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[16] = {
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[16] = {
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);
100 static gboolean gst_aac_parse_src_event (GstBaseParse * parse,
103 #define gst_aac_parse_parent_class parent_class
104 G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
107 * gst_aac_parse_class_init:
108 * @klass: #GstAacParseClass.
112 gst_aac_parse_class_init (GstAacParseClass * klass)
114 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
115 GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
117 GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
118 "AAC audio stream parser");
120 gst_element_class_add_static_pad_template (element_class, &sink_template);
121 gst_element_class_add_static_pad_template (element_class, &src_template);
123 gst_element_class_set_static_metadata (element_class,
124 "AAC audio stream parser", "Codec/Parser/Audio",
125 "Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
127 parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
128 parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
129 parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
130 parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
131 parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_handle_frame);
132 parse_class->pre_push_frame =
133 GST_DEBUG_FUNCPTR (gst_aac_parse_pre_push_frame);
134 parse_class->src_event = GST_DEBUG_FUNCPTR (gst_aac_parse_src_event);
139 * gst_aac_parse_init:
140 * @aacparse: #GstAacParse.
141 * @klass: #GstAacParseClass.
145 gst_aac_parse_init (GstAacParse * aacparse)
147 GST_DEBUG ("initialized");
148 GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
149 GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (aacparse));
151 aacparse->last_parsed_sample_rate = 0;
152 aacparse->last_parsed_channels = 0;
157 * gst_aac_parse_set_src_caps:
158 * @aacparse: #GstAacParse.
159 * @sink_caps: (proposed) caps of sink pad
161 * Set source pad caps according to current knowledge about the
164 * Returns: TRUE if caps were successfully set.
167 gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
170 GstCaps *src_caps = NULL, *allowed;
171 gboolean res = FALSE;
172 const gchar *stream_format;
173 guint8 codec_data[2];
174 guint16 codec_data_data;
175 gint sample_rate_idx;
177 GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
179 src_caps = gst_caps_copy (sink_caps);
181 src_caps = gst_caps_new_empty_simple ("audio/mpeg");
183 gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
184 "mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
186 aacparse->output_header_type = aacparse->header_type;
187 switch (aacparse->header_type) {
188 case DSPAAC_HEADER_NONE:
189 stream_format = "raw";
191 case DSPAAC_HEADER_ADTS:
192 stream_format = "adts";
194 case DSPAAC_HEADER_ADIF:
195 stream_format = "adif";
197 case DSPAAC_HEADER_LOAS:
198 stream_format = "loas";
201 stream_format = NULL;
204 /* Generate codec data to be able to set profile/level on the caps */
206 gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
207 if (sample_rate_idx < 0)
208 goto not_a_known_rate;
210 (aacparse->object_type << 11) |
211 (sample_rate_idx << 7) | (aacparse->channels << 3);
212 GST_WRITE_UINT16_BE (codec_data, codec_data_data);
213 gst_codec_utils_aac_caps_set_level_and_profile (src_caps, codec_data, 2);
215 s = gst_caps_get_structure (src_caps, 0);
216 if (aacparse->sample_rate > 0)
217 gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
218 if (aacparse->channels > 0)
219 gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
221 gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
223 allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
224 if (allowed && !gst_caps_can_intersect (src_caps, allowed)) {
225 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
226 "Caps can not intersect");
227 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
228 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
229 "Input is ADTS, trying raw");
230 gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
232 if (gst_caps_can_intersect (src_caps, allowed)) {
233 GstBuffer *codec_data_buffer;
235 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
236 "Caps can intersect, we will drop the ADTS layer");
237 aacparse->output_header_type = DSPAAC_HEADER_NONE;
239 /* The codec_data data is according to AudioSpecificConfig,
240 ISO/IEC 14496-3, 1.6.2.1 */
241 codec_data_buffer = gst_buffer_new_and_alloc (2);
242 gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
243 gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
244 codec_data_buffer, NULL);
246 } else if (aacparse->header_type == DSPAAC_HEADER_NONE) {
247 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
248 "Input is raw, trying ADTS");
249 gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
251 if (gst_caps_can_intersect (src_caps, allowed)) {
252 GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
253 "Caps can intersect, we will prepend ADTS headers");
254 aacparse->output_header_type = DSPAAC_HEADER_ADTS;
259 gst_caps_unref (allowed);
261 aacparse->last_parsed_channels = 0;
262 aacparse->last_parsed_sample_rate = 0;
264 GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
266 res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
267 gst_caps_unref (src_caps);
271 GST_ERROR_OBJECT (aacparse, "Not a known sample rate: %d",
272 aacparse->sample_rate);
273 gst_caps_unref (src_caps);
279 * gst_aac_parse_sink_setcaps:
283 * Implementation of "set_sink_caps" vmethod in #GstBaseParse class.
