Trying to convert the AudioFilter_Callback_BF706Mini example from IIR to FIR.
Got the IIR version running OK even though "iirdf1_init(stateLeft, df1_coeffs, delayLeft, NUM_STAGES);" (and the equivalent right channel version) gave a "could not be resolved" error.
Modified the IIR version to remove the pushbutton functionality and to place all the code in a single file.
Replaced IIR calls with FIR calls (still get "could not be resolved" error for "fir_init(stateLeft, coeffs, delayLeft, NUM_COEFFS, 0);" (and equivalent right channel version).
Added the coefficients for a 45 tap low pass FIR filter, Kaiser window, 48KHz sampling frequency, pass band cut off 1kHz, stop band cut off 5kHz, 60dB stop band attenuation - derived from
NI's LABVIEW software.
Modifying the AudioFilter function to provide a loop-through function (no filtering) works OK so the codec callbacks etc. seem to be working.
When I enable the filtering function of AudioFilter by adding "fir_fr32(inLeft, outLeft, NUM_SAMPLES, &stateLeft);" (and similar for the right channel version) I get no output.
What I expected to get was a similar output to that from the IIR version (which also has a 1kHz cut off frequency).
So I've messed up somewhere along the line. Could anyone point me in the right direction ? As always, any help appreciated.
