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