285 * Returns: TRUE on success.
288 gst_aac_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
290 GstAacParse *aacparse;
291 GstStructure *structure;
295 aacparse = GST_AAC_PARSE (parse);
296 structure = gst_caps_get_structure (caps, 0);
297 caps_str = gst_caps_to_string (caps);
299 GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str);
302 /* This is needed at least in case of RTP
303 * Parses the codec_data information to get ObjectType,
304 * number of channels and samplerate */
305 value = gst_structure_get_value (structure, "codec_data");
307 GstBuffer *buf = gst_value_get_buffer (value);
313 gst_buffer_map (buf, &map, GST_MAP_READ);
315 sr_idx = ((map.data[0] & 0x07) << 1) | ((map.data[1] & 0x80) >> 7);
316 aacparse->object_type = (map.data[0] & 0xf8) >> 3;
317 aacparse->sample_rate =
318 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
319 aacparse->channels = (map.data[1] & 0x78) >> 3;
320 if (aacparse->channels == 7)
321 aacparse->channels = 8;
322 else if (aacparse->channels == 11)
323 aacparse->channels = 7;
324 else if (aacparse->channels == 12 || aacparse->channels == 14)
325 aacparse->channels = 8;
326 aacparse->header_type = DSPAAC_HEADER_NONE;
327 aacparse->mpegversion = 4;
328 aacparse->frame_samples = (map.data[1] & 4) ? 960 : 1024;
329 gst_buffer_unmap (buf, &map);
331 GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
332 "samples=%d", aacparse->object_type, aacparse->sample_rate,
333 aacparse->channels, aacparse->frame_samples);
335 /* arrange for metadata and get out of the way */
336 gst_aac_parse_set_src_caps (aacparse, caps);
337 if (aacparse->header_type == aacparse->output_header_type)
338 gst_base_parse_set_passthrough (parse, TRUE);
343 /* caps info overrides */
344 gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
345 gst_structure_get_int (structure, "channels", &aacparse->channels);
347 const gchar *stream_format =
348 gst_structure_get_string (structure, "stream-format");
350 if (g_strcmp0 (stream_format, "raw") == 0) {
351 GST_ERROR_OBJECT (parse, "Need codec_data for raw AAC");
354 aacparse->sample_rate = 0;
355 aacparse->channels = 0;
356 aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
357 gst_base_parse_set_passthrough (parse, FALSE);
365 * gst_aac_parse_adts_get_frame_len:
366 * @data: block of data containing an ADTS header.
368 * This function calculates ADTS frame length from the given header.
370 * Returns: size of the ADTS frame.
373 gst_aac_parse_adts_get_frame_len (const guint8 * data)
375 return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
380 * gst_aac_parse_check_adts_frame:
381 * @aacparse: #GstAacParse.
382 * @data: Data to be checked.
383 * @avail: Amount of data passed.
384 * @framesize: If valid ADTS frame was found, this will be set to tell the
385 * found frame size in bytes.
386 * @needed_data: If frame was not found, this may be set to tell how much
387 * more data is needed in the next round to detect the frame
388 * reliably. This may happen when a frame header candidate
389 * is found but it cannot be guaranteed to be the header without
390 * peeking the following data.
392 * Check if the given data contains contains ADTS frame. The algorithm
393 * will examine ADTS frame header and calculate the frame size. Also, another
394 * consecutive ADTS frame header need to be present after the found frame.
395 * Otherwise the data is not considered as a valid ADTS frame. However, this
396 * "extra check" is omitted when EOS has been received. In this case it is
397 * enough when data[0] contains a valid ADTS header.
399 * This function may set the #needed_data to indicate that a possible frame
400 * candidate has been found, but more data (#needed_data bytes) is needed to
401 * be absolutely sure. When this situation occurs, FALSE will be returned.
