mirror of
https://github.com/JvanKatwijk/qt-dab.git
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168 lines
5.3 KiB
C++
168 lines
5.3 KiB
C++
#
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/*
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* Copyright (C) 2013 .. 2024
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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 Qt-DAB
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*
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* Qt-DAB 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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* Qt-DAB 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 Qt-DAB; 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 "faad-decoder.h"
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#include "neaacdec.h"
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#include "radio.h"
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faadDecoder::faadDecoder (RadioInterface *mr,
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RingBuffer<complex16> *buffer) {
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this -> audioBuffer = buffer;
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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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connect (this, &faadDecoder::newAudio,
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mr, &RadioInterface::newAudio);
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}
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faadDecoder::~faadDecoder () {
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NeAACDecClose (aacHandle);
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}
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int get_aac_channel_configuration (int16_t m_mpeg_surround_config,
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uint8_t aacChannelMode) {
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switch(m_mpeg_surround_config) {
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case 0: // no surround
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return aacChannelMode ? 2 : 1;
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case 1: // 5.1
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return 6;
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case 2: // 7.1
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return 7;
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default:
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return -1;
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}
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}
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bool faadDecoder::initialize (stream_parms *sp) {
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long unsigned int sample_rate;
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uint8_t channels;
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/* AudioSpecificConfig structure (the only way to select 960 transform here!)
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*
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* 00010 = AudioObjectType 2 (AAC LC)
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* xxxx = (core) sample rate index
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* xxxx = (core) channel config
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* 100 = GASpecificConfig with 960 transform
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*
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* SBR: implicit signaling sufficient - libfaad2
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* automatically assumes SBR on sample rates <= 24 kHz
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* => explicit signaling works, too, but is not necessary here
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*
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* PS: implicit signaling sufficient - libfaad2
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* therefore always uses stereo output (if PS support was enabled)
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* => explicit signaling not possible, as libfaad2 does not
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* support AudioObjectType 29 (PS)
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*/
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int core_sr_index =
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sp -> dacRate ? (sp -> sbrFlag ? 6 : 3) :
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(sp -> sbrFlag ? 8 : 5); // 24/48/16/32 kHz
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int core_ch_config =
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get_aac_channel_configuration (sp -> mpegSurround,
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sp ->aacChannelMode);
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if (core_ch_config == -1) {
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printf ("Unrecognized mpeg surround config (ignored): %d\n",
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sp -> mpegSurround);
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return false;
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}
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uint8_t asc[2];
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asc [0] = 0b00010 << 3 | core_sr_index >> 1;
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asc [1] = (core_sr_index & 0x01) << 7 | core_ch_config << 3 | 0b100;
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long int init_result = NeAACDecInit2 (aacHandle,
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asc,
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sizeof (asc),
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&sample_rate,
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&channels);
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if (init_result != 0) {
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/* If some error initializing occured, skip the file */
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printf ("Error initializing decoder library: %s\n",
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NeAACDecGetErrorMessage (-init_result));
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NeAACDecClose (aacHandle);
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return false;
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}
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return true;
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}
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int16_t faadDecoder::MP42PCM (stream_parms *sp,
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uint8_t buffer [],
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int16_t bufferLength) {
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int16_t samples;
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long unsigned int sampleRate;
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int16_t *outBuffer;
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NeAACDecFrameInfo hInfo;
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uint8_t channels;
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if (!aacInitialized) {
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if (!initialize (sp))
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return 0;
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aacInitialized = true;
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}
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outBuffer = (int16_t *)NeAACDecDecode (aacHandle,
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&hInfo, buffer, bufferLength);
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sampleRate = hInfo. samplerate;
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samples = hInfo. samples;
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if ((sampleRate == 24000) ||
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(sampleRate == 32000) ||
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(sampleRate == 48000) ||
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(sampleRate != (long unsigned)baudRate))
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baudRate = sampleRate;
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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 -1;
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}
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if (channels == 2) {
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for (int i = 0; i < samples / 2; i ++) {
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complex16 s = complex16 (outBuffer [2 * i],
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outBuffer [2 * i + 1]);
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audioBuffer -> putDataIntoBuffer (&s, 1);
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}
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if (audioBuffer -> GetRingBufferReadAvailable() > (int)sampleRate / 10)
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newAudio (sampleRate / 10, sampleRate, hInfo. ps, hInfo. sbr);
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}
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else
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if (channels == 1) {
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for (int i = 0; i < samples; i ++) {
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complex16 s = complex16 (outBuffer [i], outBuffer [i]);
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audioBuffer -> putDataIntoBuffer (&s, 1);
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}
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if (audioBuffer -> GetRingBufferReadAvailable() > (int)sampleRate / 8)
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newAudio (sampleRate / 10, sampleRate, hInfo. ps, hInfo. sbr);
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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 channels;
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}
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