1 /* GStreamer
2 * Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.net>
3 *
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
8 *
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
13 *
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
18 */
19
20 /**
21 * SECTION:element-a52dec
22 *
23 * Dolby Digital (AC-3) audio decoder.
24 *
25 * <refsect2>
26 * <title>Example launch line</title>
27 * |[
28 * gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
29 * ]| Play audio part of a dvd title.
30 * |[
31 * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
32 * ]| Decode and play a stand alone AC-3 file.
33 * </refsect2>
34 */
35
36 #ifdef HAVE_CONFIG_H
37 #include "config.h"
38 #endif
39
40 #include <string.h>
41 #include <stdlib.h>
42 #ifdef HAVE_STDINT_H
43 #include <stdint.h>
44 #endif
45
46 #include <gst/gst.h>
47
48 #include <a52dec/a52.h>
49 #if !defined(A52_ACCEL_DETECT)
50 # include <a52dec/mm_accel.h>
51 #endif
52 #include "gsta52dec.h"
53
54 #if HAVE_ORC
55 #include <orc/orc.h>
56 #endif
57
58 #ifdef LIBA52_DOUBLE
59 #define SAMPLE_WIDTH 64
60 #define SAMPLE_FORMAT GST_AUDIO_NE(F64)
61 #define SAMPLE_TYPE GST_AUDIO_FORMAT_F64
62 #else
63 #define SAMPLE_WIDTH 32
64 #define SAMPLE_FORMAT GST_AUDIO_NE(F32)
65 #define SAMPLE_TYPE GST_AUDIO_FORMAT_F32
66 #endif
67
68 GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
69 #define GST_CAT_DEFAULT (a52dec_debug)
70
71 /* A52Dec args */
72 enum
73 {
74 ARG_0,
75 ARG_DRC,
76 ARG_MODE,
77 ARG_LFE,
78 };
79
80 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
81 GST_PAD_SINK,
82 GST_PAD_ALWAYS,
83 GST_STATIC_CAPS ("audio/x-ac3; audio/ac3; audio/x-private1-ac3")
84 );
85
86 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
87 GST_PAD_SRC,
88 GST_PAD_ALWAYS,
89 GST_STATIC_CAPS ("audio/x-raw, "
90 "format = (string) " SAMPLE_FORMAT ", "
91 "layout = (string) interleaved, "
92 "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
93 );
94
95 #define gst_a52dec_parent_class parent_class
96 G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_AUDIO_DECODER);
97
98 static gboolean gst_a52dec_start (GstAudioDecoder * dec);
99 static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
100 static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
101 static GstFlowReturn gst_a52dec_parse (GstAudioDecoder * dec,
102 GstAdapter * adapter, gint * offset, gint * length);
103 static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
104 GstBuffer * buffer);
105
106 static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstObject * parent,
107 GstBuffer * buffer);
108 static void gst_a52dec_set_property (GObject * object, guint prop_id,
109 const GValue * value, GParamSpec * pspec);
110 static void gst_a52dec_get_property (GObject * object, guint prop_id,
111 GValue * value, GParamSpec * pspec);
112
113 #define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type())
114 static GType
gst_a52dec_mode_get_type(void)115 gst_a52dec_mode_get_type (void)
116 {
117 static GType a52dec_mode_type = 0;
118 static const GEnumValue a52dec_modes[] = {
119 {A52_MONO, "Mono", "mono"},
120 {A52_STEREO, "Stereo", "stereo"},
121 {A52_3F, "3 Front", "3f"},
122 {A52_2F1R, "2 Front, 1 Rear", "2f1r"},
123 {A52_3F1R, "3 Front, 1 Rear", "3f1r"},
124 {A52_2F2R, "2 Front, 2 Rear", "2f2r"},
125 {A52_3F2R, "3 Front, 2 Rear", "3f2r"},
126 {A52_DOLBY, "Dolby", "dolby"},
127 {0, NULL, NULL},
128 };
129
130 if (!a52dec_mode_type) {
131 a52dec_mode_type = g_enum_register_static ("GstA52DecMode", a52dec_modes);
132 }
133 return a52dec_mode_type;
134 }
135
136 static void
gst_a52dec_class_init(GstA52DecClass * klass)137 gst_a52dec_class_init (GstA52DecClass * klass)
138 {
139 GObjectClass *gobject_class;
140 GstElementClass *gstelement_class;
141 GstAudioDecoderClass *gstbase_class;
142 guint cpuflags = 0;
143
144 gobject_class = (GObjectClass *) klass;
145 gstelement_class = (GstElementClass *) klass;
146 gstbase_class = (GstAudioDecoderClass *) klass;
147
148 gobject_class->set_property = gst_a52dec_set_property;
149 gobject_class->get_property = gst_a52dec_get_property;
150
151 gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
152 gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
153 gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
154 gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
155 gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
156
157 /**
158 * GstA52Dec::drc
159 *
160 * Set to true to apply the recommended Dolby Digital dynamic range compression
161 * to the audio stream. Dynamic range compression makes loud sounds
162 * softer and soft sounds louder, so you can more easily listen
163 * to the stream without disturbing other people.
