summaryrefslogtreecommitdiffstats
path: root/src/dabplus-enc.cpp
blob: 2822f94688ae7b7642692a66d58f1db053a0bea7 (plain)
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
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
/* ------------------------------------------------------------------
 * Copyright (C) 2011 Martin Storsjo
 * Copyright (C) 2013,2014 Matthias P. Braendli
 *
 * Licensed under the Apache License, Version 2.0 (the "License");
 * you may not use this file except in compliance with the License.
 * You may obtain a copy of the License at
 *
 *    http://www.apache.org/licenses/LICENSE-2.0
 *
 * Unless required by applicable law or agreed to in writing, software
 * distributed under the License is distributed on an "AS IS" BASIS,
 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either
 * express or implied.
 * See the License for the specific language governing permissions
 * and limitations under the License.
 * -------------------------------------------------------------------
 */

#include "AlsaInput.h"
#include "FileInput.h"
#include "SampleQueue.h"
#include "zmq.hpp"

extern "C" {
#include "encryption.h"
#include "utils.h"
#include "wavreader.h"
}

#include <string>
#include <getopt.h>
#include <cstdio>
#include <stdint.h>
#include <time.h>
#include <unistd.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/ioctl.h>

#include "libAACenc/include/aacenc_lib.h"

extern "C" {
#include <fec.h>
}

using namespace std;

void usage(const char* name) {
    fprintf(stderr,
    "dabplus-enc %s is a HE-AACv2 encoder for DAB+\n"
    "based on fdk-aac-dabplus that can read from a ALSA or file source\n"
    "and encode to a ZeroMQ output for ODR-DabMux.\n"
    "\n"
    "The -D option enables experimental sound card clock drift compensation.\n"
    "A consumer sound card has a clock that is always a bit imprecise, and\n"
    "would drift off after some time. ODR-DabMux cannot handle such drift\n"
    "because it would have to throw away or insert a full DAB+ superframe,\n"
    "which would create audible artifacts. This drift compensation can\n"
    "make sure that the encoding rate is correct by inserting or deleting\n"
    "audio samples.\n"
    "\n"
    "When this option is enabled, you will see U and O printed in the\n"
    "console. These correspond to audio underruns and overruns caused\n"
    "by sound card clock drift. When sparse, they should not create audible\n"
    "artifacts.\n"
    "\n"
    "This encoder includes PAD (DLS and MOT Slideshow) support by\n"
    "http://rd.csp.it to be used with mot-encoder\n"
    "\n"
    "  http://opendigitalradio.org\n"
    "\nUsage:\n"
    "%s (-i file|-d alsa_device) [OPTION...]\n",
#if defined(GITVERSION)
    GITVERSION
#else
    PACKAGE_VERSION
#endif
    , name);
    fprintf(stderr,
    "   For the alsa input:\n"
    "     -d, --device=alsa_device             Set ALSA input device (default: \"default\").\n"
    "     -D, --drift-comp                     Enable ALSA sound card drift compensation.\n"
    "   For the file input:\n"
    "     -i, --input=FILENAME                 Input filename (default: stdin).\n"
    "     -f, --format={ wav, raw }            Set input file format (default: wav).\n"
    "         --fifo-silence                   Input file is fifo and encoder generates silence when fifo is empty. Ignore EOF.\n"
    "   Encoder parameters:\n"
    "     -b, --bitrate={ 8, 16, ..., 192 }    Output bitrate in kbps. Must be 8 multiple.\n"
    "     -a, --afterburner                    Turn on AAC encoder quality increaser.\n"
    "     -c, --channels={ 1, 2 }              Nb of input channels (default: 2).\n"
    "     -r, --rate={ 32000, 48000 }          Input sample rate (default: 48000).\n"
    "         --aaclc                          Force the usage of AAC-LC (no SBR, no PS)\n"
    "         --sbr                            Force the usage of SBR\n"
    "         --ps                             Force the usage of PS\n"
    "   Output and pad parameters:\n"
    "     -o, --output=URI                     Output zmq uri. (e.g. 'tcp://*:9000')\n"
    "                                     -or- Output file uri. (e.g. 'file.dab')\n"
    "                                     -or- a single dash '-' to denote stdout\n"
    "     -k, --secret-key=FILE                Enable ZMQ encryption with the given secret key.\n"
    "     -p, --pad=BYTES                      Set PAD size in bytes.\n"
    "     -P, --pad-fifo=FILENAME              Set PAD data input fifo name"
    "                                          (default: /tmp/pad.fifo).\n"
    "     -l, --level                          Show peak audio level indication.\n"
    "\n"
    "Only the tcp:// zeromq transport has been tested until now,\n"
    " but epgm:// and pgm:// are also accepted\n"
    );

