aboutsummaryrefslogtreecommitdiff
path: root/src/plugins/previewopus_extractor.c
blob: 0b9ab0bdbb015b1ddbd46cae7407c40af5131a8b (plain) (blame)
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
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
1129
1130
1131
1132
1133
1134
1135
1136
1137
1138
1139
1140
1141
1142
1143
1144
1145
1146
1147
1148
1149
1150
1151
1152
1153
1154
1155
1156
1157
1158
1159
1160
1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
1171
1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
1189
1190
1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
1205
1206
1207
1208
1209
1210
1211
1212
1213
1214
1215
1216
1217
1218
1219
1220
1221
1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235
1236
1237
1238
1239
1240
1241
1242
1243
1244
1245
1246
1247
1248
1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
1262
1263
1264
1265
1266
1267
1268
1269
1270
1271
1272
1273
1274
1275
1276
1277
1278
/*
     This file is part of libextractor.
     Copyright Copyright (C) 2008, 2013 Bruno Cabral and Christian Grothoff

     libextractor 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, or (at your
     option) any later version.

     libextractor 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 libextractor; see the file COPYING.  If not, write to the
     Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
     Boston, MA 02110-1301, USA.
 */
/**
 * @file previewopus_extractor.c
 * @author Bruno Cabral
 * @author Christian Grothoff
 * @brief this extractor produces a binary encoded
 * audio snippet of music/video files using ffmpeg libs.
 *
 * Based on ffmpeg samples.
 *
 * Note that ffmpeg has a few issues:
 * (1) there are no recent official releases of the ffmpeg libs
 * (2) ffmpeg has a history of having security issues (parser is not robust)
 *
 *  So this plugin cannot be recommended for system with high security
 *requirements.
 */
#include "platform.h"
#include "extractor.h"
#include <magic.h>

#if HAVE_LIBAVUTIL_AVUTIL_H
#include <libavutil/avutil.h>
#include <libavutil/audio_fifo.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>

#elif HAVE_FFMPEG_AVUTIL_H
#include <ffmpeg/avutil.h>
#include <ffmpeg/audio_fifo.h>
#include <ffmpeg/opt.h>
#include <ffmpeg/mathematics.h>
#endif
#if HAVE_LIBAVFORMAT_AVFORMAT_H
#include <libavformat/avformat.h>
#elif HAVE_FFMPEG_AVFORMAT_H
#include <ffmpeg/avformat.h>
#endif
#if HAVE_LIBAVCODEC_AVCODEC_H
#include <libavcodec/avcodec.h>
#elif HAVE_FFMPEG_AVCODEC_H
#include <ffmpeg/avcodec.h>
#endif
#if HAVE_LIBSWSCALE_SWSCALE_H
#include <libswscale/swscale.h>
#elif HAVE_FFMPEG_SWSCALE_H
#include <ffmpeg/swscale.h>
#endif

// TODO: Check for ffmpeg
#include <libavresample/avresample.h>


/**
 * Set to 1 to enable debug output.
 */
#define DEBUG 0

/**
 * Set to 1 to enable a output file for testing.
 */
#define OUTPUT_FILE 0


/**
 * Maximum size in bytes for the preview.
 */
#define MAX_SIZE (28 * 1024)

/**
 * HardLimit for file
 */
#define HARD_LIMIT_SIZE (50 * 1024)


/** The output bit rate in kbit/s */
#define OUTPUT_BIT_RATE 28000
/** The number of output channels */
#define OUTPUT_CHANNELS 2
/** The audio sample output format */
#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16


/** Our output buffer*/
static unsigned char *buffer;

/** Actual output buffer size */
static int totalSize;

/**
 * Convert an error code into a text message.
 * @param error Error code to be converted
 * @return Corresponding error text (not thread-safe)
 */
static char *const
get_error_text (const int error)
{
  static char error_buffer[255];
  av_strerror (error, error_buffer, sizeof(error_buffer));
  return error_buffer;
}


/**
 * Read callback.
 *
 * @param opaque the 'struct EXTRACTOR_ExtractContext'
 * @param buf where to write data
 * @param buf_size how many bytes to read
 * @return -1 on error (or for unknown file size)
 */
static int
read_cb (void *opaque,
         uint8_t *buf,
         int buf_size)
{
  struct EXTRACTOR_ExtractContext *ec = opaque;
  void *data;
  ssize_t ret;

  ret = ec->read (ec->cls, &data, buf_size);
  if (ret <= 0)
    return ret;
  memcpy (buf, data, ret);
  return ret;
}


