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., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, 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 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
47 #include <gst/base/gstbitreader.h>
48 #include "gstaacparse.h"
51 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
54 GST_STATIC_CAPS ("audio/mpeg, "
55 "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, "
56 "stream-format = (string) { raw, adts, adif, loas };"));
58 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
61 GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) { 2, 4 };"));
63 GST_DEBUG_CATEGORY_STATIC (aacparse_debug);
64 #define GST_CAT_DEFAULT aacparse_debug
67 #define ADIF_MAX_SIZE 40 /* Should be enough */
68 #define ADTS_MAX_SIZE 10 /* Should be enough */
69 #define LOAS_MAX_SIZE 3 /* Should be enough */
72 #define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
74 static const gint loas_sample_rate_table[32] = {
75 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
76 16000, 12000, 11025, 8000, 7350, 0, 0, 0
79 static const gint loas_channels_table[32] = {
80 0, 1, 2, 3, 4, 5, 6, 8,
81 0, 0, 0, 0, 0, 0, 0, 0
84 static gboolean gst_aac_parse_start (GstBaseParse * parse);
85 static gboolean gst_aac_parse_stop (GstBaseParse * parse);
87 static gboolean gst_aac_parse_sink_setcaps (GstBaseParse * parse,
89 static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
92 static GstFlowReturn gst_aac_parse_handle_frame (GstBaseParse * parse,
93 GstBaseParseFrame * frame, gint * skipsize);
95 G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
98 gst_aac_parse_get_sample_rate_from_index (guint sr_idx)
100 static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
101 32000, 24000, 22050, 16000, 12000, 11025, 8000
104 if (sr_idx < G_N_ELEMENTS (aac_sample_rates))
105 return aac_sample_rates[sr_idx];
106 GST_WARNING ("Invalid sample rate index %u", sr_idx);
111 * gst_aac_parse_class_init:
112 * @klass: #GstAacParseClass.
116 gst_aac_parse_class_init (GstAacParseClass * klass)
118 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
119 GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
121 GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
122 "AAC audio stream parser");
124 gst_element_class_add_pad_template (element_class,
125 gst_static_pad_template_get (&sink_template));
126 gst_element_class_add_pad_template (element_class,
127 gst_static_pad_template_get (&src_template));
129 gst_element_class_set_static_metadata (element_class,
130 "AAC audio stream parser", "Codec/Parser/Audio",
131 "Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
133 parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
134 parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
135 parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
136 parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
137 parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_handle_frame);
142 * gst_aac_parse_init:
143 * @aacparse: #GstAacParse.
144 * @klass: #GstAacParseClass.
148 gst_aac_parse_init (GstAacParse * aacparse)
150 GST_DEBUG ("initialized");
155 * gst_aac_parse_set_src_caps:
156 * @aacparse: #GstAacParse.
157 * @sink_caps: (proposed) caps of sink pad
159 * Set source pad caps according to current knowledge about the
162 * Returns: TRUE if caps were successfully set.
165 gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
168 GstCaps *src_caps = NULL;
169 gboolean res = FALSE;
170 const gchar *stream_format;
172 GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
174 src_caps = gst_caps_copy (sink_caps);
176 src_caps = gst_caps_new_empty_simple ("audio/mpeg");
178 gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
179 "mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
181 switch (aacparse->header_type) {
182 case DSPAAC_HEADER_NONE:
183 stream_format = "raw";
185 case DSPAAC_HEADER_ADTS:
186 stream_format = "adts";
188 case DSPAAC_HEADER_ADIF:
189 stream_format = "adif";
191 case DSPAAC_HEADER_LOAS:
192 stream_format = "loas";
195 stream_format = NULL;
198 s = gst_caps_get_structure (src_caps, 0);
199 if (aacparse->sample_rate > 0)
200 gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
201 if (aacparse->channels > 0)
202 gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
204 gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
206 GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
208 res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
209 gst_caps_unref (src_caps);
215 * gst_aac_parse_sink_setcaps:
219 * Implementation of "set_sink_caps" vmethod in #GstBaseParse class.
221 * Returns: TRUE on success.