164 */
165 g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
166 g_param_spec_boolean ("drc", "Dynamic Range Compression",
167 "Use Dynamic Range Compression", FALSE,
168 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
169 /**
170 * GstA52Dec::mode
171 *
172 * Force a particular output channel configuration from the decoder. By default,
173 * the channel downmix (if any) is chosen automatically based on the downstream
174 * capabilities of the pipeline.
175 */
176 g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE,
177 g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)",
178 GST_TYPE_A52DEC_MODE, A52_3F2R,
179 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
180 /**
181 * GstA52Dec::lfe
182 *
183 * Whether to output the LFE (Low Frequency Emitter) channel of the audio stream.
184 */
185 g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE,
186 g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE,
187 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
188
189 gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
190 gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
191 gst_element_class_set_static_metadata (gstelement_class,
192 "ATSC A/52 audio decoder", "Codec/Decoder/Audio",
193 "Decodes ATSC A/52 encoded audio streams",
194 "David I. Lehn <dlehn@users.sourceforge.net>");
195
196 GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
197 "AC3/A52 software decoder");
198
199 /* If no CPU instruction based acceleration is available, end up using the
200 * generic software djbfft based one when available in the used liba52 */
201 #ifdef MM_ACCEL_DJBFFT
202 klass->a52_cpuflags = MM_ACCEL_DJBFFT;
203 #elif defined(A52_ACCEL_DETECT)
204 klass->a52_cpuflags = A52_ACCEL_DETECT;
205 #else
206 klass->a52_cpuflags = 0;
207 #endif
208
209 #if HAVE_ORC && !defined(A52_ACCEL_DETECT)
210 cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx"));
211 if (cpuflags & ORC_TARGET_MMX_MMX)
212 klass->a52_cpuflags |= MM_ACCEL_X86_MMX;
213 if (cpuflags & ORC_TARGET_MMX_3DNOW)
214 klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW;
215 if (cpuflags & ORC_TARGET_MMX_MMXEXT)
216 klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
217 #endif
218
219 GST_LOG ("CPU flags: a52=%08x, orc=%08x", klass->a52_cpuflags, cpuflags);
220 }
221
222 static void
gst_a52dec_init(GstA52Dec * a52dec)223 gst_a52dec_init (GstA52Dec * a52dec)
224 {
225 a52dec->request_channels = A52_CHANNEL;
226 a52dec->dynamic_range_compression = FALSE;
227
228 a52dec->state = NULL;
229 a52dec->samples = NULL;
230
231 gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
232 (a52dec), TRUE);
233 GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (a52dec));
234
235 /* retrieve and intercept base class chain.
