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);
309 if (buf && gst_buffer_get_size (buf) >= 2) {
313 if (!gst_buffer_map (buf, &map, GST_MAP_READ))
316 sr_idx = ((map.data[0] & 0x07) << 1) | ((map.data[1] & 0x80) >> 7);
317 aacparse->object_type = (map.data[0] & 0xf8) >> 3;
318 aacparse->sample_rate =
319 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
320 aacparse->channels = (map.data[1] & 0x78) >> 3;
321 if (aacparse->channels == 7)
322 aacparse->channels = 8;
323 else if (aacparse->channels == 11)
324 aacparse->channels = 7;
325 else if (aacparse->channels == 12 || aacparse->channels == 14)
326 aacparse->channels = 8;
327 aacparse->header_type = DSPAAC_HEADER_NONE;
328 aacparse->mpegversion = 4;
329 aacparse->frame_samples = (map.data[1] & 4) ? 960 : 1024;
330 gst_buffer_unmap (buf, &map);
332 GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
333 "samples=%d", aacparse->object_type, aacparse->sample_rate,
334 aacparse->channels, aacparse->frame_samples);
336 /* arrange for metadata and get out of the way */
337 gst_aac_parse_set_src_caps (aacparse, caps);
338 if (aacparse->header_type == aacparse->output_header_type)
339 gst_base_parse_set_passthrough (parse, TRUE);
344 /* caps info overrides */
345 gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
346 gst_structure_get_int (structure, "channels", &aacparse->channels);
348 const gchar *stream_format =
349 gst_structure_get_string (structure, "stream-format");
351 if (g_strcmp0 (stream_format, "raw") == 0) {
352 GST_ERROR_OBJECT (parse, "Need codec_data for raw AAC");
355 aacparse->sample_rate = 0;
356 aacparse->channels = 0;
357 aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
358 gst_base_parse_set_passthrough (parse, FALSE);
366 * gst_aac_parse_adts_get_frame_len:
367 * @data: block of data containing an ADTS header.
369 * This function calculates ADTS frame length from the given header.
371 * Returns: size of the ADTS frame.
374 gst_aac_parse_adts_get_frame_len (const guint8 * data)
376 return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
381 * gst_aac_parse_check_adts_frame:
382 * @aacparse: #GstAacParse.
383 * @data: Data to be checked.
384 * @avail: Amount of data passed.
385 * @framesize: If valid ADTS frame was found, this will be set to tell the
386 * found frame size in bytes.
387 * @needed_data: If frame was not found, this may be set to tell how much
388 * more data is needed in the next round to detect the frame
389 * reliably. This may happen when a frame header candidate
390 * is found but it cannot be guaranteed to be the header without
391 * peeking the following data.
393 * Check if the given data contains contains ADTS frame. The algorithm
394 * will examine ADTS frame header and calculate the frame size. Also, another
395 * consecutive ADTS frame header need to be present after the found frame.
396 * Otherwise the data is not considered as a valid ADTS frame. However, this
397 * "extra check" is omitted when EOS has been received. In this case it is
398 * enough when data[0] contains a valid ADTS header.
400 * This function may set the #needed_data to indicate that a possible frame
401 * candidate has been found, but more data (#needed_data bytes) is needed to
402 * be absolutely sure. When this situation occurs, FALSE will be returned.
404 * When a valid frame is detected, this function will use
405 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
406 * to set the needed bytes for next frame.This way next data chunk is already
409 * Returns: TRUE if the given data contains a valid ADTS header.
412 gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
413 const guint8 * data, const guint avail, gboolean drain,
414 guint * framesize, guint * needed_data)
420 /* Absolute minimum to perform the ADTS syncword,
421 layer and sampling frequency tests */
422 if (G_UNLIKELY (avail < 3)) {
427 /* Syncword and layer tests */
428 if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
430 /* Sampling frequency test */
431 if (G_UNLIKELY ((data[2] & 0x3C) >> 2 == 15))
434 /* This looks like an ADTS frame header but
435 we need at least 6 bytes to proceed */
436 if (G_UNLIKELY (avail < 6)) {
441 *framesize = gst_aac_parse_adts_get_frame_len (data);
443 /* If frame has CRC, it needs 2 bytes
444 for it at the end of the header */
445 crc_size = (data[1] & 0x01) ? 0 : 2;
448 if (*framesize < 7 + crc_size) {
449 *needed_data = 7 + crc_size;
453 /* In EOS mode this is enough. No need to examine the data further.
