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/*
* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include <AK/FixedArray.h>
#include <AK/NoAllocationGuard.h>
#include <AK/NonnullRefPtr.h>
#include <AK/Optional.h>
#include <AK/StdLibExtras.h>
#include <AK/TypedTransfer.h>
#include <AK/Types.h>
#include <LibDSP/Music.h>
#include <LibDSP/Processor.h>
#include <LibDSP/Track.h>
namespace DSP {
bool Track::add_processor(NonnullRefPtr<Processor> new_processor)
{
m_processor_chain.append(move(new_processor));
if (!check_processor_chain_valid()) {
(void)m_processor_chain.take_last();
return false;
}
return true;
}
bool Track::check_processor_chain_valid_with_initial_type(SignalType initial_type) const
{
Processor const* previous_processor = nullptr;
for (auto& processor : m_processor_chain) {
// The first processor must have the given initial signal type as input.
if (previous_processor == nullptr) {
if (processor->input_type() != initial_type)
return false;
} else if (previous_processor->output_type() != processor->input_type())
return false;
previous_processor = processor.ptr();
}
return true;
}
NonnullRefPtr<Synthesizers::Classic> Track::synth()
{
return static_ptr_cast<Synthesizers::Classic>(m_processor_chain[0]);
}
NonnullRefPtr<Effects::Delay> Track::delay()
{
return static_ptr_cast<Effects::Delay>(m_processor_chain[1]);
}
bool AudioTrack::check_processor_chain_valid() const
{
return check_processor_chain_valid_with_initial_type(SignalType::Sample);
}
bool NoteTrack::check_processor_chain_valid() const
{
return check_processor_chain_valid_with_initial_type(SignalType::Note);
}
ErrorOr<void> Track::resize_internal_buffers_to(size_t buffer_size)
{
m_secondary_sample_buffer = TRY(FixedArray<Sample>::create(buffer_size));
return {};
}
void Track::current_signal(FixedArray<Sample>& output_signal)
{
// This is real-time code. We must NEVER EVER EVER allocate.
NoAllocationGuard guard;
VERIFY(m_secondary_sample_buffer.type() == SignalType::Sample);
VERIFY(output_signal.size() == m_secondary_sample_buffer.get<FixedArray<Sample>>().size());
compute_current_clips_signal();
Signal* source_signal = &m_current_signal;
// This provides an audio buffer of the right size. It is not allocated here, but whenever we are informed about a buffer size change.
Signal* target_signal = &m_secondary_sample_buffer;
for (auto& processor : m_processor_chain) {
// Depending on what the processor needs to have as output, we need to place either a pre-allocated note hash map or a pre-allocated sample buffer in the target signal.
if (processor->output_type() == SignalType::Note)
target_signal = &m_secondary_note_buffer;
else
target_signal = &m_secondary_sample_buffer;
processor->process(*source_signal, *target_signal);
swap(source_signal, target_signal);
}
VERIFY(source_signal->type() == SignalType::Sample);
VERIFY(output_signal.size() == source_signal->get<FixedArray<Sample>>().size());
// The last processor is the fixed mastering processor. This can write directly to the output data. We also just trust this processor that it does the right thing :^)
m_track_mastering->process_to_fixed_array(*source_signal, output_signal);
}
void NoteTrack::compute_current_clips_signal()
{
// FIXME: Handle looping properly
u32 start_time = m_transport->time();
VERIFY(m_secondary_sample_buffer.type() == SignalType::Sample);
size_t sample_count = m_secondary_sample_buffer.get<FixedArray<Sample>>().size();
u32 end_time = start_time + static_cast<u32>(sample_count);
// Find the currently playing clips.
// We can't handle more than 32 playing clips at a time, but that is a ridiculous number.
Array<RefPtr<NoteClip>, 32> playing_clips;
size_t playing_clips_index = 0;
for (auto& clip : m_clips) {
// A clip is playing if its start time or end time fall in the current time range.
// Or, if they both enclose the current time range.
if ((clip->start() <= start_time && clip->end() >= end_time)
|| (clip->start() >= start_time && clip->start() < end_time)
|| (clip->end() > start_time && clip->end() <= end_time)) {
VERIFY(playing_clips_index < playing_clips.size());
playing_clips[playing_clips_index++] = clip;
}
}
auto& current_notes = m_current_signal.get<RollNotes>();
m_current_signal.get<RollNotes>().fill({});
if (playing_clips_index == 0)
return;
for (auto const& playing_clip : playing_clips) {
if (playing_clip.is_null())
break;
for (auto const& note : playing_clip->notes()) {
if (note.is_playing_during(start_time, end_time))
current_notes[note.pitch] = note;
}
}
for (auto const& keyboard_note : m_keyboard->notes()) {
if (!keyboard_note.has_value() || !keyboard_note->is_playing_during(start_time, end_time))
continue;
// Always overwrite roll notes with keyboard notes.
current_notes[keyboard_note->pitch] = keyboard_note;
}
}
void AudioTrack::compute_current_clips_signal()
{
// This is quite involved as we need to look at multiple clips and take looping into account.
TODO();
}
Optional<RollNote> NoteTrack::note_at(u32 time, u8 pitch) const
{
for (auto& clip : m_clips) {
if (time >= clip->start() && time <= clip->end())
return clip->note_at(time, pitch);
}
return {};
}
void NoteTrack::set_note(RollNote note)
{
for (auto& clip : m_clips) {
if (clip->start() <= note.on_sample && clip->end() >= note.on_sample)
clip->set_note(note);
}
}
void NoteTrack::remove_note(RollNote note)
{
for (auto& clip : m_clips)
clip->remove_note(note);
}
void NoteTrack::add_clip(u32 start_time, u32 end_time)
{
m_clips.append(AK::make_ref_counted<NoteClip>(start_time, end_time));
}
}
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