synthio: Factor out `synth_note_into_buffer`, start adding filter buffer
This commit is contained in:
Jeff Epler
parent
9d8dcf7d33
commit
62e6de8ed5
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@ -171,6 +171,144 @@ int16_t mix_down_sample(int32_t sample) {
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return sample;
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}
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static void synth_note_into_buffer(synthio_synth_t *synth, int chan, int32_t *out_buffer32, int16_t dur) {
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mp_obj_t note_obj = synth->span.note_obj[chan];
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if (note_obj == SYNTHIO_SILENCE) {
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synth->accum[chan] = 0;
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return;
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}
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if (synth->envelope_state[chan].level == 0) {
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// note is truly finished, but we only just noticed
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synth->span.note_obj[chan] = SYNTHIO_SILENCE;
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return;
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}
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int32_t sample_rate = synth->sample_rate;
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// adjust loudness by envelope
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uint16_t loudness[2] = {synth->envelope_state[chan].level,synth->envelope_state[chan].level};
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uint32_t dds_rate;
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const int16_t *waveform = synth->waveform_bufinfo.buf;
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uint32_t waveform_length = synth->waveform_bufinfo.len / 2;
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uint32_t ring_dds_rate = 0;
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const int16_t *ring_waveform = NULL;
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uint32_t ring_waveform_length = 0;
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if (mp_obj_is_small_int(note_obj)) {
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uint8_t note = mp_obj_get_int(note_obj);
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uint8_t octave = note / 12;
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uint16_t base_freq = notes[note % 12];
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// rate = base_freq * waveform_length
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// den = sample_rate * 2 ^ (10 - octave)
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// den = sample_rate * 2 ^ 10 / 2^octave
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// dds_rate = 2^SHIFT * rate / den
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// dds_rate = 2^(SHIFT-10+octave) * base_freq * waveform_length / sample_rate
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dds_rate = (sample_rate / 2 + ((uint64_t)(base_freq * waveform_length) << (SYNTHIO_FREQUENCY_SHIFT - 10 + octave))) / sample_rate;
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} else {
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synthio_note_obj_t *note = MP_OBJ_TO_PTR(note_obj);
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int32_t frequency_scaled = synthio_note_step(note, sample_rate, dur, loudness);
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if (note->waveform_buf.buf) {
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waveform = note->waveform_buf.buf;
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waveform_length = note->waveform_buf.len / 2;
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}
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dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)frequency_scaled * waveform_length, sample_rate);
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if (note->ring_frequency_scaled != 0 && note->ring_waveform_buf.buf) {
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ring_waveform = note->ring_waveform_buf.buf;
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ring_waveform_length = note->ring_waveform_buf.len / 2;
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ring_dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)note->ring_frequency_scaled * ring_waveform_length, sample_rate);
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uint32_t lim = ring_waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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if (ring_dds_rate > lim / 2) {
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ring_dds_rate = 0; // can't ring at that frequency
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}
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}
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}
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int synth_chan = synth->channel_count;
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if (ring_dds_rate) {
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uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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uint32_t accum = synth->accum[chan];
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if (dds_rate > lim / 2) {
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// beyond nyquist, can't play note
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return;
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}
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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int32_t ring_buffer[dur];
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// first, fill with waveform
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for (uint16_t i = 0; i < dur; i++) {
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accum += dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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ring_buffer[i] = waveform[idx];
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}
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synth->accum[chan] = accum;
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// now modulate by ring and accumulate
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accum = synth->ring_accum[chan];
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lim = ring_waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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for (uint16_t i = 0, j = 0; i < dur; i++) {
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accum += ring_dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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int16_t wi = (ring_waveform[idx] * ring_buffer[i]) / 32768;
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for (int c = 0; c < synth_chan; c++) {
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out_buffer32[j] += (wi * loudness[c]) / 32768;
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j++;
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}
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}
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synth->ring_accum[chan] = accum;
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} else {
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uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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uint32_t accum = synth->accum[chan];
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if (dds_rate > lim / 2) {
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// beyond nyquist, can't play note