454 We also relax the check when we have sync, on the assumption that
455 if we're not looking at random data, we have a much higher chance
456 to get the correct sync, and this avoids losing two frames when
457 a single bit corruption happens. */
458 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
462 if (*framesize + ADTS_MAX_SIZE > avail) {
463 /* We have found a possible frame header candidate, but can't be
464 sure since we don't have enough data to check the next frame */
465 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
466 *framesize + ADTS_MAX_SIZE, avail);
467 *needed_data = *framesize + ADTS_MAX_SIZE;
468 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
469 *framesize + ADTS_MAX_SIZE);
473 if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
474 guint nextlen = gst_aac_parse_adts_get_frame_len (data + (*framesize));
476 GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
477 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
478 nextlen + ADTS_MAX_SIZE);
486 gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
489 guint8 bytes, i, byte;
492 if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
494 for (i = 0; i <= bytes; ++i) {
496 if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
504 gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
505 guint8 * audio_object_type)
507 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
509 if (*audio_object_type == 31) {
510 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
512 *audio_object_type += 32;
514 GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
519 gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
522 guint8 sampling_frequency_index;
523 if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
525 GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
526 sampling_frequency_index);
527 if (sampling_frequency_index == 0xf) {
528 guint32 sampling_rate;
529 if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
531 *sample_rate = sampling_rate;
533 *sample_rate = loas_sample_rate_table[sampling_frequency_index];
537 aacparse->last_parsed_sample_rate = *sample_rate;
541 /* See table 1.13 in ISO/IEC 14496-3 */
543 gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
544 GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
546 guint8 audio_object_type;
547 guint8 G_GNUC_UNUSED extension_audio_object_type;
548 guint8 channel_configuration, extension_channel_configuration;
549 gboolean G_GNUC_UNUSED sbr = FALSE, ps = FALSE;
551 if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
554 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
557 if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
559 *channels = loas_channels_table[channel_configuration];
560 GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
564 if (audio_object_type == 5 || audio_object_type == 29) {
565 extension_audio_object_type = 5;
567 if (audio_object_type == 29)
570 GST_LOG_OBJECT (aacparse,
571 "Audio object type 5 or 29, so rereading sampling rate...");
572 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
575 if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
578 if (audio_object_type == 22) {
579 /* extension channel configuration */
580 if (!gst_bit_reader_get_bits_uint8 (br, &extension_channel_configuration,
583 GST_LOG_OBJECT (aacparse, "extension channel_configuration: %d",
584 extension_channel_configuration);
585 *channels = loas_channels_table[extension_channel_configuration];
590 extension_audio_object_type = 0;
593 GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
594 *sample_rate, *channels);
596 /* There's LOTS of stuff next, but we ignore it for now as we have
597 what we want (sample rate and number of channels */
598 GST_DEBUG_OBJECT (aacparse,
599 "Need more code to parse humongous LOAS data, currently ignored");
602 aacparse->last_parsed_channels = *channels;
608 gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
609 guint avail, gint * sample_rate, gint * channels, gint * version)
614 /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
618 gst_bit_reader_init (&br, data, avail);
620 /* skip sync word (11 bits) and size (13 bits) */
621 if (!gst_bit_reader_skip (&br, 11 + 13))
624 /* First bit is "use last config" */
625 if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
628 GST_LOG_OBJECT (aacparse, "Frame uses previous config");
629 if (!aacparse->last_parsed_sample_rate || !aacparse->last_parsed_channels) {
630 GST_DEBUG_OBJECT (aacparse,
631 "No previous config to use. We'll look for more data.");
634 *sample_rate = aacparse->last_parsed_sample_rate;
635 *channels = aacparse->last_parsed_channels;
639 GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
641 /* audioMuxVersion */
642 if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
645 /* audioMuxVersionA */
646 if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
651 GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
653 guint8 same_time, subframes, num_program, prog;
656 /* taraBufferFullness */
657 if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
660 if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
662 if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
664 if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
666 GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
667 same_time, subframes, num_program);
669 for (prog = 0; prog <= num_program; ++prog) {
670 guint8 num_layer, layer;
671 if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
673 GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
675 for (layer = 0; layer <= num_layer; ++layer) {
676 guint8 use_same_config;
677 if (prog == 0 && layer == 0) {
680 if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
683 if (!use_same_config) {
685 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
686 sample_rate, channels, NULL))
689 guint32 bits, asc_len;
690 if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
692 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
693 sample_rate, channels, &bits))
696 if (!gst_bit_reader_skip (&br, asc_len))
702 GST_LOG_OBJECT (aacparse, "More data ignored");
704 GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
711 * gst_aac_parse_loas_get_frame_len:
712 * @data: block of data containing a LOAS header.
