0be6d925dc
Which means that reduz' beloved style which we all became used to will now be changed automatically to remove the first empty line. This makes us lean closer to 1TBS (the one true brace style) instead of hybridating it with some Allman-inspired spacing. There's still the case of braces around single-statement blocks that needs to be addressed (but clang-format can't help with that, but clang-tidy may if we agree about it). Part of #33027.
646 lines
19 KiB
C++
646 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:
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len /= 1;
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break;
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case AudioStreamSample::FORMAT_16_BITS:
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len /= 2;
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break;
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case AudioStreamSample::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 != 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(MAX(fincrement * MIX_FRAC_LEN, 1));
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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:
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len /= 1;
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break;
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case AudioStreamSample::FORMAT_16_BITS:
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len /= 2;
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break;
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case AudioStreamSample::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 float(len) / mix_rate;
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}
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void AudioStreamSample::set_data(const Vector<uint8_t> &p_data) {
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AudioServer::get_singleton()->lock();
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if (data) {
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memfree(data);
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data = nullptr;
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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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const uint8_t *r = p_data.ptr();
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int alloc_len = datalen + DATA_PAD * 2;
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data = memalloc(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, datalen);
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data_bytes = datalen;
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}
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AudioServer::get_singleton()->unlock();
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}
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Vector<uint8_t> AudioStreamSample::get_data() const {
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Vector<uint8_t> pv;
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if (data) {
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pv.resize(data_bytes);
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{
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uint8_t *w = pv.ptrw();
|
|
uint8_t *dataptr = (uint8_t *)data;
|
|
copymem(w, dataptr + DATA_PAD, data_bytes);
|
|
}
|
|
}
|
|
|
|
return pv;
|
|
}
|
|
|
|
Error AudioStreamSample::save_to_wav(const String &p_path) {
|
|
if (format == AudioStreamSample::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 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
|
|
Vector<uint8_t> data = get_data();
|
|
const uint8_t *read_data = data.ptr();
|
|
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::PACKED_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 = nullptr;
|
|
data_bytes = 0;
|
|
}
|
|
|
|
AudioStreamSample::~AudioStreamSample() {
|
|
if (data) {
|
|
memfree(data);
|
|
data = nullptr;
|
|
data_bytes = 0;
|
|
}
|
|
}
|