Sine waves are boring, MUSIC!

pull/113/head
Phil Howard 2021-03-31 14:52:32 +01:00
rodzic b5ba4ba913
commit 082d41a384
5 zmienionych plików z 464 dodań i 79 usunięć

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@ -6,6 +6,7 @@ if (TARGET pico_audio_i2s)
add_executable( add_executable(
audio audio
demo.cpp demo.cpp
synth.cpp
) )
# Pull in pico libraries that we need # Pull in pico libraries that we need

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@ -0,0 +1,64 @@
/**
* Copyright (c) 2020 Raspberry Pi (Trading) Ltd.
*
* SPDX-License-Identifier: BSD-3-Clause
*/
#pragma once
#include "pico/audio_i2s.h"
#define SAMPLES_PER_BUFFER 256
typedef int16_t (*buffer_callback)(void);
struct audio_buffer_pool *init_audio(uint32_t sample_rate, uint8_t pin_data, uint8_t pin_bclk, uint8_t pio_sm=0, uint8_t dma_ch=0) {
static audio_format_t audio_format = {
.sample_freq = sample_rate,
.format = AUDIO_BUFFER_FORMAT_PCM_S16,
.channel_count = 1,
};
static struct audio_buffer_format producer_format = {
.format = &audio_format,
.sample_stride = 2
};
struct audio_buffer_pool *producer_pool = audio_new_producer_pool(
&producer_format,
3,
SAMPLES_PER_BUFFER
);
const struct audio_format *output_format;
struct audio_i2s_config config = {
.data_pin = pin_data,
.clock_pin_base = pin_bclk,
.dma_channel = dma_ch,
.pio_sm = pio_sm,
};
output_format = audio_i2s_setup(&audio_format, &config);
if (!output_format) {
panic("PicoAudio: Unable to open audio device.\n");
}
bool status = audio_i2s_connect(producer_pool);
if (!status) {
panic("PicoAudio: Unable to connect to audio device.\n");
}
audio_i2s_set_enabled(true);
return producer_pool;
}
void update_buffer(struct audio_buffer_pool *ap, buffer_callback cb) {
struct audio_buffer *buffer = take_audio_buffer(ap, true);
int16_t *samples = (int16_t *) buffer->buffer->bytes;
for (uint i = 0; i < buffer->max_sample_count; i++) {
samples[i] = cb();
}
buffer->sample_count = buffer->max_sample_count;
give_audio_buffer(ap, buffer);
}

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@ -1,97 +1,125 @@
/**
* Copyright (c) 2020 Raspberry Pi (Trading) Ltd.
*
* SPDX-License-Identifier: BSD-3-Clause
*/
#include <stdio.h> #include <stdio.h>
#include <math.h> #include <math.h>
#if PICO_ON_DEVICE
#include "hardware/clocks.h"
#include "hardware/structs/clocks.h"
#endif
#include "pico/stdlib.h" #include "pico/stdlib.h"
#include "pico/audio_i2s.h"
#define PICO_AUDIO_PAC_I2S_DATA 9 #include "synth.hpp"
#define PICO_AUDIO_PAC_I2S_BITCLOCK 10 #include "audio.hpp"
#define SINE_WAVE_TABLE_LEN 2048
#define SAMPLES_PER_BUFFER 256
static int16_t sine_wave_table[SINE_WAVE_TABLE_LEN]; #define PICO_AUDIO_PACK_I2S_DATA 9
#define PICO_AUDIO_PACK_I2S_BCLK 10
struct audio_buffer_pool *init_audio() { #define SONG_LENGTH 384
#define HAT 20000
#define BASS 500
#define SNARE 6000
#define SUB 50
static audio_format_t audio_format = { using namespace synth;
.sample_freq = 24000,
.format = AUDIO_BUFFER_FORMAT_PCM_S16,
.channel_count = 1,
};
static struct audio_buffer_format producer_format = { synth::AudioChannel synth::channels[CHANNEL_COUNT];
.format = &audio_format,
.sample_stride = 2
};
struct audio_buffer_pool *producer_pool = audio_new_producer_pool(&producer_format, 3, // Gadgetoid's amazing masterpiece!
