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d95794ec8a
As many open source projects have started doing it, we're removing the current year from the copyright notice, so that we don't need to bump it every year. It seems like only the first year of publication is technically relevant for copyright notices, and even that seems to be something that many companies stopped listing altogether (in a version controlled codebase, the commits are a much better source of date of publication than a hardcoded copyright statement). We also now list Godot Engine contributors first as we're collectively the current maintainers of the project, and we clarify that the "exclusive" copyright of the co-founders covers the timespan before opensourcing (their further contributions are included as part of Godot Engine contributors). Also fixed "cf." Frenchism - it's meant as "refer to / see".
668 lines
19 KiB
C++
668 lines
19 KiB
C++
/**************************************************************************/
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/* audio_stream_wav.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) 2014-present Godot Engine contributors (see AUTHORS.md). */
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/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
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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_wav.h"
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#include "core/io/file_access.h"
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#include "core/io/marshalls.h"
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void AudioStreamPlaybackWAV::start(double p_from_pos) {
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if (base->format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
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//no seeking in IMA_ADPCM
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for (int i = 0; i < 2; i++) {
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ima_adpcm[i].step_index = 0;
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ima_adpcm[i].predictor = 0;
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ima_adpcm[i].loop_step_index = 0;
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ima_adpcm[i].loop_predictor = 0;
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ima_adpcm[i].last_nibble = -1;
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ima_adpcm[i].loop_pos = 0x7FFFFFFF;
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ima_adpcm[i].window_ofs = 0;
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}
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offset = 0;
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} else {
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seek(p_from_pos);
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}
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sign = 1;
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active = true;
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}
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void AudioStreamPlaybackWAV::stop() {
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active = false;
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}
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bool AudioStreamPlaybackWAV::is_playing() const {
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return active;
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}
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int AudioStreamPlaybackWAV::get_loop_count() const {
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return 0;
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}
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double AudioStreamPlaybackWAV::get_playback_position() const {
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return float(offset >> MIX_FRAC_BITS) / base->mix_rate;
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}
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void AudioStreamPlaybackWAV::seek(double p_time) {
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if (base->format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
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return; //no seeking in ima-adpcm
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}
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double max = base->get_length();
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if (p_time < 0) {
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p_time = 0;
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} else if (p_time >= max) {
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p_time = max - 0.001;
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}
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offset = uint64_t(p_time * base->mix_rate) << MIX_FRAC_BITS;
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}
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template <class Depth, bool is_stereo, bool is_ima_adpcm>
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void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &offset, int32_t &increment, uint32_t amount, IMA_ADPCM_State *ima_adpcm) {
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// this function will be compiled branchless by any decent compiler
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int32_t final, final_r, next, next_r;
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while (amount) {
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amount--;
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int64_t pos = offset >> MIX_FRAC_BITS;
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if (is_stereo && !is_ima_adpcm) {
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pos <<= 1;
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}
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if (is_ima_adpcm) {
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int64_t sample_pos = pos + ima_adpcm[0].window_ofs;
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while (sample_pos > ima_adpcm[0].last_nibble) {
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static const int16_t _ima_adpcm_step_table[89] = {
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7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
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19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
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50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
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130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
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337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
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876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
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2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
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5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
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};
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static const int8_t _ima_adpcm_index_table[16] = {
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-1, -1, -1, -1, 2, 4, 6, 8,
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-1, -1, -1, -1, 2, 4, 6, 8
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};
