1 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
3 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
4 * Copyright (C) <2006> Nokia Corporation, 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-wavparse
25 * Parse a .wav file into raw or compressed audio.
27 * Wavparse supports both push and pull mode operations, making it possible to
28 * stream from a network source.
31 * <title>Example launch line</title>
33 * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
34 * ]| Read a wav file and output to the soundcard using the ALSA element. The
35 * wav file is assumed to contain raw uncompressed samples.
37 * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
38 * ]| Stream data from a network url.
44 * http://replaygain.hydrogenaudio.org/file_format_wav.html
54 #include "gstwavparse.h"
55 #include "gst/riff/riff-media.h"
56 #include <gst/base/gsttypefindhelper.h>
57 #include <gst/gst-i18n-plugin.h>
59 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
60 #define GST_CAT_DEFAULT (wavparse_debug)
62 #define GST_RIFF_TAG_Fake GST_MAKE_FOURCC ('F','a','k','e')
64 #define GST_BWF_TAG_iXML GST_MAKE_FOURCC ('i','X','M','L')
65 #define GST_BWF_TAG_qlty GST_MAKE_FOURCC ('q','l','t','y')
66 #define GST_BWF_TAG_mext GST_MAKE_FOURCC ('m','e','x','t')
67 #define GST_BWF_TAG_levl GST_MAKE_FOURCC ('l','e','v','l')
68 #define GST_BWF_TAG_link GST_MAKE_FOURCC ('l','i','n','k')
69 #define GST_BWF_TAG_axml GST_MAKE_FOURCC ('a','x','m','l')
71 /* Data size chunk of RF64,
72 * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
73 #define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
75 static void gst_wavparse_dispose (GObject * object);
77 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
79 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
80 GstObject * parent, GstPadMode mode, gboolean active);
81 static gboolean gst_wavparse_send_event (GstElement * element,
83 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
84 GstStateChange transition);
86 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
88 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
89 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
91 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
93 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
95 static void gst_wavparse_loop (GstPad * pad);
96 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
99 static void gst_wavparse_set_property (GObject * object, guint prop_id,
100 const GValue * value, GParamSpec * pspec);
101 static void gst_wavparse_get_property (GObject * object, guint prop_id,
102 GValue * value, GParamSpec * pspec);
104 #define DEFAULT_IGNORE_LENGTH FALSE
112 static GstStaticPadTemplate sink_template_factory =
113 GST_STATIC_PAD_TEMPLATE ("sink",
116 GST_STATIC_CAPS ("audio/x-wav")
120 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
122 #define gst_wavparse_parent_class parent_class
123 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
128 /* Offset Size Description Value
129 * 0x00 4 ID unique identification value
130 * 0x04 4 Position play order position
131 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
132 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
133 * 0x10 4 Block Start Byte Offset to sample of First Channel
134 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
138 guint32 data_chunk_id;
141 guint32 sample_offset;
146 /* Offset Size Description Value
147 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
150 guint32 cue_point_id;
152 } GstWavParseLabl, GstWavParseNote;
155 gst_wavparse_class_init (GstWavParseClass * klass)
157 GstElementClass *gstelement_class;
158 GObjectClass *object_class;
159 GstPadTemplate *src_template;
161 gstelement_class = (GstElementClass *) klass;
162 object_class = (GObjectClass *) klass;
164 parent_class = g_type_class_peek_parent (klass);
166 object_class->dispose = gst_wavparse_dispose;
168 object_class->set_property = gst_wavparse_set_property;
169 object_class->get_property = gst_wavparse_get_property;
172 * GstWavParse:ignore-length:
174 * This selects whether the length found in a data chunk
175 * should be ignored. This may be useful for streamed audio
176 * where the length is unknown until the end of streaming,
177 * and various software/hardware just puts some random value
178 * in there and hopes it doesn't break too much.
180 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
181 g_param_spec_boolean ("ignore-length",
183 "Ignore length from the Wave header",
184 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
187 gstelement_class->change_state = gst_wavparse_change_state;
188 gstelement_class->send_event = gst_wavparse_send_event;
191 gst_element_class_add_pad_template (gstelement_class,
192 gst_static_pad_template_get (&sink_template_factory));
194 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
195 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
196 gst_element_class_add_pad_template (gstelement_class, src_template);
198 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
199 "Codec/Demuxer/Audio",
200 "Parse a .wav file into raw audio",
201 "Erik Walthinsen <omega@cse.ogi.edu>");
205 gst_wavparse_reset (GstWavParse * wav)
207 wav->state = GST_WAVPARSE_START;
209 /* These will all be set correctly in the fmt chunk */
223 wav->got_fmt = FALSE;
227 gst_event_unref (wav->seek_event);
228 wav->seek_event = NULL;
230 gst_adapter_clear (wav->adapter);
231 g_object_unref (wav->adapter);
235 gst_tag_list_unref (wav->tags);
238 gst_toc_unref (wav->toc);
241 g_list_free_full (wav->cues, g_free);
244 g_list_free_full (wav->labls, g_free);
247 gst_caps_unref (wav->caps);
249 if (wav->start_segment)
250 gst_event_unref (wav->start_segment);
251 wav->start_segment = NULL;
255 gst_wavparse_dispose (GObject * object)
257 GstWavParse *wav = GST_WAVPARSE (object);
259 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
260 gst_wavparse_reset (wav);
262 G_OBJECT_CLASS (parent_class)->dispose (object);
266 gst_wavparse_init (GstWavParse * wavparse)
268 gst_wavparse_reset (wavparse);
272 gst_pad_new_from_static_template (&sink_template_factory, "sink");
273 gst_pad_set_activate_function (wavparse->sinkpad,
274 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
275 gst_pad_set_activatemode_function (wavparse->sinkpad,
276 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
277 gst_pad_set_chain_function (wavparse->sinkpad,
278 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
279 gst_pad_set_event_function (wavparse->sinkpad,
280 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
281 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
285 gst_pad_new_from_template (gst_element_class_get_pad_template
286 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
287 gst_pad_use_fixed_caps (wavparse->srcpad);
288 gst_pad_set_query_function (wavparse->srcpad,
289 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
290 gst_pad_set_event_function (wavparse->srcpad,
291 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
292 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
296 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
300 if (!gst_riff_parse_file_header (element, buf, &doctype))
303 if (doctype != GST_RIFF_RIFF_WAVE)
311 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
312 ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
318 gst_wavparse_stream_init (GstWavParse * wav)
321 GstBuffer *buf = NULL;
323 if ((res = gst_pad_pull_range (wav->sinkpad,
324 wav->offset, 12, &buf)) != GST_FLOW_OK)
326 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
327 return GST_FLOW_ERROR;
335 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
337 /* -1 always maps to -1 */
343 /* 0 always maps to 0 */
350 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
352 } else if (wav->fact) {
353 guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
354 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
361 /* This function is used to perform seeks on the element.
