Update AudioFilterFIR for audio_block_samples

feature_setBlockSize
Chip Audette 8 years ago
parent f13598b968
commit 556d9d92ab
  1. 87
      AudioFilterFIR_F32.h

@ -22,38 +22,38 @@
class AudioFilterFIR_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
public:
AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
}
void begin(const float32_t *cp, int n_coeffs) {
coeff_p = cp;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], AUDIO_BLOCK_SAMPLES);
//if (arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], AUDIO_BLOCK_SAMPLES) != ARM_MATH_SUCCESS) {
// n_coeffs must be an even number, 4 or larger
//coeff_p = NULL;
//}
}
}
void end(void) {
coeff_p = NULL;
}
virtual void update(void);
void setBlockDC(void) {
//helper function that sets this up for a first-order HP filter at 20Hz
}
private:
audio_block_f32_t *inputQueueArray[1];
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
// ARM DSP Math library filter instance
arm_fir_instance_f32 fir_inst;
float32_t StateF32[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
public:
AudioFilterFIR_F32(void): AudioStream_F32(1,inputQueueArray),
coeff_p(FIR_F32_PASSTHRU), n_coeffs(1), configured_block_size(0) { }
//initialize the FIR filter by giving it the filter coefficients
void begin(const float32_t *cp, const int _n_coeffs) { begin(cp, _n_coeffs, AUDIO_BLOCK_SAMPLES); } //assume that the block size is the maximum
void begin(const float32_t *cp, const int _n_coeffs, const int block_size) { //or, you can provide it with the block size
coeff_p = cp;
n_coeffs = _n_coeffs;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != FIR_F32_PASSTHRU) && n_coeffs <= FIR_MAX_COEFFS) {
arm_fir_init_f32(&fir_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], block_size);
configured_block_size = block_size;
}
}
void end(void) { coeff_p = NULL; }
virtual void update(void);
//void setBlockDC(void) {} //helper function that sets this up for a first-order HP filter at 20Hz
private:
audio_block_f32_t *inputQueueArray[1];
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
int n_coeffs;
int configured_block_size;
// ARM DSP Math library filter instance
arm_fir_instance_f32 fir_inst;
float32_t StateF32[AUDIO_BLOCK_SAMPLES + FIR_MAX_COEFFS];
};
@ -78,14 +78,21 @@ void AudioFilterFIR_F32::update(void)
return;
}
// get a block for the FIR output
b_new = AudioStream_F32::allocate_f32();
if (b_new) {
arm_fir_f32(&fir_inst, (float32_t *)block->data, (float32_t *)b_new->data, block->length);
AudioStream_F32::transmit(b_new); // send the FIR output
AudioStream_F32::release(b_new);
}
AudioStream_F32::release(block);
// get a block for the FIR output
b_new = AudioStream_F32::allocate_f32();
if (b_new) {
//check to make sure our FIR instance has the right size
if (block->length != configured_block_size) {
//doesn't match. re-initialize
Serial.println("AudioFilterFIR_F32: block size doesn't match. Re-initializing FIR.");
begin(coeff_p, n_coeffs, block->length);
}
//apply the FIR
arm_fir_f32(&fir_inst, (float32_t *)block->data, (float32_t *)b_new->data, block->length);
AudioStream_F32::transmit(b_new); // send the FIR output
AudioStream_F32::release(b_new);
}
AudioStream_F32::release(block);
}
#endif

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