}

int prepare_aac_encoder(
        HANDLE_AACENCODER *encoder,
        int subchannel_index,
        int channels,
        int sample_rate,
        int afterburner,
        int *aot)
{
    HANDLE_AACENCODER handle = *encoder;

    CHANNEL_MODE mode;
    switch (channels) {
        case 1: mode = MODE_1; break;
        case 2: mode = MODE_2; break;
        default:
                fprintf(stderr, "Unsupported channels number %d\n", channels);
                return 1;
    }


    if (aacEncOpen(&handle, 0x01|0x02|0x04, channels) != AACENC_OK) {
        fprintf(stderr, "Unable to open encoder\n");
        return 1;
    }

    *encoder = handle;

    if (*aot == AOT_NONE) {

        if(channels == 2 && subchannel_index <= 6) {
            *aot = AOT_DABPLUS_PS;
        }
        else if((channels == 1 && subchannel_index <= 8) || subchannel_index <= 10) {
            *aot = AOT_DABPLUS_SBR;
        }
        else {
            *aot = AOT_DABPLUS_AAC_LC;
        }
    }

    fprintf(stderr, "Using %d subchannels. AAC type: %s%s%s. channels=%d, sample_rate=%d\n",
            subchannel_index,
            *aot == AOT_DABPLUS_PS ? "HE-AAC v2" : "",
            *aot == AOT_DABPLUS_SBR ? "HE-AAC" : "",
            *aot == AOT_DABPLUS_AAC_LC ? "AAC-LC" : "",
            channels, sample_rate);

    if (aacEncoder_SetParam(handle, AACENC_AOT, *aot) != AACENC_OK) {
        fprintf(stderr, "Unable to set the AOT\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_SAMPLERATE, sample_rate) != AACENC_OK) {
        fprintf(stderr, "Unable to set the sample rate\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_CHANNELMODE, mode) != AACENC_OK) {
        fprintf(stderr, "Unable to set the channel mode\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_CHANNELORDER, 1) != AACENC_OK) {
        fprintf(stderr, "Unable to set the wav channel order\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_GRANULE_LENGTH, 960) != AACENC_OK) {
        fprintf(stderr, "Unable to set the granule length\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_TRANSMUX, TT_DABPLUS) != AACENC_OK) {
        fprintf(stderr, "Unable to set the RAW transmux\n");
        return 1;
    }

    /*if (aacEncoder_SetParam(handle, AACENC_BITRATEMODE, 7 *AACENC_BR_MODE_SFR*)
     * != AACENC_OK) {
        fprintf(stderr, "Unable to set the bitrate mode\n");
        return 1;
    }*/


    fprintf(stderr, "AAC bitrate set to: %d\n", subchannel_index*8000);
    if (aacEncoder_SetParam(handle, AACENC_BITRATE, subchannel_index*8000) != AACENC_OK) {
        fprintf(stderr, "Unable to set the bitrate\n");
        return 1;
    }
    if (aacEncoder_SetParam(handle, AACENC_AFTERBURNER, afterburner) != AACENC_OK) {
        fprintf(stderr, "Unable to set the afterburner mode\n");
        return 1;
    }
    if (aacEncEncode(handle, NULL, NULL, NULL, NULL) != AACENC_OK) {
        fprintf(stderr, "Unable to initialize the encoder\n");
        return 1;
    }
    return 0;
}

#define no_argument 0
#define required_argument 1
#define optional_argument 2

#define STATUS_PAD_INSERTED 0x1
#define STATUS_OVERRUN 0x2
#define STATUS_UNDERRUN 0x4

int main(int argc, char *argv[])
{
    int subchannel_index = 8; //64kbps subchannel
    int ch=0;

    // For the ALSA input
    const char *alsa_device = NULL;

    // For the file input
    const char *infile = NULL;
    int raw_input = 0;