714 * This function calculates LOAS frame length from the given header.
716 * Returns: size of the LOAS frame.
719 gst_aac_parse_loas_get_frame_len (const guint8 * data)
721 return (((data[1] & 0x1f) << 8) | data[2]) + 3;
726 * gst_aac_parse_check_loas_frame:
727 * @aacparse: #GstAacParse.
728 * @data: Data to be checked.
729 * @avail: Amount of data passed.
730 * @framesize: If valid LOAS frame was found, this will be set to tell the
731 * found frame size in bytes.
732 * @needed_data: If frame was not found, this may be set to tell how much
733 * more data is needed in the next round to detect the frame
734 * reliably. This may happen when a frame header candidate
735 * is found but it cannot be guaranteed to be the header without
736 * peeking the following data.
738 * Check if the given data contains contains LOAS frame. The algorithm
739 * will examine LOAS frame header and calculate the frame size. Also, another
740 * consecutive LOAS frame header need to be present after the found frame.
741 * Otherwise the data is not considered as a valid LOAS frame. However, this
742 * "extra check" is omitted when EOS has been received. In this case it is
743 * enough when data[0] contains a valid LOAS header.
745 * This function may set the #needed_data to indicate that a possible frame
746 * candidate has been found, but more data (#needed_data bytes) is needed to
747 * be absolutely sure. When this situation occurs, FALSE will be returned.
749 * When a valid frame is detected, this function will use
750 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
751 * to set the needed bytes for next frame.This way next data chunk is already
754 * LOAS can have three different formats, if I read the spec correctly. Only
755 * one of them is supported here, as the two samples I have use this one.
757 * Returns: TRUE if the given data contains a valid LOAS header.
760 gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
761 const guint8 * data, const guint avail, gboolean drain,
762 guint * framesize, guint * needed_data)
767 if (G_UNLIKELY (avail < 3)) {
772 if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
773 *framesize = gst_aac_parse_loas_get_frame_len (data);
774 GST_DEBUG_OBJECT (aacparse, "Found possible %u byte LOAS frame",
777 /* In EOS mode this is enough. No need to examine the data further.
778 We also relax the check when we have sync, on the assumption that
779 if we're not looking at random data, we have a much higher chance
780 to get the correct sync, and this avoids losing two frames when
781 a single bit corruption happens. */
782 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
786 if (*framesize + LOAS_MAX_SIZE > avail) {
787 /* We have found a possible frame header candidate, but can't be
788 sure since we don't have enough data to check the next frame */
789 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
790 *framesize + LOAS_MAX_SIZE, avail);
791 *needed_data = *framesize + LOAS_MAX_SIZE;
792 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
793 *framesize + LOAS_MAX_SIZE);
797 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
798 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
800 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
801 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
802 nextlen + LOAS_MAX_SIZE);
805 GST_DEBUG_OBJECT (aacparse, "That was a false positive");
811 /* caller ensure sufficient data */
813 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
814 gint * rate, gint * channels, gint * object, gint * version)
818 gint sr_idx = (data[2] & 0x3c) >> 2;
820 *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
823 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
829 *version = (data[1] & 0x08) ? 2 : 4;
831 *object = ((data[2] & 0xc0) >> 6) + 1;
835 * gst_aac_parse_detect_stream:
836 * @aacparse: #GstAacParse.
837 * @data: A block of data that needs to be examined for stream characteristics.
838 * @avail: Size of the given datablock.
839 * @framesize: If valid stream was found, this will be set to tell the
840 * first frame size in bytes.
841 * @skipsize: If valid stream was found, this will be set to tell the first
842 * audio frame position within the given data.
844 * Examines the given piece of data and try to detect the format of it. It
845 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
846 * header. If the stream is detected, TRUE will be returned and #framesize
847 * is set to indicate the found frame size. Additionally, #skipsize might
848 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
849 * position of the frame inside given data chunk.
