1 /* GStreamer MPEG audio parser
2 * Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
3 * Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
4 * Copyright (C) 2010 Nokia Corporation. All rights reserved.
5 * Contact: Stefan Kost <stefan.kost@nokia.com>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
23 * SECTION:element-mpegaudioparse
24 * @short_description: MPEG audio parser
25 * @see_also: #GstAmrParse, #GstAACParse
27 * Parses and frames mpeg1 audio streams. Provides seeking.
30 * <title>Example launch line</title>
32 * gst-launch-1.0 filesrc location=test.mp3 ! mpegaudioparse ! mpg123audiodec
33 * ! audioconvert ! audioresample ! autoaudiosink
38 /* FIXME: we should make the base class (GstBaseParse) aware of the
39 * XING seek table somehow, so it can use it properly for things like
40 * accurate seeks. Currently it can only do a lookup via the convert function,
41 * but then doesn't know what the result represents exactly. One could either
42 * add a vfunc for index lookup, or just make mpegaudioparse populate the
43 * base class's index via the API provided.
51 #include "gstmpegaudioparse.h"
52 #include <gst/base/gstbytereader.h>
53 #include <gst/pbutils/pbutils.h>
55 GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug);
56 #define GST_CAT_DEFAULT mpeg_audio_parse_debug
58 #define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1
59 #define MPEG_AUDIO_CHANNEL_MODE_STEREO 0
60 #define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1
61 #define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2
62 #define MPEG_AUDIO_CHANNEL_MODE_MONO 3
64 #define CRC_UNKNOWN -1
65 #define CRC_PROTECTED 0
66 #define CRC_NOT_PROTECTED 1
68 #define XING_FRAMES_FLAG 0x0001
69 #define XING_BYTES_FLAG 0x0002
70 #define XING_TOC_FLAG 0x0004
71 #define XING_VBR_SCALE_FLAG 0x0008
73 #define MIN_FRAME_SIZE 6
75 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
78 GST_STATIC_CAPS ("audio/mpeg, "
79 "mpegversion = (int) 1, "
80 "layer = (int) [ 1, 3 ], "
81 "mpegaudioversion = (int) [ 1, 3], "
82 "rate = (int) [ 8000, 48000 ], "
83 "channels = (int) [ 1, 2 ], " "parsed=(boolean) true")
86 static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
89 GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1")
92 static void gst_mpeg_audio_parse_finalize (GObject * object);
94 static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
95 static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
96 static GstFlowReturn gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
97 GstBaseParseFrame * frame, gint * skipsize);
98 static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
99 GstBaseParseFrame * frame);
100 static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
101 GstFormat src_format, gint64 src_value,
102 GstFormat dest_format, gint64 * dest_value);
103 static GstCaps *gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse,
106 static void gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse *
107 mp3parse, GstBuffer * buf);
109 #define gst_mpeg_audio_parse_parent_class parent_class
110 G_DEFINE_TYPE (GstMpegAudioParse, gst_mpeg_audio_parse, GST_TYPE_BASE_PARSE);
112 #define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE \
113 (gst_mpeg_audio_channel_mode_get_type())
115 static const GEnumValue mpeg_audio_channel_mode[] = {
116 {MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
117 {MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"},
118 {MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
119 {MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
120 {MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
125 gst_mpeg_audio_channel_mode_get_type (void)
127 static GType mpeg_audio_channel_mode_type = 0;
129 if (!mpeg_audio_channel_mode_type) {
130 mpeg_audio_channel_mode_type =
131 g_enum_register_static ("GstMpegAudioChannelMode",
132 mpeg_audio_channel_mode);
134 return mpeg_audio_channel_mode_type;
138 gst_mpeg_audio_channel_mode_get_nick (gint mode)
141 for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) {
142 if (mpeg_audio_channel_mode[i].value == mode)
143 return mpeg_audio_channel_mode[i].value_nick;
149 gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
151 GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
152 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
153 GObjectClass *object_class = G_OBJECT_CLASS (klass);
155 GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
156 "MPEG1 audio stream parser");
158 object_class->finalize = gst_mpeg_audio_parse_finalize;
160 parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
161 parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
162 parse_class->handle_frame =
163 GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_handle_frame);
164 parse_class->pre_push_frame =
165 GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
166 parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
167 parse_class->get_sink_caps =