SAMPLES_PER_BUFFER); // todo correct size const int16_t notes[5][SONG_LENGTH] = {
bool __unused ok; { // melody notes
const struct audio_format *output_format; 147, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 247, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 175, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 330, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 349, 0, 0, 0, 0, 0, 0, 0, 349, 0, 330, 0, 294, 0, 220, 0, 262, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 247, 0, 0, 0, 0, 0, 0, 0, 247, 0, 220, 0, 196, 0, 147, 0, 175, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0,
struct audio_i2s_config config = { 147, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 247, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 175, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 330, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 349, 0, 0, 0, 0, 0, 0, 0, 349, 0, 330, 0, 294, 0, 220, 0, 262, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 247, 0, 0, 0, 0, 0, 0, 0, 247, 0, 220, 0, 196, 0, 147, 0, 175, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0,
.data_pin = PICO_AUDIO_PAC_I2S_DATA, 147, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 247, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 175, 0, 0, 0, 0, 0, 0, 0, 175, 0, 196, 0, 220, 0, 262, 0, 330, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 349, 0, 0, 0, 0, 0, 0, 0, 349, 0, 330, 0, 294, 0, 220, 0, 262, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, 0, 247, 0, 0, 0, 0, 0, 0, 0, 247, 0, 262, 0, 294, 0, 392, 0, 440, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
.clock_pin_base = PICO_AUDIO_PAC_I2S_BITCLOCK, },
.dma_channel = 0, { // rhythm notes
.pio_sm = 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 392, 0, 523, 0, 659, 0, 523, 0, 392, 0, 523, 0, 659, 0, 523, 0, 698, 0, 587, 0, 440, 0, 587, 0, 698, 0, 587, 0, 440, 0, 587, 0, 523, 0, 440, 0, 330, 0, 440, 0, 523, 0, 440, 0, 330, 0, 440, 0, 349, 0, 294, 0, 220, 0, 294, 0, 349, 0, 294, 0, 220, 0, 294, 0, 262, 0, 247, 0, 220, 0, 175, 0, 165, 0, 147, 0, 131, 0, 98, 0,
}; 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 392, 0, 523, 0, 659, 0, 523, 0, 392, 0, 523, 0, 659, 0, 523, 0, 698, 0, 587, 0, 440, 0, 587, 0, 698, 0, 587, 0, 440, 0, 587, 0, 523, 0, 440, 0, 330, 0, 440, 0, 523, 0, 440, 0, 330, 0, 440, 0, 349, 0, 294, 0, 220, 0, 294, 0, 349, 0, 294, 0, 220, 0, 294, 0, 262, 0, 247, 0, 220, 0, 175, 0, 165, 0, 147, 0, 131, 0, 98, 0,
294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 294, 0, 440, 0, 587, 0, 440, 0, 392, 0, 523, 0, 659, 0, 523, 0, 392, 0, 523, 0, 659, 0, 523, 0, 698, 0, 587, 0, 440, 0, 587, 0, 698, 0, 587, 0, 440, 0, 587, 0, 523, 0, 440, 0, 330, 0, 440, 0, 523, 0, 440, 0, 330, 0, 440, 0, 349, 0, 294, 0, 220, 0, 294, 0, 349, 0, 294, 0, 220, 0, 294, 0, 262, 0, 247, 0, 220, 0, 175, 0, 165, 0, 147, 0, 131, 0, 98, 0,
},
{ // drum beats
BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0,
BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0,
BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, BASS, -1, BASS, -1, 0, 0, 0, 0, 0, 0, SNARE, 0, -1, 0, 0, 0, 0, 0