/**
 * Seek callback.
 *
 * @param opaque the 'struct EXTRACTOR_ExtractContext'
 * @param offset where to seek
 * @param whence how to seek; AVSEEK_SIZE to return file size without seeking
 * @return -1 on error (or for unknown file size)
 */
static int64_t
seek_cb (void *opaque,
         int64_t offset,
         int whence)
{
  struct EXTRACTOR_ExtractContext *ec = opaque;

  if (AVSEEK_SIZE == whence)
    return ec->get_size (ec->cls);
  return ec->seek (ec->cls, offset, whence);
}


/**
 * write callback.
 *
 * @param opaque NULL
 * @param pBuffer to write
 * @param pBufferSize , amount to write
 * @return 0 on error
 */
static int
writePacket (void *opaque,
             unsigned char *pBuffer,
             int pBufferSize)
{
  int sizeToCopy = pBufferSize;

  if ( (totalSize + pBufferSize) > HARD_LIMIT_SIZE)
    sizeToCopy = HARD_LIMIT_SIZE - totalSize;

  memcpy (buffer + totalSize, pBuffer, sizeToCopy);
  totalSize += sizeToCopy;
  return sizeToCopy;
}


/**
 * Open an output file and the required encoder.
 * Also set some basic encoder parameters.
 * Some of these parameters are based on the input file's parameters.
 */
static int
open_output_file (
  AVCodecContext *input_codec_context,
  AVFormatContext **output_format_context,
  AVCodecContext **output_codec_context)
{
  AVStream *stream               = NULL;
  AVCodec *output_codec          = NULL;
  AVIOContext *io_ctx;
  int error;
  unsigned char *iob;

  if (NULL == (iob = av_malloc (16 * 1024)))
    return AVERROR_EXIT;
  if (NULL == (io_ctx = avio_alloc_context (iob, 16 * 1024,
                                            AVIO_FLAG_WRITE, NULL,
                                            NULL,
                                            &writePacket /* no writing */,
                                            NULL)))
  {
    av_free (iob);
    return AVERROR_EXIT;
  }
  if (NULL == ((*output_format_context) = avformat_alloc_context ()))
  {
    av_free (io_ctx);
    return AVERROR_EXIT;
  }
  (*output_format_context)->pb = io_ctx;

  /** Guess the desired container format based on the file extension. */
  if (! ((*output_format_context)->oformat = av_guess_format (NULL,
                                                              "file.ogg",
                                                              NULL)))
  {
#if DEBUG
    fprintf (stderr, "Could not find output file format\n");
#endif
    goto cleanup;
  }

  /** Find the encoder to be used by its name. */
  if (! (output_codec = avcodec_find_encoder (AV_CODEC_ID_OPUS)))
  {
#if DEBUG
    fprintf (stderr, "Could not find an OPUS encoder.\n");
#endif
    goto cleanup;
  }

  /** Create a new audio stream in the output file container. */
  if (! (stream = avformat_new_stream (*output_format_context, output_codec)))
  {
#if DEBUG
    fprintf (stderr, "Could not create new stream\n");
#endif
    error = AVERROR (ENOMEM);
    goto cleanup;
  }

  /** Save the encoder context for easiert access later. */
  *output_codec_context = stream->codec;

  /**
   * Set the basic encoder parameters.
   * The input file's sample rate is used to avoid a sample rate conversion.
   */
  (*output_codec_context)->channels       = OUTPUT_CHANNELS;
  (*output_codec_context)->channel_layout = av_get_default_channel_layout (
    OUTPUT_CHANNELS);
  (*output_codec_context)->sample_rate    = 48000; // Opus need 48000
  (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
  (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;

  /** Open the encoder for the audio stream to use it later. */
  if ((error = avcodec_open2 (*output_codec_context, output_codec, NULL)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not open output codec (error '%s')\n",
             get_error_text (error));
#endif
    goto cleanup;
  }
  return 0;

cleanup:
  av_free (io_ctx);
  return error < 0 ? error : AVERROR_EXIT;
}


/** Initialize one data packet for reading or writing. */
static void
init_packet (AVPacket *packet)
{
  av_init_packet (packet);
  /** Set the packet data and size so that it is recognized as being empty. */
  packet->data = NULL;
  packet->size = 0;
}