168 GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_get_sink_caps);
171 #define GST_TAG_CRC "has-crc"
172 #define GST_TAG_MODE "channel-mode"
174 gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
175 "has crc", "Using CRC", NULL);
176 gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
177 "channel mode", "MPEG audio channel mode", NULL);
179 g_type_class_ref (GST_TYPE_MPEG_AUDIO_CHANNEL_MODE);
181 gst_element_class_add_static_pad_template (element_class, &sink_template);
182 gst_element_class_add_static_pad_template (element_class, &src_template);
184 gst_element_class_set_static_metadata (element_class, "MPEG1 Audio Parser",
185 "Codec/Parser/Audio",
186 "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
187 "Jan Schmidt <thaytan@mad.scientist.com>,"
188 "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
192 gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
194 mp3parse->channels = -1;
196 mp3parse->sent_codec_tag = FALSE;
197 mp3parse->last_posted_crc = CRC_UNKNOWN;
198 mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
199 mp3parse->freerate = 0;
201 mp3parse->hdr_bitrate = 0;
203 mp3parse->xing_flags = 0;
204 mp3parse->xing_bitrate = 0;
205 mp3parse->xing_frames = 0;
206 mp3parse->xing_total_time = 0;
207 mp3parse->xing_bytes = 0;
208 mp3parse->xing_vbr_scale = 0;
209 memset (mp3parse->xing_seek_table, 0, sizeof (mp3parse->xing_seek_table));
210 memset (mp3parse->xing_seek_table_inverse, 0,
211 sizeof (mp3parse->xing_seek_table_inverse));
213 mp3parse->vbri_bitrate = 0;
214 mp3parse->vbri_frames = 0;
215 mp3parse->vbri_total_time = 0;
216 mp3parse->vbri_bytes = 0;
217 mp3parse->vbri_seek_points = 0;
218 g_free (mp3parse->vbri_seek_table);
219 mp3parse->vbri_seek_table = NULL;
221 mp3parse->encoder_delay = 0;
222 mp3parse->encoder_padding = 0;
226 gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse)
228 gst_mpeg_audio_parse_reset (mp3parse);
229 GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (mp3parse));
230 GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (mp3parse));
234 gst_mpeg_audio_parse_finalize (GObject * object)
236 G_OBJECT_CLASS (parent_class)->finalize (object);
240 gst_mpeg_audio_parse_start (GstBaseParse * parse)
242 GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
244 gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), MIN_FRAME_SIZE);
245 GST_DEBUG_OBJECT (parse, "starting");
247 gst_mpeg_audio_parse_reset (mp3parse);
253 gst_mpeg_audio_parse_stop (GstBaseParse * parse)
255 GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
257 GST_DEBUG_OBJECT (parse, "stopping");
259 gst_mpeg_audio_parse_reset (mp3parse);
264 static const guint mp3types_bitrates[2][3][16] = {
266 {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
267 {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
268 {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
271 {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
272 {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
273 {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
277 static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
278 {22050, 24000, 16000},
283 mp3_type_frame_length_from_header (GstMpegAudioParse * mp3parse, guint32 header,
284 guint * put_version, guint * put_layer, guint * put_channels,
285 guint * put_bitrate, guint * put_samplerate, guint * put_mode,
289 gulong mode, samplerate, bitrate, layer, channels, padding, crc;
293 if (header & (1 << 20)) {
294 lsf = (header & (1 << 19)) ? 0 : 1;
301 version = 1 + lsf + mpg25;
303 layer = 4 - ((header >> 17) & 0x3);
305 crc = (header >> 16) & 0x1;
307 bitrate = (header >> 12) & 0xF;
308 bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
310 GST_LOG_OBJECT (mp3parse, "using freeform bitrate");
311 bitrate = mp3parse->freerate;
314 samplerate = (header >> 10) & 0x3;
315 samplerate = mp3types_freqs[lsf + mpg25][samplerate];
317 /* force 0 length if 0 bitrate */
318 padding = (bitrate > 0) ? (header >> 9) & 0x1 : 0;
320 mode = (header >> 6) & 0x3;
321 channels = (mode == 3) ? 1 : 2;
325 length = 4 * ((bitrate * 12) / samplerate + padding);
328 length = (bitrate * 144) / samplerate + padding;
332 length = (bitrate * 144) / (samplerate << lsf) + padding;
336 GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
338 GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
339 "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
340 layer, channels, gst_mpeg_audio_channel_mode_get_nick (mode));
343 *put_version = version;
347 *put_channels = channels;
349 *put_bitrate = bitrate;
351 *put_samplerate = samplerate;
360 /* Minimum number of consecutive, valid-looking frames to consider
362 #define MIN_RESYNC_FRAMES 3
364 /* Perform extended validation to check that subsequent headers match
365 * the first header given here in important characteristics, to avoid
366 * false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
367 * frames to match their major characteristics.
