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 %u byte LOAS frame", *framesize);
750 /* In EOS mode this is enough. No need to examine the data further.
751 We also relax the check when we have sync, on the assumption that
752 if we're not looking at random data, we have a much higher chance
753 to get the correct sync, and this avoids losing two frames when
754 a single bit corruption happens. */
755 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
759 if (*framesize + LOAS_MAX_SIZE > avail) {
760 /* We have found a possible frame header candidate, but can't be
761 sure since we don't have enough data to check the next frame */
762 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
763 *framesize + LOAS_MAX_SIZE, avail);
764 *needed_data = *framesize + LOAS_MAX_SIZE;
765 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
766 *framesize + LOAS_MAX_SIZE);
770 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
771 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
773 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
774 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
775 nextlen + LOAS_MAX_SIZE);
782 /* caller ensure sufficient data */
784 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
785 gint * rate, gint * channels, gint * object, gint * version)
789 gint sr_idx = (data[2] & 0x3c) >> 2;
791 *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
794 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
800 *version = (data[1] & 0x08) ? 2 : 4;
802 *object = ((data[2] & 0xc0) >> 6) + 1;
806 * gst_aac_parse_detect_stream:
807 * @aacparse: #GstAacParse.
808 * @data: A block of data that needs to be examined for stream characteristics.
809 * @avail: Size of the given datablock.
810 * @framesize: If valid stream was found, this will be set to tell the
811 * first frame size in bytes.
812 * @skipsize: If valid stream was found, this will be set to tell the first
813 * audio frame position within the given data.
815 * Examines the given piece of data and try to detect the format of it. It
816 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
817 * header. If the stream is detected, TRUE will be returned and #framesize
818 * is set to indicate the found frame size. Additionally, #skipsize might
819 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
820 * position of the frame inside given data chunk.
822 * Returns: TRUE on success.
825 gst_aac_parse_detect_stream (GstAacParse * aacparse,
826 const guint8 * data, const guint avail, gboolean drain,
827 guint * framesize, gint * skipsize)
829 gboolean found = FALSE;
830 guint need_data_adts = 0, need_data_loas;
833 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
835 /* FIXME: No need to check for ADIF if we are not in the beginning of the
838 /* Can we even parse the header? */
839 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
840 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
844 for (i = 0; i < avail - 4; i++) {
845 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
846 ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
847 strncmp ((char *) data + i, "ADIF", 4) == 0) {
848 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
852 /* Trick: tell the parent class that we didn't find the frame yet,
853 but make it skip 'i' amount of bytes. Next time we arrive
854 here we have full frame in the beginning of the data. */
867 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
868 framesize, &need_data_adts)) {
871 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
873 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
874 &aacparse->object_type, &aacparse->mpegversion);
876 if (!channels || !framesize) {
877 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
881 aacparse->header_type = DSPAAC_HEADER_ADTS;
882 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
883 aacparse->frame_samples, 2, 2);
885 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
886 rate, channels, aacparse->object_type, aacparse->mpegversion);
888 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
893 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
894 framesize, &need_data_loas)) {
895 gint rate = 0, channels = 0;
897 GST_INFO ("LOAS, framesize: %d", *framesize);
899 aacparse->header_type = DSPAAC_HEADER_LOAS;
901 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
902 &channels, &aacparse->mpegversion)) {
903 /* This is pretty normal when skipping data at the start of
904 * random stream (MPEG-TS capture for example) */
905 GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
909 if (rate && channels) {
910 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
911 aacparse->frame_samples, 2, 2);
913 /* Don't store the sample rate and channels yet -
914 * this is just format detection. */
915 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
916 rate, channels, aacparse->object_type, aacparse->mpegversion);
919 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
924 if (need_data_adts || need_data_loas) {
925 /* This tells the parent class not to skip any data */
930 if (avail < ADIF_MAX_SIZE)
933 if (memcmp (data + i, "ADIF", 4) == 0) {
940 aacparse->header_type = DSPAAC_HEADER_ADIF;
941 aacparse->mpegversion = 4;
943 /* Skip the "ADIF" bytes */
946 /* copyright string */
948 skip_size += 9; /* skip 9 bytes */
950 bitstream_type = adif[0 + skip_size] & 0x10;
952 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
953 ((unsigned int) adif[1 + skip_size] << 11) |
954 ((unsigned int) adif[2 + skip_size] << 3) |
955 ((unsigned int) adif[3 + skip_size] & 0xe0);
958 if (bitstream_type == 0) {
960 /* Buffer fullness parsing. Currently not needed... */
964 num_elems = (adif[3 + skip_size] & 0x1e);
965 GST_INFO ("ADIF num_config_elems: %d", num_elems);
967 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
968 ((unsigned int) adif[4 + skip_size] << 11) |
969 ((unsigned int) adif[5 + skip_size] << 3) |
970 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
972 GST_INFO ("ADIF buffer fullness: %d", fullness);
974 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
975 ((adif[7 + skip_size] & 0x80) >> 7);
976 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
980 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
981 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
982 ((adif[5 + skip_size] & 0x80) >> 7);
985 /* FIXME: This gives totally wrong results. Duration calculation cannot
987 aacparse->sample_rate =
988 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
990 /* baseparse is not given any fps,
991 * so it will give up on timestamps, seeking, etc */
993 /* FIXME: Can we assume this? */
994 aacparse->channels = 2;
996 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
997 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
999 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
1001 /* arrange for metadata and get out of the way */
1002 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
1003 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
1005 gst_caps_unref (sinkcaps);
1007 /* not syncable, not easily seekable (unless we push data from start */
1008 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
1009 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
1010 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
1016 /* This should never happen */
1021 * gst_aac_parse_get_audio_profile_object_type
1022 * @aacparse: #GstAacParse.
