mirror of
https://github.com/RGBCube/serenity
synced 2025-07-26 23:17:46 +00:00
Piano+LibDSP: Move Track to LibDSP
This is a tangly commit and it fixes all the bugs that a plain move would have caused (i.e. we need to touch other logic which had wrong assumptions).
This commit is contained in:
parent
125122a9ab
commit
4941cffdd0
29 changed files with 322 additions and 413 deletions
|
@ -11,6 +11,7 @@
|
|||
#include <AK/StdLibExtras.h>
|
||||
#include <LibAudio/Sample.h>
|
||||
#include <LibDSP/Envelope.h>
|
||||
#include <LibDSP/Music.h>
|
||||
#include <LibDSP/Processor.h>
|
||||
#include <LibDSP/Synthesizers.h>
|
||||
|
||||
|
@ -39,42 +40,45 @@ void Classic::process_impl(Signal const& input_signal, [[maybe_unused]] Signal&
|
|||
// Do this for every time step and set the signal accordingly.
|
||||
for (size_t sample_index = 0; sample_index < output_samples.size(); ++sample_index) {
|
||||
Sample& out = output_samples[sample_index];
|
||||
out = {};
|
||||
u32 sample_time = m_transport->time() + sample_index;
|
||||
|
||||
SinglyLinkedList<PitchedEnvelope> playing_envelopes;
|
||||
Array<Optional<PitchedEnvelope>, note_frequencies.size()> playing_envelopes;
|
||||
|
||||
// "Press" the necessary notes in the internal representation,
|
||||
// and "release" all of the others
|
||||
for (u8 i = 0; i < note_frequencies.size(); ++i) {
|
||||
if (auto maybe_note = in.get(i); maybe_note.has_value())
|
||||
m_playing_notes.set(i, maybe_note.value());
|
||||
if (auto maybe_note = in[i]; maybe_note.has_value())
|
||||
m_playing_notes[i] = maybe_note;
|
||||
|
||||
if (m_playing_notes.contains(i)) {
|
||||
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());
|
||||
if (m_playing_notes[i].has_value()) {
|
||||
Envelope note_envelope = m_playing_notes[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());
|
||||
// There are two conditions for removing notes:
|
||||
// 1. The envelope has expired, regardless of whether the note was still given to us in the input.
|
||||
if (!note_envelope.is_active()) {
|
||||
m_playing_notes.remove(i);
|
||||
m_playing_notes[i] = {};
|
||||
continue;
|
||||
}
|
||||
// 2. The envelope has not expired, but the note was not given to us.
|
||||
// This means that the note abruptly stopped playing; i.e. the audio infrastructure didn't know the length of the notes initially.
|
||||
// That basically means we're dealing with a keyboard note. Chop its end time to end now.
