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
1279
1280
1281
1282
1283
1284
1285
1286
1287
1288
1289
1290
1291
1292
1293
1294
1295
1296
1297
1298
1299
1300
1301
1302
1303
1304
1305
1306
1307
1308
1309
1310
1311
1312
1313
1314
1315
1316
1317
1318
1319
1320
1321
1322
1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
1338
1339
1340
1341
1342
1343
1344
1345
1346
1347
1348
1349
1350
1351
1352
1353
1354
1355
1356
1357
1358
1359
1360
1361
1362
1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
1373
1374
1375
1376
1377
1378
1379
1380
1381
1382
1383
1384
1385
1386
1387
1388
1389
1390
1391
1392
1393
1394
1395
1396
1397
1398
1399
1400
1401
1402
1403
1404
1405
1406
1407
1408
1409
1410
1411
1412
1413
1414
1415
|
#define USE_G72X
//#define USE_LIBAC3
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
extern int verbose; // defined in mplayer.c
#include "stream.h"
#include "demuxer.h"
#include "codec-cfg.h"
#include "stheader.h"
#include "dec_audio.h"
#include "roqav.h"
//==========================================================================
#include "libao2/afmt.h"
#include "dll_init.h"
#include "mp3lib/mp3.h"
#ifdef USE_LIBAC3
#include "libac3/ac3.h"
#endif
#include "liba52/a52.h"
#include "liba52/mm_accel.h"
static sample_t * a52_samples;
static a52_state_t a52_state;
static uint32_t a52_accel=0;
static uint32_t a52_flags=0;
#ifdef USE_G72X
#include "g72x/g72x.h"
static G72x_DATA g72x_data;
#endif
#include "alaw.h"
#include "xa/xa_gsm.h"
#include "ac3-iec958.h"
#include "adpcm.h"
#include "cpudetect.h"
/* used for ac3surround decoder - set using -channels option */
int audio_output_channels = 2;
#ifdef USE_FAKE_MONO
int fakemono=0;
#endif
#ifdef USE_DIRECTSHOW
#include "loader/dshow/DS_AudioDecoder.h"
static DS_AudioDecoder* ds_adec=NULL;
#endif
#ifdef HAVE_OGGVORBIS
/* XXX is math.h really needed? - atmos */
#include <math.h>
#include <vorbis/codec.h>
typedef struct ov_struct_st {
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the bitstream user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
} ov_struct_t;
#endif
#ifdef USE_LIBAVCODEC
#ifdef USE_LIBAVCODEC_SO
#include <libffmpeg/avcodec.h>
#else
#include "libavcodec/avcodec.h"
#endif
static AVCodec *lavc_codec=NULL;
static AVCodecContext lavc_context;
extern int avcodec_inited;
#endif
#ifdef USE_LIBMAD
#include <mad.h>
#define MAD_SINGLE_BUFFER_SIZE 8192
#define MAD_TOTAL_BUFFER_SIZE ((MAD_SINGLE_BUFFER_SIZE)*3)
static struct mad_stream mad_stream;
static struct mad_frame mad_frame;
static struct mad_synth mad_synth;
static char* mad_in_buffer = 0; /* base pointer of buffer */
// ensure buffer is filled with some data
static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length)
{
if(sh_audio->a_in_buffer_len < length) {
int len = demux_read_data(sh_audio->ds, sh_audio->a_in_buffer+sh_audio->a_in_buffer_len, length-sh_audio->a_in_buffer_len);
sh_audio->a_in_buffer_len += len;
// printf("mad_prepare_buffer: read %d bytes\n", len);
}
}
static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms)
{
/* rotate buffer while possible, in order to reduce the overhead of endless memcpy */
int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer;
if((unsigned long)(sh_audio->a_in_buffer) - (unsigned long)mad_in_buffer <
(MAD_TOTAL_BUFFER_SIZE - MAD_SINGLE_BUFFER_SIZE - delta)) {
sh_audio->a_in_buffer += delta;
sh_audio->a_in_buffer_len -= delta;
} else {
sh_audio->a_in_buffer = mad_in_buffer;
sh_audio->a_in_buffer_len -= delta;
memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len);
}
}
static inline
signed short mad_scale(mad_fixed_t sample)
{
/* round */
sample += (1L << (MAD_F_FRACBITS - 16));
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
static void mad_sync(sh_audio_t* sh_audio, struct mad_stream* ms)
{
int len;
#if 1
int skipped = 0;
// printf("buffer len: %d\n", sh_audio->a_in_buffer_len);
while(sh_audio->a_in_buffer_len - skipped)
{
len = mp_decode_mp3_header(sh_audio->a_in_buffer+skipped);
if (len != -1)
{
// printf("Frame len=%d\n", len);
break;
}
else
skipped++;
}
if (skipped)
{
printf("Audio synced, skipped bytes: %d\n", skipped);
// ms->skiplen += skipped;
// printf("skiplen: %d (skipped: %d)\n", ms->skiplen, skipped);
// if (sh_audio->a_in_buffer_len - skipped < MAD_BUFFER_GUARD)
// printf("Mad reports: too small buffer\n");
// mad_stream_buffer(ms, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len-skipped);
// mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_len-skipped);