403 * When a valid frame is detected, this function will use
404 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
405 * to set the needed bytes for next frame.This way next data chunk is already
408 * Returns: TRUE if the given data contains a valid ADTS header.
411 gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
412 const guint8 * data, const guint avail, gboolean drain,
413 guint * framesize, guint * needed_data)
419 /* Absolute minimum to perform the ADTS syncword,
420 layer and sampling frequency tests */
421 if (G_UNLIKELY (avail < 3)) {
426 /* Syncword and layer tests */
427 if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
429 /* Sampling frequency test */
430 if (G_UNLIKELY ((data[2] & 0x3C) >> 2 == 15))
433 /* This looks like an ADTS frame header but
434 we need at least 6 bytes to proceed */
435 if (G_UNLIKELY (avail < 6)) {
440 *framesize = gst_aac_parse_adts_get_frame_len (data);
442 /* If frame has CRC, it needs 2 bytes
443 for it at the end of the header */
444 crc_size = (data[1] & 0x01) ? 0 : 2;
447 if (*framesize < 7 + crc_size) {
448 *needed_data = 7 + crc_size;
452 /* In EOS mode this is enough. No need to examine the data further.
453 We also relax the check when we have sync, on the assumption that
454 if we're not looking at random data, we have a much higher chance
455 to get the correct sync, and this avoids losing two frames when
456 a single bit corruption happens. */
457 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
461 if (*framesize + ADTS_MAX_SIZE > avail) {
462 /* We have found a possible frame header candidate, but can't be
463 sure since we don't have enough data to check the next frame */
464 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
465 *framesize + ADTS_MAX_SIZE, avail);
466 *needed_data = *framesize + ADTS_MAX_SIZE;
467 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
468 *framesize + ADTS_MAX_SIZE);
472 if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
473 guint nextlen = gst_aac_parse_adts_get_frame_len (data + (*framesize));
475 GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
476 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
477 nextlen + ADTS_MAX_SIZE);
485 gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
488 guint8 bytes, i, byte;
491 if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
493 for (i = 0; i <= bytes; ++i) {
495 if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
503 gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
504 guint8 * audio_object_type)
506 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
508 if (*audio_object_type == 31) {
509 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
511 *audio_object_type += 32;
513 GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
518 gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
521 guint8 sampling_frequency_index;
522 if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
524 GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
525 sampling_frequency_index);
526 if (sampling_frequency_index == 0xf) {
527 guint32 sampling_rate;
528 if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
530 *sample_rate = sampling_rate;
532 *sample_rate = loas_sample_rate_table[sampling_frequency_index];
536 aacparse->last_parsed_sample_rate = *sample_rate;
540 /* See table 1.13 in ISO/IEC 14496-3 */
542 gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
543 GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
545 guint8 audio_object_type, channel_configuration;
547 if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
550 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
553 if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
555 GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
556 *channels = loas_channels_table[channel_configuration];
560 if (audio_object_type == 5) {
561 GST_LOG_OBJECT (aacparse,
562 "Audio object type 5, so rereading sampling rate...");
563 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
567 GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
568 *sample_rate, *channels);
570 /* There's LOTS of stuff next, but we ignore it for now as we have
571 what we want (sample rate and number of channels */
572 GST_DEBUG_OBJECT (aacparse,
573 "Need more code to parse humongous LOAS data, currently ignored");
576 aacparse->last_parsed_channels = *channels;
582 gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
583 guint avail, gint * sample_rate, gint * channels, gint * version)
588 /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
592 gst_bit_reader_init (&br, data, avail);
594 /* skip sync word (11 bits) and size (13 bits) */
595 if (!gst_bit_reader_skip (&br, 11 + 13))
598 /* First bit is "use last config" */
599 if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
602 GST_LOG_OBJECT (aacparse, "Frame uses previous config");
603 if (!aacparse->last_parsed_sample_rate || !aacparse->last_parsed_channels) {
604 GST_DEBUG_OBJECT (aacparse,
605 "No previous config to use. We'll look for more data.");
608 *sample_rate = aacparse->last_parsed_sample_rate;
609 *channels = aacparse->last_parsed_channels;
613 GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
615 /* audioMuxVersion */
616 if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
619 /* audioMuxVersionA */
620 if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
625 GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
627 guint8 same_time, subframes, num_program, prog;
630 /* taraBufferFullness */
631 if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
634 if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
636 if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
638 if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
640 GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
641 same_time, subframes, num_program);
643 for (prog = 0; prog <= num_program; ++prog) {
644 guint8 num_layer, layer;
645 if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
647 GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
649 for (layer = 0; layer <= num_layer; ++layer) {
650 guint8 use_same_config;
651 if (prog == 0 && layer == 0) {
654 if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
657 if (!use_same_config) {
659 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
660 sample_rate, channels, NULL))
663 guint32 bits, asc_len;
664 if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
666 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
667 sample_rate, channels, &bits))
670 if (!gst_bit_reader_skip (&br, asc_len))
676 GST_LOG_OBJECT (aacparse, "More data ignored");
678 GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
685 * gst_aac_parse_loas_get_frame_len:
686 * @data: block of data containing a LOAS header.