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for (int i = 0; i < (is_stereo ? 2 : 1); i++) {
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int16_t nibble, diff, step;
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ima_adpcm[i].last_nibble++;
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const uint8_t *src_ptr = (const uint8_t *)base->data;
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src_ptr += AudioStreamWAV::DATA_PAD;
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uint8_t nbb = src_ptr[(ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i];
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nibble = (ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF);
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step = _ima_adpcm_step_table[ima_adpcm[i].step_index];
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ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
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if (ima_adpcm[i].step_index < 0) {
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ima_adpcm[i].step_index = 0;
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}
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if (ima_adpcm[i].step_index > 88) {
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ima_adpcm[i].step_index = 88;
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}
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diff = step >> 3;
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if (nibble & 1) {
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diff += step >> 2;
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}
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if (nibble & 2) {
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diff += step >> 1;
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}
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if (nibble & 4) {
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diff += step;
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}
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if (nibble & 8) {
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diff = -diff;
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}
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ima_adpcm[i].predictor += diff;
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if (ima_adpcm[i].predictor < -0x8000) {
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ima_adpcm[i].predictor = -0x8000;
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} else if (ima_adpcm[i].predictor > 0x7FFF) {
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ima_adpcm[i].predictor = 0x7FFF;
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}
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/* store loop if there */
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if (ima_adpcm[i].last_nibble == ima_adpcm[i].loop_pos) {
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ima_adpcm[i].loop_step_index = ima_adpcm[i].step_index;
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ima_adpcm[i].loop_predictor = ima_adpcm[i].predictor;
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}
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//printf("%i - %i - pred %i\n",int(ima_adpcm[i].last_nibble),int(nibble),int(ima_adpcm[i].predictor));
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}
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}
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final = ima_adpcm[0].predictor;
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if (is_stereo) {
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final_r = ima_adpcm[1].predictor;
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}
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} else {
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final = p_src[pos];
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if (is_stereo) {
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final_r = p_src[pos + 1];
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}
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if constexpr (sizeof(Depth) == 1) { /* conditions will not exist anymore when compiled! */
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final <<= 8;
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if (is_stereo) {
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final_r <<= 8;
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}
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}
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if (is_stereo) {
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next = p_src[pos + 2];
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next_r = p_src[pos + 3];
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} else {
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next = p_src[pos + 1];
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}
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if constexpr (sizeof(Depth) == 1) {
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next <<= 8;
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if (is_stereo) {
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next_r <<= 8;
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}
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}
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int32_t frac = int64_t(offset & MIX_FRAC_MASK);
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final = final + ((next - final) * frac >> MIX_FRAC_BITS);
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if (is_stereo) {
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final_r = final_r + ((next_r - final_r) * frac >> MIX_FRAC_BITS);
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}
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}
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if (!is_stereo) {
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final_r = final; //copy to right channel if stereo
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}
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p_dst->l = final / 32767.0;
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p_dst->r = final_r / 32767.0;
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p_dst++;
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offset += increment;
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}
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}
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int AudioStreamPlaybackWAV::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) {
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if (!base->data || !active) {
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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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return 0;
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}
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int len = base->data_bytes;
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switch (base->format) {
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case AudioStreamWAV::FORMAT_8_BITS:
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len /= 1;
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break;
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case AudioStreamWAV::FORMAT_16_BITS:
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len /= 2;
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break;