363 * It also works when event is NULL, in which case it will just
364 * start from the last configured segment. This technique is
365 * used when activating the element and to perform the seek in
369 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
373 GstFormat format, bformat;
375 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
376 gint64 cur, stop, upstream_size;
379 GstSegment seeksegment = { 0, };
384 GST_DEBUG_OBJECT (wav, "doing seek with event");
386 gst_event_parse_seek (event, &rate, &format, &flags,
387 &cur_type, &cur, &stop_type, &stop);
388 seqnum = gst_event_get_seqnum (event);
390 /* no negative rates yet */
394 if (format != wav->segment.format) {
395 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
396 gst_format_get_name (format),
397 gst_format_get_name (wav->segment.format));
399 if (cur_type != GST_SEEK_TYPE_NONE)
401 gst_pad_query_convert (wav->srcpad, format, cur,
402 wav->segment.format, &cur);
403 if (res && stop_type != GST_SEEK_TYPE_NONE)
405 gst_pad_query_convert (wav->srcpad, format, stop,
406 wav->segment.format, &stop);
410 format = wav->segment.format;
413 GST_DEBUG_OBJECT (wav, "doing seek without event");
416 cur_type = GST_SEEK_TYPE_SET;
417 stop_type = GST_SEEK_TYPE_SET;
420 /* in push mode, we must delegate to upstream */
421 if (wav->streaming) {
422 gboolean res = FALSE;
424 /* if streaming not yet started; only prepare initial newsegment */
425 if (!event || wav->state != GST_WAVPARSE_DATA) {
426 if (wav->start_segment)
427 gst_event_unref (wav->start_segment);
428 wav->start_segment = gst_event_new_segment (&wav->segment);
431 /* convert seek positions to byte positions in data sections */
432 if (format == GST_FORMAT_TIME) {
433 /* should not fail */
434 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
436 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
439 /* mind sample boundary and header */
441 cur -= (cur % wav->bytes_per_sample);
442 cur += wav->datastart;
445 stop -= (stop % wav->bytes_per_sample);
446 stop += wav->datastart;
448 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
449 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
451 /* BYTE seek event */
452 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
454 gst_event_set_seqnum (event, seqnum);
455 res = gst_pad_push_event (wav->sinkpad, event);
461 flush = flags & GST_SEEK_FLAG_FLUSH;
463 /* now we need to make sure the streaming thread is stopped. We do this by
464 * either sending a FLUSH_START event downstream which will cause the
465 * streaming thread to stop with a WRONG_STATE.
466 * For a non-flushing seek we simply pause the task, which will happen as soon
467 * as it completes one iteration (and thus might block when the sink is
468 * blocking in preroll). */
471 GST_DEBUG_OBJECT (wav, "sending flush start");
473 fevent = gst_event_new_flush_start ();
474 gst_event_set_seqnum (fevent, seqnum);
475 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
476 gst_pad_push_event (wav->srcpad, fevent);
478 gst_pad_pause_task (wav->sinkpad);
481 /* we should now be able to grab the streaming thread because we stopped it
482 * with the above flush/pause code */
483 GST_PAD_STREAM_LOCK (wav->sinkpad);
485 /* save current position */
486 last_stop = wav->segment.position;
488 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
490 /* copy segment, we need this because we still need the old
491 * segment when we close the current segment. */
492 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
494 /* configure the seek parameters in the seeksegment. We will then have the
495 * right values in the segment to perform the seek */
497 GST_DEBUG_OBJECT (wav, "configuring seek");
498 gst_segment_do_seek (&seeksegment, rate, format, flags,
499 cur_type, cur, stop_type, stop, &update);
502 /* figure out the last position we need to play. If it's configured (stop !=
503 * -1), use that, else we play until the total duration of the file */
504 if ((stop = seeksegment.stop) == -1)
505 stop = seeksegment.duration;
507 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
508 if ((cur_type != GST_SEEK_TYPE_NONE)) {
509 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
510 * we can just copy the last_stop. If not, we use the bps to convert TIME to
512 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
513 (gint64 *) & wav->offset))
514 wav->offset = seeksegment.position;
515 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
516 wav->offset -= (wav->offset % wav->bytes_per_sample);
517 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
518 wav->offset += wav->datastart;
519 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
521 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
525 if (stop_type != GST_SEEK_TYPE_NONE) {
526 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
527 wav->end_offset = stop;
528 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
529 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
530 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
531 wav->end_offset += wav->datastart;
532 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
534 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
538 /* make sure filesize is not exceeded due to rounding errors or so,
539 * same precaution as in _stream_headers */
540 bformat = GST_FORMAT_BYTES;
541 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
542 wav->end_offset = MIN (wav->end_offset, upstream_size);
544 /* this is the range of bytes we will use for playback */
545 wav->offset = MIN (wav->offset, wav->end_offset);
546 wav->dataleft = wav->end_offset - wav->offset;
548 GST_DEBUG_OBJECT (wav,
549 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
550 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
551 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
553 /* prepare for streaming again */
557 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
558 GST_DEBUG_OBJECT (wav, "sending flush stop");
560 fevent = gst_event_new_flush_stop (TRUE);
561 gst_event_set_seqnum (fevent, seqnum);
562 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
563 gst_pad_push_event (wav->srcpad, fevent);
566 /* now we did the seek and can activate the new segment values */
567 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
569 /* if we're doing a segment seek, post a SEGMENT_START message */
570 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
571 gst_element_post_message (GST_ELEMENT_CAST (wav),
572 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
573 wav->segment.format, wav->segment.position));
576 /* now create the newsegment */
577 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
578 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
580 /* store the newsegment event so it can be sent from the streaming thread. */
581 if (wav->start_segment)
582 gst_event_unref (wav->start_segment);
583 wav->start_segment = gst_event_new_segment (&wav->segment);
584 gst_event_set_seqnum (wav->start_segment, seqnum);
586 /* mark discont if we are going to stream from another position. */
587 if (last_stop != wav->segment.position) {
588 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
592 /* and start the streaming task again */
593 if (!wav->streaming) {
594 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
598 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
605 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
610 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
615 GST_DEBUG_OBJECT (wav,
616 "Could not determine byte position for desired time");
622 * gst_wavparse_peek_chunk_info:
623 * @wav Wavparse object
624 * @tag holder for tag
625 * @size holder for tag size
627 * Peek next chunk info (tag and size)
629 * Returns: %TRUE when the chunk info (header) is available
632 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
634 const guint8 *data = NULL;
636 if (gst_adapter_available (wav->adapter) < 8)
639 data = gst_adapter_map (wav->adapter, 8);
640 *tag = GST_READ_UINT32_LE (data);
641 *size = GST_READ_UINT32_LE (data + 4);
642 gst_adapter_unmap (wav->adapter);
644 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
645 GST_FOURCC_ARGS (*tag));
651 * gst_wavparse_peek_chunk:
652 * @wav Wavparse object
653 * @tag holder for tag
654 * @size holder for tag size
656 * Peek enough data for one full chunk
658 * Returns: %TRUE when the full chunk is available
661 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
663 guint32 peek_size = 0;
666 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
669 /* size 0 -> empty data buffer would surprise most callers,
670 * large size -> do not bother trying to squeeze that into adapter,
671 * so we throw poor man's exception, which can be caught if caller really
672 * wants to handle 0 size chunk */
673 if (!(*size) || (*size) >= (1 << 30)) {
674 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
675 *size, GST_FOURCC_ARGS (*tag));
676 /* chain should give up */
677 wav->abort_buffering = TRUE;
680 peek_size = (*size + 1) & ~1;
681 available = gst_adapter_available (wav->adapter);
683 if (available >= (8 + peek_size)) {
686 GST_LOG ("but only %u bytes available now", available);
692 * gst_wavparse_calculate_duration:
693 * @wav: wavparse object
695 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
698 * Returns: %TRUE if duration is available.