851 * Returns: TRUE on success.
854 gst_aac_parse_detect_stream (GstAacParse * aacparse,
855 const guint8 * data, const guint avail, gboolean drain,
856 guint * framesize, gint * skipsize)
858 gboolean found = FALSE;
859 guint need_data_adts = 0, need_data_loas;
862 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
864 /* FIXME: No need to check for ADIF if we are not in the beginning of the
867 /* Can we even parse the header? */
868 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
869 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
873 for (i = 0; i < avail - 4; i++) {
874 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
875 ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
876 strncmp ((char *) data + i, "ADIF", 4) == 0) {
877 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
881 /* Trick: tell the parent class that we didn't find the frame yet,
882 but make it skip 'i' amount of bytes. Next time we arrive
883 here we have full frame in the beginning of the data. */
896 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
897 framesize, &need_data_adts)) {
900 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
902 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
903 &aacparse->object_type, &aacparse->mpegversion);
905 if (!channels || !framesize) {
906 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
910 aacparse->header_type = DSPAAC_HEADER_ADTS;
911 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
912 aacparse->frame_samples, 2, 2);
914 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
915 rate, channels, aacparse->object_type, aacparse->mpegversion);
917 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
922 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
923 framesize, &need_data_loas)) {
924 gint rate = 0, channels = 0;
926 GST_INFO ("LOAS, framesize: %d", *framesize);
928 aacparse->header_type = DSPAAC_HEADER_LOAS;
930 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
931 &channels, &aacparse->mpegversion)) {
932 /* This is pretty normal when skipping data at the start of
933 * random stream (MPEG-TS capture for example) */
934 GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
938 if (rate && channels) {
939 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
940 aacparse->frame_samples, 2, 2);
942 /* Don't store the sample rate and channels yet -
943 * this is just format detection. */
944 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
945 rate, channels, aacparse->object_type, aacparse->mpegversion);
948 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
953 if (need_data_adts || need_data_loas) {
954 /* This tells the parent class not to skip any data */
959 if (avail < ADIF_MAX_SIZE)
962 if (memcmp (data + i, "ADIF", 4) == 0) {
969 aacparse->header_type = DSPAAC_HEADER_ADIF;
970 aacparse->mpegversion = 4;
972 /* Skip the "ADIF" bytes */
975 /* copyright string */
977 skip_size += 9; /* skip 9 bytes */
979 bitstream_type = adif[0 + skip_size] & 0x10;
981 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
982 ((unsigned int) adif[1 + skip_size] << 11) |
983 ((unsigned int) adif[2 + skip_size] << 3) |
984 ((unsigned int) adif[3 + skip_size] & 0xe0);
987 if (bitstream_type == 0) {
989 /* Buffer fullness parsing. Currently not needed... */
993 num_elems = (adif[3 + skip_size] & 0x1e);
994 GST_INFO ("ADIF num_config_elems: %d", num_elems);
996 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
997 ((unsigned int) adif[4 + skip_size] << 11) |
998 ((unsigned int) adif[5 + skip_size] << 3) |
999 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
1001 GST_INFO ("ADIF buffer fullness: %d", fullness);
1003 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
1004 ((adif[7 + skip_size] & 0x80) >> 7);
1005 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
1009 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
1010 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
1011 ((adif[5 + skip_size] & 0x80) >> 7);
1014 /* FIXME: This gives totally wrong results. Duration calculation cannot
1016 aacparse->sample_rate =
1017 gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
1019 /* baseparse is not given any fps,
1020 * so it will give up on timestamps, seeking, etc */
1022 /* FIXME: Can we assume this? */
1023 aacparse->channels = 2;
1025 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
1026 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
1028 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
1030 /* arrange for metadata and get out of the way */
1031 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
1032 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
1034 gst_caps_unref (sinkcaps);
1036 /* not syncable, not easily seekable (unless we push data from start */
1037 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
1038 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
1039 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
1045 /* This should never happen */
1050 * gst_aac_parse_get_audio_profile_object_type
1051 * @aacparse: #GstAacParse.
1053 * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
1054 * mpegversion and profile of @aacparse's src pad caps, according to the
1055 * values defined by table 1.A.11 in ISO/IEC 14496-3.
