mirror of
https://github.com/JvanKatwijk/dab-cmdline
synced 2025-10-05 23:52:50 +02:00
151 lines
4.5 KiB
C++
151 lines
4.5 KiB
C++
#
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/*
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* Copyright (C) 2011, 2012, 2013
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* Jan van Katwijk (J.vanKatwijk@gmail.com)
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* Lazy Chair Computing
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*
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* This file is part of the main program for the DAB library
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*
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* DAB library 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 2 of the License, or
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* (at your option) any later version.
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*
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* DAB library 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 DAB library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "audio-base.h"
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#include <stdio.h>
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/*
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* The class is the abstract sink for the data generated
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* It will handle the "dumping" though
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*/
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audioBase::audioBase (void):
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converter_16 (16000, 48000, 2 * 1600),
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converter_24 (24000, 48000, 2 * 2400),
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converter_32 (32000, 48000, 4 * 3200) {
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}
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audioBase::~audioBase (void) {
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}
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void audioBase::restart (void) {
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}
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void audioBase::stop (void) {
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}
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//
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// This one is a hack for handling different baudrates coming from
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// the aac decoder. call is from the GUI, triggered by the
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// aac decoder or the mp2 decoder
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void audioBase::audioOut (int16_t *buffer,
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int32_t amount, int32_t rate) {
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switch (rate) {
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case 16000:
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audioOut_16000 (buffer, amount / 2);
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return;
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case 24000:
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audioOut_24000 (buffer, amount / 2);
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return;
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case 32000:
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audioOut_32000 (buffer, amount / 2);
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return;
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default:
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case 48000:
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audioOut_48000 (buffer, amount / 2);
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return;
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}
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}
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//
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// scale up from 16 -> 48
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// amount gives number of pairs
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void audioBase::audioOut_16000 (int16_t *V, int32_t amount) {
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DSPCOMPLEX outputBuffer [converter_16. getOutputsize ()];
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float buffer [2 * converter_16. getOutputsize ()];
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int16_t i, j;
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int32_t result;
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for (i = 0; i < amount; i ++)
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if (converter_16. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
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V [2 * i + 1] / 32767.0),
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outputBuffer, &result)) {
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for (j = 0; j < result; j ++) {
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buffer [2 * j ] = real (outputBuffer [j]);
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buffer [2 * j + 1] = imag (outputBuffer [j]);
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}
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audioOutput (buffer, result);
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}
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}
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// scale up from 24000 -> 48000
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// amount gives number of pairs
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void audioBase::audioOut_24000 (int16_t *V, int32_t amount) {
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DSPCOMPLEX outputBuffer [converter_24. getOutputsize ()];
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float buffer [2 * converter_24. getOutputsize ()];
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int16_t i, j;
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int32_t result;
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for (i = 0; i < amount; i ++)
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if (converter_24. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
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V [2 * i + 1] / 32767.0),
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outputBuffer, &result)) {
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for (j = 0; j < result; j ++) {
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buffer [2 * j ] = real (outputBuffer [j]);
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buffer [2 * j + 1] = imag (outputBuffer [j]);
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}
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audioOutput (buffer, result);
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}
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}
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//
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// scale up from 32000 -> 48000
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// amount is number of pairs
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void audioBase::audioOut_32000 (int16_t *V, int32_t amount) {
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DSPCOMPLEX outputBuffer [converter_32. getOutputsize ()];
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float buffer [2 * converter_32. getOutputsize ()];
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int32_t i, j;
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int32_t result;
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for (i = 0; i < amount; i ++) {
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if (converter_32. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
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V [2 * i + 1] / 32767.0),
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outputBuffer, &result)) {
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for (j = 0; j < result; j ++) {
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buffer [2 * j ] = real (outputBuffer [j]);
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buffer [2 * j + 1] = imag (outputBuffer [j]);
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}
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audioOutput (buffer, result);
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}
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}
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}
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void audioBase::audioOut_48000 (int16_t *V, int32_t amount) {
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float *buffer = (float *)alloca (2 * amount * sizeof (float));
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int32_t i;
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for (i = 0; i < amount; i ++) {
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buffer [2 * i] = V [2 * i] / 32767.0;
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buffer [2 * i + 1] = V [2 * i + 1] / 32767.0;
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}
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audioOutput (buffer, amount);
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}
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//
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// The audioOut function is the one that really should be
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// reimplemented in the offsprings of this class
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void audioBase::audioOutput (float *v, int32_t amount) {
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(void)v;
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(void)amount;
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}
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