701 gst_wavparse_calculate_duration (GstWavParse * wav)
703 if (wav->duration > 0)
707 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
709 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
711 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
712 GST_TIME_ARGS (wav->duration));
714 } else if (wav->fact) {
716 gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
717 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
718 GST_TIME_ARGS (wav->duration));
725 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
730 if (wav->streaming) {
731 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
734 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
735 GST_FOURCC_ARGS (tag));
736 flush = 8 + ((size + 1) & ~1);
737 wav->offset += flush;
738 if (wav->streaming) {
739 gst_adapter_flush (wav->adapter, flush);
741 gst_buffer_unref (buf);
748 * gst_wavparse_cue_chunk:
749 * @wav GstWavParse object
750 * @data holder for data
751 * @size holder for data size
753 * Parse cue chunk from @data to wav->cues.
755 * Returns: %TRUE when cue chunk is available
758 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
765 GST_WARNING_OBJECT (wav, "found another cue's");
769 ncues = GST_READ_UINT32_LE (data);
771 if (size < 4 + ncues * 24) {
772 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
778 for (i = 0; i < ncues; i++) {
779 cue = g_new0 (GstWavParseCue, 1);
780 cue->id = GST_READ_UINT32_LE (data);
781 cue->position = GST_READ_UINT32_LE (data + 4);
782 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
783 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
784 cue->block_start = GST_READ_UINT32_LE (data + 16);
785 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
786 cues = g_list_append (cues, cue);
796 * gst_wavparse_labl_chunk:
797 * @wav GstWavParse object
798 * @data holder for data
799 * @size holder for data size
801 * Parse labl from @data to wav->labls.
803 * Returns: %TRUE when labl chunk is available
806 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
808 GstWavParseLabl *labl;
813 labl = g_new0 (GstWavParseLabl, 1);
817 labl->cue_point_id = GST_READ_UINT32_LE (data);
818 labl->text = g_memdup (data + 4, size - 4);
820 wav->labls = g_list_append (wav->labls, labl);
826 * gst_wavparse_note_chunk:
827 * @wav GstWavParse object
828 * @data holder for data
829 * @size holder for data size
831 * Parse note from @data to wav->notes.
833 * Returns: %TRUE when note chunk is available
836 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
838 GstWavParseNote *note;
843 note = g_new0 (GstWavParseNote, 1);
847 note->cue_point_id = GST_READ_UINT32_LE (data);
848 note->text = g_memdup (data + 4, size - 4);
850 wav->notes = g_list_append (wav->notes, note);
856 * gst_wavparse_smpl_chunk:
857 * @wav GstWavParse object
858 * @data holder for data
859 * @size holder for data size
861 * Parse smpl chunk from @data.
863 * Returns: %TRUE when cue chunk is available
866 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
871 manufacturer_id = GST_READ_UINT32_LE (data);
872 product_id = GST_READ_UINT32_LE (data + 4);
873 sample_period = GST_READ_UINT32_LE (data + 8);
875 note_number = GST_READ_UINT32_LE (data + 12);
877 pitch_fraction = GST_READ_UINT32_LE (data + 16);
878 SMPTE_format = GST_READ_UINT32_LE (data + 20);
879 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
880 num_sample_loops = GST_READ_UINT32_LE (data + 28);
881 List of Sample Loops, 24 bytes each
885 wav->tags = gst_tag_list_new_empty ();
886 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
887 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
892 * gst_wavparse_adtl_chunk:
893 * @wav GstWavParse object
894 * @data holder for data
895 * @size holder for data size
897 * Parse adtl from @data.
899 * Returns: %TRUE when adtl chunk is available
902 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
904 guint32 ltag, lsize, offset = 0;
907 ltag = GST_READ_UINT32_LE (data + offset);
908 lsize = GST_READ_UINT32_LE (data + offset + 4);
910 if (lsize + 8 > size) {
911 GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
916 case GST_RIFF_TAG_labl:
917 gst_wavparse_labl_chunk (wav, data + offset, size);
919 case GST_RIFF_TAG_note:
920 gst_wavparse_note_chunk (wav, data + offset, size);
923 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
924 GST_FOURCC_ARGS (ltag));
925 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
928 offset += 8 + GST_ROUND_UP_2 (lsize);
929 size -= 8 + GST_ROUND_UP_2 (lsize);
936 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
938 GstTagList *tags = NULL;
939 GstTocEntry *entry = NULL;
941 entry = gst_toc_find_entry (toc, id);
943 tags = gst_toc_entry_get_tags (entry);
945 tags = gst_tag_list_new_empty ();
946 gst_toc_entry_set_tags (entry, tags);
954 * gst_wavparse_create_toc:
955 * @wav GstWavParse object
957 * Create TOC from wav->cues and wav->labls.
960 gst_wavparse_create_toc (GstWavParse * wav)
966 GstWavParseLabl *labl;
967 GstWavParseNote *note;
970 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
972 GST_OBJECT_LOCK (wav);
974 GST_OBJECT_UNLOCK (wav);
975 GST_WARNING_OBJECT (wav, "found another TOC");
980 GST_OBJECT_UNLOCK (wav);
984 /* FIXME: send CURRENT scope toc too */
985 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
987 /* add cue edition */
988 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
989 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
990 gst_toc_append_entry (toc, entry);
992 /* add tracks in cue edition */
996 prev_subentry = cur_subentry;
997 /* previous track stop time = current track start time */
998 if (prev_subentry != NULL) {
999 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1000 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1001 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1003 id = g_strdup_printf ("%08x", cue->id);
1004 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1006 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1007 stop = wav->duration;
1008 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1009 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1010 list = g_list_next (list);
1013 /* add tags in tracks */
1017 id = g_strdup_printf ("%08x", labl->cue_point_id);
1018 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1021 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1024 list = g_list_next (list);
1029 id = g_strdup_printf ("%08x", note->cue_point_id);
1030 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1033 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1036 list = g_list_next (list);
1039 /* send data as TOC */
1042 /* send TOC event */
1044 GST_OBJECT_UNLOCK (wav);
1045 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1051 #define MAX_BUFFER_SIZE 4096
1054 parse_ds64 (GstWavParse * wav, GstBuffer * buf)
1057 guint32 dataSizeLow, dataSizeHigh;
1058 guint32 sampleCountLow, sampleCountHigh;
1060 gst_buffer_map (buf, &map, GST_MAP_READ);
1061 dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
1062 dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
1063 sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
1064 sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
1065 gst_buffer_unmap (buf, &map);
1066 if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
1067 wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
1069 if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
1070 wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
1073 GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
1074 " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
1078 static GstFlowReturn
1079 gst_wavparse_stream_headers (GstWavParse * wav)
1081 GstFlowReturn res = GST_FLOW_OK;
1082 GstBuffer *buf = NULL;
1083 gst_riff_strf_auds *header = NULL;
1085 gboolean gotdata = FALSE;
1086 GstCaps *caps = NULL;
1087 gchar *codec_name = NULL;
1088 gint64 upstream_size = 0;
1091 /* search for "_fmt" chunk, which should be first */
1092 while (!wav->got_fmt) {
1095 /* The header starts with a 'fmt ' tag */
1096 if (wav->streaming) {
1097 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1100 gst_adapter_flush (wav->adapter, 8);
1104 buf = gst_adapter_take_buffer (wav->adapter, size);
1106 gst_adapter_flush (wav->adapter, 1);
1107 wav->offset += GST_ROUND_UP_2 (size);
1109 buf = gst_buffer_new ();
1112 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1113 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1117 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1118 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1119 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1120 tag == GST_RIFF_TAG_id3 || tag == GST_RIFF_TAG_IDVX ||
1121 tag == GST_BWF_TAG_iXML || tag == GST_BWF_TAG_qlty ||
1122 tag == GST_BWF_TAG_mext || tag == GST_BWF_TAG_levl ||
1123 tag == GST_BWF_TAG_link || tag == GST_BWF_TAG_axml ||
1124 tag == GST_RIFF_TAG_Fake) {
1125 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1126 GST_FOURCC_ARGS (tag));
1127 gst_buffer_unref (buf);
1132 if (tag == GST_RS64_TAG_DS64) {
1133 if (!parse_ds64 (wav, buf))
1139 if (tag != GST_RIFF_TAG_fmt)
1142 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1144 goto parse_header_error;
1146 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1148 /* do sanity checks of header fields */
1149 if (header->channels == 0)
1151 if (header->rate == 0)
1154 GST_DEBUG_OBJECT (wav, "creating the caps");
1156 /* Note: gst_riff_create_audio_caps might need to fix values in
1157 * the header header depending on the format, so call it first */
1158 /* FIXME: Need to handle the channel reorder map */
1159 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1160 NULL, &codec_name, NULL);
1163 gst_buffer_unref (extra);
1166 goto unknown_format;
1168 /* If we got raw audio from upstream, we remove the codec_data field,
1169 * which may have been added if the wav header included an extended
1170 * chunk. We want to keep it for non raw audio.