688 * This function calculates LOAS frame length from the given header.
690 * Returns: size of the LOAS frame.
693 gst_aac_parse_loas_get_frame_len (const guint8 * data)
695 return (((data[1] & 0x1f) << 8) | data[2]) + 3;
700 * gst_aac_parse_check_loas_frame:
701 * @aacparse: #GstAacParse.
702 * @data: Data to be checked.
703 * @avail: Amount of data passed.
704 * @framesize: If valid LOAS frame was found, this will be set to tell the
705 * found frame size in bytes.
706 * @needed_data: If frame was not found, this may be set to tell how much
707 * more data is needed in the next round to detect the frame
708 * reliably. This may happen when a frame header candidate
709 * is found but it cannot be guaranteed to be the header without
710 * peeking the following data.
712 * Check if the given data contains contains LOAS frame. The algorithm
713 * will examine LOAS frame header and calculate the frame size. Also, another
714 * consecutive LOAS frame header need to be present after the found frame.
715 * Otherwise the data is not considered as a valid LOAS frame. However, this
716 * "extra check" is omitted when EOS has been received. In this case it is
717 * enough when data[0] contains a valid LOAS header.
719 * This function may set the #needed_data to indicate that a possible frame
720 * candidate has been found, but more data (#needed_data bytes) is needed to
721 * be absolutely sure. When this situation occurs, FALSE will be returned.
723 * When a valid frame is detected, this function will use
724 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
725 * to set the needed bytes for next frame.This way next data chunk is already
728 * LOAS can have three different formats, if I read the spec correctly. Only
729 * one of them is supported here, as the two samples I have use this one.
731 * Returns: TRUE if the given data contains a valid LOAS header.
734 gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
735 const guint8 * data, const guint avail, gboolean drain,
736 guint * framesize, guint * needed_data)
741 if (G_UNLIKELY (avail < 3)) {
746 if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
747 *framesize = gst_aac_parse_loas_get_frame_len (data);
748 GST_DEBUG_OBJECT (aacparse, "Found possible %u byte LOAS frame",
751 /* In EOS mode this is enough. No need to examine the data further.
752 We also relax the check when we have sync, on the assumption that
753 if we're not looking at random data, we have a much higher chance
754 to get the correct sync, and this avoids losing two frames when
755 a single bit corruption happens. */
756 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
760 if (*framesize + LOAS_MAX_SIZE > avail) {
761 /* We have found a possible frame header candidate, but can't be
762 sure since we don't have enough data to check the next frame */
763 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
764 *framesize + LOAS_MAX_SIZE, avail);
765 *needed_data = *framesize + LOAS_MAX_SIZE;
766 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
767 *framesize + LOAS_MAX_SIZE);
771 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
772 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
774 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
775 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
776 nextlen + LOAS_MAX_SIZE);
779 GST_DEBUG_OBJECT (aacparse, "That was a false positive");
785 /* caller ensure sufficient data */
787 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
788 gint * rate, gint * channels, gint * object, gint * version)
792 gint sr_idx = (data[2] & 0x3c) >> 2;
794 *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
797 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
803 *version = (data[1] & 0x08) ? 2 : 4;
805 *object = ((data[2] & 0xc0) >> 6) + 1;
809 * gst_aac_parse_detect_stream:
810 * @aacparse: #GstAacParse.
811 * @data: A block of data that needs to be examined for stream characteristics.
812 * @avail: Size of the given datablock.
813 * @framesize: If valid stream was found, this will be set to tell the
814 * first frame size in bytes.
815 * @skipsize: If valid stream was found, this will be set to tell the first
816 * audio frame position within the given data.