/* move frame to the beginning of the buffer and fill up to a_in_buffer_size */
sh_audio->a_in_buffer_len -= skipped;
memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len);
mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_size);
mad_stream_buffer(ms, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
// printf("bufflen: %d\n", sh_audio->a_in_buffer_len);
// len = mp_decode_mp3_header(sh_audio->a_in_buffer);
// printf("len: %d\n", len);
ms->md_len = len;
}
#else
len = mad_stream_sync(&ms);
if (len == -1)
{
printf("Mad sync failed\n");
}
#endif
}
static void mad_print_error(struct mad_stream *mad_stream)
{
printf("error (0x%x): ", mad_stream->error);
switch(mad_stream->error)
{
case MAD_ERROR_BUFLEN: printf("buffer too small"); break;
case MAD_ERROR_BUFPTR: printf("invalid buffer pointer"); break;
case MAD_ERROR_NOMEM: printf("not enought memory"); break;
case MAD_ERROR_LOSTSYNC: printf("lost sync"); break;
case MAD_ERROR_BADLAYER: printf("bad layer"); break;
case MAD_ERROR_BADBITRATE: printf("bad bitrate"); break;
case MAD_ERROR_BADSAMPLERATE: printf("bad samplerate"); break;
case MAD_ERROR_BADEMPHASIS: printf("bad emphasis"); break;
case MAD_ERROR_BADCRC: printf("bad crc"); break;
case MAD_ERROR_BADBITALLOC: printf("forbidden bit alloc val"); break;
case MAD_ERROR_BADSCALEFACTOR: printf("bad scalefactor index"); break;
case MAD_ERROR_BADFRAMELEN: printf("bad frame length"); break;
case MAD_ERROR_BADBIGVALUES: printf("bad bigvalues count"); break;
case MAD_ERROR_BADBLOCKTYPE: printf("reserved blocktype"); break;
case MAD_ERROR_BADSCFSI: printf("bad scalefactor selinfo"); break;
case MAD_ERROR_BADDATAPTR: printf("bad maindatabegin ptr"); break;
case MAD_ERROR_BADPART3LEN: printf("bad audio data len"); break;
case MAD_ERROR_BADHUFFTABLE: printf("bad huffman table sel"); break;
case MAD_ERROR_BADHUFFDATA: printf("huffman data overrun"); break;
case MAD_ERROR_BADSTEREO: printf("incomp. blocktype for JS"); break;
default:
printf("unknown error");
}
printf("\n");
}
#endif
static int a52_fillbuff(sh_audio_t *sh_audio){
int length=0;
int flags=0;
int sample_rate=0;
int bit_rate=0;
sh_audio->a_in_buffer_len=0;
// sync frame:
while(1){
while(sh_audio->a_in_buffer_len<7){
int c=demux_getc(sh_audio->ds);
if(c<0) return -1; // EOF
sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
}
length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
if(length>=7 && length<=3840) break; // we're done.
// bad file => resync
memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6);
--sh_audio->a_in_buffer_len;
}
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
sh_audio->samplerate=sample_rate;
sh_audio->i_bps=bit_rate/8;
demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+7,length-7);
if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
return length;
}
// returns: number of available channels
static int a52_printinfo(sh_audio_t *sh_audio){
int flags, sample_rate, bit_rate;
char* mode="unknown";
int channels=0;
a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
switch(flags&A52_CHANNEL_MASK){
case A52_CHANNEL: mode="channel"; channels=2; break;
case A52_MONO: mode="mono"; channels=1; break;
case A52_STEREO: mode="stereo"; channels=2; break;
case A52_3F: mode="3f";channels=3;break;
case A52_2F1R: mode="2f+1r";channels=3;break;
case A52_3F1R: mode="3f+1r";channels=4;break;
case A52_2F2R: mode="2f+2r";channels=4;break;
case A52_3F2R: mode="3f+2r";channels=5;break;
case A52_CHANNEL1: mode="channel1"; channels=2; break;
case A52_CHANNEL2: mode="channel2"; channels=2; break;
case A52_DOLBY: mode="dolby"; channels=2; break;
}
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
channels, (flags&A52_LFE)?1:0,
mode, (flags&A52_LFE)?"+lfe":"",
sample_rate, bit_rate*0.001f);
return (flags&A52_LFE) ? (channels+1) : channels;
}
int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen);
static sh_audio_t* dec_audio_sh=NULL;
#ifdef USE_LIBAC3
// AC3 decoder buffer callback:
static void ac3_fill_buffer(uint8_t **start,uint8_t **end){
int len=ds_get_packet(dec_audio_sh->ds,start);
//printf("<ac3:%d>\n",len);
if(len<0)
*start = *end = NULL;
else
*end = *start + len;
}
#endif
// MP3 decoder buffer callback:
int mplayer_audio_read(char *buf,int size){
int len;
len=demux_read_data(dec_audio_sh->ds,buf,size);
return len;
}
int init_audio(sh_audio_t *sh_audio){
int driver=sh_audio->codec->driver;
sh_audio->samplesize=2;
#ifdef WORDS_BIGENDIAN
