1 /* GStreamer AAC parser plugin
2 * Copyright (C) 2008 Nokia Corporation. All rights reserved.
4 * Contact: Stefan Kost <stefan.kost@nokia.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-aacparse
24 * @short_description: AAC parser
25 * @see_also: #GstAmrParse
27 * This is an AAC parser which handles both ADIF and ADTS stream formats.
29 * As ADIF format is not framed, it is not seekable and stream duration cannot
30 * be determined either. However, ADTS format AAC clips can be seeked, and parser
31 * can also estimate playback position and clip duration.
34 * <title>Example launch line</title>
36 * gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
47 #include <gst/base/gstbitreader.h>
48 #include <gst/pbutils/pbutils.h>
49 #include "gstaacparse.h"
52 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
55 GST_STATIC_CAPS ("audio/mpeg, "
56 "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, "
57 "stream-format = (string) { raw, adts, adif, loas };"));
59 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
62 GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) { 2, 4 };"));
64 GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
65 #define GST_CAT_DEFAULT aacparse_debug
68 #define ADIF_MAX_SIZE 40 /* Should be enough */
69 #define ADTS_MAX_SIZE 10 /* Should be enough */
70 #define LOAS_MAX_SIZE 3 /* Should be enough */
72 #define ADTS_HEADERS_LENGTH 7UL /* Total byte-length of fixed and variable
73 headers prepended during raw to ADTS
76 #define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
78 static const gint loas_sample_rate_table[32] = {
79 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
80 16000, 12000, 11025, 8000, 7350, 0, 0, 0
83 static const gint loas_channels_table[32] = {
84 0, 1, 2, 3, 4, 5, 6, 8,
85 0, 0, 0, 7, 8, 0, 8, 0
88 static gboolean gst_aac_parse_start (GstBaseParse * parse);
89 static gboolean gst_aac_parse_stop (GstBaseParse * parse);
91 static gboolean gst_aac_parse_sink_setcaps (GstBaseParse * parse,
93 static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
96 static GstFlowReturn gst_aac_parse_handle_frame (GstBaseParse * parse,
97 GstBaseParseFrame * frame, gint * skipsize);
98 static GstFlowReturn gst_aac_parse_pre_push_frame (GstBaseParse * parse,
99 GstBaseParseFrame * frame);
101 G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
104 * gst_aac_parse_class_init:
105 * @klass: #GstAacParseClass.
109 gst_aac_parse_class_init (GstAacParseClass * klass)
111 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
112 GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
114 GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
115 "AAC audio stream parser");
117 gst_element_class_add_pad_template (element_class,
118 gst_static_pad_template_get (&sink_template));
119 gst_element_class_add_pad_template (element_class,
120 gst_static_pad_template_get (&src_template));
122 gst_element_class_set_static_metadata (element_class,
123 "AAC audio stream parser", "Codec/Parser/Audio",
124 "Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
126 parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
127 parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
128 parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
129 parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
130 parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_handle_frame);
131 parse_class->pre_push_frame =
132 GST_DEBUG_FUNCPTR (gst_aac_parse_pre_push_frame);
137 * gst_aac_parse_init:
138 * @aacparse: #GstAacParse.
139 * @klass: #GstAacParseClass.
143 gst_aac_parse_init (GstAacParse * aacparse)
145 GST_DEBUG ("initialized");
146 GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
147 GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (aacparse));
152 * gst_aac_parse_set_src_caps:
153 * @aacparse: #GstAacParse.
154 * @sink_caps: (proposed) caps of sink pad
156 * Set source pad caps according to current knowledge about the
159 * Returns: TRUE if caps were successfully set.
