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Fixed AudioSynthWaveform. Uses LUT with linear interpolation for sine…
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…wave.

Square, triangle and sawtooth are generated directly.
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@el-supremo
el-supremo committed Apr 16, 2014
1 parent 677c072 commit 0c07318
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Showing 3 changed files with 315 additions and 26 deletions.
217 changes: 194 additions & 23 deletions synth_waveform.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -29,9 +29,95 @@
#include "utility/dspinst.h"


#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
/******************************************************************/
// PAH - add ramp-up and ramp-down to the onset of the wave
// the length is specified in samples
void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
{
if(r_length < 0) {
ramp_length = 0;
return;
}
// Don't set the ramp length longer than about 4 milliseconds
if(r_length > 44*4) {
ramp_length = 44*4;
return;
}
ramp_length = r_length;
}

void AudioSynthWaveform::update(void)
{
audio_block_t *block;
uint32_t i, ph, inc, index, scale;
int32_t val1, val2, val3;

//Serial.println("AudioSynthWaveform::update");
if (((magnitude > 0) || ramp_down) && (block = allocate()) != NULL) {
ph = phase;
inc = phase_increment;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
val1 = wavetable[index];
val2 = wavetable[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0xFFFF - scale;
val3 = (val1 + val2) >> 16;

// The value of ramp_up is always initialized to RAMP_LENGTH and then is
// decremented each time through here until it reaches zero.
// The value of ramp_up is used to generate a Q15 fraction which varies
// from [0 - 1), and multiplies this by the current sample
if(ramp_up) {
// ramp up to the new magnitude
// ramp_mag is the Q15 representation of the fraction
// Since ramp_up can't be zero, this cannot generate +1
ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
ramp_up--;
block->data[i] = (val3 * ((ramp_mag * magnitude)>>15)) >> 15;

} else if(ramp_down) {
// ramp down to zero from the last magnitude
// The value of ramp_down is always initialized to RAMP_LENGTH and then is
// decremented each time through here until it reaches zero.
// The value of ramp_down is used to generate a Q15 fraction which varies
// from (1 - 0], and multiplies this by the current sample
// avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
// cannot represent +1
ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
ramp_down--;
block->data[i] = (val3 * ((ramp_mag * last_magnitude)>>15)) >> 15;
} else {
block->data[i] = (val3 * magnitude) >> 15;
}

//Serial.print(block->data[i]);
//Serial.print(", ");
//if ((i % 12) == 11) Serial.println();
ph += inc;
}
//Serial.println();
phase = ph;
transmit(block);
release(block);
} else {
// is this numerical overflow ok?
phase += phase_increment * AUDIO_BLOCK_SAMPLES;
}
}

#else

/******************************************************************/
// PAH 140415 - change sin to use Paul's interpolation which is much
// faster than arm's sin function
// PAH 140316 - fix calculation of sample (amplitude error)
// PAH 140314 - change t_hi from int to float
// PAH - add ramp-up and ramp-down to the onset of the wave
// the length is specified in samples

void AudioSynthWaveform::set_ramp_length(int16_t r_length)
{
if(r_length < 0) {
Expand All @@ -47,17 +133,15 @@ void AudioSynthWaveform::set_ramp_length(int16_t r_length)
}


boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
boolean AudioSynthWaveform::begin(float t_amp,float t_hi,short type)
{
tone_type = type;
// tone_amp = t_amp;
amplitude(t_amp);
tone_freq = t_hi;
if(t_hi < 1)return false;
tone_freq = t_hi > 0.0;
if(t_hi <= 0.0)return false;
if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false;
tone_phase = 0;
// tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
frequency(t_hi);
if(0) {
Serial.print("AudioSynthWaveform.begin(tone_amp = ");
Serial.print(t_amp);
Expand All @@ -72,19 +156,19 @@ boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
return(true);
}

// PAH - 140313 fixed the calculation of the tone so that its spectrum
// is much improved
// PAH - 140313 fixed a problem with ramping
void AudioSynthWaveform::update(void)
{
audio_block_t *block;
short *bp;
// temporary for ramp in sine
uint32_t ramp_mag;
int32_t val1, val2, val3;
uint32_t index, scale;