818 * Examines the given piece of data and try to detect the format of it. It
819 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
820 * header. If the stream is detected, TRUE will be returned and #framesize
821 * is set to indicate the found frame size. Additionally, #skipsize might
822 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
823 * position of the frame inside given data chunk.
825 * Returns: TRUE on success.
828 gst_aac_parse_detect_stream (GstAacParse * aacparse,
829 const guint8 * data, const guint avail, gboolean drain,
830 guint * framesize, gint * skipsize)
832 gboolean found = FALSE;
833 guint need_data_adts = 0, need_data_loas;
836 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
838 /* FIXME: No need to check for ADIF if we are not in the beginning of the
841 /* Can we even parse the header? */
842 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
843 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
847 for (i = 0; i < avail - 4; i++) {
848 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
849 ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
850 strncmp ((char *) data + i, "ADIF", 4) == 0) {
851 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
855 /* Trick: tell the parent class that we didn't find the frame yet,
856 but make it skip 'i' amount of bytes. Next time we arrive
857 here we have full frame in the beginning of the data. */
870 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
871 framesize, &need_data_adts)) {
874 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
876 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
877 &aacparse->object_type, &aacparse->mpegversion);
879 if (!channels || !framesize) {
880 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
884 aacparse->header_type = DSPAAC_HEADER_ADTS;
885 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
886 aacparse->frame_samples, 2, 2);
888 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
889 rate, channels, aacparse->object_type, aacparse->mpegversion);
891 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
896 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
897 framesize, &need_data_loas)) {
898 gint rate = 0, channels = 0;
900 GST_INFO ("LOAS, framesize: %d", *framesize);
902 aacparse->header_type = DSPAAC_HEADER_LOAS;
904 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
905 &channels, &aacparse->mpegversion)) {
906 /* This is pretty normal when skipping data at the start of
907 * random stream (MPEG-TS capture for example) */
908 GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
912 if (rate && channels) {
913 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
914 aacparse->frame_samples, 2, 2);
916 /* Don't store the sample rate and channels yet -
917 * this is just format detection. */
918 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
919 rate, channels, aacparse->object_type, aacparse->mpegversion);
922 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
927 if (need_data_adts || need_data_loas) {
928 /* This tells the parent class not to skip any data */
933 if (avail < ADIF_MAX_SIZE)
936 if (memcmp (data + i, "ADIF", 4) == 0) {
943 aacparse->header_type = DSPAAC_HEADER_ADIF;
944 aacparse->mpegversion = 4;
946 /* Skip the "ADIF" bytes */
949 /* copyright string */
951 skip_size += 9; /* skip 9 bytes */
953 bitstream_type = adif[0 + skip_size] & 0x10;
955 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
956 ((unsigned int) adif[1 + skip_size] << 11) |
957 ((unsigned int) adif[2 + skip_size] << 3) |
958 ((unsigned int) adif[3 + skip_size] & 0xe0);
961 if (bitstream_type == 0) {
963 /* Buffer fullness parsing. Currently not needed... */
967 num_elems = (adif[3 + skip_size] & 0x1e);
968 GST_INFO ("ADIF num_config_elems: %d", num_elems);
970 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
971 ((unsigned int) adif[4 + skip_size] << 11) |
972 ((unsigned int) adif[5 + skip_size] << 3) |
973 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
975 GST_INFO ("ADIF buffer fullness: %d", fullness);
977 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
978 ((adif[7 + skip_size] & 0x80) >> 7);
979 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
983 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
984 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
985 ((adif[5 + skip_size] & 0x80) >> 7);
988 /* FIXME: This gives totally wrong results. Duration calculation cannot
990 aacparse->sample_rate =
991 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
993 /* baseparse is not given any fps,
994 * so it will give up on timestamps, seeking, etc */
996 /* FIXME: Can we assume this? */
997 aacparse->channels = 2;
999 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
1000 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
1002 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
1004 /* arrange for metadata and get out of the way */
1005 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
1006 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
1008 gst_caps_unref (sinkcaps);
1010 /* not syncable, not easily seekable (unless we push data from start */
1011 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
1012 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
1013 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
1019 /* This should never happen */
1024 * gst_aac_parse_get_audio_profile_object_type
1025 * @aacparse: #GstAacParse.