224 gst_aac_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
226 GstAacParse *aacparse;
227 GstStructure *structure;
231 aacparse = GST_AAC_PARSE (parse);
232 structure = gst_caps_get_structure (caps, 0);
233 caps_str = gst_caps_to_string (caps);
235 GST_DEBUG_OBJECT (aacparse, "setcaps: %s", caps_str);
238 /* This is needed at least in case of RTP
239 * Parses the codec_data information to get ObjectType,
240 * number of channels and samplerate */
241 value = gst_structure_get_value (structure, "codec_data");
243 GstBuffer *buf = gst_value_get_buffer (value);
249 gst_buffer_map (buf, &map, GST_MAP_READ);
251 sr_idx = ((map.data[0] & 0x07) << 1) | ((map.data[1] & 0x80) >> 7);
252 aacparse->object_type = (map.data[0] & 0xf8) >> 3;
253 aacparse->sample_rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
254 aacparse->channels = (map.data[1] & 0x78) >> 3;
255 aacparse->header_type = DSPAAC_HEADER_NONE;
256 aacparse->mpegversion = 4;
257 aacparse->frame_samples = (map.data[1] & 4) ? 960 : 1024;
258 gst_buffer_unmap (buf, &map);
260 GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
261 "samples=%d", aacparse->object_type, aacparse->sample_rate,
262 aacparse->channels, aacparse->frame_samples);
264 /* arrange for metadata and get out of the way */
265 gst_aac_parse_set_src_caps (aacparse, caps);
266 gst_base_parse_set_passthrough (parse, TRUE);
270 /* caps info overrides */
271 gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
272 gst_structure_get_int (structure, "channels", &aacparse->channels);
274 gst_base_parse_set_passthrough (parse, FALSE);
282 * gst_aac_parse_adts_get_frame_len:
283 * @data: block of data containing an ADTS header.
285 * This function calculates ADTS frame length from the given header.
287 * Returns: size of the ADTS frame.
290 gst_aac_parse_adts_get_frame_len (const guint8 * data)
292 return ((data[3] & 0x03) << 11) | (data[4] << 3) | ((data[5] & 0xe0) >> 5);
297 * gst_aac_parse_check_adts_frame:
298 * @aacparse: #GstAacParse.
299 * @data: Data to be checked.
300 * @avail: Amount of data passed.
301 * @framesize: If valid ADTS frame was found, this will be set to tell the
302 * found frame size in bytes.
303 * @needed_data: If frame was not found, this may be set to tell how much
304 * more data is needed in the next round to detect the frame
305 * reliably. This may happen when a frame header candidate
306 * is found but it cannot be guaranteed to be the header without
307 * peeking the following data.
309 * Check if the given data contains contains ADTS frame. The algorithm
310 * will examine ADTS frame header and calculate the frame size. Also, another
311 * consecutive ADTS frame header need to be present after the found frame.
312 * Otherwise the data is not considered as a valid ADTS frame. However, this
313 * "extra check" is omitted when EOS has been received. In this case it is
314 * enough when data[0] contains a valid ADTS header.
316 * This function may set the #needed_data to indicate that a possible frame
317 * candidate has been found, but more data (#needed_data bytes) is needed to
318 * be absolutely sure. When this situation occurs, FALSE will be returned.
320 * When a valid frame is detected, this function will use
321 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
322 * to set the needed bytes for next frame.This way next data chunk is already
325 * Returns: TRUE if the given data contains a valid ADTS header.
328 gst_aac_parse_check_adts_frame (GstAacParse * aacparse,
329 const guint8 * data, const guint avail, gboolean drain,
330 guint * framesize, guint * needed_data)
334 if (G_UNLIKELY (avail < 2))
337 if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
338 *framesize = gst_aac_parse_adts_get_frame_len (data);
340 /* In EOS mode this is enough. No need to examine the data further.
