feat: updated engine version to 4.4-rc1
This commit is contained in:
Sara
parent
ee00efde1f
commit
21ba8e33af
5459 changed files with 1128836 additions and 198305 deletions
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@ -1,4 +1,5 @@
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#!/usr/bin/env python
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from misc.utility.scons_hints import *
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Import("env")
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@ -60,11 +60,22 @@ float AudioEffectAmplify::get_volume_db() const {
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return volume_db;
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}
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void AudioEffectAmplify::set_volume_linear(float p_volume) {
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set_volume_db(Math::linear_to_db(p_volume));
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}
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float AudioEffectAmplify::get_volume_linear() const {
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return Math::db_to_linear(get_volume_db());
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}
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void AudioEffectAmplify::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_volume_db", "volume"), &AudioEffectAmplify::set_volume_db);
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ClassDB::bind_method(D_METHOD("get_volume_db"), &AudioEffectAmplify::get_volume_db);
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ClassDB::bind_method(D_METHOD("set_volume_linear", "volume"), &AudioEffectAmplify::set_volume_linear);
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ClassDB::bind_method(D_METHOD("get_volume_linear"), &AudioEffectAmplify::get_volume_linear);
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "volume_db", PROPERTY_HINT_RANGE, "-80,24,0.01,suffix:dB"), "set_volume_db", "get_volume_db");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "volume_linear", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NONE), "set_volume_linear", "get_volume_linear");
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}
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AudioEffectAmplify::AudioEffectAmplify() {
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@ -60,6 +60,9 @@ public:
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void set_volume_db(float p_volume);
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float get_volume_db() const;
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void set_volume_linear(float p_volume);
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float get_volume_linear() const;
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AudioEffectAmplify();
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};
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@ -30,6 +30,8 @@
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#include "audio_effect_capture.h"
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#include "servers/audio_server.h"
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bool AudioEffectCapture::can_get_buffer(int p_frames) const {
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return buffer.data_left() >= p_frames;
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}
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@ -31,13 +31,10 @@
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#ifndef AUDIO_EFFECT_CAPTURE_H
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#define AUDIO_EFFECT_CAPTURE_H
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#include "core/config/engine.h"
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#include "core/math/audio_frame.h"
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#include "core/object/ref_counted.h"
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#include "core/templates/ring_buffer.h"
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#include "core/templates/vector.h"
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#include "servers/audio/audio_effect.h"
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#include "servers/audio_server.h"
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class AudioEffectCapture;
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@ -29,7 +29,6 @@
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/**************************************************************************/
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#include "audio_effect_phaser.h"
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#include "core/math/math_funcs.h"
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#include "servers/audio_server.h"
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void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
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@ -286,6 +286,11 @@ void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
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/* clang-format on */
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void AudioEffectPitchShiftInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
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// Avoid distortion by skipping processing if pitch_scale is 1.0.
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if (Math::is_equal_approx(base->pitch_scale, 1.0f)) {
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return;
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}
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float sample_rate = AudioServer::get_singleton()->get_mix_rate();
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float *in_l = (float *)p_src_frames;
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@ -77,7 +77,7 @@ class AudioEffectPitchShift : public AudioEffect {
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public:
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friend class AudioEffectPitchShiftInstance;
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enum FFTSize {
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enum FFTSize : unsigned int {
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FFT_SIZE_256,
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FFT_SIZE_512,
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FFT_SIZE_1024,
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@ -30,10 +30,7 @@
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#include "audio_effect_record.h"
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#ifdef TOOLS_ENABLED
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// FIXME: This file shouldn't depend on editor stuff.
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#include "editor/import/resource_importer_wav.h"
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#endif
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#include "core/io/marshalls.h"
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void AudioEffectRecordInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
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if (!is_recording) {
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@ -125,7 +122,7 @@ Ref<AudioEffectInstance> AudioEffectRecord::instantiate() {
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ins.instantiate();
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ins->is_recording = false;
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//Re-using the buffer size calculations from audio_effect_delay.cpp
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// Reusing the buffer size calculations from audio_effect_delay.cpp.
