void AMDemod::update(void)
{
if (!pass) return;
audio_block_t *blocki, *blockq;
int16_t *pi, *pq, *end;
int32_t sum;
blocki = receiveWritable(0); // receives I, end result is written in channel 0
blockq = receiveReadOnly(1); // receives Q
if (!blocki) {
if (blockq) release(blockq);
return;
}
if (!blockq) {
release(blocki);
return;
}
pi = (int16_t *)(blocki->data);
pq = (int16_t *)(blockq->data);
end = pi + AUDIO_BLOCK_SAMPLES;
while (pi < end) {
// square and sum I and Q
sum = *pi * *pi;
sum += *pq * *pq;
// The result of squaring a 1.15 is 2.30.
// Shift the sum up one bit to make it 1.31 (Q31)
// and then that becomes the input to the arm sqrt function
sum <<= 1;
arm_sqrt_q31((q31_t)sum,(q31_t *) &sum);
// The result is in the high order 16 bits of sum
*pi++ = sum >> 16;
pq++;
}
transmit(blocki);
release(blocki);
release(blockq);
}
void AudioSynthDigiFilter::update(void)
{
audio_block_t *block, *modBlock;
uint16_t baseFreq,envIn;
int16_t modIn;
uint8_t pot1value;
block = receiveReadOnly(0);
modBlock = receiveReadOnly(1);
if (block)
{
if (modBlock)
{
//final frequency = baseFreq + envIn * envAmt + modIn * modAmt;
//following implementation isn't correct yet: negative envIns aren't possible
envIn = (block->data[64] * (envAmt)) >> 15;//32768 is the middle here, so we can have inverting and non in
modIn = (modBlock->data[64] * modAmt) >> 15;
baseFreq = freq;
int32_t totalfreq = baseFreq + envIn + modIn;
if(totalfreq > 65535) totalfreq = 65535;
if(totalfreq < 0) totalfreq = 0;
pot1value = totalfreq >> 8;
outvalue = pot1value;
digitalPotWrite(pot1value);
release(modBlock);
}
release(block);
}
}
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);
}
void AudioOutputI2SQuad::update(void)
{
audio_block_t *block;
block = receiveReadOnly(0);
if (block) release(block);
block = receiveReadOnly(1);
if (block) release(block);
block = receiveReadOnly(2);
if (block) release(block);
block = receiveReadOnly(3);
if (block) release(block);
}
void AudioAnalyzePeak::update(void)
{
audio_block_t *block;
const int16_t *p, *end;
int32_t min, max;
block = receiveReadOnly();
if (!block) {
return;
}
p = block->data;
end = p + AUDIO_BLOCK_SAMPLES;
min = min_sample;
max = max_sample;
do {
int16_t d=*p++;
// TODO: can we speed this up with SSUB16 and SEL
// http://www.m4-unleashed.com/parallel-comparison/
if (d<min) min=d;
if (d>max) max=d;
} while (p < end);
min_sample = min;
max_sample = max;
new_output = true;
release(block);
}
void AudioMixer2::update(void)
{
audio_block_t *in, *out=NULL;
unsigned int channel;
for (channel=0; channel < 2; channel++) {
if (!out) {
out = receiveWritable(channel);
if (out) {
int32_t mult = multiplier[channel];
if (mult != 65536) applyGain(out->data, mult);
}
} else {
in = receiveReadOnly(channel);
if (in) {
applyGainThenAdd(out->data, in->data, multiplier[channel]);
release(in);
}
}
}
if (out) {
transmit(out);
release(out);
}
}
void AudioOutputI2S::update(void)
{
// null audio device: discard all incoming data
//if (!active) return;
//audio_block_t *block = receiveReadOnly();
//if (block) release(block);
audio_block_t *block;
block = receiveReadOnly(0); // input 0 = left channel
if (block) {
__disable_irq();
if (block_left_1st == NULL) {
block_left_1st = block;
block_left_offset = 0;
__enable_irq();
} else if (block_left_2nd == NULL) {
block_left_2nd = block;
__enable_irq();
} else {
audio_block_t *tmp = block_left_1st;
block_left_1st = block_left_2nd;
block_left_2nd = block;
block_left_offset = 0;
__enable_irq();
release(tmp);
}
}
block = receiveReadOnly(1); // input 1 = right channel
if (block) {
__disable_irq();
if (block_right_1st == NULL) {
block_right_1st = block;
block_right_offset = 0;
__enable_irq();
} else if (block_right_2nd == NULL) {
block_right_2nd = block;
__enable_irq();
} else {
audio_block_t *tmp = block_right_1st;
block_right_1st = block_right_2nd;
block_right_2nd = block;
block_right_offset = 0;
__enable_irq();
release(tmp);
}
}
}
void AudioAnalyzePrint::update(void)
{
audio_block_t *block;
uint32_t offset = 0;
uint32_t remain, n;
block = receiveReadOnly();
if (!block) return;
while (offset < AUDIO_BLOCK_SAMPLES) {
remain = AUDIO_BLOCK_SAMPLES - offset;
switch (state) {
case STATE_WAIT_TRIGGER:
// TODO: implement this....
