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
|
/* GStreamer FAAD (Free AAC Decoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-faad
* @seealso: faac
*
* faad decodes AAC (MPEG-4 part 3) stream.
*
* <refsect2>
* <title>Example launch lines</title>
* |[
* gst-launch filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play aac from mp4 file.
* |[
* gst-launch filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play standalone aac bitstream.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/audio/audio.h>
/* These are the correct types for these functions, as defined in the source,
* with types changed to match glib types, since those are defined for us.
* However, upstream FAAD is distributed with a broken header file that defined
* these wrongly (in a way which was broken on 64 bit systems).
*
* Upstream CVS still has the bug, but has also renamed all the public symbols
* for Better Corporate Branding (or whatever), so we need to take that
* (FAAD_IS_NEAAC) into account as well.
*
* We must call them using these definitions. Most distributions now have the
* corrected header file (they distribute a patch along with the source),
* but not all, hence this Truly Evil Hack.
*
* Note: The prototypes don't need to be defined conditionaly, as the cpp will
* do that for us.
*/
#if FAAD2_MINOR_VERSION < 7
#ifdef FAAD_IS_NEAAC
#define NeAACDecInit NeAACDecInit_no_definition
#define NeAACDecInit2 NeAACDecInit2_no_definition
#else
#define faacDecInit faacDecInit_no_definition
#define faacDecInit2 faacDecInit2_no_definition
#endif
#endif /* FAAD2_MINOR_VERSION < 7 */
#include "gstfaad.h"
#if FAAD2_MINOR_VERSION < 7
#ifdef FAAD_IS_NEAAC
#undef NeAACDecInit
#undef NeAACDecInit2
#else
#undef faacDecInit
#undef faacDecInit2
#endif
extern long faacDecInit (faacDecHandle, guint8 *, guint32, guint32 *, guint8 *);
extern gint8 faacDecInit2 (faacDecHandle, guint8 *, guint32,
guint32 *, guint8 *);
#endif /* FAAD2_MINOR_VERSION < 7 */
GST_DEBUG_CATEGORY_STATIC (faad_debug);
#define GST_CAT_DEFAULT faad_debug
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
);
#define STATIC_RAW_CAPS(format) \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE(format)", " \
"layout = (string) interleaved, " \
"rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 8 ]"
/*
* All except 16-bit integer are disabled until someone fixes FAAD.
* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
* for 1 frame (1024 samples) of 6 channel (5.1) 16-bit integer 16bpp
* audio, but not for any other. You'll get random segfaults, crashes
* and even valgrind goes crazy.
*/
#define STATIC_CAPS \
STATIC_RAW_CAPS (S16)
#if 0
#define NOTUSED "; " \
STATIC_RAW_CAPS (S24) \
"; " \
STATIC_RAW_CAPS (S32) \
"; " \
STATIC_RAW_CAPS (F32) \
"; " \
STATIC_RAW_CAPS (F64)
#endif
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (STATIC_CAPS)
);
static void gst_faad_reset (GstFaad * faad);
static gboolean gst_faad_start (GstAudioDecoder * dec);
static gboolean gst_faad_stop (GstAudioDecoder * dec);
static gboolean gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps);
static gboolean gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length);
static GstFlowReturn gst_faad_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard);
static gboolean gst_faad_open_decoder (GstFaad * faad);
static void gst_faad_close_decoder (GstFaad * faad);
#define gst_faad_parent_class parent_class
G_DEFINE_TYPE (GstFaad, gst_faad, GST_TYPE_AUDIO_DECODER);
static void
gst_faad_class_init (GstFaadClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class, "AAC audio decoder",
"Codec/Decoder/Audio",
"Free MPEG-2/4 AAC decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush);
GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
}
static void
gst_faad_init (GstFaad * faad)
{
gst_faad_reset (faad);
}
static void
gst_faad_reset_stream_state (GstFaad * faad)
{
if (faad->handle)
faacDecPostSeekReset (faad->handle, 0);
}
static void
gst_faad_reset (GstFaad * faad)
{
faad->samplerate = -1;
faad->channels = -1;
faad->init = FALSE;
faad->packetised = FALSE;
g_free (faad->channel_positions);
