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_DEBUG_OBJECT (aacparse, "Frame uses previous config");
585 if (!aacparse->sample_rate || !aacparse->channels) {
586 GST_WARNING_OBJECT (aacparse, "No previous config to use");
588 *sample_rate = aacparse->sample_rate;
589 *channels = aacparse->channels;
593 GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
595 if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
598 if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
603 GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
605 guint8 same_time, subframes, num_program, prog;
608 if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
611 if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
613 if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
615 if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
617 GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
618 same_time, subframes, num_program);
620 for (prog = 0; prog <= num_program; ++prog) {
621 guint8 num_layer, layer;
622 if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
624 GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
626 for (layer = 0; layer <= num_layer; ++layer) {
627 guint8 use_same_config;
628 if (prog == 0 && layer == 0) {
631 if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
634 if (!use_same_config) {
636 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
637 sample_rate, channels, NULL))
640 guint32 bits, asc_len;
641 if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
643 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
644 sample_rate, channels, &bits))
647 if (!gst_bit_reader_skip (&br, asc_len))
653 GST_LOG_OBJECT (aacparse, "More data ignored");
655 GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
661 * gst_aac_parse_loas_get_frame_len:
662 * @data: block of data containing a LOAS header.
664 * This function calculates LOAS frame length from the given header.
666 * Returns: size of the LOAS frame.
669 gst_aac_parse_loas_get_frame_len (const guint8 * data)
671 return (((data[1] & 0x1f) << 8) | data[2]) + 3;
676 * gst_aac_parse_check_loas_frame:
677 * @aacparse: #GstAacParse.
678 * @data: Data to be checked.
679 * @avail: Amount of data passed.
680 * @framesize: If valid LOAS frame was found, this will be set to tell the
681 * found frame size in bytes.
682 * @needed_data: If frame was not found, this may be set to tell how much
683 * more data is needed in the next round to detect the frame
684 * reliably. This may happen when a frame header candidate
685 * is found but it cannot be guaranteed to be the header without
686 * peeking the following data.
688 * Check if the given data contains contains LOAS frame. The algorithm
689 * will examine LOAS frame header and calculate the frame size. Also, another
690 * consecutive LOAS frame header need to be present after the found frame.
691 * Otherwise the data is not considered as a valid LOAS frame. However, this
692 * "extra check" is omitted when EOS has been received. In this case it is
693 * enough when data[0] contains a valid LOAS header.
695 * This function may set the #needed_data to indicate that a possible frame
696 * candidate has been found, but more data (#needed_data bytes) is needed to
697 * be absolutely sure. When this situation occurs, FALSE will be returned.
699 * When a valid frame is detected, this function will use
700 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
701 * to set the needed bytes for next frame.This way next data chunk is already
704 * LOAS can have three different formats, if I read the spec correctly. Only
705 * one of them is supported here, as the two samples I have use this one.
707 * Returns: TRUE if the given data contains a valid LOAS header.
710 gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
711 const guint8 * data, const guint avail, gboolean drain,
712 guint * framesize, guint * needed_data)
717 if (G_UNLIKELY (avail < 3)) {
722 if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
723 *framesize = gst_aac_parse_loas_get_frame_len (data);
724 GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize);
726 /* In EOS mode this is enough. No need to examine the data further.
727 We also relax the check when we have sync, on the assumption that
728 if we're not looking at random data, we have a much higher chance
729 to get the correct sync, and this avoids losing two frames when
730 a single bit corruption happens. */
731 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
735 if (*framesize + LOAS_MAX_SIZE > avail) {
736 /* We have found a possible frame header candidate, but can't be
737 sure since we don't have enough data to check the next frame */
738 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
739 *framesize + LOAS_MAX_SIZE, avail);
740 *needed_data = *framesize + LOAS_MAX_SIZE;
741 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
742 *framesize + LOAS_MAX_SIZE);
746 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
747 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
749 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
750 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
751 nextlen + LOAS_MAX_SIZE);
758 /* caller ensure sufficient data */
760 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
761 gint * rate, gint * channels, gint * object, gint * version)
765 gint sr_idx = (data[2] & 0x3c) >> 2;
767 *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
770 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
776 *version = (data[1] & 0x08) ? 2 : 4;
778 *object = ((data[2] & 0xc0) >> 6) + 1;
782 * gst_aac_parse_detect_stream:
783 * @aacparse: #GstAacParse.