236 * Quite HACKish, but that's dvd specs/caps for you,
237 * since one buffer needs to be split into 2 frames */
238 a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
239 gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
240 GST_DEBUG_FUNCPTR (gst_a52dec_chain));
241 }
242
243 static gboolean
gst_a52dec_start(GstAudioDecoder * dec)244 gst_a52dec_start (GstAudioDecoder * dec)
245 {
246 GstA52Dec *a52dec = GST_A52DEC (dec);
247 GstA52DecClass *klass;
248 static GMutex init_mutex;
249
250 GST_DEBUG_OBJECT (dec, "start");
251
252 klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
253 g_mutex_lock (&init_mutex);
254 #if defined(A52_ACCEL_DETECT)
255 a52dec->state = a52_init ();
256 /* This line is just to avoid being accused of not using klass */
257 a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT);
258 #else
259 a52dec->state = a52_init (klass->a52_cpuflags);
260 #endif
261 g_mutex_unlock (&init_mutex);
262
263 if (!a52dec->state) {
264 GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL),
265 ("failed to initialize a52 state"));
266 return FALSE;
267 }
268
269 a52dec->samples = a52_samples (a52dec->state);
270 a52dec->bit_rate = -1;
271 a52dec->sample_rate = -1;
272 a52dec->stream_channels = A52_CHANNEL;
273 a52dec->using_channels = A52_CHANNEL;
274 a52dec->level = 1;
275 a52dec->bias = 0;
276 a52dec->flag_update = TRUE;
277
278 /* call upon legacy upstream byte support (e.g. seeking) */
279 gst_audio_decoder_set_estimate_rate (dec, TRUE);
280
281 return TRUE;
282 }
283
284 static gboolean
gst_a52dec_stop(GstAudioDecoder * dec)285 gst_a52dec_stop (GstAudioDecoder * dec)
286 {
287 GstA52Dec *a52dec = GST_A52DEC (dec);
288
289 GST_DEBUG_OBJECT (dec, "stop");
290
291 a52dec->samples = NULL;
292 if (a52dec->state) {
293 a52_free (a52dec->state);
294 a52dec->state = NULL;
295 }
296
297 return TRUE;
298 }
299
300 static GstFlowReturn
gst_a52dec_parse(GstAudioDecoder * bdec,GstAdapter * adapter,gint * _offset,gint * len)301 gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
302 gint * _offset, gint * len)
303 {
304 GstA52Dec *a52dec;
305 const guint8 *data;
306 gint av, size;
307 gint length = 0, flags, sample_rate, bit_rate;
308 GstFlowReturn result = GST_FLOW_EOS;
309
310 a52dec = GST_A52DEC (bdec);
311
312 size = av = gst_adapter_available (adapter);
313 data = (const guint8 *) gst_adapter_map (adapter, av);
314
315 /* find and read header */
316 bit_rate = a52dec->bit_rate;
317 sample_rate = a52dec->sample_rate;
318 flags = 0;
319 while (size >= 7) {
320 length = a52_syncinfo ((guint8 *) data, &flags, &sample_rate, &bit_rate);
321
322 if (length == 0) {
323 /* shift window to re-find sync */
324 data++;
325 size--;
326 } else if (length <= size) {
327 GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
328 result = GST_FLOW_OK;
329 break;
330 } else {
331 GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
332 length, size);
333 break;
334 }
335 }
336 gst_adapter_unmap (adapter);
337
338 *_offset = av - size;
339 *len = length;
340
341 return result;
342 }
343
344 static gint
gst_a52dec_channels(int flags,GstAudioChannelPosition * pos)345 gst_a52dec_channels (int flags, GstAudioChannelPosition * pos)
346 {
347 gint chans = 0;
348
349 if (flags & A52_LFE) {
350 chans += 1;
351 if (pos) {
352 pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE1;
353 }
354 }
355 flags &= A52_CHANNEL_MASK;
356 switch (flags) {
357 case A52_3F2R:
358 if (pos) {
359 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
360 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
361 pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
362 pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
363 pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
364 }
365 chans += 5;
366 break;
367 case A52_2F2R:
368 if (pos) {
369 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
370 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
371 pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
372 pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
373 }
374 chans += 4;
375 break;
376 case A52_3F1R:
377 if (pos) {
378 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
379 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
380 pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
381 pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
382 }
383 chans += 4;
384 break;
385 case A52_2F1R:
386 if (pos) {
387 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
388 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
389 pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
390 }
391 chans += 3;
392 break;
393 case A52_3F:
394 if (pos) {
395 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
396 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