162 gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
165 GstCaps *src_caps = NULL, *allowed;
166 gboolean res = FALSE;
167 const gchar *stream_format;
168 guint8 codec_data[2];
169 guint16 codec_data_data;
170 gint sample_rate_idx;
172 GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
174 src_caps = gst_caps_copy (sink_caps);
176 src_caps = gst_caps_new_empty_simple ("audio/mpeg");
178 gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
179 "mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
181 aacparse->output_header_type = aacparse->header_type;
182 switch (aacparse->header_type) {
183 case DSPAAC_HEADER_NONE:
184 stream_format = "raw";
186 case DSPAAC_HEADER_ADTS:
187 stream_format = "adts";
189 case DSPAAC_HEADER_ADIF:
190 stream_format = "adif";
192 case DSPAAC_HEADER_LOAS:
193 stream_format = "loas";
196 stream_format = NULL;
199 /* Generate codec data to be able to set profile/level on the caps */
201 gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
202 if (sample_rate_idx < 0)
203 goto not_a_known_rate;
205 (aacparse->object_type << 11) |
206 (sample_rate_idx << 7) | (aacparse->channels << 3);
207 GST_WRITE_UINT16_BE (codec_data, codec_data_data);
208 gst_codec_utils_aac_caps_set_level_and_profile (src_caps, codec_data, 2);
210 s = gst_caps_get_structure (src_caps, 0);
211 if (aacparse->sample_rate > 0)
212 gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
213 if (aacparse->channels > 0)
214 gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
216 gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
218 allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
219 if (allowed && !gst_caps_can_intersect (src_caps, allowed)) {
220 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
221 "Caps can not intersect");
222 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
223 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
224 "Input is ADTS, trying raw");
225 gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
227 if (gst_caps_can_intersect (src_caps, allowed)) {
228 GstBuffer *codec_data_buffer;
230 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
231 "Caps can intersect, we will drop the ADTS layer");
232 aacparse->output_header_type = DSPAAC_HEADER_NONE;
234 /* The codec_data data is according to AudioSpecificConfig,
235 ISO/IEC 14496-3, 1.6.2.1 */
236 codec_data_buffer = gst_buffer_new_and_alloc (2);
237 gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
238 gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
239 codec_data_buffer, NULL);
241 } else if (aacparse->header_type == DSPAAC_HEADER_NONE) {
242 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
243 "Input is raw, trying ADTS");
244 gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
246 if (gst_caps_can_intersect (src_caps, allowed)) {
247 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
248 "Caps can intersect, we will prepend ADTS headers");
249 aacparse->output_header_type = DSPAAC_HEADER_ADTS;
254 gst_caps_unref (allowed);
256 GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
258 res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
259 gst_caps_unref (src_caps);
263 GST_ERROR_OBJECT (aacparse, "Not a known sample rate: %d",
264 aacparse->sample_rate);
265 gst_caps_unref (src_caps);
271 * gst_aac_parse_sink_setcaps:
275 * Implementation of "set_sink_caps" vmethod in #GstBaseParse class.
277 * Returns: TRUE on success.
280 gst_aac_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
282 GstAacParse *aacparse;
283 GstStructure *structure;
287 aacparse = GST_AAC_PARSE (parse);
288 structure = gst_caps_get_structure (caps, 0);
289 caps_str = gst_caps_to_string (caps);
291 GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str);
294 /* This is needed at least in case of RTP
295 * Parses the codec_data information to get ObjectType,
296 * number of channels and samplerate */
297 value = gst_structure_get_value (structure, "codec_data");
299 GstBuffer *buf = gst_value_get_buffer (value);
305 gst_buffer_map (buf, &map, GST_MAP_READ);
307 sr_idx = ((map.data[0] & 0x07) << 1) | ((map.data[1] & 0x80) >> 7);
308 aacparse->object_type = (map.data[0] & 0xf8) >> 3;
309 aacparse->sample_rate =
310 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
311 aacparse->channels = (map.data[1] & 0x78) >> 3;
312 if (aacparse->channels == 7)
313 aacparse->channels = 8;
314 else if (aacparse->channels == 11)
315 aacparse->channels = 7;
316 else if (aacparse->channels == 12 || aacparse->channels == 14)
317 aacparse->channels = 8;
318 aacparse->header_type = DSPAAC_HEADER_NONE;
319 aacparse->mpegversion = 4;
320 aacparse->frame_samples = (map.data[1] & 4) ? 960 : 1024;
321 gst_buffer_unmap (buf, &map);
323 GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
324 "samples=%d", aacparse->object_type, aacparse->sample_rate,
325 aacparse->channels, aacparse->frame_samples);
327 /* arrange for metadata and get out of the way */
328 gst_aac_parse_set_src_caps (aacparse, caps);
329 if (aacparse->header_type == aacparse->output_header_type)
330 gst_base_parse_set_passthrough (parse, TRUE);
334 /* caps info overrides */
335 gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
336 gst_structure_get_int (structure, "channels", &aacparse->channels);
338 aacparse->sample_rate = 0;
339 aacparse->channels = 0;
340 aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
341 gst_base_parse_set_passthrough (parse, FALSE);
349 * gst_aac_parse_adts_get_frame_len:
350 * @data: block of data containing an ADTS header.
