduce further if you wish.  There were numerous devices using this technique, and I recall one in either Wireless World or Practical Electronics in the early 80s.


From: Synth-diy [mailto:synth-diy-bounces at synth-diy.org] On Behalf Of hannes.janetzek--- via Synth-diy
Sent: 26 June 2022 21:25
To: SDIY
Subject: Re: [sdiy] Using dual BBD chips for higher clock frequency

Hello everyone! - it's my first post to this list. I'm also quite new to learning electronics from studying old hardware. I still recall BBDs were one of the things that quite stunned me when first encoutering them in the Casio CT-5000 ensemble effect - from this year :)

The most recent finding in this regards is the digital reverb of the Technics SX-E66 and SX-G7 which uses 16 M5225L chips for modulation/demodulation with 8KB of ram. Schematic and diagram is here https://imgur.com/a/n4S4JsF https://imgur.com/a/HYFyNbU if anyone is curious. This chip is only mentioned by one patent (Low-pitched sound creator) and seems to do 'adaptive differential modulation'.
I haven't investigated this module in detail yet but was wondering if someone knows this chip and if this technique is used elsewhere. (Just posting this here since similar approaches have been suggested and it's appears to be ok to extend on the original topic)

Best
Hannes

Am So., 26. Juni 2022 um 16:46 Uhr schrieb Michael E Caloroso via Synth-diy <synth-diy at synth-diy.org<mailto:synth-diy at synth-diy.org>>:
A BBD delay is never going to have stellar fidelity.

Well, neither are echos or modulated delay effects that occur in nature.  In a real room or in valleys, the reflections of the original signal is going to be bandwidth limited and distorted.

My Yamaha E1010 sucks at modulated delays and the serial BBD architecture (four at longest delay!) with heavy filtering hurt the fidelity of the original signal.  But i exploit that "defect" for Haas effects - by panning the direct signal hard right and panning the 15-25ms delayed signal hard left, this creates a pseudo "stereo" image.  Haas effects work really well with guitars as it creates a "hole" in the middle of the stereo field which lets vocals be heard better without the mix getting overwhelmingly loud.  Since the delayed signal is sonically different (there's a REASON why Yamaha put EQ controls on the E1010), the Haas effect sounds better than on my other delay units, even digital.  It gets closer to a single guitar sounding like two distinct guitars.  There's a REASON why Yamaha included a direct output on the rear panel.

Many chorus algorithms bandwidth limit the delayed signal intentionally.  Frequencies below 100hz tend to muddy the chorus effect in a mix.  Limiting upper frequencies can improve the effect, this is a subjective area based on instrument (or vocals) and player preferences.

Imperfections >can< be a good thing.

MC

On Sat, Jun 25, 2022 at 6:29 AM Richie Burnett <rburnett at richieburnett.co.uk<mailto:rburnett at richieburnett.co.uk>> wrote:
The BBD with fixed-length and variable- sample-rate does respond to modulation differently to a fixed-rate variable-length delay. There's a DAFX paper about this, and others too from what I can remember. Our very own Tom Wiltshire also did an excellent analysis of how a BBD responds to modulation. Although it stopped short of specifically discussing how to model this digitally in a fixed sample-rate system.

From my experience it's not hard to model a short BBD with slow gentle modulation using a variable-length digital delay and fixed sample-rate. You can get very close to a typical BBD chorus effect with a fixed rate DSP with the right IIR filtering, etc. However...

It's the cases with a longer delay and deep, fast modulation that do the interesting stuff! This is harder to model digitally because the modulation changes whilst the signal is still shuffling through the buckets of the BBD! One paper suggested using a standard fixed rate digital delay-line for the audio alongside a "virtual BBD" that stored time-stamps of when the virtual bucket's samples were taken rather than actual audio data. The appropriate output samples could then be generated by using the time-stamps to extract values from the fixed-rate delay using interpolation. I haven't tried it but think I understand how it works!?

-Richie,

---- brianw wrote ----

>I don't know whether any company ever shipped a BBD-based product that clocked high enough to even get 20 kHz bandwidth, much less 500 kHz. Most BBD products have less than full bandwidth, such that even us old folks can hear the loss.
>
>The MAX1326 ADC clocks up to 526 kS/s and is 14 bits.
>
>The ADS7951 ADC clocks up to 1 MSPS and is 12 bits.
>
>The THS5671 DAC clocks up to 125 MSPS and is also 14 bits.
>
>I've long been curious about how much of a factor the continuously-variable sample rate of a BBD (or non-audio CODEC) is compared to the sound of a fixed-sample-rate digital design (even with software SRC to simulate variable sample rates). i.e. I wonder if the 'sound' of a BBD isn't its analog nature, or the sample rate, per se, but the fact that the sample rate changes are continuous.
>
>Brian
>
>
>On Jun 16, 2022, at 6:01 PM, Jay Schwichtenberg via Synth-diy <synth-diy at synth-diy.org<mailto:synth-diy at synth-diy.org>> wrote:
>> Isn't one of the advantages of using a BBD over a digital solution is the clocking can go to 500 KHz to 1MHz?
>>
>> So depending on the clock freq you can get the aliasing filters higher up and have better freq range.
>>
>> Jay S.
>>
>> On 6/16/2022 4:38 PM, Tom Wiltshire wrote:
>>> On 16 Jun 2022, at 23:11, Mike Bryant <mbryant at futurehorizons.com<mailto:mbryant at futurehorizons.com>> wrote:
>>>> You could add more BBDs in parallel and sum the outputs to reduce it again.
>>> I think this is what's referred to in the SAD1024 datasheet. That chip was actually a dual 512-stage device, and one suggestion was to use the two delay lines in parallel for reduced noise. I don't remember the details. Could be that simply summing the signals gives you a +6dB boost in the signal, but not so much in the noise, or it could be that it recommended a differential signal. Can't remember, sorry. I found an "Archer"-branded version of the datasheet here which shows various possibilities:
>>>
>>> http://www.pmerecords.com/Docs/Archer_SAD-1024_Tech_Data.pdf
>>>
>>>> If lowest clock speed is 20k, you really need to be filtering out everything about 10k.
>>> BBD datasheets generally recommend a conservative bandwidth of 1/3rd of the clock frequency, so 20K would be 6.67KHz. Poor, but much better than 3.3KHz!!
>>>
>>> I totally agree that 18KHz filtering with 20KHz clock is not realistic. No practical filter is that sharp.
>>>
>>> Tom
>
>
>_______________________________________________
>Synth-diy mailing list
>Synth-diy at synth-diy.org<mailto:Synth-diy at synth-diy.org>
>http://synth-diy.org/mailman/listinfo/synth-diy
>Selling or trading? Use marketplace at synth-diy.org<mailto:marketplace at synth-diy.org>

_______________________________________________
Synth-diy mailing list
Synth-diy at synth-diy.org<mailto:Synth-diy at synth-diy.org>
http://synth-diy.org/mailman/listinfo/synth-diy
Selling or trading? Use marketplace at synth-diy.org<mailto:marketplace at synth-diy.org>
_______________________________________________
Synth-diy mailing list
Synth-diy at synth-diy.org<mailto:Synth-diy at synth-diy.org>
http://synth-diy.org/mailman/listinfo/synth-diy
Selling or trading? Use marketplace at synth-diy.org<mailto:marketplace at synth-diy.org>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://synth-diy.org/pipermail/synth-diy/attachments/20220626/fd8cd840/attachment.htm>


More information about the Synth-diy mailing list