mirror of
https://github.com/JvanKatwijk/qt-dab.git
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143 lines
4.1 KiB
Plaintext
Executable File
143 lines
4.1 KiB
Plaintext
Executable File
#
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/*
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* Copyright (C) 2013
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* Jan van Katwijk (J.vanKatwijk@gmail.com)
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* Lazy Chair Programming
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*
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* This file is part of the SDR-J (JSDR).
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* SDR-J 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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* SDR-J 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 SDR-J; 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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* This file will be included in mp4processor
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*/
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#
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#include "neaacdec.h"
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#include "audiosink.h"
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#include "newconverter.h"
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class faadDecoder {
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private:
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bool processorOK;
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bool aacInitialized;
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uint32_t aacCap;
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NeAACDecHandle aacHandle;
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NeAACDecConfigurationPtr aacConf;
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NeAACDecFrameInfo hInfo;
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audioSink *ourSink;
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int32_t baudRate;
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newConverter *myConverter;
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DSPCOMPLEX *tempBuffer;
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bool corrector;
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//
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public:
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faadDecoder (audioSink *as, bool corrector) {
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ourSink = as;
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this -> corrector = corrector;
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aacCap = NeAACDecGetCapabilities ();
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aacHandle = NeAACDecOpen ();
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aacConf = NeAACDecGetCurrentConfiguration (aacHandle);
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aacInitialized = false;
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baudRate = 48000;
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myConverter = new newConverter ();
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// fprintf (stderr, "for 4096 samples in krijgen we %d samples uit\n",
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// myConverter -> getOutputSize ());
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tempBuffer = new DSPCOMPLEX [480];
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}
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~faadDecoder (void) {
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NeAACDecClose (aacHandle);
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delete myConverter;
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delete [] tempBuffer;
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}
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int16_t MP42PCM (uint8_t buffer [], int16_t bufferLength) {
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int16_t len;
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int16_t i;
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int16_t samples;
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uint8_t channels;
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long unsigned int sample_rate;
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int16_t *outBuffer;
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NeAACDecFrameInfo hInfo;
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if (!aacInitialized) {
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len = NeAACDecInit (aacHandle,
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buffer, bufferLength, &sample_rate, &channels);
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if (len < 0) {
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fprintf (stderr, "Cannot handle this frame\n");
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return 0;
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}
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outBuffer = (int16_t *)NeAACDecDecode (aacHandle,
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&hInfo,
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&buffer [len],
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(uint64_t)(bufferLength - len));
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aacInitialized = true;
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}
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else
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outBuffer = (int16_t *)NeAACDecDecode (aacHandle,
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&hInfo,
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buffer,
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(uint64_t)bufferLength);
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sample_rate = hInfo. samplerate;
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samples = hInfo. samples;
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if ((sample_rate == 24000) ||
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(sample_rate == 48000) ||
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(sample_rate != baudRate))
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baudRate = sample_rate;
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// fprintf (stderr, "bytes consumed %d\n", (int)(hInfo. bytesconsumed));
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// fprintf (stderr, "samplerate = %d, samples = %d, channels = %d, error = %d, sbr = %d\n", sample_rate, samples,
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// hInfo. channels,
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// hInfo. error,
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// hInfo. sbr);
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// fprintf (stderr, "header = %d\n", hInfo. header_type);
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channels = hInfo. channels;
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if (hInfo. error != 0) {
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fprintf (stderr, "Warning: %s\n",
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faacDecGetErrorMessage (hInfo. error));
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return 0;
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}
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if (channels == 2 && corrector) {
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int16_t amount;
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for (i = 0; i < samples / 2; i ++) {
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if (myConverter -> add (outBuffer [2 * i],
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outBuffer [2 * i + 1],
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tempBuffer, &amount)) {
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ourSink -> putSamples (tempBuffer, amount);
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}
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}
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}
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else
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if (channels == 2)
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ourSink -> audioOut (outBuffer, samples / 2);
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else
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if (channels == 1) {
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int16_t *buffer = (int16_t *)alloca (2 * samples);
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int16_t i;
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for (i = 0; i < samples; i ++) {
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buffer [2 * i] = ((int16_t *)outBuffer) [i];
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buffer [2 * i + 1] = buffer [2 * i];
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}
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ourSink -> audioOut (buffer, samples);
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}
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else
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fprintf (stderr, "Cannot handle these channels\n");
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return samples / 2;
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}
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};
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