godot/servers/audio/audio_stream.cpp
Hugo Locurcio 35d788cff5
Remove warning when playing random no-repeat sound with only 1 sound in pool
This makes setting up sounds for random pitch/volume faster, as you
don't have to change the mode from Random (Avoid Repeats) to Random
anymore if you only care about randomizing pitch/volume but want
to prevent a warning message from appearing on every playback.
2022-12-09 23:06:51 +01:00

825 lines
26 KiB
C++

/*************************************************************************/
/* audio_stream.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_stream.h"
#include "core/config/project_settings.h"
#include "core/os/os.h"
void AudioStreamPlayback::start(double p_from_pos) {
if (GDVIRTUAL_CALL(_start, p_from_pos)) {
return;
}
ERR_FAIL_MSG("AudioStreamPlayback::start unimplemented!");
}
void AudioStreamPlayback::stop() {
if (GDVIRTUAL_CALL(_stop)) {
return;
}
ERR_FAIL_MSG("AudioStreamPlayback::stop unimplemented!");
}
bool AudioStreamPlayback::is_playing() const {
bool ret;
if (GDVIRTUAL_CALL(_is_playing, ret)) {
return ret;
}
ERR_FAIL_V_MSG(false, "AudioStreamPlayback::is_playing unimplemented!");
}
int AudioStreamPlayback::get_loop_count() const {
int ret = 0;
GDVIRTUAL_CALL(_get_loop_count, ret);
return ret;
}
double AudioStreamPlayback::get_playback_position() const {
double ret;
if (GDVIRTUAL_CALL(_get_playback_position, ret)) {
return ret;
}
ERR_FAIL_V_MSG(0, "AudioStreamPlayback::get_playback_position unimplemented!");
}
void AudioStreamPlayback::seek(double p_time) {
GDVIRTUAL_CALL(_seek, p_time);
}
int AudioStreamPlayback::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
int ret = 0;
GDVIRTUAL_REQUIRED_CALL(_mix, p_buffer, p_rate_scale, p_frames, ret);
return ret;
}
void AudioStreamPlayback::tag_used_streams() {
GDVIRTUAL_CALL(_tag_used_streams);
}
void AudioStreamPlayback::_bind_methods() {
GDVIRTUAL_BIND(_start, "from_pos")
GDVIRTUAL_BIND(_stop)
GDVIRTUAL_BIND(_is_playing)
GDVIRTUAL_BIND(_get_loop_count)
GDVIRTUAL_BIND(_get_playback_position)
GDVIRTUAL_BIND(_seek, "position")
GDVIRTUAL_BIND(_mix, "buffer", "rate_scale", "frames");
GDVIRTUAL_BIND(_tag_used_streams);
}
//////////////////////////////
void AudioStreamPlaybackResampled::begin_resample() {
//clear cubic interpolation history
internal_buffer[0] = AudioFrame(0.0, 0.0);
internal_buffer[1] = AudioFrame(0.0, 0.0);
internal_buffer[2] = AudioFrame(0.0, 0.0);
internal_buffer[3] = AudioFrame(0.0, 0.0);
//mix buffer
_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
mix_offset = 0;
}
int AudioStreamPlaybackResampled::_mix_internal(AudioFrame *p_buffer, int p_frames) {
int ret = 0;
GDVIRTUAL_REQUIRED_CALL(_mix_resampled, p_buffer, p_frames, ret);
return ret;
}
float AudioStreamPlaybackResampled::get_stream_sampling_rate() {
float ret = 0;
GDVIRTUAL_REQUIRED_CALL(_get_stream_sampling_rate, ret);
return ret;
}
void AudioStreamPlaybackResampled::_bind_methods() {
ClassDB::bind_method(D_METHOD("begin_resample"), &AudioStreamPlaybackResampled::begin_resample);
GDVIRTUAL_BIND(_mix_resampled, "dst_buffer", "frame_count");
GDVIRTUAL_BIND(_get_stream_sampling_rate);
}
int AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
float target_rate = AudioServer::get_singleton()->get_mix_rate();
float playback_speed_scale = AudioServer::get_singleton()->get_playback_speed_scale();
uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale * playback_speed_scale) / double(target_rate)) * double(FP_LEN));
int mixed_frames_total = -1;
int i;
for (i = 0; i < p_frames; i++) {
uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
//standard cubic interpolation (great quality/performance ratio)
//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
float mu = (mix_offset & FP_MASK) / float(FP_LEN);
AudioFrame y0 = internal_buffer[idx - 3];
AudioFrame y1 = internal_buffer[idx - 2];
AudioFrame y2 = internal_buffer[idx - 1];
AudioFrame y3 = internal_buffer[idx - 0];
if (idx >= internal_buffer_end && mixed_frames_total == -1) {
// The internal buffer ends somewhere in this range, and we haven't yet recorded the number of good frames we have.