    // For the file output
    FILE *out_fh = NULL;

    const char *outuri = NULL;
    int sample_rate=48000, channels=2;
    const int bytes_per_sample = 2;
    void *rs_handler = NULL;
    bool afterburner = false;
    bool inFifoSilence = false;
    bool drift_compensation = false;
    AACENC_InfoStruct info = { 0 };
    int aot = AOT_NONE;

    /* Keep track of peaks */
    int peak_left  = 0;
    int peak_right = 0;

    /* For MOT Slideshow and DLS insertion */
    const char* pad_fifo = "/tmp/pad.fifo";
    int pad_fd;
    unsigned char pad_buf[128];
    int padlen = 0;

    /* Encoder status, see the above STATUS macros */
    int status = 0;

    /* Whether to show the 'sox'-like measurement */
    int show_level = 0;

    /* Data for ZMQ CURVE authentication */
    char* keyfile = NULL;
    char secretkey[CURVE_KEYLEN+1];

    const struct option longopts[] = {
        {"bitrate",       required_argument,  0, 'b'},
        {"channels",      required_argument,  0, 'c'},
        {"device",        required_argument,  0, 'd'},
        {"format",        required_argument,  0, 'f'},
        {"input",         required_argument,  0, 'i'},
        {"output",        required_argument,  0, 'o'},
        {"pad",           required_argument,  0, 'p'},
        {"pad-fifo",      required_argument,  0, 'P'},
        {"rate",          required_argument,  0, 'r'},
        {"secret-key",    required_argument,  0, 'k'},
        {"afterburner",   no_argument,        0, 'a'},
        {"drift-comp",    no_argument,        0, 'D'},
        {"help",          no_argument,        0, 'h'},
        {"level",         no_argument,        0, 'l'},
        {"aaclc",         no_argument,        0, 0  },
        {"sbr",           no_argument,        0, 1  },
        {"ps",            no_argument,        0, 2  },
        {"fifo-silence",   no_argument,        0, 3  },
        {0,0,0,0},
    };

    if (argc < 2) {
        usage(argv[0]);
        return 1;
    }

    int index;
    while(ch != -1) {
        ch = getopt_long(argc, argv, "ahDlb:c:f:i:k:o:r:d:p:P:", longopts, &index);
        switch (ch) {
        case 0: // AAC-LC
            aot = AOT_DABPLUS_AAC_LC;
            break;
        case 1: // SBR
            aot = AOT_DABPLUS_SBR;
            break;
        case 2: // PS
            aot = AOT_DABPLUS_PS;
            break;
        case 3: // FIFO SILENCE
            inFifoSilence = true;
            break;
        case 'a':
            afterburner = true;
            break;
        case 'b':
            subchannel_index = atoi(optarg) / 8;
            break;
        case 'c':
            channels = atoi(optarg);
            break;
        case 'd':
            alsa_device = optarg;
            break;
        case 'D':
            drift_compensation = true;
            break;
        case 'f':
            if(strcmp(optarg, "raw")==0) {
                raw_input = 1;
            } else if(strcmp(optarg, "wav")!=0)
                usage(argv[0]);
            break;
        case 'i':
            infile = optarg;
            break;
        case 'k':
            keyfile = optarg;
            break;
        case 'l':
            show_level = 1;
            break;
        case 'o':
            outuri = optarg;
            break;
        case 'p':
            padlen = atoi(optarg);
            break;
        case 'P':
            pad_fifo = optarg;
            break;
        case 'r':
            sample_rate = atoi(optarg);
            break;
        case '?':
        case 'h':
            usage(argv[0]);
            return 1;
        }
    }

    if (alsa_device && infile) {
        fprintf(stderr, "You must define either alsa or file input, not both\n");
        return 1;
    }

    if (subchannel_index < 1 || subchannel_index > 24) {
        fprintf(stderr, "Bad subchannels number: %d, try other bitrate.\n",
                subchannel_index);
        return 1;
    }

    if ( ! (sample_rate == 32000 || sample_rate == 48000)) {
        fprintf(stderr, "Invalid sample rate. Possible values are: 32000, 48000.\n");
        return 1;
    }

    zmq::context_t zmq_ctx;
    zmq::socket_t zmq_sock(zmq_ctx, ZMQ_PUB);

    if (outuri) {
        if (strcmp(outuri, "-") == 0) {
            out_fh = stdout;
        }
        else if ((strncmp(outuri, "tcp://", 6) == 0) ||
                (strncmp(outuri, "pgm://", 6) == 0) ||
                (strncmp(outuri, "epgm://", 7) == 0)) {
            if (keyfile) {
                fprintf(stderr, "Enabling encryption\n");

