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/*
Copyright (C) 2014 Matthias P. Braendli (http://www.opendigitalradio.org)
This program 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 3 of the License, or
(at your option) any later version.
This program 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 this program. If not, see <http://www.gnu.org/licenses/>.
faad_decoder.cpp
Use libfaad to decode the AAC content of the DAB+ subchannel
Authors:
Matthias P. Braendli <matthias@mpb.li>
*/
#include "faad_decoder.h"
#include "wavfile.h"
#include "utils.h"
#include <stdio.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <cassert>
#include <string>
#include <sstream>
#include <vector>
using namespace std;
FaadDecoder::FaadDecoder() :
m_data_len(0),
m_fd(NULL),
m_aac(NULL),
m_initialised(false)
{
}
void FaadDecoder::open(string filename, bool ps_flag, bool aac_channel_mode,
bool dac_rate, bool sbr_flag, int mpeg_surround_config)
{
m_filename = filename;
m_ps_flag = ps_flag;
m_aac_channel_mode = aac_channel_mode;
m_dac_rate = dac_rate;
m_sbr_flag = sbr_flag;
m_mpeg_surround_config = mpeg_surround_config;
stringstream ss;
ss << filename << ".aac";
m_aac = fopen(ss.str().c_str(), "w");
}
bool FaadDecoder::decode(vector<vector<uint8_t> > aus)
{
/* ADTS header creation taken from SDR-J */
adts_fixed_header fh;
adts_variable_header vh;
fh.syncword = 0xfff;
fh.id = 0;
fh.layer = 0;
fh.protection_absent = 1;
fh.profile_objecttype = 0; // aac main - 1
fh.private_bit = 0; // ignored when decoding
fh.original_copy = 0;
fh.home = 0;
vh.copyright_id_bit = 0;
vh.copyright_id_start = 0;
vh.adts_buffer_fullness = 1999; // ? according to OpenDab
vh.no_raw_data_blocks = 0;
uint8_t d_header[10];
d_header[0] = fh.syncword >> 4;
d_header[1] = (fh.syncword & 0xf) << 4;
d_header[1] |= fh.id << 3;
d_header[1] |= fh.layer << 1;
d_header[1] |= fh.protection_absent;
d_header[2] = fh.profile_objecttype << 6;
// sampling frequency index filled in dynamically
d_header[2] |= fh.private_bit << 1;
// channel configuration filled in dynamically
d_header[3] = fh.original_copy << 5;
d_header[3] |= fh.home << 4;
d_header[3] |= vh.copyright_id_bit << 3;
d_header[3] |= vh.copyright_id_start << 2;
// framelength filled in dynamically
d_header[4] = 0;
d_header[5] = vh.adts_buffer_fullness >> 6;
d_header[6] = (vh.adts_buffer_fullness & 0x3f) << 2;
d_header[6] |= vh.no_raw_data_blocks;
if (!m_dac_rate && m_sbr_flag) fh.sampling_freq_idx = 8;
// AAC core sampling rate 16 kHz
else if (m_dac_rate && m_sbr_flag) fh.sampling_freq_idx = 6;
// AAC core sampling rate 24 kHz
else if (!m_dac_rate && !m_sbr_flag) fh.sampling_freq_idx = 5;
// AAC core sampling rate 32 kHz
else if (m_dac_rate && !m_sbr_flag) fh.sampling_freq_idx = 3;
// AAC core sampling rate 48 kHz
setBits (&d_header[2], fh.sampling_freq_idx, 2, 4);
if (m_mpeg_surround_config == 0) {
if (m_sbr_flag && !m_aac_channel_mode && m_ps_flag)
fh.channel_conf = 2;
else
fh.channel_conf = 1 << (m_aac_channel_mode ? 1 : 0);
}
else if (m_mpeg_surround_config == 1) {
fh.channel_conf = 6;
}
else {
printf("Unrecognized mpeg surround config (ignored)\n");
return false;
}
setBits (&d_header[2], fh.channel_conf, 7, 3);
for (size_t au_ix = 0; au_ix < aus.size(); au_ix++) {
vector<uint8_t>& au = aus[au_ix];
uint8_t helpBuffer[960];
memset(helpBuffer, 0, sizeof(helpBuffer));
// Set length in header (header + au)
vh.aac_frame_length = 7 + au.size();
setBits(&d_header[3], vh.aac_frame_length, 6, 13);
memcpy(helpBuffer, d_header, 7 * sizeof(uint8_t));
memcpy(&helpBuffer[7],
&au[0], au.size() * sizeof (uint8_t));
fwrite(helpBuffer, 1, vh.aac_frame_length, m_aac);
NeAACDecFrameInfo hInfo;
int16_t* outBuffer;
if (!m_initialised) {
long unsigned samplerate;
unsigned char channels;
int len;
if ((len = NeAACDecInit(m_faad_handle.decoder, helpBuffer,
vh.aac_frame_length, &samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
printf("Error initializing decoder library (%d).\n",
len);
NeAACDecClose(m_faad_handle.decoder);
return false;
}
m_initialised = true;
outBuffer = (int16_t *)NeAACDecDecode(
m_faad_handle.decoder, &hInfo,
helpBuffer + len, vh.aac_frame_length - len );
}
else {
outBuffer = (int16_t *)NeAACDecDecode(
m_faad_handle.decoder, &hInfo,
helpBuffer, vh.aac_frame_length );
}
assert(outBuffer != NULL);
m_sample_rate = hInfo.samplerate;
m_channels = hInfo.channels;
size_t samples = hInfo.samples;
#if 0
printf("bytes consumed %d\n", (int)(hInfo.bytesconsumed));
printf("samplerate = %d, samples = %zu, channels = %d,"
" error = %d, sbr = %d\n", m_sample_rate, samples,
m_channels, hInfo.error, hInfo.sbr);
printf("header = %d\n", hInfo.header_type);
#endif
if (hInfo.error != 0) {
printf("FAAD Warning: %s\n",
faacDecGetErrorMessage(hInfo.error));
return false;
}
if (m_fd == NULL) {
stringstream ss;
ss << m_filename << ".wav";
m_fd = wavfile_open(ss.str().c_str(), m_sample_rate);
}
if (samples) {
if (m_channels == 1) {
int16_t *buffer = (int16_t *)alloca (2 * samples);
size_t i;
for (i = 0; i < samples; i ++) {
buffer [2 * i] = ((int16_t *)outBuffer) [i];
buffer [2 * i + 1] = buffer [2 * i];
}
wavfile_write(m_fd, buffer, 2*samples);
}
else if (m_channels == 2) {
wavfile_write(m_fd, outBuffer, samples);
}
else {
printf("Cannot handle %d channels\n", m_channels);
}
}
}
return true;
}
void FaadDecoder::close()
{
if (m_initialised) {
wavfile_close(m_fd);
}
}
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