1024 * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
1025 * mpegversion and profile of @aacparse's src pad caps, according to the
1026 * values defined by table 1.A.11 in ISO/IEC 14496-3.
1028 * Returns: the profile or object type value corresponding to @aacparse's src
1029 * pad caps, if such a value exists; otherwise G_MAXUINT8.
1032 gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
1035 GstStructure *srcstruct;
1036 const gchar *profile;
1039 srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
1040 if (G_UNLIKELY (srccaps == NULL)) {
1044 srcstruct = gst_caps_get_structure (srccaps, 0);
1045 profile = gst_structure_get_string (srcstruct, "profile");
1046 if (G_UNLIKELY (profile == NULL)) {
1047 gst_caps_unref (srccaps);
1051 if (g_strcmp0 (profile, "main") == 0) {
1053 } else if (g_strcmp0 (profile, "lc") == 0) {
1055 } else if (g_strcmp0 (profile, "ssr") == 0) {
1057 } else if (g_strcmp0 (profile, "ltp") == 0) {
1058 if (G_LIKELY (aacparse->mpegversion == 4))
1061 ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
1066 gst_caps_unref (srccaps);
1071 * gst_aac_parse_get_audio_channel_configuration
1072 * @num_channels: number of audio channels.
1074 * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
1075 * 14496-3, for a given number of audio channels.
1077 * Returns: the Channel Configuration value corresponding to @num_channels, if
1078 * such a value exists; otherwise G_MAXUINT8.
1081 gst_aac_parse_get_audio_channel_configuration (gint num_channels)
1083 if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
1084 return (guint8) num_channels;
1085 else if (num_channels == 8) /* 7.1 */
1090 /* FIXME: Add support for configurations 11, 12 and 14 from
1091 * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
1096 * gst_aac_parse_get_audio_sampling_frequency_index:
1097 * @sample_rate: audio sampling rate.
1099 * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
1100 * 14496-3, for a given sampling rate.
1102 * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
1103 * if such a value exists; otherwise G_MAXUINT8.
1106 gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
1108 switch (sample_rate) {
1141 * gst_aac_parse_prepend_adts_headers:
1142 * @aacparse: #GstAacParse.
1143 * @frame: raw AAC frame to which ADTS headers shall be prepended.
1145 * Prepends ADTS headers to a raw AAC audio frame.
1147 * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
1150 gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
1151 GstBaseParseFrame * frame)
1154 guint8 *adts_headers;
1157 guint8 id, profile, channel_configuration, sampling_frequency_index;
1159 id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
1160 profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
1161 if (profile == G_MAXUINT8) {
1162 GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
1165 channel_configuration =
1166 gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
1167 if (channel_configuration == G_MAXUINT8) {
1168 GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
1171 sampling_frequency_index =
1172 gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
1173 if (sampling_frequency_index == G_MAXUINT8) {
1174 GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
1178 frame->out_buffer = gst_buffer_copy (frame->buffer);
1179 buf_size = gst_buffer_get_size (frame->out_buffer);
1180 frame_size = buf_size + ADTS_HEADERS_LENGTH;
1182 if (G_UNLIKELY (frame_size >= 0x4000)) {
1183 GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
1187 adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
1189 /* Note: no error correction bits are added to the resulting ADTS frames */
1190 adts_headers[0] = 0xFFU;
1191 adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
1192 adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
1193 (channel_configuration & 0x4U);
1194 adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
1195 (guint8) (frame_size >> 11);
1196 adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
1197 adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
1198 adts_headers[6] = 0xFCU;
1200 mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
1201 ADTS_HEADERS_LENGTH, adts_headers, g_free);
1202 gst_buffer_prepend_memory (frame->out_buffer, mem);
1208 * gst_aac_parse_check_valid_frame:
1209 * @parse: #GstBaseParse.