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return;
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}
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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for (uint16_t i = 0, j = 0; i < dur; i++) {
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accum += dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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int16_t wi = waveform[idx];
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for (int c = 0; c < synth_chan; c++) {
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out_buffer32[j] += (wi * loudness[c]) / 65536;
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j++;
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}
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}
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synth->accum[chan] = accum;
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}
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}
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void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t *buffer_length, uint8_t channel) {
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if (channel == synth->other_channel) {
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@ -186,145 +324,14 @@ void synthio_synth_synthesize(synthio_synth_t *synth, uint8_t **bufptr, uint32_t
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uint16_t dur = MIN(SYNTHIO_MAX_DUR, synth->span.dur);
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synth->span.dur -= dur;
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int32_t sample_rate = synth->sample_rate;
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int32_t out_buffer32[dur * synth->channel_count];
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memset(out_buffer32, 0, sizeof(out_buffer32));
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for (int chan = 0; chan < CIRCUITPY_SYNTHIO_MAX_CHANNELS; chan++) {
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mp_obj_t note_obj = synth->span.note_obj[chan];
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if (note_obj == SYNTHIO_SILENCE) {
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synth->accum[chan] = 0;
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continue;
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}
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if (synth->envelope_state[chan].level == 0) {
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// note is truly finished, but we only just noticed
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synth->span.note_obj[chan] = SYNTHIO_SILENCE;
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continue;
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}
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// adjust loudness by envelope
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uint16_t loudness[2] = {synth->envelope_state[chan].level,synth->envelope_state[chan].level};
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uint32_t dds_rate;
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const int16_t *waveform = synth->waveform;
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uint32_t waveform_length = synth->waveform_length;
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uint32_t ring_dds_rate = 0;
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const int16_t *ring_waveform = NULL;
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uint32_t ring_waveform_length = 0;
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if (mp_obj_is_small_int(note_obj)) {
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uint8_t note = mp_obj_get_int(note_obj);
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uint8_t octave = note / 12;
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uint16_t base_freq = notes[note % 12];
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// rate = base_freq * waveform_length
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// den = sample_rate * 2 ^ (10 - octave)
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// den = sample_rate * 2 ^ 10 / 2^octave
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// dds_rate = 2^SHIFT * rate / den
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// dds_rate = 2^(SHIFT-10+octave) * base_freq * waveform_length / sample_rate
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dds_rate = (sample_rate / 2 + ((uint64_t)(base_freq * waveform_length) << (SYNTHIO_FREQUENCY_SHIFT - 10 + octave))) / sample_rate;
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} else {
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synthio_note_obj_t *note = MP_OBJ_TO_PTR(note_obj);
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int32_t frequency_scaled = synthio_note_step(note, sample_rate, dur, loudness);
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if (note->waveform_buf.buf) {
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waveform = note->waveform_buf.buf;
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waveform_length = note->waveform_buf.len / 2;
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}
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dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)frequency_scaled * waveform_length, sample_rate);
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if (note->ring_frequency_scaled != 0 && note->ring_waveform_buf.buf) {
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ring_waveform = note->ring_waveform_buf.buf;
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ring_waveform_length = note->ring_waveform_buf.len / 2;
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ring_dds_rate = synthio_frequency_convert_scaled_to_dds((uint64_t)note->ring_frequency_scaled * ring_waveform_length, sample_rate);
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uint32_t lim = ring_waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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if (ring_dds_rate > lim / 2) {
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ring_dds_rate = 0; // can't ring at that frequency
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}
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}
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}
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int synth_chan = synth->channel_count;
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if (ring_dds_rate) {
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uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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uint32_t accum = synth->accum[chan];
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if (dds_rate > lim / 2) {
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// beyond nyquist, can't play note
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continue;
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}
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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int32_t ring_buffer[dur];
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// first, fill with waveform
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for (uint16_t i = 0; i < dur; i++) {
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accum += dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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ring_buffer[i] = waveform[idx];
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}
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synth->accum[chan] = accum;
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// now modulate by ring and accumulate
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accum = synth->ring_accum[chan];