784 * @data: A block of data that needs to be examined for stream characteristics.
785 * @avail: Size of the given datablock.
786 * @framesize: If valid stream was found, this will be set to tell the
787 * first frame size in bytes.
788 * @skipsize: If valid stream was found, this will be set to tell the first
789 * audio frame position within the given data.
791 * Examines the given piece of data and try to detect the format of it. It
792 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
793 * header. If the stream is detected, TRUE will be returned and #framesize
794 * is set to indicate the found frame size. Additionally, #skipsize might
795 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
796 * position of the frame inside given data chunk.
798 * Returns: TRUE on success.
801 gst_aac_parse_detect_stream (GstAacParse * aacparse,
802 const guint8 * data, const guint avail, gboolean drain,
803 guint * framesize, gint * skipsize)
805 gboolean found = FALSE;
806 guint need_data_adts = 0, need_data_loas;
809 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
811 /* FIXME: No need to check for ADIF if we are not in the beginning of the
814 /* Can we even parse the header? */
815 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
816 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
820 for (i = 0; i < avail - 4; i++) {
821 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
822 ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
823 strncmp ((char *) data + i, "ADIF", 4) == 0) {
824 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
828 /* Trick: tell the parent class that we didn't find the frame yet,
829 but make it skip 'i' amount of bytes. Next time we arrive
830 here we have full frame in the beginning of the data. */
843 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
844 framesize, &need_data_adts)) {
847 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
849 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
850 &aacparse->object_type, &aacparse->mpegversion);
852 if (!channels || !framesize) {
853 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
857 aacparse->header_type = DSPAAC_HEADER_ADTS;
858 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
859 aacparse->frame_samples, 2, 2);
861 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
862 rate, channels, aacparse->object_type, aacparse->mpegversion);
864 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
869 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
870 framesize, &need_data_loas)) {
873 GST_INFO ("LOAS, framesize: %d", *framesize);
875 aacparse->header_type = DSPAAC_HEADER_LOAS;
877 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
878 &channels, &aacparse->mpegversion)) {
879 GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
883 if (rate && channels) {
884 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
885 aacparse->frame_samples, 2, 2);
887 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
888 rate, channels, aacparse->object_type, aacparse->mpegversion);
889 aacparse->sample_rate = rate;
890 aacparse->channels = channels;
893 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
898 if (need_data_adts || need_data_loas) {
899 /* This tells the parent class not to skip any data */
904 if (avail < ADIF_MAX_SIZE)
907 if (memcmp (data + i, "ADIF", 4) == 0) {
914 aacparse->header_type = DSPAAC_HEADER_ADIF;
915 aacparse->mpegversion = 4;
917 /* Skip the "ADIF" bytes */
920 /* copyright string */
922 skip_size += 9; /* skip 9 bytes */
924 bitstream_type = adif[0 + skip_size] & 0x10;
926 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
927 ((unsigned int) adif[1 + skip_size] << 11) |
928 ((unsigned int) adif[2 + skip_size] << 3) |
929 ((unsigned int) adif[3 + skip_size] & 0xe0);
932 if (bitstream_type == 0) {
934 /* Buffer fullness parsing. Currently not needed... */
938 num_elems = (adif[3 + skip_size] & 0x1e);
939 GST_INFO ("ADIF num_config_elems: %d", num_elems);
941 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
942 ((unsigned int) adif[4 + skip_size] << 11) |
943 ((unsigned int) adif[5 + skip_size] << 3) |
944 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
946 GST_INFO ("ADIF buffer fullness: %d", fullness);
948 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
949 ((adif[7 + skip_size] & 0x80) >> 7);
950 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
954 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
955 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
956 ((adif[5 + skip_size] & 0x80) >> 7);
959 /* FIXME: This gives totally wrong results. Duration calculation cannot
961 aacparse->sample_rate =
962 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
964 /* baseparse is not given any fps,
965 * so it will give up on timestamps, seeking, etc */
967 /* FIXME: Can we assume this? */
968 aacparse->channels = 2;
970 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
971 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
973 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
975 /* arrange for metadata and get out of the way */
976 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
977 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
979 gst_caps_unref (sinkcaps);
981 /* not syncable, not easily seekable (unless we push data from start */
982 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
983 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
984 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
990 /* This should never happen */
995 * gst_aac_parse_get_audio_profile_object_type
996 * @aacparse: #GstAacParse.