369 * If at_eos is set to TRUE, we just check that we don't find any invalid
370 * frames in whatever data is available, rather than requiring a full
371 * MIN_RESYNC_FRAMES of data.
373 * Returns TRUE if we've seen enough data to validate or reject the frame.
374 * If TRUE is returned, then *valid contains TRUE if it validated, or false
375 * if we decided it was false sync.
376 * If FALSE is returned, then *valid contains minimum needed data.
379 gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
380 guint32 header, int bpf, gboolean at_eos, gint * valid)
385 int frames_found = 1;
388 gst_buffer_map (buf, &map, GST_MAP_READ);
390 while (frames_found < MIN_RESYNC_FRAMES) {
391 /* Check if we have enough data for all these frames, plus the next
393 if (map.size < offset + 4) {
395 /* Running out of data at EOS is fine; just accept it */
405 next_header = GST_READ_UINT32_BE (map.data + offset);
406 GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
407 offset, (unsigned int) header, (unsigned int) next_header, bpf);
409 /* mask the bits which are allowed to differ between frames */
410 #define HDRMASK ~((0xF << 12) /* bitrate */ | \
411 (0x1 << 9) /* padding */ | \
412 (0xf << 4) /* mode|mode extension */ | \
413 (0xf)) /* copyright|emphasis */
415 if ((next_header & HDRMASK) != (header & HDRMASK)) {
416 /* If any of the unmasked bits don't match, then it's not valid */
417 GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
418 "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
419 (guint) header, (guint) header & HDRMASK, (guint) next_header,
420 (guint) next_header & HDRMASK, bpf);
423 } else if (((next_header >> 12) & 0xf) == 0xf) {
424 /* The essential parts were the same, but the bitrate held an
425 invalid value - also reject */
426 GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
431 bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
432 NULL, NULL, NULL, NULL, NULL, NULL, NULL);
434 /* if no bitrate, and no freeform rate known, then fail */
435 if (G_UNLIKELY (!bpf)) {
436 GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate 0)");
448 gst_buffer_unmap (buf, &map);
453 gst_mpeg_audio_parse_head_check (GstMpegAudioParse * mp3parse,
456 GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
457 /* if it's not a valid sync */
458 if ((head & 0xffe00000) != 0xffe00000) {
459 GST_WARNING_OBJECT (mp3parse, "invalid sync");
462 /* if it's an invalid MPEG version */
463 if (((head >> 19) & 3) == 0x1) {
464 GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
468 /* if it's an invalid layer */
469 if (!((head >> 17) & 3)) {
470 GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
473 /* if it's an invalid bitrate */
474 if (((head >> 12) & 0xf) == 0xf) {
475 GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
478 /* if it's an invalid samplerate */
479 if (((head >> 10) & 0x3) == 0x3) {
480 GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
485 if ((head & 0x3) == 0x2) {
486 /* Ignore this as there are some files with emphasis 0x2 that can
487 * be played fine. See BGO #537235 */
488 GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
494 /* Determines possible freeform frame rate/size by looking for next
495 * header with valid bitrate (0 or otherwise valid) (and sufficiently
496 * matching current header).
498 * Returns TRUE if we've found such one, and *rate then contains rate
499 * (or *rate contains 0 if decided no freeframe size could be determined).
500 * If not enough data, returns FALSE.
