768 lines
23 KiB
C++
768 lines
23 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 <typename Depth, bool is_stereo, bool is_ima_adpcm, bool is_qoa>
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void AudioStreamPlaybackWAV::do_resample(const Depth *p_src, AudioFrame *p_dst, int64_t &p_offset, int32_t &p_increment, uint32_t p_amount, IMA_ADPCM_State *p_ima_adpcm, QOA_State *p_qoa) {
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// this function will be compiled branchless by any decent compiler
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int32_t final = 0, final_r = 0, next = 0, next_r = 0;
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while (p_amount) {
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p_amount--;
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int64_t pos = p_offset >> MIX_FRAC_BITS;
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if (is_stereo && !is_ima_adpcm && !is_qoa) {
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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 + p_ima_adpcm[0].window_ofs;
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while (sample_pos > p_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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p_ima_adpcm[i].last_nibble++;
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uint8_t nbb = p_src[(p_ima_adpcm[i].last_nibble >> 1) * (is_stereo ? 2 : 1) + i];
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nibble = (p_ima_adpcm[i].last_nibble & 1) ? (nbb >> 4) : (nbb & 0xF);
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step = _ima_adpcm_step_table[p_ima_adpcm[i].step_index];
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p_ima_adpcm[i].step_index += _ima_adpcm_index_table[nibble];
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if (p_ima_adpcm[i].step_index < 0) {
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p_ima_adpcm[i].step_index = 0;
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}
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if (p_ima_adpcm[i].step_index > 88) {
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p_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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p_ima_adpcm[i].predictor += diff;
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if (p_ima_adpcm[i].predictor < -0x8000) {
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p_ima_adpcm[i].predictor = -0x8000;
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} else if (p_ima_adpcm[i].predictor > 0x7FFF) {
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p_ima_adpcm[i].predictor = 0x7FFF;
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}
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/* store loop if there */
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if (p_ima_adpcm[i].last_nibble == p_ima_adpcm[i].loop_pos) {
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p_ima_adpcm[i].loop_step_index = p_ima_adpcm[i].step_index;
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p_ima_adpcm[i].loop_predictor = p_ima_adpcm[i].predictor;
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}
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//printf("%i - %i - pred %i\n",int(p_ima_adpcm[i].last_nibble),int(nibble),int(p_ima_adpcm[i].predictor));
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}
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}
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final = p_ima_adpcm[0].predictor;
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if (is_stereo) {
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final_r = p_ima_adpcm[1].predictor;
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}
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} else {
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if (is_qoa) {
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if (pos != p_qoa->cache_pos) { // Prevents triple decoding on lower mix rates.
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for (int i = 0; i < 2; i++) {
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// Sign operations prevent triple decoding on backward loops, maxing prevents pop.
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uint32_t interp_pos = MIN(pos + (i * sign) + (sign < 0), p_qoa->desc.samples - 1);
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uint32_t new_data_ofs = 8 + interp_pos / QOA_FRAME_LEN * p_qoa->frame_len;
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if (p_qoa->data_ofs != new_data_ofs) {
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p_qoa->data_ofs = new_data_ofs;
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const uint8_t *ofs_src = (uint8_t *)p_src + p_qoa->data_ofs;
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qoa_decode_frame(ofs_src, p_qoa->frame_len, &p_qoa->desc, p_qoa->dec.ptr(), &p_qoa->dec_len);
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}
