virtualx-engine/servers/audio/audio_stream.cpp

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/**************************************************************************/
/* audio_stream.cpp */
/**************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/**************************************************************************/
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* 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. */
/**************************************************************************/
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#include "audio_stream.h"
#include "core/config/project_settings.h"
#include "core/os/os.h"
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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 {
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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) {
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GDVIRTUAL_CALL(_seek, p_time);
}
int AudioStreamPlayback::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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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);
}
Implement audio stream playback parameters. Implements a way for audio stream playback to be configured via parameters directly in the edited AudioStreamPlayer[2D/3D]. Currently, configuring the playback stream is not possible (or is sometimes hacky as the user has to obtain the currently played stream, which is not always immediately available). This PR only implements this new feature to control looping in stream playback instances (a commonly requested feature, which was lost in the transition from Godot 2 to Godot 3). But the idea is that it can do a lot more: * If effects are bundled to the stream, control per playback instance parameters such as cutoff or resoance, or any other exposed effect parameter per playback instance. * For the upcoming interactive music PR (#64488), this exposes an easy way to change the active clip, which was not possible before. * For the upcoming parametrizable audio support (https://github.com/godotengine/godot-proposals/issues/3394) this allows editing and animating audio graph parameters. In any case, this PR is required to complete #64488. Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update doc/classes/AudioStream.xml Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com>
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void AudioStreamPlayback::set_parameter(const StringName &p_name, const Variant &p_value) {
GDVIRTUAL_CALL(_set_parameter, p_name, p_value);
}
Variant AudioStreamPlayback::get_parameter(const StringName &p_name) const {
Variant ret;
GDVIRTUAL_CALL(_get_parameter, p_name, ret);
return ret;
}
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);
Implement audio stream playback parameters. Implements a way for audio stream playback to be configured via parameters directly in the edited AudioStreamPlayer[2D/3D]. Currently, configuring the playback stream is not possible (or is sometimes hacky as the user has to obtain the currently played stream, which is not always immediately available). This PR only implements this new feature to control looping in stream playback instances (a commonly requested feature, which was lost in the transition from Godot 2 to Godot 3). But the idea is that it can do a lot more: * If effects are bundled to the stream, control per playback instance parameters such as cutoff or resoance, or any other exposed effect parameter per playback instance. * For the upcoming interactive music PR (#64488), this exposes an easy way to change the active clip, which was not possible before. * For the upcoming parametrizable audio support (https://github.com/godotengine/godot-proposals/issues/3394) this allows editing and animating audio graph parameters. In any case, this PR is required to complete #64488. Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update doc/classes/AudioStream.xml Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com>
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GDVIRTUAL_BIND(_set_parameter, "name", "value");
GDVIRTUAL_BIND(_get_parameter, "name");
}
//////////////////////////////
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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;
}
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int AudioStreamPlaybackResampled::_mix_internal(AudioFrame *p_buffer, int p_frames) {
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int ret = 0;
GDVIRTUAL_REQUIRED_CALL(_mix_resampled, p_buffer, p_frames, ret);
return ret;
}
float AudioStreamPlaybackResampled::get_stream_sampling_rate() {
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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();
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uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale * playback_speed_scale) / double(target_rate)) * double(FP_LEN));
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int mixed_frames_total = -1;
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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];
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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;
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AudioFrame a0 = 3 * y1 - 3 * y2 + y3 - y0;
AudioFrame a1 = 2 * y0 - 5 * y1 + 4 * y2 - y3;
AudioFrame a2 = y2 - y0;
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AudioFrame a3 = 2 * y1;
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p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3) / 2;
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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];
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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 {
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// The internal buffer does not contain the first frame of silence.
