Eric Brombaugh (@emeb)
I've seen schematics for active noise cancellation done entirely in analog and it's remarkably simple to get basic operation. I am curious how it's done in consumer-grade...
There are a lot of optimizations you can make but most of them are only necessary if you're working in hardware (ASIC or FPGA design) and need to minimize resources....
For the curve fitting approach: You need to generate a family of responses, basically a 2D surface with FFT bin weights in one axis and cutoff frequency setting...
Option 1) is attractive as it would imply simply shifting a single fixed response up or down. You'd want to evaluate your filter response across its whole range...
This sent me down a rabbit-hole. For 1-bit -> 16-bit you'd need a 4^15 decimation rate - roughly 2 billion, so that's clearly not what's going on in a PDM->PCM...
Something doesn't smell right here: Wouldn't a factor of 64 bandwidth reduction only gain you 3 bits of precision?
I haven't designed 2-arm IIR filter style Hilbert transforms from first principles myself - I assume most folks looking for such would likely begin with common...
It's common to use a pair of allpass IIR filters with +/-45deg shifts as an analytic signal converter (aka Hilbert transform) that can generate a complex (quadrature)...
By what mechanism do you anticipate crosstalk?All signal processing after the ADC is done in the digital domain. Unless there's something seriously wrong with the...
Here's how I'd approach this: Assuming that your ADC is sampling just a real signal, the complete spectrum of this can be viewed as 0-500MHz of the real signal,...
Yes, I grabbed the clip you posted and listened to it. I also looked at it in time & frequency with Audacity. It seems to have a fairly strong peak around 377Hz...
I just built a musical instrument tuner that's based on phase vocoder principles and it seems to work pretty well on irregular waveforms for estimating frequency....
It would be very helpful if you could tell us which development tools you used that generated these errors. I do a lot of DSP work on STM32 and I've never come...
Seems like you really need to understand why your overlap/save code isn't running on the STM32 properly. Add some diagnostics to the audio processing code - toggle...
ARM's Cortex-M4F claims to have "DSP instructions" but it's nowhere near as cycle-efficient as a "real" DSP would be - no simultaneous operand buses, no auto looping...
Not strictly a true DSP, but I use STM32 ARM processors for a lot of audio DSP. Any of the families with Cortex M4F do quite nicely - even the low end STM32F3xx...
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