[sdiy] Novation peak NCOs
Rutger Vlek
rutgervlek at gmail.com
Fri Apr 21 17:19:29 CEST 2017
Hi Richie,
I was referring to this article in Sound on Sound: http://www.soundonsound.com/techniques/mqa-time-domain-accuracy-digital-audio-quality. One of the best articles I read in a long time!
Regards,
Rutger
On 21 apr 2017, at 14:49, rburnett at richieburnett.co.uk wrote:
> Which issue of SOS?
>
> On 2017-04-21 13:02, Rutger Vlek wrote:
>> Hi Richie,
>> Indeed it concerns FM, mostly. And perhaps, theoretically, the low
>> end tightness. I recently read a great article in sound on sound about
>> a new HD audio codec that aims to improve aliasing as well as low
>> frequency phase issues and ringing that normally come with a typical
>> anti alias filter in or post a dac. I assume novation's approach also
>> improves that. Although i have no idea how therotical this is: if it
>> can be measured, if it makes an audible difference.
>> Best,
>> Rutger
>> Op 21 apr. 2017 13:13 schreef <rburnett at richieburnett.co.uk>:
>>> Yeah, it's an interesting way to tackle the aliasing issue. As Tom
>>> and others pointed out, raising the sample rate up to 24MHz doesn't
>>> completely eliminate aliasing for naïve generated sawtooths and
>>> pulses, but the high sample rate means that by the time the
>>> harmonics alias back into the audio spectrum they're at least 60dB
>>> down, and probably more like 75dB down for the highest pitches that
>>> people typically play in a music intended for human consumption
>>> (4kHz/24Mhz).
>>> The NCO/DDS technique isn't new. It was used in the SID, Paul
>>> Maddox's excellent Monowave and 002 synths, and I believe it was
>>> used in the "DCO" chip in Roland's alpha-juno series of hybrid
>>> synths back in 1985. I've had discussions with Tom W about this
>>> off-list and did some tests on my alpha juno and we came to the
>>> conclusion that the "DCO" chip in these synths generated the
>>> waveforms digitally using an NCO outputting at a sample rate of
>>> 6MHz. So I guess that the increase of sample rate from 6MHz to
>>> 24Mhz in 32 years isn't unfeasible!
>>> With the alpha juno you can clearly see the aliased harmonics of
>>> very high pitched notes on a spectrum analyser at about -58dB
>>> relative to the fundamental. However, I believe there's also a
>>> trick where you can apply a *very tiny* amount of random Frequency
>>> Modulation to the NCO that causes imperceptible pitch modulation in
>>> the audio part of the spectrum. However, it is enough to spread the
>>> energy in those upper harmonics around 24MHz (that alias into the
>>> audio band) so that they blend into each other and produce a lower
>>> and more even white noise floor rather than a series of
>>> objectionable inharmonic spikes across the audio spectrum. I would
>>> imagine that Novation are doing this too, since -60dB alias
>>> suppression is nothing spectacular compared to what can be achieved
>>> at a sample rate of 96kHz using BLIT/BLEP/whatever on a general
>>> purpose DSP.
>>> Rutger wrote "I'm particularly interested in how high-frequency
>>> interactions between NCOs are going to sound on this machine. I
>>> suspect FM and RingMod will be pretty smooth, as these interactions
>>> make aliasing effects more pronounced." I think you are right and
>>> this is where the technique actually shines. RingMod isn't a
>>> problem, but FM produces an infinite spectrum of harmonics, so
>>> whatever sample rate you use it will alias to some extent, but the
>>> higher the better obviously.
>>> The part that I thought was particularly clever was the dithering of
>>> the oscillator waveforms at 24MHz sample-rate down to a single-bit
>>> "bitstream" / "DSD" output within the FPGA, to mitigate the need for
>>> a high-speed multi-bit DAC that can settle fast enough to handle the
>>> 24MHz sample-rate. Again, the dithering and noise-shaping stuff
>>> isn't anything new, but it all seems to fit well together as a
>>> package for their application.
>>> As for the reverb being done in the FPGA, that's almost surely the
>>> best way. General purpose DSP chips aren't optimised for reverb
>>> (except maybe the FV-1 and Wavefront chips) but you can build the
>>> DSP core that you wish you had with an FPGA and make it optimised
>>> for hundreds of allpass filters or whatever you want for your
>>> reverb.
>>> Definitely an interesting time for synth design :-)
>>> -Richie,
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