341 We also relax the check when we have sync, on the assumption that
342 if we're not looking at random data, we have a much higher chance
343 to get the correct sync, and this avoids losing two frames when
344 a single bit corruption happens. */
345 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
349 if (*framesize + ADTS_MAX_SIZE > avail) {
350 /* We have found a possible frame header candidate, but can't be
351 sure since we don't have enough data to check the next frame */
352 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
353 *framesize + ADTS_MAX_SIZE, avail);
354 *needed_data = *framesize + ADTS_MAX_SIZE;
355 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
356 *framesize + ADTS_MAX_SIZE);
360 if ((data[*framesize] == 0xff) && ((data[*framesize + 1] & 0xf6) == 0xf0)) {
361 guint nextlen = gst_aac_parse_adts_get_frame_len (data + (*framesize));
363 GST_LOG ("ADTS frame found, len: %d bytes", *framesize);
364 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
365 nextlen + ADTS_MAX_SIZE);
373 gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br,
376 guint8 bytes, i, byte;
379 if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
381 for (i = 0; i < bytes; ++i) {
383 if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
391 gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br,
392 guint8 * audio_object_type)
394 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5))
396 if (*audio_object_type == 31) {
397 if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6))
399 *audio_object_type += 32;
401 GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type);
406 gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
409 guint8 sampling_frequency_index;
410 if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4))
412 GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u",
413 sampling_frequency_index);
414 if (sampling_frequency_index == 0xf) {
415 guint32 sampling_rate;
416 if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24))
418 *sample_rate = sampling_rate;
420 *sample_rate = loas_sample_rate_table[sampling_frequency_index];
427 /* See table 1.13 in ISO/IEC 14496-3 */
429 gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
430 GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
432 guint8 audio_object_type, channel_configuration;
434 if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
437 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
440 if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
442 GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
443 *channels = loas_channels_table[channel_configuration];
447 if (audio_object_type == 5) {
448 GST_LOG_OBJECT (aacparse,
449 "Audio object type 5, so rereading sampling rate...");
450 if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
454 GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
455 *sample_rate, *channels);
457 /* There's LOTS of stuff next, but we ignore it for now as we have
458 what we want (sample rate and number of channels */
459 GST_DEBUG_OBJECT (aacparse,
460 "Need more code to parse humongous LOAS data, currently ignored");
468 gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
469 guint avail, gint * sample_rate, gint * channels, gint * version)
474 /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */
478 gst_bit_reader_init (&br, data, avail);
480 /* skip sync word (11 bits) and size (13 bits) */
481 if (!gst_bit_reader_skip (&br, 11 + 13))
484 /* First bit is "use last config" */
485 if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
488 GST_DEBUG_OBJECT (aacparse, "Frame uses previous config");
489 if (!aacparse->sample_rate || !aacparse->channels) {
490 GST_WARNING_OBJECT (aacparse, "No previous config to use");
492 *sample_rate = aacparse->sample_rate;
493 *channels = aacparse->channels;
497 GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
499 if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
502 if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
507 GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA);
509 guint8 same_time, subframes, num_program, prog;
512 if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
515 if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1))
517 if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6))
519 if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4))
521 GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d",
522 same_time, subframes, num_program);
524 for (prog = 0; prog <= num_program; ++prog) {
525 guint8 num_layer, layer;
526 if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3))
528 GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer);
530 for (layer = 0; layer <= num_layer; ++layer) {
531 guint8 use_same_config;
532 if (prog == 0 && layer == 0) {
535 if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1))
538 if (!use_same_config) {
540 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
541 sample_rate, channels, NULL))
544 guint32 bits, asc_len;
545 if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
547 if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
548 sample_rate, channels, &bits))
551 if (!gst_bit_reader_skip (&br, asc_len))
557 GST_WARNING_OBJECT (aacparse, "More data ignored");
559 GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
565 * gst_aac_parse_loas_get_frame_len:
566 * @data: block of data containing a LOAS header.
568 * This function calculates LOAS frame length from the given header.
570 * Returns: size of the LOAS frame.
573 gst_aac_parse_loas_get_frame_len (const guint8 * data)
575 return (((data[1] & 0x1f) << 8) | data[2]) + 3;
580 * gst_aac_parse_check_loas_frame:
581 * @aacparse: #GstAacParse.
582 * @data: Data to be checked.
583 * @avail: Amount of data passed.
584 * @framesize: If valid LOAS frame was found, this will be set to tell the
585 * found frame size in bytes.
586 * @needed_data: If frame was not found, this may be set to tell how much
587 * more data is needed in the next round to detect the frame
588 * reliably. This may happen when a frame header candidate
589 * is found but it cannot be guaranteed to be the header without
590 * peeking the following data.
592 * Check if the given data contains contains LOAS frame. The algorithm
593 * will examine LOAS frame header and calculate the frame size. Also, another
594 * consecutive LOAS frame header need to be present after the found frame.