mixed_frames_total = i;
}
float mu2 = mu * mu;
AudioFrame a0 = 3 * y1 - 3 * y2 + y3 - y0;
AudioFrame a1 = 2 * y0 - 5 * y1 + 4 * y2 - y3;
AudioFrame a2 = y2 - y0;
AudioFrame a3 = 2 * y1;
p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3) / 2;
mix_offset += mix_increment;
while ((mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN) {
internal_buffer[0] = internal_buffer[INTERNAL_BUFFER_LEN + 0];
internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1];
internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2];
internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3];
int mixed_frames = _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
if (mixed_frames != INTERNAL_BUFFER_LEN) {
// internal_buffer[mixed_frames] is the first frame of silence.
internal_buffer_end = mixed_frames;
} else {
// The internal buffer does not contain the first frame of silence.
internal_buffer_end = -1;
}
mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
}
}
if (mixed_frames_total == -1 && i == p_frames) {
mixed_frames_total = p_frames;
}
return mixed_frames_total;
}
////////////////////////////////
Ref<AudioStreamPlayback> AudioStream::instantiate_playback() {
Ref<AudioStreamPlayback> ret;
if (GDVIRTUAL_CALL(_instantiate_playback, ret)) {
return ret;
}
ERR_FAIL_V_MSG(Ref<AudioStreamPlayback>(), "Method must be implemented!");
}
String AudioStream::get_stream_name() const {
String ret;
GDVIRTUAL_CALL(_get_stream_name, ret);
return ret;
}
double AudioStream::get_length() const {
double ret = 0;
GDVIRTUAL_CALL(_get_length, ret);
return ret;
}
bool AudioStream::is_monophonic() const {
bool ret = true;
GDVIRTUAL_CALL(_is_monophonic, ret);
return ret;
}
double AudioStream::get_bpm() const {
double ret = 0;
GDVIRTUAL_CALL(_get_bpm, ret);
return ret;
}
bool AudioStream::has_loop() const {
bool ret = 0;
GDVIRTUAL_CALL(_has_loop, ret);
return ret;
}
int AudioStream::get_bar_beats() const {
int ret = 0;
GDVIRTUAL_CALL(_get_bar_beats, ret);
return ret;
}
int AudioStream::get_beat_count() const {
int ret = 0;
GDVIRTUAL_CALL(_get_beat_count, ret);
return ret;
}
void AudioStream::tag_used(float p_offset) {
if (tagged_frame != AudioServer::get_singleton()->get_mixed_frames()) {
offset_count = 0;
tagged_frame = AudioServer::get_singleton()->get_mixed_frames();
}
if (offset_count < MAX_TAGGED_OFFSETS) {
tagged_offsets[offset_count++] = p_offset;
}
}
uint64_t AudioStream::get_tagged_frame() const {
return tagged_frame;
}
uint32_t AudioStream::get_tagged_frame_count() const {
return offset_count;
}
float AudioStream::get_tagged_frame_offset(int p_index) const {
ERR_FAIL_INDEX_V(p_index, MAX_TAGGED_OFFSETS, 0);
return tagged_offsets[p_index];
}
void AudioStream::_bind_methods() {
ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length);
ClassDB::bind_method(D_METHOD("is_monophonic"), &AudioStream::is_monophonic);
ClassDB::bind_method(D_METHOD("instantiate_playback"), &AudioStream::instantiate_playback);
GDVIRTUAL_BIND(_instantiate_playback);
GDVIRTUAL_BIND(_get_stream_name);
GDVIRTUAL_BIND(_get_length);
GDVIRTUAL_BIND(_is_monophonic);
GDVIRTUAL_BIND(_get_bpm)
GDVIRTUAL_BIND(_get_beat_count)
}
////////////////////////////////
Ref<AudioStreamPlayback> AudioStreamMicrophone::instantiate_playback() {
Ref<AudioStreamPlaybackMicrophone> playback;
playback.instantiate();
playbacks.insert(playback.ptr());
playback->microphone = Ref<AudioStreamMicrophone>((AudioStreamMicrophone *)this);
playback->active = false;
return playback;
}
String AudioStreamMicrophone::get_stream_name() const {