1210 * @frame: #GstBaseParseFrame.
1211 * @skipsize: How much data parent class should skip in order to find the
1214 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
1216 * Also determines frame overhead.
1217 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
1218 * a per-frame header. LOAS has 3 bytes.
1220 * We're making a couple of simplifying assumptions:
1222 * 1. We count Program Configuration Elements rather than searching for them
1223 * in the streams to discount them - the overhead is negligible.
1225 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
1226 * bits, which should still not be significant enough to warrant the
1227 * additional parsing through the headers
1229 * Returns: a #GstFlowReturn.
1231 static GstFlowReturn
1232 gst_aac_parse_handle_frame (GstBaseParse * parse,
1233 GstBaseParseFrame * frame, gint * skipsize)
1236 GstAacParse *aacparse;
1237 gboolean ret = FALSE;
1241 gint rate = 0, channels = 0;
1243 aacparse = GST_AAC_PARSE (parse);
1244 buffer = frame->buffer;
1246 gst_buffer_map (buffer, &map, GST_MAP_READ);
1249 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
1251 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
1252 aacparse->header_type == DSPAAC_HEADER_NONE) {
1253 /* There is nothing to parse */
1254 framesize = map.size;
1257 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
1259 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
1260 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
1262 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1263 guint needed_data = 1024;
1265 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
1266 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1268 if (!ret && needed_data) {
1269 GST_DEBUG ("buffer didn't contain valid frame");
1271 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1275 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1276 guint needed_data = 1024;
1278 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
1279 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1281 if (!ret && needed_data) {
1282 GST_DEBUG ("buffer didn't contain valid frame");
1284 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1289 GST_DEBUG ("buffer didn't contain valid frame");
1290 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1294 if (G_UNLIKELY (!ret))
1297 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1299 frame->overhead = 7;
1301 gst_aac_parse_parse_adts_header (aacparse, map.data,
1302 &rate, &channels, NULL, NULL);
1304 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
1306 if (G_UNLIKELY (rate != aacparse->sample_rate
1307 || channels != aacparse->channels)) {
1308 aacparse->sample_rate = rate;
1309 aacparse->channels = channels;
1311 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1312 /* If linking fails, we need to return appropriate error */
1313 ret = GST_FLOW_NOT_LINKED;
1316 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1317 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1319 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1320 gboolean setcaps = FALSE;
1323 frame->overhead = 3;
1325 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1326 &channels, NULL) || !rate || !channels) {
1327 /* This is pretty normal when skipping data at the start of
1328 * random stream (MPEG-TS capture for example) */
1329 GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
1330 /* Since we don't fully parse the LOAS config, we don't know for sure
1331 * how much to skip. Just skip 1 to end up to the next marker and
1332 * resume parsing from there */
1337 if (G_UNLIKELY (rate != aacparse->sample_rate
1338 || channels != aacparse->channels)) {
1339 aacparse->sample_rate = rate;
1340 aacparse->channels = channels;
1342 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1346 /* We want to set caps both at start, and when rate/channels change.
1347 Since only some LOAS frames have that info, we may receive frames
1348 before knowing about rate/channels. */
1350 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1351 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1352 /* If linking fails, we need to return appropriate error */
1353 ret = GST_FLOW_NOT_LINKED;
1356 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1357 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1361 if (aacparse->header_type == DSPAAC_HEADER_NONE
1362 && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
1363 if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
1364 GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
1365 ret = GST_FLOW_ERROR;
1370 gst_buffer_unmap (buffer, &map);
1373 /* found, skip if needed */
1382 if (ret && framesize <= map.size) {
1383 return gst_base_parse_finish_frame (parse, frame, framesize);
1389 static GstFlowReturn
1390 gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
1392 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1394 if (!aacparse->sent_codec_tag) {
1395 GstTagList *taglist;
1399 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1401 if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
1402 GST_INFO_OBJECT (parse, "Src pad is flushing");
1403 return GST_FLOW_FLUSHING;
1405 GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
1406 return GST_FLOW_NOT_NEGOTIATED;
1410 taglist = gst_tag_list_new_empty ();
1411 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1412 GST_TAG_AUDIO_CODEC, caps);
1413 gst_caps_unref (caps);
1415 gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
1416 gst_tag_list_unref (taglist);
1418 /* also signals the end of first-frame processing */
1419 aacparse->sent_codec_tag = TRUE;
1422 /* As a special case, we can remove the ADTS framing and output raw AAC. */
1423 if (aacparse->header_type == DSPAAC_HEADER_ADTS
1424 && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
1427 gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
1428 header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
1429 gst_buffer_unmap (frame->buffer, &map);
1430 gst_buffer_resize (frame->buffer, header_size,
1431 gst_buffer_get_size (frame->buffer) - header_size);
1439 * gst_aac_parse_start:
1440 * @parse: #GstBaseParse.