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case AudioStreamWAV::FORMAT_IMA_ADPCM:
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len *= 2;
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break;
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}
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if (base->stereo) {
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len /= 2;
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}
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/* some 64-bit fixed point precaches */
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int64_t loop_begin_fp = ((int64_t)base->loop_begin << MIX_FRAC_BITS);
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int64_t loop_end_fp = ((int64_t)base->loop_end << MIX_FRAC_BITS);
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int64_t length_fp = ((int64_t)len << MIX_FRAC_BITS);
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int64_t begin_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_begin_fp : 0;
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int64_t end_limit = (base->loop_mode != AudioStreamWAV::LOOP_DISABLED) ? loop_end_fp : length_fp;
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bool is_stereo = base->stereo;
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int32_t todo = p_frames;
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if (base->loop_mode == AudioStreamWAV::LOOP_BACKWARD) {
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sign = -1;
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}
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float base_rate = AudioServer::get_singleton()->get_mix_rate();
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float srate = base->mix_rate;
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srate *= p_rate_scale;
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float playback_speed_scale = AudioServer::get_singleton()->get_playback_speed_scale();
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float fincrement = (srate * playback_speed_scale) / base_rate;
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int32_t increment = int32_t(MAX(fincrement * MIX_FRAC_LEN, 1));
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increment *= sign;
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//looping
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AudioStreamWAV::LoopMode loop_format = base->loop_mode;
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AudioStreamWAV::Format format = base->format;
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/* audio data */
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uint8_t *dataptr = (uint8_t *)base->data;
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const void *data = dataptr + AudioStreamWAV::DATA_PAD;
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AudioFrame *dst_buff = p_buffer;
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if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
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if (loop_format != AudioStreamWAV::LOOP_DISABLED) {
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ima_adpcm[0].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
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ima_adpcm[1].loop_pos = loop_begin_fp >> MIX_FRAC_BITS;
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loop_format = AudioStreamWAV::LOOP_FORWARD;
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}
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}
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while (todo > 0) {
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int64_t limit = 0;
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int32_t target = 0, aux = 0;
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/** LOOP CHECKING **/
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if (increment < 0) {
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/* going backwards */
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if (loop_format != AudioStreamWAV::LOOP_DISABLED && offset < loop_begin_fp) {
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/* loopstart reached */
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if (loop_format == AudioStreamWAV::LOOP_PINGPONG) {
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/* bounce ping pong */
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offset = loop_begin_fp + (loop_begin_fp - offset);
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increment = -increment;
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sign *= -1;
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} else {
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/* go to loop-end */
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offset = loop_end_fp - (loop_begin_fp - offset);
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}
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} else {
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/* check for sample not reaching beginning */
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if (offset < 0) {
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active = false;
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break;
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}
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}
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} else {
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/* going forward */
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if (loop_format != AudioStreamWAV::LOOP_DISABLED && offset >= loop_end_fp) {
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/* loopend reached */
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if (loop_format == AudioStreamWAV::LOOP_PINGPONG) {
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/* bounce ping pong */
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offset = loop_end_fp - (offset - loop_end_fp);
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increment = -increment;
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sign *= -1;
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} else {
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/* go to loop-begin */
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if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
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for (int i = 0; i < 2; i++) {
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ima_adpcm[i].step_index = ima_adpcm[i].loop_step_index;
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ima_adpcm[i].predictor = ima_adpcm[i].loop_predictor;
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ima_adpcm[i].last_nibble = loop_begin_fp >> MIX_FRAC_BITS;
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}
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offset = loop_begin_fp;
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} else {
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offset = loop_begin_fp + (offset - loop_end_fp);
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}
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}
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} else {
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/* no loop, check for end of sample */
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if (offset >= length_fp) {
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active = false;