//if (audio_stream.is_valid()) {
//return "Random: " + audio_stream->get_name();
//}
return "Microphone";
}
double AudioStreamMicrophone::get_length() const {
return 0;
}
bool AudioStreamMicrophone::is_monophonic() const {
return true;
}
void AudioStreamMicrophone::_bind_methods() {
}
AudioStreamMicrophone::AudioStreamMicrophone() {
}
int AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_frames) {
AudioDriver::get_singleton()->lock();
Vector<int32_t> buf = AudioDriver::get_singleton()->get_input_buffer();
unsigned int input_size = AudioDriver::get_singleton()->get_input_size();
int mix_rate = AudioDriver::get_singleton()->get_mix_rate();
unsigned int playback_delay = MIN(((50 * mix_rate) / 1000) * 2, buf.size() >> 1);
#ifdef DEBUG_ENABLED
unsigned int input_position = AudioDriver::get_singleton()->get_input_position();
#endif
int mixed_frames = p_frames;
if (playback_delay > input_size) {
for (int i = 0; i < p_frames; i++) {
p_buffer[i] = AudioFrame(0.0f, 0.0f);
}
input_ofs = 0;
} else {
for (int i = 0; i < p_frames; i++) {
if (input_size > input_ofs && (int)input_ofs < buf.size()) {
float l = (buf[input_ofs++] >> 16) / 32768.f;
if ((int)input_ofs >= buf.size()) {
input_ofs = 0;
}
float r = (buf[input_ofs++] >> 16) / 32768.f;
if ((int)input_ofs >= buf.size()) {
input_ofs = 0;
}
p_buffer[i] = AudioFrame(l, r);
} else {
if (mixed_frames == p_frames) {
mixed_frames = i;
}
p_buffer[i] = AudioFrame(0.0f, 0.0f);
}
}
}
#ifdef DEBUG_ENABLED
if (input_ofs > input_position && (int)(input_ofs - input_position) < (p_frames * 2)) {
print_verbose(String(get_class_name()) + " buffer underrun: input_position=" + itos(input_position) + " input_ofs=" + itos(input_ofs) + " input_size=" + itos(input_size));
}
#endif
AudioDriver::get_singleton()->unlock();
return mixed_frames;
}
int AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
return AudioStreamPlaybackResampled::mix(p_buffer, p_rate_scale, p_frames);
}
float AudioStreamPlaybackMicrophone::get_stream_sampling_rate() {
return AudioDriver::get_singleton()->get_mix_rate();
}
void AudioStreamPlaybackMicrophone::start(double p_from_pos) {
if (active) {
return;
}
if (!GLOBAL_GET("audio/driver/enable_input")) {
WARN_PRINT("Need to enable Project settings > Audio > Enable Audio Input option to use capturing.");
return;
}
input_ofs = 0;
if (AudioDriver::get_singleton()->capture_start() == OK) {
active = true;
begin_resample();
}
}
void AudioStreamPlaybackMicrophone::stop() {
if (active) {
AudioDriver::get_singleton()->capture_stop();
active = false;
}
}
bool AudioStreamPlaybackMicrophone::is_playing() const {
return active;
}
int AudioStreamPlaybackMicrophone::get_loop_count() const {
return 0;
}
double AudioStreamPlaybackMicrophone::get_playback_position() const {
return 0;
}
void AudioStreamPlaybackMicrophone::seek(double p_time) {
// Can't seek a microphone input
}
void AudioStreamPlaybackMicrophone::tag_used_streams() {
microphone->tag_used(0);
}
AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() {
microphone->playbacks.erase(this);
stop();
}
AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() {
}
////////////////////////////////
void AudioStreamRandomizer::add_stream(int p_index) {
if (p_index < 0) {
p_index = audio_stream_pool.size();
}
ERR_FAIL_COND(p_index > audio_stream_pool.size());
PoolEntry entry{ nullptr, 1.0f };
audio_stream_pool.insert(p_index, entry);
emit_signal(SNAME("changed"));
notify_property_list_changed();
}
// p_index_to is relative to the array prior to the removal of from.