sh_audio->sample_format=AFMT_S16_BE;
#else
sh_audio->sample_format=AFMT_S16_LE;
#endif
sh_audio->samplerate=0;
//sh_audio->pcm_bswap=0;
sh_audio->o_bps=0;
sh_audio->a_buffer_size=0;
sh_audio->a_buffer=NULL;
sh_audio->a_in_buffer_len=0;
// setup required min. in/out buffer size:
sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM
switch(driver){
case AFM_ACM:
#ifndef USE_WIN32DLL
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport);
driver=0;
#else
// Win32 ACM audio codec:
if(init_acm_audio_codec(sh_audio)){
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
sh_audio->channels=sh_audio->o_wf.nChannels;
sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec;
// if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384;
// sh_audio->a_buffer_size=sh_audio->audio_out_minsize;
// if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST)
// sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST;
} else {
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror);
driver=0;
}
#endif
break;
case AFM_DSHOW:
#ifndef USE_DIRECTSHOW
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio);
driver=0;
#else
// Win32 DShow audio codec:
// printf("DShow_audio: channs=%d rate=%d\n",sh_audio->channels,sh_audio->samplerate);
if(!(ds_adec=DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf))){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll);
driver=0;
} else {
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign;
if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192;
sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
sh_audio->a_in_buffer_len=0;
sh_audio->audio_out_minsize=16384;
}
#endif
break;
case AFM_VORBIS:
#ifndef HAVE_OGGVORBIS
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis);
driver=0;
#else
/* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#endif
break;
case AFM_PCM:
case AFM_DVDPCM:
case AFM_ALAW:
// PCM, aLaw
sh_audio->audio_out_minsize=2048;
break;
case AFM_AC3:
case AFM_A52:
// Dolby AC3 audio:
// however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame
sh_audio->audio_out_minsize=audio_output_channels*2*256*6;
break;
case AFM_HWAC3:
// Dolby AC3 audio:
sh_audio->audio_out_minsize=4*256*6;
// sh_audio->sample_format = AFMT_AC3;
// sh_audio->sample_format = AFMT_S16_LE;
sh_audio->channels=2;
break;
case AFM_GSM:
// MS-GSM audio codec:
sh_audio->audio_out_minsize=4*320;
break;
case AFM_IMAADPCM:
sh_audio->audio_out_minsize=4096;
sh_audio->ds->ss_div=IMA_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels;
break;
case AFM_MSADPCM:
sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign * 8;
sh_audio->ds->ss_div = MS_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
break;
case AFM_DK4ADPCM:
sh_audio->audio_out_minsize=DK4_ADPCM_SAMPLES_PER_BLOCK * 4;
sh_audio->ds->ss_div=DK4_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=sh_audio->wf->nBlockAlign;
break;
case AFM_DK3ADPCM:
sh_audio->audio_out_minsize=DK3_ADPCM_SAMPLES_PER_BLOCK * 4;
sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE;
break;
case AFM_ROQAUDIO:
// minsize was stored in wf->nBlockAlign by the RoQ demuxer
sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign;
sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK;
sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE;
sh_audio->context = roq_decode_audio_init();
break;
case AFM_MPEG:
// MPEG Audio:
sh_audio->audio_out_minsize=4608;
break;
#ifdef USE_G72X
case AFM_G72X:
// g72x_reader_init(&g72x_data,G723_16_BITS_PER_SAMPLE);
g72x_reader_init(&g72x_data,G723_24_BITS_PER_SAMPLE);
// g72x_reader_init(&g72x_data,G721_32_BITS_PER_SAMPLE);
// g72x_reader_init(&g72x_data,G721_40_BITS_PER_SAMPLE);
sh_audio->audio_out_minsize=g72x_data.samplesperblock*4;
break;
#endif
case AFM_FFMPEG:
#ifndef USE_LIBAVCODEC
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport);
return 0;
#else
// FFmpeg Audio:
sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
break;
#endif
#ifdef USE_LIBMAD
case AFM_MAD:
printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__);
sh_audio->audio_out_minsize=4608;
if(sh_audio->audio_in_minsize<MAD_SINGLE_BUFFER_SIZE) sh_audio->audio_in_minsize=MAD_SINGLE_BUFFER_SIZE;
sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
mad_in_buffer = sh_audio->a_in_buffer = malloc(MAD_TOTAL_BUFFER_SIZE);
sh_audio->a_in_buffer_len=0;
break;
#endif
}
if(!driver) return 0;
// allocate audio out buffer:
sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc.
mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n",
sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size);
sh_audio->a_buffer=malloc(sh_audio->a_buffer_size);
if(!sh_audio->a_buffer){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf);
return 0;
}
memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size);
sh_audio->a_buffer_len=0;
switch(driver){
#ifdef USE_WIN32DLL
case AFM_ACM: {
int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size);
if(ret<0){
mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret);
driver=0;
}
sh_audio->a_buffer_len=ret;
break;
}
#endif
case AFM_PCM: {
// AVI PCM Audio:
WAVEFORMATEX *h=sh_audio->wf;
sh_audio->i_bps=h->nAvgBytesPerSec;
sh_audio->channels=h->nChannels;
sh_audio->samplerate=h->nSamplesPerSec;
sh_audio->samplesize=(h->wBitsPerSample+7)/8;
switch(sh_audio->format){ // hardware formats:
case 0x6: sh_audio->sample_format=AFMT_A_LAW;break;
case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break;
case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break;
case 0x50: sh_audio->sample_format=AFMT_MPEG;break;
case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break;
// case 0x2000: sh_audio->sample_format=AFMT_AC3;
default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8;
}
break;
}
case AFM_DVDPCM: {
// DVD PCM Audio:
sh_audio->channels=2;
sh_audio->samplerate=48000;
sh_audio->i_bps=2*2*48000;
// sh_audio->pcm_bswap=1;
break;
}
case AFM_AC3: {
#ifndef USE_LIBAC3
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"WARNING: libac3 support is disabled. (hint: upgrade codecs.conf)\n");
driver=0;
#else
// Dolby AC3 audio:
dec_audio_sh=sh_audio; // save sh_audio for the callback:
ac3_config.fill_buffer_callback = ac3_fill_buffer;
ac3_config.num_output_ch = audio_output_channels;
ac3_config.flags = 0;
if(gCpuCaps.hasMMX){
ac3_config.flags |= AC3_MMX_ENABLE;
}
if(gCpuCaps.has3DNow){
ac3_config.flags |= AC3_3DNOW_ENABLE;
}
ac3_init();
sh_audio->ac3_frame = ac3_decode_frame();
if(sh_audio->ac3_frame){
ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame;
sh_audio->samplerate=fr->sampling_rate;
sh_audio->channels=ac3_config.num_output_ch;
// 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples
//sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256);
sh_audio->i_bps=fr->bit_rate*(1000/8);
} else {
driver=0; // bad frame -> disable audio
}
#endif
break;
}
case AFM_A52: {
sample_t level=1, bias=384;
int flags=0;
// Dolby AC3 audio:
if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
a52_samples=a52_init (a52_accel);
if (a52_samples == NULL) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
driver=0;break;
}
sh_audio->a_in_buffer_size=3840;
sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
sh_audio->a_in_buffer_len=0;
if(a52_fillbuff(sh_audio)<0){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
driver=0;break;
}
// 'a52 cannot upmix' hotfix:
a52_printinfo(sh_audio);
// if(audio_output_channels<sh_audio->channels)
// sh_audio->channels=audio_output_channels;
// channels setup:
sh_audio->channels=audio_output_channels;
while(sh_audio->channels>0){
switch(sh_audio->channels){
case 1: a52_flags=A52_MONO; break;
// case 2: a52_flags=A52_STEREO; break;
case 2: a52_flags=A52_DOLBY; break;
// case 3: a52_flags=A52_3F; break;
case 3: a52_flags=A52_2F1R; break;
case 4: a52_flags=A52_2F2R; break; // 2+2
case 5: a52_flags=A52_3F2R; break;
case 6: a52_flags=A52_3F2R|A52_LFE; break; // 5.1
}
// test:
flags=a52_flags|A52_ADJUST_LEVEL;
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
driver=0;break;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
// frame decoded, let's init resampler:
if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
--sh_audio->channels; // try to decrease no. of channels
}
if(sh_audio->channels<=0){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
driver=0;break;
}
break;
}
case AFM_HWAC3: {
// Dolby AC3 passthrough:
a52_samples=a52_init (a52_accel);
if (a52_samples == NULL) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
driver=0;break;
}
sh_audio->a_in_buffer_size=3840;
sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
sh_audio->a_in_buffer_len=0;
if(a52_fillbuff(sh_audio)<0) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
driver=0;break;
}
//sh_audio->samplerate=ai.samplerate; // SET by a52_fillbuff()
//sh_audio->samplesize=ai.framesize;
//sh_audio->i_bps=ai.bitrate*(1000/8); // SET by a52_fillbuff()
//sh_audio->ac3_frame=malloc(6144);
//sh_audio->o_bps=sh_audio->i_bps; // XXX FIXME!!! XXX
// o_bps is calculated from samplesize*channels*samplerate
// a single ac3 frame is always translated to 6144 byte packet. (zero padding)
sh_audio->channels=2;
sh_audio->samplesize=2; // 2*2*(6*256) = 6144 (very TRICKY!)