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break;
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}
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}
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}
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/** MIXCOUNT COMPUTING **/
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/* next possible limit (looppoints or sample begin/end */
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limit = (increment < 0) ? begin_limit : end_limit;
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/* compute what is shorter, the todo or the limit? */
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aux = (limit - offset) / increment + 1;
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target = (aux < todo) ? aux : todo; /* mix target is the shorter buffer */
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/* check just in case */
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if (target <= 0) {
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active = false;
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break;
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}
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todo -= target;
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switch (base->format) {
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case AudioStreamWAV::FORMAT_8_BITS: {
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if (is_stereo) {
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do_resample<int8_t, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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} else {
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do_resample<int8_t, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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}
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} break;
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case AudioStreamWAV::FORMAT_16_BITS: {
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if (is_stereo) {
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do_resample<int16_t, true, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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} else {
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do_resample<int16_t, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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}
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} break;
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case AudioStreamWAV::FORMAT_IMA_ADPCM: {
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if (is_stereo) {
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do_resample<int8_t, true, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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} else {
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do_resample<int8_t, false, true>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm);
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}
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} break;
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}
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dst_buff += target;
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}
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if (todo) {
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int mixed_frames = p_frames - todo;
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//bit was missing from mix
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int todo_ofs = p_frames - todo;
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for (int i = todo_ofs; i < p_frames; i++) {
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p_buffer[i] = AudioFrame(0, 0);
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}
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return mixed_frames;
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}
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return p_frames;
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}
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void AudioStreamPlaybackWAV::tag_used_streams() {
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base->tag_used(get_playback_position());
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}
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AudioStreamPlaybackWAV::AudioStreamPlaybackWAV() {}
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/////////////////////
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void AudioStreamWAV::set_format(Format p_format) {
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format = p_format;
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}
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AudioStreamWAV::Format AudioStreamWAV::get_format() const {
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return format;
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}
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void AudioStreamWAV::set_loop_mode(LoopMode p_loop_mode) {
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loop_mode = p_loop_mode;
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}
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AudioStreamWAV::LoopMode AudioStreamWAV::get_loop_mode() const {
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return loop_mode;
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}
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void AudioStreamWAV::set_loop_begin(int p_frame) {
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loop_begin = p_frame;
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}
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int AudioStreamWAV::get_loop_begin() const {
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return loop_begin;
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}
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void AudioStreamWAV::set_loop_end(int p_frame) {
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loop_end = p_frame;
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}
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int AudioStreamWAV::get_loop_end() const {
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return loop_end;
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}
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void AudioStreamWAV::set_mix_rate(int p_hz) {
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ERR_FAIL_COND(p_hz == 0);
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mix_rate = p_hz;
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}
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int AudioStreamWAV::get_mix_rate() const {
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return mix_rate;
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}
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void AudioStreamWAV::set_stereo(bool p_enable) {
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stereo = p_enable;
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}
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bool AudioStreamWAV::is_stereo() const {
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return stereo;
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}
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double AudioStreamWAV::get_length() const {
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int len = data_bytes;