// Example: [0, 1, 2, 3], move(1, 3) => [0, 2, 1, 3]
void AudioStreamRandomizer::move_stream(int p_index_from, int p_index_to) {
ERR_FAIL_INDEX(p_index_from, audio_stream_pool.size());
// p_index_to == audio_stream_pool.size() is valid (move to end).
ERR_FAIL_COND(p_index_to < 0);
ERR_FAIL_COND(p_index_to > audio_stream_pool.size());
audio_stream_pool.insert(p_index_to, audio_stream_pool[p_index_from]);
// If 'from' is strictly after 'to' we need to increment the index by one because of the insertion.
if (p_index_from > p_index_to) {
p_index_from++;
}
audio_stream_pool.remove_at(p_index_from);
emit_signal(SNAME("changed"));
notify_property_list_changed();
}
void AudioStreamRandomizer::remove_stream(int p_index) {
ERR_FAIL_INDEX(p_index, audio_stream_pool.size());
audio_stream_pool.remove_at(p_index);
emit_signal(SNAME("changed"));
notify_property_list_changed();
}
void AudioStreamRandomizer::set_stream(int p_index, Ref<AudioStream> p_stream) {
ERR_FAIL_INDEX(p_index, audio_stream_pool.size());
audio_stream_pool.write[p_index].stream = p_stream;
emit_signal(SNAME("changed"));
}
Ref<AudioStream> AudioStreamRandomizer::get_stream(int p_index) const {
ERR_FAIL_INDEX_V(p_index, audio_stream_pool.size(), nullptr);
return audio_stream_pool[p_index].stream;
}
void AudioStreamRandomizer::set_stream_probability_weight(int p_index, float p_weight) {
ERR_FAIL_INDEX(p_index, audio_stream_pool.size());
audio_stream_pool.write[p_index].weight = p_weight;
emit_signal(SNAME("changed"));
}
float AudioStreamRandomizer::get_stream_probability_weight(int p_index) const {
ERR_FAIL_INDEX_V(p_index, audio_stream_pool.size(), 0);
return audio_stream_pool[p_index].weight;
}
void AudioStreamRandomizer::set_streams_count(int p_count) {
audio_stream_pool.resize(p_count);
}
int AudioStreamRandomizer::get_streams_count() const {
return audio_stream_pool.size();
}
void AudioStreamRandomizer::set_random_pitch(float p_pitch) {
if (p_pitch < 1) {
p_pitch = 1;
}
random_pitch_scale = p_pitch;
}
float AudioStreamRandomizer::get_random_pitch() const {
return random_pitch_scale;
}
void AudioStreamRandomizer::set_random_volume_offset_db(float p_volume_offset_db) {
if (p_volume_offset_db < 0) {
p_volume_offset_db = 0;
}
random_volume_offset_db = p_volume_offset_db;
}
float AudioStreamRandomizer::get_random_volume_offset_db() const {
return random_volume_offset_db;
}
void AudioStreamRandomizer::set_playback_mode(PlaybackMode p_playback_mode) {
playback_mode = p_playback_mode;
}
AudioStreamRandomizer::PlaybackMode AudioStreamRandomizer::get_playback_mode() const {
return playback_mode;
}
Ref<AudioStreamPlayback> AudioStreamRandomizer::instance_playback_random() {
Ref<AudioStreamPlaybackRandomizer> playback;
playback.instantiate();
playbacks.insert(playback.ptr());
playback->randomizer = Ref<AudioStreamRandomizer>((AudioStreamRandomizer *)this);
double total_weight = 0;
Vector<PoolEntry> local_pool;
for (const PoolEntry &entry : audio_stream_pool) {
if (entry.stream.is_valid() && entry.weight > 0) {
local_pool.push_back(entry);
total_weight += entry.weight;
}
}
if (local_pool.is_empty()) {
return playback;
}
double chosen_cumulative_weight = Math::random(0.0, total_weight);
double cumulative_weight = 0;
for (PoolEntry &entry : local_pool) {