internal_buffer_end = -1;
}
mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
}
}
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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;
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GDVIRTUAL_CALL(_get_stream_name, ret);
return ret;
}
double AudioStream::get_length() const {
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double ret = 0;
GDVIRTUAL_CALL(_get_length, ret);
return ret;
}
bool AudioStream::is_monophonic() const {
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bool ret = true;
GDVIRTUAL_CALL(_is_monophonic, ret);
return ret;
}
double AudioStream::get_bpm() const {
double ret = 0;
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GDVIRTUAL_CALL(_get_bpm, ret);
return ret;
}
bool AudioStream::has_loop() const {
bool ret = 0;
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GDVIRTUAL_CALL(_has_loop, ret);
return ret;
}
int AudioStream::get_bar_beats() const {
int ret = 0;
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GDVIRTUAL_CALL(_get_bar_beats, ret);
return ret;
}
int AudioStream::get_beat_count() const {
int ret = 0;
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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];
}
Implement audio stream playback parameters. Implements a way for audio stream playback to be configured via parameters directly in the edited AudioStreamPlayer[2D/3D]. Currently, configuring the playback stream is not possible (or is sometimes hacky as the user has to obtain the currently played stream, which is not always immediately available). This PR only implements this new feature to control looping in stream playback instances (a commonly requested feature, which was lost in the transition from Godot 2 to Godot 3). But the idea is that it can do a lot more: * If effects are bundled to the stream, control per playback instance parameters such as cutoff or resoance, or any other exposed effect parameter per playback instance. * For the upcoming interactive music PR (#64488), this exposes an easy way to change the active clip, which was not possible before. * For the upcoming parametrizable audio support (https://github.com/godotengine/godot-proposals/issues/3394) this allows editing and animating audio graph parameters. In any case, this PR is required to complete #64488. Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update doc/classes/AudioStream.xml Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com>
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void AudioStream::get_parameter_list(List<Parameter> *r_parameters) {
TypedArray<Dictionary> ret;
GDVIRTUAL_CALL(_get_parameter_list, ret);
for (int i = 0; i < ret.size(); i++) {
Dictionary d = ret[i];
ERR_CONTINUE(!d.has("default_value"));
r_parameters->push_back(Parameter(PropertyInfo::from_dict(d), d["default_value"]));
}
}
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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)
Implement audio stream playback parameters. Implements a way for audio stream playback to be configured via parameters directly in the edited AudioStreamPlayer[2D/3D]. Currently, configuring the playback stream is not possible (or is sometimes hacky as the user has to obtain the currently played stream, which is not always immediately available). This PR only implements this new feature to control looping in stream playback instances (a commonly requested feature, which was lost in the transition from Godot 2 to Godot 3). But the idea is that it can do a lot more: * If effects are bundled to the stream, control per playback instance parameters such as cutoff or resoance, or any other exposed effect parameter per playback instance. * For the upcoming interactive music PR (#64488), this exposes an easy way to change the active clip, which was not possible before. * For the upcoming parametrizable audio support (https://github.com/godotengine/godot-proposals/issues/3394) this allows editing and animating audio graph parameters. In any case, this PR is required to complete #64488. Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/minimp3/audio_stream_mp3.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update modules/vorbis/audio_stream_ogg_vorbis.h Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com> Update doc/classes/AudioStream.xml Co-authored-by: A Thousand Ships <96648715+AThousandShips@users.noreply.github.com>
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GDVIRTUAL_BIND(_get_parameter_list)
ADD_SIGNAL(MethodInfo("parameter_list_changed"));
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}
////////////////////////////////
Ref<AudioStreamPlayback> AudioStreamMicrophone::instantiate_playback() {
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Ref<AudioStreamPlaybackMicrophone> playback;
playback.instantiate();
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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 {
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return 0;
}
bool AudioStreamMicrophone::is_monophonic() const {
return true;
}
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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);
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#ifdef DEBUG_ENABLED
unsigned int input_position = AudioDriver::get_singleton()->get_input_position();
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#endif
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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);
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}
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);
