Hello Kevin
I am not convinced that your application totally compares to a
continously changed sampling rate, but anyway:
The maths stays the same, so you will have to respect Nyquist and take
the artifacts of your AA filter as well as your signal processing into
account. This means you might use a sampling rate significantly higher
than the highest frequency to be represented correctly and this is the
edge frequency of the stop band of your AA-filter.
For a wave form generator in an industrial device, having similar
demands, we are using something like DSD internally and perform a
continous downsampling / filtering. According to the fully digital
representation no further aliasing occurs. There is only the alias from
the primary sampling process, held low because of the high input rate.
What you can / must do is an internal upsampling, since I expect to
operate with normal 192kHz/24Bit input (?)
Regarding your concerns: It is a difference if you playback the stream
with a multiple of the sampling frequency, especially with the same
frequency, performing modulation mathematically or if you perform a
slight variation of the output frequency, such as with an analog PLL
with modulation taking the values from a FIFO. In the first case, there
is a convolution with the filter behaviour of you processing, in the
second case, there is also a spreading, according to the individual
ratio to the new sampling frequency.
From the view of a musical application, case 2 is preferred, because
any harmonics included in the stream , such as the wave table, can be
preprocess, easier controlled and are a "musical" harmonic. In one of my
synths I operate this way, that all primary frequencies come from a PLL
buffered 2 stage DDS accesssing the wave table with 100% each so there
are no gaps and jumps in the wave table as with classical DDS.
j
Am 01.06.2018 um 04:03 schrieb Kevin Chi:
Dear List,
Long time lurker here, learned a lot from the random posts, so thanks
for those.
Maybe somebody can help me out what is the best practice for realtime
applications to minimize aliasing when
scanning a waveform by changing speed or constantly modulating the delay
time on a delay (it's like the
resampling rate is changing by every step)?
I guess the smaller the change, the smaller the aliasing effect is, but
what if the change can be fast enough to make it
audible?
If you can suggest a paper or site or keywords that I should look for,
I'd appreciate the help!
Kevin
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