503 gst_mp3parse_find_freerate (GstMpegAudioParse * mp3parse, GstMapInfo * map,
504 guint32 header, gboolean at_eos, gint * _rate)
510 gulong samplerate, rate, layer, padding;
514 available = map->size;
519 /* pick apart header again partially */
520 if (header & (1 << 20)) {
521 lsf = (header & (1 << 19)) ? 0 : 1;
527 layer = 4 - ((header >> 17) & 0x3);
528 samplerate = (header >> 10) & 0x3;
529 samplerate = mp3types_freqs[lsf + mpg25][samplerate];
530 padding = (header >> 9) & 0x1;
532 for (; offset < available; ++offset) {
533 /* Check if we have enough data for all these frames, plus the next
535 if (available < offset + 4) {
537 /* Running out of data; failed to determine size */
545 next_header = GST_READ_UINT32_BE (data + offset);
546 if ((next_header & 0xFFE00000) != 0xFFE00000)
549 GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X",
550 offset, (unsigned int) header, (unsigned int) next_header);
552 if ((next_header & HDRMASK) != (header & HDRMASK)) {
553 /* If any of the unmasked bits don't match, then it's not valid */
554 GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
555 "(header=%08X (%08X), header2=%08X (%08X))",
556 (guint) header, (guint) header & HDRMASK, (guint) next_header,
557 (guint) next_header & HDRMASK);
559 } else if (((next_header >> 12) & 0xf) == 0xf) {
560 /* The essential parts were the same, but the bitrate held an
561 invalid value - also reject */
562 GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
569 /* almost accept as free frame */
571 rate = samplerate * (offset - 4 * padding + 4) / 48000;
573 rate = samplerate * (offset - padding + 1) / (144 >> lsf) / 1000;
577 GST_LOG_OBJECT (mp3parse, "calculated rate %lu", rate * 1000);
578 if (rate < 8 || (layer == 3 && rate > 640)) {
579 GST_DEBUG_OBJECT (mp3parse, "rate invalid");
581 /* maybe some hope */
584 GST_DEBUG_OBJECT (mp3parse, "aborting");
589 *_rate = rate * 1000;
592 /* avoid indefinite searching */
594 GST_DEBUG_OBJECT (mp3parse, "exceeded sanity rate; aborting");
604 gst_mpeg_audio_parse_handle_frame (GstBaseParse * parse,
605 GstBaseParseFrame * frame, gint * skipsize)
607 GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
608 GstBuffer *buf = frame->buffer;
609 GstByteReader reader;
611 gboolean lost_sync, draining, valid, caps_change;
613 guint bitrate, layer, rate, channels, version, mode, crc;
615 gboolean res = FALSE;
617 gst_buffer_map (buf, &map, GST_MAP_READ);
618 if (G_UNLIKELY (map.size < 6)) {
623 gst_byte_reader_init (&reader, map.data, map.size);
625 off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
628 GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
630 /* didn't find anything that looks like a sync word, skip */
632 *skipsize = map.size - 3;
636 /* possible frame header, but not at offset 0? skip bytes before sync */
642 /* make sure the values in the frame header look sane */
643 header = GST_READ_UINT32_BE (map.data);
644 if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
649 GST_LOG_OBJECT (parse, "got frame");
651 lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
652 draining = GST_BASE_PARSE_DRAINING (parse);
654 if (G_UNLIKELY (lost_sync))
655 mp3parse->freerate = 0;
657 bpf = mp3_type_frame_length_from_header (mp3parse, header,
658 &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
660 if (channels != mp3parse->channels || rate != mp3parse->rate ||
661 layer != mp3parse->layer || version != mp3parse->version)
666 /* maybe free format */
668 GST_LOG_OBJECT (mp3parse, "possibly free format");
669 if (lost_sync || mp3parse->freerate == 0) {
670 GST_DEBUG_OBJECT (mp3parse, "finding free format rate");
671 if (!gst_mp3parse_find_freerate (mp3parse, &map, header, draining,
673 /* not enough data */
674 gst_base_parse_set_min_frame_size (parse, valid);
678 GST_DEBUG_OBJECT (parse, "determined freeform size %d", valid);
679 mp3parse->freerate = valid;
683 bpf = mp3_type_frame_length_from_header (mp3parse, header,
684 &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
686 /* did not come up with valid freeform length, reject after all */
692 if (!draining && (lost_sync || caps_change)) {
693 if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
695 /* not enough data */
696 gst_base_parse_set_min_frame_size (parse, valid);
705 } else if (draining && lost_sync && caps_change && mp3parse->rate > 0) {
706 /* avoid caps jitter that we can't be sure of */
711 /* restore default minimum */
712 gst_base_parse_set_min_frame_size (parse, MIN_FRAME_SIZE);
716 /* metadata handling */
717 if (G_UNLIKELY (caps_change)) {
718 GstCaps *caps = gst_caps_new_simple ("audio/mpeg",
719 "mpegversion", G_TYPE_INT, 1,
720 "mpegaudioversion", G_TYPE_INT, version,
721 "layer", G_TYPE_INT, layer,
722 "rate", G_TYPE_INT, rate,
723 "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
724 gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
725 gst_caps_unref (caps);
727 mp3parse->rate = rate;
728 mp3parse->channels = channels;
729 mp3parse->layer = layer;
730 mp3parse->version = version;
732 /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
733 if (mp3parse->layer == 1)
735 else if (mp3parse->layer == 2)
736 mp3parse->spf = 1152;
737 else if (mp3parse->version == 1) {
738 mp3parse->spf = 1152;
740 /* MPEG-2 or "2.5" */
745 * We start pushing 9 frames earlier (29 frames for MPEG2) than
746 * segment start to be able to decode the first frame we want.