},
{ // hi-hat
HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1,
HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1,
HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1, HAT, -1
},
{ // bass notes under bass drum
SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0,
SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0,
SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, SUB, -1, SUB, -1, 0, 0, 0, 0, 0, 0, 0, -1, 0, 0, 0, 0, 0, 0
},
};
output_format = audio_i2s_setup(&audio_format, &config); void update_playback(void) {
if (!output_format) { static uint16_t prev_beat = 1;
panic("PicoAudio: Unable to open audio device.\n"); static uint16_t beat = 0;
absolute_time_t at = get_absolute_time();
uint64_t tick_ms = to_us_since_boot(at) / 1000;
beat = (tick_ms / 100) % SONG_LENGTH; // 100ms per beat
if (beat == prev_beat) return;
prev_beat = beat;
for(uint8_t i = 0; i < 5; i++) {
if(notes[i][beat] > 0) {
channels[i].frequency = notes[i][beat];
channels[i].trigger_attack();
} else if (notes[i][beat] == -1) {
channels[i].trigger_release();
} }
}
ok = audio_i2s_connect(producer_pool);
assert(ok);
audio_i2s_set_enabled(true);
return producer_pool;
} }
int main() { int main() {
stdio_init_all(); stdio_init_all();
struct audio_buffer_pool *ap = init_audio(synth::sample_rate, PICO_AUDIO_PACK_I2S_DATA, PICO_AUDIO_PACK_I2S_BCLK);
for (int i = 0; i < SINE_WAVE_TABLE_LEN; i++) { // configure voices
sine_wave_table[i] = 32767 * cosf(i * 2 * (float) (M_PI / SINE_WAVE_TABLE_LEN));
}
struct audio_buffer_pool *ap = init_audio(); // melody track
uint32_t step = 0x200000; channels[0].waveforms = Waveform::TRIANGLE | Waveform::SQUARE;
uint32_t pos = 0; channels[0].attack_ms = 16;
uint32_t pos_max = 0x10000 * SINE_WAVE_TABLE_LEN; channels[0].decay_ms = 168;
uint vol = 128; channels[0].sustain = 0xafff;
while (true) { channels[0].release_ms = 168;
int c = getchar_timeout_us(0); channels[0].volume = 10000;
if (c >= 0) {
if (c == '-' && vol) vol -= 4; // rhythm track
if ((c == '=' || c == '+') && vol < 255) vol += 4; channels[1].waveforms = Waveform::SINE | Waveform::SQUARE;
if (c == '[' && step > 0x10000) step -= 0x10000; channels[1].attack_ms = 38;
if (c == ']' && step < (SINE_WAVE_TABLE_LEN / 16) * 0x20000) step += 0x10000; channels[1].decay_ms = 300;
if (c == 'q') break; channels[1].sustain = 0;
printf("vol = %d, step = %d \r", vol, step >> 16); channels[1].release_ms = 0;
} channels[1].volume = 12000;
struct audio_buffer *buffer = take_audio_buffer(ap, true);
int16_t *samples = (int16_t *) buffer->buffer->bytes; // drum track
for (uint i = 0; i < buffer->max_sample_count; i++) { channels[2].waveforms = Waveform::NOISE;
samples[i] = (vol * sine_wave_table[pos >> 16u]) >> 8u; channels[2].attack_ms = 5;
pos += step; channels[2].decay_ms = 10;
if (pos >= pos_max) pos -= pos_max; channels[2].sustain = 16000;
} channels[2].release_ms = 100;
buffer->sample_count = buffer->max_sample_count; channels[2].volume = 18000;
give_audio_buffer(ap, buffer);
} // hi-hat track
puts("\n"); channels[3].waveforms = Waveform::NOISE;
return 0; channels[3].attack_ms = 5;
channels[3].decay_ms = 5;