1027 * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
1028 * mpegversion and profile of @aacparse's src pad caps, according to the
1029 * values defined by table 1.A.11 in ISO/IEC 14496-3.
1031 * Returns: the profile or object type value corresponding to @aacparse's src
1032 * pad caps, if such a value exists; otherwise G_MAXUINT8.
1035 gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
1038 GstStructure *srcstruct;
1039 const gchar *profile;
1042 srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
1043 if (G_UNLIKELY (srccaps == NULL)) {
1047 srcstruct = gst_caps_get_structure (srccaps, 0);
1048 profile = gst_structure_get_string (srcstruct, "profile");
1049 if (G_UNLIKELY (profile == NULL)) {
1050 gst_caps_unref (srccaps);
1054 if (g_strcmp0 (profile, "main") == 0) {
1056 } else if (g_strcmp0 (profile, "lc") == 0) {
1058 } else if (g_strcmp0 (profile, "ssr") == 0) {
1060 } else if (g_strcmp0 (profile, "ltp") == 0) {
1061 if (G_LIKELY (aacparse->mpegversion == 4))
1064 ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
1069 gst_caps_unref (srccaps);
1074 * gst_aac_parse_get_audio_channel_configuration
1075 * @num_channels: number of audio channels.
1077 * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
1078 * 14496-3, for a given number of audio channels.
1080 * Returns: the Channel Configuration value corresponding to @num_channels, if
1081 * such a value exists; otherwise G_MAXUINT8.
1084 gst_aac_parse_get_audio_channel_configuration (gint num_channels)
1086 if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
1087 return (guint8) num_channels;
1088 else if (num_channels == 8) /* 7.1 */
1093 /* FIXME: Add support for configurations 11, 12 and 14 from
1094 * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
1099 * gst_aac_parse_get_audio_sampling_frequency_index:
1100 * @sample_rate: audio sampling rate.
1102 * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
1103 * 14496-3, for a given sampling rate.
1105 * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
1106 * if such a value exists; otherwise G_MAXUINT8.
1109 gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
1111 switch (sample_rate) {
1144 * gst_aac_parse_prepend_adts_headers:
1145 * @aacparse: #GstAacParse.
1146 * @frame: raw AAC frame to which ADTS headers shall be prepended.
1148 * Prepends ADTS headers to a raw AAC audio frame.
1150 * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
1153 gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
1154 GstBaseParseFrame * frame)
1157 guint8 *adts_headers;
1160 guint8 id, profile, channel_configuration, sampling_frequency_index;
1162 id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
1163 profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
1164 if (profile == G_MAXUINT8) {
1165 GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
1168 channel_configuration =
1169 gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
1170 if (channel_configuration == G_MAXUINT8) {
1171 GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
1174 sampling_frequency_index =
1175 gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
1176 if (sampling_frequency_index == G_MAXUINT8) {
1177 GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
1181 frame->out_buffer = gst_buffer_copy (frame->buffer);
1182 buf_size = gst_buffer_get_size (frame->out_buffer);
1183 frame_size = buf_size + ADTS_HEADERS_LENGTH;
1185 if (G_UNLIKELY (frame_size >= 0x4000)) {
1186 GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
1190 adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
1192 /* Note: no error correction bits are added to the resulting ADTS frames */
1193 adts_headers[0] = 0xFFU;
1194 adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
1195 adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
1196 (channel_configuration & 0x4U);
1197 adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
1198 (guint8) (frame_size >> 11);
1199 adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
1200 adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
1201 adts_headers[6] = 0xFCU;
1203 mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
1204 ADTS_HEADERS_LENGTH, adts_headers, g_free);
1205 gst_buffer_prepend_memory (frame->out_buffer, mem);
1211 * gst_aac_parse_check_valid_frame:
1212 * @parse: #GstBaseParse.
1213 * @frame: #GstBaseParseFrame.
1214 * @skipsize: How much data parent class should skip in order to find the
1217 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
1219 * Also determines frame overhead.
1220 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
1221 * a per-frame header. LOAS has 3 bytes.
1223 * We're making a couple of simplifying assumptions:
1225 * 1. We count Program Configuration Elements rather than searching for them
1226 * in the streams to discount them - the overhead is negligible.
1228 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
1229 * bits, which should still not be significant enough to warrant the
1230 * additional parsing through the headers
1232 * Returns: a #GstFlowReturn.