1172 s = gst_caps_get_structure (caps, 0);
1173 if (s && gst_structure_has_name (s, "audio/x-raw")) {
1174 gst_structure_remove_field (s, "codec_data");
1177 /* do more sanity checks of header fields
1178 * (these can be sanitized by gst_riff_create_audio_caps()
1180 wav->format = header->format;
1181 wav->rate = header->rate;
1182 wav->channels = header->channels;
1183 wav->blockalign = header->blockalign;
1184 wav->depth = header->bits_per_sample;
1185 wav->av_bps = header->av_bps;
1191 /* do format specific handling */
1192 switch (wav->format) {
1193 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1194 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1196 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1197 * bitrate inside the mpeg stream */
1198 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1202 case GST_RIFF_WAVE_FORMAT_PCM:
1203 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1204 goto invalid_blockalign;
1207 if (wav->av_bps > wav->blockalign * wav->rate)
1209 /* use the configured bps */
1210 wav->bps = wav->av_bps;
1214 wav->width = (wav->blockalign * 8) / wav->channels;
1215 wav->bytes_per_sample = wav->channels * wav->width / 8;
1217 if (wav->bytes_per_sample <= 0)
1218 goto no_bytes_per_sample;
1220 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1221 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1222 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1223 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1224 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1225 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1226 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1228 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1229 * formats). This will make the element output a BYTE format segment and
1230 * will not timestamp the outgoing buffers.
1232 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1234 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1236 /* create pad later so we can sniff the first few bytes
1237 * of the real data and correct our caps if necessary */
1238 gst_caps_replace (&wav->caps, caps);
1239 gst_caps_replace (&caps, NULL);
1241 wav->got_fmt = TRUE;
1244 wav->tags = gst_tag_list_new_empty ();
1246 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1247 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1249 g_free (codec_name);
1255 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1256 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1258 /* loop headers until we get data */
1260 if (wav->streaming) {
1261 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1268 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1269 &buf)) != GST_FLOW_OK)
1270 goto header_read_error;
1271 gst_buffer_map (buf, &map, GST_MAP_READ);
1272 tag = GST_READ_UINT32_LE (map.data);
1273 size = GST_READ_UINT32_LE (map.data + 4);
1274 gst_buffer_unmap (buf, &map);
1277 GST_INFO_OBJECT (wav,
1278 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
1279 G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
1281 /* Maximum valid size is INT_MAX */
1282 if (size & 0x80000000) {
1283 GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
1287 /* Clip to upstream size if known */
1288 if (wav->datasize > 0 && size + wav->offset > wav->datasize) {
1289 GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
1290 size = wav->datasize - wav->offset;
1293 /* wav is a st00pid format, we don't know for sure where data starts.
1294 * So we have to go bit by bit until we find the 'data' header
1297 case GST_RIFF_TAG_data:{
1300 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1302 if (wav->ignore_length) {
1303 GST_DEBUG_OBJECT (wav, "Ignoring length");
1306 if (wav->streaming) {
1307 gst_adapter_flush (wav->adapter, 8);
1310 gst_buffer_unref (buf);
1313 wav->datastart = wav->offset;
1314 /* use size from ds64 chunk if available */
1315 if (size64 == -1 && wav->datasize > 0) {
1316 GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
1317 size64 = wav->datasize;
1319 /* If size is zero, then the data chunk probably actually extends to
1320 the end of the file */
1321 if (size64 == 0 && upstream_size) {
1322 size64 = upstream_size - wav->datastart;
1324 /* Or the file might be truncated */
1325 else if (upstream_size) {
1326 size64 = MIN (size64, (upstream_size - wav->datastart));
1328 wav->datasize = size64;
1329 wav->dataleft = size64;
1330 wav->end_offset = size64 + wav->datastart;
1331 if (!wav->streaming) {
1332 /* We will continue parsing tags 'till end */
1333 wav->offset += size64;
1335 GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
1338 case GST_RIFF_TAG_fact:{
1339 if (wav->fact == 0 &&
1340 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1341 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1342 const guint data_size = 4;
1344 GST_INFO_OBJECT (wav, "Have fact chunk");
1345 if (size < data_size) {
1346 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1347 /* need more data */
1350 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1354 /* number of samples (for compressed formats) */
1355 if (wav->streaming) {
1356 const guint8 *data = NULL;
1358 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1361 gst_adapter_flush (wav->adapter, 8);
1362 data = gst_adapter_map (wav->adapter, data_size);
1363 wav->fact = GST_READ_UINT32_LE (data);
1364 gst_adapter_unmap (wav->adapter);
1365 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1367 gst_buffer_unref (buf);
1370 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1371 data_size, &buf)) != GST_FLOW_OK)
1372 goto header_read_error;
1373 gst_buffer_extract (buf, 0, &wav->fact, 4);
1374 wav->fact = GUINT32_FROM_LE (wav->fact);
1375 gst_buffer_unref (buf);
1377 GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
1378 wav->offset += 8 + GST_ROUND_UP_2 (size);
1381 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1382 /* need more data */
1388 case GST_RIFF_TAG_acid:{
1389 const gst_riff_acid *acid = NULL;
1390 const guint data_size = sizeof (gst_riff_acid);
1393 GST_INFO_OBJECT (wav, "Have acid chunk");
1394 if (size < data_size) {
1395 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1396 /* need more data */
1399 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1403 if (wav->streaming) {
1404 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1407 gst_adapter_flush (wav->adapter, 8);
1408 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1410 tempo = acid->tempo;
1411 gst_adapter_unmap (wav->adapter);
1414 gst_buffer_unref (buf);
1417 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1418 size, &buf)) != GST_FLOW_OK)
1419 goto header_read_error;
1420 gst_buffer_map (buf, &map, GST_MAP_READ);
1421 acid = (const gst_riff_acid *) map.data;
1422 tempo = acid->tempo;
1423 gst_buffer_unmap (buf, &map);
1425 /* send data as tags */
1427 wav->tags = gst_tag_list_new_empty ();
1428 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1429 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1431 size = GST_ROUND_UP_2 (size);
1432 if (wav->streaming) {
1433 gst_adapter_flush (wav->adapter, size);
1435 gst_buffer_unref (buf);
1437 wav->offset += 8 + size;
1440 /* FIXME: all list tags after data are ignored in streaming mode */
1441 case GST_RIFF_TAG_LIST:{
1444 if (wav->streaming) {
1445 const guint8 *data = NULL;
1447 if (gst_adapter_available (wav->adapter) < 12) {
1450 data = gst_adapter_map (wav->adapter, 12);
1451 ltag = GST_READ_UINT32_LE (data + 8);
1452 gst_adapter_unmap (wav->adapter);
1454 gst_buffer_unref (buf);
1457 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1458 &buf)) != GST_FLOW_OK)
1459 goto header_read_error;
1460 gst_buffer_extract (buf, 8, <ag, 4);
1461 ltag = GUINT32_FROM_LE (ltag);
1464 case GST_RIFF_LIST_INFO:{
