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LibDSP+Piano: Convert DSP APIs to accept entire sample ranges
This has mainly performance benefits, so that we only need to call into all processors once for every audio buffer segment. It requires adjusting quite some logic in most processors and in Track, as we have to consider a larger collection of notes and samples at each step. There's some cautionary TODOs in the currently unused LibDSP tracks because they don't do things properly yet.
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9 changed files with 123 additions and 75 deletions
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@ -31,38 +31,41 @@ Classic::Classic(NonnullRefPtr<Transport> transport)
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m_parameters.append(m_release);
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}
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Signal Classic::process_impl(Signal const& input_signal)
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void Classic::process_impl(Signal const& input_signal, [[maybe_unused]] Signal& output_signal)
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{
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auto& in = input_signal.get<RollNotes>();
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auto const& in = input_signal.get<RollNotes>();
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auto& output_samples = output_signal.get<FixedArray<Sample>>();
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Sample out;
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// Do this for every time step and set the signal accordingly.
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for (size_t sample_index = 0; sample_index < output_samples.size(); ++sample_index) {
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Sample& out = output_samples[sample_index];
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u32 sample_time = m_transport->time() + sample_index;
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SinglyLinkedList<PitchedEnvelope> playing_envelopes;
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SinglyLinkedList<PitchedEnvelope> playing_envelopes;
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// "Press" the necessary notes in the internal representation,
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// and "release" all of the others
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for (u8 i = 0; i < note_frequencies.size(); ++i) {
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if (auto maybe_note = in.get(i); maybe_note.has_value())
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m_playing_notes.set(i, maybe_note.value());
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// "Press" the necessary notes in the internal representation,
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// and "release" all of the others
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for (u8 i = 0; i < note_frequencies.size(); ++i) {
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if (auto maybe_note = in.get(i); maybe_note.has_value())
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m_playing_notes.set(i, maybe_note.value());
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if (m_playing_notes.contains(i)) {
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Envelope note_envelope = m_playing_notes.get(i)->to_envelope(m_transport->time(), m_attack * m_transport->ms_sample_rate(), m_decay * m_transport->ms_sample_rate(), m_release * m_transport->ms_sample_rate());
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if (!note_envelope.is_active()) {
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m_playing_notes.remove(i);
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continue;
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if (m_playing_notes.contains(i)) {
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Envelope note_envelope = m_playing_notes.get(i)->to_envelope(sample_time, m_attack * m_transport->ms_sample_rate(), m_decay * m_transport->ms_sample_rate(), m_release * m_transport->ms_sample_rate());
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if (!note_envelope.is_active()) {
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m_playing_notes.remove(i);
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continue;
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}
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playing_envelopes.append(PitchedEnvelope { note_envelope, i });
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}
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}
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playing_envelopes.append(PitchedEnvelope { note_envelope, i });
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for (auto envelope : playing_envelopes) {
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double volume = volume_from_envelope(envelope);
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double wave = wave_position(envelope.note);
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out += volume * wave;
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}
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}
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for (auto envelope : playing_envelopes) {
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double volume = volume_from_envelope(envelope);
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double wave = wave_position(envelope.note);
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out += volume * wave;
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}
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return out;
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}
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// Linear ADSR envelope with no peak adjustment.
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