/*************************************************************************/ /* audio_rb_resampler.cpp */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2022 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 VideoStreamPlayer 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.set((rb_read_pos.get() + 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; const size_t array_size = rb_len * (size_t)p_channels; rb = memnew_arr(float, array_size); read_buf = memnew_arr(float, array_size); } src_mix_rate = p_src_mix_rate; target_mix_rate = p_target_mix_rate; offset = 0; rb_read_pos.set(0); rb_write_pos.set(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 = nullptr; offset = 0; rb_read_pos.set(0); rb_write_pos.set(0); read_buf = nullptr; } AudioRBResampler::AudioRBResampler() { rb = nullptr; offset = 0; read_buf = nullptr; rb_read_pos.set(0); rb_write_pos.set(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); } }