1234 static GstFlowReturn
1235 gst_aac_parse_handle_frame (GstBaseParse * parse,
1236 GstBaseParseFrame * frame, gint * skipsize)
1239 GstAacParse *aacparse;
1240 gboolean ret = FALSE;
1244 gint rate = 0, channels = 0;
1246 aacparse = GST_AAC_PARSE (parse);
1247 buffer = frame->buffer;
1249 gst_buffer_map (buffer, &map, GST_MAP_READ);
1252 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
1254 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
1255 aacparse->header_type == DSPAAC_HEADER_NONE) {
1256 /* There is nothing to parse */
1257 framesize = map.size;
1260 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
1262 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
1263 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
1265 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1266 guint needed_data = 1024;
1268 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
1269 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1271 if (!ret && needed_data) {
1272 GST_DEBUG ("buffer didn't contain valid frame");
1274 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1278 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1279 guint needed_data = 1024;
1281 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
1282 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1284 if (!ret && needed_data) {
1285 GST_DEBUG ("buffer didn't contain valid frame");
1287 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1292 GST_DEBUG ("buffer didn't contain valid frame");
1293 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1297 if (G_UNLIKELY (!ret))
1300 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1302 frame->overhead = 7;
1304 gst_aac_parse_parse_adts_header (aacparse, map.data,
1305 &rate, &channels, NULL, NULL);
1307 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
1309 if (G_UNLIKELY (rate != aacparse->sample_rate
1310 || channels != aacparse->channels)) {
1311 aacparse->sample_rate = rate;
1312 aacparse->channels = channels;
1314 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1315 /* If linking fails, we need to return appropriate error */
1316 ret = GST_FLOW_NOT_LINKED;
1319 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1320 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1322 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1323 gboolean setcaps = FALSE;
1326 frame->overhead = 3;
1328 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1329 &channels, NULL) || !rate || !channels) {
1330 /* This is pretty normal when skipping data at the start of
1331 * random stream (MPEG-TS capture for example) */
1332 GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
1333 /* Since we don't fully parse the LOAS config, we don't know for sure
1334 * how much to skip. Just skip 1 to end up to the next marker and
1335 * resume parsing from there */
1340 if (G_UNLIKELY (rate != aacparse->sample_rate
1341 || channels != aacparse->channels)) {
1342 aacparse->sample_rate = rate;
1343 aacparse->channels = channels;
1345 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1349 /* We want to set caps both at start, and when rate/channels change.
1350 Since only some LOAS frames have that info, we may receive frames
1351 before knowing about rate/channels. */
1353 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1354 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1355 /* If linking fails, we need to return appropriate error */
1356 ret = GST_FLOW_NOT_LINKED;
1359 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1360 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1364 if (aacparse->header_type == DSPAAC_HEADER_NONE
1365 && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
1366 if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
1367 GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
1368 ret = GST_FLOW_ERROR;
1373 gst_buffer_unmap (buffer, &map);
1376 /* found, skip if needed */
1385 if (ret && framesize <= map.size) {
1386 return gst_base_parse_finish_frame (parse, frame, framesize);
1392 static GstFlowReturn
1393 gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
1395 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1397 if (!aacparse->sent_codec_tag) {
1398 GstTagList *taglist;
1402 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1404 if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
1405 GST_INFO_OBJECT (parse, "Src pad is flushing");
1406 return GST_FLOW_FLUSHING;
1408 GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
1409 return GST_FLOW_NOT_NEGOTIATED;
1413 taglist = gst_tag_list_new_empty ();
1414 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1415 GST_TAG_AUDIO_CODEC, caps);
1416 gst_caps_unref (caps);
1418 gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
1419 gst_tag_list_unref (taglist);
1421 /* also signals the end of first-frame processing */
1422 aacparse->sent_codec_tag = TRUE;
1425 /* As a special case, we can remove the ADTS framing and output raw AAC. */
1426 if (aacparse->header_type == DSPAAC_HEADER_ADTS
1427 && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
1430 gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
1431 header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
1432 gst_buffer_unmap (frame->buffer, &map);
1433 gst_buffer_resize (frame->buffer, header_size,
1434 gst_buffer_get_size (frame->buffer) - header_size);
1442 * gst_aac_parse_start:
1443 * @parse: #GstBaseParse.
1445 * Implementation of "start" vmethod in #GstBaseParse class.