|
||||
if (!note_envelope.is_release() && !in.get(i).has_value()) {
|
||||
if (!note_envelope.is_release() && !in[i].has_value()) {
|
||||
// dbgln("note {} not released, setting release phase, envelope={}", i, note_envelope.envelope);
|
||||
note_envelope.set_release(0);
|
||||
auto real_note = *m_playing_notes.get(i);
|
||||
auto real_note = *m_playing_notes[i];
|
||||
real_note.off_sample = sample_time;
|
||||
m_playing_notes.set(i, real_note);
|
||||
m_playing_notes[i] = real_note;
|
||||
}
|
||||
|
||||
playing_envelopes.append(PitchedEnvelope { note_envelope, i });
|
||||
playing_envelopes[i] = PitchedEnvelope { note_envelope, i };
|
||||
}
|
||||
}
|
||||
|
||||
for (auto envelope : playing_envelopes) {
|
||||
double volume = volume_from_envelope(envelope);
|
||||
double wave = wave_position(envelope.note);
|
||||
if (!envelope.has_value())
|
||||
continue;
|
||||
double volume = volume_from_envelope(*envelope);
|
||||
double wave = wave_position(sample_time, envelope->note);
|
||||
out += volume * wave;
|
||||
}
|
||||
}
|
||||
|
@ -100,19 +104,19 @@ double Classic::volume_from_envelope(Envelope const& envelope) const
|
|||
VERIFY_NOT_REACHED();
|
||||
}
|
||||
|
||||
double Classic::wave_position(u8 note)
|
||||
double Classic::wave_position(u32 sample_time, u8 note)
|
||||
{
|
||||
switch (m_waveform) {
|
||||
case Sine:
|
||||
return sin_position(note);
|
||||
return sin_position(sample_time, note);
|
||||
case Triangle:
|
||||
return triangle_position(note);
|
||||
return triangle_position(sample_time, note);
|
||||
case Square:
|
||||
return square_position(note);
|
||||
return square_position(sample_time, note);
|
||||
case Saw:
|
||||
return saw_position(note);
|
||||
return saw_position(sample_time, note);
|
||||
case Noise:
|
||||
return noise_position(note);
|
||||
return noise_position(sample_time, note);
|
||||
}
|
||||
VERIFY_NOT_REACHED();
|
||||
}
|
||||
|
@ -122,43 +126,43 @@ double Classic::samples_per_cycle(u8 note) const
|
|||
return m_transport->sample_rate() / note_frequencies[note];
|
||||
}
|
||||
|
||||
double Classic::sin_position(u8 note) const
|
||||
double Classic::sin_position(u32 sample_time, u8 note) const
|
||||
{
|
||||
double spc = samples_per_cycle(note);
|
||||
double cycle_pos = m_transport->time() / spc;
|
||||
double cycle_pos = sample_time / spc;
|
||||
return AK::sin(cycle_pos * 2 * AK::Pi<double>);
|
||||
}
|
||||
|
||||
// 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) const
|
||||
double Classic::triangle_position(u32 sample_time, u8 note) const
|
||||
{
|
||||
double saw = saw_position(note);
|
||||
double saw = saw_position(sample_time, 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) const
|
||||
double Classic::square_position(u32 sample_time, u8 note) const
|
||||
{
|
||||
double spc = samples_per_cycle(note);
|
||||
double progress = AK::fmod(static_cast<double>(m_transport->time()), spc) / spc;
|
||||
double progress = AK::fmod(static_cast<double>(sample_time), spc) / spc;
|
||||
return progress >= 0.5 ? -1 : 1;
|
||||
}
|
||||
|
||||
// Modulus creates inverse saw, which we need to flip and scale.
|
||||
double Classic::saw_position(u8 note) const
|
||||
double Classic::saw_position(u32 sample_time, u8 note) const
|
||||
{
|
||||
double spc = samples_per_cycle(note);
|
||||
double unscaled = spc - AK::fmod(static_cast<double>(m_transport->time()), spc);
|
||||
double unscaled = spc - AK::fmod(static_cast<double>(sample_time), spc);
|
||||
return unscaled / (samples_per_cycle(note) / 2.) - 1;
|
||||
}
|
||||
|
||||
// We resample the noise twenty times per cycle.
|
||||
double Classic::noise_position(u8 note)
|
||||
double Classic::noise_position(u32 sample_time, u8 note)
|
||||
{
|
||||
double spc = samples_per_cycle(note);
|
||||
u32 getrandom_interval = max(static_cast<u32>(spc / 2), 1);
|
||||
// Note that this code only works well if the processor is called for every increment of time.
|
||||
if (m_transport->time() % getrandom_interval == 0)
|
||||
if (sample_time % getrandom_interval == 0)
|
||||
last_random[note] = (get_random<u16>() / static_cast<double>(NumericLimits<u16>::max()) - .5) * 2;
|
||||
return last_random[note];
|
||||
}
|
||||
|
|
Loading…
Add table
Add a link
Reference in a new issue