channels[3].sustain = 8000;
channels[3].release_ms = 40;
channels[3].volume = 8000;
// bass track
channels[4].waveforms = Waveform::SQUARE;
channels[4].attack_ms = 10;
channels[4].decay_ms = 100;
channels[4].sustain = 0;
channels[4].release_ms = 500;
channels[4].volume = 12000;
while (true) {
update_playback();
update_buffer(ap, get_audio_frame);
}
return 0;
} }

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#include "synth.hpp"
namespace synth {
uint32_t prng_xorshift_state = 0x32B71700;
uint32_t prng_xorshift_next() {
uint32_t x = prng_xorshift_state;
x ^= x << 13;
x ^= x >> 17;
x ^= x << 5;
prng_xorshift_state = x;
return x;
}
int32_t prng_normal() {
// rough approximation of a normal distribution
uint32_t r0 = prng_xorshift_next();
uint32_t r1 = prng_xorshift_next();
uint32_t n = ((r0 & 0xffff) + (r1 & 0xffff) + (r0 >> 16) + (r1 >> 16)) / 2;
return n - 0xffff;
}
uint16_t volume = 0xffff;
const int16_t sine_waveform[256] = {-32768,-32758,-32729,-32679,-32610,-32522,-32413,-32286,-32138,-31972,-31786,-31581,-31357,-31114,-30853,-30572,-30274,-29957,-29622,-29269,-28899,-28511,-28106,-27684,-27246,-26791,-26320,-25833,-25330,-24812,-24279,-23732,-23170,-22595,-22006,-21403,-20788,-20160,-19520,-18868,-18205,-17531,-16846,-16151,-15447,-14733,-14010,-13279,-12540,-11793,-11039,-10279,-9512,-8740,-7962,-7180,-6393,-5602,-4808,-4011,-3212,-2411,-1608,-804,0,804,1608,2411,3212,4011,4808,5602,6393,7180,7962,8740,9512,10279,11039,11793,12540,13279,14010,14733,15447,16151,16846,17531,18205,18868,19520,20160,20788,21403,22006,22595,23170,23732,24279,24812,25330,25833,26320,26791,27246,27684,28106,28511,28899,29269,29622,29957,30274,30572,30853,31114,31357,31581,31786,31972,32138,32286,32413,32522,32610,32679,32729,32758,32767,32758,32729,32679,32610,32522,32413,32286,32138,31972,31786,31581,31357,31114,30853,30572,30274,29957,29622,29269,28899,28511,28106,27684,27246,26791,26320,25833,25330,24812,24279,23732,23170,22595,22006,21403,20788,20160,19520,18868,18205,17531,16846,16151,15447,14733,14010,13279,12540,11793,11039,10279,9512,8740,7962,7180,6393,5602,4808,4011,3212,2411,1608,804,0,-804,-1608,-2411,-3212,-4011,-4808,-5602,-6393,-7180,-7962,-8740,-9512,-10279,-11039,-11793,-12540,-13279,-14010,-14733,-15447,-16151,-16846,-17531,-18205,-18868,-19520,-20160,-20788,-21403,-22006,-22595,-23170,-23732,-24279,-24812,-25330,-25833,-26320,-26791,-27246,-27684,-28106,-28511,-28899,-29269,-29622,-29957,-30274,-30572,-30853,-31114,-31357,-31581,-31786,-31972,-32138,-32286,-32413,-32522,-32610,-32679,-32729,-32758};
bool is_audio_playing() {
if(volume == 0) {
return false;
}
bool any_channel_playing = false;
for(int c = 0; c < CHANNEL_COUNT; c++) {
if(channels[c].volume > 0 && channels[c].adsr_phase != ADSRPhase::OFF) {
any_channel_playing = true;
}
}
return any_channel_playing;
}
int16_t get_audio_frame() {
int32_t sample = 0; // used to combine channel output
for(int c = 0; c < CHANNEL_COUNT; c++) {
auto &channel = channels[c];
// increment the waveform position counter. this provides an
// Q16 fixed point value representing how far through
// the current waveform we are