break;
}
case AFM_ALAW: {
// aLaw audio codec:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate;
break;
}
#ifdef USE_G72X
case AFM_G72X: {
// GSM 723 audio codec:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=(sh_audio->samplerate/g72x_data.samplesperblock)*g72x_data.blocksize;
break;
}
#endif
#ifdef USE_LIBAVCODEC
case AFM_FFMPEG: {
int x;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
if(!avcodec_inited){
avcodec_init();
avcodec_register_all();
avcodec_inited=1;
}
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
return 0;
}
memset(&lavc_context, 0, sizeof(lavc_context));
/* open it */
if (avcodec_open(&lavc_context, lavc_codec) < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
// Decode at least 1 byte: (to get header filled)
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
if(x>0) sh_audio->a_buffer_len=x;
#if 1
sh_audio->channels=lavc_context.channels;
sh_audio->samplerate=lavc_context.sample_rate;
sh_audio->i_bps=lavc_context.bit_rate/8;
#else
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
#endif
break;
}
#endif
case AFM_GSM: {
// MS-GSM audio codec:
GSM_Init();
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
// decodes 65 byte -> 320 short
// 1 sec: sh_audio->channels*sh_audio->samplerate samples
// 1 frame: 320 samples
sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320; // 1:10
break;
}
case AFM_IMAADPCM:
// IMA-ADPCM 4:1 audio codec:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
// decodes 34 byte -> 64 short
sh_audio->i_bps=IMA_ADPCM_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/IMA_ADPCM_SAMPLES_PER_BLOCK; // 1:4
break;
case AFM_MSADPCM:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps = sh_audio->wf->nBlockAlign *
(sh_audio->channels*sh_audio->samplerate) / MS_ADPCM_SAMPLES_PER_BLOCK;
break;
case AFM_DK4ADPCM:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps = sh_audio->wf->nBlockAlign *
(sh_audio->channels*sh_audio->samplerate) / DK4_ADPCM_SAMPLES_PER_BLOCK;
break;
case AFM_DK3ADPCM:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=DK3_ADPCM_BLOCK_SIZE*
(sh_audio->channels*sh_audio->samplerate) / DK3_ADPCM_SAMPLES_PER_BLOCK;
break;
case AFM_ROQAUDIO:
sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps = (sh_audio->channels * 22050) / 2;
break;
case AFM_MPEG: {
// MPEG Audio:
dec_audio_sh=sh_audio; // save sh_audio for the callback:
#ifdef USE_FAKE_MONO
MP3_Init(fakemono);
#else
MP3_Init();
#endif
MP3_samplerate=MP3_channels=0;
sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1);
sh_audio->channels=2; // hack
sh_audio->samplerate=MP3_samplerate;
sh_audio->i_bps=MP3_bitrate*(1000/8);
MP3_PrintHeader();
break;
}
#ifdef HAVE_OGGVORBIS
case AFM_VORBIS: {
// OggVorbis Audio:
#if 0 /* just here for reference - atmos */
ogg_sync_state oy; /* sync and verify incoming physical bitstream */
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
ogg_packet op; /* one raw packet of data for decode */
vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the bitstream user comments */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
#else
/* nix, nada, rien, nothing, nem, nüx */
#endif
uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header,
maybe use ogg_uint32_t */
//int i;
int ret;
char *buffer;
ogg_packet hdr;
//ov_struct_t *s=&sh_audio->ov;
sh_audio->ov=malloc(sizeof(ov_struct_t));
//s=&sh_audio->ov;
vorbis_info_init(&sh_audio->ov->vi);
vorbis_comment_init(&sh_audio->ov->vc);
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize);
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t));
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]);
/*for(i=12; i <= 40; i+=2) { // header bruteforce :)
memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t));
printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]);
}*/
/* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos
hdr.packet=NULL;
hdr.b_o_s = 1; /* beginning of stream for first packet */
hdr.bytes = hdrsizes[0];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n");
hdr.b_o_s = 0;
hdr.bytes = hdrsizes[1];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n");
hdr.bytes = hdrsizes[2];
hdr.packet = realloc(hdr.packet,hdr.bytes);
memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes);
if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n");
hdr.bytes=0;
hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */
/* done with the headers */
/* Throw the comments plus a few lines about the bitstream we're
decoding */
{
char **ptr=sh_audio->ov->vc.user_comments;
while(*ptr){
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
++ptr;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",sh_audio->ov->vi.channels,sh_audio->ov->vi.rate,sh_audio->ov->vi.bitrate_nominal/1000, (sh_audio->ov->vi.bitrate_lower!=sh_audio->ov->vi.bitrate_nominal)||(sh_audio->ov->vi.bitrate_upper!=sh_audio->ov->vi.bitrate_nominal)?'V':'C');
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor);
}
sh_audio->channels=sh_audio->ov->vi.channels;
sh_audio->samplerate=sh_audio->ov->vi.rate;
sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8;
// printf("[\n");
// sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels;
// printf("]\n");
/* OK, got and parsed all three headers. Initialize the Vorbis
packet->PCM decoder. */
vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */
vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode
so multiple block decodes can
proceed in parallel. We could init
multiple vorbis_block structures
for vd here */
//printf("OggVorbis: synthesis and block init done.\n");
ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
if(ret == -1)
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
else
if(ret == 0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n");
/* submit a a_buffer_len block to libvorbis' Ogg layer */
buffer=ogg_sync_buffer(&sh_audio->ov->oy,256);
ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256));
}
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n");
ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here
as they'll also become apparent
at packetout */
/* Get the serial number and set up the rest of decode. */
/* serialno first; use it to set up a logical stream */
ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og));
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
break;
}
#endif
#ifdef USE_LIBMAD
case AFM_MAD:
{
printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);
printf(__FILE__ ":%d:mad: initialising\n", __LINE__);
mad_frame_init(&mad_frame);
mad_stream_init(&mad_stream);
printf(__FILE__ ":%d:mad: preparing buffer\n", __LINE__);
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len);
// mad_stream_sync(&mad_stream);
mad_sync(sh_audio, &mad_stream);
mad_synth_init(&mad_synth);
if(mad_frame_decode(&mad_frame, &mad_stream) == 0)
{
printf(__FILE__ ":%d:mad: post processing buffer\n", __LINE__);
mad_postprocess_buffer(sh_audio, &mad_stream);
}
else
{
printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__);
mad_print_error(&mad_stream);
}
switch (mad_frame.header.mode)
{
case MAD_MODE_SINGLE_CHANNEL:
sh_audio->channels=1;
break;
case MAD_MODE_DUAL_CHANNEL:
case MAD_MODE_JOINT_STEREO:
case MAD_MODE_STEREO:
sh_audio->channels=2;
break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n");
}
mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n",
sh_audio->channels, mad_frame.header.mode);
/* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */
#if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13)
sh_audio->samplerate=mad_frame.header.samplerate;
#else
sh_audio->samplerate=mad_frame.header.sfreq;
#endif
sh_audio->i_bps=mad_frame.header.bitrate;
printf(__FILE__ ":%d:mad: continuing\n", __LINE__);
break;
}
#endif
}
if(!sh_audio->channels || !sh_audio->samplerate){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio);
driver=0;
}
if(!driver){
if(sh_audio->a_buffer) free(sh_audio->a_buffer);
sh_audio->a_buffer=NULL;
return 0;
}
if(!sh_audio->o_bps)
sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize;
return driver;
}
// Audio decoding:
// Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc)
// buffer length is 'maxlen' bytes, it shouldn't be exceeded...
int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
int len=-1;
switch(sh_audio->codec->driver){
#ifdef USE_LIBAVCODEC
case AFM_FFMPEG: {
unsigned char *start=NULL;
int y;
while(len<minlen){
int len2=0;
int x=ds_get_packet(sh_audio->ds,&start);
if(x<=0) break; // error
y=avcodec_decode_audio(&lavc_context,(INT16*)buf,&len2,start,x);
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
}
}
break;
#endif
case AFM_MPEG: // MPEG layer 2 or 3
len=MP3_DecodeFrame(buf,-1);
// len=MP3_DecodeFrame(buf,3);
break;
#ifdef HAVE_OGGVORBIS
case AFM_VORBIS: { // OggVorbis
/* note: good minlen would be 4k or 8k IMHO - atmos */
int ret;
char *buffer;
int bytes;
int samples;
float **pcm;
//ogg_int16_t convbuffer[4096];
// int convsize;
int readlen=1024;
len=0;
// convsize=minlen/sh_audio->ov->vi.channels;
while(len < minlen) { /* double loop allows for break in inner loop */
while(len < minlen) { /* without aborting the outer loop - atmos */
ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op);
if(ret==0) {
int xxx=0;
//printf("OggVorbis: Packetout: need more data, paging!\n");
while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
if(ret == -1)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
else
if(ret == 0) {
//printf("OggVorbis: Pageout: need more data to verify page, reading more data.\n");
/* submit a readlen k block to libvorbis' Ogg layer */
buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen);
bytes=demux_read_data(sh_audio->ds,buffer,readlen);
xxx+=bytes;
ogg_sync_wrote(&sh_audio->ov->oy,bytes);
if(bytes==0)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n");
}
}
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx);
//printf("OggVorbis: Pageout: successfull, pagin in.\n");
if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0)
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n");
break;
} else if(ret<0) {
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Packetout: missing or corrupt data, skipping packet!\n");
break;
} else {
/* we have a packet. Decode it */
if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */
vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->vb);
/* **pcm is a multichannel float vector. In stereo, for
example, pcm[0] is left, and pcm[1] is right. samples is
the size of each channel. Convert the float values
(-1.<=range<=1.) to whatever PCM format and write it out */
while((samples=vorbis_synthesis_pcmout(&sh_audio->ov->vd,&pcm))>0){
int i,j;
int clipflag=0;
int convsize=(maxlen-len)/(2*sh_audio->ov->vi.channels); // max size!
int bout=(samples<convsize?samples:convsize);
if(bout<=0) break;
/* convert floats to 16 bit signed ints (host order) and
interleave */
for(i=0;i<sh_audio->ov->vi.channels;i++){
ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
ogg_int16_t *ptr=convbuffer+i;
float *mono=pcm[i];
for(j=0;j<bout;j++){
#if 1
int val=mono[j]*32767.f;
#else /* optional dither */
int val=mono[j]*32767.f+drand48()-0.5f;
#endif
/* might as well guard against clipping */
if(val>32767){
val=32767;
clipflag=1;
}
if(val<-32768){
val=-32768;
clipflag=1;
}
*ptr=val;
ptr+=sh_audio->ov->vi.channels;
}
}
if(clipflag)
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence));
//fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging
//memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout);
len+=2*sh_audio->ov->vi.channels*bout;
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how
many samples we
actually consumed */
}
} // from else, packetout ok
} // while len
} // outer while len
if(ogg_page_eos(&sh_audio->ov->og))
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos
mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len);
break;
}
#endif
case AFM_PCM: // AVI PCM
len=demux_read_data(sh_audio->ds,buf,minlen);
break;
case AFM_DVDPCM: // DVD PCM
{ int j;
len=demux_read_data(sh_audio->ds,buf,minlen);
//if(i&1){ printf("Warning! pcm_audio_size&1 !=0 (%d)\n",i);i&=~1; }
// swap endian:
for(j=0;j<len;j+=2){
char x=buf[j];
buf[j]=buf[j+1];
buf[j+1]=x;
}
break;
}
case AFM_ALAW: // aLaw decoder
{ int l=demux_read_data(sh_audio->ds,buf,minlen/2);
unsigned short *d=(unsigned short *) buf;
unsigned char *s=buf;
len=2*l;
if(sh_audio->format==6){
// aLaw
while(l>0){ --l; d[l]=alaw2short[s[l]]; }
} else {
// uLaw
while(l>0){ --l; d[l]=ulaw2short[s[l]]; }
}
break;
}
case AFM_GSM: // MS-GSM decoder
{ unsigned char ibuf[65]; // 65 bytes / frame
if(demux_read_data(sh_audio->ds,ibuf,65)!=65) break; // EOF
XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short
