Hi!
This is my first post here, although I've spent many hour browsing this forum over the years and learned lots of things here!
Anyway, here is my issue:
I’m contemplating a project that requires one or two simple delay lines with decent audio quality. My problem is that I need a minimum delay time of ~0.5ms or shorter, including the latency of the system itself. Also, the delay time needs to be continuously variable with high resolution, so I’ll probably have to sample the input as fast as possible. From what I’ve gathered, this rules out the audio library and it’s block based processing scheme, as well as the audio board, because of the delay in the ADCs/DACs filters.
Since all the filtering and mixing will be done with analogue circuitry, there’s not a lot left to do for the processor. Get a value from the ADC, feed it into a ring buffer, lookup the correct position for the read pointer, interpolate between two adjacent samples and pass the result to a DAC, that’s about it.
Here’s my questions:
Do you think that per-sample processing at, let’s say 196kHz is a realistic goal for a otherwise simple task like this? If so, what kind of converters should I be looking at?
Apart from dabbling around with the teensy+audio board or the Axoloti, I don’t have any experience with embedded DSP, especially when it comes to the hardware setup, so I’d be happy for every suggestion!
Thanks in advance,
Robin
This is my first post here, although I've spent many hour browsing this forum over the years and learned lots of things here!
Anyway, here is my issue:
I’m contemplating a project that requires one or two simple delay lines with decent audio quality. My problem is that I need a minimum delay time of ~0.5ms or shorter, including the latency of the system itself. Also, the delay time needs to be continuously variable with high resolution, so I’ll probably have to sample the input as fast as possible. From what I’ve gathered, this rules out the audio library and it’s block based processing scheme, as well as the audio board, because of the delay in the ADCs/DACs filters.
Since all the filtering and mixing will be done with analogue circuitry, there’s not a lot left to do for the processor. Get a value from the ADC, feed it into a ring buffer, lookup the correct position for the read pointer, interpolate between two adjacent samples and pass the result to a DAC, that’s about it.
Here’s my questions:
Do you think that per-sample processing at, let’s say 196kHz is a realistic goal for a otherwise simple task like this? If so, what kind of converters should I be looking at?
Apart from dabbling around with the teensy+audio board or the Axoloti, I don’t have any experience with embedded DSP, especially when it comes to the hardware setup, so I’d be happy for every suggestion!
Thanks in advance,
Robin