faad->channel_positions = NULL;
faad->last_header = 0;
gst_faad_reset_stream_state (faad);
}
static gboolean
gst_faad_start (GstAudioDecoder * dec)
{
GstFaad *faad = GST_FAAD (dec);
GST_DEBUG_OBJECT (dec, "start");
gst_faad_reset (faad);
/* call upon legacy upstream byte support (e.g. seeking) */
gst_audio_decoder_set_byte_time (dec, TRUE);
/* never mind a few errors */
gst_audio_decoder_set_max_errors (dec, 10);
return TRUE;
}
static gboolean
gst_faad_stop (GstAudioDecoder * dec)
{
GstFaad *faad = GST_FAAD (dec);
GST_DEBUG_OBJECT (dec, "stop");
gst_faad_reset (faad);
gst_faad_close_decoder (faad);
return TRUE;
}
static gint
aac_rate_idx (gint rate)
{
if (92017 <= rate)
return 0;
else if (75132 <= rate)
return 1;
else if (55426 <= rate)
return 2;
else if (46009 <= rate)
return 3;
else if (37566 <= rate)
return 4;
else if (27713 <= rate)
return 5;
else if (23004 <= rate)
return 6;
else if (18783 <= rate)
return 7;
else if (13856 <= rate)
return 8;
else if (11502 <= rate)
return 9;
else if (9391 <= rate)
return 10;
else
return 11;
}
static gboolean
gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstFaad *faad = GST_FAAD (dec);
GstStructure *str = gst_caps_get_structure (caps, 0);
GstBuffer *buf;
const GValue *value;
GstMapInfo map;
guint8 *cdata;
gsize csize;
/* clean up current decoder, rather than trying to reconfigure */
gst_faad_close_decoder (faad);
/* Assume raw stream */
faad->packetised = FALSE;
if ((value = gst_structure_get_value (str, "codec_data"))) {
#if FAAD2_MINOR_VERSION >= 7
unsigned long samplerate;
#else
guint32 samplerate;
#endif
guint8 channels;
/* We have codec data, means packetised stream */
faad->packetised = TRUE;
buf = gst_value_get_buffer (value);
g_return_val_if_fail (buf != NULL, FALSE);
gst_buffer_map (buf, &map, GST_MAP_READ);
cdata = map.data;
csize = map.size;
if (csize < 2)
goto wrong_length;
GST_DEBUG_OBJECT (faad,
"codec_data: object_type=%d, sample_rate=%d, channels=%d",
((cdata[0] & 0xf8) >> 3),
(((cdata[0] & 0x07) << 1) | ((cdata[1] & 0x80) >> 7)),
((cdata[1] & 0x78) >> 3));
if (!gst_faad_open_decoder (faad))
goto open_failed;
/* someone forgot that char can be unsigned when writing the API */
if ((gint8) faacDecInit2 (faad->handle, cdata, csize, &samplerate,
&channels) < 0)
goto init_failed;
if (channels != ((cdata[1] & 0x78) >> 3)) {
/* https://bugs.launchpad.net/ubuntu/+source/faad2/+bug/290259 */
GST_WARNING_OBJECT (faad,
"buggy faad version, wrong nr of channels %d instead of %d", channels,
((cdata[1] & 0x78) >> 3));
}
GST_DEBUG_OBJECT (faad, "codec_data init: channels=%u, rate=%u", channels,
(guint32) samplerate);
/* not updating these here, so they are updated in the
* chain function, and new caps are created etc. */
faad->samplerate = 0;
faad->channels = 0;
faad->init = TRUE;
gst_buffer_unmap (buf, &map);
} else if ((value = gst_structure_get_value (str, "framed")) &&
g_value_get_boolean (value) == TRUE) {
faad->packetised = TRUE;
faad->init = FALSE;
GST_DEBUG_OBJECT (faad, "we have packetized audio");
} else {
faad->init = FALSE;
}
faad->fake_codec_data[0] = 0;
faad->fake_codec_data[1] = 0;
if (faad->packetised && !faad->init) {
gint rate, channels;
if (gst_structure_get_int (str, "rate", &rate) &&
gst_structure_get_int (str, "channels", &channels)) {
gint rate_idx, profile;
profile = 3; /* 0=MAIN, 1=LC, 2=SSR, 3=LTP */
rate_idx = aac_rate_idx (rate);
faad->fake_codec_data[0] = ((profile + 1) << 3) | ((rate_idx & 0xE) >> 1);
faad->fake_codec_data[1] = ((rate_idx & 0x1) << 7) | (channels << 3);
GST_LOG_OBJECT (faad, "created fake codec data (%u,%u): 0x%x 0x%x", rate,
channels, (int) faad->fake_codec_data[0],
(int) faad->fake_codec_data[1]);
}
}
return TRUE;
/* ERRORS */
wrong_length:
{
GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
gst_object_unref (faad);
gst_buffer_unmap (buf, &map);
return FALSE;
}
open_failed:
{
GST_DEBUG_OBJECT (faad, "failed to create decoder");
gst_object_unref (faad);
gst_buffer_unmap (buf, &map);
return FALSE;
}
init_failed:
{
GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
gst_object_unref (faad);
gst_buffer_unmap (buf, &map);
