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synthio: reorganize the order of operations
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Apply envelope & panning after biquad filtering.

This may fix the weird popping problem. It also reduces the number
of operations that are done "in stereo", so it could help performance.

It also fixes a previously unnoticed problem where a ring-modulated
waveform had 2x the amplitude of an un-modulated waveform.

The test differences look large but it's because some values got changed
in the LSB after the mathematical divisions were moved around.
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@jepler
jepler committed May 31, 2023
1 parent 30b69a8 commit a999e40
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Showing 10 changed files with 5,127 additions and 5,142 deletions.
45 changes: 20 additions & 25 deletions shared-module/synthio/Biquad.c
View file
Expand Up @@ -111,38 +111,33 @@ void synthio_biquad_filter_assign(biquad_filter_state *st, mp_obj_t biquad_obj)
}

void synthio_biquad_filter_reset(biquad_filter_state *st) {
memset(&st->x, 0, 8 * sizeof(int16_t));
memset(&st->x, 0, 4 * sizeof(int16_t));
}

void synthio_biquad_filter_samples(biquad_filter_state *st, int32_t *out0, const int32_t *in0, size_t n0, size_t n_channels) {
void synthio_biquad_filter_samples(biquad_filter_state *st, int32_t *buffer, size_t n_samples) {
int32_t a1 = st->a1;
int32_t a2 = st->a2;
int32_t b0 = st->b0;
int32_t b1 = st->b1;
int32_t b2 = st->b2;

for (size_t i = 0; i < n_channels; i++) {
const int32_t *in = in0 + i;
int32_t *out = out0 + i;

int32_t x0 = st->x[i][0];
int32_t x1 = st->x[i][1];
int32_t y0 = st->y[i][0];
int32_t y1 = st->y[i][1];

for (size_t n = n0; n; --n, in += n_channels, out += n_channels) {
int32_t input = *in;
int32_t output = (b0 * input + b1 * x0 + b2 * x1 - a1 * y0 - a2 * y1 + (1 << (BIQUAD_SHIFT - 1))) >> BIQUAD_SHIFT;

x1 = x0;
x0 = input;
y1 = y0;
y0 = output;
*out += output;
}
st->x[i][0] = x0;
st->x[i][1] = x1;
st->y[i][0] = y0;
st->y[i][1] = y1;
int32_t x0 = st->x[0];
int32_t x1 = st->x[1];
int32_t y0 = st->y[0];
int32_t y1 = st->y[1];

for (size_t n = n_samples; n; --n, ++buffer) {
int32_t input = *buffer;
int32_t output = (b0 * input + b1 * x0 + b2 * x1 - a1 * y0 - a2 * y1 + (1 << (BIQUAD_SHIFT - 1))) >> BIQUAD_SHIFT;

x1 = x0;
x0 = input;
y1 = y0;
y0 = output;
*buffer = output;
}
st->x[0] = x0;
st->x[1] = x1;
st->y[0] = y0;
st->y[1] = y1;
}
4 changes: 2 additions & 2 deletions shared-module/synthio/Biquad.h
View file
Expand Up @@ -30,9 +30,9 @@

typedef struct {
int32_t a1, a2, b0, b1, b2;
int32_t x[2][2], y[2][2];
int32_t x[2], y[2];
} biquad_filter_state;

void synthio_biquad_filter_assign(biquad_filter_state *st, mp_obj_t biquad_obj);
void synthio_biquad_filter_reset(biquad_filter_state *st);
void synthio_biquad_filter_samples(biquad_filter_state *st, int32_t *out, const int32_t *in, size_t n_samples, size_t n_channels);
void synthio_biquad_filter_samples(biquad_filter_state *st, int32_t *buffer, size_t n_samples);
150 changes: 70 additions & 80 deletions shared-module/synthio/__init__.c
View file
Expand Up @@ -172,24 +172,11 @@ int16_t mix_down_sample(int32_t sample) {
return sample;
}

static void synth_note_into_buffer(synthio_synth_t *synth, int chan, int32_t *out_buffer32, int16_t dur) {
static bool synth_note_into_buffer(synthio_synth_t *synth, int chan, int32_t *out_buffer32, int16_t dur, uint16_t loudness[2]) {
mp_obj_t note_obj = synth->span.note_obj[chan];

if (note_obj == SYNTHIO_SILENCE) {
synth->accum[chan] = 0;
return;
}

if (synth->envelope_state[chan].level == 0) {
// note is truly finished, but we only just noticed
synth->span.note_obj[chan] = SYNTHIO_SILENCE;
return;
}

int32_t sample_rate = synth->sample_rate;

// adjust loudness by envelope
uint16_t loudness[2] = {synth->envelope_state[chan].level,synth->envelope_state[chan].level};

uint32_t dds_rate;
const int16_t *waveform = synth->waveform_bufinfo.buf;
Expand Down Expand Up @@ -228,33 +215,37 @@ static void synth_note_into_buffer(synthio_synth_t *synth, int chan, int32_t *ou
}
}

int synth_chan = synth->channel_count;
if (ring_dds_rate) {
uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
uint32_t accum = synth->accum[chan];
uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
uint32_t accum = synth->accum[chan];

if (dds_rate > lim / 2) {
// beyond nyquist, can't play note
return;
}
if (dds_rate > lim / 2) {
// beyond nyquist, can't play note
return false;
}

// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
if (accum > lim) {
accum %= lim;
}

// first, fill with waveform
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;
accum -= lim;
}
int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
out_buffer32[i] = waveform[idx];
}
synth->accum[chan] = accum;

int32_t ring_buffer[dur];
// first, fill with waveform
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;
ring_buffer[i] = waveform[idx];
if (ring_dds_rate) {
if (ring_dds_rate > lim / 2) {
// beyond nyquist, can't play ring (but did synth main sound so
// return true)
return true;
}
synth->accum[chan] = accum;

// now modulate by ring and accumulate
accum = synth->ring_accum[chan];
Expand All @@ -265,49 +256,19 @@ static void synth_note_into_buffer(synthio_synth_t *synth, int chan, int32_t *ou
accum %= lim;
}

for (uint16_t i = 0, j = 0; i < dur; i++) {
for (uint16_t i = 0; i < dur; i++) {
accum += ring_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;
int16_t wi = (ring_waveform[idx] * ring_buffer[i]) / 32768;
for (int c = 0; c < synth_chan; c++) {
out_buffer32[j] += (wi * loudness[c]) / 32768;
j++;
}
int16_t wi = (ring_waveform[idx] * out_buffer32[i]) / 32768;
out_buffer32[i] = wi;
}
synth->ring_accum[chan] = accum;
} else {
uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
uint32_t accum = synth->accum[chan];

if (dds_rate > lim / 2) {
// beyond nyquist, can't play note
return;
}

// 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, j = 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;
int16_t wi = waveform[idx];
for (int c = 0; c < synth_chan; c++) {
out_buffer32[j] += (wi * loudness[c]) / 65536;
j++;
}
}
synth->accum[chan] = accum;
}
return true;
}

STATIC mp_obj_t synthio_synth_get_note_filter(mp_obj_t note_obj) {
Expand All @@ -321,6 +282,19 @@ STATIC mp_obj_t synthio_synth_get_note_filter(mp_obj_t note_obj) {
return mp_const_none;
}

STATIC void sum_with_loudness(int32_t *out_buffer32, int32_t *tmp_buffer32, uint16_t loudness[2], size_t dur, int synth_chan) {
if (synth_chan == 1) {
for (size_t i = 0; i < dur; i++) {
*out_buffer32++ += (*tmp_buffer32++ *loudness[0]) >> 16;
}
} else {
for (size_t i = 0; i < dur; i++) {
*out_buffer32++ += (*tmp_buffer32 * loudness[0]) >> 16;
*out_buffer32++ += (*tmp_buffer32++ *loudness[1]) >> 16;
}
}
}

void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t *buffer_length, uint8_t channel) {

if (channel == synth->other_channel) {
Expand All @@ -338,28 +312,44 @@ void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t
uint16_t dur = MIN(SYNTHIO_MAX_DUR, synth->span.dur);
synth->span.dur -= dur;

int32_t out_buffer32[dur * synth->channel_count];
memset(out_buffer32, 0, sizeof(out_buffer32));
int32_t out_buffer32[SYNTHIO_MAX_DUR * synth->channel_count];
int32_t tmp_buffer32[SYNTHIO_MAX_DUR];
memset(out_buffer32, 0, synth->channel_count * dur * sizeof(int32_t));

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) {
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;
}

uint16_t loudness[2] = {synth->envelope_state[chan].level,synth->envelope_state[chan].level};

if (!synth_note_into_buffer(synth, chan, tmp_buffer32, dur, loudness)) {
// for some other reason, such as being above nyquist, note
// couldn't be synthed, so don't filter or sum it in
continue;
}

mp_obj_t filter_obj = synthio_synth_get_note_filter(note_obj);
if (filter_obj == mp_const_none) {
synth_note_into_buffer(synth, chan, out_buffer32, dur);
} else {
if (filter_obj != mp_const_none) {
synthio_note_obj_t *note = MP_OBJ_TO_PTR(note_obj);
int32_t filter_buffer32[dur * synth->channel_count];
memset(filter_buffer32, 0, sizeof(filter_buffer32));

synth_note_into_buffer(synth, chan, filter_buffer32, dur);
synthio_biquad_filter_samples(&note->filter_state, out_buffer32, filter_buffer32, dur, synth->channel_count);
synthio_biquad_filter_samples(&note->filter_state, tmp_buffer32, dur);
}

// adjust loudness by envelope
sum_with_loudness(out_buffer32, tmp_buffer32, loudness, dur, synth->channel_count);
}

int16_t *out_buffer16 = (int16_t *)(void *)synth->buffers[synth->buffer_index];

// mix down audio
for (size_t i = 0; i < MP_ARRAY_SIZE(out_buffer32); i++) {
for (size_t i = 0; i < dur * synth->channel_count; i++) {
int32_t sample = out_buffer32[i];
out_buffer16[i] = mix_down_sample(sample);
}
Expand Down
4 changes: 2 additions & 2 deletions tests/circuitpython/miditrack.py.exp
View file
@@ -1,4 +1,4 @@
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