397 pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
398 }
399 chans += 3;
400 break;
401 case A52_CHANNEL: /* Dual mono. Should really be handled as 2 src pads */
402 case A52_STEREO:
403 case A52_DOLBY:
404 if (pos) {
405 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
406 pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
407 }
408 chans += 2;
409 break;
410 case A52_MONO:
411 if (pos) {
412 pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_MONO;
413 }
414 chans += 1;
415 break;
416 default:
417 /* error, caller should post error message */
418 return 0;
419 }
420
421 return chans;
422 }
423
424 static gboolean
gst_a52dec_reneg(GstA52Dec * a52dec)425 gst_a52dec_reneg (GstA52Dec * a52dec)
426 {
427 gint channels;
428 gboolean result = FALSE;
429 GstAudioChannelPosition from[6], to[6];
430 GstAudioInfo info;
431
432 channels = gst_a52dec_channels (a52dec->using_channels, from);
433
434 if (!channels)
435 goto done;
436
437 GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d",
438 channels, a52dec->sample_rate);
439
440 memcpy (to, from, sizeof (GstAudioChannelPosition) * channels);
441 gst_audio_channel_positions_to_valid_order (to, channels);
442 gst_audio_get_channel_reorder_map (channels, from, to,
443 a52dec->channel_reorder_map);
444
445 gst_audio_info_init (&info);
446 gst_audio_info_set_format (&info,
447 SAMPLE_TYPE, a52dec->sample_rate, channels, (channels > 1 ? to : NULL));
448
449 if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (a52dec), &info))
450 goto done;
451
452 result = TRUE;
453
454 done:
455 return result;
456 }
457
458 static void
gst_a52dec_update_streaminfo(GstA52Dec * a52dec)459 gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
460 {
461 GstTagList *taglist;
462
463 taglist = gst_tag_list_new_empty ();
464 gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
465 (guint) a52dec->bit_rate, NULL);
466
467 gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (a52dec), taglist,
468 GST_TAG_MERGE_REPLACE);
469 gst_tag_list_unref (taglist);
470 }
471
472 static GstFlowReturn
gst_a52dec_handle_frame(GstAudioDecoder * bdec,GstBuffer * buffer)473 gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
474 {
475 GstA52Dec *a52dec;
476 gint channels, i;
477 gboolean need_reneg = FALSE;
478 gint chans;
479 gint length = 0, flags, sample_rate, bit_rate;
480 GstMapInfo map;
481 GstFlowReturn result = GST_FLOW_OK;
482 GstBuffer *outbuf;
483 const gint num_blocks = 6;
484
485 a52dec = GST_A52DEC (bdec);
486
487 /* no fancy draining */
488 if (G_UNLIKELY (!buffer))
489 return GST_FLOW_OK;
490
491 /* parsed stuff already, so this should work out fine */
492 gst_buffer_map (buffer, &map, GST_MAP_READ);
493 g_assert (map.size >= 7);
494
495 /* re-obtain some sync header info,
496 * should be same as during _parse and could also be cached there,
497 * but anyway ... */
498 bit_rate = a52dec->bit_rate;
499 sample_rate = a52dec->sample_rate;
500 flags = 0;
501 length = a52_syncinfo (map.data, &flags, &sample_rate, &bit_rate);
502 g_assert (length == map.size);
503
504 /* update stream information, renegotiate or re-streaminfo if needed */
505 need_reneg = FALSE;
506 if (a52dec->sample_rate != sample_rate) {
507 GST_DEBUG_OBJECT (a52dec, "sample rate changed");
508 need_reneg = TRUE;
509 a52dec->sample_rate = sample_rate;
510 }
511
512 if (flags) {
513 if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) {
514 GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update");
515 a52dec->flag_update = TRUE;
516 }
517 a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
518 }
519
520 if (bit_rate != a52dec->bit_rate) {
521 a52dec->bit_rate = bit_rate;
522 gst_a52dec_update_streaminfo (a52dec);
523 }
524
525 /* If we haven't had an explicit number of channels chosen through properties
526 * at this point, choose what to downmix to now, based on what the peer will
527 * accept - this allows a52dec to do downmixing in preference to a
528 * downstream element such as audioconvert.
529 */
530 if (a52dec->request_channels != A52_CHANNEL) {
531 flags = a52dec->request_channels;
532 } else if (a52dec->flag_update) {
533 GstCaps *caps;
534
535 a52dec->flag_update = FALSE;
536
537 caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
538 if (caps && gst_caps_get_size (caps) > 0) {
539 GstCaps *copy = gst_caps_copy_nth (caps, 0);
540 GstStructure *structure = gst_caps_get_structure (copy, 0);
541 gint orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6;
542 gint fixed_channels = 0;
543 const int a52_channels[6] = {
544 A52_MONO,
545 A52_STEREO,
546 A52_STEREO | A52_LFE,
547 A52_2F2R,
548 A52_2F2R | A52_LFE,
549 A52_3F2R | A52_LFE,
550 };
551
552 /* Prefer the original number of channels, but fixate to something
553 * preferred (first in the caps) downstream if possible.