return FALSE;
}
}
static gboolean
gst_faad_chanpos_to_gst (GstFaad * faad, guchar * fpos,
GstAudioChannelPosition * pos, guint num)
{
guint n;
gboolean unknown_channel = FALSE;
/* special handling for the common cases for mono and stereo */
if (num == 1 && fpos[0] == FRONT_CHANNEL_CENTER) {
GST_DEBUG_OBJECT (faad, "mono common case; won't set channel positions");
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
return TRUE;
} else if (num == 2 && fpos[0] == FRONT_CHANNEL_LEFT
&& fpos[1] == FRONT_CHANNEL_RIGHT) {
GST_DEBUG_OBJECT (faad, "stereo common case; won't set channel positions");
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
return TRUE;
}
for (n = 0; n < num; n++) {
GST_DEBUG_OBJECT (faad, "faad channel %d as %d", n, fpos[n]);
switch (fpos[n]) {
case FRONT_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
break;
case FRONT_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
case FRONT_CHANNEL_CENTER:
/* argh, mono = center */
if (num == 1)
pos[n] = GST_AUDIO_CHANNEL_POSITION_MONO;
else
pos[n] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
break;
case SIDE_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT;
break;
case SIDE_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT;
break;
case BACK_CHANNEL_LEFT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
break;
case BACK_CHANNEL_RIGHT:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
break;
case BACK_CHANNEL_CENTER:
pos[n] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
break;
case LFE_CHANNEL:
pos[n] = GST_AUDIO_CHANNEL_POSITION_LFE1;
break;
default:
GST_DEBUG_OBJECT (faad, "unknown channel %d at %d", fpos[n], n);
unknown_channel = TRUE;
break;
}
}
if (unknown_channel) {
switch (num) {
case 1:{
GST_DEBUG_OBJECT (faad,
"FAAD reports unknown 1 channel mapping. Forcing to mono");
pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO;
break;
}
case 2:{
GST_DEBUG_OBJECT (faad,
"FAAD reports unknown 2 channel mapping. Forcing to stereo");
pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
break;
}
default:{
GST_WARNING_OBJECT (faad,
"Unsupported FAAD channel position 0x%x encountered", fpos[n]);
return FALSE;
break;
}
}
}
return TRUE;
}
static gboolean
gst_faad_update_caps (GstFaad * faad, faacDecFrameInfo * info)
{
gboolean ret;
GstCaps *caps;
gboolean fmt_change = FALSE;
GstAudioInfo ainfo;
/* see if we need to renegotiate */
if (info->samplerate != faad->samplerate ||
info->channels != faad->channels || !faad->channel_positions) {
fmt_change = TRUE;
} else {
gint i;
for (i = 0; i < info->channels; i++) {
if (info->channel_position[i] != faad->channel_positions[i]) {
fmt_change = TRUE;
break;
}
}
}
if (G_LIKELY (gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (faad))
&& !fmt_change))
return TRUE;
/* store new negotiation information */
faad->samplerate = info->samplerate;
faad->channels = info->channels;
g_free (faad->channel_positions);
faad->channel_positions = g_memdup (info->channel_position, faad->channels);
/* FIXME: Use the GstAudioInfo of GstAudioDecoder for all of this */
gst_audio_info_init (&ainfo);
gst_audio_info_set_format (&ainfo, GST_AUDIO_FORMAT_S16, faad->samplerate,
faad->channels, NULL);
faad->bps = 16 / 8;
if (!gst_faad_chanpos_to_gst (faad, faad->channel_positions,
faad->aac_positions, faad->channels)) {
GST_DEBUG_OBJECT (faad, "Could not map channel positions");
return FALSE;
}
memcpy (ainfo.position, faad->aac_positions,
faad->channels * sizeof (GstAudioChannelPosition));
gst_audio_channel_positions_to_valid_order (ainfo.position, faad->channels);
memcpy (faad->gst_positions, ainfo.position,
faad->channels * sizeof (GstAudioChannelPosition));
/* Unset UNPOSITIONED flag */
if (ainfo.position[0] != GST_AUDIO_CHANNEL_POSITION_NONE)
ainfo.flags &= ~GST_AUDIO_FLAG_UNPOSITIONED;
caps = gst_audio_info_to_caps (&ainfo);
GST_DEBUG_OBJECT (faad, "New output caps: %" GST_PTR_FORMAT, caps);
ret = gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (faad), caps);
gst_caps_unref (caps);
return ret;
}
/*
* Find syncpoint in ADTS/ADIF stream. Doesn't work for raw,
* packetized streams. Be careful when calling.
