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83 lines
2.5 KiB
83 lines
2.5 KiB
/*
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* AudioHelpers.cpp
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*
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* Created on: January 1, 2018
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* Author: slascos
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation, either version 3 of the License, or
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* (at your option) any later version.*
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "Audio.h"
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#include "LibBasicFunctions.h"
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namespace BALibrary {
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size_t calcAudioSamples(float milliseconds)
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{
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return (size_t)((milliseconds*(AUDIO_SAMPLE_RATE_EXACT/1000.0f))+0.5f);
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}
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float calcAudioTimeMs(size_t numSamples)
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{
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return ((float)(numSamples) / AUDIO_SAMPLE_RATE_EXACT) * 1000.0f;
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}
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QueuePosition calcQueuePosition(size_t numSamples)
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{
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QueuePosition queuePosition;
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queuePosition.index = (int)(numSamples / AUDIO_BLOCK_SAMPLES);
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queuePosition.offset = numSamples % AUDIO_BLOCK_SAMPLES;
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return queuePosition;
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}
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QueuePosition calcQueuePosition(float milliseconds) {
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size_t numSamples = (int)((milliseconds*(AUDIO_SAMPLE_RATE_EXACT/1000.0f))+0.5f);
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return calcQueuePosition(numSamples);
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}
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size_t calcOffset(QueuePosition position)
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{
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return (position.index*AUDIO_BLOCK_SAMPLES) + position.offset;
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}
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void alphaBlend(audio_block_t *out, audio_block_t *dry, audio_block_t* wet, float mix)
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{
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// ARM DSP optimized
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int16_t wetBuffer[AUDIO_BLOCK_SAMPLES];
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int16_t dryBuffer[AUDIO_BLOCK_SAMPLES];
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int16_t scaleFractWet = (int16_t)(mix * 32767.0f);
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int16_t scaleFractDry = 32767-scaleFractWet;
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arm_scale_q15(dry->data, scaleFractDry, 0, dryBuffer, AUDIO_BLOCK_SAMPLES);
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arm_scale_q15(wet->data, scaleFractWet, 0, wetBuffer, AUDIO_BLOCK_SAMPLES);
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arm_add_q15(wetBuffer, dryBuffer, out->data, AUDIO_BLOCK_SAMPLES);
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}
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void gainAdjust(audio_block_t *out, audio_block_t *in, float vol, int coeffShift)
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{
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int16_t scale = (int16_t)(vol * 32767.0f);
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arm_scale_q15(in->data, scale, coeffShift, out->data, AUDIO_BLOCK_SAMPLES);
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}
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void combine(audio_block_t *out, audio_block_t *in0, audio_block_t *in1)
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{
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arm_add_q15 (in0->data, in1->data, out->data, AUDIO_BLOCK_SAMPLES);
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}
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void clearAudioBlock(audio_block_t *block)
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{
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memset(block->data, 0, sizeof(int16_t)*AUDIO_BLOCK_SAMPLES);
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}
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}
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