554 */
555 gst_structure_fixate_field_nearest_int (structure, "channels",
556 orig_channels);
557
558 if (gst_structure_get_int (structure, "channels", &fixed_channels)
559 && fixed_channels <= 6) {
560 if (fixed_channels < orig_channels)
561 flags = a52_channels[fixed_channels - 1];
562 } else {
563 flags = a52_channels[5];
564 }
565
566 gst_caps_unref (copy);
567 } else if (flags)
568 flags = a52dec->stream_channels;
569 else
570 flags = A52_3F2R | A52_LFE;
571
572 if (caps)
573 gst_caps_unref (caps);
574 } else {
575 flags = a52dec->using_channels;
576 }
577
578 /* process */
579 flags |= A52_ADJUST_LEVEL;
580 a52dec->level = 1;
581 if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) {
582 gst_buffer_unmap (buffer, &map);
583 GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
584 ("a52_frame error"), result);
585 goto exit;
586 }
587 gst_buffer_unmap (buffer, &map);
588
589 channels = flags & (A52_CHANNEL_MASK | A52_LFE);
590 if (a52dec->using_channels != channels) {
591 need_reneg = TRUE;
592 a52dec->using_channels = channels;
593 }
594
595 /* negotiate if required */
596 if (need_reneg) {
597 GST_DEBUG_OBJECT (a52dec,
598 "a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
599 a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
600 if (!gst_a52dec_reneg (a52dec))
601 goto failed_negotiation;
602 }
603
604 if (a52dec->dynamic_range_compression == FALSE) {
605 a52_dynrng (a52dec->state, NULL, NULL);
606 }
607
608 flags &= (A52_CHANNEL_MASK | A52_LFE);
609 chans = gst_a52dec_channels (flags, NULL);
610 if (!chans)
611 goto invalid_flags;
612
613 /* handle decoded data;
614 * each frame has 6 blocks, one block is 256 samples, ea */
615 outbuf =
616 gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks);
617
618 gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
619 {
620 guint8 *ptr = map.data;
621 for (i = 0; i < num_blocks; i++) {
622 if (a52_block (a52dec->state)) {
623 /* also marks discont */
624 GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
625 ("error decoding block %d", i), result);
626 if (result != GST_FLOW_OK) {
627 gst_buffer_unmap (outbuf, &map);
628 gst_buffer_unref (outbuf);
629 goto exit;
630 }
631 } else {
632 gint n, c;
633 gint *reorder_map = a52dec->channel_reorder_map;
634
635 for (n = 0; n < 256; n++) {
636 for (c = 0; c < chans; c++) {
637 ((sample_t *) ptr)[n * chans + reorder_map[c]] =
638 a52dec->samples[c * 256 + n];
639 }
640 }
641 }
642 ptr += 256 * chans * (SAMPLE_WIDTH / 8);
643 }
644 }
645 gst_buffer_unmap (outbuf, &map);
646
647 result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
648
649 exit:
650 return result;
651
652 /* ERRORS */
653 failed_negotiation:
654 {
655 GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
656 return GST_FLOW_ERROR;
657 }
658 invalid_flags:
659 {
660 GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
661 ("Invalid channel flags: %d", flags));
662 return GST_FLOW_ERROR;
663 }
664 }
665
666 static gboolean
gst_a52dec_set_format(GstAudioDecoder * bdec,GstCaps * caps)667 gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
668 {
669 GstA52Dec *a52dec = GST_A52DEC (bdec);
670 GstStructure *structure;
671
672 structure = gst_caps_get_structure (caps, 0);
673
674 if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3"))
675 a52dec->dvdmode = TRUE;
676 else
677 a52dec->dvdmode = FALSE;
678
679 return TRUE;
680 }
681
682 static GstFlowReturn
gst_a52dec_chain(GstPad * pad,GstObject * parent,GstBuffer * buf)683 gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
684 {
685 GstA52Dec *a52dec = GST_A52DEC (parent);
686 GstFlowReturn ret = GST_FLOW_OK;
687 gint first_access;
688
689 if (a52dec->dvdmode) {
690 gsize size;
691 guint8 data[2];
692 gint offset;
693 gint len;
694 GstBuffer *subbuf;
695
696 size = gst_buffer_get_size (buf);
697 if (size < 2)
698 goto not_enough_data;
699
700 gst_buffer_extract (buf, 0, data, 2);