cumulative_weight += entry.weight;
if (cumulative_weight > chosen_cumulative_weight) {
playback->playback = entry.stream->instantiate_playback();
last_playback = entry.stream;
break;
}
}
if (playback->playback.is_null()) {
// This indicates a floating point error. Take the last element.
last_playback = local_pool[local_pool.size() - 1].stream;
playback->playback = local_pool.write[local_pool.size() - 1].stream->instantiate_playback();
}
return playback;
}
Ref<AudioStreamPlayback> AudioStreamRandomizer::instance_playback_no_repeats() {
Ref<AudioStreamPlaybackRandomizer> playback;
double total_weight = 0;
Vector<PoolEntry> local_pool;
for (const PoolEntry &entry : audio_stream_pool) {
if (entry.stream == last_playback) {
continue;
}
if (entry.stream.is_valid() && entry.weight > 0) {
local_pool.push_back(entry);
total_weight += entry.weight;
}
}
if (local_pool.is_empty()) {
// There is only one sound to choose from.
// Always play a random sound while allowing repeats (which always plays the same sound).
playback = instance_playback_random();
return playback;
}
playback.instantiate();
playbacks.insert(playback.ptr());
playback->randomizer = Ref<AudioStreamRandomizer>((AudioStreamRandomizer *)this);
double chosen_cumulative_weight = Math::random(0.0, total_weight);
double cumulative_weight = 0;
for (PoolEntry &entry : local_pool) {
cumulative_weight += entry.weight;
if (cumulative_weight > chosen_cumulative_weight) {
last_playback = entry.stream;
playback->playback = entry.stream->instantiate_playback();
break;
}
}
if (playback->playback.is_null()) {
// This indicates a floating point error. Take the last element.
last_playback = local_pool[local_pool.size() - 1].stream;
playback->playback = local_pool.write[local_pool.size() - 1].stream->instantiate_playback();
}
return playback;
}
Ref<AudioStreamPlayback> AudioStreamRandomizer::instance_playback_sequential() {
Ref<AudioStreamPlaybackRandomizer> playback;
playback.instantiate();
playbacks.insert(playback.ptr());
playback->randomizer = Ref<AudioStreamRandomizer>((AudioStreamRandomizer *)this);
Vector<Ref<AudioStream>> local_pool;
for (const PoolEntry &entry : audio_stream_pool) {
if (entry.stream.is_null()) {
continue;
}
if (local_pool.find(entry.stream) != -1) {
WARN_PRINT("Duplicate stream in sequential playback pool");
continue;
}
local_pool.push_back(entry.stream);
}
if (local_pool.is_empty()) {
return playback;
}
bool found_last_stream = false;
for (Ref<AudioStream> &entry : local_pool) {
if (found_last_stream) {
last_playback = entry;
playback->playback = entry->instantiate_playback();
break;
}
if (entry == last_playback) {
found_last_stream = true;
}
}
if (playback->playback.is_null()) {
// Wrap around
last_playback = local_pool[0];
playback->playback = local_pool.write[0]->instantiate_playback();
}
return playback;
}
Ref<AudioStreamPlayback> AudioStreamRandomizer::instantiate_playback() {
switch (playback_mode) {
case PLAYBACK_RANDOM:
return instance_playback_random();
case PLAYBACK_RANDOM_NO_REPEATS:
return instance_playback_no_repeats();
case PLAYBACK_SEQUENTIAL:
return instance_playback_sequential();
default:
ERR_FAIL_V_MSG(nullptr, "Unhandled playback mode.");
}
}
String AudioStreamRandomizer::get_stream_name() const {
return "Randomizer";
}