352 * This function calculates ADTS frame length from the given header.
354 * Returns: size of the ADTS frame.
357 gst_aac_parse_adts_get_frame_len (const guint8 * data)
359 return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
364 * gst_aac_parse_check_adts_frame:
365 * @aacparse: #GstAacParse.
366 * @data: Data to be checked.
367 * @avail: Amount of data passed.
368 * @framesize: If valid ADTS frame was found, this will be set to tell the
369 * found frame size in bytes.
370 * @needed_data: If frame was not found, this may be set to tell how much
371 * more data is needed in the next round to detect the frame
372 * reliably. This may happen when a frame header candidate
373 * is found but it cannot be guaranteed to be the header without
374 * peeking the following data.
376 * Check if the given data contains contains ADTS frame. The algorithm
377 * will examine ADTS frame header and calculate the frame size. Also, another
378 * consecutive ADTS frame header need to be present after the found frame.
379 * Otherwise the data is not considered as a valid ADTS frame. However, this
380 * "extra check" is omitted when EOS has been received. In this case it is
381 * enough when data[0] contains a valid ADTS header.
383 * This function may set the #needed_data to indicate that a possible frame
384 * candidate has been found, but more data (#needed_data bytes) is needed to
385 * be absolutely sure. When this situation occurs, FALSE will be returned.
387 * When a valid frame is detected, this function will use
388 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
389 * to set the needed bytes for next frame.This way next data chunk is already
392 * Returns: TRUE if the given data contains a valid ADTS header.
395 gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
396 const guint8 * data, const guint avail, gboolean drain,
397 guint * framesize, guint * needed_data)
403 /* Absolute minimum to perform the ADTS syncword,
404 layer and sampling frequency tests */
405 if (G_UNLIKELY (avail < 3)) {
410 /* Syncword and layer tests */
411 if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
413 /* Sampling frequency test */
414 if (G_UNLIKELY ((data[2] & 0x3C) >> 2 == 15))
417 /* This looks like an ADTS frame header but
418 we need at least 6 bytes to proceed */
419 if (G_UNLIKELY (avail < 6)) {
424 *framesize = gst_aac_parse_adts_get_frame_len (data);
426 /* If frame has CRC, it needs 2 bytes
427 for it at the end of the header */
428 crc_size = (data[1] & 0x01) ? 0 : 2;
431 if (*framesize < 7 + crc_size) {
432 *needed_data = 7 + crc_size;
436 /* In EOS mode this is enough. No need to examine the data further.
437 We also relax the check when we have sync, on the assumption that
438 if we're not looking at random data, we have a much higher chance
439 to get the correct sync, and this avoids losing two frames when
440 a single bit corruption happens. */
441 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
445 if (*framesize + ADTS_MAX_SIZE > avail) {
446 /* We have found a possible frame header candidate, but can't be
447 sure since we don't have enough data to check the next frame */
448 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
449 *framesize + ADTS_MAX_SIZE, avail);
450 *needed_data = *framesize + ADTS_MAX_SIZE;
451 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
452 *framesize + ADTS_MAX_SIZE);
456 if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
457 guint nextlen = gst_aac_parse_adts_get_frame_len (data + (*framesize));
459 GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
460 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
461 nextlen + ADTS_MAX_SIZE);
469 gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
472 guint8 bytes, i, byte;
475 if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
477 for (i = 0; i < bytes; ++i) {
479 if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
487 gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
488 guint8 * audio_object_type)
490 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
492 if (*audio_object_type == 31) {
493 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
495 *audio_object_type += 32;
497 GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
502 gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
505 guint8 sampling_frequency_index;
506 if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
508 GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
509 sampling_frequency_index);
510 if (sampling_frequency_index == 0xf) {
511 guint32 sampling_rate;
512 if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
514 *sample_rate = sampling_rate;
516 *sample_rate = loas_sample_rate_table[sampling_frequency_index];
523 /* See table 1.13 in ISO/IEC 14496-3 */
525 gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
526 GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
528 guint8 audio_object_type, channel_configuration;
530 if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
533 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
536 if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
538 GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