                int rc = readkey(keyfile, secretkey);
                if (rc) {
                    fprintf(stderr, "Error reading secret key\n");
                    return 2;
                }

                const int yes = 1;
                zmq_sock.setsockopt(ZMQ_CURVE_SERVER,
                        &yes, sizeof(yes));

                zmq_sock.setsockopt(ZMQ_CURVE_SECRETKEY,
                        secretkey, CURVE_KEYLEN);
            }
            zmq_sock.connect(outuri);
        }
        else { // We assume it's a file name
            out_fh = fopen(outuri, "wb");

            if (!out_fh) {
                fprintf(stderr, "Can't open output file!\n");
                return 1;
            }
        }
    }
    else {
        fprintf(stderr, "Output URI not defined\n");
        return 1;
    }

    if (padlen != 0) {
        int flags;
        if (mkfifo(pad_fifo, S_IWUSR | S_IRUSR | S_IRGRP | S_IROTH) != 0) {
            if (errno != EEXIST) {
                fprintf(stderr, "Can't create pad file: %d!\n", errno);
                return 1;
            }
        }
        pad_fd = open(pad_fifo, O_RDONLY | O_NONBLOCK);
        if (pad_fd == -1) {
            fprintf(stderr, "Can't open pad file!\n");
            return 1;
        }
        flags = fcntl(pad_fd, F_GETFL, 0);
        if (fcntl(pad_fd, F_SETFL, flags | O_NONBLOCK)) {
            fprintf(stderr, "Can't set non-blocking mode in pad file!\n");
            return 1;
        }
    }


    HANDLE_AACENCODER encoder;

    if (prepare_aac_encoder(&encoder, subchannel_index, channels,
                sample_rate, afterburner, &aot) != 0) {
        fprintf(stderr, "Encoder preparation failed\n");
        return 2;
    }

    /* We assume that we need to call the encoder
     * enc_calls_per_output before it gives us one encoded audio
     * frame. This information is used when the alsa drift compensation
     * is active
     */
    const int enc_calls_per_output =
        (aot == AOT_DABPLUS_AAC_LC) ? sample_rate / 8000 : sample_rate / 16000;


    if (aacEncInfo(encoder, &info) != AACENC_OK) {
        fprintf(stderr, "Unable to get the encoder info\n");
        return 1;
    }

    // Each DAB+ frame will need input_size audio bytes
    const int input_size = channels * bytes_per_sample * info.frameLength;
    fprintf(stderr, "DAB+ Encoding: framelen=%d (%dB)\n",
            info.frameLength,
            input_size);

    uint8_t input_buf[input_size];

    int max_size = 2*input_size + NUM_SAMPLES_PER_CALL;

    SampleQueue<uint8_t> queue(BYTES_PER_SAMPLE, channels, max_size);

    /* symsize=8, gfpoly=0x11d, fcr=0, prim=1, nroots=10, pad=135 */
    rs_handler = init_rs_char(8, 0x11d, 0, 1, 10, 135);
    if (rs_handler == NULL) {
        perror("init_rs_char failed");
        return 1;
    }

    /* No input defined ? default to alsa "default" */
    if (!alsa_device) {
        alsa_device = "default";
    }

    // We'll use one of the tree possible inputs
    AlsaInputThreaded alsa_in_threaded(alsa_device, channels, sample_rate, queue);
    AlsaInputDirect   alsa_in_direct(alsa_device, channels, sample_rate);
    FileInput         file_in(infile, raw_input, sample_rate);

    if (infile) {
        if (file_in.prepare() != 0) {
            fprintf(stderr, "File input preparation failed\n");
            return 1;
        }
    }
    else if (drift_compensation) {
        if (alsa_in_threaded.prepare() != 0) {
            fprintf(stderr, "Alsa preparation failed\n");
            return 1;
        }

        fprintf(stderr, "Start ALSA capture thread\n");
        alsa_in_threaded.start();
    }
    else {
        if (alsa_in_direct.prepare() != 0) {
            fprintf(stderr, "Alsa preparation failed\n");
            return 1;
        }
    }

    int outbuf_size = subchannel_index*120;
    uint8_t zmqframebuf[2048];
    zmq_frame_header_t *zmq_frame_header = (zmq_frame_header_t*)zmqframebuf;

    uint8_t outbuf[2048];

    if(outbuf_size % 5 != 0) {
        fprintf(stderr, "(outbuf_size mod 5) = %d\n", outbuf_size % 5);
    }

    fprintf(stderr, "Starting encoding\n");

    int send_error_count = 0;
    struct timespec tp_next;
    clock_gettime(CLOCK_MONOTONIC, &tp_next);

    int calls = 0; // for checking
    while (1) {
        int in_identifier[] = {IN_AUDIO_DATA, IN_ANCILLRY_DATA};
        int out_identifier = OUT_BITSTREAM_DATA;