1465 const gint data_size = size - 4;
1468 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1469 if (wav->streaming) {
1470 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1473 gst_adapter_flush (wav->adapter, 12);
1475 if (data_size > 0) {
1476 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1478 gst_adapter_flush (wav->adapter, 1);
1482 gst_buffer_unref (buf);
1484 if (data_size > 0) {
1486 gst_pad_pull_range (wav->sinkpad, wav->offset,
1487 data_size, &buf)) != GST_FLOW_OK)
1488 goto header_read_error;
1491 if (data_size > 0) {
1493 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1495 GstTagList *old = wav->tags;
1497 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1499 gst_tag_list_unref (old);
1500 gst_tag_list_unref (new);
1502 gst_buffer_unref (buf);
1503 wav->offset += GST_ROUND_UP_2 (data_size);
1507 case GST_RIFF_LIST_adtl:{
1508 const gint data_size = size - 4;
1510 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1511 if (wav->streaming) {
1512 const guint8 *data = NULL;
1514 gst_adapter_flush (wav->adapter, 12);
1516 data = gst_adapter_map (wav->adapter, data_size);
1517 gst_wavparse_adtl_chunk (wav, data, data_size);
1518 gst_adapter_unmap (wav->adapter);
1522 gst_buffer_unref (buf);
1526 gst_pad_pull_range (wav->sinkpad, wav->offset,
1527 data_size, &buf)) != GST_FLOW_OK)
1528 goto header_read_error;
1529 gst_buffer_map (buf, &map, GST_MAP_READ);
1530 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1532 gst_buffer_unmap (buf, &map);
1534 wav->offset += GST_ROUND_UP_2 (data_size);
1538 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1539 GST_FOURCC_ARGS (ltag));
1540 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1541 /* need more data */
1547 case GST_RIFF_TAG_cue:{
1548 const guint data_size = size;
1550 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1551 if (wav->streaming) {
1552 const guint8 *data = NULL;
1554 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1557 gst_adapter_flush (wav->adapter, 8);
1559 data = gst_adapter_map (wav->adapter, data_size);
1560 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1561 goto header_read_error;
1563 gst_adapter_unmap (wav->adapter);
1568 gst_buffer_unref (buf);
1571 gst_pad_pull_range (wav->sinkpad, wav->offset,
1572 data_size, &buf)) != GST_FLOW_OK)
1573 goto header_read_error;
1574 gst_buffer_map (buf, &map, GST_MAP_READ);
1575 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1577 goto header_read_error;
1579 gst_buffer_unmap (buf, &map);
1581 size = GST_ROUND_UP_2 (size);
1582 if (wav->streaming) {
1583 gst_adapter_flush (wav->adapter, size);
1585 gst_buffer_unref (buf);
1587 size = GST_ROUND_UP_2 (size);
1588 wav->offset += size;
1591 case GST_RIFF_TAG_smpl:{
1592 const gint data_size = size;
1594 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1595 if (wav->streaming) {
1596 const guint8 *data = NULL;
1598 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1601 gst_adapter_flush (wav->adapter, 8);
1603 data = gst_adapter_map (wav->adapter, data_size);
1604 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1605 goto header_read_error;
1607 gst_adapter_unmap (wav->adapter);
1612 gst_buffer_unref (buf);
1615 gst_pad_pull_range (wav->sinkpad, wav->offset,
1616 data_size, &buf)) != GST_FLOW_OK)
1617 goto header_read_error;
1618 gst_buffer_map (buf, &map, GST_MAP_READ);
1619 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1621 goto header_read_error;
1623 gst_buffer_unmap (buf, &map);
1625 size = GST_ROUND_UP_2 (size);
1626 if (wav->streaming) {
1627 gst_adapter_flush (wav->adapter, size);
1629 gst_buffer_unref (buf);
1631 size = GST_ROUND_UP_2 (size);
1632 wav->offset += size;
1636 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1637 GST_FOURCC_ARGS (tag));
1638 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1639 /* need more data */
1644 if (upstream_size && (wav->offset >= upstream_size)) {
1645 /* Now we are gone through the whole file */
1650 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1652 if (wav->bps <= 0 && wav->fact) {
1654 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1656 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1657 (guint64) wav->fact);
1658 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1663 if (gst_wavparse_calculate_duration (wav)) {
1664 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1665 if (!wav->ignore_length)
1666 wav->segment.duration = wav->duration;
1668 gst_wavparse_create_toc (wav);
1670 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1671 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1672 if (!wav->ignore_length)
1673 wav->segment.duration = wav->datasize;
1676 /* now we have all the info to perform a pending seek if any, if no
1677 * event, this will still do the right thing and it will also send
1678 * the right newsegment event downstream. */
1679 gst_wavparse_perform_seek (wav, wav->seek_event);
1680 /* remove pending event */
1681 gst_event_replace (&wav->seek_event, NULL);
1683 /* we just started, we are discont */
1684 wav->discont = TRUE;
1686 wav->state = GST_WAVPARSE_DATA;
1688 /* determine reasonable max buffer size,
1689 * that is, buffers not too small either size or time wise
1690 * so we do not end up with too many of them */
1692 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1693 wav->max_buf_size = upstream_size;
1695 wav->max_buf_size = 0;
1696 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1697 if (wav->blockalign > 0)
1698 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1700 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1707 g_free (codec_name);
1710 gst_caps_unref (caps);
1715 res = GST_FLOW_ERROR;
1720 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1721 ("Invalid WAV header (no fmt at start): %"
1722 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1727 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1728 ("Couldn't parse audio header"));
1733 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1734 ("Stream claims to contain no channels - invalid data"));
1739 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1740 ("Stream with sample_rate == 0 - invalid data"));
1745 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1746 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1747 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1752 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1753 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1754 wav->av_bps, wav->blockalign * wav->rate));
1757 no_bytes_per_sample:
1759 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1760 ("Could not caluclate bytes per sample - invalid data"));
1765 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1766 ("No caps found for format 0x%x, %u channels, %u Hz",
1767 wav->format, wav->channels, wav->rate));
1772 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1773 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1779 * Read WAV file tag when streaming
1781 static GstFlowReturn
1782 gst_wavparse_parse_stream_init (GstWavParse * wav)
1784 if (gst_adapter_available (wav->adapter) >= 12) {
1787 /* _take flushes the data */
1788 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1790 GST_DEBUG ("Parsing wav header");
1791 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1792 return GST_FLOW_ERROR;
1795 /* Go to next state */
1796 wav->state = GST_WAVPARSE_HEADER;
1801 /* handle an event sent directly to the element.
1803 * This event can be sent either in the READY state or the
1804 * >READY state. The only event of interest really is the seek
1807 * In the READY state we can only store the event and try to
1808 * respect it when going to PAUSED. We assume we are in the
1809 * READY state when our parsing state != GST_WAVPARSE_DATA.
