virtualx-engine/servers/audio/audio_stream.cpp
Rémi Verschelde fe52458154
Update copyright statements to 2022
Happy new year to the wonderful Godot community!
2022-01-03 21:27:34 +01:00

491 lines
15 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(float 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;
if (GDVIRTUAL_CALL(_get_loop_count, ret)) {
return ret;
}
return 0;
}
float AudioStreamPlayback::get_playback_position() const {
float ret;
if (GDVIRTUAL_CALL(_get_playback_position, ret)) {
return ret;
}
ERR_FAIL_V_MSG(0, "AudioStreamPlayback::get_playback_position unimplemented!");
}
void AudioStreamPlayback::seek(float p_time) {
if (GDVIRTUAL_CALL(_seek, p_time)) {
return;
}
}
int AudioStreamPlayback::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
int ret;
if (GDVIRTUAL_CALL(_mix, p_buffer, p_rate_scale, p_frames, ret)) {
return ret;
}
WARN_PRINT_ONCE("AudioStreamPlayback::mix unimplemented!");
return 0;
}
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");
}
//////////////////////////////
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(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 = p_frames;
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];
if (idx <= internal_buffer_end && idx >= internal_buffer_end && mixed_frames_total == p_frames) {
// 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];
if (is_playing()) {
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;
}
} 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);
}
}
return mixed_frames_total;
}
////////////////////////////////
Ref<AudioStreamPlayback> AudioStream::instance_playback() {
Ref<AudioStreamPlayback> ret;
if (GDVIRTUAL_CALL(_instance_playback, ret)) {
return ret;
}
ERR_FAIL_V_MSG(Ref<AudioStreamPlayback>(), "Method must be implemented!");
}
String AudioStream::get_stream_name() const {
String ret;
if (GDVIRTUAL_CALL(_get_stream_name, ret)) {
return ret;
}
return String();
}
float AudioStream::get_length() const {
float ret;
if (GDVIRTUAL_CALL(_get_length, ret)) {
return ret;
}
return 0;
}
bool AudioStream::is_monophonic() const {
bool ret;
if (GDVIRTUAL_CALL(_is_monophonic, ret)) {
return ret;
}
return true;
}
void AudioStream::_bind_methods() {
ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length);
ClassDB::bind_method(D_METHOD("is_monophonic"), &AudioStream::is_monophonic);
GDVIRTUAL_BIND(_instance_playback);
GDVIRTUAL_BIND(_get_stream_name);
GDVIRTUAL_BIND(_get_length);
GDVIRTUAL_BIND(_is_monophonic);
}
////////////////////////////////
Ref<AudioStreamPlayback> AudioStreamMicrophone::instance_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";
}
float 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(float 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;
}
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.instantiate();
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;
}
bool AudioStreamRandomPitch::is_monophonic() const {
if (audio_stream.is_valid()) {
return audio_stream->is_monophonic();
}
return true; // It doesn't really matter what we return here, but no sense instancing a many playbacks of a null stream.
}
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::FLOAT, "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);
}
}
int AudioStreamPlaybackRandomPitch::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;
}
}
AudioStreamPlaybackRandomPitch::~AudioStreamPlaybackRandomPitch() {
random_pitch->playbacks.erase(this);
}
/////////////////////////////////////////////