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LibDSP: Improve const correctness

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This commit is contained in:
kleines Filmröllchen 2022-05-11 21:49:41 +02:00 • committed by Linus Groh
parent 4a6ebb8beb
commit bcb331b862
8 changed files with 23 additions and 22 deletions
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2
Userland/Libraries/LibDSP/Clip.h
View file

@ -47,7 +47,7 @@ class NoteClip final : public Clip {
public:
void set_note(RollNote note);
Array<SinglyLinkedList<RollNote>, note_count>& notes() { return m_notes; }
Array<SinglyLinkedList<RollNote>, note_count> const& notes() const { return m_notes; }
private:
Array<SinglyLinkedList<RollNote>, note_count> m_notes;

11
Userland/Libraries/LibDSP/Music.h
View file

@ -15,10 +15,9 @@
namespace LibDSP {
// FIXME: Audio::Frame is 64-bit float, which is quite large for long clips.
using Sample = Audio::Sample;
Sample const SAMPLE_OFF = { 0.0, 0.0 };
constexpr Sample const SAMPLE_OFF = { 0.0, 0.0 };
struct RollNote {
constexpr u32 length() const { return (off_sample - on_sample) + 1; }
@ -28,7 +27,7 @@ struct RollNote {
u8 pitch;
i8 velocity;
Envelope to_envelope(u32 time, u32 attack_samples, u32 decay_samples, u32 release_samples)
constexpr Envelope to_envelope(u32 time, u32 attack_samples, u32 decay_samples, u32 release_samples) const
{
i64 time_since_end = static_cast<i64>(time) - static_cast<i64>(off_sample);
// We're before the end of this note.
@ -58,7 +57,7 @@ struct RollNote {
return Envelope::from_release(static_cast<double>(time_since_end) / static_cast<double>(release_samples));
}
constexpr bool is_playing(u32 time) { return on_sample <= time && time <= off_sample; }
constexpr bool is_playing(u32 time) const { return on_sample <= time && time <= off_sample; }
};
enum class SignalType : u8 {
@ -178,8 +177,8 @@ constexpr double note_frequencies[] = {
3729.3100921447249,
3951.0664100489994,
};
constexpr size_t note_count = array_size(note_frequencies);
constexpr size_t const note_count = array_size(note_frequencies);
constexpr double middle_c = note_frequencies[36];
constexpr double const middle_c = note_frequencies[36];
}

1
Userland/Libraries/LibDSP/Processor.h
View file

@ -34,6 +34,7 @@ public:
SignalType input_type() const { return m_input_type; }
SignalType output_type() const { return m_output_type; }
Vector<ProcessorParameter&>& parameters() { return m_parameters; }
Vector<ProcessorParameter&> const& parameters() const { return m_parameters; }
private:
SignalType const m_input_type;

12
Userland/Libraries/LibDSP/Synthesizers.cpp
View file

@ -66,7 +66,7 @@ Signal Classic::process_impl(Signal const& input_signal)
}
// Linear ADSR envelope with no peak adjustment.
double Classic::volume_from_envelope(Envelope const& envelope)
double Classic::volume_from_envelope(Envelope const& envelope) const
{
switch (static_cast<EnvelopeState>(envelope)) {
case EnvelopeState::Off:
@ -102,12 +102,12 @@ double Classic::wave_position(u8 note)
VERIFY_NOT_REACHED();
}
double Classic::samples_per_cycle(u8 note)
double Classic::samples_per_cycle(u8 note) const
{
return m_transport->sample_rate() / note_frequencies[note];
}
double Classic::sin_position(u8 note)
double Classic::sin_position(u8 note) const
{
double spc = samples_per_cycle(note);
double cycle_pos = m_transport->time() / spc;
@ -115,14 +115,14 @@ double Classic::sin_position(u8 note)
}
// Absolute value of the saw wave "flips" the negative portion into the positive, creating a ramp up and down.
double Classic::triangle_position(u8 note)
double Classic::triangle_position(u8 note) const
{
double saw = saw_position(note);
return AK::fabs(saw) * 2 - 1;
}
// The first half of the cycle period is 1, the other half -1.
double Classic::square_position(u8 note)
double Classic::square_position(u8 note) const
{
double spc = samples_per_cycle(note);
double progress = AK::fmod(static_cast<double>(m_transport->time()), spc) / spc;
@ -130,7 +130,7 @@ double Classic::square_position(u8 note)
}
// Modulus creates inverse saw, which we need to flip and scale.
double Classic::saw_position(u8 note)
double Classic::saw_position(u8 note) const
{
double spc = samples_per_cycle(note);
double unscaled = spc - AK::fmod(static_cast<double>(m_transport->time()), spc);

12
Userland/Libraries/LibDSP/Synthesizers.h
View file

@ -49,13 +49,13 @@ public:
private:
virtual Signal process_impl(Signal const&) override;
double volume_from_envelope(Envelope const&);
double volume_from_envelope(Envelope const&) const;
double wave_position(u8 note);
double samples_per_cycle(u8 note);
double sin_position(u8 note);
double triangle_position(u8 note);
double square_position(u8 note);
double saw_position(u8 note);
double samples_per_cycle(u8 note) const;
double sin_position(u8 note) const;
double triangle_position(u8 note) const;
double square_position(u8 note) const;
double saw_position(u8 note) const;
double noise_position(u8 note);
double get_random_from_seed(u64 note);

4
Userland/Libraries/LibDSP/Track.cpp
View file

@ -79,8 +79,8 @@ void NoteTrack::compute_current_clips_signal()
return;
// FIXME: performance?
for (auto& note_list : playing_clip->notes()) {
for (auto& note : note_list) {
for (auto const& note_list : playing_clip->notes()) {
for (auto const& note : note_list) {
if (note.on_sample >= time && note.off_sample >= time)
break;
if (note.on_sample <= time && note.off_sample >= time)

2
Userland/Libraries/LibDSP/Track.h
View file

@ -27,7 +27,7 @@ public:
Sample current_signal();
NonnullRefPtrVector<Processor> const& processor_chain() const { return m_processor_chain; }
NonnullRefPtr<Transport> const transport() const { return m_transport; }
NonnullRefPtr<Transport const> transport() const { return m_transport; }
protected:
Track(NonnullRefPtr<Transport> transport)

1
Userland/Libraries/LibDSP/Transport.h
View file

@ -16,6 +16,7 @@ namespace LibDSP {
class Transport final : public RefCounted<Transport> {
public:
constexpr u32& time() { return m_time; }
constexpr u32 time() const { return m_time; }
constexpr u16 beats_per_minute() const { return m_beats_per_minute; }
constexpr double current_second() const { return static_cast<double>(m_time) / m_sample_rate; }
constexpr double samples_per_measure() const { return (1.0 / m_beats_per_minute) * 60.0 * m_sample_rate; }