offset = AUDIO_BLOCK_SAMPLES;
break;
case STATE_DELAY:
//Serial.printf("STATE_DELAY, count = %u\n", count);
if (remain < count) {
count -= remain;
offset = AUDIO_BLOCK_SAMPLES;
} else {
offset += count;
count = print_length;
state = STATE_PRINTING;
}
break;
case STATE_PRINTING:
n = count;
if (n > remain) n = remain;
count -= n;
while (n > 0) {
Serial.println(block->data[offset++]);
n--;
}
if (count == 0) state = STATE_IDLE;
break;
default: // STATE_IDLE
offset = AUDIO_BLOCK_SAMPLES;
break;
}
}
release(block);
}
void AudioAnalyzeFFT256::update(void)
{
audio_block_t *block;
block = receiveReadOnly();
if (!block) return;
if (!prevblock) {
prevblock = block;
return;
}
copy_to_fft_buffer(buffer, prevblock->data);
copy_to_fft_buffer(buffer+256, block->data);
//window = AudioWindowBlackmanNuttall256;
//window = NULL;
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
// G. Heinzel's paper says we're supposed to average the magnitude
// squared, then do the square root at the end.
if (count == 0) {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] = magsq / naverage;
}
} else {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] += magsq / naverage;
}
}
if (++count == naverage) {
count = 0;
for (int i=0; i < 128; i++) {
output[i] = sqrt_uint32_approx(sum[i]);
}
outputflag = true;
}
release(prevblock);
prevblock = block;
}
void AudioOutputAnalog::update(void)
{
audio_block_t *block;
block = receiveReadOnly(0); // input 0
if (block) {
__disable_irq();
if (block_left_1st == NULL) {
block_left_1st = block;
__enable_irq();
} else if (block_left_2nd == NULL) {
block_left_2nd = block;
__enable_irq();
} else {
audio_block_t *tmp = block_left_1st;
block_left_1st = block_left_2nd;
block_left_2nd = block;
__enable_irq();
release(tmp);
}
}
}
void Beat_detector::update()
{
audio_block_t *block = receiveReadOnly();
if (!block) {
return;
}
const int16_t *p = block->data;
const int16_t *end = p + AUDIO_BLOCK_SAMPLES;
do {
m_energy += *p * *p;
if (m_count >= ENERGY_SAMPLES) {
memmove(m_history + 1, m_history,
sizeof(m_history) - sizeof(m_history[0]));
m_history[0] = m_energy;
m_energy = 0;
m_count = 0;
}
++p;
} while (p < end);
release(block);
}
void AudioOutputI2SQuad::update(void)
{
audio_block_t *block, *tmp;
block = receiveReadOnly(0); // channel 1
if (block) {
__disable_irq();
if (block_ch1_1st == NULL) {
block_ch1_1st = block;
ch1_offset = 0;
__enable_irq();
} else if (block_ch1_2nd == NULL) {
block_ch1_2nd = block;
__enable_irq();
} else {
tmp = block_ch1_1st;
block_ch1_1st = block_ch1_2nd;
block_ch1_2nd = block;
ch1_offset = 0;
__enable_irq();
release(tmp);
}
}
block = receiveReadOnly(1); // channel 2
if (block) {
__disable_irq();
if (block_ch2_1st == NULL) {
block_ch2_1st = block;
ch2_offset = 0;
__enable_irq();
} else if (block_ch2_2nd == NULL) {
block_ch2_2nd = block;
__enable_irq();
} else {
