33b5c57199
- Renames PackedIntArray to PackedInt32Array. - Renames PackedFloatArray to PackedFloat32Array. - Adds PackedInt64Array and PackedFloat64Array. - Renames Variant::REAL to Variant::FLOAT for consistency. Packed arrays are for storing large amount of data and creating stuff like meshes, buffers. textures, etc. Forcing them to be 64 is a huge waste of memory. That said, many users requested the ability to have 64 bits packed arrays for their games, so this is just an optional added type. For Variant, the float datatype is always 64 bits, and exposed as `float`. We still have `real_t` which is the datatype that can change from 32 to 64 bits depending on a compile flag (not entirely working right now, but that's the idea). It affects math related datatypes and code only. Neither Variant nor PackedArray make use of real_t, which is only intended for math precision, so the term is removed from there to keep only float.
370 lines
11 KiB
C++
370 lines
11 KiB
C++
/*************************************************************************/
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/* audio_stream.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2020 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2020 Godot Engine contributors (cf. AUTHORS.md). */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "audio_stream.h"
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#include "core/os/os.h"
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#include "core/project_settings.h"
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//////////////////////////////
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void AudioStreamPlaybackResampled::_begin_resample() {
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//clear cubic interpolation history
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internal_buffer[0] = AudioFrame(0.0, 0.0);
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internal_buffer[1] = AudioFrame(0.0, 0.0);
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internal_buffer[2] = AudioFrame(0.0, 0.0);
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internal_buffer[3] = AudioFrame(0.0, 0.0);
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//mix buffer
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_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
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mix_offset = 0;
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}
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void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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float target_rate = AudioServer::get_singleton()->get_mix_rate();
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float global_rate_scale = AudioServer::get_singleton()->get_global_rate_scale();
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uint64_t mix_increment = uint64_t(((get_stream_sampling_rate() * p_rate_scale) / double(target_rate * global_rate_scale)) * double(FP_LEN));
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for (int i = 0; i < p_frames; i++) {
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uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS);
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//standard cubic interpolation (great quality/performance ratio)
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//this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory.
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float mu = (mix_offset & FP_MASK) / float(FP_LEN);
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AudioFrame y0 = internal_buffer[idx - 3];
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AudioFrame y1 = internal_buffer[idx - 2];
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AudioFrame y2 = internal_buffer[idx - 1];
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AudioFrame y3 = internal_buffer[idx - 0];
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float mu2 = mu * mu;
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AudioFrame a0 = y3 - y2 - y0 + y1;
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AudioFrame a1 = y0 - y1 - a0;
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AudioFrame a2 = y2 - y0;
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AudioFrame a3 = y1;
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p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3);
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mix_offset += mix_increment;
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while ((mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN) {
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internal_buffer[0] = internal_buffer[INTERNAL_BUFFER_LEN + 0];
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internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1];
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internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2];
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internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3];
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if (is_playing()) {
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_mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN);
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} else {
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//fill with silence, not playing
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for (int j = 0; j < INTERNAL_BUFFER_LEN; ++j) {
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internal_buffer[j + 4] = AudioFrame(0, 0);
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}
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}
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mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS);
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}
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}
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}
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////////////////////////////////
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void AudioStream::_bind_methods() {
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ClassDB::bind_method(D_METHOD("get_length"), &AudioStream::get_length);
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}
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////////////////////////////////
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Ref<AudioStreamPlayback> AudioStreamMicrophone::instance_playback() {
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Ref<AudioStreamPlaybackMicrophone> playback;
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playback.instance();
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playbacks.insert(playback.ptr());
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playback->microphone = Ref<AudioStreamMicrophone>((AudioStreamMicrophone *)this);
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playback->active = false;
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return playback;
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}
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String AudioStreamMicrophone::get_stream_name() const {
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//if (audio_stream.is_valid()) {
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//return "Random: " + audio_stream->get_name();
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//}
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return "Microphone";
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}
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float AudioStreamMicrophone::get_length() const {
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return 0;
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}
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void AudioStreamMicrophone::_bind_methods() {
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}
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AudioStreamMicrophone::AudioStreamMicrophone() {
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}
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void AudioStreamPlaybackMicrophone::_mix_internal(AudioFrame *p_buffer, int p_frames) {
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AudioDriver::get_singleton()->lock();
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Vector<int32_t> buf = AudioDriver::get_singleton()->get_input_buffer();
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unsigned int input_size = AudioDriver::get_singleton()->get_input_size();
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int mix_rate = AudioDriver::get_singleton()->get_mix_rate();
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unsigned int playback_delay = MIN(((50 * mix_rate) / 1000) * 2, buf.size() >> 1);
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#ifdef DEBUG_ENABLED
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unsigned int input_position = AudioDriver::get_singleton()->get_input_position();
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#endif
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if (playback_delay > input_size) {
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for (int i = 0; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0.0f, 0.0f);
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}
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input_ofs = 0;
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} else {
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for (int i = 0; i < p_frames; i++) {
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if (input_size > input_ofs && (int)input_ofs < buf.size()) {
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float l = (buf[input_ofs++] >> 16) / 32768.f;
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if ((int)input_ofs >= buf.size()) {
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input_ofs = 0;
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}
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float r = (buf[input_ofs++] >> 16) / 32768.f;
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if ((int)input_ofs >= buf.size()) {
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input_ofs = 0;
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}
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p_buffer[i] = AudioFrame(l, r);
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} else {
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p_buffer[i] = AudioFrame(0.0f, 0.0f);
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}
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}
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}
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#ifdef DEBUG_ENABLED
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if (input_ofs > input_position && (int)(input_ofs - input_position) < (p_frames * 2)) {
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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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}
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#endif
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AudioDriver::get_singleton()->unlock();
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}
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void AudioStreamPlaybackMicrophone::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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AudioStreamPlaybackResampled::mix(p_buffer, p_rate_scale, p_frames);
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}
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float AudioStreamPlaybackMicrophone::get_stream_sampling_rate() {