539 *channels = loas_channels_table[channel_configuration];
543 if (audio_object_type == 5) {
544 GST_LOG_OBJECT (aacparse,
545 "Audio object type 5, so rereading sampling rate...");
546 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
550 GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
551 *sample_rate, *channels);
553 /* There's LOTS of stuff next, but we ignore it for now as we have
554 what we want (sample rate and number of channels */
555 GST_DEBUG_OBJECT (aacparse,
556 "Need more code to parse humongous LOAS data, currently ignored");
564 gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
565 guint avail, gint * sample_rate, gint * channels, gint * version)
570 /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
574 gst_bit_reader_init (&br, data, avail);
576 /* skip sync word (11 bits) and size (13 bits) */
577 if (!gst_bit_reader_skip (&br, 11 + 13))
580 /* First bit is "use last config" */
581 if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
584 GST_LOG_OBJECT (aacparse, "Frame uses previous config");
585 if (!aacparse->sample_rate || !aacparse->channels) {
586 GST_DEBUG_OBJECT (aacparse,
587 "No previous config to use. We'll look for more data.");
590 *sample_rate = aacparse->sample_rate;
591 *channels = aacparse->channels;
595 GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
597 if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
600 if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
605 GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
607 guint8 same_time, subframes, num_program, prog;
610 if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
613 if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
615 if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
617 if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
619 GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
620 same_time, subframes, num_program);
622 for (prog = 0; prog <= num_program; ++prog) {
623 guint8 num_layer, layer;
624 if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
626 GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
628 for (layer = 0; layer <= num_layer; ++layer) {
629 guint8 use_same_config;
630 if (prog == 0 && layer == 0) {
633 if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
636 if (!use_same_config) {
638 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
639 sample_rate, channels, NULL))
642 guint32 bits, asc_len;
643 if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
645 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
646 sample_rate, channels, &bits))
649 if (!gst_bit_reader_skip (&br, asc_len))
655 GST_LOG_OBJECT (aacparse, "More data ignored");
657 GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
664 * gst_aac_parse_loas_get_frame_len:
665 * @data: block of data containing a LOAS header.
667 * This function calculates LOAS frame length from the given header.
669 * Returns: size of the LOAS frame.
672 gst_aac_parse_loas_get_frame_len (const guint8 * data)
674 return (((data[1] & 0x1f) << 8) | data[2]) + 3;
679 * gst_aac_parse_check_loas_frame:
680 * @aacparse: #GstAacParse.
681 * @data: Data to be checked.
682 * @avail: Amount of data passed.
683 * @framesize: If valid LOAS frame was found, this will be set to tell the
684 * found frame size in bytes.
685 * @needed_data: If frame was not found, this may be set to tell how much
686 * more data is needed in the next round to detect the frame
687 * reliably. This may happen when a frame header candidate
688 * is found but it cannot be guaranteed to be the header without
689 * peeking the following data.
691 * Check if the given data contains contains LOAS frame. The algorithm
692 * will examine LOAS frame header and calculate the frame size. Also, another
693 * consecutive LOAS frame header need to be present after the found frame.
694 * Otherwise the data is not considered as a valid LOAS frame. However, this
695 * "extra check" is omitted when EOS has been received. In this case it is
696 * enough when data[0] contains a valid LOAS header.
698 * This function may set the #needed_data to indicate that a possible frame
699 * candidate has been found, but more data (#needed_data bytes) is needed to
700 * be absolutely sure. When this situation occurs, FALSE will be returned.
702 * When a valid frame is detected, this function will use
703 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
704 * to set the needed bytes for next frame.This way next data chunk is already
707 * LOAS can have three different formats, if I read the spec correctly. Only
708 * one of them is supported here, as the two samples I have use this one.
710 * Returns: TRUE if the given data contains a valid LOAS header.
713 gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
714 const guint8 * data, const guint avail, gboolean drain,
715 guint * framesize, guint * needed_data)
720 if (G_UNLIKELY (avail < 3)) {
725 if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
726 *framesize = gst_aac_parse_loas_get_frame_len (data);
727 GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize);
729 /* In EOS mode this is enough. No need to examine the data further.