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float ring_buffer_max_size = IO_BUFFER_SIZE_MS;
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ring_buffer_max_size /= 1000.0; //convert to seconds
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ring_buffer_max_size *= AudioServer::get_singleton()->get_mix_rate();
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@ -149,7 +146,7 @@ Ref<AudioEffectInstance> AudioEffectRecord::instantiate() {
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ensure_thread_stopped();
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bool is_currently_recording = false;
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if (current_instance != nullptr) {
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if (current_instance.is_valid()) {
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is_currently_recording = current_instance->is_recording;
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}
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if (is_currently_recording) {
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@ -161,28 +158,28 @@ Ref<AudioEffectInstance> AudioEffectRecord::instantiate() {
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}
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void AudioEffectRecord::ensure_thread_stopped() {
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if (current_instance != nullptr) {
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if (current_instance.is_valid()) {
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current_instance->finish();
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}
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}
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void AudioEffectRecord::set_recording_active(bool p_record) {
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if (p_record) {
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if (current_instance == nullptr) {
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if (current_instance.is_null()) {
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WARN_PRINT("Recording should not be set as active before Godot has initialized.");
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return;
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}
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ensure_thread_stopped();
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current_instance->init();
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} else {
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if (current_instance != nullptr) {
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if (current_instance.is_valid()) {
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current_instance->is_recording = false;
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}
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}
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}
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bool AudioEffectRecord::is_recording_active() const {
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if (current_instance == nullptr) {
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if (current_instance.is_null()) {
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return false;
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} else {
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return current_instance->is_recording;
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@ -241,12 +238,8 @@ Ref<AudioStreamWAV> AudioEffectRecord::get_recording() const {
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Vector<uint8_t> bleft;
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Vector<uint8_t> bright;
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#ifdef TOOLS_ENABLED
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ResourceImporterWAV::_compress_ima_adpcm(left, bleft);
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ResourceImporterWAV::_compress_ima_adpcm(right, bright);
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#else
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ERR_PRINT("AudioEffectRecord cannot do IMA ADPCM compression at runtime.");
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#endif
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AudioStreamWAV::_compress_ima_adpcm(left, bleft);
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AudioStreamWAV::_compress_ima_adpcm(right, bright);
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int dl = bleft.size();
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dst_data.resize(dl * 2);
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@ -259,6 +252,12 @@ Ref<AudioStreamWAV> AudioEffectRecord::get_recording() const {
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w[i * 2 + 0] = rl[i];
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w[i * 2 + 1] = rr[i];
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}
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} else if (dst_format == AudioStreamWAV::FORMAT_QOA) {
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qoa_desc desc = {};
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desc.samples = current_instance->recording_data.size() / 2;
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desc.samplerate = AudioServer::get_singleton()->get_mix_rate();
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desc.channels = 2;
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AudioStreamWAV::_compress_qoa(current_instance->recording_data, dst_data, &desc);
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} else {
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ERR_PRINT("Format not implemented.");
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}
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@ -283,7 +282,7 @@ void AudioEffectRecord::_bind_methods() {
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ClassDB::bind_method(D_METHOD("get_format"), &AudioEffectRecord::get_format);
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ClassDB::bind_method(D_METHOD("get_recording"), &AudioEffectRecord::get_recording);
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ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA-ADPCM"), "set_format", "get_format");
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ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA ADPCM,Quite OK Audio"), "set_format", "get_format");
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}
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AudioEffectRecord::AudioEffectRecord() {
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@ -31,9 +31,6 @@
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#ifndef AUDIO_EFFECT_RECORD_H
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#define AUDIO_EFFECT_RECORD_H
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#include "core/io/file_access.h"
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#include "core/io/marshalls.h"
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#include "core/os/os.h"
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#include "core/os/thread.h"
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#include "scene/resources/audio_stream_wav.h"
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#include "servers/audio/audio_effect.h"
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@ -145,8 +145,8 @@ void AudioEffectSpectrumAnalyzerInstance::process(const AudioFrame *p_src_frames
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}
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//determine time of capture