tmp = block_ch2_1st;
block_ch2_1st = block_ch2_2nd;
block_ch2_2nd = block;
ch2_offset = 0;
__enable_irq();
release(tmp);
}
}
block = receiveReadOnly(2); // channel 3
if (block) {
__disable_irq();
if (block_ch3_1st == NULL) {
block_ch3_1st = block;
ch3_offset = 0;
__enable_irq();
} else if (block_ch3_2nd == NULL) {
block_ch3_2nd = block;
__enable_irq();
} else {
tmp = block_ch3_1st;
block_ch3_1st = block_ch3_2nd;
block_ch3_2nd = block;
ch3_offset = 0;
__enable_irq();
release(tmp);
}
}
block = receiveReadOnly(3); // channel 4
if (block) {
__disable_irq();
if (block_ch4_1st == NULL) {
block_ch4_1st = block;
ch4_offset = 0;
__enable_irq();
} else if (block_ch4_2nd == NULL) {
block_ch4_2nd = block;
__enable_irq();
} else {
tmp = block_ch4_1st;
block_ch4_1st = block_ch4_2nd;
block_ch4_2nd = block;
ch4_offset = 0;
__enable_irq();
release(tmp);
}
}
}
void AudioAnalyzeFFT256::update(void)
{
audio_block_t *block;
block = receiveReadOnly();
if (!block) return;
#if AUDIO_BLOCK_SAMPLES == 128
if (!prevblock) {
prevblock = block;
return;
}
copy_to_fft_buffer(buffer, prevblock->data);
copy_to_fft_buffer(buffer+256, block->data);
//window = AudioWindowBlackmanNuttall256;
//window = NULL;
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
// G. Heinzel's paper says we're supposed to average the magnitude
// squared, then do the square root at the end.
if (count == 0) {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] = magsq / naverage;
}
} else {
for (int i=0; i < 128; i++) {
uint32_t tmp = *((uint32_t *)buffer + i);
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
sum[i] += magsq / naverage;
}
}
if (++count == naverage) {
count = 0;
for (int i=0; i < 128; i++) {
output[i] = sqrt_uint32_approx(sum[i]);
}
outputflag = true;
}
release(prevblock);
prevblock = block;
#elif AUDIO_BLOCK_SAMPLES == 64
if (prevblocks[2] == NULL) {
prevblocks[2] = prevblocks[1];
prevblocks[1] = prevblocks[0];
prevblocks[0] = block;
return;
}
if (count == 0) {
count = 1;
copy_to_fft_buffer(buffer, prevblocks[2]->data);
copy_to_fft_buffer(buffer+128, prevblocks[1]->data);
copy_to_fft_buffer(buffer+256, prevblocks[1]->data);
copy_to_fft_buffer(buffer+384, block->data);
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
} else {
count = 2;
const uint32_t *p = (uint32_t *)buffer;
for (int i=0; i < 128; i++) {
uint32_t tmp = *p++;
int16_t v1 = tmp & 0xFFFF;
int16_t v2 = tmp >> 16;
output[i] = sqrt_uint32_approx(v1 * v1 + v2 * v2);
}
}
release(prevblocks[2]);
prevblocks[2] = prevblocks[1];
prevblocks[1] = prevblocks[0];
prevblocks[0] = block;
#endif
}
void AudioSynthKS::update(void){
audio_block_t *block, *excitement;
int32_t s0, s1, s2, s3;
int32_t acc;
int i;
if (!running) return;
block = allocate();
if (block == NULL) return;
// If buflen has been reduced by a wavelength change & cursor is now outside of bounds, reposition cursor.