730 We also relax the check when we have sync, on the assumption that
731 if we're not looking at random data, we have a much higher chance
732 to get the correct sync, and this avoids losing two frames when
733 a single bit corruption happens. */
734 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
738 if (*framesize + LOAS_MAX_SIZE > avail) {
739 /* We have found a possible frame header candidate, but can't be
740 sure since we don't have enough data to check the next frame */
741 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
742 *framesize + LOAS_MAX_SIZE, avail);
743 *needed_data = *framesize + LOAS_MAX_SIZE;
744 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
745 *framesize + LOAS_MAX_SIZE);
749 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
750 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
752 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
753 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
754 nextlen + LOAS_MAX_SIZE);
761 /* caller ensure sufficient data */
763 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
764 gint * rate, gint * channels, gint * object, gint * version)
768 gint sr_idx = (data[2] & 0x3c) >> 2;
770 *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
773 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
779 *version = (data[1] & 0x08) ? 2 : 4;
781 *object = ((data[2] & 0xc0) >> 6) + 1;
785 * gst_aac_parse_detect_stream:
786 * @aacparse: #GstAacParse.
787 * @data: A block of data that needs to be examined for stream characteristics.
788 * @avail: Size of the given datablock.
789 * @framesize: If valid stream was found, this will be set to tell the
790 * first frame size in bytes.
791 * @skipsize: If valid stream was found, this will be set to tell the first
792 * audio frame position within the given data.
794 * Examines the given piece of data and try to detect the format of it. It
795 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
796 * header. If the stream is detected, TRUE will be returned and #framesize
797 * is set to indicate the found frame size. Additionally, #skipsize might
798 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
799 * position of the frame inside given data chunk.
801 * Returns: TRUE on success.
804 gst_aac_parse_detect_stream (GstAacParse * aacparse,
805 const guint8 * data, const guint avail, gboolean drain,
806 guint * framesize, gint * skipsize)
808 gboolean found = FALSE;
809 guint need_data_adts = 0, need_data_loas;
812 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
814 /* FIXME: No need to check for ADIF if we are not in the beginning of the
817 /* Can we even parse the header? */
818 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
819 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
823 for (i = 0; i < avail - 4; i++) {
824 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
825 ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
826 strncmp ((char *) data + i, "ADIF", 4) == 0) {
827 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
831 /* Trick: tell the parent class that we didn't find the frame yet,
832 but make it skip 'i' amount of bytes. Next time we arrive
833 here we have full frame in the beginning of the data. */
846 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
847 framesize, &need_data_adts)) {
850 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
852 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
853 &aacparse->object_type, &aacparse->mpegversion);
855 if (!channels || !framesize) {
856 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
860 aacparse->header_type = DSPAAC_HEADER_ADTS;
861 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
862 aacparse->frame_samples, 2, 2);
864 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
865 rate, channels, aacparse->object_type, aacparse->mpegversion);
867 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
872 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
873 framesize, &need_data_loas)) {
874 gint rate = 0, channels = 0;
876 GST_INFO ("LOAS, framesize: %d", *framesize);
878 aacparse->header_type = DSPAAC_HEADER_LOAS;
880 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
881 &channels, &aacparse->mpegversion)) {
882 /* This is pretty normal when skipping data at the start of
883 * random stream (MPEG-TS capture for example) */
884 GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
888 if (rate && channels) {
889 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
890 aacparse->frame_samples, 2, 2);
892 /* Don't store the sample rate and channels yet -
893 * this is just format detection. */
894 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
895 rate, channels, aacparse->object_type, aacparse->mpegversion);
898 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
903 if (need_data_adts || need_data_loas) {
904 /* This tells the parent class not to skip any data */
909 if (avail < ADIF_MAX_SIZE)
912 if (memcmp (data + i, "ADIF", 4) == 0) {
919 aacparse->header_type = DSPAAC_HEADER_ADIF;
920 aacparse->mpegversion = 4;
922 /* Skip the "ADIF" bytes */
925 /* copyright string */
927 skip_size += 9; /* skip 9 bytes */
929 bitstream_type = adif[0 + skip_size] & 0x10;
931 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
932 ((unsigned int) adif[1 + skip_size] << 11) |
933 ((unsigned int) adif[2 + skip_size] << 3) |
934 ((unsigned int) adif[3 + skip_size] & 0xe0);
937 if (bitstream_type == 0) {
939 /* Buffer fullness parsing. Currently not needed... */
943 num_elems = (adif[3 + skip_size] & 0x1e);
944 GST_INFO ("ADIF num_config_elems: %d", num_elems);
946 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
947 ((unsigned int) adif[4 + skip_size] << 11) |
948 ((unsigned int) adif[5 + skip_size] << 3) |
949 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
951 GST_INFO ("ADIF buffer fullness: %d", fullness);
953 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
954 ((adif[7 + skip_size] & 0x80) >> 7);
955 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
959 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
960 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
961 ((adif[5 + skip_size] & 0x80) >> 7);
964 /* FIXME: This gives totally wrong results. Duration calculation cannot
966 aacparse->sample_rate =
967 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
969 /* baseparse is not given any fps,
970 * so it will give up on timestamps, seeking, etc */
972 /* FIXME: Can we assume this? */
973 aacparse->channels = 2;
975 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
976 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
978 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
980 /* arrange for metadata and get out of the way */
981 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
982 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
984 gst_caps_unref (sinkcaps);
986 /* not syncable, not easily seekable (unless we push data from start */
987 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
988 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
989 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
995 /* This should never happen */
1000 * gst_aac_parse_get_audio_profile_object_type
1001 * @aacparse: #GstAacParse.