/** Initialize one audio frame for reading from the input file */
static int
init_input_frame (AVFrame **frame)
{
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
  *frame = av_frame_alloc ();
#else
  *frame = avcodec_alloc_frame ();
#endif
  if (NULL == *frame)
  {
#if DEBUG
    fprintf (stderr, "Could not allocate input frame\n");
#endif
    return AVERROR (ENOMEM);
  }
  return 0;
}


/**
 * Initialize the audio resampler based on the input and output codec settings.
 * If the input and output sample formats differ, a conversion is required
 * libavresample takes care of this, but requires initialization.
 */
static int
init_resampler (AVCodecContext *input_codec_context,
                AVCodecContext *output_codec_context,
                AVAudioResampleContext  **resample_context)
{
  /**
   * Only initialize the resampler if it is necessary, i.e.,
   * if and only if the sample formats differ.
   */
  if ((input_codec_context->sample_fmt != output_codec_context->sample_fmt) ||
      (input_codec_context->channels != output_codec_context->channels) )
  {
    int error;

    /** Create a resampler context for the conversion. */
    if (! (*resample_context = avresample_alloc_context ()))
    {
#if DEBUG
      fprintf (stderr, "Could not allocate resample context\n");
#endif
      return AVERROR (ENOMEM);
    }


    /**
     * Set the conversion parameters.
     * Default channel layouts based on the number of channels
     * are assumed for simplicity (they are sometimes not detected
     * properly by the demuxer and/or decoder).
     */av_opt_set_int (*resample_context, "in_channel_layout",
                    av_get_default_channel_layout (
                      input_codec_context->channels), 0);
    av_opt_set_int (*resample_context, "out_channel_layout",
                    av_get_default_channel_layout (
                      output_codec_context->channels), 0);
    av_opt_set_int (*resample_context, "in_sample_rate",
                    input_codec_context->sample_rate, 0);
    av_opt_set_int (*resample_context, "out_sample_rate",
                    output_codec_context->sample_rate, 0);
    av_opt_set_int (*resample_context, "in_sample_fmt",
                    input_codec_context->sample_fmt, 0);
    av_opt_set_int (*resample_context, "out_sample_fmt",
                    output_codec_context->sample_fmt, 0);

    /** Open the resampler with the specified parameters. */
    if ((error = avresample_open (*resample_context)) < 0)
    {
#if DEBUG
      fprintf (stderr, "Could not open resample context\n");
#endif
      avresample_free (resample_context);
      return error;
    }
  }
  return 0;
}


/** Initialize a FIFO buffer for the audio samples to be encoded. */
static int
init_fifo (AVAudioFifo **fifo)
{
  /** Create the FIFO buffer based on the specified output sample format. */
  if (! (*fifo = av_audio_fifo_alloc (OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS,
                                      1)))
  {
#if DEBUG
    fprintf (stderr, "Could not allocate FIFO\n");
#endif
    return AVERROR (ENOMEM);
  }
  return 0;
}


/** Write the header of the output file container. */
static int
write_output_file_header (AVFormatContext *output_format_context)
{
  int error;
  if ((error = avformat_write_header (output_format_context, NULL)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not write output file header (error '%s')\n",
             get_error_text (error));
#endif
    return error;
  }
  return 0;
}


/** Decode one audio frame from the input file. */
static int
decode_audio_frame (AVFrame *frame,
                    AVFormatContext *input_format_context,
                    AVCodecContext *input_codec_context, int audio_stream_index,
                    int *data_present, int *finished)
{
  /** Packet used for temporary storage. */
  AVPacket input_packet;
  int error;
  init_packet (&input_packet);