double AudioStreamRandomizer::get_length() const {
return 0;
}
bool AudioStreamRandomizer::is_monophonic() const {
for (const PoolEntry &entry : audio_stream_pool) {
if (entry.stream.is_valid() && entry.stream->is_monophonic()) {
return true;
}
}
return false;
}
bool AudioStreamRandomizer::_get(const StringName &p_name, Variant &r_ret) const {
if (AudioStream::_get(p_name, r_ret)) {
return true;
}
Vector<String> components = String(p_name).split("/", true, 2);
if (components.size() == 2 && components[0].begins_with("stream_") && components[0].trim_prefix("stream_").is_valid_int()) {
int index = components[0].trim_prefix("stream_").to_int();
if (index < 0 || index >= (int)audio_stream_pool.size()) {
return false;
}
if (components[1] == "stream") {
r_ret = get_stream(index);
return true;
} else if (components[1] == "weight") {
r_ret = get_stream_probability_weight(index);
return true;
} else {
return false;
}
}
return false;
}
bool AudioStreamRandomizer::_set(const StringName &p_name, const Variant &p_value) {
if (AudioStream::_set(p_name, p_value)) {
return true;
}
Vector<String> components = String(p_name).split("/", true, 2);
if (components.size() == 2 && components[0].begins_with("stream_") && components[0].trim_prefix("stream_").is_valid_int()) {
int index = components[0].trim_prefix("stream_").to_int();
if (index < 0 || index >= (int)audio_stream_pool.size()) {
return false;
}
if (components[1] == "stream") {
set_stream(index, p_value);
return true;
} else if (components[1] == "weight") {
set_stream_probability_weight(index, p_value);
return true;
} else {
return false;
}
}
return false;
}
void AudioStreamRandomizer::_get_property_list(List<PropertyInfo> *p_list) const {
AudioStream::_get_property_list(p_list); // Define the trivial scalar properties.
p_list->push_back(PropertyInfo(Variant::NIL, "Streams", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_GROUP));
for (int i = 0; i < audio_stream_pool.size(); i++) {
p_list->push_back(PropertyInfo(Variant::OBJECT, vformat("stream_%d/stream", i), PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"));
p_list->push_back(PropertyInfo(Variant::FLOAT, vformat("stream_%d/weight", i), PROPERTY_HINT_RANGE, "0,100,0.001,or_greater"));
}
}
void AudioStreamRandomizer::_bind_methods() {
ClassDB::bind_method(D_METHOD("add_stream", "index"), &AudioStreamRandomizer::add_stream);
ClassDB::bind_method(D_METHOD("move_stream", "index_from", "index_to"), &AudioStreamRandomizer::move_stream);
ClassDB::bind_method(D_METHOD("remove_stream", "index"), &AudioStreamRandomizer::remove_stream);
ClassDB::bind_method(D_METHOD("set_stream", "index", "stream"), &AudioStreamRandomizer::set_stream);
ClassDB::bind_method(D_METHOD("get_stream", "index"), &AudioStreamRandomizer::get_stream);
ClassDB::bind_method(D_METHOD("set_stream_probability_weight", "index", "weight"), &AudioStreamRandomizer::set_stream_probability_weight);
ClassDB::bind_method(D_METHOD("get_stream_probability_weight", "index"), &AudioStreamRandomizer::get_stream_probability_weight);
ClassDB::bind_method(D_METHOD("set_streams_count", "count"), &AudioStreamRandomizer::set_streams_count);
ClassDB::bind_method(D_METHOD("get_streams_count"), &AudioStreamRandomizer::get_streams_count);