747 * 9 (29) frames are the theoretical maximum of frames that contain
748 * data for the current frame (bit reservoir).
751 * Some mp3 streams have an offset in the timestamps, for which we have to
752 * push the frame *after* the end position in order for the decoder to be
753 * able to decode everything up until the segment.stop position. */
754 gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf,
755 (version == 1) ? 10 : 30, 2);
758 mp3parse->hdr_bitrate = bitrate;
760 /* For first frame; check for seek tables and output a codec tag */
761 gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
763 /* store some frame info for later processing */
764 mp3parse->last_crc = crc;
765 mp3parse->last_mode = mode;
768 gst_buffer_unmap (buf, &map);
770 if (res && bpf <= map.size) {
771 return gst_base_parse_finish_frame (parse, frame, bpf);
778 gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse,
781 const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
782 const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
783 const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */
784 const guint32 lame_id = 0x4c414d45; /* 'LAME' in hex */
785 gint offset_xing, offset_vbri;
787 gint64 upstream_total_bytes = 0;
788 guint32 read_id_xing = 0, read_id_vbri = 0;
793 if (mp3parse->sent_codec_tag)
796 /* Check first frame for Xing info */
797 if (mp3parse->version == 1) { /* MPEG-1 file */
798 if (mp3parse->channels == 1)
802 } else { /* MPEG-2 header */
803 if (mp3parse->channels == 1)
809 /* The VBRI tag is always at offset 0x20 */
812 /* Skip the 4 bytes of the MP3 header too */
816 /* Check if we have enough data to read the Xing header */
817 gst_buffer_map (buf, &map, GST_MAP_READ);
821 if (avail >= offset_xing + 4) {
822 read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
824 if (avail >= offset_vbri + 4) {
825 read_id_vbri = GST_READ_UINT32_BE (data + offset_vbri);
828 /* obtain real upstream total bytes */
829 if (!gst_pad_peer_query_duration (GST_BASE_PARSE_SINK_PAD (mp3parse),
830 GST_FORMAT_BYTES, &upstream_total_bytes))
831 upstream_total_bytes = 0;
833 if (read_id_xing == xing_id || read_id_xing == info_id) {
835 guint bytes_needed = offset_xing + 8;
837 GstClockTime total_time;
839 GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
841 /* Move data after Xing header */
842 data += offset_xing + 4;
844 /* Read 4 base bytes of flags, big-endian */
845 xing_flags = GST_READ_UINT32_BE (data);
847 if (xing_flags & XING_FRAMES_FLAG)
849 if (xing_flags & XING_BYTES_FLAG)
851 if (xing_flags & XING_TOC_FLAG)
853 if (xing_flags & XING_VBR_SCALE_FLAG)
855 if (avail < bytes_needed) {
856 GST_DEBUG_OBJECT (mp3parse,
857 "Not enough data to read Xing header (need %d)", bytes_needed);
861 GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
862 mp3parse->xing_flags = xing_flags;
864 if (xing_flags & XING_FRAMES_FLAG) {
865 mp3parse->xing_frames = GST_READ_UINT32_BE (data);
866 if (mp3parse->xing_frames == 0) {
867 GST_WARNING_OBJECT (mp3parse,
868 "Invalid number of frames in Xing header");
869 mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
871 mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
872 (guint64) (mp3parse->xing_frames) * (mp3parse->spf),
878 mp3parse->xing_frames = 0;
879 mp3parse->xing_total_time = 0;
882 if (xing_flags & XING_BYTES_FLAG) {
883 mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
884 if (mp3parse->xing_bytes == 0) {
885 GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
886 mp3parse->xing_flags &= ~XING_BYTES_FLAG;
890 mp3parse->xing_bytes = 0;
893 /* If we know the upstream size and duration, compute the
894 * total bitrate, rounded up to the nearest kbit/sec */
895 if ((total_time = mp3parse->xing_total_time) &&
896 (total_bytes = mp3parse->xing_bytes)) {
897 mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
898 8 * GST_SECOND, total_time);
899 mp3parse->xing_bitrate += 500;
900 mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
903 if (xing_flags & XING_TOC_FLAG) {
905 guchar *table = mp3parse->xing_seek_table;
910 GST_DEBUG_OBJECT (mp3parse,
911 "Subtracting initial offset of %d bytes from Xing TOC", first);
913 /* xing seek table: percent time -> 1/256 bytepos */
914 for (i = 0; i < 100; i++) {
915 new = data[i] - first;
917 GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
918 mp3parse->xing_flags &= ~XING_TOC_FLAG;
921 mp3parse->xing_seek_table[i] = old = new;
924 /* build inverse table: 1/256 bytepos -> 1/100 percent time */
925 for (i = 0; i < 256; i++) {