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switch (format) {
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case AudioStreamWAV::FORMAT_8_BITS:
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len /= 1;
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break;
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case AudioStreamWAV::FORMAT_16_BITS:
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len /= 2;
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break;
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case AudioStreamWAV::FORMAT_IMA_ADPCM:
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len *= 2;
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break;
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}
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if (stereo) {
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len /= 2;
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}
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|
return double(len) / mix_rate;
|
|
}
|
|
|
|
bool AudioStreamWAV::is_monophonic() const {
|
|
return false;
|
|
}
|
|
|
|
void AudioStreamWAV::set_data(const Vector<uint8_t> &p_data) {
|
|
AudioServer::get_singleton()->lock();
|
|
if (data) {
|
|
memfree(data);
|
|
data = nullptr;
|
|
data_bytes = 0;
|
|
}
|
|
|
|
int datalen = p_data.size();
|
|
if (datalen) {
|
|
const uint8_t *r = p_data.ptr();
|
|
int alloc_len = datalen + DATA_PAD * 2;
|
|
data = memalloc(alloc_len); //alloc with some padding for interpolation
|
|
memset(data, 0, alloc_len);
|
|
uint8_t *dataptr = (uint8_t *)data;
|
|
memcpy(dataptr + DATA_PAD, r, datalen);
|
|
data_bytes = datalen;
|
|
}
|
|
|
|
AudioServer::get_singleton()->unlock();
|
|
}
|
|
|
|
Vector<uint8_t> AudioStreamWAV::get_data() const {
|
|
Vector<uint8_t> pv;
|
|
|
|
if (data) {
|
|
pv.resize(data_bytes);
|
|
{
|
|
uint8_t *w = pv.ptrw();
|
|
uint8_t *dataptr = (uint8_t *)data;
|
|
memcpy(w, dataptr + DATA_PAD, data_bytes);
|
|
}
|
|
}
|
|
|
|
return pv;
|
|
}
|
|
|
|
Error AudioStreamWAV::save_to_wav(const String &p_path) {
|
|
if (format == AudioStreamWAV::FORMAT_IMA_ADPCM) {
|
|
WARN_PRINT("Saving IMA_ADPC samples are not supported yet");
|
|
return ERR_UNAVAILABLE;
|
|
}
|
|
|
|
int sub_chunk_2_size = data_bytes; //Subchunk2Size = Size of data in bytes
|
|
|
|
// Format code
|
|
// 1:PCM format (for 8 or 16 bit)
|
|
// 3:IEEE float format
|
|
int format_code = (format == FORMAT_IMA_ADPCM) ? 3 : 1;
|
|
|
|
int n_channels = stereo ? 2 : 1;
|
|
|
|
long sample_rate = mix_rate;
|
|
|
|
int byte_pr_sample = 0;
|
|
switch (format) {
|
|
case AudioStreamWAV::FORMAT_8_BITS:
|
|
byte_pr_sample = 1;
|
|
break;
|
|
case AudioStreamWAV::FORMAT_16_BITS:
|
|
byte_pr_sample = 2;
|
|
break;
|
|
case AudioStreamWAV::FORMAT_IMA_ADPCM:
|
|
byte_pr_sample = 4;
|
|
break;
|
|
}
|
|
|
|
String file_path = p_path;
|
|
if (!(file_path.substr(file_path.length() - 4, 4) == ".wav")) {
|
|
file_path += ".wav";
|
|
}
|
|
|
|
Ref<FileAccess> file = FileAccess::open(file_path, FileAccess::WRITE); //Overrides existing file if present
|
|
|
|
ERR_FAIL_COND_V(file.is_null(), ERR_FILE_CANT_WRITE);
|
|
|
|
// Create WAV Header
|
|
file->store_string("RIFF"); //ChunkID
|
|
file->store_32(sub_chunk_2_size + 36); //ChunkSize = 36 + SubChunk2Size (size of entire file minus the 8 bits for this and previous header)
|
|
file->store_string("WAVE"); //Format
|
|
file->store_string("fmt "); //Subchunk1ID
|
|
file->store_32(16); //Subchunk1Size = 16
|
|
file->store_16(format_code); //AudioFormat
|
|
file->store_16(n_channels); //Number of Channels
|
|
file->store_32(sample_rate); //SampleRate
|
|
file->store_32(sample_rate * n_channels * byte_pr_sample); //ByteRate
|
|
file->store_16(n_channels * byte_pr_sample); //BlockAlign = NumChannels * BytePrSample
|
|
file->store_16(byte_pr_sample * 8); //BitsPerSample
|
|
file->store_string("data"); //Subchunk2ID
|
|
file->store_32(sub_chunk_2_size); //Subchunk2Size
|
|
|
|
// Add data
|
|
Vector<uint8_t> stream_data = get_data();
|
|
const uint8_t *read_data = stream_data.ptr();
|
|
switch (format) {
|
|
case AudioStreamWAV::FORMAT_8_BITS:
|
|
for (unsigned int i = 0; i < data_bytes; i++) {
|
|
uint8_t data_point = (read_data[i] + 128);
|
|
file->store_8(data_point);
|
|
}
|
|
break;
|
|
case AudioStreamWAV::FORMAT_16_BITS:
|
|
for (unsigned int i = 0; i < data_bytes / 2; i++) {
|
|
uint16_t data_point = decode_uint16(&read_data[i * 2]);
|
|
file->store_16(data_point);
|
|
}
|
|
break;
|
|
case AudioStreamWAV::FORMAT_IMA_ADPCM:
|
|
//Unimplemented
|
|
break;
|
|
}
|
|
|
|
return OK;
|
|
}
|
|
|
|
Ref<AudioStreamPlayback> AudioStreamWAV::instantiate_playback() {
|
|
Ref<AudioStreamPlaybackWAV> sample;
|
|
sample.instantiate();
|
|
sample->base = Ref<AudioStreamWAV>(this);
|
|
return sample;
|
|
}
|
|
|
|
String AudioStreamWAV::get_stream_name() const {
|
|
return "";
|
|
}
|
|
|
|
void AudioStreamWAV::_bind_methods() {
|
|
ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamWAV::set_data);
|
|
ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamWAV::get_data);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_format", "format"), &AudioStreamWAV::set_format);
|
|
ClassDB::bind_method(D_METHOD("get_format"), &AudioStreamWAV::get_format);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_mode", "loop_mode"), &AudioStreamWAV::set_loop_mode);
|
|
ClassDB::bind_method(D_METHOD("get_loop_mode"), &AudioStreamWAV::get_loop_mode);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_begin", "loop_begin"), &AudioStreamWAV::set_loop_begin);
|
|
ClassDB::bind_method(D_METHOD("get_loop_begin"), &AudioStreamWAV::get_loop_begin);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_end", "loop_end"), &AudioStreamWAV::set_loop_end);
|
|
ClassDB::bind_method(D_METHOD("get_loop_end"), &AudioStreamWAV::get_loop_end);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_mix_rate", "mix_rate"), &AudioStreamWAV::set_mix_rate);
|
|
ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamWAV::get_mix_rate);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_stereo", "stereo"), &AudioStreamWAV::set_stereo);
|
|
ClassDB::bind_method(D_METHOD("is_stereo"), &AudioStreamWAV::is_stereo);
|
|
|
|
ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamWAV::save_to_wav);
|
|
|
|
ADD_PROPERTY(PropertyInfo(Variant::PACKED_BYTE_ARRAY, "data", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NO_EDITOR), "set_data", "get_data");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "format", PROPERTY_HINT_ENUM, "8-Bit,16-Bit,IMA-ADPCM"), "set_format", "get_format");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_mode", PROPERTY_HINT_ENUM, "Disabled,Forward,Ping-Pong,Backward"), "set_loop_mode", "get_loop_mode");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_begin"), "set_loop_begin", "get_loop_begin");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "loop_end"), "set_loop_end", "get_loop_end");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "mix_rate"), "set_mix_rate", "get_mix_rate");
|
|
ADD_PROPERTY(PropertyInfo(Variant::BOOL, "stereo"), "set_stereo", "is_stereo");
|
|
|
|
BIND_ENUM_CONSTANT(FORMAT_8_BITS);
|
|
BIND_ENUM_CONSTANT(FORMAT_16_BITS);
|
|
BIND_ENUM_CONSTANT(FORMAT_IMA_ADPCM);
|
|
|
|
BIND_ENUM_CONSTANT(LOOP_DISABLED);
|
|
BIND_ENUM_CONSTANT(LOOP_FORWARD);
|
|
BIND_ENUM_CONSTANT(LOOP_PINGPONG);
|
|
BIND_ENUM_CONSTANT(LOOP_BACKWARD);
|
|
}
|
|
|
|
AudioStreamWAV::AudioStreamWAV() {}
|
|
|
|
AudioStreamWAV::~AudioStreamWAV() {
|
|
if (data) {
|
|
memfree(data);
|
|
data = nullptr;
|
|
data_bytes = 0;
|
|
}
|
|
}
|