        AACENC_BufDesc in_buf = { 0 }, out_buf = { 0 };
        AACENC_InArgs in_args = { 0 };
        AACENC_OutArgs out_args = { 0 };
        void *in_ptr[2], *out_ptr;
        int in_size[2], in_elem_size[2];
        int out_size, out_elem_size;


        // -------------- wait the right amount of time
        if (drift_compensation) {
            struct timespec tp_now;
            clock_gettime(CLOCK_MONOTONIC, &tp_now);

            unsigned long time_now  = (1000000000ul * tp_now.tv_sec) +
                tp_now.tv_nsec;
            unsigned long time_next = (1000000000ul * tp_next.tv_sec) +
                tp_next.tv_nsec;

            const unsigned long dabplus_superframe_nsec = 120000000ul;

            const unsigned long wait_time =
                dabplus_superframe_nsec / enc_calls_per_output;

            unsigned long waiting = wait_time - (time_now - time_next);
            if ((time_now - time_next) < wait_time) {
                //printf("Sleep %zuus\n", waiting / 1000);
                usleep(waiting / 1000);
            }

            // Move our time_counter 60ms into the future.
            // The encoder needs two calls for one frame
            tp_next.tv_nsec += wait_time;
            if (tp_next.tv_nsec >  1000000000L) {
                tp_next.tv_nsec -= 1000000000L;
                tp_next.tv_sec  += 1;
            }
        }

        // --------------- Read data from the PAD fifo
        int ret;
        if (padlen != 0) {
            ret = read(pad_fd, pad_buf, padlen);
        }
        else {
            ret = 0;
        }


        if(ret < 0 && errno == EAGAIN) {
            // If this condition passes, there is no data to be read
            in_buf.numBufs = 1;    // Samples;
        }
        else if(ret >= 0) {
            // Otherwise, you're good to go and buffer should contain "count" bytes.
            in_buf.numBufs = 2;    // Samples + Data;
            if (ret > 0)
                status |= STATUS_PAD_INSERTED;
        }
        else {
            // Some other error occurred during read.
            fprintf(stderr, "Unable to read from PAD!\n");
                        break;
        }

        // -------------- Read Data
        memset(outbuf, 0x00, outbuf_size);
        memset(input_buf, 0x00, input_size);

        ssize_t read;
        if (infile) {
            read = file_in.read(input_buf, input_size);
            if (read < 0) {
                break;
            }
            else if (read != input_size) {
                if (inFifoSilence && file_in.eof()) {
                    memset(input_buf, 0, input_size);
                    read = input_size;
                    usleep((long)input_size * 1000000 /
                            (bytes_per_sample * channels * sample_rate));
                }
                else {
                    fprintf(stderr, "Short file read !\n");
                    break;
                }
            }
        }
        else if (drift_compensation) {
            if (alsa_in_threaded.fault_detected()) {
                fprintf(stderr, "Detected fault in alsa input!\n");
                break;
            }

            size_t overruns;
            read = queue.pop(input_buf, input_size, &overruns); // returns bytes

            if (read != input_size) {
                status |= STATUS_UNDERRUN;
            }

            if (overruns) {
                status |= STATUS_OVERRUN;
            }
        }
        else {
            read = alsa_in_direct.read(input_buf, input_size);
            if (read < 0) {
                break;
            }
            else if (read != input_size) {
                fprintf(stderr, "Short alsa read !\n");
            }
        }

        for (int i = 0; i < read; i+=4) {
            int16_t l = input_buf[i] | (input_buf[i+1] << 8);
            int16_t r = input_buf[i+2] | (input_buf[i+3] << 8);
            peak_left  = MAX(peak_left,  l);
            peak_right = MAX(peak_right, r);
        }