1811 * When we are steaming, we can simply perform the seek right
1815 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1817 GstWavParse *wav = GST_WAVPARSE (element);
1818 gboolean res = FALSE;
1820 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1822 switch (GST_EVENT_TYPE (event)) {
1823 case GST_EVENT_SEEK:
1824 if (wav->state == GST_WAVPARSE_DATA) {
1825 /* we can handle the seek directly when streaming data */
1826 res = gst_wavparse_perform_seek (wav, event);
1828 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1830 gst_event_replace (&wav->seek_event, event);
1832 /* we always return true */
1839 gst_event_unref (event);
1844 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1848 s = gst_caps_get_structure (caps, 0);
1849 if (!gst_structure_has_name (s, "audio/x-dts"))
1851 /* typefind behavior for DTS:
1852 * MAXIMUM: multiple frame syncs detected, certainly DTS
1853 * LIKELY: single frame sync at offset 0. Maybe DTS?
1854 * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
1856 if (prob > GST_TYPE_FIND_LIKELY)
1858 if (prob <= GST_TYPE_FIND_POSSIBLE)
1860 /* for maybe, check for at least a valid-looking rate and channels */
1861 if (!gst_structure_has_field (s, "channels"))
1863 /* and for extra assurance we could also check the rate from the DTS frame
1864 * against the one in the wav header, but for now let's not do that */
1865 return gst_structure_has_field (s, "rate");
1869 gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
1871 GstTagList *tags = NULL;
1876 while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
1877 gst_event_parse_tag (ev, &tags);
1878 if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
1879 tags = gst_tag_list_copy (tags);
1880 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1881 gst_event_unref (ev);
1885 gst_event_unref (ev);
1891 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1894 GstTagList *tags, *utags;
1896 GST_DEBUG_OBJECT (wav, "adding src pad");
1898 g_assert (wav->caps != NULL);
1900 s = gst_caps_get_structure (wav->caps, 0);
1901 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1902 GstTypeFindProbability prob;
1905 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1906 if (tf_caps != NULL) {
1907 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1908 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1909 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1910 gst_caps_unref (wav->caps);
1911 wav->caps = tf_caps;
1913 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1914 GST_TAG_AUDIO_CODEC, "dts", NULL);
1916 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1917 "marked as raw PCM audio, but ignoring for now", tf_caps);
1918 gst_caps_unref (tf_caps);
1923 gst_pad_set_caps (wav->srcpad, wav->caps);
1924 gst_caps_replace (&wav->caps, NULL);
1926 if (wav->start_segment) {
1927 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1928 gst_pad_push_event (wav->srcpad, wav->start_segment);
1929 wav->start_segment = NULL;
1932 /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
1933 * that there'll be only one scope/type of tag list from upstream, if any */
1934 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
1936 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
1938 /* if there's a tag upstream it's probably been added to override the
1939 * tags from inside the wav header, so keep upstream tags if in doubt */
1940 tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
1942 if (wav->tags != NULL) {
1943 gst_tag_list_unref (wav->tags);
1948 gst_tag_list_unref (utags);
1950 /* send tags downstream, if any */
1952 gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
1955 static GstFlowReturn
1956 gst_wavparse_stream_data (GstWavParse * wav)
1958 GstBuffer *buf = NULL;
1959 GstFlowReturn res = GST_FLOW_OK;
1960 guint64 desired, obtained;
1961 GstClockTime timestamp, next_timestamp, duration;
1962 guint64 pos, nextpos;
1965 GST_LOG_OBJECT (wav,
1966 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1967 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1969 /* Get the next n bytes and output them */
1970 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1973 /* scale the amount of data by the segment rate so we get equal
1974 * amounts of data regardless of the playback rate */
1976 MIN (gst_guint64_to_gdouble (wav->dataleft),
1977 wav->max_buf_size * ABS (wav->segment.rate));
1979 if (desired >= wav->blockalign && wav->blockalign > 0)
1980 desired -= (desired % wav->blockalign);
1982 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1983 "from the sinkpad", desired);
1985 if (wav->streaming) {
1986 guint avail = gst_adapter_available (wav->adapter);
1989 /* flush some bytes if evil upstream sends segment that starts
1990 * before data or does is not send sample aligned segment */
1991 if (G_LIKELY (wav->offset >= wav->datastart)) {
1992 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1994 extra = wav->datastart - wav->offset;
1997 if (G_UNLIKELY (extra)) {
1998 extra = wav->bytes_per_sample - extra;
1999 if (extra <= avail) {
2000 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2001 gst_adapter_flush (wav->adapter, extra);
2002 wav->offset += extra;
2003 wav->dataleft -= extra;
2004 goto iterate_adapter;
2006 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2007 gst_adapter_clear (wav->adapter);
2008 wav->offset += avail;
2009 wav->dataleft -= avail;
2014 if (avail < desired) {
2015 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2019 buf = gst_adapter_take_buffer (wav->adapter, desired);
2021 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2022 desired, &buf)) != GST_FLOW_OK)
2025 /* we may get a short buffer at the end of the file */
2026 if (gst_buffer_get_size (buf) < desired) {
2027 gsize size = gst_buffer_get_size (buf);
2029 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2030 if (size >= wav->blockalign) {
2031 if (wav->blockalign > 0) {
2032 buf = gst_buffer_make_writable (buf);
2033 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2036 gst_buffer_unref (buf);
2042 obtained = gst_buffer_get_size (buf);
2044 /* our positions in bytes */
2045 pos = wav->offset - wav->datastart;
2046 nextpos = pos + obtained;
2048 /* update offsets, does not overflow. */
2049 buf = gst_buffer_make_writable (buf);
2050 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2051 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2053 /* first chunk of data? create the source pad. We do this only here so
2054 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2055 if (G_UNLIKELY (wav->first)) {
2057 /* this will also push the segment events */
2058 gst_wavparse_add_src_pad (wav, buf);
2060 /* If we have a pending start segment, send it now. */
2061 if (G_UNLIKELY (wav->start_segment != NULL)) {
2062 gst_pad_push_event (wav->srcpad, wav->start_segment);
2063 wav->start_segment = NULL;
2068 /* and timestamps if we have a bitrate, be careful for overflows */
2070 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2072 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2073 duration = next_timestamp - timestamp;
2075 /* update current running segment position */
2076 if (G_LIKELY (next_timestamp >= wav->segment.start))
2077 wav->segment.position = next_timestamp;
2078 } else if (wav->fact) {
2080 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2081 /* and timestamps if we have a bitrate, be careful for overflows */
2082 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2083 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2084 duration = next_timestamp - timestamp;
2086 /* no bitrate, all we know is that the first sample has timestamp 0, all
2087 * other positions and durations have unknown timestamp. */
2091 timestamp = GST_CLOCK_TIME_NONE;
2092 duration = GST_CLOCK_TIME_NONE;
2093 /* update current running segment position with byte offset */
2094 if (G_LIKELY (nextpos >= wav->segment.start))
2095 wav->segment.position = nextpos;
2097 if ((pos > 0) && wav->vbr) {
2098 /* don't set timestamps for VBR files if it's not the first buffer */
2099 timestamp = GST_CLOCK_TIME_NONE;
2100 duration = GST_CLOCK_TIME_NONE;
2103 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2104 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2105 wav->discont = FALSE;
2108 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2109 GST_BUFFER_DURATION (buf) = duration;
2111 GST_LOG_OBJECT (wav,