  /** Read one audio frame from the input file into a temporary packet. */
  while (1)
  {
    if ((error = av_read_frame (input_format_context, &input_packet)) < 0)
    {
      /** If we are the the end of the file, flush the decoder below. */
      if (error == AVERROR_EOF)
      {
#if DEBUG
        fprintf (stderr, "EOF in decode_audio\n");
#endif
        *finished = 1;
      }
      else
      {
#if DEBUG
        fprintf (stderr, "Could not read frame (error '%s')\n",
                 get_error_text (error));
#endif
        return error;
      }
    }

    if (input_packet.stream_index == audio_stream_index)
      break;
  }

  /**
   * Decode the audio frame stored in the temporary packet.
   * The input audio stream decoder is used to do this.
   * If we are at the end of the file, pass an empty packet to the decoder
   * to flush it.
   */if ((error = avcodec_decode_audio4 (input_codec_context, frame,
                                      data_present, &input_packet)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not decode frame (error '%s')\n",
             get_error_text (error));
#endif
    av_free_packet (&input_packet);
    return error;
  }

  /**
   * If the decoder has not been flushed completely, we are not finished,
   * so that this function has to be called again.
   */
  if (*finished && *data_present)
    *finished = 0;
  av_free_packet (&input_packet);
  return 0;
}


/**
 * Initialize a temporary storage for the specified number of audio samples.
 * The conversion requires temporary storage due to the different format.
 * The number of audio samples to be allocated is specified in frame_size.
 */
static int
init_converted_samples (uint8_t ***converted_input_samples, int*out_linesize,
                        AVCodecContext *output_codec_context,
                        int frame_size)
{
  int error;

  /**
   * Allocate as many pointers as there are audio channels.
   * Each pointer will later point to the audio samples of the corresponding
   * channels (although it may be NULL for interleaved formats).
   */if (! (*converted_input_samples = calloc (output_codec_context->channels,
                                            sizeof(**converted_input_samples))))
  {
#if DEBUG
    fprintf (stderr, "Could not allocate converted input sample pointers\n");
#endif
    return AVERROR (ENOMEM);
  }

  /**
   * Allocate memory for the samples of all channels in one consecutive
   * block for convenience.
   */
  if ((error = av_samples_alloc (*converted_input_samples, out_linesize,
                                 output_codec_context->channels,
                                 frame_size,
                                 output_codec_context->sample_fmt, 0)) < 0)
  {
#if DEBUG
    fprintf (stderr,
             "Could not allocate converted input samples (error '%s')\n",
             get_error_text (error));
#endif
    av_freep (&(*converted_input_samples)[0]);
    free (*converted_input_samples);
    return error;
  }
  return 0;
}


/**
 * Convert the input audio samples into the output sample format.
 * The conversion happens on a per-frame basis, the size of which is specified
 * by frame_size.
 */
static int
convert_samples (uint8_t **input_data,
                 uint8_t **converted_data, const int in_sample, const int
                 out_sample, const int out_linesize,
                 AVAudioResampleContext  *resample_context)
{
  int error;

  /** Convert the samples using the resampler. */
  if ((error = avresample_convert (resample_context, converted_data,
                                   out_linesize,
                                   out_sample, input_data, 0, in_sample)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not convert input samples (error '%s')\n",
             get_error_text (error));
#endif
    return error;
  }


  /**
   * Perform a sanity check so that the number of converted samples is
   * not greater than the number of samples to be converted.
   * If the sample rates differ, this case has to be handled differently
   */if (avresample_available (resample_context))
  {
#if DEBUG
    fprintf (stderr, "%i Converted samples left over\n",avresample_available (
               resample_context));
#endif
  }


  return 0;
}


/** Add converted input audio samples to the FIFO buffer for later processing. */
static int
add_samples_to_fifo (AVAudioFifo *fifo,
                     uint8_t **converted_input_samples,
                     const int frame_size)
{
  int error;

  /**
   * Make the FIFO as large as it needs to be to hold both,
   * the old and the new samples.
   */
  if ((error = av_audio_fifo_realloc (fifo, av_audio_fifo_size (fifo)
                                      + frame_size)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not reallocate FIFO\n");
#endif
    return error;
  }

  /** Store the new samples in the FIFO buffer. */
  if (av_audio_fifo_write (fifo, (void **) converted_input_samples,
                           frame_size) < frame_size)
  {
#if DEBUG
    fprintf (stderr, "Could not write data to FIFO\n");
#endif
    return AVERROR_EXIT;
  }
  return 0;
}


