synthio: implement a range compressor with hard knee

This really improves the loudness of the output with multiple notes
while being a nice simple algorithm to implement.
This commit is contained in:
Jeff Epler 2023-05-08 09:20:46 -05:00
parent 9a9f3229fa
commit c031bda5dd
No known key found for this signature in database
GPG Key ID: D5BF15AB975AB4DE
5 changed files with 160 additions and 137 deletions

View File

@ -145,20 +145,30 @@ STATIC synthio_envelope_definition_t *synthio_synth_get_note_envelope(synthio_sy
}
STATIC uint32_t synthio_synth_sum_envelope(synthio_synth_t *synth) {
uint32_t result = 0;
for (int chan = 0; chan < CIRCUITPY_SYNTHIO_MAX_CHANNELS; chan++) {
mp_obj_t note_obj = synth->span.note_obj[chan];
if (note_obj != SYNTHIO_SILENCE) {
synthio_envelope_state_t *state = &synth->envelope_state[chan];
if (state->state == SYNTHIO_ENVELOPE_STATE_ATTACK) {
result += state->level;
} else {
result += synthio_synth_get_note_envelope(synth, note_obj)->attack_level;
}
}
#define RANGE_LOW (-28000)
#define RANGE_HIGH (28000)
#define RANGE_SHIFT (16)
#define RANGE_SCALE (0xfffffff / (32768 * CIRCUITPY_SYNTHIO_MAX_CHANNELS - RANGE_HIGH))
// dynamic range compression via a downward compressor with hard knee
//
// When the output value is within the range +-28000 (about 85% of full scale),
// it is unchanged. Otherwise, it undergoes a gain reduction so that the
// largest possible values, (+32768,-32767) * CIRCUITPY_SYNTHIO_MAX_CHANNELS,
// still fit within the output range
//
// This produces a much louder overall volume with multiple voices, without
// much additional processing.
//
// https://en.wikipedia.org/wiki/Dynamic_range_compression
STATIC
int16_t mix_down_sample(int32_t sample) {
if (sample < RANGE_LOW) {
sample = (((sample - RANGE_LOW) * RANGE_SCALE) >> RANGE_SHIFT) + RANGE_LOW;
} else if (sample > RANGE_HIGH) {
sample = (((sample - RANGE_HIGH) * RANGE_SCALE) >> RANGE_SHIFT) + RANGE_HIGH;
}
return result;
return sample;
}
void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t *buffer_length, uint8_t channel) {
@ -172,89 +182,83 @@ void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t
synth->buffer_index = !synth->buffer_index;
synth->other_channel = 1 - channel;
synth->other_buffer_index = synth->buffer_index;
int16_t *out_buffer = (int16_t *)(void *)synth->buffers[synth->buffer_index];
uint16_t dur = MIN(SYNTHIO_MAX_DUR, synth->span.dur);
synth->span.dur -= dur;
memset(out_buffer, 0, synth->buffer_length);
int32_t sample_rate = synth->sample_rate;
uint32_t total_envelope = synthio_synth_sum_envelope(synth);
if (total_envelope < synth->total_envelope) {
// total envelope is decreasing. Slowly let remaining notes get louder
// the time constant is arbitrary, on the order of 1s at 48kHz
total_envelope = synth->total_envelope = (
total_envelope + synth->total_envelope * 255) / 256;
} else {
// total envelope is steady or increasing, so just store this as
// the high water mark
synth->total_envelope = total_envelope;
}
if (total_envelope > 0) {
uint16_t ovl_loudness = 0x7fffffff / MAX(0x8000, total_envelope);
int32_t out_buffer32[dur];
for (int chan = 0; chan < CIRCUITPY_SYNTHIO_MAX_CHANNELS; chan++) {
mp_obj_t note_obj = synth->span.note_obj[chan];
if (note_obj == SYNTHIO_SILENCE) {
synth->accum[chan] = 0;
continue;
}
if (synth->envelope_state[chan].level == 0) {
// note is truly finished, but we only just noticed
synth->span.note_obj[chan] = SYNTHIO_SILENCE;
continue;
}
// adjust loudness by envelope
uint16_t loudness = (ovl_loudness * synth->envelope_state[chan].level) >> 16;
uint32_t dds_rate;
const int16_t *waveform = synth->waveform;
uint32_t waveform_length = synth->waveform_length;
if (mp_obj_is_small_int(note_obj)) {
uint8_t note = mp_obj_get_int(note_obj);
uint8_t octave = note / 12;
uint16_t base_freq = notes[note % 12];
// rate = base_freq * waveform_length
// den = sample_rate * 2 ^ (10 - octave)
// den = sample_rate * 2 ^ 10 / 2^octave
// dds_rate = 2^SHIFT * rate / den
// dds_rate = 2^(SHIFT-10+octave) * base_freq * waveform_length / sample_rate
dds_rate = (sample_rate / 2 + ((uint64_t)(base_freq * waveform_length) << (SYNTHIO_FREQUENCY_SHIFT - 10 + octave))) / sample_rate;
} else {
synthio_note_obj_t *note = MP_OBJ_TO_PTR(note_obj);
int32_t frequency_scaled = synthio_note_step(note, sample_rate, dur, &loudness);
if (note->waveform_buf.buf) {
waveform = note->waveform_buf.buf;
waveform_length = note->waveform_buf.len / 2;
}
dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)frequency_scaled * waveform_length, sample_rate);
}
uint32_t accum = synth->accum[chan];
uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
if (dds_rate > lim / 2) {
// beyond nyquist, can't play note
continue;
}
// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
if (accum > lim) {
accum %= lim;
}
for (uint16_t i = 0; i < dur; i++) {
accum += dds_rate;
// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
if (accum > lim) {
accum -= lim;
}
int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
out_buffer[i] += (waveform[idx] * loudness) / 65536;
}
synth->accum[chan] = accum;
memset(out_buffer32, 0, sizeof(out_buffer32));
for (int chan = 0; chan < CIRCUITPY_SYNTHIO_MAX_CHANNELS; chan++) {
mp_obj_t note_obj = synth->span.note_obj[chan];
if (note_obj == SYNTHIO_SILENCE) {
synth->accum[chan] = 0;
continue;
}
if (synth->envelope_state[chan].level == 0) {
// note is truly finished, but we only just noticed
synth->span.note_obj[chan] = SYNTHIO_SILENCE;
continue;
}
// adjust loudness by envelope
uint16_t loudness = synth->envelope_state[chan].level;
uint32_t dds_rate;
const int16_t *waveform = synth->waveform;
uint32_t waveform_length = synth->waveform_length;
if (mp_obj_is_small_int(note_obj)) {
uint8_t note = mp_obj_get_int(note_obj);
uint8_t octave = note / 12;
uint16_t base_freq = notes[note % 12];
// rate = base_freq * waveform_length
// den = sample_rate * 2 ^ (10 - octave)
// den = sample_rate * 2 ^ 10 / 2^octave
// dds_rate = 2^SHIFT * rate / den
// dds_rate = 2^(SHIFT-10+octave) * base_freq * waveform_length / sample_rate
dds_rate = (sample_rate / 2 + ((uint64_t)(base_freq * waveform_length) << (SYNTHIO_FREQUENCY_SHIFT - 10 + octave))) / sample_rate;
} else {
synthio_note_obj_t *note = MP_OBJ_TO_PTR(note_obj);
int32_t frequency_scaled = synthio_note_step(note, sample_rate, dur, &loudness);
if (note->waveform_buf.buf) {
waveform = note->waveform_buf.buf;
waveform_length = note->waveform_buf.len / 2;
}
dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)frequency_scaled * waveform_length, sample_rate);
}
uint32_t accum = synth->accum[chan];
uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
if (dds_rate > lim / 2) {
// beyond nyquist, can't play note
continue;
}
// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
if (accum > lim) {
accum %= lim;
}
for (uint16_t i = 0; i < dur; i++) {
accum += dds_rate;
// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
if (accum > lim) {
accum -= lim;
}
int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
out_buffer32[i] += (waveform[idx] * loudness) / 65536;
}
synth->accum[chan] = accum;
}
int16_t *out_buffer16 = (int16_t *)(void *)synth->buffers[synth->buffer_index];
// mix down audio
for (size_t i = 0; i < dur; i++) {
int32_t sample = out_buffer32[i];
out_buffer16[i] = mix_down_sample(sample);
}
// advance envelope states
@ -267,7 +271,7 @@ void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t
}
*buffer_length = synth->last_buffer_length = dur * SYNTHIO_BYTES_PER_SAMPLE;
*bufptr = (uint8_t *)out_buffer;
*bufptr = (uint8_t *)out_buffer16;
}
void synthio_synth_reset_buffer(synthio_synth_t *synth, bool single_channel_output, uint8_t channel) {