595 * Otherwise the data is not considered as a valid LOAS frame. However, this
596 * "extra check" is omitted when EOS has been received. In this case it is
597 * enough when data[0] contains a valid LOAS header.
599 * This function may set the #needed_data to indicate that a possible frame
600 * candidate has been found, but more data (#needed_data bytes) is needed to
601 * be absolutely sure. When this situation occurs, FALSE will be returned.
603 * When a valid frame is detected, this function will use
604 * gst_base_parse_set_min_frame_size() function from #GstBaseParse class
605 * to set the needed bytes for next frame.This way next data chunk is already
608 * LOAS can have three different formats, if I read the spec correctly. Only
609 * one of them is supported here, as the two samples I have use this one.
611 * Returns: TRUE if the given data contains a valid LOAS header.
614 gst_aac_parse_check_loas_frame (GstAacParse * aacparse,
615 const guint8 * data, const guint avail, gboolean drain,
616 guint * framesize, guint * needed_data)
621 if (G_UNLIKELY (avail < 3))
624 if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
625 *framesize = gst_aac_parse_loas_get_frame_len (data);
626 GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize);
628 /* In EOS mode this is enough. No need to examine the data further.
629 We also relax the check when we have sync, on the assumption that
630 if we're not looking at random data, we have a much higher chance
631 to get the correct sync, and this avoids losing two frames when
632 a single bit corruption happens. */
633 if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) {
637 if (*framesize + LOAS_MAX_SIZE > avail) {
638 /* We have found a possible frame header candidate, but can't be
639 sure since we don't have enough data to check the next frame */
640 GST_DEBUG ("NEED MORE DATA: we need %d, available %d",
641 *framesize + LOAS_MAX_SIZE, avail);
642 *needed_data = *framesize + LOAS_MAX_SIZE;
643 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
644 *framesize + LOAS_MAX_SIZE);
648 if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) {
649 guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize));
651 GST_LOG ("LOAS frame found, len: %d bytes", *framesize);
652 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
653 nextlen + LOAS_MAX_SIZE);
660 /* caller ensure sufficient data */
662 gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
663 gint * rate, gint * channels, gint * object, gint * version)
667 gint sr_idx = (data[2] & 0x3c) >> 2;
669 *rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
672 *channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
675 *version = (data[1] & 0x08) ? 2 : 4;
677 *object = (data[2] & 0xc0) >> 6;
681 * gst_aac_parse_detect_stream:
682 * @aacparse: #GstAacParse.
683 * @data: A block of data that needs to be examined for stream characteristics.
684 * @avail: Size of the given datablock.
685 * @framesize: If valid stream was found, this will be set to tell the
686 * first frame size in bytes.
687 * @skipsize: If valid stream was found, this will be set to tell the first
688 * audio frame position within the given data.
690 * Examines the given piece of data and try to detect the format of it. It
691 * checks for "ADIF" header (in the beginning of the clip) and ADTS frame
692 * header. If the stream is detected, TRUE will be returned and #framesize
693 * is set to indicate the found frame size. Additionally, #skipsize might
694 * be set to indicate the number of bytes that need to be skipped, a.k.a. the
695 * position of the frame inside given data chunk.
697 * Returns: TRUE on success.