/**
 * Read one audio frame from the input file, decodes, converts and stores
 * it in the FIFO buffer.
 */
static int
read_decode_convert_and_store (AVAudioFifo *fifo,
                               AVFormatContext *input_format_context,
                               AVCodecContext *input_codec_context,
                               AVCodecContext *output_codec_context,
                               AVAudioResampleContext  *resampler_context, int
                               audio_stream_index,
                               int *finished)
{
  /** Temporary storage of the input samples of the frame read from the file. */
  AVFrame *input_frame = NULL;
  /** Temporary storage for the converted input samples. */
  uint8_t **converted_input_samples = NULL;
  int data_present;
  int ret = AVERROR_EXIT;

  /** Initialize temporary storage for one input frame. */
  if (init_input_frame (&input_frame))
  {
#if DEBUG
    fprintf (stderr, "Failed at init frame\n");
#endif
    goto cleanup;

  }
  /** Decode one frame worth of audio samples. */
  if (decode_audio_frame (input_frame, input_format_context,
                          input_codec_context, audio_stream_index,
                          &data_present,  finished))
  {
#if DEBUG
    fprintf (stderr, "Failed at decode audio\n");
#endif

    goto cleanup;

  }
  /**
   * If we are at the end of the file and there are no more samples
   * in the decoder which are delayed, we are actually finished.
   * This must not be treated as an error.
   */if (*finished && ! data_present)
  {
    ret = 0;
#if DEBUG
    fprintf (stderr, "Failed at finished or no data\n");
#endif
    goto cleanup;
  }
  /** If there is decoded data, convert and store it */
  if (data_present)
  {
    int out_linesize;
    // FIX ME: I'm losing samples, but can't get it to work.
    int out_samples = avresample_available (resampler_context)
                      + avresample_get_delay (resampler_context)
                      + input_frame->nb_samples;


    // fprintf(stderr, "Input nbsamples %i out_samples: %i \n",input_frame->nb_samples,out_samples);

    /** Initialize the temporary storage for the converted input samples. */
    if (init_converted_samples (&converted_input_samples, &out_linesize,
                                output_codec_context,
                                out_samples))
    {
#if DEBUG
      fprintf (stderr, "Failed at init_converted_samples\n");
#endif
      goto cleanup;
    }

    /**
     * Convert the input samples to the desired output sample format.
     * This requires a temporary storage provided by converted_input_samples.
     */
    if (convert_samples (input_frame->extended_data, converted_input_samples,
                         input_frame->nb_samples, out_samples, out_linesize,
                         resampler_context))
    {


#if DEBUG
      fprintf (stderr, "Failed at convert_samples, input frame %i \n",
               input_frame->nb_samples);
#endif
      goto cleanup;
    }
    /** Add the converted input samples to the FIFO buffer for later processing. */
    if (add_samples_to_fifo (fifo, converted_input_samples,
                             out_samples))
    {
#if DEBUG
      fprintf (stderr, "Failed at add_samples_to_fifo\n");
#endif
      goto cleanup;
    }
    ret = 0;
  }
  ret = 0;

cleanup:
  if (converted_input_samples)
  {
    av_freep (&converted_input_samples[0]);
    free (converted_input_samples);
  }
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
  av_frame_free (&input_frame);
#else
  avcodec_free_frame (&input_frame);
#endif

  return ret;
}


/**
 * Initialize one input frame for writing to the output file.
 * The frame will be exactly frame_size samples large.
 */
static int
init_output_frame (AVFrame **frame,
                   AVCodecContext *output_codec_context,
                   int frame_size)
{
  int error;

  /** Create a new frame to store the audio samples. */
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
  *frame = av_frame_alloc ();
#else
  *frame = avcodec_alloc_frame ();
#endif
  if (NULL == *frame)
  {
#if DEBUG
    fprintf (stderr, "Could not allocate output frame\n");
#endif
    return AVERROR_EXIT;
  }

  /**
   * Set the frame's parameters, especially its size and format.
   * av_frame_get_buffer needs this to allocate memory for the
   * audio samples of the frame.
   * Default channel layouts based on the number of channels
   * are assumed for simplicity.
   */(*frame)->nb_samples  = frame_size;
  (*frame)->channel_layout = output_codec_context->channel_layout;
  (*frame)->format         = output_codec_context->sample_fmt;
  (*frame)->sample_rate    = output_codec_context->sample_rate;


