forked from forks/qmk_firmware
start morphing wavetable
This commit is contained in:
parent
474d100b56
commit
bfe468ef1d
|
@ -38,6 +38,33 @@
|
|||
#define DAC_SAMPLE_MAX 4095U
|
||||
#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
|
||||
* lower will sacrifice perceptible audio quality. Any higher will limit the
|
||||
|
@ -66,16 +93,8 @@
|
|||
#endif
|
||||
|
||||
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};
|
||||
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_note = false;
|
||||
|
@ -87,10 +106,8 @@ uint32_t note_position = 0;
|
|||
float (* notes_pointer)[][2];
|
||||
uint16_t notes_count;
|
||||
bool notes_repeat;
|
||||
bool note_resting = false;
|
||||
|
||||
uint16_t current_note = 0;
|
||||
uint8_t rest_counter = 0;
|
||||
|
||||
#ifdef VIBRATO_ENABLE
|
||||
float vibrato_counter = 0;
|
||||
|
@ -192,23 +209,25 @@ static const dacsample_t dac_buffer_square[DAC_BUFFER_SIZE] = {
|
|||
|
||||
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};
|
||||
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) {
|
||||
|
||||
(void)dacp;
|
||||
(void)dac_buffer;
|
||||
// (void)dac_buffer_triangle;
|
||||
(void)dac_buffer_square;
|
||||
|
||||
__attribute__ ((weak))
|
||||
uint16_t generate_sample(void) {
|
||||
uint16_t sample = DAC_OFF_VALUE;
|
||||
uint8_t working_voices = voices;
|
||||
if (working_voices > DAC_VOICES_MAX)
|
||||
working_voices = DAC_VOICES_MAX;
|
||||
|
||||
for (uint8_t s = 0; s < sample_count; s++) {
|
||||
if (working_voices > 0) {
|
||||
uint16_t sample_sum = 0;
|
||||
for (uint8_t i = 0; i < working_voices; i++) {
|
||||
|
@ -219,23 +238,51 @@ static void dac_end(DACDriver * dacp, dacsample_t * sample_p, size_t sample_coun
|
|||
while (dac_if[i] >= (DAC_BUFFER_SIZE))
|
||||
dac_if[i] = dac_if[i] - DAC_BUFFER_SIZE;
|
||||
|
||||
(void)dac_buffer;
|
||||
(void)dac_buffer_square;
|
||||
(void)dac_buffer_triangle;
|
||||
|
||||
#define DAC_MORPH_SPEED 3000
|
||||
#define DAC_SAMPLE_CUSTOM_LENGTH 64
|
||||
#define DAC_MORPH_SPEED_COMPUTED (DAC_SAMPLE_RATE / DAC_SAMPLE_CUSTOM_LENGTH * DAC_MORPH_SPEED / 1000)
|
||||
|
||||
uint16_t dac_i = (uint16_t)dac_if[i];
|
||||
// Wavetable generation/lookup
|
||||
// SINE
|
||||
sample_sum += dac_buffer[dac_i] / working_voices / 3;
|
||||
// sample_sum += dac_buffer[dac_i] / working_voices / 3;
|
||||
// TRIANGLE
|
||||
sample_sum += dac_buffer_triangle[dac_i] / working_voices / 3;
|
||||
// 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_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;
|
||||
}
|
||||
|
||||
}
|
||||
sample_p[s] = sample_sum;
|
||||
} else {
|
||||
sample_p[s] = DAC_OFF_VALUE;
|
||||
}
|
||||
/**
|
||||
* 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) {
|
||||
|
@ -359,8 +406,6 @@ void stop_all_notes() {
|
|||
|
||||
playing_notes = false;
|
||||
playing_note = false;
|
||||
frequency = 0;
|
||||
frequency_alt = 0;
|
||||
|
||||
for (uint8_t i = 0; i < 8; i++)
|
||||
{
|
||||
|
@ -393,12 +438,7 @@ void stop_note(float freq) {
|
|||
if (voices < 0) {
|
||||
voices = 0;
|
||||
}
|
||||
if (voice_place >= voices) {
|
||||
voice_place = 0;
|
||||
}
|
||||
if (voices == 0) {
|
||||
frequency = 0;
|
||||
frequency_alt = 0;
|
||||
playing_note = false;
|
||||
}
|
||||
}
|
||||
|
|
2178
quantum/audio/wavetable.h
Normal file
2178
quantum/audio/wavetable.h
Normal file
File diff suppressed because it is too large
Load diff
23
util/wav_parser.py
Normal file
23
util/wav_parser.py
Normal file
|
@ -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)
|
Loading…
Reference in a new issue