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double remainer_sec = (temporal_fft_pos / mix_rate); //subtract remainder from mix time
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last_fft_time = time - uint64_t(remainer_sec * 1000000.0);
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double remainder_sec = (temporal_fft_pos / mix_rate); //subtract remainder from mix time
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last_fft_time = time - uint64_t(remainder_sec * 1000000.0);
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}
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void AudioEffectSpectrumAnalyzerInstance::_bind_methods() {
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@ -39,34 +39,34 @@ void AudioEffectStereoEnhanceInstance::process(const AudioFrame *p_src_frames, A
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unsigned int delay_frames = (base->time_pullout / 1000.0) * AudioServer::get_singleton()->get_mix_rate();
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for (int i = 0; i < p_frame_count; i++) {
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float l = p_src_frames[i].left;
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float r = p_src_frames[i].right;
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float left = p_src_frames[i].left;
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float right = p_src_frames[i].right;
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float center = (l + r) / 2.0f;
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float center = (left + right) / 2.0f;
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l = (center + (l - center) * intensity);
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r = (center + (r - center) * intensity);
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left = (center + (left - center) * intensity);
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right = (center + (right - center) * intensity);
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if (surround_mode) {
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float val = (l + r) / 2.0;
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float val = (left + right) / 2.0;
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delay_ringbuff[ringbuff_pos & ringbuff_mask] = val;
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float out = delay_ringbuff[(ringbuff_pos - delay_frames) & ringbuff_mask] * surround_amount;
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l += out;
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r += -out;
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left += out;
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right += -out;
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} else {
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float val = r;
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float val = right;
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delay_ringbuff[ringbuff_pos & ringbuff_mask] = val;
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//r is delayed
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r = delay_ringbuff[(ringbuff_pos - delay_frames) & ringbuff_mask];
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// The right channel is delayed.
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right = delay_ringbuff[(ringbuff_pos - delay_frames) & ringbuff_mask];
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}
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p_dst_frames[i].left = l;
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p_dst_frames[i].right = r;
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p_dst_frames[i].left = left;
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p_dst_frames[i].right = right;
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ringbuff_pos++;
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}
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}
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@ -38,6 +38,15 @@ float AudioStreamGenerator::get_mix_rate() const {
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return mix_rate;
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}
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void AudioStreamGenerator::set_mix_rate_mode(AudioStreamGenerator::AudioStreamGeneratorMixRate p_mix_rate_mode) {
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ERR_FAIL_INDEX(p_mix_rate_mode, AudioStreamGeneratorMixRate::MIX_RATE_MAX);
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mix_rate_mode = p_mix_rate_mode;
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}
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AudioStreamGenerator::AudioStreamGeneratorMixRate AudioStreamGenerator::get_mix_rate_mode() const {
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return mix_rate_mode;
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}
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void AudioStreamGenerator::set_buffer_length(float p_seconds) {
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buffer_len = p_seconds;
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}
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@ -46,11 +55,22 @@ float AudioStreamGenerator::get_buffer_length() const {
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return buffer_len;
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}
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float AudioStreamGenerator::_get_target_rate() const {
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switch (mix_rate_mode) {
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case AudioStreamGeneratorMixRate::MIX_RATE_OUTPUT:
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return AudioServer::get_singleton()->get_mix_rate();
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case AudioStreamGeneratorMixRate::MIX_RATE_INPUT:
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return AudioServer::get_singleton()->get_input_mix_rate();
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default:
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return mix_rate;
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}
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}
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Ref<AudioStreamPlayback> AudioStreamGenerator::instantiate_playback() {
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Ref<AudioStreamGeneratorPlayback> playback;
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playback.instantiate();
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playback->generator = this;
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int target_buffer_size = mix_rate * buffer_len;
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int target_buffer_size = _get_target_rate() * buffer_len;
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playback->buffer.resize(nearest_shift(target_buffer_size));
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playback->buffer.clear();
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return playback;
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@ -72,16 +92,20 @@ void AudioStreamGenerator::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_mix_rate", "hz"), &AudioStreamGenerator::set_mix_rate);
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ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamGenerator::get_mix_rate);
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ClassDB::bind_method(D_METHOD("set_mix_rate_mode", "mode"), &AudioStreamGenerator::set_mix_rate_mode);