1447 * Returns: TRUE if startup succeeded.
1450 gst_aac_parse_start (GstBaseParse * parse)
1452 GstAacParse *aacparse;
1454 aacparse = GST_AAC_PARSE (parse);
1455 GST_DEBUG ("start");
1456 aacparse->frame_samples = 1024;
1457 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1458 aacparse->sent_codec_tag = FALSE;
1459 aacparse->last_parsed_channels = 0;
1460 aacparse->last_parsed_sample_rate = 0;
1466 * gst_aac_parse_stop:
1467 * @parse: #GstBaseParse.
1469 * Implementation of "stop" vmethod in #GstBaseParse class.
1471 * Returns: TRUE is stopping succeeded.
1474 gst_aac_parse_stop (GstBaseParse * parse)
1481 remove_fields (GstCaps * caps)
1485 n = gst_caps_get_size (caps);
1486 for (i = 0; i < n; i++) {
1487 GstStructure *s = gst_caps_get_structure (caps, i);
1489 gst_structure_remove_field (s, "framed");
1494 add_conversion_fields (GstCaps * caps)
1498 n = gst_caps_get_size (caps);
1499 for (i = 0; i < n; i++) {
1500 GstStructure *s = gst_caps_get_structure (caps, i);
1502 if (gst_structure_has_field (s, "stream-format")) {
1503 const GValue *v = gst_structure_get_value (s, "stream-format");
1505 if (G_VALUE_HOLDS_STRING (v)) {
1506 const gchar *str = g_value_get_string (v);
1508 if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
1509 GValue va = G_VALUE_INIT;
1510 GValue vs = G_VALUE_INIT;
1512 g_value_init (&va, GST_TYPE_LIST);
1513 g_value_init (&vs, G_TYPE_STRING);
1514 g_value_set_string (&vs, "adts");
1515 gst_value_list_append_value (&va, &vs);
1516 g_value_set_string (&vs, "raw");
1517 gst_value_list_append_value (&va, &vs);
1518 gst_structure_set_value (s, "stream-format", &va);
1519 g_value_unset (&va);
1520 g_value_unset (&vs);
1522 } else if (GST_VALUE_HOLDS_LIST (v)) {
1523 gboolean contains_raw = FALSE;
1524 gboolean contains_adts = FALSE;
1525 guint m = gst_value_list_get_size (v), j;
1527 for (j = 0; j < m; j++) {
1528 const GValue *ve = gst_value_list_get_value (v, j);
1531 if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
1532 if (strcmp (str, "adts") == 0)
1533 contains_adts = TRUE;
1534 else if (strcmp (str, "raw") == 0)
1535 contains_raw = TRUE;
1539 if (contains_adts || contains_raw) {
1540 GValue va = G_VALUE_INIT;
1541 GValue vs = G_VALUE_INIT;
1543 g_value_init (&va, GST_TYPE_LIST);
1544 g_value_init (&vs, G_TYPE_STRING);
1545 g_value_copy (v, &va);
1547 if (!contains_raw) {
1548 g_value_set_string (&vs, "raw");
1549 gst_value_list_append_value (&va, &vs);
1551 if (!contains_adts) {
1552 g_value_set_string (&vs, "adts");
1553 gst_value_list_append_value (&va, &vs);
1556 gst_structure_set_value (s, "stream-format", &va);
1558 g_value_unset (&vs);
1559 g_value_unset (&va);
1567 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1569 GstCaps *peercaps, *templ;
1572 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1575 GstCaps *fcopy = gst_caps_copy (filter);
1576 /* Remove the fields we convert */
1577 remove_fields (fcopy);
1578 add_conversion_fields (fcopy);
1579 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1580 gst_caps_unref (fcopy);
1582 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1585 peercaps = gst_caps_make_writable (peercaps);
1586 /* Remove the fields we convert */
1587 remove_fields (peercaps);
1588 add_conversion_fields (peercaps);
1590 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1591 gst_caps_unref (peercaps);
1592 gst_caps_unref (templ);
1598 GstCaps *intersection;
1601 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1602 gst_caps_unref (res);
1610 gst_aac_parse_src_event (GstBaseParse * parse, GstEvent * event)
1612 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1614 if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
1615 aacparse->last_parsed_channels = 0;
1616 aacparse->last_parsed_sample_rate = 0;
1619 return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);