// while (cursor >= buflen) {
// cursor -= buflen;
// }
if (triggering) {
// We've just been asked to excite our buffer
triggering = false;
triggered = true;
triggerPoint = cursorPlus(buflen);
Serial.print("triggerPoint: ");
Serial.println(triggerPoint, DEC);
}
if (triggered) { //DEBUG
// read a block of input excitement (noise, or whatever you got)
excitement = receiveReadOnly(_exciter); // could be null?
if (excitement == NULL) {
Serial.println("z"); // DEBUG
release(block);
return; //DEBUG
}
}
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
if (triggered) {
block->data[i] = buffer[cursor] = excitement->data[i];
} else {
// K/S algorithm at its simplest:
// load the cursor sample & the one just behind it
// (In our travelling-buffer implementation, cursor-buflen == cursor)
s0 = buffer[cursorMinus(1)];
s1 = buffer[cursorMinus(buflen)];
// TODO: put a knob on this bug:
// Overshooting the beginning of the buffer throws tiny specs of random bugnoise into the tube ... which ends up sounding kind of nice
// Seems like values past 32 become insipid ... adjust exponentially.
//s1 = buffer[cursorMinus(buflen + 1)];
// Adding a third element creates ring-mod-type stuff, and also brings about some tremolo related to _bufmax.
// That would be a nice effect to control ...
s2 = buffer[cursorMinus(_magic1)];
//
// Also: when this is a fixed value (not relative to buflen) then crazy FM-sounding shit happens ...
// TODO: an absolute/pitched toggle for these knob vals.
//s2 = buffer[cursorMinus(128)];
// average them (comb filter),
// then mix them back into the buffer (using decayBalance),
// and put that in the output.
// block->data[i] = buffer[cursor] = ((s0 + s1) / 2);
// "drum variant" according to K and S?
acc = ((s0 + s1) / 2) ;
if (random(0,128) >= _drumness) // flip a coin
acc = 0 - acc; // negate the samplE
block->data[i] = buffer[cursor] = acc;
/* workingish but parked for now?
block->data[i] = buffer[cursor] = (int16_t)
// how much of the original sample (s0) to mix?
(((65535 - _decayBalance) * s0)>>16) // will be zero when decayBalance = 0xffff
+
// how much of the new value to mix?
((_decayBalance * ((s0 + s1) / 2) )>>16); // will be zero when decayBalance = 0;
*/
// This ought to do the same thing, but it causes noise and crashes ... wtf?
//block->data[i] = buffer[cursor] = (int16_t)((s0 + s1) * 0.5);
// Other things we could do instead:
// If we subtract instead of adding, the whole thing drops an octave & also sounds slightly more lowpass filtered ...
// I believe the octave thing is because the entire waveform is inverted on every pass, so it becomes a half-wave.
// The filter thing may be math rounding issues or similar ... otherwise i don't know. At any rate, this does sound nice.
// TODO: my kingdom for a 'scope ...
// block->data[i] = buffer[cursor] = ((s0 - s1) / 2);
// Here's the same thing redrawn in the more 'general' form of a digital comb filter:
// (feedbackGain * buffer[cursor - feedbackDistance]) + (feedforwardGain * buffer[cursor + feedforwardDistance])
// (with feedforwardDistance = 0, feedbackDistance = 1, and both gains at 0.5.