926 while (percent < 99 && table[percent + 1] <= i)
929 if (table[percent] == i) {
930 mp3parse->xing_seek_table_inverse[i] = percent * 100;
931 } else if (percent < 99 && table[percent]) {
933 gint a = percent, b = percent + 1;
937 fx = (b - a) / (fb - fa) * (i - fa) + a;
938 mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
939 } else if (percent == 99) {
941 gint a = percent, b = 100;
945 fx = (b - a) / (fb - fa) * (i - fa) + a;
946 mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
952 memset (mp3parse->xing_seek_table, 0, sizeof (mp3parse->xing_seek_table));
953 memset (mp3parse->xing_seek_table_inverse, 0,
954 sizeof (mp3parse->xing_seek_table_inverse));
957 if (xing_flags & XING_VBR_SCALE_FLAG) {
958 mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
961 mp3parse->xing_vbr_scale = 0;
963 GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
964 GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
965 GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
966 mp3parse->xing_vbr_scale);
968 /* check for truncated file */
969 if (upstream_total_bytes && mp3parse->xing_bytes &&
970 mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
971 GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
972 "invalidating Xing header duration and size");
973 mp3parse->xing_flags &= ~XING_BYTES_FLAG;
974 mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
977 /* Optional LAME tag? */
978 if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) {
979 gchar lame_version[10] = { 0, };
981 guint32 encoder_delay, encoder_padding;
983 memcpy (lame_version, data, 9);
985 tag_rev = data[0] >> 4;
986 GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'",
987 tag_rev, lame_version);
989 /* Skip all the information we're not interested in */
991 /* Encoder delay and end padding */
992 encoder_delay = GST_READ_UINT24_BE (data);
993 encoder_delay >>= 12;
994 encoder_padding = GST_READ_UINT24_BE (data);
995 encoder_padding &= 0x000fff;
997 mp3parse->encoder_delay = encoder_delay;
998 mp3parse->encoder_padding = encoder_padding;
1000 GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
1001 encoder_delay, encoder_padding);
1003 } else if (read_id_vbri == vbri_id) {
1004 gint64 total_bytes, total_frames;
1005 GstClockTime total_time;
1006 guint16 nseek_points;
1008 GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
1010 if (avail < offset_vbri + 26) {
1011 GST_DEBUG_OBJECT (mp3parse,
1012 "Not enough data to read VBRI header (need %d)", offset_vbri + 26);
1016 GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
1018 /* Move data after VBRI header */
1019 data += offset_vbri + 4;
1021 if (GST_READ_UINT16_BE (data) != 0x0001) {
1022 GST_WARNING_OBJECT (mp3parse,
1023 "Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
1028 /* Skip encoder delay */
1034 total_bytes = GST_READ_UINT32_BE (data);
1035 if (total_bytes != 0)
1036 mp3parse->vbri_bytes = total_bytes;
1039 total_frames = GST_READ_UINT32_BE (data);
1040 if (total_frames != 0) {
1041 mp3parse->vbri_frames = total_frames;
1042 mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
1043 (guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
1047 /* If we know the upstream size and duration, compute the
1048 * total bitrate, rounded up to the nearest kbit/sec */
1049 if ((total_time = mp3parse->vbri_total_time) &&
1050 (total_bytes = mp3parse->vbri_bytes)) {
1051 mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
1052 8 * GST_SECOND, total_time);
1053 mp3parse->vbri_bitrate += 500;
1054 mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
1057 nseek_points = GST_READ_UINT16_BE (data);
1060 if (nseek_points > 0) {
1061 guint scale, seek_bytes, seek_frames;
1064 mp3parse->vbri_seek_points = nseek_points;
1066 scale = GST_READ_UINT16_BE (data);
1069 seek_bytes = GST_READ_UINT16_BE (data);
1072 seek_frames = GST_READ_UINT16_BE (data);
1074 if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
1075 GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
1079 if (avail < offset_vbri + 26 + nseek_points * seek_bytes) {
1080 GST_WARNING_OBJECT (mp3parse,
1081 "Not enough data to read VBRI seek table (need %d)",
1082 offset_vbri + 26 + nseek_points * seek_bytes);
1086 if (seek_frames * nseek_points < total_frames - seek_frames ||
1087 seek_frames * nseek_points > total_frames + seek_frames) {
1088 GST_WARNING_OBJECT (mp3parse,
1089 "VBRI seek table doesn't cover the complete file");
1094 data += offset_vbri + 26;
1096 /* VBRI seek table: frame/seek_frames -> byte */
1097 mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
