/*************************************************************************/ /* audio_stream.cpp */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* http://www.godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */ /* */ /* 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_stream.h" ////////////////////////////// void AudioStreamPlaybackResampled::_begin_resample() { //clear cubic interpolation history internal_buffer[0] = AudioFrame(0.0, 0.0); internal_buffer[1] = AudioFrame(0.0, 0.0); internal_buffer[2] = AudioFrame(0.0, 0.0); internal_buffer[3] = AudioFrame(0.0, 0.0); //mix buffer _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN); mix_offset = 0; } void AudioStreamPlaybackResampled::mix(AudioFrame *p_buffer, float p_rate_scale, int p_frames) { float target_rate = AudioServer::get_singleton()->get_mix_rate() * p_rate_scale; uint64_t mix_increment = uint64_t((get_stream_sampling_rate() / double(target_rate)) * double(FP_LEN)); for (int i = 0; i < p_frames; i++) { uint32_t idx = CUBIC_INTERP_HISTORY + uint32_t(mix_offset >> FP_BITS); //standard cubic interpolation (great quality/performance ratio) //this used to be moved to a LUT for greater performance, but nowadays CPU speed is generally faster than memory. float mu = (mix_offset & FP_MASK) / float(FP_LEN); AudioFrame y0 = internal_buffer[idx - 3]; AudioFrame y1 = internal_buffer[idx - 2]; AudioFrame y2 = internal_buffer[idx - 1]; AudioFrame y3 = internal_buffer[idx - 0]; float mu2 = mu * mu; AudioFrame a0 = y3 - y2 - y0 + y1; AudioFrame a1 = y0 - y1 - a0; AudioFrame a2 = y2 - y0; AudioFrame a3 = y1; p_buffer[i] = (a0 * mu * mu2 + a1 * mu2 + a2 * mu + a3); mix_offset += mix_increment; while ((mix_offset >> FP_BITS) >= INTERNAL_BUFFER_LEN) { internal_buffer[0] = internal_buffer[INTERNAL_BUFFER_LEN + 0]; internal_buffer[1] = internal_buffer[INTERNAL_BUFFER_LEN + 1]; internal_buffer[2] = internal_buffer[INTERNAL_BUFFER_LEN + 2]; internal_buffer[3] = internal_buffer[INTERNAL_BUFFER_LEN + 3]; _mix_internal(internal_buffer + 4, INTERNAL_BUFFER_LEN); mix_offset -= (INTERNAL_BUFFER_LEN << FP_BITS); } } }