void AudioFilterFIR::update(void)
{
audio_block_t *block,*b_new;
// If there's no coefficient table, give up.
if(coeff_p == NULL)return;
// do passthru
if(coeff_p == FIR_PASSTHRU) {
// Just passthrough
block = receiveReadOnly(0);
if(block) {
transmit(block,0);
release(block);
}
return;
}
// Left Channel
block = receiveReadOnly(0);
// get a block for the FIR output
b_new = allocate();
if(block && b_new) {
if(arm_fast)
arm_fir_fast_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
else
arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
// send the FIR output to the left channel
transmit(b_new,0);
}
if(block)release(block);
if(b_new)release(b_new);
}
// the core dsp function
void dsp(int16_t* buffer, int length)
{
// only enable the filter if the user button is pressed
if (user_mode & 1)
{
// we initiate the filter only if needed to prevent clitches at the beginning of new buffers
if (firstStart == false || old_user_mode != user_mode)
{
initFilter();
old_user_mode = user_mode;
firstStart = true;
}
// process with FIR
arm_fir_fast_q15(&FIR, buffer, outSignal, BLOCKSIZE);
// copy the result
arm_copy_q15(outSignal, buffer, length);
}
}