998 * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
999 * mpegversion and profile of @aacparse's src pad caps, according to the
1000 * values defined by table 1.A.11 in ISO/IEC 14496-3.
1002 * Returns: the profile or object type value corresponding to @aacparse's src
1003 * pad caps, if such a value exists; otherwise G_MAXUINT8.
1006 gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
1009 GstStructure *srcstruct;
1010 const gchar *profile;
1013 srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
1014 srcstruct = gst_caps_get_structure (srccaps, 0);
1015 profile = gst_structure_get_string (srcstruct, "profile");
1016 if (G_UNLIKELY (profile == NULL)) {
1017 gst_caps_unref (srccaps);
1021 if (g_strcmp0 (profile, "main") == 0) {
1023 } else if (g_strcmp0 (profile, "lc") == 0) {
1025 } else if (g_strcmp0 (profile, "ssr") == 0) {
1027 } else if (g_strcmp0 (profile, "ltp") == 0) {
1028 if (G_LIKELY (aacparse->mpegversion == 4))
1031 ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
1036 gst_caps_unref (srccaps);
1041 * gst_aac_parse_get_audio_channel_configuration
1042 * @num_channels: number of audio channels.
1044 * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
1045 * 14496-3, for a given number of audio channels.
1047 * Returns: the Channel Configuration value corresponding to @num_channels, if
1048 * such a value exists; otherwise G_MAXUINT8.
1051 gst_aac_parse_get_audio_channel_configuration (gint num_channels)
1053 if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
1054 return (guint8) num_channels;
1055 else if (num_channels == 8) /* 7.1 */
1060 /* FIXME: Add support for configurations 11, 12 and 14 from
1061 * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
1066 * gst_aac_parse_get_audio_sampling_frequency_index:
1067 * @sample_rate: audio sampling rate.
1069 * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
1070 * 14496-3, for a given sampling rate.
1072 * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
1073 * if such a value exists; otherwise G_MAXUINT8.
1076 gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
1078 switch (sample_rate) {
1111 * gst_aac_parse_prepend_adts_headers:
1112 * @aacparse: #GstAacParse.
1113 * @frame: raw AAC frame to which ADTS headers shall be prepended.
1115 * Prepends ADTS headers to a raw AAC audio frame.
1117 * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
1120 gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
1121 GstBaseParseFrame * frame)
1124 guint8 *adts_headers;
1127 guint8 id, profile, channel_configuration, sampling_frequency_index;
1129 id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
1130 profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
1131 if (profile == G_MAXUINT8) {
1132 GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
1135 channel_configuration =
1136 gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
1137 if (channel_configuration == G_MAXUINT8) {
1138 GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
1141 sampling_frequency_index =
1142 gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
1143 if (sampling_frequency_index == G_MAXUINT8) {
1144 GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
1148 frame->out_buffer = gst_buffer_copy (frame->buffer);
1149 buf_size = gst_buffer_get_size (frame->out_buffer);
1150 frame_size = buf_size + ADTS_HEADERS_LENGTH;
1152 if (G_UNLIKELY (frame_size >= 0x4000)) {
1153 GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
1157 adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
1159 /* Note: no error correction bits are added to the resulting ADTS frames */
1160 adts_headers[0] = 0xFFU;
1161 adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
1162 adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
1163 (channel_configuration & 0x4U);
1164 adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
1165 (guint8) (frame_size >> 11);
1166 adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
1167 adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
1168 adts_headers[6] = 0xFCU;
1170 mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
1171 ADTS_HEADERS_LENGTH, adts_headers, g_free);
1172 gst_buffer_prepend_memory (frame->out_buffer, mem);
1178 * gst_aac_parse_check_valid_frame:
1179 * @parse: #GstBaseParse.
1180 * @frame: #GstBaseParseFrame.
