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
26 * Parse a .wav file into raw or compressed audio.
28 * Wavparse supports both push and pull mode operations, making it possible to
29 * stream from a network source.
31 * ## Example launch line
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/pbutils/descriptions.h>
58 #include <glib/gi18n-lib.h>
60 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
61 #define GST_CAT_DEFAULT (wavparse_debug)
63 /* Data size chunk of RF64,
64 * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
65 #define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
67 static void gst_wavparse_dispose (GObject * object);
69 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
71 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
72 GstObject * parent, GstPadMode mode, gboolean active);
73 static gboolean gst_wavparse_send_event (GstElement * element,
75 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
76 GstStateChange transition);
78 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
80 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
81 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
83 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
85 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
87 static void gst_wavparse_loop (GstPad * pad);
88 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
91 static void gst_wavparse_set_property (GObject * object, guint prop_id,
92 const GValue * value, GParamSpec * pspec);
93 static void gst_wavparse_get_property (GObject * object, guint prop_id,
94 GValue * value, GParamSpec * pspec);
96 #define DEFAULT_IGNORE_LENGTH FALSE
104 static GstStaticPadTemplate sink_template_factory =
105 GST_STATIC_PAD_TEMPLATE ("sink",
108 GST_STATIC_CAPS ("audio/x-wav;audio/x-rf64")
112 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
114 #define gst_wavparse_parent_class parent_class
115 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
118 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (wavparse, "wavparse", GST_RANK_PRIMARY,
119 GST_TYPE_WAVPARSE, gst_riff_init ();
124 /* Offset Size Description Value
125 * 0x00 4 ID unique identification value
126 * 0x04 4 Position play order position
127 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
128 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
129 * 0x10 4 Block Start Byte Offset to sample of First Channel
130 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
134 guint32 data_chunk_id;
137 guint32 sample_offset;
142 /* Offset Size Description Value
143 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
146 guint32 cue_point_id;
148 } GstWavParseLabl, GstWavParseNote;
151 gst_wavparse_class_init (GstWavParseClass * klass)
153 GstElementClass *gstelement_class;
154 GObjectClass *object_class;
155 GstPadTemplate *src_template;
157 gstelement_class = (GstElementClass *) klass;
158 object_class = (GObjectClass *) klass;
160 parent_class = g_type_class_peek_parent (klass);
162 object_class->dispose = gst_wavparse_dispose;
164 object_class->set_property = gst_wavparse_set_property;
165 object_class->get_property = gst_wavparse_get_property;
168 * GstWavParse:ignore-length:
170 * This selects whether the length found in a data chunk
171 * should be ignored. This may be useful for streamed audio
172 * where the length is unknown until the end of streaming,
173 * and various software/hardware just puts some random value
174 * in there and hopes it doesn't break too much.
176 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
177 g_param_spec_boolean ("ignore-length",
179 "Ignore length from the Wave header",
180 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
183 gstelement_class->change_state = gst_wavparse_change_state;
184 gstelement_class->send_event = gst_wavparse_send_event;
187 gst_element_class_add_static_pad_template (gstelement_class,
188 &sink_template_factory);
190 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
191 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
192 gst_element_class_add_pad_template (gstelement_class, src_template);
194 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
195 "Codec/Demuxer/Audio",
196 "Parse a .wav file into raw audio",
197 "Erik Walthinsen <omega@cse.ogi.edu>");
201 gst_wavparse_notes_free (GstWavParseNote * note)
209 gst_wavparse_labls_free (GstWavParseLabl * labl)
217 gst_wavparse_reset (GstWavParse * wav)
219 wav->state = GST_WAVPARSE_START;
221 /* These will all be set correctly in the fmt chunk */
236 wav->got_fmt = FALSE;
240 gst_event_unref (wav->seek_event);
241 wav->seek_event = NULL;
243 gst_adapter_clear (wav->adapter);
244 g_object_unref (wav->adapter);
248 gst_tag_list_unref (wav->tags);
251 gst_toc_unref (wav->toc);
254 g_list_free_full (wav->cues, g_free);
257 g_list_free_full (wav->labls, (GDestroyNotify) gst_wavparse_labls_free);
260 g_list_free_full (wav->notes, (GDestroyNotify) gst_wavparse_notes_free);
263 gst_caps_unref (wav->caps);