1442 * Implementation of "start" vmethod in #GstBaseParse class.
1444 * Returns: TRUE if startup succeeded.
1447 gst_aac_parse_start (GstBaseParse * parse)
1449 GstAacParse *aacparse;
1451 aacparse = GST_AAC_PARSE (parse);
1452 GST_DEBUG ("start");
1453 aacparse->frame_samples = 1024;
1454 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1455 aacparse->sent_codec_tag = FALSE;
1456 aacparse->last_parsed_channels = 0;
1457 aacparse->last_parsed_sample_rate = 0;
1463 * gst_aac_parse_stop:
1464 * @parse: #GstBaseParse.
1466 * Implementation of "stop" vmethod in #GstBaseParse class.
1468 * Returns: TRUE is stopping succeeded.
1471 gst_aac_parse_stop (GstBaseParse * parse)
1478 remove_fields (GstCaps * caps)
1482 n = gst_caps_get_size (caps);
1483 for (i = 0; i < n; i++) {
1484 GstStructure *s = gst_caps_get_structure (caps, i);
1486 gst_structure_remove_field (s, "framed");
1491 add_conversion_fields (GstCaps * caps)
1495 n = gst_caps_get_size (caps);
1496 for (i = 0; i < n; i++) {
1497 GstStructure *s = gst_caps_get_structure (caps, i);
1499 if (gst_structure_has_field (s, "stream-format")) {
1500 const GValue *v = gst_structure_get_value (s, "stream-format");
1502 if (G_VALUE_HOLDS_STRING (v)) {
1503 const gchar *str = g_value_get_string (v);
1505 if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
1506 GValue va = G_VALUE_INIT;
1507 GValue vs = G_VALUE_INIT;
1509 g_value_init (&va, GST_TYPE_LIST);
1510 g_value_init (&vs, G_TYPE_STRING);
1511 g_value_set_string (&vs, "adts");
1512 gst_value_list_append_value (&va, &vs);
1513 g_value_set_string (&vs, "raw");
1514 gst_value_list_append_value (&va, &vs);
1515 gst_structure_set_value (s, "stream-format", &va);
1516 g_value_unset (&va);
1517 g_value_unset (&vs);
1519 } else if (GST_VALUE_HOLDS_LIST (v)) {
1520 gboolean contains_raw = FALSE;
1521 gboolean contains_adts = FALSE;
1522 guint m = gst_value_list_get_size (v), j;
1524 for (j = 0; j < m; j++) {
1525 const GValue *ve = gst_value_list_get_value (v, j);
1528 if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
1529 if (strcmp (str, "adts") == 0)
1530 contains_adts = TRUE;
1531 else if (strcmp (str, "raw") == 0)
1532 contains_raw = TRUE;
1536 if (contains_adts || contains_raw) {
1537 GValue va = G_VALUE_INIT;
1538 GValue vs = G_VALUE_INIT;
1540 g_value_init (&va, GST_TYPE_LIST);
1541 g_value_init (&vs, G_TYPE_STRING);
1542 g_value_copy (v, &va);
1544 if (!contains_raw) {
1545 g_value_set_string (&vs, "raw");
1546 gst_value_list_append_value (&va, &vs);
1548 if (!contains_adts) {
1549 g_value_set_string (&vs, "adts");
1550 gst_value_list_append_value (&va, &vs);
1553 gst_structure_set_value (s, "stream-format", &va);
1555 g_value_unset (&vs);
1556 g_value_unset (&va);
1564 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1566 GstCaps *peercaps, *templ;
1569 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1572 GstCaps *fcopy = gst_caps_copy (filter);
1573 /* Remove the fields we convert */
1574 remove_fields (fcopy);
1575 add_conversion_fields (fcopy);
1576 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1577 gst_caps_unref (fcopy);
1579 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1582 peercaps = gst_caps_make_writable (peercaps);
1583 /* Remove the fields we convert */
1584 remove_fields (peercaps);
1585 add_conversion_fields (peercaps);
1587 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1588 gst_caps_unref (peercaps);
1589 gst_caps_unref (templ);
1595 GstCaps *intersection;
1598 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1599 gst_caps_unref (res);
1607 gst_aac_parse_src_event (GstBaseParse * parse, GstEvent * event)
1609 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1611 if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
1612 aacparse->last_parsed_channels = 0;
1613 aacparse->last_parsed_sample_rate = 0;
1616 return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);