701 first_access = (data[0] << 8) | data[1];
702
703 /* Skip the first_access header */
704 offset = 2;
705
706 if (first_access > 1) {
707 /* Length of data before first_access */
708 len = first_access - 1;
709
710 if (len <= 0 || offset + len > size)
711 goto bad_first_access_parameter;
712
713 subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
714 GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
715 ret = a52dec->base_chain (pad, parent, subbuf);
716 if (ret != GST_FLOW_OK) {
717 gst_buffer_unref (buf);
718 goto done;
719 }
720
721 offset += len;
722 len = size - offset;
723
724 if (len > 0) {
725 subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len);
726 GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
727
728 ret = a52dec->base_chain (pad, parent, subbuf);
729 }
730 gst_buffer_unref (buf);
731 } else {
732 /* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
733 subbuf =
734 gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset,
735 size - offset);
736 GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
737 gst_buffer_unref (buf);
738 ret = a52dec->base_chain (pad, parent, subbuf);
739 }
740 } else {
741 ret = a52dec->base_chain (pad, parent, buf);
742 }
743
744 done:
745 return ret;
746
747 /* ERRORS */
748 not_enough_data:
749 {
750 GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
751 ("Insufficient data in buffer. Can't determine first_acess"));
752 gst_buffer_unref (buf);
753 return GST_FLOW_ERROR;
754 }
755 bad_first_access_parameter:
756 {
757 GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
758 ("Bad first_access parameter (%d) in buffer", first_access));
759 gst_buffer_unref (buf);
760 return GST_FLOW_ERROR;
761 }
762 }
763
764 static void
gst_a52dec_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)765 gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
766 GParamSpec * pspec)
767 {
768 GstA52Dec *src = GST_A52DEC (object);
769
770 switch (prop_id) {
771 case ARG_DRC:
772 GST_OBJECT_LOCK (src);
773 src->dynamic_range_compression = g_value_get_boolean (value);
774 GST_OBJECT_UNLOCK (src);
775 break;
776 case ARG_MODE:
777 GST_OBJECT_LOCK (src);
778 src->request_channels &= ~A52_CHANNEL_MASK;
779 src->request_channels |= g_value_get_enum (value);
780 GST_OBJECT_UNLOCK (src);
781 break;
782 case ARG_LFE:
783 GST_OBJECT_LOCK (src);
784 src->request_channels &= ~A52_LFE;
785 src->request_channels |= g_value_get_boolean (value) ? A52_LFE : 0;
786 GST_OBJECT_UNLOCK (src);
787 break;
788 default:
789 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
790 break;
791 }
792 }
793
794 static void
gst_a52dec_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)795 gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
796 GParamSpec * pspec)
797 {
798 GstA52Dec *src = GST_A52DEC (object);
799
800 switch (prop_id) {
801 case ARG_DRC:
802 GST_OBJECT_LOCK (src);
803 g_value_set_boolean (value, src->dynamic_range_compression);
804 GST_OBJECT_UNLOCK (src);
805 break;
806 case ARG_MODE:
807 GST_OBJECT_LOCK (src);
808 g_value_set_enum (value, src->request_channels & A52_CHANNEL_MASK);
809 GST_OBJECT_UNLOCK (src);
810 break;
811 case ARG_LFE:
812 GST_OBJECT_LOCK (src);
813 g_value_set_boolean (value, src->request_channels & A52_LFE);
814 GST_OBJECT_UNLOCK (src);
815 break;
816 default:
817 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
818 break;
819 }
820 }
821
822 static gboolean
plugin_init(GstPlugin * plugin)823 plugin_init (GstPlugin * plugin)
824 {
825 #if HAVE_ORC
826 orc_init ();
827 #endif
828
829 if (!gst_element_register (plugin, "a52dec", GST_RANK_SECONDARY,
830 GST_TYPE_A52DEC))
831 return FALSE;
832
833 return TRUE;
834 }
835
836 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
837 GST_VERSION_MINOR,
838 a52dec,
839 "Decodes ATSC A/52 encoded audio streams",
840 plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
841