1057 * Returns: the profile or object type value corresponding to @aacparse's src
1058 * pad caps, if such a value exists; otherwise G_MAXUINT8.
1061 gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
1064 GstStructure *srcstruct;
1065 const gchar *profile;
1068 srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
1069 if (G_UNLIKELY (srccaps == NULL)) {
1073 srcstruct = gst_caps_get_structure (srccaps, 0);
1074 profile = gst_structure_get_string (srcstruct, "profile");
1075 if (G_UNLIKELY (profile == NULL)) {
1076 gst_caps_unref (srccaps);
1080 if (g_strcmp0 (profile, "main") == 0) {
1082 } else if (g_strcmp0 (profile, "lc") == 0) {
1084 } else if (g_strcmp0 (profile, "ssr") == 0) {
1086 } else if (g_strcmp0 (profile, "ltp") == 0) {
1087 if (G_LIKELY (aacparse->mpegversion == 4))
1090 ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
1095 gst_caps_unref (srccaps);
1100 * gst_aac_parse_get_audio_channel_configuration
1101 * @num_channels: number of audio channels.
1103 * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
1104 * 14496-3, for a given number of audio channels.
1106 * Returns: the Channel Configuration value corresponding to @num_channels, if
1107 * such a value exists; otherwise G_MAXUINT8.
1110 gst_aac_parse_get_audio_channel_configuration (gint num_channels)
1112 if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
1113 return (guint8) num_channels;
1114 else if (num_channels == 8) /* 7.1 */
1119 /* FIXME: Add support for configurations 11, 12 and 14 from
1120 * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
1125 * gst_aac_parse_get_audio_sampling_frequency_index:
1126 * @sample_rate: audio sampling rate.
1128 * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
1129 * 14496-3, for a given sampling rate.
1131 * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
1132 * if such a value exists; otherwise G_MAXUINT8.
1135 gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
1137 switch (sample_rate) {
1170 * gst_aac_parse_prepend_adts_headers:
1171 * @aacparse: #GstAacParse.
1172 * @frame: raw AAC frame to which ADTS headers shall be prepended.
1174 * Prepends ADTS headers to a raw AAC audio frame.
1176 * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
1179 gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
1180 GstBaseParseFrame * frame)
1183 guint8 *adts_headers;
1186 guint8 id, profile, channel_configuration, sampling_frequency_index;
1188 id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
1189 profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
1190 if (profile == G_MAXUINT8) {
1191 GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
1194 channel_configuration =
1195 gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
1196 if (channel_configuration == G_MAXUINT8) {
1197 GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
1200 sampling_frequency_index =
1201 gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
1202 if (sampling_frequency_index == G_MAXUINT8) {
1203 GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
1207 frame->out_buffer = gst_buffer_copy (frame->buffer);
1208 buf_size = gst_buffer_get_size (frame->out_buffer);
1209 frame_size = buf_size + ADTS_HEADERS_LENGTH;
1211 if (G_UNLIKELY (frame_size >= 0x4000)) {
1212 GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
1216 adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
1218 /* Note: no error correction bits are added to the resulting ADTS frames */
1219 adts_headers[0] = 0xFFU;
1220 adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
1221 adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
1222 (channel_configuration & 0x4U);
1223 adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
1224 (guint8) (frame_size >> 11);
1225 adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
1226 adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
1227 adts_headers[6] = 0xFCU;
1229 mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
1230 ADTS_HEADERS_LENGTH, adts_headers, g_free);
1231 gst_buffer_prepend_memory (frame->out_buffer, mem);
1237 * gst_aac_parse_check_valid_frame:
1238 * @parse: #GstBaseParse.
1239 * @frame: #GstBaseParseFrame.
1240 * @skipsize: How much data parent class should skip in order to find the
1243 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
1245 * Also determines frame overhead.
1246 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
1247 * a per-frame header. LOAS has 3 bytes.
1249 * We're making a couple of simplifying assumptions:
1251 * 1. We count Program Configuration Elements rather than searching for them
1252 * in the streams to discount them - the overhead is negligible.
1254 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
1255 * bits, which should still not be significant enough to warrant the
1256 * additional parsing through the headers
1258 * Returns: a #GstFlowReturn.
