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SDR-DAB_dab-cmdline/example-1/audio-base.cpp
2017-05-28 12:06:43 +02:00

151 lines
4.5 KiB
C++

#
/*
* Copyright (C) 2011, 2012, 2013
* Jan van Katwijk (J.vanKatwijk@gmail.com)
* Lazy Chair Computing
*
* This file is part of the main program for the DAB library
*
* DAB library is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* DAB library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with DAB library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "audio-base.h"
#include <stdio.h>
/*
* The class is the abstract sink for the data generated
* It will handle the "dumping" though
*/
audioBase::audioBase (void):
converter_16 (16000, 48000, 2 * 1600),
converter_24 (24000, 48000, 2 * 2400),
converter_32 (32000, 48000, 4 * 3200) {
}
audioBase::~audioBase (void) {
}
void audioBase::restart (void) {
}
void audioBase::stop (void) {
}
//
// This one is a hack for handling different baudrates coming from
// the aac decoder. call is from the GUI, triggered by the
// aac decoder or the mp2 decoder
void audioBase::audioOut (int16_t *buffer,
int32_t amount, int32_t rate) {
switch (rate) {
case 16000:
audioOut_16000 (buffer, amount / 2);
return;
case 24000:
audioOut_24000 (buffer, amount / 2);
return;
case 32000:
audioOut_32000 (buffer, amount / 2);
return;
default:
case 48000:
audioOut_48000 (buffer, amount / 2);
return;
}
}
//
// scale up from 16 -> 48
// amount gives number of pairs
void audioBase::audioOut_16000 (int16_t *V, int32_t amount) {
DSPCOMPLEX outputBuffer [converter_16. getOutputsize ()];
float buffer [2 * converter_16. getOutputsize ()];
int16_t i;
int32_t result;
for (i = 0; i < amount; i ++)
if (converter_16. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
V [2 * i + 1] / 32767.0),
outputBuffer, &result)) {
for (i = 0; i < result; i ++) {
buffer [2 * i ] = real (outputBuffer [i]);
buffer [2 * i + 1] = imag (outputBuffer [i]);
}
audioOutput (buffer, result);
}
}
// scale up from 24000 -> 48000
// amount gives number of pairs
void audioBase::audioOut_24000 (int16_t *V, int32_t amount) {
DSPCOMPLEX outputBuffer [converter_24. getOutputsize ()];
float buffer [2 * converter_24. getOutputsize ()];
int16_t i;
int32_t result;
for (i = 0; i < amount; i ++)
if (converter_24. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
V [2 * i + 1] / 32767.0),
outputBuffer, &result)) {
for (i = 0; i < result; i ++) {
buffer [2 * i ] = real (outputBuffer [i]);
buffer [2 * i + 1] = imag (outputBuffer [i]);
}
audioOutput (buffer, result);
}
}
//
// scale up from 32000 -> 48000
// amount is number of pairs
void audioBase::audioOut_32000 (int16_t *V, int32_t amount) {
DSPCOMPLEX outputBuffer [converter_32. getOutputsize ()];
float buffer [2 * converter_32. getOutputsize ()];
int32_t i;
int32_t result;
for (i = 0; i < amount; i ++) {
if (converter_32. convert (DSPCOMPLEX (V [2 * i] / 32767.0,
V [2 * i + 1] / 32767.0),
outputBuffer, &result)) {
for (i = 0; i < result; i ++) {
buffer [2 * i ] = real (outputBuffer [i]);
buffer [2 * i + 1] = imag (outputBuffer [i]);
}
audioOutput (buffer, result);
}
}
}
void audioBase::audioOut_48000 (int16_t *V, int32_t amount) {
float *buffer = (float *)alloca (2 * amount * sizeof (float));
int32_t i;
for (i = 0; i < amount; i ++) {
buffer [2 * i] = V [2 * i] / 32767.0;
buffer [2 * i + 1] = V [2 * i + 1] / 32767.0;
}
audioOutput (buffer, amount);
}
//
// The audioOut function is the one that really should be
// reimplemented in the offsprings of this class
void audioBase::audioOutput (float *v, int32_t amount) {
(void)v;
(void)amount;
}