ClassDB::bind_method(D_METHOD("set_random_pitch", "scale"), &AudioStreamRandomizer::set_random_pitch);
ClassDB::bind_method(D_METHOD("get_random_pitch"), &AudioStreamRandomizer::get_random_pitch);
ClassDB::bind_method(D_METHOD("set_random_volume_offset_db", "db_offset"), &AudioStreamRandomizer::set_random_volume_offset_db);
ClassDB::bind_method(D_METHOD("get_random_volume_offset_db"), &AudioStreamRandomizer::get_random_volume_offset_db);
ClassDB::bind_method(D_METHOD("set_playback_mode", "mode"), &AudioStreamRandomizer::set_playback_mode);
ClassDB::bind_method(D_METHOD("get_playback_mode"), &AudioStreamRandomizer::get_playback_mode);
ADD_ARRAY("streams", "stream_");
ADD_PROPERTY(PropertyInfo(Variant::INT, "streams_count", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NO_EDITOR), "set_streams_count", "get_streams_count");
ADD_PROPERTY(PropertyInfo(Variant::INT, "playback_mode", PROPERTY_HINT_ENUM, "Random (Avoid Repeats),Random,Sequential"), "set_playback_mode", "get_playback_mode");
ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "random_pitch", PROPERTY_HINT_RANGE, "1,16,0.01"), "set_random_pitch", "get_random_pitch");
ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "random_volume_offset_db", PROPERTY_HINT_RANGE, "0,40,0.01,suffix:dB"), "set_random_volume_offset_db", "get_random_volume_offset_db");
BIND_ENUM_CONSTANT(PLAYBACK_RANDOM_NO_REPEATS);
BIND_ENUM_CONSTANT(PLAYBACK_RANDOM);
BIND_ENUM_CONSTANT(PLAYBACK_SEQUENTIAL);
}
AudioStreamRandomizer::AudioStreamRandomizer() {}
void AudioStreamPlaybackRandomizer::start(double p_from_pos) {
playing = playback;
{
float range_from = 1.0 / randomizer->random_pitch_scale;
float range_to = randomizer->random_pitch_scale;
pitch_scale = range_from + Math::randf() * (range_to - range_from);
}
{
float range_from = -randomizer->random_volume_offset_db;
float range_to = randomizer->random_volume_offset_db;
float volume_offset_db = range_from + Math::randf() * (range_to - range_from);
volume_scale = Math::db_to_linear(volume_offset_db);
}
if (playing.is_valid()) {
playing->start(p_from_pos);
}
}
void AudioStreamPlaybackRandomizer::stop() {
if (playing.is_valid()) {
playing->stop();
}
}
bool AudioStreamPlaybackRandomizer::is_playing() const {
if (playing.is_valid()) {
return playing->is_playing();
}
return false;
}
int AudioStreamPlaybackRandomizer::get_loop_count() const {
if (playing.is_valid()) {
return playing->get_loop_count();
}
return 0;
}
double AudioStreamPlaybackRandomizer::get_playback_position() const {
if (playing.is_valid()) {
return playing->get_playback_position();
}
return 0;
}
void AudioStreamPlaybackRandomizer::seek(double p_time) {
if (playing.is_valid()) {
playing->seek(p_time);
}
}
void AudioStreamPlaybackRandomizer::tag_used_streams() {
Ref<AudioStreamPlayback> p = playing; // Thread safety
if (p.is_valid()) {
p->tag_used_streams();
}
randomizer->tag_used(0);
}
int AudioStreamPlaybackRandomizer::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
if (playing.is_valid()) {
return playing->mix(p_buffer, p_rate_scale * pitch_scale, p_frames);
} else {
for (int i = 0; i < p_frames; i++) {
p_buffer[i] = AudioFrame(0, 0);
}
return p_frames;
}
}
AudioStreamPlaybackRandomizer::~AudioStreamPlaybackRandomizer() {
randomizer->playbacks.erase(this);
}
/////////////////////////////////////////////