700 gst_aac_parse_detect_stream (GstAacParse * aacparse,
701 const guint8 * data, const guint avail, gboolean drain,
702 guint * framesize, gint * skipsize)
704 gboolean found = FALSE;
705 guint need_data_adts = 0, need_data_loas;
708 GST_DEBUG_OBJECT (aacparse, "Parsing header data");
710 /* FIXME: No need to check for ADIF if we are not in the beginning of the
713 /* Can we even parse the header? */
714 if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) {
715 GST_DEBUG_OBJECT (aacparse, "Not enough data to check");
719 for (i = 0; i < avail - 4; i++) {
720 if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
721 ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) ||
722 strncmp ((char *) data + i, "ADIF", 4) == 0) {
723 GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
727 /* Trick: tell the parent class that we didn't find the frame yet,
728 but make it skip 'i' amount of bytes. Next time we arrive
729 here we have full frame in the beginning of the data. */
742 if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain,
743 framesize, &need_data_adts)) {
746 GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
748 gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
749 &aacparse->object_type, &aacparse->mpegversion);
751 if (!channels || !framesize) {
752 GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
756 aacparse->header_type = DSPAAC_HEADER_ADTS;
757 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
758 aacparse->frame_samples, 2, 2);
760 GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
761 rate, channels, aacparse->object_type, aacparse->mpegversion);
763 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
768 if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
769 framesize, &need_data_loas)) {
772 GST_INFO ("LOAS, framesize: %d", *framesize);
774 aacparse->header_type = DSPAAC_HEADER_LOAS;
776 if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
777 &channels, &aacparse->mpegversion)) {
778 GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
782 if (rate && channels) {
783 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
784 aacparse->frame_samples, 2, 2);
786 GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
787 rate, channels, aacparse->object_type, aacparse->mpegversion);
788 aacparse->sample_rate = rate;
789 aacparse->channels = channels;
792 gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
797 if (need_data_adts || need_data_loas) {
798 /* This tells the parent class not to skip any data */
803 if (avail < ADIF_MAX_SIZE)
806 if (memcmp (data + i, "ADIF", 4) == 0) {
813 aacparse->header_type = DSPAAC_HEADER_ADIF;
814 aacparse->mpegversion = 4;
816 /* Skip the "ADIF" bytes */
819 /* copyright string */
821 skip_size += 9; /* skip 9 bytes */
823 bitstream_type = adif[0 + skip_size] & 0x10;
825 ((unsigned int) (adif[0 + skip_size] & 0x0f) << 19) |
826 ((unsigned int) adif[1 + skip_size] << 11) |
827 ((unsigned int) adif[2 + skip_size] << 3) |
828 ((unsigned int) adif[3 + skip_size] & 0xe0);
831 if (bitstream_type == 0) {
833 /* Buffer fullness parsing. Currently not needed... */
837 num_elems = (adif[3 + skip_size] & 0x1e);
838 GST_INFO ("ADIF num_config_elems: %d", num_elems);
840 fullness = ((unsigned int) (adif[3 + skip_size] & 0x01) << 19) |
841 ((unsigned int) adif[4 + skip_size] << 11) |
842 ((unsigned int) adif[5 + skip_size] << 3) |
843 ((unsigned int) (adif[6 + skip_size] & 0xe0) >> 5);
845 GST_INFO ("ADIF buffer fullness: %d", fullness);
847 aacparse->object_type = ((adif[6 + skip_size] & 0x01) << 1) |
848 ((adif[7 + skip_size] & 0x80) >> 7);
849 sr_idx = (adif[7 + skip_size] & 0x78) >> 3;
853 aacparse->object_type = (adif[4 + skip_size] & 0x18) >> 3;
854 sr_idx = ((adif[4 + skip_size] & 0x07) << 1) |
855 ((adif[5 + skip_size] & 0x80) >> 7);
858 /* FIXME: This gives totally wrong results. Duration calculation cannot
860 aacparse->sample_rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
862 /* baseparse is not given any fps,
863 * so it will give up on timestamps, seeking, etc */
865 /* FIXME: Can we assume this? */
866 aacparse->channels = 2;
868 GST_INFO ("ADIF: br=%d, samplerate=%d, objtype=%d",
869 aacparse->bitrate, aacparse->sample_rate, aacparse->object_type);
871 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 512);
873 /* arrange for metadata and get out of the way */
874 sinkcaps = gst_pad_get_current_caps (GST_BASE_PARSE_SINK_PAD (aacparse));
875 gst_aac_parse_set_src_caps (aacparse, sinkcaps);
877 gst_caps_unref (sinkcaps);
879 /* not syncable, not easily seekable (unless we push data from start */
880 gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (aacparse), FALSE);
881 gst_base_parse_set_passthrough (GST_BASE_PARSE_CAST (aacparse), TRUE);
882 gst_base_parse_set_average_bitrate (GST_BASE_PARSE_CAST (aacparse), 0);
888 /* This should never happen */
894 * gst_aac_parse_check_valid_frame:
895 * @parse: #GstBaseParse.
896 * @frame: #GstBaseParseFrame.
897 * @skipsize: How much data parent class should skip in order to find the
900 * Implementation of "handle_frame" vmethod in #GstBaseParse class.
902 * Also determines frame overhead.