2112 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2113 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2114 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2116 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2119 if (obtained < wav->dataleft) {
2120 wav->offset += obtained;
2121 wav->dataleft -= obtained;
2123 wav->offset += wav->dataleft;
2127 /* Iterate until need more data, so adapter size won't grow */
2128 if (wav->streaming) {
2129 GST_LOG_OBJECT (wav,
2130 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2132 goto iterate_adapter;
2139 GST_DEBUG_OBJECT (wav, "found EOS");
2140 return GST_FLOW_EOS;
2144 /* check if we got EOS */
2145 if (res == GST_FLOW_EOS)
2148 GST_WARNING_OBJECT (wav,
2149 "Error getting %" G_GINT64_FORMAT " bytes from the "
2150 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2155 GST_INFO_OBJECT (wav,
2156 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2157 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2158 gst_pad_is_linked (wav->srcpad));
2164 gst_wavparse_loop (GstPad * pad)
2167 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2171 GST_LOG_OBJECT (wav, "process data");
2173 switch (wav->state) {
2174 case GST_WAVPARSE_START:
2175 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2176 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2180 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2181 event = gst_event_new_stream_start (stream_id);
2182 gst_event_set_group_id (event, gst_util_group_id_next ());
2183 gst_pad_push_event (wav->srcpad, event);
2186 wav->state = GST_WAVPARSE_HEADER;
2189 case GST_WAVPARSE_HEADER:
2190 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2191 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2194 wav->state = GST_WAVPARSE_DATA;
2195 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2198 case GST_WAVPARSE_DATA:
2199 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2203 g_assert_not_reached ();
2210 const gchar *reason = gst_flow_get_name (ret);
2212 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2213 gst_pad_pause_task (pad);
2215 if (ret == GST_FLOW_EOS) {
2216 /* handle end-of-stream/segment */
2217 /* so align our position with the end of it, if there is one
2218 * this ensures a subsequent will arrive at correct base/acc time */
2219 if (wav->segment.format == GST_FORMAT_TIME) {
2220 if (wav->segment.rate > 0.0 &&
2221 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2222 wav->segment.position = wav->segment.stop;
2223 else if (wav->segment.rate < 0.0)
2224 wav->segment.position = wav->segment.start;
2226 if (wav->state == GST_WAVPARSE_START) {
2227 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2228 ("No valid input found before end of stream"));
2229 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2231 /* add pad before we perform EOS */
2232 if (G_UNLIKELY (wav->first)) {
2234 gst_wavparse_add_src_pad (wav, NULL);
2237 /* perform EOS logic */
2238 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2241 if ((stop = wav->segment.stop) == -1)
2242 stop = wav->segment.duration;
2244 gst_element_post_message (GST_ELEMENT_CAST (wav),
2245 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2246 wav->segment.format, stop));
2247 gst_pad_push_event (wav->srcpad,
2248 gst_event_new_segment_done (wav->segment.format, stop));
2250 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2253 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2254 /* for fatal errors we post an error message, post the error
2255 * first so the app knows about the error first. */
2256 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2257 (_("Internal data flow error.")),
2258 ("streaming task paused, reason %s (%d)", reason, ret));
2259 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2265 static GstFlowReturn
2266 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2269 GstWavParse *wav = GST_WAVPARSE (parent);
2271 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2272 gst_buffer_get_size (buf));
2274 gst_adapter_push (wav->adapter, buf);
2276 switch (wav->state) {
2277 case GST_WAVPARSE_START:
2278 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2279 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2282 if (wav->state != GST_WAVPARSE_HEADER)
2285 /* otherwise fall-through */
2286 case GST_WAVPARSE_HEADER:
2287 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2288 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2291 if (!wav->got_fmt || wav->datastart == 0)
2294 wav->state = GST_WAVPARSE_DATA;
2295 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2298 case GST_WAVPARSE_DATA:
2299 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2300 wav->discont = TRUE;
2301 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2305 g_return_val_if_reached (GST_FLOW_ERROR);
2308 if (G_UNLIKELY (wav->abort_buffering)) {
2309 wav->abort_buffering = FALSE;
2310 ret = GST_FLOW_ERROR;
2311 /* sort of demux/parse error */
2312 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2318 static GstFlowReturn
2319 gst_wavparse_flush_data (GstWavParse * wav)
2321 GstFlowReturn ret = GST_FLOW_OK;
2324 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2325 ret = gst_wavparse_stream_data (wav);
2332 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2334 GstWavParse *wav = GST_WAVPARSE (parent);
2335 gboolean ret = TRUE;
2337 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2339 switch (GST_EVENT_TYPE (event)) {
2340 case GST_EVENT_CAPS:
2342 /* discard, we'll come up with proper src caps */
2343 gst_event_unref (event);
2346 case GST_EVENT_SEGMENT:
2348 gint64 start, stop, offset = 0, end_offset = -1;
2351 /* some debug output */
2352 gst_event_copy_segment (event, &segment);
2353 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2356 if (wav->state != GST_WAVPARSE_DATA) {
2357 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2361 /* now we are either committed to TIME or BYTE format,
2362 * and we only expect a BYTE segment, e.g. following a seek */
2363 if (segment.format == GST_FORMAT_BYTES) {
2364 /* handle (un)signed issues */
2365 start = segment.start;
2366 stop = segment.stop;
2369 start -= wav->datastart;
2370 start = MAX (start, 0);
2374 stop -= wav->datastart;
2375 stop = MAX (stop, 0);
2377 if (wav->segment.format == GST_FORMAT_TIME) {
2378 guint64 bps = wav->bps;
2380 /* operating in format TIME, so we can convert */
2381 if (!bps && wav->fact)
2383 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2387 gst_util_uint64_scale_ceil (start, GST_SECOND,
2388 (guint64) wav->bps);
2391 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2392 (guint64) wav->bps);
2396 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2400 segment.start = start;
2401 segment.stop = stop;
2403 /* accept upstream's notion of segment and distribute along */
2404 segment.format = wav->segment.format;
2405 segment.time = segment.position = segment.start;
2406 segment.duration = wav->segment.duration;
2407 segment.base = gst_segment_to_running_time (&wav->segment,
2408 GST_FORMAT_TIME, wav->segment.position);
2410 gst_segment_copy_into (&segment, &wav->segment);
2412 /* also store the newsegment event for the streaming thread */
2413 if (wav->start_segment)
2414 gst_event_unref (wav->start_segment);
2415 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2416 wav->start_segment = gst_event_new_segment (&segment);
2418 /* stream leftover data in current segment */
2419 gst_wavparse_flush_data (wav);
2420 /* and set up streaming thread for next one */
2421 wav->offset = offset;
2422 wav->end_offset = end_offset;
2423 if (wav->end_offset != -1) {
2424 wav->dataleft = wav->end_offset - wav->offset;
2426 /* infinity; upstream will EOS when done */
2427 wav->dataleft = G_MAXUINT64;
2430 gst_event_unref (event);
2434 if (wav->state == GST_WAVPARSE_START) {
2435 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2436 ("No valid input found before end of stream"));
2438 /* add pad if needed so EOS is seen downstream */
2439 if (G_UNLIKELY (wav->first)) {
2441 gst_wavparse_add_src_pad (wav, NULL);
2443 /* stream leftover data in current segment */
2444 gst_wavparse_flush_data (wav);
2449 case GST_EVENT_FLUSH_STOP:
2453 gst_adapter_clear (wav->adapter);
2454 wav->discont = TRUE;
2455 dur = wav->segment.duration;
2456 gst_segment_init (&wav->segment, wav->segment.format);
2457 wav->segment.duration = dur;
2461 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2469 /* convert and query stuff */
2470 static const GstFormat *