265 if (wav->start_segment)
266 gst_event_unref (wav->start_segment);
267 wav->start_segment = NULL;
271 gst_wavparse_dispose (GObject * object)
273 GstWavParse *wav = GST_WAVPARSE (object);
275 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
276 gst_wavparse_reset (wav);
278 G_OBJECT_CLASS (parent_class)->dispose (object);
282 gst_wavparse_init (GstWavParse * wavparse)
284 gst_wavparse_reset (wavparse);
288 gst_pad_new_from_static_template (&sink_template_factory, "sink");
289 gst_pad_set_activate_function (wavparse->sinkpad,
290 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
291 gst_pad_set_activatemode_function (wavparse->sinkpad,
292 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
293 gst_pad_set_chain_function (wavparse->sinkpad,
294 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
295 gst_pad_set_event_function (wavparse->sinkpad,
296 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
297 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
301 gst_pad_new_from_template (gst_element_class_get_pad_template
302 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
303 gst_pad_use_fixed_caps (wavparse->srcpad);
304 gst_pad_set_query_function (wavparse->srcpad,
305 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
306 gst_pad_set_event_function (wavparse->srcpad,
307 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
308 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
312 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
316 if (!gst_riff_parse_file_header (element, buf, &doctype))
319 if (doctype != GST_RIFF_RIFF_WAVE)
327 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
328 ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
334 gst_wavparse_stream_init (GstWavParse * wav)
337 GstBuffer *buf = NULL;
339 if ((res = gst_pad_pull_range (wav->sinkpad,
340 wav->offset, 12, &buf)) != GST_FLOW_OK)
342 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
343 return GST_FLOW_ERROR;
351 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
353 /* -1 always maps to -1 */
359 /* 0 always maps to 0 */
366 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
368 } else if (wav->fact) {
369 guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
370 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
377 /* This function is used to perform seeks on the element.
379 * It also works when event is NULL, in which case it will just
380 * start from the last configured segment. This technique is
381 * used when activating the element and to perform the seek in
385 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
389 GstFormat format, bformat;
391 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
392 gint64 cur, stop, upstream_size;
395 GstSegment seeksegment = { 0, };
397 guint32 seqnum = GST_SEQNUM_INVALID;
400 GST_DEBUG_OBJECT (wav, "doing seek with event");
402 gst_event_parse_seek (event, &rate, &format, &flags,
403 &cur_type, &cur, &stop_type, &stop);
404 seqnum = gst_event_get_seqnum (event);
406 /* no negative rates yet */
410 if (format != wav->segment.format) {
411 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
412 gst_format_get_name (format),
413 gst_format_get_name (wav->segment.format));
415 if (cur_type != GST_SEEK_TYPE_NONE)
417 gst_pad_query_convert (wav->srcpad, format, cur,
418 wav->segment.format, &cur);
419 if (res && stop_type != GST_SEEK_TYPE_NONE)
421 gst_pad_query_convert (wav->srcpad, format, stop,
422 wav->segment.format, &stop);
426 format = wav->segment.format;
429 GST_DEBUG_OBJECT (wav, "doing seek without event");
432 cur_type = GST_SEEK_TYPE_SET;
433 stop_type = GST_SEEK_TYPE_SET;
436 /* in push mode, we must delegate to upstream */
437 if (wav->streaming) {
438 gboolean res = FALSE;
440 /* if streaming not yet started; only prepare initial newsegment */
441 if (!event || wav->state != GST_WAVPARSE_DATA) {
442 if (wav->start_segment)
443 gst_event_unref (wav->start_segment);
444 wav->start_segment = gst_event_new_segment (&wav->segment);
447 /* convert seek positions to byte positions in data sections */
448 if (format == GST_FORMAT_TIME) {
449 /* should not fail */
450 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
452 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
455 /* mind sample boundary and header */
457 cur -= (cur % wav->bytes_per_sample);
458 cur += wav->datastart;
461 stop -= (stop % wav->bytes_per_sample);
462 stop += wav->datastart;
464 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
465 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
467 /* BYTE seek event */
468 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
470 if (seqnum != GST_SEQNUM_INVALID)
471 gst_event_set_seqnum (event, seqnum);
472 res = gst_pad_push_event (wav->sinkpad, event);
478 flush = flags & GST_SEEK_FLAG_FLUSH;
480 /* now we need to make sure the streaming thread is stopped. We do this by
481 * either sending a FLUSH_START event downstream which will cause the
482 * streaming thread to stop with a WRONG_STATE.