* Returns FALSE on no-sync, fills offset/length if one/two
* syncpoints are found, only returns TRUE when it finds two
* subsequent syncpoints (similar to mp3 typefinding in
* gst/typefind/) for ADTS because 12 bits isn't very reliable.
*/
static gboolean
gst_faad_sync (GstFaad * faad, const guint8 * data, guint size, gboolean next,
gint * off, gint * length)
{
guint n = 0;
gint snc;
gboolean ret = FALSE;
guint len = 0;
GST_LOG_OBJECT (faad, "Finding syncpoint");
/* check for too small a buffer */
if (size < 3)
goto exit;
for (n = 0; n < size - 3; n++) {
snc = GST_READ_UINT16_BE (&data[n]);
if ((snc & 0xfff6) == 0xfff0) {
/* we have an ADTS syncpoint. Parse length and find
* next syncpoint. */
GST_LOG_OBJECT (faad,
"Found one ADTS syncpoint at offset 0x%x, tracing next...", n);
if (size - n < 5) {
GST_LOG_OBJECT (faad, "Not enough data to parse ADTS header");
break;
}
len = ((data[n + 3] & 0x03) << 11) |
(data[n + 4] << 3) | ((data[n + 5] & 0xe0) >> 5);
if (n + len + 2 >= size) {
GST_LOG_OBJECT (faad, "Frame size %d, next frame is not within reach",
len);
if (next) {
break;
} else if (n + len <= size) {
GST_LOG_OBJECT (faad, "but have complete frame and no next frame; "
"accept ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
ret = TRUE;
break;
}
}
snc = GST_READ_UINT16_BE (&data[n + len]);
if ((snc & 0xfff6) == 0xfff0) {
GST_LOG_OBJECT (faad,
"Found ADTS syncpoint at offset 0x%x (framelen %u)", n, len);
ret = TRUE;
break;
}
GST_LOG_OBJECT (faad, "No next frame found... (should be at 0x%x)",
n + len);
} else if (!memcmp (&data[n], "ADIF", 4)) {
/* we have an ADIF syncpoint. 4 bytes is enough. */
GST_LOG_OBJECT (faad, "Found ADIF syncpoint at offset 0x%x", n);
ret = TRUE;
break;
}
}
exit:
*off = n;
if (ret) {
*length = len;
} else {
GST_LOG_OBJECT (faad, "Found no syncpoint");
}
return ret;
}
static gboolean
looks_like_valid_header (guint8 * input_data, guint input_size)
{
if (input_size < 4)
return FALSE;
if (input_data[0] == 'A'
&& input_data[1] == 'D' && input_data[2] == 'I' && input_data[3] == 'F')
/* ADIF type header */
return TRUE;
if (input_data[0] == 0xff && (input_data[1] >> 4) == 0xf)
/* ADTS type header */
return TRUE;
return FALSE;
}
static GstFlowReturn
gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
GstFaad *faad;
const guint8 *data;
guint size;
gboolean sync, eos;
faad = GST_FAAD (dec);
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
gst_audio_decoder_get_parse_state (dec, &sync, &eos);
if (faad->packetised) {
*offset = 0;
*length = size;
return GST_FLOW_OK;
} else {
gboolean ret;
data = gst_adapter_map (adapter, size);
ret = gst_faad_sync (faad, data, size, !eos, offset, length);
gst_adapter_unmap (adapter);
return (ret ? GST_FLOW_OK : GST_FLOW_EOS);
}
}
static GstFlowReturn
gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstFaad *faad;
GstFlowReturn ret = GST_FLOW_OK;
GstMapInfo map;
gsize input_size;
guchar *input_data;
GstBuffer *outbuf;
faacDecFrameInfo info;
void *out;
faad = GST_FAAD (dec);
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
gst_buffer_map (buffer, &map, GST_MAP_READ);
input_data = map.data;
input_size = map.size;
init:
/* init if not already done during capsnego */
if (!faad->init) {
#if FAAD2_MINOR_VERSION >= 7
unsigned long rate;
#else
guint32 rate;
#endif
guint8 ch;
GST_DEBUG_OBJECT (faad, "initialising ...");
if (!gst_faad_open_decoder (faad))
goto open_failed;
/* We check if the first data looks like it might plausibly contain
* appropriate initialisation info... if not, we use our fake_codec_data
*/
if (looks_like_valid_header (input_data, input_size) || !faad->packetised) {
if (faacDecInit (faad->handle, input_data, input_size, &rate, &ch) < 0)
goto init_failed;