// XA_GSM_Decoder(buf,(unsigned short *) &sh_audio->a_buffer[sh_audio->a_buffer_len]); // decodes 33 byte -> 160 short
len=2*320;
break;
}
#ifdef USE_G72X
case AFM_G72X: // GSM 723 decoder
{ if(demux_read_data(sh_audio->ds,g72x_data.block, g72x_data.blocksize)!=g72x_data.blocksize) break; // EOF
g72x_decode_block(&g72x_data);
len=2*g72x_data.samplesperblock;
memcpy(buf,g72x_data.samples,len);
break;
}
#endif
case AFM_IMAADPCM:
{ unsigned char ibuf[IMA_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame
if (demux_read_data(sh_audio->ds, ibuf,
IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) !=
IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels)
break; // EOF
len=2*ima_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels);
break;
}
case AFM_MSADPCM:
{ static unsigned char *ibuf = NULL;
if (!ibuf)
ibuf = (unsigned char *)malloc
(sh_audio->wf->nBlockAlign * sh_audio->wf->nChannels);
if (demux_read_data(sh_audio->ds, ibuf,
sh_audio->wf->nBlockAlign) !=
sh_audio->wf->nBlockAlign)
break; // EOF
len= 2 * ms_adpcm_decode_block(
(unsigned short*)buf,ibuf, sh_audio->wf->nChannels,
sh_audio->wf->nBlockAlign);
break;
}
case AFM_DK4ADPCM:
{ static unsigned char *ibuf = NULL;
if (!ibuf)
ibuf = (unsigned char *)malloc(sh_audio->wf->nBlockAlign);
if (demux_read_data(sh_audio->ds, ibuf, sh_audio->wf->nBlockAlign) !=
sh_audio->wf->nBlockAlign)
break; // EOF
len=2*dk4_adpcm_decode_block((unsigned short*)buf,ibuf,
sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
break;
}
case AFM_DK3ADPCM:
{ unsigned char ibuf[DK3_ADPCM_BLOCK_SIZE * 2]; // bytes / stereo frame
if (demux_read_data(sh_audio->ds, ibuf,
DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) !=
DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels)
break; // EOF
len = 2 * dk3_adpcm_decode_block(
(unsigned short*)buf,ibuf);
break;
}
case AFM_ROQAUDIO:
{
static unsigned char *ibuf = NULL;
unsigned char header_data[6];
int read_len;
if (!ibuf)
ibuf = (unsigned char *)malloc(sh_audio->audio_out_minsize / 2);
// figure out how much data to read
if (demux_read_data(sh_audio->ds, header_data, 6) != 6)
break; // EOF
read_len = (header_data[5] << 24) | (header_data[4] << 16) |
(header_data[3] << 8) | header_data[2];
read_len += 2; // 16-bit arguments
if (demux_read_data(sh_audio->ds, ibuf, read_len) != read_len)
break;
len = 2 * roq_decode_audio((unsigned short *)buf, ibuf,
read_len, sh_audio->channels, sh_audio->context);
break;
}
#ifdef USE_LIBAC3
case AFM_AC3: // AC3 decoder
//printf("{1:%d}",avi_header.idx_pos);fflush(stdout);
if(!sh_audio->ac3_frame) sh_audio->ac3_frame=ac3_decode_frame();
//printf("{2:%d}",avi_header.idx_pos);fflush(stdout);
if(sh_audio->ac3_frame){
len = 256 * 6 *sh_audio->channels*sh_audio->samplesize;
memcpy(buf,((ac3_frame_t*)sh_audio->ac3_frame)->audio_data,len);
sh_audio->ac3_frame=NULL;
}
//printf("{3:%d}",avi_header.idx_pos);fflush(stdout);
break;
#endif
case AFM_A52: { // AC3 decoder
sample_t level=1, bias=384;
int flags=a52_flags|A52_ADJUST_LEVEL;
int i;
if(!sh_audio->a_in_buffer_len)
if(a52_fillbuff(sh_audio)<0) break; // EOF
sh_audio->a_in_buffer_len=0;
if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
break;
}
// a52_dynrng (&state, NULL, NULL); // disable dynamic range compensation
// frame decoded, let's resample:
//a52_resample_init(a52_accel,flags,sh_audio->channels);
len=0;
for (i = 0; i < 6; i++) {
if (a52_block (&a52_state, a52_samples)){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
break;
}
len+=2*a52_resample(a52_samples,&buf[len]);
}
// printf("len = %d \n",len); // 6144 on all vobs I tried so far... (5.1 and 2.0) ::atmos
break;
}
case AFM_HWAC3: // AC3 through SPDIF
if(!sh_audio->a_in_buffer_len)
if((len=a52_fillbuff(sh_audio))<0) break; //EOF
sh_audio->a_in_buffer_len=0;
len = ac3_iec958_build_burst(len, 0x01, 1, sh_audio->a_in_buffer, buf);
// len = 6144 = 4*(6*256)
break;
#ifdef USE_WIN32DLL
case AFM_ACM:
// len=sh_audio->audio_out_minsize; // optimal decoded fragment size
// if(len<minlen) len=minlen; else
// if(len>maxlen) len=maxlen;
// len=acm_decode_audio(sh_audio,buf,len);
len=acm_decode_audio(sh_audio,buf,minlen,maxlen);
break;
#endif
#ifdef USE_DIRECTSHOW
case AFM_DSHOW: // DirectShow
{ int size_in=0;
int size_out=0;
int srcsize=DS_AudioDecoder_GetSrcSize(ds_adec, maxlen);
mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DShow says: srcsize=%d (buffsize=%d) out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen);
if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!!
