1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
|
/*
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_initialised(false)
{
m_aac = NULL;
}
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], vh.aac_frame_length * 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;
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);
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);
int16_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);
}
}
|