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ClassDB::bind_method(D_METHOD("get_mix_rate_mode"), &AudioStreamGenerator::get_mix_rate_mode);
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ClassDB::bind_method(D_METHOD("set_buffer_length", "seconds"), &AudioStreamGenerator::set_buffer_length);
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ClassDB::bind_method(D_METHOD("get_buffer_length"), &AudioStreamGenerator::get_buffer_length);
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ADD_PROPERTY(PropertyInfo(Variant::INT, "mix_rate_mode", PROPERTY_HINT_ENUM, "System Output Rate,System Input Rate,Custom Rate"), "set_mix_rate_mode", "get_mix_rate_mode");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "mix_rate", PROPERTY_HINT_RANGE, "20,192000,1,suffix:Hz"), "set_mix_rate", "get_mix_rate");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "buffer_length", PROPERTY_HINT_RANGE, "0.01,10,0.01,suffix:s"), "set_buffer_length", "get_buffer_length");
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}
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AudioStreamGenerator::AudioStreamGenerator() {
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mix_rate = 44100;
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buffer_len = 0.5;
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BIND_ENUM_CONSTANT(MIX_RATE_OUTPUT);
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BIND_ENUM_CONSTANT(MIX_RATE_INPUT);
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BIND_ENUM_CONSTANT(MIX_RATE_CUSTOM);
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BIND_ENUM_CONSTANT(MIX_RATE_MAX);
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}
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////////////////
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@ -162,12 +186,12 @@ int AudioStreamGeneratorPlayback::_mix_internal(AudioFrame *p_buffer, int p_fram
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skips++;
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}
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mixed += p_frames / generator->get_mix_rate();
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mixed += p_frames / generator->_get_target_rate();
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return p_frames;
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}
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float AudioStreamGeneratorPlayback::get_stream_sampling_rate() {
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return generator->get_mix_rate();
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return generator->_get_target_rate();
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}
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void AudioStreamGeneratorPlayback::start(double p_from_pos) {
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@ -37,16 +37,31 @@
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class AudioStreamGenerator : public AudioStream {
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GDCLASS(AudioStreamGenerator, AudioStream);
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float mix_rate;
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float buffer_len;
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public:
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enum AudioStreamGeneratorMixRate {
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MIX_RATE_OUTPUT,
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MIX_RATE_INPUT,
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MIX_RATE_CUSTOM,
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MIX_RATE_MAX,
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};
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private:
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AudioStreamGeneratorMixRate mix_rate_mode = MIX_RATE_CUSTOM;
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float mix_rate = 44100;
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float buffer_len = 0.5;
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protected:
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static void _bind_methods();
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public:
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float _get_target_rate() const;
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void set_mix_rate(float p_mix_rate);
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float get_mix_rate() const;
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void set_mix_rate_mode(AudioStreamGeneratorMixRate p_mix_rate_mode);
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AudioStreamGeneratorMixRate get_mix_rate_mode() const;
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void set_buffer_length(float p_seconds);
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float get_buffer_length() const;
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@ -55,7 +70,7 @@ public:
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virtual double get_length() const override;
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virtual bool is_monophonic() const override;
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AudioStreamGenerator();
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AudioStreamGenerator() {}
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};
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class AudioStreamGeneratorPlayback : public AudioStreamPlaybackResampled {
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@ -96,4 +111,6 @@ public:
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AudioStreamGeneratorPlayback();
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};
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VARIANT_ENUM_CAST(AudioStreamGenerator::AudioStreamGeneratorMixRate);
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#endif // AUDIO_STREAM_GENERATOR_H
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@ -31,7 +31,7 @@
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#include "eq_filter.h"
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#include "core/error/error_macros.h"
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#include "core/math/math_funcs.h"
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#include "core/math/math_defs.h"
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#include <math.h>
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@ -32,7 +32,6 @@
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#define EQ_FILTER_H
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#include "core/templates/vector.h"
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#include "core/typedefs.h"
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class EQ {
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public:
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@ -30,7 +30,8 @@
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#include "reverb_filter.h"
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#include "core/math/math_funcs.h"
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#include "core/math/audio_frame.h"
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#include "core/os/memory.h"
|
||||
|
||||
#include <math.h>
|
||||
|
||||
|
|
|
|||
|
|
@ -31,10 +31,6 @@
|
|||
#ifndef REVERB_FILTER_H
|
||||
#define REVERB_FILTER_H
|
||||
|
||||
#include "core/math/audio_frame.h"
|
||||
#include "core/os/memory.h"
|
||||
#include "core/typedefs.h"
|
||||
|
||||
class Reverb {
|
||||
public:
|
||||
enum {
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue