a7f49ac9a1
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667 lines
19 KiB
C++
667 lines
19 KiB
C++
/*************************************************************************/
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/* audio_stream_sample.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_sample.h"
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#include "core/io/marshalls.h"
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#include "core/os/file_access.h"
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void AudioStreamPlaybackSample::start(float p_from_pos) {
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if (base->format == AudioStreamSample::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 AudioStreamPlaybackSample::stop() {
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active = false;
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}
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bool AudioStreamPlaybackSample::is_playing() const {
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return active;
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}
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int AudioStreamPlaybackSample::get_loop_count() const {
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return 0;
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}
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float AudioStreamPlaybackSample::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 AudioStreamPlaybackSample::seek(float p_time) {
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if (base->format == AudioStreamSample::FORMAT_IMA_ADPCM)
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return; //no seeking in ima-adpcm
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float 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 AudioStreamPlaybackSample::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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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 += AudioStreamSample::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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if (ima_adpcm[i].step_index > 88)
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ima_adpcm[i].step_index = 88;
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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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if (nibble & 2)
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diff += step >> 1;
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if (nibble & 4)
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diff += step;
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if (nibble & 8)
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diff = -diff;
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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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/* 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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if (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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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 (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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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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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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void AudioStreamPlaybackSample::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;
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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 AudioStreamSample::FORMAT_8_BITS: len /= 1; break;
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case AudioStreamSample::FORMAT_16_BITS: len /= 2; break;
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case AudioStreamSample::FORMAT_IMA_ADPCM: len *= 2; 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 != AudioStreamSample::LOOP_DISABLED) ? loop_begin_fp : 0;
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int64_t end_limit = (base->loop_mode != AudioStreamSample::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 == AudioStreamSample::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 fincrement = srate / base_rate;
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int32_t increment = int32_t(fincrement * MIX_FRAC_LEN);
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increment *= sign;
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//looping
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AudioStreamSample::LoopMode loop_format = base->loop_mode;
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AudioStreamSample::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 + AudioStreamSample::DATA_PAD;
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AudioFrame *dst_buff = p_buffer;
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if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
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if (loop_format != AudioStreamSample::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 = AudioStreamSample::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 != AudioStreamSample::LOOP_DISABLED && offset < loop_begin_fp) {
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/* loopstart reached */
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if (loop_format == AudioStreamSample::LOOP_PING_PONG) {
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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 != AudioStreamSample::LOOP_DISABLED && offset >= loop_end_fp) {
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/* loopend reached */
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if (loop_format == AudioStreamSample::LOOP_PING_PONG) {
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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 == AudioStreamSample::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 AudioStreamSample::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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} break;
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case AudioStreamSample::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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} break;
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case AudioStreamSample::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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} 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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//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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}
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}
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AudioStreamPlaybackSample::AudioStreamPlaybackSample() {
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active = false;
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offset = 0;
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sign = 1;
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}
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/////////////////////
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void AudioStreamSample::set_format(Format p_format) {
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format = p_format;
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}
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AudioStreamSample::Format AudioStreamSample::get_format() const {
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return format;
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}
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void AudioStreamSample::set_loop_mode(LoopMode p_loop_mode) {
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loop_mode = p_loop_mode;
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}
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AudioStreamSample::LoopMode AudioStreamSample::get_loop_mode() const {
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return loop_mode;
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}
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void AudioStreamSample::set_loop_begin(int p_frame) {
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loop_begin = p_frame;
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}
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int AudioStreamSample::get_loop_begin() const {
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return loop_begin;
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}
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void AudioStreamSample::set_loop_end(int p_frame) {
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loop_end = p_frame;
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}
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int AudioStreamSample::get_loop_end() const {
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return loop_end;
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}
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void AudioStreamSample::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 AudioStreamSample::get_mix_rate() const {
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return mix_rate;
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}
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void AudioStreamSample::set_stereo(bool p_enable) {
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stereo = p_enable;
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}
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bool AudioStreamSample::is_stereo() const {
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return stereo;
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}
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float AudioStreamSample::get_length() const {
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int len = data_bytes;
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switch (format) {
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case AudioStreamSample::FORMAT_8_BITS: len /= 1; break;
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case AudioStreamSample::FORMAT_16_BITS: len /= 2; break;
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case AudioStreamSample::FORMAT_IMA_ADPCM: len *= 2; 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 float(len) / mix_rate;
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}
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void AudioStreamSample::set_data(const PoolVector<uint8_t> &p_data) {
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AudioServer::get_singleton()->lock();
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if (data) {
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AudioServer::get_singleton()->audio_data_free(data);
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data = NULL;
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data_bytes = 0;
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}
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int datalen = p_data.size();
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if (datalen) {
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PoolVector<uint8_t>::Read r = p_data.read();
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int alloc_len = datalen + DATA_PAD * 2;
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data = AudioServer::get_singleton()->audio_data_alloc(alloc_len); //alloc with some padding for interpolation
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zeromem(data, alloc_len);
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uint8_t *dataptr = (uint8_t *)data;
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copymem(dataptr + DATA_PAD, r.ptr(), datalen);
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data_bytes = datalen;
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}
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AudioServer::get_singleton()->unlock();
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}
|
|
PoolVector<uint8_t> AudioStreamSample::get_data() const {
|
|
|
|
PoolVector<uint8_t> pv;
|
|
|
|
if (data) {
|
|
pv.resize(data_bytes);
|
|
{
|
|
|
|
PoolVector<uint8_t>::Write w = pv.write();
|
|
uint8_t *dataptr = (uint8_t *)data;
|
|
copymem(w.ptr(), dataptr + DATA_PAD, data_bytes);
|
|
}
|
|
}
|
|
|
|
return pv;
|
|
}
|
|
|
|
Error AudioStreamSample::save_to_wav(const String &p_path) {
|
|
if (format == AudioStreamSample::FORMAT_IMA_ADPCM) {
|
|
WARN_PRINTS("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 AudioStreamSample::FORMAT_8_BITS: byte_pr_sample = 1; break;
|
|
case AudioStreamSample::FORMAT_16_BITS: byte_pr_sample = 2; break;
|
|
case AudioStreamSample::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";
|
|
}
|
|
|
|
FileAccessRef file = FileAccess::open(file_path, FileAccess::WRITE); //Overrides existing file if present
|
|
|
|
ERR_FAIL_COND_V(!file, 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
|
|
PoolVector<uint8_t> data = get_data();
|
|
PoolVector<uint8_t>::Read read_data = data.read();
|
|
switch (format) {
|
|
case AudioStreamSample::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 AudioStreamSample::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 AudioStreamSample::FORMAT_IMA_ADPCM:
|
|
//Unimplemented
|
|
break;
|
|
}
|
|
|
|
file->close();
|
|
|
|
return OK;
|
|
}
|
|
|
|
Ref<AudioStreamPlayback> AudioStreamSample::instance_playback() {
|
|
|
|
Ref<AudioStreamPlaybackSample> sample;
|
|
sample.instance();
|
|
sample->base = Ref<AudioStreamSample>(this);
|
|
return sample;
|
|
}
|
|
|
|
String AudioStreamSample::get_stream_name() const {
|
|
|
|
return "";
|
|
}
|
|
|
|
void AudioStreamSample::_bind_methods() {
|
|
|
|
ClassDB::bind_method(D_METHOD("set_data", "data"), &AudioStreamSample::set_data);
|
|
ClassDB::bind_method(D_METHOD("get_data"), &AudioStreamSample::get_data);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_format", "format"), &AudioStreamSample::set_format);
|
|
ClassDB::bind_method(D_METHOD("get_format"), &AudioStreamSample::get_format);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_mode", "loop_mode"), &AudioStreamSample::set_loop_mode);
|
|
ClassDB::bind_method(D_METHOD("get_loop_mode"), &AudioStreamSample::get_loop_mode);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_begin", "loop_begin"), &AudioStreamSample::set_loop_begin);
|
|
ClassDB::bind_method(D_METHOD("get_loop_begin"), &AudioStreamSample::get_loop_begin);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_loop_end", "loop_end"), &AudioStreamSample::set_loop_end);
|
|
ClassDB::bind_method(D_METHOD("get_loop_end"), &AudioStreamSample::get_loop_end);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_mix_rate", "mix_rate"), &AudioStreamSample::set_mix_rate);
|
|
ClassDB::bind_method(D_METHOD("get_mix_rate"), &AudioStreamSample::get_mix_rate);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_stereo", "stereo"), &AudioStreamSample::set_stereo);
|
|
ClassDB::bind_method(D_METHOD("is_stereo"), &AudioStreamSample::is_stereo);
|
|
|
|
ClassDB::bind_method(D_METHOD("save_to_wav", "path"), &AudioStreamSample::save_to_wav);
|
|
|
|
ADD_PROPERTY(PropertyInfo(Variant::POOL_BYTE_ARRAY, "data", PROPERTY_HINT_NONE, "", PROPERTY_USAGE_NOEDITOR), "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_PING_PONG);
|
|
BIND_ENUM_CONSTANT(LOOP_BACKWARD);
|
|
}
|
|
|
|
AudioStreamSample::AudioStreamSample() {
|
|
format = FORMAT_8_BITS;
|
|
loop_mode = LOOP_DISABLED;
|
|
stereo = false;
|
|
loop_begin = 0;
|
|
loop_end = 0;
|
|
mix_rate = 44100;
|
|
data = NULL;
|
|
data_bytes = 0;
|
|
}
|
|
AudioStreamSample::~AudioStreamSample() {
|
|
|
|
if (data) {
|
|
AudioServer::get_singleton()->audio_data_free(data);
|
|
data = NULL;
|
|
data_bytes = 0;
|
|
}
|
|
}
|