# /* * 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 /* * 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, j; 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 (j = 0; j < result; j ++) { buffer [2 * j ] = real (outputBuffer [j]); buffer [2 * j + 1] = imag (outputBuffer [j]); } 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, j; 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 (j = 0; j < result; j ++) { buffer [2 * j ] = real (outputBuffer [j]); buffer [2 * j + 1] = imag (outputBuffer [j]); } 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, j; 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 (j = 0; j < result; j ++) { buffer [2 * j ] = real (outputBuffer [j]); buffer [2 * j + 1] = imag (outputBuffer [j]); } 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; }