View File

@ -84,6 +84,25 @@ def synthesize4(synth):
yield 36
def synthesize5(synth):
notes = [
synthio.Note(
frequency=synthio.midi_to_hz(60 + i + o),
waveform=sine,
envelope=envelope,
)
for i in [0, 4, 7]
for o in [0, -12, 12]
]
for n in notes:
print(n)
synth.press((n,))
yield 120
synth.release_all()
yield 36
def chain(*args):
for a in args:
yield from a
@ -94,7 +113,7 @@ with wave.open("tune-noenv.wav", "w") as f:
f.setnchannels(1)
f.setsampwidth(2)
f.setframerate(48000)
for n in chain(synthesize2(synth), synthesize3(synth), synthesize4(synth)):
for n in chain(synthesize5(synth)):
for i in range(n):
result, data = audiocore.get_buffer(synth)
f.writeframes(data)

View File

@ -1,4 +1,4 @@
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View File

@ -1,30 +1,30 @@
()
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View File

@ -1,30 +1,30 @@
()
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(Note(frequency=830.6076004423605, amplitude=1.0, tremolo_rate=0.0, tremolo_depth=0.0, vibrato_rate=0.0, vibrato_depth=0.0, waveform=None, envelope=None),)
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(Note(frequency=830.6076004423605, amplitude=1.0, tremolo_rate=0.0, tremolo_depth=0.0, vibrato_rate=0.0, vibrato_depth=0.0, waveform=None, envelope=None), Note(frequency=830.6076004423605, amplitude=1.0, tremolo_rate=0.0, tremolo_depth=0.0, vibrato_rate=0.0, vibrato_depth=0.0, waveform=None, envelope=None))
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(Note(frequency=830.6076004423605, amplitude=1.0, tremolo_rate=0.0, tremolo_depth=0.0, vibrato_rate=0.0, vibrato_depth=0.0, waveform=None, envelope=None),)
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