1260 static GstFlowReturn
1261 gst_aac_parse_handle_frame (GstBaseParse * parse,
1262 GstBaseParseFrame * frame, gint * skipsize)
1265 GstAacParse *aacparse;
1266 gboolean ret = FALSE;
1270 gint rate = 0, channels = 0;
1272 aacparse = GST_AAC_PARSE (parse);
1273 buffer = frame->buffer;
1275 gst_buffer_map (buffer, &map, GST_MAP_READ);
1278 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
1280 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
1281 aacparse->header_type == DSPAAC_HEADER_NONE) {
1282 /* There is nothing to parse */
1283 framesize = map.size;
1286 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
1288 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
1289 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
1291 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1292 guint needed_data = 1024;
1294 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
1295 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1297 if (!ret && needed_data) {
1298 GST_DEBUG ("buffer didn't contain valid frame");
1300 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1304 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1305 guint needed_data = 1024;
1307 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
1308 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
1310 if (!ret && needed_data) {
1311 GST_DEBUG ("buffer didn't contain valid frame");
1313 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1318 GST_DEBUG ("buffer didn't contain valid frame");
1319 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
1323 if (G_UNLIKELY (!ret))
1326 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
1328 frame->overhead = 7;
1330 gst_aac_parse_parse_adts_header (aacparse, map.data,
1331 &rate, &channels, NULL, NULL);
1333 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
1335 if (G_UNLIKELY (rate != aacparse->sample_rate
1336 || channels != aacparse->channels)) {
1337 aacparse->sample_rate = rate;
1338 aacparse->channels = channels;
1340 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1341 /* If linking fails, we need to return appropriate error */
1342 ret = GST_FLOW_NOT_LINKED;
1345 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1346 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1348 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1349 gboolean setcaps = FALSE;
1352 frame->overhead = 3;
1354 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1355 &channels, NULL) || !rate || !channels) {
1356 /* This is pretty normal when skipping data at the start of
1357 * random stream (MPEG-TS capture for example) */
1358 GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
1359 /* Since we don't fully parse the LOAS config, we don't know for sure
1360 * how much to skip. Just skip 1 to end up to the next marker and
1361 * resume parsing from there */
1366 if (G_UNLIKELY (rate != aacparse->sample_rate
1367 || channels != aacparse->channels)) {
1368 aacparse->sample_rate = rate;
1369 aacparse->channels = channels;
1371 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1375 /* We want to set caps both at start, and when rate/channels change.
1376 Since only some LOAS frames have that info, we may receive frames
1377 before knowing about rate/channels. */
1379 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1380 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1381 /* If linking fails, we need to return appropriate error */
1382 ret = GST_FLOW_NOT_LINKED;
1385 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1386 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1390 if (aacparse->header_type == DSPAAC_HEADER_NONE
1391 && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
1392 if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
1393 GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
1394 ret = GST_FLOW_ERROR;
1399 gst_buffer_unmap (buffer, &map);
1402 /* found, skip if needed */
1411 if (ret && framesize <= map.size) {
1412 return gst_base_parse_finish_frame (parse, frame, framesize);
1418 static GstFlowReturn
1419 gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
1421 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1423 if (!aacparse->sent_codec_tag) {
1424 GstTagList *taglist;
1428 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1430 if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
1431 GST_INFO_OBJECT (parse, "Src pad is flushing");
1432 return GST_FLOW_FLUSHING;
1434 GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
1435 return GST_FLOW_NOT_NEGOTIATED;
1439 taglist = gst_tag_list_new_empty ();
1440 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1441 GST_TAG_AUDIO_CODEC, caps);
1442 gst_caps_unref (caps);
1444 gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
1445 gst_tag_list_unref (taglist);
1447 /* also signals the end of first-frame processing */
1448 aacparse->sent_codec_tag = TRUE;
1451 /* As a special case, we can remove the ADTS framing and output raw AAC. */
1452 if (aacparse->header_type == DSPAAC_HEADER_ADTS
1453 && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
1456 frame->out_buffer = gst_buffer_make_writable (frame->buffer);
1457 frame->buffer = NULL;
1458 gst_buffer_map (frame->out_buffer, &map, GST_MAP_READ);
1459 header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
1460 gst_buffer_unmap (frame->out_buffer, &map);
1461 gst_buffer_resize (frame->out_buffer, header_size,
1462 gst_buffer_get_size (frame->out_buffer) - header_size);
1470 * gst_aac_parse_start:
1471 * @parse: #GstBaseParse.