channel.waveform_offset += ((channel.frequency * 256) << 8) / sample_rate;
if(channel.adsr_phase == ADSRPhase::OFF) {
continue;
}
if ((channel.adsr_frame >= channel.adsr_end_frame) && (channel.adsr_phase != ADSRPhase::SUSTAIN)) {
switch (channel.adsr_phase) {
case ADSRPhase::ATTACK:
channel.trigger_decay();
break;
case ADSRPhase::DECAY:
channel.trigger_sustain();
break;
case ADSRPhase::RELEASE:
channel.off();
break;
default:
break;
}
}
channel.adsr += channel.adsr_step;
channel.adsr_frame++;
if(channel.waveform_offset & 0x10000) {
// if the waveform offset overflows then generate a new
// random noise sample
channel.noise = prng_normal();
}
channel.waveform_offset &= 0xffff;
// check if any waveforms are active for this channel
if(channel.waveforms) {
uint8_t waveform_count = 0;
int32_t channel_sample = 0;
if(channel.waveforms & Waveform::NOISE) {
channel_sample += channel.noise;
waveform_count++;
}
if(channel.waveforms & Waveform::SAW) {
channel_sample += (int32_t)channel.waveform_offset - 0x7fff;
waveform_count++;
}
// creates a triangle wave of ^
if (channel.waveforms & Waveform::TRIANGLE) {
if (channel.waveform_offset < 0x7fff) { // initial quarter up slope
channel_sample += int32_t(channel.waveform_offset * 2) - int32_t(0x7fff);
}
else { // final quarter up slope
channel_sample += int32_t(0x7fff) - ((int32_t(channel.waveform_offset) - int32_t(0x7fff)) * 2);
}
waveform_count++;
}
if (channel.waveforms & Waveform::SQUARE) {
channel_sample += (channel.waveform_offset < channel.pulse_width) ? 0x7fff : -0x7fff;
waveform_count++;
}
if(channel.waveforms & Waveform::SINE) {
// the sine_waveform sample contains 256 samples in
// total so we'll just use the most significant bits
// of the current waveform position to index into it
channel_sample += sine_waveform[channel.waveform_offset >> 8];
waveform_count++;
}
if(channel.waveforms & Waveform::WAVE) {
channel_sample += channel.wave_buffer[channel.wave_buf_pos];
if (++channel.wave_buf_pos == 64) {
channel.wave_buf_pos = 0;
if(channel.wave_buffer_callback)
channel.wave_buffer_callback(channel);
}
waveform_count++;
}
channel_sample = channel_sample / waveform_count;
channel_sample = (int64_t(channel_sample) * int32_t(channel.adsr >> 8)) >> 16;
// apply channel volume
channel_sample = (int64_t(channel_sample) * int32_t(channel.volume)) >> 16;
// apply channel filter
//if (channel.filter_enable) {
//float filter_epow = 1 - expf(-(1.0f / 22050.0f) * 2.0f * pi * int32_t(channel.filter_cutoff_frequency));
//channel_sample += (channel_sample - channel.filter_last_sample) * filter_epow;
//}
//channel.filter_last_sample = channel_sample;
// combine channel sample into the final sample
sample += channel_sample;
}
}
sample = (int64_t(sample) * int32_t(volume)) >> 16;
// clip result to 16-bit
sample = sample <= -0x8000 ? -0x8000 : (sample > 0x7fff ? 0x7fff : sample);
return sample;
}
}

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#pragma once
#include <cstdint>
namespace synth {
// The duration a note is played is determined by the amount of attack,
// decay, and release, combined with the length of the note as defined by
// the user.