483 * For a non-flushing seek we simply pause the task, which will happen as soon
484 * as it completes one iteration (and thus might block when the sink is
485 * blocking in preroll). */
488 GST_DEBUG_OBJECT (wav, "sending flush start");
490 fevent = gst_event_new_flush_start ();
491 if (seqnum != GST_SEQNUM_INVALID)
492 gst_event_set_seqnum (fevent, seqnum);
493 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
494 gst_pad_push_event (wav->srcpad, fevent);
496 gst_pad_pause_task (wav->sinkpad);
499 /* we should now be able to grab the streaming thread because we stopped it
500 * with the above flush/pause code */
501 GST_PAD_STREAM_LOCK (wav->sinkpad);
503 /* save current position */
504 last_stop = wav->segment.position;
506 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
508 /* copy segment, we need this because we still need the old
509 * segment when we close the current segment. */
510 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
512 /* configure the seek parameters in the seeksegment. We will then have the
513 * right values in the segment to perform the seek */
515 GST_DEBUG_OBJECT (wav, "configuring seek");
516 gst_segment_do_seek (&seeksegment, rate, format, flags,
517 cur_type, cur, stop_type, stop, &update);
520 /* figure out the last position we need to play. If it's configured (stop !=
521 * -1), use that, else we play until the total duration of the file */
522 if ((stop = seeksegment.stop) == -1)
523 stop = seeksegment.duration;
525 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
526 if ((cur_type != GST_SEEK_TYPE_NONE)) {
527 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
528 * we can just copy the last_stop. If not, we use the bps to convert TIME to
530 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
531 (gint64 *) & wav->offset))
532 wav->offset = seeksegment.position;
533 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
534 wav->offset -= (wav->offset % wav->bytes_per_sample);
535 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
536 wav->offset += wav->datastart;
537 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
539 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
543 if (stop_type != GST_SEEK_TYPE_NONE) {
544 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
545 wav->end_offset = stop;
546 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
547 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
548 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
549 wav->end_offset += wav->datastart;
550 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
552 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
556 /* make sure filesize is not exceeded due to rounding errors or so,
557 * same precaution as in _stream_headers */
558 bformat = GST_FORMAT_BYTES;
559 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
560 wav->end_offset = MIN (wav->end_offset, upstream_size);
562 if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
563 wav->end_offset = wav->datastart + wav->datasize;
565 /* this is the range of bytes we will use for playback */
566 wav->offset = MIN (wav->offset, wav->end_offset);
567 wav->dataleft = wav->end_offset - wav->offset;
569 GST_DEBUG_OBJECT (wav,
570 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
571 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
572 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
574 /* prepare for streaming again */
578 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
579 GST_DEBUG_OBJECT (wav, "sending flush stop");
581 fevent = gst_event_new_flush_stop (TRUE);
582 if (seqnum != GST_SEQNUM_INVALID)
583 gst_event_set_seqnum (fevent, seqnum);
584 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
585 gst_pad_push_event (wav->srcpad, fevent);
588 /* now we did the seek and can activate the new segment values */
589 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
591 /* if we're doing a segment seek, post a SEGMENT_START message */
592 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
593 gst_element_post_message (GST_ELEMENT_CAST (wav),
594 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
595 wav->segment.format, wav->segment.position));
598 /* now create the newsegment */
599 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
600 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
602 /* store the newsegment event so it can be sent from the streaming thread. */
603 if (wav->start_segment)
604 gst_event_unref (wav->start_segment);
605 wav->start_segment = gst_event_new_segment (&wav->segment);
606 if (seqnum != GST_SEQNUM_INVALID)
607 gst_event_set_seqnum (wav->start_segment, seqnum);
609 /* mark discont if we are going to stream from another position. */
610 if (last_stop != wav->segment.position) {
611 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
615 /* and start the streaming task again */
616 if (!wav->streaming) {
617 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
621 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
628 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
633 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
638 GST_DEBUG_OBJECT (wav,
639 "Could not determine byte position for desired time");
645 * gst_wavparse_peek_chunk_info:
646 * @wav Wavparse object
647 * @tag holder for tag
648 * @size holder for tag size
650 * Peek next chunk info (tag and size)
652 * Returns: %TRUE when the chunk info (header) is available
655 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
657 const guint8 *data = NULL;
659 if (gst_adapter_available (wav->adapter) < 8)
662 data = gst_adapter_map (wav->adapter, 8);
663 *tag = GST_READ_UINT32_LE (data);
664 *size = GST_READ_UINT32_LE (data + 4);
665 gst_adapter_unmap (wav->adapter);
667 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
668 GST_FOURCC_ARGS (*tag));
674 * gst_wavparse_peek_chunk:
675 * @wav Wavparse object
676 * @tag holder for tag
677 * @size holder for tag size
679 * Peek enough data for one full chunk
681 * Returns: %TRUE when the full chunk is available
684 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
686 guint32 peek_size = 0;
689 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
692 /* size 0 -> empty data buffer would surprise most callers,
693 * large size -> do not bother trying to squeeze that into adapter,
694 * so we throw poor man's exception, which can be caught if caller really
695 * wants to handle 0 size chunk */
696 if (!(*size) || (*size) >= (1 << 30)) {
697 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
698 *size, GST_FOURCC_ARGS (*tag));
699 /* chain should give up */
700 wav->abort_buffering = TRUE;
703 peek_size = (*size + 1) & ~1;
704 available = gst_adapter_available (wav->adapter);
706 if (available >= (8 + peek_size)) {
709 GST_LOG ("but only %u bytes available now", available);
715 * gst_wavparse_calculate_duration:
716 * @wav: wavparse object
718 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
721 * Returns: %TRUE if duration is available.