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lim = ring_waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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for (uint16_t i = 0, j = 0; i < dur; i++) {
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accum += ring_dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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int16_t wi = (ring_waveform[idx] * ring_buffer[i]) / 32768;
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for (int c = 0; c < synth_chan; c++) {
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out_buffer32[j] += (wi * loudness[c]) / 32768;
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j++;
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}
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}
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synth->ring_accum[chan] = accum;
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} else {
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uint32_t lim = waveform_length << SYNTHIO_FREQUENCY_SHIFT;
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uint32_t accum = synth->accum[chan];
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if (dds_rate > lim / 2) {
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// beyond nyquist, can't play note
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continue;
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}
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// can happen if note waveform gets set mid-note, but the expensive modulo is usually avoided
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if (accum > lim) {
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accum %= lim;
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}
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for (uint16_t i = 0, j = 0; i < dur; i++) {
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accum += dds_rate;
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// because dds_rate is low enough, the subtraction is guaranteed to go back into range, no expensive modulo needed
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if (accum > lim) {
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accum -= lim;
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}
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int16_t idx = accum >> SYNTHIO_FREQUENCY_SHIFT;
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int16_t wi = waveform[idx];
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for (int c = 0; c < synth_chan; c++) {
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out_buffer32[j] += (wi * loudness[c]) / 65536;
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j++;
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}
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}
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synth->accum[chan] = accum;
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}
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synth_note_into_buffer(synth, chan, out_buffer32, dur);
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}
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int16_t *out_buffer16 = (int16_t *)(void *)synth->buffers[synth->buffer_index];
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// mix down audio
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@ -358,8 +365,7 @@ bool synthio_synth_deinited(synthio_synth_t *synth) {
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}
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void synthio_synth_deinit(synthio_synth_t *synth) {
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m_del(uint8_t, synth->buffers[0], synth->buffer_length);
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m_del(uint8_t, synth->buffers[1], synth->buffer_length);
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synth->filter_buffer = NULL;
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synth->buffers[0] = NULL;
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synth->buffers[1] = NULL;
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}
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@ -374,15 +380,19 @@ mp_obj_t synthio_synth_envelope_get(synthio_synth_t *synth) {
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}
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void synthio_synth_init(synthio_synth_t *synth, uint32_t sample_rate, int channel_count, mp_obj_t waveform_obj, mp_obj_t filter_obj, mp_obj_t envelope_obj) {
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synthio_synth_parse_waveform(&synth->waveform_bufinfo, waveform_obj);
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synthio_synth_parse_filter(&synth->filter_bufinfo, filter_obj);
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mp_arg_validate_int_range(channel_count, 1, 2, MP_QSTR_channel_count);
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synth->buffer_length = SYNTHIO_MAX_DUR * SYNTHIO_BYTES_PER_SAMPLE * channel_count;
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synth->buffers[0] = m_malloc(synth->buffer_length, false);
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synth->buffers[1] = m_malloc(synth->buffer_length, false);
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if (synth->filter_bufinfo.len) {
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synth->filter_buffer_length = (synth->filter_bufinfo.len + SYNTHIO_MAX_DUR) * channel_count * sizeof(int32_t);
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synth->filter_buffer = m_malloc(synth->filter_buffer_length, false);
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}
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synth->channel_count = channel_count;
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synth->other_channel = -1;
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synth->waveform_obj = waveform_obj;
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synthio_synth_parse_waveform(&synth->waveform_bufinfo, waveform_obj);
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synthio_synth_parse_filter(&synth->filter_bufinfo, filter_obj);
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synth->sample_rate = sample_rate;
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synthio_synth_envelope_set(synth, envelope_obj);
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@ -64,12 +64,11 @@ typedef struct synthio_synth {
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uint32_t sample_rate;
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uint32_t total_envelope;
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int16_t *buffers[2];
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const int16_t *waveform;
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int32_t *filter_buffer;
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uint8_t channel_count;
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uint16_t buffer_length;
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uint16_t buffer_length, filter_buffer_length;
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uint16_t last_buffer_length;
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uint8_t other_channel, buffer_index, other_buffer_index;
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uint16_t waveform_length;
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mp_buffer_info_t waveform_bufinfo, filter_bufinfo;
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synthio_envelope_definition_t global_envelope_definition;
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mp_obj_t waveform_obj, filter_obj, envelope_obj;
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