1003 * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
1004 * mpegversion and profile of @aacparse's src pad caps, according to the
1005 * values defined by table 1.A.11 in ISO/IEC 14496-3.
1007 * Returns: the profile or object type value corresponding to @aacparse's src
1008 * pad caps, if such a value exists; otherwise G_MAXUINT8.
1011 gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
1014 GstStructure *srcstruct;
1015 const gchar *profile;
1018 srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
1019 srcstruct = gst_caps_get_structure (srccaps, 0);
1020 profile = gst_structure_get_string (srcstruct, "profile");
1021 if (G_UNLIKELY (profile == NULL)) {
1022 gst_caps_unref (srccaps);
1026 if (g_strcmp0 (profile, "main") == 0) {
1028 } else if (g_strcmp0 (profile, "lc") == 0) {
1030 } else if (g_strcmp0 (profile, "ssr") == 0) {
1032 } else if (g_strcmp0 (profile, "ltp") == 0) {
1033 if (G_LIKELY (aacparse->mpegversion == 4))
1036 ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
1041 gst_caps_unref (srccaps);
1046 * gst_aac_parse_get_audio_channel_configuration
1047 * @num_channels: number of audio channels.
1049 * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
1050 * 14496-3, for a given number of audio channels.
1052 * Returns: the Channel Configuration value corresponding to @num_channels, if
1053 * such a value exists; otherwise G_MAXUINT8.
1056 gst_aac_parse_get_audio_channel_configuration (gint num_channels)
1058 if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
1059 return (guint8) num_channels;
1060 else if (num_channels == 8) /* 7.1 */
1065 /* FIXME: Add support for configurations 11, 12 and 14 from
1066 * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
1071 * gst_aac_parse_get_audio_sampling_frequency_index:
1072 * @sample_rate: audio sampling rate.
1074 * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
1075 * 14496-3, for a given sampling rate.
1077 * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
1078 * if such a value exists; otherwise G_MAXUINT8.
1081 gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
1083 switch (sample_rate) {
1116 * gst_aac_parse_prepend_adts_headers:
1117 * @aacparse: #GstAacParse.
1118 * @frame: raw AAC frame to which ADTS headers shall be prepended.
1120 * Prepends ADTS headers to a raw AAC audio frame.
1122 * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
1125 gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
1126 GstBaseParseFrame * frame)
1129 guint8 *adts_headers;
1132 guint8 id, profile, channel_configuration, sampling_frequency_index;
1134 id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
1135 profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
1136 if (profile == G_MAXUINT8) {
1137 GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
1140 channel_configuration =
1141 gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
1142 if (channel_configuration == G_MAXUINT8) {
1143 GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
1146 sampling_frequency_index =
1147 gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
1148 if (sampling_frequency_index == G_MAXUINT8) {
1149 GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
1153 frame->out_buffer = gst_buffer_copy (frame->buffer);
1154 buf_size = gst_buffer_get_size (frame->out_buffer);
1155 frame_size = buf_size + ADTS_HEADERS_LENGTH;
1157 if (G_UNLIKELY (frame_size >= 0x4000)) {
1158 GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
1162 adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
1164 /* Note: no error correction bits are added to the resulting ADTS frames */
1165 adts_headers[0] = 0xFFU;
1166 adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
1167 adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
1168 (channel_configuration & 0x4U);
1169 adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
1170 (guint8) (frame_size >> 11);
1171 adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
1172 adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
1173 adts_headers[6] = 0xFCU;
1175 mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
1176 ADTS_HEADERS_LENGTH, adts_headers, g_free);
1177 gst_buffer_prepend_memory (frame->out_buffer, mem);
1183 * gst_aac_parse_check_valid_frame:
1184 * @parse: #GstBaseParse.