  // fprintf(stderr, "%i %i  \n",frame_size , (*frame)->format,(*frame)->sample_rate);

  /**
   * Allocate the samples of the created frame. This call will make
   * sure that the audio frame can hold as many samples as specified.
   */
  if ((error = av_frame_get_buffer (*frame, 0)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could allocate output frame samples (error '%s')\n",
             get_error_text (error));
#endif
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
    av_frame_free (frame);
#else
    avcodec_free_frame (frame);
#endif
    return error;
  }

  return 0;
}


/** Encode one frame worth of audio to the output file. */
static int
encode_audio_frame (AVFrame *frame,
                    AVFormatContext *output_format_context,
                    AVCodecContext *output_codec_context,
                    int *data_present)
{
  /** Packet used for temporary storage. */
  AVPacket output_packet;
  int error;
  init_packet (&output_packet);

  /**
   * Encode the audio frame and store it in the temporary packet.
   * The output audio stream encoder is used to do this.
   */
  if ((error = avcodec_encode_audio2 (output_codec_context, &output_packet,
                                      frame, data_present)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not encode frame (error '%s')\n",
             get_error_text (error));
#endif
    av_free_packet (&output_packet);
    return error;
  }

  /** Write one audio frame from the temporary packet to the output file. */
  if (*data_present)
  {
    if ((error = av_write_frame (output_format_context, &output_packet)) < 0)
    {
#if DEBUG
      fprintf (stderr, "Could not write frame (error '%s')\n",
               get_error_text (error));
#endif

      av_free_packet (&output_packet);
      return error;
    }

    av_free_packet (&output_packet);
  }

  return 0;
}


/**
 * Load one audio frame from the FIFO buffer, encode and write it to the
 * output file.
 */
static int
load_encode_and_write (AVAudioFifo *fifo,
                       AVFormatContext *output_format_context,
                       AVCodecContext *output_codec_context)
{
  /** Temporary storage of the output samples of the frame written to the file. */
  AVFrame *output_frame;
  /**
   * Use the maximum number of possible samples per frame.
   * If there is less than the maximum possible frame size in the FIFO
   * buffer use this number. Otherwise, use the maximum possible frame size
   */const int frame_size = FFMIN (av_audio_fifo_size (fifo),
                                output_codec_context->frame_size);
  int data_written;

  /** Initialize temporary storage for one output frame. */
  if (init_output_frame (&output_frame, output_codec_context, frame_size))
    return AVERROR_EXIT;

  /**
   * Read as many samples from the FIFO buffer as required to fill the frame.
   * The samples are stored in the frame temporarily.
   */
  if (av_audio_fifo_read (fifo, (void **) output_frame->data, frame_size) <
      frame_size)
  {
#if DEBUG
    fprintf (stderr, "Could not read data from FIFO\n");
#endif
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
    av_frame_free (&output_frame);
#else
    avcodec_free_frame (&output_frame);
#endif
    return AVERROR_EXIT;
  }

  /** Encode one frame worth of audio samples. */
  if (encode_audio_frame (output_frame, output_format_context,
                          output_codec_context, &data_written))
  {
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
    av_frame_free (&output_frame);
#else
    avcodec_free_frame (&output_frame);
#endif
    return AVERROR_EXIT;
  }
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
  av_frame_free (&output_frame);
#else
  avcodec_free_frame (&output_frame);
#endif
  return 0;
}


/** Write the trailer of the output file container. */
static int
write_output_file_trailer (AVFormatContext *output_format_context)
{
  int error;
  if ((error = av_write_trailer (output_format_context)) < 0)
  {
#if DEBUG
    fprintf (stderr, "Could not write output file trailer (error '%s')\n",
             get_error_text (error));
#endif
    return error;
  }
  return 0;
}


