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236 lines
7.1 KiB
C++
236 lines
7.1 KiB
C++
/*************************************************************************/
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/* audio_rb_resampler.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_rb_resampler.h"
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#include "core/math/math_funcs.h"
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#include "core/os/os.h"
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#include "servers/audio_server.h"
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int AudioRBResampler::get_channel_count() const {
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if (!rb)
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return 0;
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return channels;
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}
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// Linear interpolation based sample rate conversion (low quality)
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// Note that AudioStreamPlaybackResampled::mix has better algorithm,
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// but it wasn't obvious to integrate that with VideoPlayer
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template <int C>
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uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_increment) {
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uint32_t read = offset & MIX_FRAC_MASK;
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for (int i = 0; i < p_todo; i++) {
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offset = (offset + p_increment) & (((1 << (rb_bits + MIX_FRAC_BITS)) - 1));
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read += p_increment;
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uint32_t pos = offset >> MIX_FRAC_BITS;
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float frac = float(offset & MIX_FRAC_MASK) / float(MIX_FRAC_LEN);
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ERR_FAIL_COND_V(pos >= rb_len, 0);
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uint32_t pos_next = (pos + 1) & rb_mask;
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// since this is a template with a known compile time value (C), conditionals go away when compiling.
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if (C == 1) {
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float v0 = rb[pos];
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float v0n = rb[pos_next];
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v0 += (v0n - v0) * frac;
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p_dest[i] = AudioFrame(v0, v0);
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}
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if (C == 2) {
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float v0 = rb[(pos << 1) + 0];
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float v1 = rb[(pos << 1) + 1];
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float v0n = rb[(pos_next << 1) + 0];
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float v1n = rb[(pos_next << 1) + 1];
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v0 += (v0n - v0) * frac;
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v1 += (v1n - v1) * frac;
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p_dest[i] = AudioFrame(v0, v1);
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}
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// This will probably never be used, but added anyway
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if (C == 4) {
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float v0 = rb[(pos << 2) + 0];
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float v1 = rb[(pos << 2) + 1];
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float v0n = rb[(pos_next << 2) + 0];
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float v1n = rb[(pos_next << 2) + 1];
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v0 += (v0n - v0) * frac;
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v1 += (v1n - v1) * frac;
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p_dest[i] = AudioFrame(v0, v1);
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}
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if (C == 6) {
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float v0 = rb[(pos * 6) + 0];
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float v1 = rb[(pos * 6) + 1];
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float v0n = rb[(pos_next * 6) + 0];
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float v1n = rb[(pos_next * 6) + 1];
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v0 += (v0n - v0) * frac;
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v1 += (v1n - v1) * frac;
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p_dest[i] = AudioFrame(v0, v1);
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}
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}
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return read >> MIX_FRAC_BITS; //rb_read_pos = offset >> MIX_FRAC_BITS;
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}
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bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
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if (!rb)
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return false;
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int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
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int read_space = get_reader_space();
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int target_todo = MIN(get_num_of_ready_frames(), p_frames);
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{
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int src_read = 0;
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switch (channels) {
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case 1: src_read = _resample<1>(p_dest, target_todo, increment); break;
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case 2: src_read = _resample<2>(p_dest, target_todo, increment); break;
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case 4: src_read = _resample<4>(p_dest, target_todo, increment); break;
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case 6: src_read = _resample<6>(p_dest, target_todo, increment); break;
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}
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if (src_read > read_space)
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src_read = read_space;
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rb_read_pos = (rb_read_pos + src_read) & rb_mask;
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// Create fadeout effect for the end of stream (note that it can be because of slow writer)
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if (p_frames - target_todo > 0) {
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for (int i = 0; i < target_todo; i++) {
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p_dest[i] = p_dest[i] * float(target_todo - i) / float(target_todo);
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}
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}
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// Fill zeros (silence) for the rest of frames
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for (int i = target_todo; i < p_frames; i++) {
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p_dest[i] = AudioFrame(0, 0);
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}
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}
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return true;
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}
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int AudioRBResampler::get_num_of_ready_frames() {
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if (!is_ready())
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return 0;
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int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
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int read_space = get_reader_space();
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return (int64_t(read_space) << MIX_FRAC_BITS) / increment;
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}
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Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_mix_rate, int p_buffer_msec, int p_minbuff_needed) {
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ERR_FAIL_COND_V(p_channels != 1 && p_channels != 2 && p_channels != 4 && p_channels != 6, ERR_INVALID_PARAMETER);
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int desired_rb_bits = nearest_shift(MAX((p_buffer_msec / 1000.0) * p_src_mix_rate, p_minbuff_needed));
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bool recreate = !rb;
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if (rb && (uint32_t(desired_rb_bits) != rb_bits || channels != uint32_t(p_channels))) {
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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recreate = true;
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}
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if (recreate) {
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channels = p_channels;
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rb_bits = desired_rb_bits;
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rb_len = (1 << rb_bits);
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rb_mask = rb_len - 1;
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rb = memnew_arr(float, rb_len *p_channels);
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read_buf = memnew_arr(float, rb_len *p_channels);
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}
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src_mix_rate = p_src_mix_rate;
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target_mix_rate = p_target_mix_rate;
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offset = 0;
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rb_read_pos = 0;
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rb_write_pos = 0;
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//avoid maybe strange noises upon load
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for (unsigned int i = 0; i < (rb_len * channels); i++) {
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rb[i] = 0;
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read_buf[i] = 0;
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}
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return OK;
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}
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void AudioRBResampler::clear() {
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if (!rb)
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return;
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//should be stopped at this point but just in case
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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rb = nullptr;
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offset = 0;
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rb_read_pos = 0;
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rb_write_pos = 0;
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read_buf = nullptr;
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}
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AudioRBResampler::AudioRBResampler() {
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rb = nullptr;
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offset = 0;
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read_buf = nullptr;
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rb_read_pos = 0;
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rb_write_pos = 0;
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rb_bits = 0;
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rb_len = 0;
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rb_mask = 0;
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read_buff_len = 0;
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channels = 0;
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src_mix_rate = 0;
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target_mix_rate = 0;
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}
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AudioRBResampler::~AudioRBResampler() {
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if (rb) {
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memdelete_arr(rb);
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memdelete_arr(read_buf);
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}
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}
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