724 gst_wavparse_calculate_duration (GstWavParse * wav)
726 if (wav->duration > 0)
730 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
732 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
734 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
735 GST_TIME_ARGS (wav->duration));
737 } else if (wav->fact) {
739 gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
740 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
741 GST_TIME_ARGS (wav->duration));
748 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
753 if (wav->streaming) {
754 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
757 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
758 GST_FOURCC_ARGS (tag));
759 flush = 8 + ((size + 1) & ~1);
760 wav->offset += flush;
761 if (wav->streaming) {
762 gst_adapter_flush (wav->adapter, flush);
764 gst_buffer_unref (buf);
771 * gst_wavparse_cue_chunk:
772 * @wav GstWavParse object
773 * @data holder for data
774 * @size holder for data size
776 * Parse cue chunk from @data to wav->cues.
778 * Returns: %TRUE when cue chunk is available
781 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
788 GST_WARNING_OBJECT (wav, "found another cue's");
792 ncues = GST_READ_UINT32_LE (data);
794 if (size < 4 + ncues * 24) {
795 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
801 for (i = 0; i < ncues; i++) {
802 cue = g_new0 (GstWavParseCue, 1);
803 cue->id = GST_READ_UINT32_LE (data);
804 cue->position = GST_READ_UINT32_LE (data + 4);
805 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
806 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
807 cue->block_start = GST_READ_UINT32_LE (data + 16);
808 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
809 cues = g_list_append (cues, cue);
819 * gst_wavparse_labl_chunk:
820 * @wav GstWavParse object
821 * @data holder for data
822 * @size holder for data size
824 * Parse labl from @data to wav->labls.
826 * Returns: %TRUE when labl chunk is available
829 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
831 GstWavParseLabl *labl;
836 labl = g_new0 (GstWavParseLabl, 1);
839 labl->cue_point_id = GST_READ_UINT32_LE (data);
840 labl->text = g_strndup ((const gchar *) data + 4, size - 4);
842 wav->labls = g_list_append (wav->labls, labl);
848 * gst_wavparse_note_chunk:
849 * @wav GstWavParse object
850 * @data holder for data
851 * @size holder for data size
853 * Parse note from @data to wav->notes.
855 * Returns: %TRUE when note chunk is available
858 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
860 GstWavParseNote *note;
865 note = g_new0 (GstWavParseNote, 1);
868 note->cue_point_id = GST_READ_UINT32_LE (data);
869 note->text = g_strndup ((const gchar *) data + 4, size - 4);
871 wav->notes = g_list_append (wav->notes, note);
877 * gst_wavparse_smpl_chunk:
878 * @wav GstWavParse object
879 * @data holder for data
880 * @size holder for data size
882 * Parse smpl chunk from @data.
884 * Returns: %TRUE when cue chunk is available
887 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
892 manufacturer_id = GST_READ_UINT32_LE (data);
893 product_id = GST_READ_UINT32_LE (data + 4);
894 sample_period = GST_READ_UINT32_LE (data + 8);
896 note_number = GST_READ_UINT32_LE (data + 12);
898 pitch_fraction = GST_READ_UINT32_LE (data + 16);
899 SMPTE_format = GST_READ_UINT32_LE (data + 20);
900 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
901 num_sample_loops = GST_READ_UINT32_LE (data + 28);
902 List of Sample Loops, 24 bytes each
906 wav->tags = gst_tag_list_new_empty ();
907 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
908 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
913 * gst_wavparse_adtl_chunk:
914 * @wav GstWavParse object
915 * @data holder for data
916 * @size holder for data size
918 * Parse adtl from @data.