1181 * @skipsize: How much data parent class should skip in order to find the
1184 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
1186 * Also determines frame overhead.
1187 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
1188 * a per-frame header. LOAS has 3 bytes.
1190 * We're making a couple of simplifying assumptions:
1192 * 1. We count Program Configuration Elements rather than searching for them
1193 * in the streams to discount them - the overhead is negligible.
1195 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
1196 * bits, which should still not be significant enough to warrant the
1197 * additional parsing through the headers
1199 * Returns: a #GstFlowReturn.
1201 static GstFlowReturn
1202 gst_aac_parse_handle_frame (GstBaseParse * parse,
1203 GstBaseParseFrame * frame, gint * skipsize)
1206 GstAacParse *aacparse;
1207 gboolean ret = FALSE;
1211 gint rate, channels;
1213 aacparse = GST_AAC_PARSE (parse);
1214 buffer = frame->buffer;
1216 gst_buffer_map (buffer, &map, GST_MAP_READ);
1219 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
1221 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
1222 aacparse->header_type == DSPAAC_HEADER_NONE) {
1223 /* There is nothing to parse */
1224 framesize = map.size;
1227 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
1229 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
1230 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
1232 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1233 guint needed_data = 1024;
1235 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
1236 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1238 if (!ret && needed_data) {
1239 GST_DEBUG ("buffer didn't contain valid frame");
1241 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1245 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1246 guint needed_data = 1024;
1248 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
1249 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1251 if (!ret && needed_data) {
1252 GST_DEBUG ("buffer didn't contain valid frame");
1254 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1259 GST_DEBUG ("buffer didn't contain valid frame");
1260 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1264 if (G_UNLIKELY (!ret))
1267 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1269 frame->overhead = 7;
1271 gst_aac_parse_parse_adts_header (aacparse, map.data,
1272 &rate, &channels, NULL, NULL);
1274 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
1276 if (G_UNLIKELY (rate != aacparse->sample_rate
1277 || channels != aacparse->channels)) {
1278 aacparse->sample_rate = rate;
1279 aacparse->channels = channels;
1281 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1282 /* If linking fails, we need to return appropriate error */
1283 ret = GST_FLOW_NOT_LINKED;
1286 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1287 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1289 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1290 gboolean setcaps = FALSE;
1293 frame->overhead = 3;
1295 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1297 GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
1298 } else if (G_UNLIKELY (rate != aacparse->sample_rate
1299 || channels != aacparse->channels)) {
1300 aacparse->sample_rate = rate;
1301 aacparse->channels = channels;
1303 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1307 /* We want to set caps both at start, and when rate/channels change.
1308 Since only some LOAS frames have that info, we may receive frames
1309 before knowing about rate/channels. */
1311 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1312 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1313 /* If linking fails, we need to return appropriate error */
1314 ret = GST_FLOW_NOT_LINKED;
1317 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1318 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1322 if (aacparse->header_type == DSPAAC_HEADER_NONE
1323 && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
1324 if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
1325 GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
1326 ret = GST_FLOW_ERROR;
1331 gst_buffer_unmap (buffer, &map);
1334 /* found, skip if needed */
1343 if (ret && framesize <= map.size) {
1344 return gst_base_parse_finish_frame (parse, frame, framesize);
1350 static GstFlowReturn
1351 gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
1353 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1355 if (!aacparse->sent_codec_tag) {
1356 GstTagList *taglist;
1359 taglist = gst_tag_list_new_empty ();
1362 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1363 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1364 GST_TAG_AUDIO_CODEC, caps);
1365 gst_caps_unref (caps);
1367 gst_pad_push_event (GST_BASE_PARSE_SRC_PAD (aacparse),
1368 gst_event_new_tag (taglist));
1370 /* also signals the end of first-frame processing */
1371 aacparse->sent_codec_tag = TRUE;
1374 /* As a special case, we can remove the ADTS framing and output raw AAC. */
1375 if (aacparse->header_type == DSPAAC_HEADER_ADTS
1376 && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
1379 gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
1380 header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
1381 gst_buffer_unmap (frame->buffer, &map);
1382 gst_buffer_resize (frame->buffer, header_size,
1383 gst_buffer_get_size (frame->buffer) - header_size);
1391 * gst_aac_parse_start:
1392 * @parse: #GstBaseParse.