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return AudioDriver::get_singleton()->get_mix_rate();
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}
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void AudioStreamPlaybackMicrophone::start(float p_from_pos) {
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if (active) {
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return;
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}
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if (!GLOBAL_GET("audio/enable_audio_input")) {
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WARN_PRINT("Need to enable Project settings > Audio > Enable Audio Input option to use capturing.");
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return;
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}
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input_ofs = 0;
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if (AudioDriver::get_singleton()->capture_start() == OK) {
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active = true;
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_begin_resample();
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}
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}
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void AudioStreamPlaybackMicrophone::stop() {
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if (active) {
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AudioDriver::get_singleton()->capture_stop();
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active = false;
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}
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}
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bool AudioStreamPlaybackMicrophone::is_playing() const {
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return active;
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}
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int AudioStreamPlaybackMicrophone::get_loop_count() const {
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return 0;
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}
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float AudioStreamPlaybackMicrophone::get_playback_position() const {
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return 0;
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}
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void AudioStreamPlaybackMicrophone::seek(float p_time) {
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// Can't seek a microphone input
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}
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AudioStreamPlaybackMicrophone::~AudioStreamPlaybackMicrophone() {
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microphone->playbacks.erase(this);
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stop();
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}
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AudioStreamPlaybackMicrophone::AudioStreamPlaybackMicrophone() {
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}
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////////////////////////////////
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void AudioStreamRandomPitch::set_audio_stream(const Ref<AudioStream> &p_audio_stream) {
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audio_stream = p_audio_stream;
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if (audio_stream.is_valid()) {
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for (Set<AudioStreamPlaybackRandomPitch *>::Element *E = playbacks.front(); E; E = E->next()) {
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E->get()->playback = audio_stream->instance_playback();
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}
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}
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}
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Ref<AudioStream> AudioStreamRandomPitch::get_audio_stream() const {
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return audio_stream;
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}
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void AudioStreamRandomPitch::set_random_pitch(float p_pitch) {
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if (p_pitch < 1)
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p_pitch = 1;
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random_pitch = p_pitch;
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}
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float AudioStreamRandomPitch::get_random_pitch() const {
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return random_pitch;
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}
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Ref<AudioStreamPlayback> AudioStreamRandomPitch::instance_playback() {
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Ref<AudioStreamPlaybackRandomPitch> playback;
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playback.instance();
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if (audio_stream.is_valid())
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playback->playback = audio_stream->instance_playback();
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playbacks.insert(playback.ptr());
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playback->random_pitch = Ref<AudioStreamRandomPitch>((AudioStreamRandomPitch *)this);
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return playback;
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}
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String AudioStreamRandomPitch::get_stream_name() const {
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if (audio_stream.is_valid()) {
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return "Random: " + audio_stream->get_name();
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}
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return "RandomPitch";
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}
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float AudioStreamRandomPitch::get_length() const {
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if (audio_stream.is_valid()) {
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return audio_stream->get_length();
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}
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return 0;
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}
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void AudioStreamRandomPitch::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_audio_stream", "stream"), &AudioStreamRandomPitch::set_audio_stream);
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ClassDB::bind_method(D_METHOD("get_audio_stream"), &AudioStreamRandomPitch::get_audio_stream);
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ClassDB::bind_method(D_METHOD("set_random_pitch", "scale"), &AudioStreamRandomPitch::set_random_pitch);
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ClassDB::bind_method(D_METHOD("get_random_pitch"), &AudioStreamRandomPitch::get_random_pitch);
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ADD_PROPERTY(PropertyInfo(Variant::OBJECT, "audio_stream", PROPERTY_HINT_RESOURCE_TYPE, "AudioStream"), "set_audio_stream", "get_audio_stream");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "random_pitch", PROPERTY_HINT_RANGE, "1,16,0.01"), "set_random_pitch", "get_random_pitch");
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}
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AudioStreamRandomPitch::AudioStreamRandomPitch() {
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random_pitch = 1.1;
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}
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void AudioStreamPlaybackRandomPitch::start(float p_from_pos) {
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playing = playback;
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float range_from = 1.0 / random_pitch->random_pitch;
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float range_to = random_pitch->random_pitch;
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pitch_scale = range_from + Math::randf() * (range_to - range_from);
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if (playing.is_valid()) {
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playing->start(p_from_pos);
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}
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}
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void AudioStreamPlaybackRandomPitch::stop() {
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if (playing.is_valid()) {
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playing->stop();
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;
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}
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}
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bool AudioStreamPlaybackRandomPitch::is_playing() const {
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if (playing.is_valid()) {
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return playing->is_playing();
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}
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return false;
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}
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int AudioStreamPlaybackRandomPitch::get_loop_count() const {
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if (playing.is_valid()) {
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return playing->get_loop_count();
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}
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return 0;
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}
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float AudioStreamPlaybackRandomPitch::get_playback_position() const {
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if (playing.is_valid()) {
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return playing->get_playback_position();
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}
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return 0;
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}
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void AudioStreamPlaybackRandomPitch::seek(float p_time) {
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if (playing.is_valid()) {
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playing->seek(p_time);
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}
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}
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void AudioStreamPlaybackRandomPitch::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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if (playing.is_valid()) {
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playing->mix(p_buffer, p_rate_scale * pitch_scale, p_frames);
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} else {
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for (int i = 0; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0, 0);
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}
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}
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}
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AudioStreamPlaybackRandomPitch::~AudioStreamPlaybackRandomPitch() {
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random_pitch->playbacks.erase(this);
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}
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