1185 * @frame: #GstBaseParseFrame.
1186 * @skipsize: How much data parent class should skip in order to find the
1189 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
1191 * Also determines frame overhead.
1192 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
1193 * a per-frame header. LOAS has 3 bytes.
1195 * We're making a couple of simplifying assumptions:
1197 * 1. We count Program Configuration Elements rather than searching for them
1198 * in the streams to discount them - the overhead is negligible.
1200 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
1201 * bits, which should still not be significant enough to warrant the
1202 * additional parsing through the headers
1204 * Returns: a #GstFlowReturn.
1206 static GstFlowReturn
1207 gst_aac_parse_handle_frame (GstBaseParse * parse,
1208 GstBaseParseFrame * frame, gint * skipsize)
1211 GstAacParse *aacparse;
1212 gboolean ret = FALSE;
1216 gint rate = 0, channels = 0;
1218 aacparse = GST_AAC_PARSE (parse);
1219 buffer = frame->buffer;
1221 gst_buffer_map (buffer, &map, GST_MAP_READ);
1224 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
1226 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
1227 aacparse->header_type == DSPAAC_HEADER_NONE) {
1228 /* There is nothing to parse */
1229 framesize = map.size;
1232 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
1234 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
1235 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
1237 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1238 guint needed_data = 1024;
1240 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
1241 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1243 if (!ret && needed_data) {
1244 GST_DEBUG ("buffer didn't contain valid frame");
1246 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1250 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1251 guint needed_data = 1024;
1253 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
1254 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1256 if (!ret && needed_data) {
1257 GST_DEBUG ("buffer didn't contain valid frame");
1259 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1264 GST_DEBUG ("buffer didn't contain valid frame");
1265 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1269 if (G_UNLIKELY (!ret))
1272 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1274 frame->overhead = 7;
1276 gst_aac_parse_parse_adts_header (aacparse, map.data,
1277 &rate, &channels, NULL, NULL);
1279 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
1281 if (G_UNLIKELY (rate != aacparse->sample_rate
1282 || channels != aacparse->channels)) {
1283 aacparse->sample_rate = rate;
1284 aacparse->channels = channels;
1286 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1287 /* If linking fails, we need to return appropriate error */
1288 ret = GST_FLOW_NOT_LINKED;
1291 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1292 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1294 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1295 gboolean setcaps = FALSE;
1298 frame->overhead = 3;
1300 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1301 &channels, NULL) || !rate || !channels) {
1302 /* This is pretty normal when skipping data at the start of
1303 * random stream (MPEG-TS capture for example) */
1304 GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
1305 /* Since we don't fully parse the LOAS config, we don't know for sure
1306 * how much to skip. Just skip 1 to end up to the next marker and
1307 * resume parsing from there */
1312 if (G_UNLIKELY (rate != aacparse->sample_rate
1313 || channels != aacparse->channels)) {
1314 aacparse->sample_rate = rate;
1315 aacparse->channels = channels;
1317 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1321 /* We want to set caps both at start, and when rate/channels change.
1322 Since only some LOAS frames have that info, we may receive frames
1323 before knowing about rate/channels. */
1325 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1326 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1327 /* If linking fails, we need to return appropriate error */
1328 ret = GST_FLOW_NOT_LINKED;
1331 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1332 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1336 if (aacparse->header_type == DSPAAC_HEADER_NONE
1337 && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
1338 if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
1339 GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
1340 ret = GST_FLOW_ERROR;
1345 gst_buffer_unmap (buffer, &map);
1348 /* found, skip if needed */
1357 if (ret && framesize <= map.size) {
1358 return gst_base_parse_finish_frame (parse, frame, framesize);
1364 static GstFlowReturn
1365 gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
1367 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1369 if (!aacparse->sent_codec_tag) {
1370 GstTagList *taglist;
1373 taglist = gst_tag_list_new_empty ();
1376 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1377 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1378 GST_TAG_AUDIO_CODEC, caps);
1379 gst_caps_unref (caps);
1381 gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
1382 gst_tag_list_unref (taglist);
1384 /* also signals the end of first-frame processing */
1385 aacparse->sent_codec_tag = TRUE;
1388 /* As a special case, we can remove the ADTS framing and output raw AAC. */
1389 if (aacparse->header_type == DSPAAC_HEADER_ADTS
1390 && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
1393 gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
1394 header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
1395 gst_buffer_unmap (frame->buffer, &map);
1396 gst_buffer_resize (frame->buffer, header_size,
1397 gst_buffer_get_size (frame->buffer) - header_size);
1405 * gst_aac_parse_start:
1406 * @parse: #GstBaseParse.