if(sh_audio->a_in_buffer_len<srcsize){
sh_audio->a_in_buffer_len+=
demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len],
srcsize-sh_audio->a_in_buffer_len);
}
DS_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len,
buf,maxlen, &size_in,&size_out);
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DShow: audio %d -> %d converted (in_buf_len=%d of %d) %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds));
if(size_in>=sh_audio->a_in_buffer_len){
sh_audio->a_in_buffer_len=0;
} else {
sh_audio->a_in_buffer_len-=size_in;
memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len);
}
len=size_out;
break;
}
#endif
#ifdef USE_LIBMAD
case AFM_MAD:
{
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
// mad_stream_sync(&mad_stream);
mad_sync(sh_audio, &mad_stream);
if(mad_frame_decode(&mad_frame, &mad_stream) == 0)
{
mad_synth_frame(&mad_synth, &mad_frame);
mad_postprocess_buffer(sh_audio, &mad_stream);
/* and fill buffer */
{
int i;
int end_size = mad_synth.pcm.length;
signed short* samples = (signed short*)buf;
if(end_size > maxlen/4)
end_size=maxlen/4;
for(i=0; i<mad_synth.pcm.length; ++i) {
*samples++ = mad_scale(mad_synth.pcm.samples[0][i]);
*samples++ = mad_scale(mad_synth.pcm.samples[0][i]);
// *buf++ = mad_scale(mad_synth.pcm.sampAles[1][i]);
}
len = end_size*4;
}
}
else
{
printf(__FILE__ ":%d:mad: frame decoding failed (error: %d)\n", __LINE__,
mad_stream.error);
mad_print_error(&mad_stream);
}
break;
}
#endif
}
return len;
}
void resync_audio_stream(sh_audio_t *sh_audio){
switch(sh_audio->codec->driver){
case AFM_MPEG:
MP3_DecodeFrame(NULL,-2); // resync
MP3_DecodeFrame(NULL,-2); // resync
MP3_DecodeFrame(NULL,-2); // resync
break;
#ifdef HAVE_OGGVORBIS
case AFM_VORBIS:
//printf("OggVorbis: resetting stream.\n");
ogg_sync_reset(&sh_audio->ov->oy);
ogg_stream_reset(&sh_audio->ov->os);
break;
#endif
#ifdef USE_LIBAC3
case AFM_AC3:
ac3_bitstream_reset(); // reset AC3 bitstream buffer
// if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);}
sh_audio->ac3_frame=ac3_decode_frame(); // resync
// if(verbose) printf(" OK!\n");
break;
#endif
case AFM_A52:
case AFM_ACM:
case AFM_DSHOW:
case AFM_HWAC3:
sh_audio->a_in_buffer_len=0; // reset ACM/DShow audio buffer
break;
#ifdef USE_LIBMAD
case AFM_MAD:
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
// mad_stream_sync(&mad_stream);
mad_sync(sh_audio, &mad_stream);
mad_postprocess_buffer(sh_audio, &mad_stream);
break;
#endif
}
}
void skip_audio_frame(sh_audio_t *sh_audio){
switch(sh_audio->codec->driver){
case AFM_MPEG: MP3_DecodeFrame(NULL,-2);break; // skip MPEG frame
#ifdef USE_LIBAC3
case AFM_AC3: sh_audio->ac3_frame=ac3_decode_frame();break; // skip AC3 frame
#endif
case AFM_HWAC3:
case AFM_A52: a52_fillbuff(sh_audio);break; // skip AC3 frame
case AFM_ACM:
case AFM_DSHOW: {
int skip=sh_audio->wf->nBlockAlign;
if(skip<16){
skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7);
if(skip<16) skip=16;
}
demux_read_data(sh_audio->ds,NULL,skip);
break;
}
case AFM_PCM:
case AFM_DVDPCM:
case AFM_ALAW: {
int skip=sh_audio->i_bps/16;
skip=skip&(~3);
demux_read_data(sh_audio->ds,NULL,skip);
break;
}
#ifdef USE_LIBMAD
case AFM_MAD:
{
mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
mad_stream_skip(&mad_stream, 2);
// mad_stream_sync(&mad_stream);
mad_sync(sh_audio, &mad_stream);
mad_postprocess_buffer(sh_audio, &mad_stream);
break;
}
#endif
default: ds_fill_buffer(sh_audio->ds); // skip PCM frame
}
}
|