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start morphing wavetable

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This commit is contained in:
Jack Humbert 2019-06-04 02:32:07 -04:00
parent 474d100b56
commit bfe468ef1d
3 changed files with 2290 additions and 49 deletions
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138
quantum/audio/audio_arm.c
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@ -38,6 +38,33 @@
#define DAC_SAMPLE_MAX 4095U #define DAC_SAMPLE_MAX 4095U
#endif #endif
#define DAC_LOW_QUALITY
/**
* These presets allow you to quickly switch between quality/voice settings for
* the DAC. The sample rate and number of voices roughly has an inverse
* relationship - slightly higher sample rates may be possible.
*/
#ifdef DAC_VERY_LOW_QUALITY
#define DAC_SAMPLE_RATE 11025U
#define DAC_VOICES_MAX 8
#endif
#ifdef DAC_LOW_QUALITY
#define DAC_SAMPLE_RATE 22050U
#define DAC_VOICES_MAX 4
#endif
#ifdef DAC_HIGH_QUALITY
#define DAC_SAMPLE_RATE 44100U
#define DAC_VOICES_MAX 2
#endif
#ifdef DAC_VERY_HIGH_QUALITY
#define DAC_SAMPLE_RATE 88200U
#define DAC_VOICES_MAX 1
#endif
/** /**
* Effective bitrate of the DAC. 44.1khz is the standard for most audio - any * Effective bitrate of the DAC. 44.1khz is the standard for most audio - any
* lower will sacrifice perceptible audio quality. Any higher will limit the * lower will sacrifice perceptible audio quality. Any higher will limit the
@ -66,16 +93,8 @@
#endif #endif
int voices = 0; int voices = 0;
int voice_place = 0;
float frequency = 0;
float frequency_alt = 0;
float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0}; float frequencies[8] = {0, 0, 0, 0, 0, 0, 0, 0};
int volumes[8] = {0, 0, 0, 0, 0, 0, 0, 0}; int volumes[8] = {0, 0, 0, 0, 0, 0, 0, 0};
bool sliding = false;
uint8_t * sample;
uint16_t sample_length = 0;
bool playing_notes = false; bool playing_notes = false;
bool playing_note = false; bool playing_note = false;
@ -87,10 +106,8 @@ uint32_t note_position = 0;
float (* notes_pointer)[][2]; float (* notes_pointer)[][2];
uint16_t notes_count; uint16_t notes_count;
bool notes_repeat; bool notes_repeat;
bool note_resting = false;
uint16_t current_note = 0; uint16_t current_note = 0;
uint8_t rest_counter = 0;
#ifdef VIBRATO_ENABLE #ifdef VIBRATO_ENABLE
float vibrato_counter = 0; float vibrato_counter = 0;
@ -192,50 +209,80 @@ static const dacsample_t dac_buffer_square[DAC_BUFFER_SIZE] = {
static dacsample_t dac_buffer_empty[DAC_BUFFER_SIZE] = { DAC_OFF_VALUE }; static dacsample_t dac_buffer_empty[DAC_BUFFER_SIZE] = { DAC_OFF_VALUE };
#include "wavetable.h"
float dac_if[8] = {0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0}; float dac_if[8] = {0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0};
uint8_t dac_voice = 0;
uint8_t dac_voice_flipped = 0;
uint16_t dac_voice_counter = 0;
float dac_voice_count_flipped = 0;
/** /**
* DAC streaming callback. Does all of the main computing for sound synthesis. * Generation of the sample being passed to the callback. Declared weak so users
* can override it with their own waveforms/noises.
*/ */
static void dac_end(DACDriver * dacp, dacsample_t * sample_p, size_t sample_count) { __attribute__ ((weak))
uint16_t generate_sample(void) {
(void)dacp; uint16_t sample = DAC_OFF_VALUE;
(void)dac_buffer;
// (void)dac_buffer_triangle;
(void)dac_buffer_square;
uint8_t working_voices = voices; uint8_t working_voices = voices;
if (working_voices > DAC_VOICES_MAX) if (working_voices > DAC_VOICES_MAX)
working_voices = DAC_VOICES_MAX; working_voices = DAC_VOICES_MAX;
for (uint8_t s = 0; s < sample_count; s++) { if (working_voices > 0) {
if (working_voices > 0) { uint16_t sample_sum = 0;
uint16_t sample_sum = 0; for (uint8_t i = 0; i < working_voices; i++) {
for (uint8_t i = 0; i < working_voices; i++) { dac_if[i] = dac_if[i] + ((frequencies[i]*DAC_BUFFER_SIZE)/DAC_SAMPLE_RATE);
dac_if[i] = dac_if[i] + ((frequencies[i]*DAC_BUFFER_SIZE)/DAC_SAMPLE_RATE);
// Needed because % doesn't work with floats // Needed because % doesn't work with floats