1408 * Implementation of "start" vmethod in #GstBaseParse class.
1410 * Returns: TRUE if startup succeeded.
1413 gst_aac_parse_start (GstBaseParse * parse)
1415 GstAacParse *aacparse;
1417 aacparse = GST_AAC_PARSE (parse);
1418 GST_DEBUG ("start");
1419 aacparse->frame_samples = 1024;
1420 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1421 aacparse->sent_codec_tag = FALSE;
1427 * gst_aac_parse_stop:
1428 * @parse: #GstBaseParse.
1430 * Implementation of "stop" vmethod in #GstBaseParse class.
1432 * Returns: TRUE is stopping succeeded.
1435 gst_aac_parse_stop (GstBaseParse * parse)
1442 remove_fields (GstCaps * caps)
1446 n = gst_caps_get_size (caps);
1447 for (i = 0; i < n; i++) {
1448 GstStructure *s = gst_caps_get_structure (caps, i);
1450 gst_structure_remove_field (s, "framed");
1455 add_conversion_fields (GstCaps * caps)
1459 n = gst_caps_get_size (caps);
1460 for (i = 0; i < n; i++) {
1461 GstStructure *s = gst_caps_get_structure (caps, i);
1463 if (gst_structure_has_field (s, "stream-format")) {
1464 const GValue *v = gst_structure_get_value (s, "stream-format");
1466 if (G_VALUE_HOLDS_STRING (v)) {
1467 const gchar *str = g_value_get_string (v);
1469 if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
1470 GValue va = G_VALUE_INIT;
1471 GValue vs = G_VALUE_INIT;
1473 g_value_init (&va, GST_TYPE_LIST);
1474 g_value_init (&vs, G_TYPE_STRING);
1475 g_value_set_string (&vs, "adts");
1476 gst_value_list_append_value (&va, &vs);
1477 g_value_set_string (&vs, "raw");
1478 gst_value_list_append_value (&va, &vs);
1479 gst_structure_set_value (s, "stream-format", &va);
1480 g_value_unset (&va);
1481 g_value_unset (&vs);
1483 } else if (GST_VALUE_HOLDS_LIST (v)) {
1484 gboolean contains_raw = FALSE;
1485 gboolean contains_adts = FALSE;
1486 guint m = gst_value_list_get_size (v), j;
1488 for (j = 0; j < m; j++) {
1489 const GValue *ve = gst_value_list_get_value (v, j);
1492 if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
1493 if (strcmp (str, "adts") == 0)
1494 contains_adts = TRUE;
1495 else if (strcmp (str, "raw") == 0)
1496 contains_raw = TRUE;
1500 if (contains_adts || contains_raw) {
1501 GValue va = G_VALUE_INIT;
1502 GValue vs = G_VALUE_INIT;
1504 g_value_init (&va, GST_TYPE_LIST);
1505 g_value_init (&vs, G_TYPE_STRING);
1506 g_value_copy (v, &va);
1508 if (!contains_raw) {
1509 g_value_set_string (&vs, "raw");
1510 gst_value_list_append_value (&va, &vs);
1512 if (!contains_adts) {
1513 g_value_set_string (&vs, "adts");
1514 gst_value_list_append_value (&va, &vs);
1517 gst_structure_set_value (s, "stream-format", &va);
1519 g_value_unset (&vs);
1520 g_value_unset (&va);
1528 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1530 GstCaps *peercaps, *templ;
1533 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1536 GstCaps *fcopy = gst_caps_copy (filter);
1537 /* Remove the fields we convert */
1538 remove_fields (fcopy);
1539 add_conversion_fields (fcopy);
1540 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1541 gst_caps_unref (fcopy);
1543 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1546 peercaps = gst_caps_make_writable (peercaps);
1547 /* Remove the fields we convert */
1548 remove_fields (peercaps);
1549 add_conversion_fields (peercaps);
1551 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1552 gst_caps_unref (peercaps);
1553 gst_caps_unref (templ);
1559 GstCaps *intersection;
1562 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1563 gst_caps_unref (res);