1473 * Implementation of "start" vmethod in #GstBaseParse class.
1475 * Returns: TRUE if startup succeeded.
1478 gst_aac_parse_start (GstBaseParse * parse)
1480 GstAacParse *aacparse;
1482 aacparse = GST_AAC_PARSE (parse);
1483 GST_DEBUG ("start");
1484 aacparse->frame_samples = 1024;
1485 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1486 aacparse->sent_codec_tag = FALSE;
1487 aacparse->last_parsed_channels = 0;
1488 aacparse->last_parsed_sample_rate = 0;
1494 * gst_aac_parse_stop:
1495 * @parse: #GstBaseParse.
1497 * Implementation of "stop" vmethod in #GstBaseParse class.
1499 * Returns: TRUE is stopping succeeded.
1502 gst_aac_parse_stop (GstBaseParse * parse)
1509 remove_fields (GstCaps * caps)
1513 n = gst_caps_get_size (caps);
1514 for (i = 0; i < n; i++) {
1515 GstStructure *s = gst_caps_get_structure (caps, i);
1517 gst_structure_remove_field (s, "framed");
1522 add_conversion_fields (GstCaps * caps)
1526 n = gst_caps_get_size (caps);
1527 for (i = 0; i < n; i++) {
1528 GstStructure *s = gst_caps_get_structure (caps, i);
1530 if (gst_structure_has_field (s, "stream-format")) {
1531 const GValue *v = gst_structure_get_value (s, "stream-format");
1533 if (G_VALUE_HOLDS_STRING (v)) {
1534 const gchar *str = g_value_get_string (v);
1536 if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
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_set_string (&vs, "adts");
1543 gst_value_list_append_value (&va, &vs);
1544 g_value_set_string (&vs, "raw");
1545 gst_value_list_append_value (&va, &vs);
1546 gst_structure_set_value (s, "stream-format", &va);
1547 g_value_unset (&va);
1548 g_value_unset (&vs);
1550 } else if (GST_VALUE_HOLDS_LIST (v)) {
1551 gboolean contains_raw = FALSE;
1552 gboolean contains_adts = FALSE;
1553 guint m = gst_value_list_get_size (v), j;
1555 for (j = 0; j < m; j++) {
1556 const GValue *ve = gst_value_list_get_value (v, j);
1559 if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
1560 if (strcmp (str, "adts") == 0)
1561 contains_adts = TRUE;
1562 else if (strcmp (str, "raw") == 0)
1563 contains_raw = TRUE;
1567 if (contains_adts || contains_raw) {
1568 GValue va = G_VALUE_INIT;
1569 GValue vs = G_VALUE_INIT;
1571 g_value_init (&va, GST_TYPE_LIST);
1572 g_value_init (&vs, G_TYPE_STRING);
1573 g_value_copy (v, &va);
1575 if (!contains_raw) {
1576 g_value_set_string (&vs, "raw");
1577 gst_value_list_append_value (&va, &vs);
1579 if (!contains_adts) {
1580 g_value_set_string (&vs, "adts");
1581 gst_value_list_append_value (&va, &vs);
1584 gst_structure_set_value (s, "stream-format", &va);
1586 g_value_unset (&vs);
1587 g_value_unset (&va);
1595 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1597 GstCaps *peercaps, *templ;
1600 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1603 GstCaps *fcopy = gst_caps_copy (filter);
1604 /* Remove the fields we convert */
1605 remove_fields (fcopy);
1606 add_conversion_fields (fcopy);
1607 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1608 gst_caps_unref (fcopy);
1610 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1613 peercaps = gst_caps_make_writable (peercaps);
1614 /* Remove the fields we convert */
1615 remove_fields (peercaps);
1616 add_conversion_fields (peercaps);
1618 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1619 gst_caps_unref (peercaps);
1620 gst_caps_unref (templ);
1626 GstCaps *intersection;
1629 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1630 gst_caps_unref (res);
1638 gst_aac_parse_src_event (GstBaseParse * parse, GstEvent * event)
1640 GstAacParse *aacparse = GST_AAC_PARSE (parse);
1642 if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
1643 aacparse->last_parsed_channels = 0;
1644 aacparse->last_parsed_sample_rate = 0;
1647 return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);