// 0.5 less than the size because we use round() later // 0.5 less than the size because we use round() later
while (dac_if[i] >= (DAC_BUFFER_SIZE)) while (dac_if[i] >= (DAC_BUFFER_SIZE))
dac_if[i] = dac_if[i] - DAC_BUFFER_SIZE; dac_if[i] = dac_if[i] - DAC_BUFFER_SIZE;
uint16_t dac_i = (uint16_t)dac_if[i]; (void)dac_buffer;
// Wavetable generation/lookup (void)dac_buffer_square;
// SINE (void)dac_buffer_triangle;
sample_sum += dac_buffer[dac_i] / working_voices / 3;
// TRIANGLE
sample_sum += dac_buffer_triangle[dac_i] / working_voices / 3;
// RISING TRIANGLE
// sample_sum += (uint16_t)round((dac_if[i] * DAC_SAMPLE_MAX) / DAC_BUFFER_SIZE / working_voices );
// SQUARE
// sample_sum += ((dac_if[i] > (DAC_BUFFER_SIZE / 2)) ? DAC_SAMPLE_MAX / working_voices: 0);
sample_sum += dac_buffer_square[dac_i] / working_voices / 3;
} #define DAC_MORPH_SPEED 3000
sample_p[s] = sample_sum; #define DAC_SAMPLE_CUSTOM_LENGTH 64
} else { #define DAC_MORPH_SPEED_COMPUTED (DAC_SAMPLE_RATE / DAC_SAMPLE_CUSTOM_LENGTH * DAC_MORPH_SPEED / 1000)
sample_p[s] = DAC_OFF_VALUE;
uint16_t dac_i = (uint16_t)dac_if[i];
// Wavetable generation/lookup
// SINE
// sample_sum += dac_buffer[dac_i] / working_voices / 3;
// TRIANGLE
// sample_sum += dac_buffer_triangle[dac_i] / working_voices / 3;
// RISING TRIANGLE
// sample_sum += (uint16_t)round((dac_if[i] * DAC_SAMPLE_MAX) / DAC_BUFFER_SIZE / working_voices );
// SQUARE
// sample_sum += ((dac_if[i] > (DAC_BUFFER_SIZE / 2)) ? DAC_SAMPLE_MAX / working_voices: 0);
// sample_sum += dac_buffer_square[dac_i] / working_voices / 3;
// sample_sum += dac_buffer_custom[dac_voice_flipped][dac_i] / working_voices / 2 * ((dac_voice >= 63) ? 6400 - dac_voice_counter : dac_voice_counter) / 6400;
// sample_sum += dac_buffer_custom[dac_voice_flipped + 1][dac_i] / working_voices / 2 * ((dac_voice >= 63) ? dac_voice_counter : 6400 - dac_voice_counter) / 6400;
sample_sum += dac_buffer_custom[dac_voice][dac_i] / working_voices / 2 * (DAC_MORPH_SPEED_COMPUTED - dac_voice_counter) / DAC_MORPH_SPEED_COMPUTED;
sample_sum += dac_buffer_custom[dac_voice + 1][dac_i] / working_voices / 2 * dac_voice_counter / DAC_MORPH_SPEED_COMPUTED;
} }
sample = sample_sum;
dac_voice_counter++;
if (dac_voice_counter >= DAC_MORPH_SPEED_COMPUTED) {
dac_voice_counter = 0;
// dac_voice = (dac_voice + 1) % 125;
// dac_voice_flipped = ((dac_voice >= 63) ? (125 - dac_voice) : dac_voice);
dac_voice = (dac_voice + 1) % (DAC_SAMPLE_CUSTOM_LENGTH - 1);
}
}
return sample;
}
/**
* DAC streaming callback. Does all of the main computing for playing songs.
*/
static void dac_end(DACDriver * dacp, dacsample_t * sample_p, size_t sample_count) {
(void)dacp;
for (uint8_t s = 0; s < sample_count; s++) {
sample_p[s] = generate_sample();
} }
if (playing_notes) { if (playing_notes) {
@ -359,8 +406,6 @@ void stop_all_notes() {
playing_notes = false; playing_notes = false;
playing_note = false; playing_note = false;
frequency = 0;
frequency_alt = 0;
for (uint8_t i = 0; i < 8; i++) for (uint8_t i = 0; i < 8; i++)
{ {
@ -393,12 +438,7 @@ void stop_note(float freq) {
if (voices < 0) { if (voices < 0) {
voices = 0; voices = 0;
} }
if (voice_place >= voices) {
voice_place = 0;
}
if (voices == 0) { if (voices == 0) {
frequency = 0;
frequency_alt = 0;
playing_note = false; playing_note = false;
} }
} }

2178
quantum/audio/wavetable.h Normal file
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File diff suppressed because it is too large Load diff

23
util/wav_parser.py Normal file
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@ -0,0 +1,23 @@
#! /bin/python
import wave, struct, sys
waveFile = wave.open(sys.argv[1], 'r')
length = waveFile.getnframes()
out = "static const dacsample_t dac_buffer_custom[" + str(int(length / 256)) + "][256] = {"
for i in range(0,length):
if (i % 8 == 0):
out += "\n "
if (i % 256 == 0):
out = out[:-2]
out += "{\n "
waveData = waveFile.readframes(1)
data = struct.unpack("<h", waveData)
out += str(int((int(data[0]) + 0x8000) / 16)) + ", "
if (i % 256 == 255):
out = out[:-2]
out += "\n },"
out = out[:-1]
out += "\n};"
print(out)