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uint32_t dec_idx = (interp_pos % QOA_FRAME_LEN) * p_qoa->desc.channels;
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if ((sign > 0 && i == 0) || (sign < 0 && i == 1)) {
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final = p_qoa->dec[dec_idx];
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p_qoa->cache[0] = final;
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if (is_stereo) {
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final_r = p_qoa->dec[dec_idx + 1];
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p_qoa->cache_r[0] = final_r;
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}
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} else {
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next = p_qoa->dec[dec_idx];
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p_qoa->cache[1] = next;
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if (is_stereo) {
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next_r = p_qoa->dec[dec_idx + 1];
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p_qoa->cache_r[1] = next_r;
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}
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}
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}
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p_qoa->cache_pos = pos;
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} else {
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final = p_qoa->cache[0];
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if (is_stereo) {
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final_r = p_qoa->cache_r[0];
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}
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next = p_qoa->cache[1];
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if (is_stereo) {
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next_r = p_qoa->cache_r[1];
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}
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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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}
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int32_t frac = int64_t(p_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->left = final / 32767.0;
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p_dst->right = final_r / 32767.0;
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p_dst++;
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p_offset += p_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.is_empty() || !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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case AudioStreamWAV::FORMAT_QOA:
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len = qoa.desc.samples * qoa.desc.channels;
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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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const uint8_t *data = base->data.ptr() + 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, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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} else {
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do_resample<int8_t, false, false, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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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, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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} else {
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do_resample<int16_t, false, false, false>((int16_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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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, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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} else {
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do_resample<int8_t, false, true, false>((int8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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}
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} break;
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case AudioStreamWAV::FORMAT_QOA: {
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if (is_stereo) {
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do_resample<uint8_t, true, false, true>((uint8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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} else {
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do_resample<uint8_t, false, false, true>((uint8_t *)data, dst_buff, offset, increment, target, ima_adpcm, &qoa);
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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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void AudioStreamPlaybackWAV::set_is_sample(bool p_is_sample) {
|
|
_is_sample = p_is_sample;
|
|
}
|
|
|
|
bool AudioStreamPlaybackWAV::get_is_sample() const {
|
|