        // -------------- AAC Encoding

        int calculated_padlen = ret > 0 ? padlen : 0;


        in_ptr[0] = input_buf;
        in_ptr[1] = pad_buf;
        in_size[0] = read;
        in_size[1] = calculated_padlen;
        in_elem_size[0] = BYTES_PER_SAMPLE;
        in_elem_size[1] = sizeof(uint8_t);
        in_args.numInSamples = input_size/BYTES_PER_SAMPLE;
        in_args.numAncBytes = calculated_padlen;

        in_buf.bufs = (void**)&in_ptr;
        in_buf.bufferIdentifiers = in_identifier;
        in_buf.bufSizes = in_size;
        in_buf.bufElSizes = in_elem_size;

        out_ptr = outbuf;
        out_size = sizeof(outbuf);
        out_elem_size = 1;
        out_buf.numBufs = 1;
        out_buf.bufs = &out_ptr;
        out_buf.bufferIdentifiers = &out_identifier;
        out_buf.bufSizes = &out_size;
        out_buf.bufElSizes = &out_elem_size;

        AACENC_ERROR err;
        if ((err = aacEncEncode(encoder, &in_buf, &out_buf, &in_args, &out_args))
                != AACENC_OK) {
            if (err == AACENC_ENCODE_EOF) {
                fprintf(stderr, "encoder error: EOF reached\n");
                break;
            }
            fprintf(stderr, "Encoding failed (%d)\n", err);
            break;
        }
        calls++;

        /* Check if the encoder has generated output data */
        if (out_args.numOutBytes != 0)
        {
            // Our timing code depends on this
            if (calls != enc_calls_per_output) {
                fprintf(stderr, "INTERNAL ERROR! calls=%d"
                        ", expected %d\n",
                        calls, enc_calls_per_output);
            }
            calls = 0;

            // ----------- RS encoding
            int row, col;
            unsigned char buf_to_rs_enc[110];
            unsigned char rs_enc[10];
            for(row=0; row < subchannel_index; row++) {
                for(col=0;col < 110; col++) {
                    buf_to_rs_enc[col] = outbuf[subchannel_index * col + row];
                }

                encode_rs_char(rs_handler, buf_to_rs_enc, rs_enc);

                for(col=110; col<120; col++) {
                    outbuf[subchannel_index * col + row] = rs_enc[col-110];
                    assert(subchannel_index * col + row < outbuf_size);
                }
            }

            if (out_fh) {
                fwrite(outbuf, 1, outbuf_size, out_fh);
            }
            else {
                // ------------ ZeroMQ transmit
                try {
                    zmq_frame_header->version = 1;
                    zmq_frame_header->encoder = ZMQ_ENCODER_FDK;
                    zmq_frame_header->datasize = outbuf_size;
                    zmq_frame_header->audiolevel_left = peak_left;
                    zmq_frame_header->audiolevel_right = peak_right;

                    memcpy(ZMQ_FRAME_DATA(zmq_frame_header),
                            outbuf, outbuf_size);

                    zmq_sock.send(zmqframebuf, ZMQ_FRAME_SIZE(zmq_frame_header),
                            ZMQ_DONTWAIT);
                }
                catch (zmq::error_t& e) {
                    fprintf(stderr, "ZeroMQ send error !\n");
                    send_error_count ++;
                }

                if (send_error_count > 10)
                {
                    fprintf(stderr, "ZeroMQ send failed ten times, aborting!\n");
                    break;
                }
            }

            if (out_args.numOutBytes + row*10 == outbuf_size) {
                if (show_level) {
                    fprintf(stderr, "\rIn: [%6s|%-6s] %1s %1s %1s",
                            level(0, &peak_left),
                            level(1, &peak_right),
                            status & STATUS_PAD_INSERTED ? "P" : " ",
                            status & STATUS_UNDERRUN ? "U" : " ",
                            status & STATUS_OVERRUN ? "O" : " ");
                }
                else {
                    if (status & STATUS_OVERRUN) {
                        fprintf(stderr, "O");
                    }

                    if (status & STATUS_UNDERRUN) {
                        fprintf(stderr, "U");
                    }
                }
            }

            status = 0;
        }

        fflush(stdout);
    }
    fprintf(stderr, "\n");

    if (out_fh) {
        fclose(out_fh);
    }

    zmq_sock.close();
    free_rs_char(rs_handler);

    aacEncClose(&encoder);
}