// block->data[i] = buffer[cursor] = int( (0.5 * buffer[cursorMinus(1)]) + (0.5 * buffer[cursorMinus(buflen + 0)]));
// much obliged: https://www.dsprelated.com/freebooks/filters/Analysis_Digital_Comb_Filter.html
// three element version:
//block->data[i] = buffer[cursor] = (int16_t)((s0 + s1 + s2) / 3);
//block->data[i] = buffer[cursor] = ((s0 + s1 + s2) / 3);
// four element version:
//s3 = buffer[cursorMinus(23)];
//block->data[i] = buffer[cursor] = ((s0 + s1 + s2 + s3) / 4);
}
// increment cursor
if (++cursor == _bufmax) {
cursor = 0;
}
if (cursor == triggerPoint) {
triggered = false;
}
}
if (excitement != NULL) {
release(excitement);
}
transmit(block);
release(block);
}
void BandPassFilter::update()
{/*
if(enabled)
{
AudioFilterStateVariable::update();
}*/
audio_block_t *input_block=NULL, *control_block=NULL;
audio_block_t *lowpass_block=NULL, *bandpass_block=NULL, *highpass_block=NULL;
input_block = receiveReadOnly(0);
control_block = receiveReadOnly(1);
if (!input_block) {
if (control_block) release(control_block);
return;
}
if(enabled)
{
lowpass_block = allocate();
if (!lowpass_block) {
release(input_block);
if (control_block) release(control_block);
return;
}
bandpass_block = allocate();
if (!bandpass_block) {
release(input_block);
release(lowpass_block);
if (control_block) release(control_block);
return;
}
highpass_block = allocate();
if (!highpass_block) {
release(input_block);
release(lowpass_block);
release(bandpass_block);
if (control_block) release(control_block);
return;
}
if (control_block) {
AudioFilterStateVariable::update_variable(input_block->data,
control_block->data,
lowpass_block->data,
bandpass_block->data,
highpass_block->data);
release(control_block);
} else {
AudioFilterStateVariable::update_fixed(input_block->data,
lowpass_block->data,
bandpass_block->data,
highpass_block->data);
}
release(input_block);
transmit(lowpass_block, 0);
release(lowpass_block);
transmit(bandpass_block, 1);
release(bandpass_block);
transmit(highpass_block, 2);
release(highpass_block);
return;
}
else
{
transmit(input_block, 0);
transmit(input_block, 1);
transmit(input_block, 2);
release(input_block);
}
}
void AudioAnalyzeFFT1024::update(void)
{
audio_block_t *block;
block = receiveReadOnly();
if (!block) return;
switch (state) {
case 0:
blocklist[0] = block;
state = 1;
break;
case 1:
blocklist[1] = block;
state = 2;
break;
case 2:
blocklist[2] = block;
state = 3;
break;
case 3:
blocklist[3] = block;
state = 4;
break;
case 4:
blocklist[4] = block;
state = 5;
break;
case 5:
blocklist[5] = block;
state = 6;
break;
case 6:
blocklist[6] = block;
state = 7;
break;
case 7:
blocklist[7] = block;
// TODO: perhaps distribute the work over multiple update() ??
// github pull requsts welcome......
copy_to_fft_buffer(buffer+0x000, blocklist[0]->data);
copy_to_fft_buffer(buffer+0x100, blocklist[1]->data);
copy_to_fft_buffer(buffer+0x200, blocklist[2]->data);
copy_to_fft_buffer(buffer+0x300, blocklist[3]->data);
copy_to_fft_buffer(buffer+0x400, blocklist[4]->data);
copy_to_fft_buffer(buffer+0x500, blocklist[5]->data);
copy_to_fft_buffer(buffer+0x600, blocklist[6]->data);
copy_to_fft_buffer(buffer+0x700, blocklist[7]->data);
if (window) apply_window_to_fft_buffer(buffer, window);
arm_cfft_radix4_q15(&fft_inst, buffer);
// TODO: support averaging multiple copies
for (int i=0; i < 512; i++) {
uint32_t tmp = *((uint32_t *)buffer + i); // real & imag
uint32_t magsq = multiply_16tx16t_add_16bx16b(tmp, tmp);
output[i] = sqrt_uint32_approx(magsq);
}
outputflag = true;
release(blocklist[0]);
release(blocklist[1]);
release(blocklist[2]);
release(blocklist[3]);
blocklist[0] = blocklist[4];
blocklist[1] = blocklist[5];
blocklist[2] = blocklist[6];
blocklist[3] = blocklist[7];
state = 4;
break;
}
}