}
}
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}
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#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));
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}
#endif
AudioDriver::get_singleton()->unlock();
return mixed_frames;
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}
int AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
return AudioStreamPlaybackResampled::mix(p_buffer, p_rate_scale, p_frames);
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}
float AudioStreamPlaybackMicrophone::get_stream_sampling_rate() {
return AudioDriver::get_singleton()->get_mix_rate();
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}
void AudioStreamPlaybackMicrophone::start(double p_from_pos) {
if (active) {
return;
}
if (!GLOBAL_GET("audio/driver/enable_input")) {
WARN_PRINT("You must enable the project setting \"audio/driver/enable_input\" to use audio capture.");
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return;
}
input_ofs = 0;
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if (AudioDriver::get_singleton()->input_start() == OK) {
active = true;
begin_resample();
}
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}
void AudioStreamPlaybackMicrophone::stop() {
if (active) {
AudioDriver::get_singleton()->input_stop();
active = false;
}
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}
bool AudioStreamPlaybackMicrophone::is_playing() const {
return active;
}
int AudioStreamPlaybackMicrophone::get_loop_count() const {
return 0;
}
double AudioStreamPlaybackMicrophone::get_playback_position() const {
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return 0;
}
void AudioStreamPlaybackMicrophone::seek(double p_time) {
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// Can't seek a microphone input
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}
void AudioStreamPlaybackMicrophone::tag_used_streams() {
microphone->tag_used(0);
}
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AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() {
microphone->playbacks.erase(this);
stop();
}
AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() {
}
////////////////////////////////
void AudioStreamRandomizer::add_stream(int p_index, Ref<AudioStream> p_stream, float p_weight) {
if (p_index < 0) {
p_index = audio_stream_pool.size();
}
ERR_FAIL_COND(p_index > audio_stream_pool.size());
PoolEntry entry{ p_stream, p_weight };
audio_stream_pool.insert(p_index, entry);
emit_signal(SNAME("changed"));
notify_property_list_changed();
}
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// 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) {
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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) {
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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) {
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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 {
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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) {
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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 {
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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.has(entry.stream)) {
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.");
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}
}
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String AudioStreamRandomizer::get_stream_name() const {
return "Randomizer";
}
double AudioStreamRandomizer::get_length() const {
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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;
}
void AudioStreamRandomizer::_bind_methods() {
ClassDB::bind_method(D_METHOD("add_stream", "index", "stream", "weight"), &AudioStreamRandomizer::add_stream, DEFVAL(1.0));
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_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");
ADD_ARRAY("streams", "stream_");
ADD_PROPERTY(PropertyInfo(Variant::INT, "streams_count", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NO_EDITOR), "set_streams_count", "get_streams_count");
BIND_ENUM_CONSTANT(PLAYBACK_RANDOM_NO_REPEATS);
BIND_ENUM_CONSTANT(PLAYBACK_RANDOM);
BIND_ENUM_CONSTANT(PLAYBACK_SEQUENTIAL);
PoolEntry defaults;
base_property_helper.set_prefix("stream_");
base_property_helper.set_array_length_getter(&AudioStreamRandomizer::get_streams_count);
base_property_helper.register_property(PropertyInfo(Variant::OBJECT, "stream", PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"), defaults.stream, &AudioStreamRandomizer::set_stream, &AudioStreamRandomizer::get_stream);
base_property_helper.register_property(PropertyInfo(Variant::FLOAT, "weight", PROPERTY_HINT_RANGE, "0,100,0.001,or_greater"), defaults.weight, &AudioStreamRandomizer::set_stream_probability_weight, &AudioStreamRandomizer::get_stream_probability_weight);
}
AudioStreamRandomizer::AudioStreamRandomizer() {
property_helper.setup_for_instance(base_property_helper, this);
}
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()) {
int mixed_samples = playing->mix(p_buffer, p_rate_scale * pitch_scale, p_frames);
for (int samp = 0; samp < mixed_samples; samp++) {
p_buffer[samp] *= volume_scale;
}
return mixed_samples;
} else {
for (int i = 0; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0, 0);
}
return p_frames;
}
}
AudioStreamPlaybackRandomizer::~AudioStreamPlaybackRandomizer() {
randomizer->playbacks.erase(this);
}
/////////////////////////////////////////////