visus
/
teensy-audio
Segui 1
Vota 0
Forka 0
Codice Problemi 0 Pull Requests 0 Rilasci 0 Wiki Attività
Sfoglia il codice sorgente

Remove old version of AudioSynthWaveform. Fix sine wave generation to clean up its specturm and also fix ramping.

dds
Pete (El Supremo) 10 anni fa
parent
28f93925d2
commit
519822e76c
2 ha cambiato i file con 19 aggiunte e 251 eliminazioni
Visualizzazione separata Mostra Diff Stats
  1. +16
    -174
      synth_waveform.cpp
  2. +3
    -77
      synth_waveform.h

+ 16
- 174
synth_waveform.cpp Vedi File

@@ -29,90 +29,10 @@
#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)
void AudioSynthWaveform::set_ramp_length(int16_t r_length)
{
if(r_length < 0) {
ramp_length = 0;
@@ -135,7 +55,8 @@ boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
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;
// tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
if(0) {
Serial.print("AudioSynthWaveform.begin(tone_amp = ");
Serial.print(t_amp);
@@ -150,7 +71,9 @@ 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;
@@ -182,31 +105,35 @@ void AudioSynthWaveform::update(void)
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;
// calculate the sample
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
*bp++ = (tmp_amp * 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
// from [0 - 1), 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;
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * last_tone_amp) >> 17);
*bp++ = (tmp_amp * 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);
tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
*bp++ = tmp_amp;
}
// phase and incr are both unsigned 32-bit fractions
tone_phase += tone_incr;
// If tone_phase has overflowed, truncate the top bit
if(tone_phase & 0x80000000)tone_phase &= 0x7fffffff;
}
break;
@@ -254,88 +181,3 @@ 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






+ 3
- 77
synth_waveform.h Vedi File

@@ -30,58 +30,6 @@
#include "AudioStream.h"
#include "arm_math.h"

// waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
extern const int16_t AudioWaveformTriangle[257];
extern const int16_t AudioWaveformSquare[257];
extern const int16_t AudioWaveformSawtooth[257];
}

#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
class AudioSynthWaveform : public AudioStream
{
public:
AudioSynthWaveform(const int16_t *waveform)
: AudioStream(0, NULL), wavetable(waveform), magnitude(0), phase(0)
, ramp_down(0), ramp_up(0), ramp_mag(0), ramp_length(0)
{ }
void frequency(float freq) {
if (freq > AUDIO_SAMPLE_RATE_EXACT / 2 || freq < 0.0) return;
phase_increment = (freq / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
}
void amplitude(float n) { // 0 to 1.0
if (n < 0) n = 0;
else if (n > 1.0) n = 1.0;
// Ramp code
if(magnitude && (n == 0)) {
ramp_down = ramp_length;
ramp_up = 0;
last_magnitude = magnitude;
}
else if((magnitude == 0) && n) {
ramp_up = ramp_length;
ramp_down = 0;
}
// set new magnitude
magnitude = n * 32767.0;
}
virtual void update(void);
void set_ramp_length(uint16_t r_length);
private:
const int16_t *wavetable;
uint16_t magnitude;
uint16_t last_magnitude;
uint32_t phase;
uint32_t phase_increment;
uint32_t ramp_down;
uint32_t ramp_up;
uint32_t ramp_mag;
uint16_t ramp_length;
};

#else

#define AUDIO_SAMPLE_RATE_ROUNDED (44118)

@@ -106,7 +54,8 @@ public:
// change between two frequencies.
void frequency(int t_hi)
{
tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
// tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
}
// If ramp_length is non-zero this will set up
// either a rmap up or a ramp down when a wave
@@ -138,7 +87,7 @@ public:
}
boolean begin(float t_amp,int t_hi,short t_type);
virtual void update(void);
void set_ramp_length(uint16_t r_length);
void set_ramp_length(int16_t r_length);

private:
short tone_amp;
@@ -152,27 +101,4 @@ private:
uint32_t ramp_up;
uint16_t ramp_length;
};

#endif



#if 0
class AudioSineWaveMod : public AudioStream
{
public:
AudioSineWaveMod() : AudioStream(1, inputQueueArray) {}
void frequency(float freq);
//void amplitude(q15 n);
virtual void update(void);
private:
uint32_t phase;
uint32_t phase_increment;
uint32_t modulation_factor;
audio_block_t *inputQueueArray[1];
};
#endif



#endif

Write Preview
Loading…
Annulla
Salva