// temporaries for TRIANGLE
uint32_t mag;
short tmp_amp;


if(tone_freq == 0)return;
// L E F T C H A N N E L O N L Y
Expand All @@ -94,6 +178,14 @@ void AudioSynthWaveform::update(void)
switch(tone_type) {
case TONE_TYPE_SINE:
for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
// Calculate interpolated sin
index = tone_phase >> 23;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (tone_phase >> 7) & 0xFFFF;
val2 *= scale;
val1 *= 0xFFFF - scale;
val3 = (val1 + val2) >> 16;
// The value of ramp_up is always initialized to RAMP_LENGTH and then is
// decremented each time through here until it reaches zero.
// The value of ramp_up is used to generate a Q15 fraction which varies
Expand All @@ -106,8 +198,8 @@ void AudioSynthWaveform::update(void)
ramp_up--;
// adjust tone_phase to Q15 format and then adjust the result
// of the multiplication
// calculate the sample
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
// calculate the sample
tmp_amp = (short)((val3 * tone_amp) >> 15);
*bp++ = (tmp_amp * ramp_mag)>>15;
}
else if(ramp_down) {
Expand All @@ -120,15 +212,10 @@ void AudioSynthWaveform::update(void)
// cannot represent +1
ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
ramp_down--;
// adjust tone_phase to Q15 format and then adjust the result
// of the multiplication
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * last_tone_amp) >> 17);
tmp_amp = (short)((val3 * last_tone_amp) >> 15);
*bp++ = (tmp_amp * ramp_mag)>>15;
} else {
// adjust tone_phase to Q15 format and then adjust the result
// of the multiplication
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
*bp++ = tmp_amp;
*bp++ = (short)((val3 * tone_amp) >> 15);
}

// phase and incr are both unsigned 32-bit fractions
Expand All @@ -140,7 +227,7 @@ void AudioSynthWaveform::update(void)

case TONE_TYPE_SQUARE:
for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
if(tone_phase & 0x80000000)*bp++ = -tone_amp;
if(tone_phase & 0x40000000)*bp++ = -tone_amp;
else *bp++ = tone_amp;
// phase and incr are both unsigned 32-bit fractions
tone_phase += tone_incr;
Expand All @@ -149,10 +236,9 @@ void AudioSynthWaveform::update(void)

case TONE_TYPE_SAWTOOTH:
for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
*bp = ((short)(tone_phase>>16)*tone_amp) >> 15;
bp++;
*bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
// phase and incr are both unsigned 32-bit fractions
tone_phase += tone_incr;
tone_phase += tone_incr;
}
break;

Expand All @@ -173,7 +259,7 @@ void AudioSynthWaveform::update(void)
mag = ~mag + 1;
}
*bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
tone_phase += tone_incr;
tone_phase += 2*tone_incr;
}
break;
}
Expand All @@ -182,3 +268,88 @@ void AudioSynthWaveform::update(void)
release(block);
}
}


#endif








#if 0
void AudioSineWaveMod::frequency(float f)
{
if (f > AUDIO_SAMPLE_RATE_EXACT / 2 || f < 0.0) return;
phase_increment = (f / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
}

void AudioSineWaveMod::update(void)
{
audio_block_t *block, *modinput;
uint32_t i, ph, inc, index, scale;
int32_t val1, val2;

//Serial.println("AudioSineWave::update");
modinput = receiveReadOnly();
ph = phase;
inc = phase_increment;
block = allocate();
if (!block) {
// unable to allocate memory, so we'll send nothing
if (modinput) {
// but if we got modulation data, update the phase
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph += inc + modinput->data[i] * modulation_factor;
}
release(modinput);
} else {
ph += phase_increment * AUDIO_BLOCK_SAMPLES;
}
phase = ph;
return;
}
if (modinput) {
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
val1 = sine_table[index];
val2 = sine_table[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0xFFFF - scale;
block->data[i] = (val1 + val2) >> 16;
//Serial.print(block->data[i]);
//Serial.print(", ");
//if ((i % 12) == 11) Serial.println();
ph += inc + modinput->data[i] * modulation_factor;
}
release(modinput);
} else {
ph = phase;
inc = phase_increment;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
index = ph >> 24;
val1 = sine_table[index];
val2 = sine_table[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0xFFFF - scale;
block->data[i] = (val1 + val2) >> 16;
//Serial.print(block->data[i]);
//Serial.print(", ");
//if ((i % 12) == 11) Serial.println();
ph += inc;
}
}
phase = ph;
transmit(block);
release(block);
}
#endif





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