1098 if (seek_bytes == 4)
1099 for (i = 0; i < nseek_points; i++) {
1100 mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
1102 } else if (seek_bytes == 3)
1103 for (i = 0; i < nseek_points; i++) {
1104 mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
1106 } else if (seek_bytes == 2)
1107 for (i = 0; i < nseek_points; i++) {
1108 mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
1110 } else /* seek_bytes == 1 */
1111 for (i = 0; i < nseek_points; i++) {
1112 mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
1118 GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
1119 GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
1120 GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
1122 /* check for truncated file */
1123 if (upstream_total_bytes && mp3parse->vbri_bytes &&
1124 mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
1125 GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
1126 "invalidating VBRI header duration and size");
1127 mp3parse->vbri_valid = FALSE;
1129 mp3parse->vbri_valid = TRUE;
1132 GST_DEBUG_OBJECT (mp3parse,
1133 "Xing, LAME or VBRI header not found in first frame");
1136 /* set duration if tables provided a valid one */
1137 if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
1138 gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
1139 mp3parse->xing_total_time, 0);
1141 if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
1142 gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
1143 mp3parse->vbri_total_time, 0);
1146 /* tell baseclass how nicely we can seek, and a bitrate if one found */
1147 /* FIXME: fill index with seek table */
1149 seekable = GST_BASE_PARSE_SEEK_DEFAULT;
1150 if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes &&
1151 mp3parse->xing_total_time)
1152 seekable = GST_BASE_PARSE_SEEK_TABLE;
1154 if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes &&
1155 mp3parse->vbri_total_time)
1156 seekable = GST_BASE_PARSE_SEEK_TABLE;
1159 if (mp3parse->xing_bitrate)
1160 bitrate = mp3parse->xing_bitrate;
1161 else if (mp3parse->vbri_bitrate)
1162 bitrate = mp3parse->vbri_bitrate;
1166 gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
1169 gst_buffer_unmap (buf, &map);
1173 gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse,
1174 GstClockTime ts, gint64 * bytepos)
1177 GstClockTime total_time;
1179 /* If XING seek table exists use this for time->byte conversion */
1180 if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
1181 (total_bytes = mp3parse->xing_bytes) &&
1182 (total_time = mp3parse->xing_total_time)) {
1185 CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
1186 gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
1187 gint index = CLAMP (percent, 0, 99);
1189 fa = mp3parse->xing_seek_table[index];
1191 fb = mp3parse->xing_seek_table[index + 1];
1195 fx = fa + (fb - fa) * (percent - index);
1197 *bytepos = (1.0 / 256.0) * fx * total_bytes;
1202 if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) &&
1203 (total_time = mp3parse->vbri_total_time)) {
1205 gdouble a, b, fa, fb;
1207 i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
1208 i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
1210 a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
1211 mp3parse->vbri_seek_points));
1213 for (j = i; j >= 0; j--)
1214 fa += mp3parse->vbri_seek_table[j];
1216 if (i + 1 < mp3parse->vbri_seek_points) {
1217 b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
1218 mp3parse->vbri_seek_points));
1219 fb = fa + mp3parse->vbri_seek_table[i + 1];
1221 b = gst_guint64_to_gdouble (total_time);
1225 *bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
1234 gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse,
1235 gint64 bytepos, GstClockTime * ts)
1238 GstClockTime total_time;
1240 /* If XING seek table exists use this for byte->time conversion */
1241 if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
1242 (total_bytes = mp3parse->xing_bytes) &&
1243 (total_time = mp3parse->xing_total_time)) {
1248 pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
1249 index = CLAMP (pos, 0, 255);
1250 fa = mp3parse->xing_seek_table_inverse[index];
1252 fb = mp3parse->xing_seek_table_inverse[index + 1];
1256 fx = fa + (fb - fa) * (pos - index);
1258 *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time);
1263 if (mp3parse->vbri_seek_table &&
1264 (total_bytes = mp3parse->vbri_bytes) &&
1265 (total_time = mp3parse->vbri_total_time)) {
1268 gdouble a, b, fa, fb;
1271 sum += mp3parse->vbri_seek_table[i];
