#include "Audio.h" #include "arm_math.h" #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 - 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; } boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type) { tone_type = type; // tone_amp = t_amp; amplitude(t_amp); tone_freq = t_hi; if(t_hi < 1)return false; if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false; tone_phase = 0; tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT; if(0) { Serial.print("AudioSynthWaveform.begin(tone_amp = "); Serial.print(t_amp); Serial.print(", tone_hi = "); Serial.print(t_hi); Serial.print(", tone_incr = "); Serial.print(tone_incr,HEX); // Serial.print(", tone_hi = "); // Serial.print(t_hi); Serial.println(")"); } return(true); } void AudioSynthWaveform::update(void) { audio_block_t *block; short *bp; // temporary for ramp in sine uint32_t ramp_mag; // 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 block = allocate(); if(block) { bp = block->data; switch(tone_type) { case TONE_TYPE_SINE: for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) { // 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--; // adjust tone_phase to Q15 format and then adjust the result // of the multiplication *bp = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15); *bp++ = (*bp * ramp_mag)>>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--; // adjust tone_phase to Q15 format and then adjust the result // of the multiplication *bp = (short)((arm_sin_q15(tone_phase>>17) * last_tone_amp) >> 15); *bp++ = (*bp * ramp_mag)>>15; } else { // adjust tone_phase to Q15 format and then adjust the result // of the multiplication *bp++ = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15); } // phase and incr are both unsigned 32-bit fractions tone_phase += tone_incr; } break; case TONE_TYPE_SQUARE: for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) { if(tone_phase & 0x80000000)*bp++ = -tone_amp; else *bp++ = tone_amp; // phase and incr are both unsigned 32-bit fractions tone_phase += tone_incr; } break; case TONE_TYPE_SAWTOOTH: for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) { *bp = ((short)(tone_phase>>16)*tone_amp) >> 15; bp++; // phase and incr are both unsigned 32-bit fractions tone_phase += tone_incr; } break; case TONE_TYPE_TRIANGLE: for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) { if(tone_phase & 0x80000000) { // negative half-cycle tmp_amp = -tone_amp; } else { // positive half-cycle tmp_amp = tone_amp; } mag = tone_phase << 2; // Determine which quadrant if(tone_phase & 0x40000000) { // negate the magnitude mag = ~mag + 1; } *bp++ = ((short)(mag>>17)*tmp_amp) >> 15; tone_phase += tone_incr; } break; } // send the samples to the left channel transmit(block,0); 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