1394 * Implementation of "start" vmethod in #GstBaseParse class.
1396 * Returns: TRUE if startup succeeded.
1399 gst_aac_parse_start (GstBaseParse * parse)
1401 GstAacParse *aacparse;
1403 aacparse = GST_AAC_PARSE (parse);
1404 GST_DEBUG ("start");
1405 aacparse->frame_samples = 1024;
1406 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1407 aacparse->sent_codec_tag = FALSE;
1413 * gst_aac_parse_stop:
1414 * @parse: #GstBaseParse.
1416 * Implementation of "stop" vmethod in #GstBaseParse class.
1418 * Returns: TRUE is stopping succeeded.
1421 gst_aac_parse_stop (GstBaseParse * parse)
1428 remove_fields (GstCaps * caps)
1432 n = gst_caps_get_size (caps);
1433 for (i = 0; i < n; i++) {
1434 GstStructure *s = gst_caps_get_structure (caps, i);
1436 gst_structure_remove_field (s, "framed");
1441 add_conversion_fields (GstCaps * caps)
1445 n = gst_caps_get_size (caps);
1446 for (i = 0; i < n; i++) {
1447 GstStructure *s = gst_caps_get_structure (caps, i);
1449 if (gst_structure_has_field (s, "stream-format")) {
1450 const GValue *v = gst_structure_get_value (s, "stream-format");
1452 if (G_VALUE_HOLDS_STRING (v)) {
1453 const gchar *str = g_value_get_string (v);
1455 if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
1456 GValue va = G_VALUE_INIT;
1457 GValue vs = G_VALUE_INIT;
1459 g_value_init (&va, GST_TYPE_LIST);
1460 g_value_init (&vs, G_TYPE_STRING);
1461 g_value_set_string (&vs, "adts");
1462 gst_value_list_append_value (&va, &vs);
1463 g_value_set_string (&vs, "raw");
1464 gst_value_list_append_value (&va, &vs);
1465 gst_structure_set_value (s, "stream-format", &va);
1466 g_value_unset (&va);
1467 g_value_unset (&vs);
1469 } else if (GST_VALUE_HOLDS_LIST (v)) {
1470 gboolean contains_raw = FALSE;
1471 gboolean contains_adts = FALSE;
1472 guint m = gst_value_list_get_size (v), j;
1474 for (j = 0; j < m; j++) {
1475 const GValue *ve = gst_value_list_get_value (v, j);
1478 if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
1479 if (strcmp (str, "adts") == 0)
1480 contains_adts = TRUE;
1481 else if (strcmp (str, "raw") == 0)
1482 contains_raw = TRUE;
1486 if (contains_adts || contains_raw) {
1487 GValue va = G_VALUE_INIT;
1488 GValue vs = G_VALUE_INIT;
1490 g_value_init (&va, GST_TYPE_LIST);
1491 g_value_init (&vs, G_TYPE_STRING);
1492 g_value_copy (v, &va);
1494 if (!contains_raw) {
1495 g_value_set_string (&vs, "raw");
1496 gst_value_list_append_value (&va, &vs);
1498 if (!contains_adts) {
1499 g_value_set_string (&vs, "adts");
1500 gst_value_list_append_value (&va, &vs);
1503 gst_structure_set_value (s, "stream-format", &va);
1505 g_value_unset (&vs);
1506 g_value_unset (&va);
1514 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1516 GstCaps *peercaps, *templ;
1519 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1522 GstCaps *fcopy = gst_caps_copy (filter);
1523 /* Remove the fields we convert */
1524 remove_fields (fcopy);
1525 add_conversion_fields (fcopy);
1526 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1527 gst_caps_unref (fcopy);
1529 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1532 peercaps = gst_caps_make_writable (peercaps);
1533 /* Remove the fields we convert */
1534 remove_fields (peercaps);
1535 add_conversion_fields (peercaps);
1537 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1538 gst_caps_unref (peercaps);
1539 gst_caps_unref (templ);
1545 GstCaps *intersection;
1548 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1549 gst_caps_unref (res);