2471 gst_wavparse_get_formats (GstPad * pad)
2473 static const GstFormat formats[] = {
2476 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2485 gst_wavparse_pad_convert (GstPad * pad,
2486 GstFormat src_format, gint64 src_value,
2487 GstFormat * dest_format, gint64 * dest_value)
2489 GstWavParse *wavparse;
2490 gboolean res = TRUE;
2492 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2494 if (*dest_format == src_format) {
2495 *dest_value = src_value;
2499 if ((wavparse->bps == 0) && !wavparse->fact)
2502 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2503 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2505 switch (src_format) {
2506 case GST_FORMAT_BYTES:
2507 switch (*dest_format) {
2508 case GST_FORMAT_DEFAULT:
2509 *dest_value = src_value / wavparse->bytes_per_sample;
2510 /* make sure we end up on a sample boundary */
2511 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2513 case GST_FORMAT_TIME:
2514 /* src_value + datastart = offset */
2515 GST_INFO_OBJECT (wavparse,
2516 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2518 if (wavparse->bps > 0)
2519 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2520 (guint64) wavparse->bps);
2521 else if (wavparse->fact) {
2522 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2523 wavparse->rate, wavparse->fact);
2526 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2537 case GST_FORMAT_DEFAULT:
2538 switch (*dest_format) {
2539 case GST_FORMAT_BYTES:
2540 *dest_value = src_value * wavparse->bytes_per_sample;
2542 case GST_FORMAT_TIME:
2543 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2544 (guint64) wavparse->rate);
2552 case GST_FORMAT_TIME:
2553 switch (*dest_format) {
2554 case GST_FORMAT_BYTES:
2555 if (wavparse->bps > 0)
2556 *dest_value = gst_util_uint64_scale (src_value,
2557 (guint64) wavparse->bps, GST_SECOND);
2559 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2560 wavparse->rate, wavparse->fact);
2562 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2564 /* make sure we end up on a sample boundary */
2565 *dest_value -= *dest_value % wavparse->blockalign;
2567 case GST_FORMAT_DEFAULT:
2568 *dest_value = gst_util_uint64_scale (src_value,
2569 (guint64) wavparse->rate, GST_SECOND);
2588 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2594 /* handle queries for location and length in requested format */
2596 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2598 gboolean res = TRUE;
2599 GstWavParse *wav = GST_WAVPARSE (parent);
2601 /* only if we know */
2602 if (wav->state != GST_WAVPARSE_DATA) {
2606 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2608 switch (GST_QUERY_TYPE (query)) {
2609 case GST_QUERY_POSITION:
2615 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2616 curb = wav->offset - wav->datastart;
2617 gst_query_parse_position (query, &format, NULL);
2618 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2621 case GST_FORMAT_BYTES:
2622 format = GST_FORMAT_BYTES;
2626 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2631 gst_query_set_position (query, format, cur);
2634 case GST_QUERY_DURATION:
2636 gint64 duration = 0;
2639 if (wav->ignore_length) {
2644 gst_query_parse_duration (query, &format, NULL);
2647 case GST_FORMAT_BYTES:{
2648 format = GST_FORMAT_BYTES;
2649 duration = wav->datasize;
2652 case GST_FORMAT_TIME:
2653 if ((res = gst_wavparse_calculate_duration (wav))) {
2654 duration = wav->duration;
2662 gst_query_set_duration (query, format, duration);
2665 case GST_QUERY_CONVERT:
2667 gint64 srcvalue, dstvalue;
2668 GstFormat srcformat, dstformat;
2670 gst_query_parse_convert (query, &srcformat, &srcvalue,
2671 &dstformat, &dstvalue);
2672 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2673 &dstformat, &dstvalue);
2675 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2678 case GST_QUERY_SEEKING:{
2680 gboolean seekable = FALSE;
2682 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2683 if (fmt == wav->segment.format) {
2684 if (wav->streaming) {
2687 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2688 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2689 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2690 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2692 gst_query_unref (q);
2694 GST_LOG_OBJECT (wav, "looping => seekable");
2698 } else if (fmt == GST_FORMAT_TIME) {
2702 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2707 res = gst_pad_query_default (pad, parent, query);
2714 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2716 GstWavParse *wavparse = GST_WAVPARSE (parent);
2717 gboolean res = FALSE;
2719 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2721 switch (GST_EVENT_TYPE (event)) {
2722 case GST_EVENT_SEEK:
2723 /* can only handle events when we are in the data state */
2724 if (wavparse->state == GST_WAVPARSE_DATA) {
2725 res = gst_wavparse_perform_seek (wavparse, event);
2727 gst_event_unref (event);
2730 case GST_EVENT_TOC_SELECT:
2733 GstTocEntry *entry = NULL;
2734 GstEvent *seek_event;
2737 if (!wavparse->toc) {
2738 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2741 gst_event_parse_toc_select (event, &uid);
2743 GST_OBJECT_LOCK (wavparse);
2744 entry = gst_toc_find_entry (wavparse->toc, uid);
2745 if (entry == NULL) {
2746 GST_OBJECT_UNLOCK (wavparse);
2747 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2751 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2752 GST_OBJECT_UNLOCK (wavparse);
2753 seek_event = gst_event_new_seek (1.0,
2755 GST_SEEK_FLAG_FLUSH,
2756 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2757 res = gst_wavparse_perform_seek (wavparse, seek_event);
2758 gst_event_unref (seek_event);
2762 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2766 gst_event_unref (event);
2771 res = gst_pad_push_event (wavparse->sinkpad, event);
2778 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2780 GstWavParse *wav = GST_WAVPARSE (parent);
2785 gst_adapter_clear (wav->adapter);
2786 g_object_unref (wav->adapter);
2787 wav->adapter = NULL;
2790 query = gst_query_new_scheduling ();
2792 if (!gst_pad_peer_query (sinkpad, query)) {
2793 gst_query_unref (query);
2797 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2798 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2799 gst_query_unref (query);
2804 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2805 wav->streaming = FALSE;
2806 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2810 GST_DEBUG_OBJECT (sinkpad, "activating push");
2811 wav->streaming = TRUE;
2812 wav->adapter = gst_adapter_new ();
2813 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2819 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2820 GstPadMode mode, gboolean active)
2825 case GST_PAD_MODE_PUSH:
2828 case GST_PAD_MODE_PULL:
2830 /* if we have a scheduler we can start the task */
2831 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2834 res = gst_pad_stop_task (sinkpad);
2844 static GstStateChangeReturn
2845 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2847 GstStateChangeReturn ret;
2848 GstWavParse *wav = GST_WAVPARSE (element);
2850 switch (transition) {
2851 case GST_STATE_CHANGE_NULL_TO_READY:
2853 case GST_STATE_CHANGE_READY_TO_PAUSED:
2854 gst_wavparse_reset (wav);
2856 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2862 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2864 switch (transition) {
2865 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2867 case GST_STATE_CHANGE_PAUSED_TO_READY:
2868 gst_wavparse_reset (wav);
2870 case GST_STATE_CHANGE_READY_TO_NULL:
2879 gst_wavparse_set_property (GObject * object, guint prop_id,
2880 const GValue * value, GParamSpec * pspec)
2884 g_return_if_fail (GST_IS_WAVPARSE (object));
2885 self = GST_WAVPARSE (object);
2888 case PROP_IGNORE_LENGTH:
2889 self->ignore_length = g_value_get_boolean (value);
2892 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2898 gst_wavparse_get_property (GObject * object, guint prop_id,
2899 GValue * value, GParamSpec * pspec)
2903 g_return_if_fail (GST_IS_WAVPARSE (object));
2904 self = GST_WAVPARSE (object);
2907 case PROP_IGNORE_LENGTH:
2908 g_value_set_boolean (value, self->ignore_length);
2911 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2916 plugin_init (GstPlugin * plugin)
2920 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2924 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2927 "Parse a .wav file into raw audio",
2928 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)