1273 } while (i + 1 < mp3parse->vbri_seek_points
1274 && sum + mp3parse->vbri_seek_table[i] < bytepos);
1277 a = gst_guint64_to_gdouble (sum);
1278 fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
1279 mp3parse->vbri_seek_points));
1281 if (i + 1 < mp3parse->vbri_seek_points) {
1282 b = a + mp3parse->vbri_seek_table[i + 1];
1283 fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
1284 mp3parse->vbri_seek_points));
1287 fb = gst_guint64_to_gdouble (total_time);
1290 *ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
1299 gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format,
1300 gint64 src_value, GstFormat dest_format, gint64 * dest_value)
1302 GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
1303 gboolean res = FALSE;
1305 if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES)
1307 gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value);
1308 else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME)
1309 res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value,
1310 (GstClockTime *) dest_value);
1312 /* if no tables, fall back to default estimated rate based conversion */
1314 return gst_base_parse_convert_default (parse, src_format, src_value,
1315 dest_format, dest_value);
1320 static GstFlowReturn
1321 gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
1322 GstBaseParseFrame * frame)
1324 GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
1325 GstTagList *taglist = NULL;
1327 /* we will create a taglist (if any of the parameters has changed)
1328 * to add the tags that changed */
1329 if (mp3parse->last_posted_crc != mp3parse->last_crc) {
1333 taglist = gst_tag_list_new_empty ();
1335 mp3parse->last_posted_crc = mp3parse->last_crc;
1336 if (mp3parse->last_posted_crc == CRC_PROTECTED) {
1341 gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
1345 if (mp3parse->last_posted_channel_mode != mp3parse->last_mode) {
1347 taglist = gst_tag_list_new_empty ();
1349 mp3parse->last_posted_channel_mode = mp3parse->last_mode;
1351 gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
1352 gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL);
1355 /* tag sending done late enough in hook to ensure pending events
1356 * have already been sent */
1357 if (taglist != NULL || !mp3parse->sent_codec_tag) {
1360 if (taglist == NULL)
1361 taglist = gst_tag_list_new_empty ();
1364 caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
1365 if (G_UNLIKELY (caps == NULL)) {
1366 gst_tag_list_unref (taglist);
1368 if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
1369 GST_INFO_OBJECT (parse, "Src pad is flushing");
1370 return GST_FLOW_FLUSHING;
1372 GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
1373 return GST_FLOW_NOT_NEGOTIATED;
1376 gst_pb_utils_add_codec_description_to_tag_list (taglist,
1377 GST_TAG_AUDIO_CODEC, caps);
1378 gst_caps_unref (caps);
1380 if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
1381 mp3parse->vbri_bitrate == 0) {
1382 /* We don't have a VBR bitrate, so post the available bitrate as
1383 * nominal and let baseparse calculate the real bitrate */
1384 gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
1385 GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
1388 /* also signals the end of first-frame processing */
1389 mp3parse->sent_codec_tag = TRUE;
1392 /* if the taglist exists, we need to update it so it gets sent out */
1394 gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
1395 gst_tag_list_unref (taglist);
1398 /* usual clipping applies */
1399 frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
1405 remove_fields (GstCaps * caps)
1409 n = gst_caps_get_size (caps);
1410 for (i = 0; i < n; i++) {
1411 GstStructure *s = gst_caps_get_structure (caps, i);
1413 gst_structure_remove_field (s, "parsed");
1418 gst_mpeg_audio_parse_get_sink_caps (GstBaseParse * parse, GstCaps * filter)
1420 GstCaps *peercaps, *templ;
1423 templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
1425 GstCaps *fcopy = gst_caps_copy (filter);
1426 /* Remove the fields we convert */
1427 remove_fields (fcopy);
1428 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
1429 gst_caps_unref (fcopy);
1431 peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
1434 /* Remove the parsed field */
1435 peercaps = gst_caps_make_writable (peercaps);
1436 remove_fields (peercaps);
1438 res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
1439 gst_caps_unref (peercaps);
1440 gst_caps_unref (templ);
1446 GstCaps *intersection;
1449 gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
1450 gst_caps_unref (res);