return _is_sample;
|
|
}
|
|
|
|
Ref<AudioSamplePlayback> AudioStreamPlaybackWAV::get_sample_playback() const {
|
|
return sample_playback;
|
|
}
|
|
|
|
void AudioStreamPlaybackWAV::set_sample_playback(const Ref<AudioSamplePlayback> &p_playback) {
|
|
sample_playback = p_playback;
|
|
if (sample_playback.is_valid()) {
|
|
sample_playback->stream_playback = Ref<AudioStreamPlayback>(this);
|
|
}
|
|
}
|
|
|
|
AudioStreamPlaybackWAV::AudioStreamPlaybackWAV() {}
|
|
|
|
AudioStreamPlaybackWAV::~AudioStreamPlaybackWAV() {}
|
|
|
|
/////////////////////
|
|
|
|
void AudioStreamWAV::set_format(Format p_format) {
|
|
format = p_format;
|
|
}
|
|
|
|
AudioStreamWAV::Format AudioStreamWAV::get_format() const {
|
|
return format;
|
|
}
|
|
|
|
void AudioStreamWAV::set_loop_mode(LoopMode p_loop_mode) {
|
|
loop_mode = p_loop_mode;
|
|
}
|
|
|
|
AudioStreamWAV::LoopMode AudioStreamWAV::get_loop_mode() const {
|
|
return loop_mode;
|
|
}
|
|
|
|
void AudioStreamWAV::set_loop_begin(int p_frame) {
|
|
loop_begin = p_frame;
|
|
}
|
|
|
|
int AudioStreamWAV::get_loop_begin() const {
|
|
return loop_begin;
|
|
}
|
|
|
|
void AudioStreamWAV::set_loop_end(int p_frame) {
|
|
loop_end = p_frame;
|
|
}
|
|
|
|
int AudioStreamWAV::get_loop_end() const {
|
|
return loop_end;
|
|
}
|
|
|
|
void AudioStreamWAV::set_mix_rate(int p_hz) {
|
|
ERR_FAIL_COND(p_hz == 0);
|
|
mix_rate = p_hz;
|
|
}
|
|
|
|
int AudioStreamWAV::get_mix_rate() const {
|
|
return mix_rate;
|
|
}
|
|
|
|
void AudioStreamWAV::set_stereo(bool p_enable) {
|
|
stereo = p_enable;
|
|
}
|
|
|
|
bool AudioStreamWAV::is_stereo() const {
|
|
return stereo;
|
|
}
|
|
|
|
double AudioStreamWAV::get_length() const {
|
|
int len = data_bytes;
|
|
switch (format) {
|
|
case AudioStreamWAV::FORMAT_8_BITS:
|
|
len /= 1;
|
|
break;
|
|
case AudioStreamWAV::FORMAT_16_BITS:
|
|
len /= 2;
|
|
break;
|
|
case AudioStreamWAV::FORMAT_IMA_ADPCM:
|
|
len *= 2;
|
|
break;
|
|
case AudioStreamWAV::FORMAT_QOA:
|
|
qoa_desc desc = {};
|
|
qoa_decode_header(data.ptr() + DATA_PAD, data_bytes, &desc);
|
|
len = desc.samples * desc.channels;
|
|
break;
|
|
}
|
|
|
|
if (stereo) {
|
|
len /= 2;
|
|
}
|
|
|
|
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();
|
|
|
|
int src_data_len = p_data.size();
|
|
|
|
data.clear();
|
|
|
|
int alloc_len = src_data_len + DATA_PAD * 2;
|
|
data.resize(alloc_len);
|
|
memset(data.ptr(), 0, alloc_len);
|
|
memcpy(data.ptr() + DATA_PAD, p_data.ptr(), src_data_len);
|
|
data_bytes = src_data_len;
|
|
|
|
AudioServer::get_singleton()->unlock();
|
|
}
|
|
|
|
Vector<uint8_t> AudioStreamWAV::get_data() const {
|
|
Vector<uint8_t> pv;
|
|
|
|
if (!data.is_empty()) {
|
|
pv.resize(data_bytes);
|
|
memcpy(pv.ptrw(), data.ptr() + DATA_PAD, data_bytes);
|
|
}
|
|
|
|
return pv;
|
|
}
|
|
|
|
Error AudioStreamWAV::save_to_wav(const String &p_path) {
|
|
if (format == AudioStreamWAV::FORMAT_IMA_ADPCM || format == AudioStreamWAV::FORMAT_QOA) {
|
|
WARN_PRINT("Saving IMA_ADPCM and QOA samples is 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:
|
|
case AudioStreamWAV::FORMAT_QOA:
|
|
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:
|
|
case AudioStreamWAV::FORMAT_QOA:
|
|
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);
|
|
|
|
if (format == AudioStreamWAV::FORMAT_QOA) {
|
|
uint32_t ffp = qoa_decode_header(data.ptr() + DATA_PAD, data_bytes, &sample->qoa.desc);
|
|
ERR_FAIL_COND_V(ffp != 8, Ref<AudioStreamPlaybackWAV>());
|
|
sample->qoa.frame_len = qoa_max_frame_size(&sample->qoa.desc);
|
|
int samples_len = (sample->qoa.desc.samples > QOA_FRAME_LEN ? QOA_FRAME_LEN : sample->qoa.desc.samples);
|
|
int dec_len = sample->qoa.desc.channels * samples_len;
|
|
sample->qoa.dec.resize(dec_len);
|
|
}
|
|
|
|
return sample;
|
|
}
|
|
|
|
String AudioStreamWAV::get_stream_name() const {
|
|
return "";
|
|
}
|
|
|
|
Ref<AudioSample> AudioStreamWAV::generate_sample() const {
|
|
Ref<AudioSample> sample;
|
|
sample.instantiate();
|
|
sample->stream = this;
|
|
switch (loop_mode) {
|
|
case AudioStreamWAV::LoopMode::LOOP_DISABLED: {
|
|
sample->loop_mode = AudioSample::LoopMode::LOOP_DISABLED;
|
|
} break;
|
|
|
|
case AudioStreamWAV::LoopMode::LOOP_FORWARD: {
|
|
sample->loop_mode = AudioSample::LoopMode::LOOP_FORWARD;
|
|
} break;
|
|
|
|
case AudioStreamWAV::LoopMode::LOOP_PINGPONG: {
|
|
sample->loop_mode = AudioSample::LoopMode::LOOP_PINGPONG;
|
|
} break;
|
|
|
|
case AudioStreamWAV::LoopMode::LOOP_BACKWARD: {
|
|
sample->loop_mode = AudioSample::LoopMode::LOOP_BACKWARD;
|
|
} break;
|
|
}
|
|
sample->loop_begin = loop_begin;
|
|
sample->loop_end = loop_end;
|
|
sample->sample_rate = mix_rate;
|
|
return sample;
|
|
}
|
|
|
|
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,Quite OK Audio"), "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(FORMAT_QOA);
|
|
|
|
BIND_ENUM_CONSTANT(LOOP_DISABLED);
|
|
BIND_ENUM_CONSTANT(LOOP_FORWARD);
|
|
BIND_ENUM_CONSTANT(LOOP_PINGPONG);
|
|
BIND_ENUM_CONSTANT(LOOP_BACKWARD);
|
|
}
|
|
|
|
AudioStreamWAV::AudioStreamWAV() {}
|
|
|
|
AudioStreamWAV::~AudioStreamWAV() {}
|