mirror of
https://github.com/Relintai/pandemonium_engine.git
synced 2024-12-26 21:57:16 +01:00
359 lines
11 KiB
C++
359 lines
11 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/os/os.h"
|
|
#include "core/project_settings.h"
|
|
|
|
//////////////////////////////
|
|
|
|
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;
|
|
}
|
|
|
|
void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
|
|
float target_rate = AudioServer::get_singleton()->get_mix_rate();
|
|
float global_rate_scale = AudioServer::get_singleton()->get_global_rate_scale();
|
|
|
|
uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale) / double(target_rate * global_rate_scale)) * double(FP_LEN));
|
|
|
|
for (int 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];
|
|
|
|
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];
|
|
if (is_playing()) {
|
|
_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
|
|
} else {
|
|
//fill with silence, not playing
|
|
for (int j = 0; j < INTERNAL_BUFFER_LEN; ++j) {
|
|
internal_buffer[j + 4] = AudioFrame(0, 0);
|
|
}
|
|
}
|
|
mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
|
|
}
|
|
}
|
|
}
|
|
|
|
////////////////////////////////
|
|
|
|
void AudioStream::_bind_methods() {
|
|
ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length);
|
|
}
|
|
|
|
////////////////////////////////
|
|
|
|
Ref<AudioStreamPlayback> AudioStreamMicrophone::instance_playback() {
|
|
Ref<AudioStreamPlaybackMicrophone> playback;
|
|
playback.instance();
|
|
|
|
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";
|
|
}
|
|
|
|
float AudioStreamMicrophone::get_length() const {
|
|
return 0;
|
|
}
|
|
|
|
void AudioStreamMicrophone::_bind_methods() {
|
|
}
|
|
|
|
AudioStreamMicrophone::AudioStreamMicrophone() {
|
|
}
|
|
|
|
void 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
|
|
|
|
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 {
|
|
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();
|
|
}
|
|
|
|
void AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
|
|
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(float p_from_pos) {
|
|
if (active) {
|
|
return;
|
|
}
|
|
|
|
if (!GLOBAL_GET("audio/enable_audio_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;
|
|
}
|
|
|
|
float AudioStreamPlaybackMicrophone::get_playback_position() const {
|
|
return 0;
|
|
}
|
|
|
|
void AudioStreamPlaybackMicrophone::seek(float p_time) {
|
|
// Can't seek a microphone input
|
|
}
|
|
|
|
AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() {
|
|
microphone->playbacks.erase(this);
|
|
stop();
|
|
}
|
|
|
|
AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() {
|
|
}
|
|
|
|
////////////////////////////////
|
|
|
|
void AudioStreamRandomPitch::set_audio_stream(const Ref<AudioStream> &p_audio_stream) {
|
|
audio_stream = p_audio_stream;
|
|
if (audio_stream.is_valid()) {
|
|
for (Set<AudioStreamPlaybackRandomPitch *>::Element *E = playbacks.front(); E; E = E->next()) {
|
|
E->get()->playback = audio_stream->instance_playback();
|
|
}
|
|
}
|
|
}
|
|
|
|
Ref<AudioStream> AudioStreamRandomPitch::get_audio_stream() const {
|
|
return audio_stream;
|
|
}
|
|
|
|
void AudioStreamRandomPitch::set_random_pitch(float p_pitch) {
|
|
if (p_pitch < 1) {
|
|
p_pitch = 1;
|
|
}
|
|
random_pitch = p_pitch;
|
|
}
|
|
|
|
float AudioStreamRandomPitch::get_random_pitch() const {
|
|
return random_pitch;
|
|
}
|
|
|
|
Ref<AudioStreamPlayback> AudioStreamRandomPitch::instance_playback() {
|
|
Ref<AudioStreamPlaybackRandomPitch> playback;
|
|
playback.instance();
|
|
if (audio_stream.is_valid()) {
|
|
playback->playback = audio_stream->instance_playback();
|
|
}
|
|
|
|
playbacks.insert(playback.ptr());
|
|
playback->random_pitch = Ref<AudioStreamRandomPitch>((AudioStreamRandomPitch *)this);
|
|
return playback;
|
|
}
|
|
|
|
String AudioStreamRandomPitch::get_stream_name() const {
|
|
if (audio_stream.is_valid()) {
|
|
return "Random: " + audio_stream->get_name();
|
|
}
|
|
return "RandomPitch";
|
|
}
|
|
|
|
float AudioStreamRandomPitch::get_length() const {
|
|
if (audio_stream.is_valid()) {
|
|
return audio_stream->get_length();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
void AudioStreamRandomPitch::_bind_methods() {
|
|
ClassDB::bind_method(D_METHOD("set_audio_stream", "stream"), &AudioStreamRandomPitch::set_audio_stream);
|
|
ClassDB::bind_method(D_METHOD("get_audio_stream"), &AudioStreamRandomPitch::get_audio_stream);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_random_pitch", "scale"), &AudioStreamRandomPitch::set_random_pitch);
|
|
ClassDB::bind_method(D_METHOD("get_random_pitch"), &AudioStreamRandomPitch::get_random_pitch);
|
|
|
|
ADD_PROPERTY(PropertyInfo(Variant::OBJECT, "audio_stream", PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"), "set_audio_stream", "get_audio_stream");
|
|
ADD_PROPERTY(PropertyInfo(Variant::REAL, "random_pitch", PROPERTY_HINT_RANGE, "1,16,0.01"), "set_random_pitch", "get_random_pitch");
|
|
}
|
|
|
|
AudioStreamRandomPitch::AudioStreamRandomPitch() {
|
|
random_pitch = 1.1;
|
|
}
|
|
|
|
void AudioStreamPlaybackRandomPitch::start(float p_from_pos) {
|
|
playing = playback;
|
|
float range_from = 1.0 / random_pitch->random_pitch;
|
|
float range_to = random_pitch->random_pitch;
|
|
|
|
pitch_scale = range_from + Math::randf() * (range_to - range_from);
|
|
|
|
if (playing.is_valid()) {
|
|
playing->start(p_from_pos);
|
|
}
|
|
}
|
|
|
|
void AudioStreamPlaybackRandomPitch::stop() {
|
|
if (playing.is_valid()) {
|
|
playing->stop();
|
|
;
|
|
}
|
|
}
|
|
bool AudioStreamPlaybackRandomPitch::is_playing() const {
|
|
if (playing.is_valid()) {
|
|
return playing->is_playing();
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
int AudioStreamPlaybackRandomPitch::get_loop_count() const {
|
|
if (playing.is_valid()) {
|
|
return playing->get_loop_count();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
float AudioStreamPlaybackRandomPitch::get_playback_position() const {
|
|
if (playing.is_valid()) {
|
|
return playing->get_playback_position();
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
void AudioStreamPlaybackRandomPitch::seek(float p_time) {
|
|
if (playing.is_valid()) {
|
|
playing->seek(p_time);
|
|
}
|
|
}
|
|
|
|
void AudioStreamPlaybackRandomPitch::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
|
|
if (playing.is_valid()) {
|
|
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);
|
|
}
|
|
}
|
|
}
|
|
|
|
AudioStreamPlaybackRandomPitch::~AudioStreamPlaybackRandomPitch() {
|
|
random_pitch->playbacks.erase(this);
|
|
}
|