//
// - Attack: number of milliseconds it takes for a note to hit full volume
// - Decay: number of milliseconds it takes for a note to settle to sustain volume
// - Sustain: percentage of full volume that the note sustains at (duration implied by other factors)
// - Release: number of milliseconds it takes for a note to reduce to zero volume after it has ended
//
// Attack (750ms) - Decay (500ms) -------- Sustain ----- Release (250ms)
//
// + + + +
// | | | |
// | | | |
// | | | |
// v v v v
// 0ms 1000ms 2000ms 3000ms 4000ms
//
// | XXXX | | | |
// | X X|XX | | |
// | X | XXX | | |
// | X | XXXXXXXXXXXXXX|XXXXXXXXXXXXXXXXXXX| |
// | X | | |X |
// | X | | |X |
// | X | | | X |
// | X | | | X |
// | X | | | X |
// | X | | | X |
// | X | | | X |
// | X | | | X |
// | X + + + | + + + | + + + | + + + | +
// | X | | | | | | | | | | | | | | | | |
// |X | | | | | | | | | | | | | | | | |
// +----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+----+--->
#define CHANNEL_COUNT 8
constexpr float pi = 3.14159265358979323846f;
const uint32_t sample_rate = 44100;
extern uint16_t volume;
enum Waveform {
NOISE = 128,
SQUARE = 64,
SAW = 32,
TRIANGLE = 16,
SINE = 8,
WAVE = 1
};
enum class ADSRPhase : uint8_t {
ATTACK,
DECAY,
SUSTAIN,
RELEASE,
OFF
};
struct AudioChannel {
uint8_t waveforms = 0; // bitmask for enabled waveforms (see AudioWaveform enum for values)
uint16_t frequency = 660; // frequency of the voice (Hz)
uint16_t volume = 0xffff; // channel volume (default 50%)
uint16_t attack_ms = 2; // attack period
uint16_t decay_ms = 6; // decay period
uint16_t sustain = 0xffff; // sustain volume
uint16_t release_ms = 1; // release period
uint16_t pulse_width = 0x7fff; // duty cycle of square wave (default 50%)
int16_t noise = 0; // current noise value
uint32_t waveform_offset = 0; // voice offset (Q8)
int32_t filter_last_sample = 0;
bool filter_enable = false;
uint16_t filter_cutoff_frequency = 0;
uint32_t adsr_frame = 0; // number of frames into the current ADSR phase
uint32_t adsr_end_frame = 0; // frame target at which the ADSR changes to the next phase
uint32_t adsr = 0;
int32_t adsr_step = 0;
ADSRPhase adsr_phase = ADSRPhase::OFF;
uint8_t wave_buf_pos = 0; //
int16_t wave_buffer[64]; // buffer for arbitrary waveforms. small as it's filled by user callback
void *user_data = nullptr;
void (*wave_buffer_callback)(AudioChannel &channel);
void trigger_attack() {
adsr_frame = 0;
adsr_phase = ADSRPhase::ATTACK;
adsr_end_frame = (attack_ms * sample_rate) / 1000;
adsr_step = (int32_t(0xffffff) - int32_t(adsr)) / int32_t(adsr_end_frame);
}
void trigger_decay() {
adsr_frame = 0;
adsr_phase = ADSRPhase::DECAY;
adsr_end_frame = (decay_ms * sample_rate) / 1000;
adsr_step = (int32_t(sustain << 8) - int32_t(adsr)) / int32_t(adsr_end_frame);
}
void trigger_sustain() {
adsr_frame = 0;
adsr_phase = ADSRPhase::SUSTAIN;
adsr_end_frame = 0;
adsr_step = 0;
}
void trigger_release() {
adsr_frame = 0;
adsr_phase = ADSRPhase::RELEASE;
adsr_end_frame = (release_ms * sample_rate) / 1000;
adsr_step = (int32_t(0) - int32_t(adsr)) / int32_t(adsr_end_frame);
}
void off() {
adsr_frame = 0;
adsr_phase = ADSRPhase::OFF;
adsr_step = 0;
}
};
extern AudioChannel channels[CHANNEL_COUNT];
int16_t get_audio_frame();
bool is_audio_playing();
}