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SDR-DAB_Qt-DAB/sources/backend/audio/faad-decoder.cpp
2025-07-16 20:32:47 +02:00

168 lines
5.3 KiB
C++

#
/*
* Copyright (C) 2013 .. 2024
* Jan van Katwijk (J.vanKatwijk@gmail.com)
* Lazy Chair Computing
*
* This file is part of Qt-DAB
*
* Qt-DAB 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.
*
* Qt-DAB 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 Qt-DAB; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "faad-decoder.h"
#include "neaacdec.h"
#include "radio.h"
faadDecoder::faadDecoder (RadioInterface *mr,
RingBuffer<complex16> *buffer) {
this -> audioBuffer = buffer;
aacCap = NeAACDecGetCapabilities();
aacHandle = NeAACDecOpen();
aacConf = NeAACDecGetCurrentConfiguration (aacHandle);
aacInitialized = false;
baudRate = 48000;
connect (this, &faadDecoder::newAudio,
mr, &RadioInterface::newAudio);
}
faadDecoder::~faadDecoder () {
NeAACDecClose (aacHandle);
}
int get_aac_channel_configuration (int16_t m_mpeg_surround_config,
uint8_t aacChannelMode) {
switch(m_mpeg_surround_config) {
case 0: // no surround
return aacChannelMode ? 2 : 1;
case 1: // 5.1
return 6;
case 2: // 7.1
return 7;
default:
return -1;
}
}
bool faadDecoder::initialize (stream_parms *sp) {
long unsigned int sample_rate;
uint8_t channels;
/* AudioSpecificConfig structure (the only way to select 960 transform here!)
*
* 00010 = AudioObjectType 2 (AAC LC)
* xxxx = (core) sample rate index
* xxxx = (core) channel config
* 100 = GASpecificConfig with 960 transform
*
* SBR: implicit signaling sufficient - libfaad2
* automatically assumes SBR on sample rates <= 24 kHz
* => explicit signaling works, too, but is not necessary here
*
* PS: implicit signaling sufficient - libfaad2
* therefore always uses stereo output (if PS support was enabled)
* => explicit signaling not possible, as libfaad2 does not
* support AudioObjectType 29 (PS)
*/
int core_sr_index =
sp -> dacRate ? (sp -> sbrFlag ? 6 : 3) :
(sp -> sbrFlag ? 8 : 5); // 24/48/16/32 kHz
int core_ch_config =
get_aac_channel_configuration (sp -> mpegSurround,
sp ->aacChannelMode);
if (core_ch_config == -1) {
printf ("Unrecognized mpeg surround config (ignored): %d\n",
sp -> mpegSurround);
return false;
}
uint8_t asc[2];
asc [0] = 0b00010 << 3 | core_sr_index >> 1;
asc [1] = (core_sr_index & 0x01) << 7 | core_ch_config << 3 | 0b100;
long int init_result = NeAACDecInit2 (aacHandle,
asc,
sizeof (asc),
&sample_rate,
&channels);
if (init_result != 0) {
/* If some error initializing occured, skip the file */
printf ("Error initializing decoder library: %s\n",
NeAACDecGetErrorMessage (-init_result));
NeAACDecClose (aacHandle);
return false;
}
return true;
}
int16_t faadDecoder::MP42PCM (stream_parms *sp,
uint8_t buffer [],
int16_t bufferLength) {
int16_t samples;
long unsigned int sampleRate;
int16_t *outBuffer;
NeAACDecFrameInfo hInfo;
uint8_t channels;
if (!aacInitialized) {
if (!initialize (sp))
return 0;
aacInitialized = true;
}
outBuffer = (int16_t *)NeAACDecDecode (aacHandle,
&hInfo, buffer, bufferLength);
sampleRate = hInfo. samplerate;
samples = hInfo. samples;
if ((sampleRate == 24000) ||
(sampleRate == 32000) ||
(sampleRate == 48000) ||
(sampleRate != (long unsigned)baudRate))
baudRate = sampleRate;
channels = hInfo. channels;
if (hInfo. error != 0) {
fprintf (stderr, "Warning: %s\n",
faacDecGetErrorMessage (hInfo. error));
return -1;
}
if (channels == 2) {
for (int i = 0; i < samples / 2; i ++) {
complex16 s = complex16 (outBuffer [2 * i],
outBuffer [2 * i + 1]);
audioBuffer -> putDataIntoBuffer (&s, 1);
}
if (audioBuffer -> GetRingBufferReadAvailable() > (int)sampleRate / 10)
newAudio (sampleRate / 10, sampleRate, hInfo. ps, hInfo. sbr);
}
else
if (channels == 1) {
for (int i = 0; i < samples; i ++) {
complex16 s = complex16 (outBuffer [i], outBuffer [i]);
audioBuffer -> putDataIntoBuffer (&s, 1);
}
if (audioBuffer -> GetRingBufferReadAvailable() > (int)sampleRate / 8)
newAudio (sampleRate / 10, sampleRate, hInfo. ps, hInfo. sbr);
}
else
fprintf (stderr, "Cannot handle these channels\n");
return channels;
}