903 * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
904 * a per-frame header. LOAS has 3 bytes.
906 * We're making a couple of simplifying assumptions:
908 * 1. We count Program Configuration Elements rather than searching for them
909 * in the streams to discount them - the overhead is negligible.
911 * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
912 * bits, which should still not be significant enough to warrant the
913 * additional parsing through the headers
915 * Returns: a #GstFlowReturn.
918 gst_aac_parse_handle_frame (GstBaseParse * parse,
919 GstBaseParseFrame * frame, gint * skipsize)
922 GstAacParse *aacparse;
923 gboolean ret = FALSE;
929 aacparse = GST_AAC_PARSE (parse);
930 buffer = frame->buffer;
932 gst_buffer_map (buffer, &map, GST_MAP_READ);
935 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
937 if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
938 aacparse->header_type == DSPAAC_HEADER_NONE) {
939 /* There is nothing to parse */
940 framesize = map.size;
943 } else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
945 ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
946 GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
948 } else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
949 guint needed_data = 1024;
951 ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
952 GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
955 GST_DEBUG ("buffer didn't contain valid frame");
956 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
960 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
961 guint needed_data = 1024;
963 ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
964 map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
967 GST_DEBUG ("buffer didn't contain valid frame");
968 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
973 GST_DEBUG ("buffer didn't contain valid frame");
974 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
978 if (G_UNLIKELY (!ret))
981 if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
985 gst_aac_parse_parse_adts_header (aacparse, map.data,
986 &rate, &channels, NULL, NULL);
988 GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
990 if (G_UNLIKELY (rate != aacparse->sample_rate
991 || channels != aacparse->channels)) {
992 aacparse->sample_rate = rate;
993 aacparse->channels = channels;
995 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
996 /* If linking fails, we need to return appropriate error */
997 ret = GST_FLOW_NOT_LINKED;
1000 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1001 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1003 } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) {
1004 gboolean setcaps = FALSE;
1007 frame->overhead = 3;
1009 if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
1011 GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
1012 } else if (G_UNLIKELY (rate != aacparse->sample_rate
1013 || channels != aacparse->channels)) {
1014 aacparse->sample_rate = rate;
1015 aacparse->channels = channels;
1017 GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
1021 /* We want to set caps both at start, and when rate/channels change.
1022 Since only some LOAS frames have that info, we may receive frames
1023 before knowing about rate/channels. */
1025 || !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
1026 if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
1027 /* If linking fails, we need to return appropriate error */
1028 ret = GST_FLOW_NOT_LINKED;
1031 gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
1032 aacparse->sample_rate, aacparse->frame_samples, 2, 2);
1037 gst_buffer_unmap (buffer, &map);
1040 /* found, skip if needed */
1049 if (ret && framesize <= map.size) {
1050 return gst_base_parse_finish_frame (parse, frame, framesize);
1058 * gst_aac_parse_start:
1059 * @parse: #GstBaseParse.
1061 * Implementation of "start" vmethod in #GstBaseParse class.
1063 * Returns: TRUE if startup succeeded.
1066 gst_aac_parse_start (GstBaseParse * parse)
1068 GstAacParse *aacparse;
1070 aacparse = GST_AAC_PARSE (parse);
1071 GST_DEBUG ("start");
1072 aacparse->frame_samples = 1024;
1073 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
1079 * gst_aac_parse_stop:
1080 * @parse: #GstBaseParse.
1082 * Implementation of "stop" vmethod in #GstBaseParse class.
1084 * Returns: TRUE is stopping succeeded.
1087 gst_aac_parse_stop (GstBaseParse * parse)
1094 gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
1096 GstCaps *peercaps, *templ;
1099 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1100 peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
1104 /* Remove the framed field */
1105 peercaps = gst_caps_make_writable (peercaps);
1106 n = gst_caps_get_size (peercaps);
1107 for (i = 0; i < n; i++) {
1108 GstStructure *s = gst_caps_get_structure (peercaps, i);
1110 gst_structure_remove_field (s, "framed");
1113 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1114 gst_caps_unref (peercaps);
1116 /* Append the template caps because we still want to accept
1117 * caps without any fields in the case upstream does not
1120 gst_caps_append (res, templ);
1126 GstCaps *intersection;
1129 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1130 gst_caps_unref (res);