godot/servers/audio/audio_rb_resampler.cpp
Rémi Verschelde a7f49ac9a1 Update copyright statements to 2020
Happy new year to the wonderful Godot community!

We're starting a new decade with a well-established, non-profit, free
and open source game engine, and tons of further improvements in the
pipeline from hundreds of contributors.

Godot will keep getting better, and we're looking forward to all the
games that the community will keep developing and releasing with it.
2020-01-01 11:16:22 +01:00

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/*************************************************************************/
/* audio_rb_resampler.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2020 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2020 Godot Engine contributors (cf. AUTHORS.md). */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_rb_resampler.h"
#include "core/math/math_funcs.h"
#include "core/os/os.h"
#include "servers/audio_server.h"
int AudioRBResampler::get_channel_count() const {
if (!rb)
return 0;
return channels;
}
// Linear interpolation based sample rate conversion (low quality)
// Note that AudioStreamPlaybackResampled::mix has better algorithm,
// but it wasn't obvious to integrate that with VideoPlayer
template <int C>
uint32_t AudioRBResampler::_resample(AudioFrame *p_dest, int p_todo, int32_t p_increment) {
uint32_t read = offset & MIX_FRAC_MASK;
for (int i = 0; i < p_todo; i++) {
offset = (offset + p_increment) & (((1 << (rb_bits + MIX_FRAC_BITS)) - 1));
read += p_increment;
uint32_t pos = offset >> MIX_FRAC_BITS;
float frac = float(offset & MIX_FRAC_MASK) / float(MIX_FRAC_LEN);
ERR_FAIL_COND_V(pos >= rb_len, 0);
uint32_t pos_next = (pos + 1) & rb_mask;
// since this is a template with a known compile time value (C), conditionals go away when compiling.
if (C == 1) {
float v0 = rb[pos];
float v0n = rb[pos_next];
v0 += (v0n - v0) * frac;
p_dest[i] = AudioFrame(v0, v0);
}
if (C == 2) {
float v0 = rb[(pos << 1) + 0];
float v1 = rb[(pos << 1) + 1];
float v0n = rb[(pos_next << 1) + 0];
float v1n = rb[(pos_next << 1) + 1];
v0 += (v0n - v0) * frac;
v1 += (v1n - v1) * frac;
p_dest[i] = AudioFrame(v0, v1);
}
// This will probably never be used, but added anyway
if (C == 4) {
float v0 = rb[(pos << 2) + 0];
float v1 = rb[(pos << 2) + 1];
float v0n = rb[(pos_next << 2) + 0];
float v1n = rb[(pos_next << 2) + 1];
v0 += (v0n - v0) * frac;
v1 += (v1n - v1) * frac;
p_dest[i] = AudioFrame(v0, v1);
}
if (C == 6) {
float v0 = rb[(pos * 6) + 0];
float v1 = rb[(pos * 6) + 1];
float v0n = rb[(pos_next * 6) + 0];
float v1n = rb[(pos_next * 6) + 1];
v0 += (v0n - v0) * frac;
v1 += (v1n - v1) * frac;
p_dest[i] = AudioFrame(v0, v1);
}
}
return read >> MIX_FRAC_BITS; //rb_read_pos = offset >> MIX_FRAC_BITS;
}
bool AudioRBResampler::mix(AudioFrame *p_dest, int p_frames) {
if (!rb)
return false;
int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
int read_space = get_reader_space();
int target_todo = MIN(get_num_of_ready_frames(), p_frames);
{
int src_read = 0;
switch (channels) {
case 1: src_read = _resample<1>(p_dest, target_todo, increment); break;
case 2: src_read = _resample<2>(p_dest, target_todo, increment); break;
case 4: src_read = _resample<4>(p_dest, target_todo, increment); break;
case 6: src_read = _resample<6>(p_dest, target_todo, increment); break;
}
if (src_read > read_space)
src_read = read_space;
rb_read_pos = (rb_read_pos + src_read) & rb_mask;
// Create fadeout effect for the end of stream (note that it can be because of slow writer)
if (p_frames - target_todo > 0) {
for (int i = 0; i < target_todo; i++) {
p_dest[i] = p_dest[i] * float(target_todo - i) / float(target_todo);
}
}
// Fill zeros (silence) for the rest of frames
for (int i = target_todo; i < p_frames; i++) {
p_dest[i] = AudioFrame(0, 0);
}
}
return true;
}
int AudioRBResampler::get_num_of_ready_frames() {
if (!is_ready())
return 0;
int32_t increment = (src_mix_rate * MIX_FRAC_LEN) / target_mix_rate;
int read_space = get_reader_space();
return (int64_t(read_space) << MIX_FRAC_BITS) / increment;
}
Error AudioRBResampler::setup(int p_channels, int p_src_mix_rate, int p_target_mix_rate, int p_buffer_msec, int p_minbuff_needed) {
ERR_FAIL_COND_V(p_channels != 1 && p_channels != 2 && p_channels != 4 && p_channels != 6, ERR_INVALID_PARAMETER);
int desired_rb_bits = nearest_shift(MAX((p_buffer_msec / 1000.0) * p_src_mix_rate, p_minbuff_needed));
bool recreate = !rb;
if (rb && (uint32_t(desired_rb_bits) != rb_bits || channels != uint32_t(p_channels))) {
memdelete_arr(rb);
memdelete_arr(read_buf);
recreate = true;
}
if (recreate) {
channels = p_channels;
rb_bits = desired_rb_bits;
rb_len = (1 << rb_bits);
rb_mask = rb_len - 1;
rb = memnew_arr(float, rb_len *p_channels);
read_buf = memnew_arr(float, rb_len *p_channels);
}
src_mix_rate = p_src_mix_rate;
target_mix_rate = p_target_mix_rate;
offset = 0;
rb_read_pos = 0;
rb_write_pos = 0;
//avoid maybe strange noises upon load
for (unsigned int i = 0; i < (rb_len * channels); i++) {
rb[i] = 0;
read_buf[i] = 0;
}
return OK;
}
void AudioRBResampler::clear() {
if (!rb)
return;
//should be stopped at this point but just in case
memdelete_arr(rb);
memdelete_arr(read_buf);
rb = NULL;
offset = 0;
rb_read_pos = 0;
rb_write_pos = 0;
read_buf = NULL;
}
AudioRBResampler::AudioRBResampler() {
rb = NULL;
offset = 0;
read_buf = NULL;
rb_read_pos = 0;
rb_write_pos = 0;
rb_bits = 0;
rb_len = 0;
rb_mask = 0;
read_buff_len = 0;
channels = 0;
src_mix_rate = 0;
target_mix_rate = 0;
}
AudioRBResampler::~AudioRBResampler() {
if (rb) {
memdelete_arr(rb);
memdelete_arr(read_buf);
}
}