#define ENUM_CODEC_ID enum AVCodecID


/**
 * Perform the audio snippet extraction
 *
 * @param ec extraction context to use
 */
static void
extract_audio (struct EXTRACTOR_ExtractContext *ec)
{
  AVIOContext *io_ctx;
  struct AVFormatContext *format_ctx;
  AVCodecContext *codec_ctx;
  AVFormatContext *output_format_context = NULL;
  AVCodec *codec;
  AVDictionary *options;
  AVFrame *frame;
  AVCodecContext*output_codec_context = NULL;
  AVAudioResampleContext  *resample_context = NULL;
  AVAudioFifo *fifo = NULL;

  int audio_stream_index;
  int i;
  int err;
  int duration;
  unsigned char *iob;


  totalSize = 0;
  if (NULL == (iob = av_malloc (16 * 1024)))
    return;
  if (NULL == (io_ctx = avio_alloc_context (iob,
                                            16 * 1024,
                                            0, ec,
                                            &read_cb,
                                            NULL /* no writing */,
                                            &seek_cb)))
  {
    av_free (iob);
    return;
  }
  if (NULL == (format_ctx = avformat_alloc_context ()))
  {
    av_free (io_ctx);
    return;
  }
  format_ctx->pb = io_ctx;
  options = NULL;
  if (0 != avformat_open_input (&format_ctx, "<no file>", NULL, &options))
  {
    av_free (io_ctx);
    return;
  }
  av_dict_free (&options);
  if (0 > avformat_find_stream_info (format_ctx, NULL))
  {
#if DEBUG
    fprintf (stderr,
             "Failed to read stream info\n");
#endif
    avformat_close_input (&format_ctx);
    av_free (io_ctx);
    return;
  }
  codec = NULL;
  codec_ctx = NULL;
  audio_stream_index = -1;
  for (i = 0; i<format_ctx->nb_streams; i++)
  {
    codec_ctx = format_ctx->streams[i]->codec;
    if (AVMEDIA_TYPE_AUDIO != codec_ctx->codec_type)
      continue;
    if (NULL == (codec = avcodec_find_decoder (codec_ctx->codec_id)))
      continue;
    options = NULL;
    if (0 != (err = avcodec_open2 (codec_ctx, codec, &options)))
    {
      codec = NULL;
      continue;
    }
    av_dict_free (&options);
    audio_stream_index = i;
    break;
  }
  if ( (-1 == audio_stream_index) ||
       (0 == codec_ctx->channels) )
  {
#if DEBUG
    fprintf (stderr,
             "No audio streams or no suitable codec found\n");
#endif
    if (NULL != codec)
      avcodec_close (codec_ctx);
    avformat_close_input (&format_ctx);
    av_free (io_ctx);
    return;
  }

#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT (55,28,1)
  frame = av_frame_alloc ();
#else
  frame = avcodec_alloc_frame ();
#endif
  if (NULL == frame)
  {
#if DEBUG
    fprintf (stderr,
             "Failed to allocate frame\n");
#endif
    avcodec_close (codec_ctx);
    avformat_close_input (&format_ctx);
    av_free (io_ctx);
    return;
  }


  if (! (buffer = malloc (HARD_LIMIT_SIZE)))
    goto cleanup;


  /** Open the output file for writing. */
  if (open_output_file (codec_ctx,
                        &output_format_context,
                        &output_codec_context))
    goto cleanup;
  /** Initialize the resampler to be able to convert audio sample formats. */
  if (init_resampler (codec_ctx,
                      output_codec_context,
                      &resample_context))
    goto cleanup;
  /** Initialize the FIFO buffer to store audio samples to be encoded. */
  if (init_fifo (&fifo))
    goto cleanup;

  /** Write the header of the output file container. */
  if (write_output_file_header (output_format_context))
    goto cleanup;


  if (format_ctx->duration == AV_NOPTS_VALUE)
  {
    duration = -1;
#if DEBUG
    fprintf (stderr,
             "Duration unknown\n");
#endif
  }
  else
  {
#if DEBUG
    duration = format_ctx->duration;
    fprintf (stderr,
             "Duration: %lld\n",
             format_ctx->duration);
#endif
  }

  /* if duration is known, seek to first tried,
   * else use 10 sec into stream */

  if (-1 != duration)
    err = av_seek_frame (format_ctx, -1, (duration / 3), 0);
  else
    err = av_seek_frame (format_ctx, -1, 10 * AV_TIME_BASE, 0);


  if (err >= 0)
    avcodec_flush_buffers (codec_ctx);


  /**
   * Loop as long as we have input samples to read or output samples
   * to write; abort as soon as we have neither.
   */
  while (1)
  {
    /** Use the encoder's desired frame size for processing. */
    const int output_frame_size = output_codec_context->frame_size;
    int finished                = 0;