GST_DEBUG_OBJECT (faad, "faacDecInit() ok: rate=%u,channels=%u",
(guint32) rate, ch);
} else {
if ((gint8) faacDecInit2 (faad->handle, faad->fake_codec_data, 2,
&rate, &ch) < 0) {
goto init2_failed;
}
GST_DEBUG_OBJECT (faad, "faacDecInit2() ok: rate=%u,channels=%u",
(guint32) rate, ch);
}
faad->init = TRUE;
/* make sure we create new caps below */
faad->samplerate = 0;
faad->channels = 0;
}
/* decode cycle */
info.error = 0;
do {
if (!faad->packetised) {
/* faad only really parses ADTS header at Init time, not when decoding,
* so monitor for changes and kick faad when needed */
if (GST_READ_UINT32_BE (input_data) >> 4 != faad->last_header >> 4) {
GST_DEBUG_OBJECT (faad, "ADTS header changed, forcing Init");
faad->last_header = GST_READ_UINT32_BE (input_data);
/* kick hard */
gst_faad_close_decoder (faad);
faad->init = FALSE;
goto init;
}
}
out = faacDecDecode (faad->handle, &info, input_data, input_size);
if (info.error > 0) {
/* give up on frame and bail out */
gst_audio_decoder_finish_frame (dec, NULL, 1);
goto decode_failed;
}
GST_LOG_OBJECT (faad, "%d bytes consumed, %d samples decoded",
(guint) info.bytesconsumed, (guint) info.samples);
if (out && info.samples > 0) {
if (!gst_faad_update_caps (faad, &info))
goto negotiation_failed;
/* C's lovely propensity for int overflow.. */
if (info.samples > G_MAXUINT / faad->bps)
goto sample_overflow;
/* note: info.samples is total samples, not per channel */
/* FIXME, add bufferpool and allocator support to the base class */
outbuf = gst_buffer_new_allocate (NULL, info.samples * faad->bps, 0);
gst_buffer_fill (outbuf, 0, out, info.samples * faad->bps);
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
faad->channels, faad->aac_positions, faad->gst_positions);
ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
}
} while (FALSE);
out:
gst_buffer_unmap (buffer, &map);
return ret;
/* ERRORS */
open_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to open decoder"));
ret = GST_FLOW_ERROR;
goto out;
}
init_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Failed to init decoder from stream"));
ret = GST_FLOW_ERROR;
goto out;
}
init2_failed:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("%s() failed", (faad->handle) ? "faacDecInit2" : "faacDecOpen"));
ret = GST_FLOW_ERROR;
goto out;
}
decode_failed:
{
GST_AUDIO_DECODER_ERROR (faad, 1, STREAM, DECODE, (NULL),
("decoding error: %s", faacDecGetErrorMessage (info.error)), ret);
goto out;
}
negotiation_failed:
{
GST_ELEMENT_ERROR (faad, CORE, NEGOTIATION, (NULL),
("Setting caps on source pad failed"));
ret = GST_FLOW_ERROR;
goto out;
}
sample_overflow:
{
GST_ELEMENT_ERROR (faad, STREAM, DECODE, (NULL),
("Output buffer too large"));
ret = GST_FLOW_ERROR;
goto out;
}
}
static void
gst_faad_flush (GstAudioDecoder * dec, gboolean hard)
{
gst_faad_reset_stream_state (GST_FAAD (dec));
}
static gboolean
gst_faad_open_decoder (GstFaad * faad)
{
faacDecConfiguration *conf;
faad->handle = faacDecOpen ();
if (faad->handle == NULL) {
GST_WARNING_OBJECT (faad, "faacDecOpen() failed");
return FALSE;
}
conf = faacDecGetCurrentConfiguration (faad->handle);
conf->defObjectType = LC;
conf->dontUpSampleImplicitSBR = 1;
conf->outputFormat = FAAD_FMT_16BIT;
if (faacDecSetConfiguration (faad->handle, conf) == 0) {
GST_WARNING_OBJECT (faad, "faacDecSetConfiguration() failed");
return FALSE;
}
return TRUE;
}
static void
gst_faad_close_decoder (GstFaad * faad)
{
if (faad->handle) {
faacDecClose (faad->handle);
faad->handle = NULL;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "faad", GST_RANK_PRIMARY, GST_TYPE_FAAD);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"faad",
"Free AAC Decoder (FAAD)",
plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|