Laurent le faucheur (@laurentlefaucheur)

PhD in Speech and Signal Processing Working for consumer markets since 1990 Firmware programmer on ARM/DSP Founder at http://firmware-developments.com/

Re: IIR filters

Reply posted 6 years ago (12/22/2017)
As mentioned by  dgshaw6 audio telephony Codecs are using IIR, the main reason being the complexity and lower group-delay compared to FIR interpolator/decimator....

Re: Audio speed changer without changing the pitch

Reply posted 6 years ago (12/22/2017)
Hi, you should look at PSOLA (pitch synchronous overlap-and-add) description (https://en.wikipedia.org/wiki/PSOLA).
You can measure DR using a sinewave of fixed frequency and varying amplitude from MAX to 0. You use a notch filter (or FFT) to analyze the processed output.The...

Re: How to calculate time delay estimation?

Reply posted 7 years ago (04/10/2017)

Re: How to calculate time delay estimation?

Reply posted 7 years ago (04/10/2017)
HiI filtered your data to keep the bands 800-1200 and 2400-3400. Then downsampled it to 11.025kHz. The program below gives a correlation peak at 18k samples (4190x48k/11,...

Re: PDM asynchronous sample rate conversion

Reply posted 7 years ago (03/22/2017)
Hi Andrew,In a 3MHz PDM stream there is high energy content above 20kHz (the "noise transfer function" of the PDM modulator). Each time you drop PDM samples, you...

Re: Downsampling from 2.0 MHz to 192kHz

Reply posted 8 years ago (10/26/2016)
Hi Tom,As suggested by mike your decimation uses the ratio 12/125.We are describing our solution at https://community.arm.com/people/laurentlefaucheur...In your...

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