    /**
     * Make sure that there is one frame worth of samples in the FIFO
     * buffer so that the encoder can do its work.
     * Since the decoder's and the encoder's frame size may differ, we
     * need to FIFO buffer to store as many frames worth of input samples
     * that they make up at least one frame worth of output samples.
     */while ((av_audio_fifo_size (fifo) < output_frame_size))
    {
      /**
       * Decode one frame worth of audio samples, convert it to the
       * output sample format and put it into the FIFO buffer.
       */
      if (read_decode_convert_and_store (fifo,
                                         format_ctx,
                                         codec_ctx,
                                         output_codec_context,
                                         resample_context,
                                         audio_stream_index,
                                         &finished))
      {
        goto cleanup;
      }

      /**
       * If we are at the end of the input file, we continue
       * encoding the remaining audio samples to the output file.
       */
      if (finished)
        break;
    }

    /* Already over our limit*/
    if (totalSize >= MAX_SIZE)
      finished = 1;

    /**
     * If we have enough samples for the encoder, we encode them.
     * At the end of the file, we pass the remaining samples to
     * the encoder.
     */while (av_audio_fifo_size (fifo) >= output_frame_size ||
           (finished && av_audio_fifo_size (fifo) > 0))
    {
      /**
       * Take one frame worth of audio samples from the FIFO buffer,
       * encode it and write it to the output file.
       */
      if (load_encode_and_write (fifo,
                                 output_format_context,
                                 output_codec_context))
        goto cleanup;
    }
    /**
     * If we are at the end of the input file and have encoded
     * all remaining samples, we can exit this loop and finish.
     */
    if (finished)
    {
      int data_written;
      /** Flush the encoder as it may have delayed frames. */
      do {
        encode_audio_frame (NULL,
                            output_format_context,
                            output_codec_context,
                            &data_written);
      } while (data_written);
      break;
    }
  }

  /** Write the trailer of the output file container. */
  if (write_output_file_trailer (output_format_context))
    goto cleanup;
  ec->proc (ec->cls,
            "previewopus",
            EXTRACTOR_METATYPE_AUDIO_PREVIEW,
            EXTRACTOR_METAFORMAT_BINARY,
            "audio/opus",
            buffer,
            totalSize);

#if OUTPUT_FILE
  {
    FILE *f;

    f = fopen ("example.opus", "wb");
    if (! f)
    {
      fprintf (stderr, "Could not open %s\n", "file");
      exit (1);
    }
    fwrite (buffer, 1, totalSize, f);
    fclose (f);
  }
#endif

cleanup:
  av_free (frame);
  free (buffer);

  if (fifo)
    av_audio_fifo_free (fifo);
  if (resample_context)
  {
    avresample_close (resample_context);
    avresample_free (&resample_context);
  }
  if (output_codec_context)
    avcodec_close (output_codec_context);

  if (codec_ctx)
    avcodec_close (codec_ctx);
  if (format_ctx)
    avformat_close_input (&format_ctx);
  av_free (io_ctx);
}


/**
 * Main method for the opus-preview plugin.
 *
 * @param ec extraction context
 */
void
EXTRACTOR_previewopus_extract_method (struct EXTRACTOR_ExtractContext *ec)
{
  ssize_t iret;
  void *data;


  if (-1 == (iret = ec->read (ec->cls,
                              &data,
                              16 * 1024)))
    return;

  if (0 != ec->seek (ec->cls, 0, SEEK_SET))
    return;

  extract_audio (ec);
}


/**
 * Log callback.  Does nothing.
 *
 * @param ptr NULL
 * @param level log level
 * @param format format string
 * @param ap arguments for format
 */
static void
previewopus_av_log_callback (void*ptr,
                             int level,
                             const char *format,
                             va_list ap)
{
#if DEBUG
  vfprintf (stderr, format, ap);
#endif
}


/**
 * Initialize av-libs
 */
void __attribute__ ((constructor))
previewopus_lib_init (void)
{
  av_log_set_callback (&previewopus_av_log_callback);
  av_register_all ();

}


/**
 * Destructor for the library, cleans up.
 */
void __attribute__ ((destructor))
previewopus_ltdl_fini ()
{

}


/* end of previewopus_extractor.c */