920 * Returns: %TRUE when adtl chunk is available
923 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
925 guint32 ltag, lsize, offset = 0;
928 ltag = GST_READ_UINT32_LE (data + offset);
929 lsize = GST_READ_UINT32_LE (data + offset + 4);
931 if (lsize > (G_MAXUINT - 8) || lsize + 8 > size) {
932 GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
937 case GST_RIFF_TAG_labl:
938 gst_wavparse_labl_chunk (wav, data + offset + 8, lsize);
940 case GST_RIFF_TAG_note:
941 gst_wavparse_note_chunk (wav, data + offset + 8, lsize);
944 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
945 GST_FOURCC_ARGS (ltag));
946 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
949 offset += 8 + GST_ROUND_UP_2 (lsize);
950 size -= 8 + GST_ROUND_UP_2 (lsize);
957 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
959 GstTagList *tags = NULL;
960 GstTocEntry *entry = NULL;
962 entry = gst_toc_find_entry (toc, id);
964 tags = gst_toc_entry_get_tags (entry);
966 tags = gst_tag_list_new_empty ();
967 gst_toc_entry_set_tags (entry, tags);
975 * gst_wavparse_create_toc:
976 * @wav GstWavParse object
978 * Create TOC from wav->cues and wav->labls.
981 gst_wavparse_create_toc (GstWavParse * wav)
987 GstWavParseLabl *labl;
988 GstWavParseNote *note;
991 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
993 GST_OBJECT_LOCK (wav);
995 GST_OBJECT_UNLOCK (wav);
996 GST_WARNING_OBJECT (wav, "found another TOC");
1001 GST_OBJECT_UNLOCK (wav);
1005 /* FIXME: send CURRENT scope toc too */
1006 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
1008 /* add cue edition */
1009 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
1010 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
1011 gst_toc_append_entry (toc, entry);
1013 /* add tracks in cue edition */
1017 prev_subentry = cur_subentry;
1018 /* previous track stop time = current track start time */
1019 if (prev_subentry != NULL) {
1020 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1021 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1022 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1024 id = g_strdup_printf ("%08x", cue->id);
1025 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1027 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1028 stop = wav->duration;
1029 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1030 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1031 list = g_list_next (list);
1034 /* add tags in tracks */
1038 id = g_strdup_printf ("%08x", labl->cue_point_id);
1039 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1042 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1045 list = g_list_next (list);
1050 id = g_strdup_printf ("%08x", note->cue_point_id);
1051 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1054 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1057 list = g_list_next (list);
1060 /* send data as TOC */
1063 /* send TOC event */
1065 GST_OBJECT_UNLOCK (wav);
1066 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1072 #define MAX_BUFFER_SIZE 4096
1075 parse_ds64 (GstWavParse * wav, GstBuffer * buf)
1078 guint32 dataSizeLow, dataSizeHigh;
1079 guint32 sampleCountLow, sampleCountHigh;
1081 gst_buffer_map (buf, &map, GST_MAP_READ);
1082 dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
1083 dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
1084 sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
1085 sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
1086 gst_buffer_unmap (buf, &map);
1087 if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
1088 wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
1090 if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
1091 wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
1094 GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
1095 " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
1099 static GstFlowReturn
1100 gst_wavparse_stream_headers (GstWavParse * wav)
1102 GstFlowReturn res = GST_FLOW_OK;
1103 GstBuffer *buf = NULL;
1104 gst_riff_strf_auds *header = NULL;
1106 gboolean gotdata = FALSE;
1107 GstCaps *caps = NULL;
1108 gchar *codec_name = NULL;
1109 gint64 upstream_size = 0;
1112 /* search for "_fmt" chunk, which must be before "data" */
1113 while (!wav->got_fmt) {
1116 if (wav->streaming) {
1117 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1120 gst_adapter_flush (wav->adapter, 8);
1124 buf = gst_adapter_take_buffer (wav->adapter, size);
1126 gst_adapter_flush (wav->adapter, 1);
1127 wav->offset += GST_ROUND_UP_2 (size);
1129 buf = gst_buffer_new ();
1132 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1133 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1137 if (tag == GST_RS64_TAG_DS64) {
1138 if (!parse_ds64 (wav, buf))
1144 if (tag != GST_RIFF_TAG_fmt) {
1145 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1146 GST_FOURCC_ARGS (tag));
1147 gst_buffer_unref (buf);
1152 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1154 goto parse_header_error;
1156 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1158 /* do sanity checks of header fields */
1159 if (header->channels == 0)
1161 if (header->rate == 0)
1164 GST_DEBUG_OBJECT (wav, "creating the caps");
1166 /* Note: gst_riff_create_audio_caps might need to fix values in
1167 * the header header depending on the format, so call it first */
1168 /* FIXME: Need to handle the channel reorder map */
1169 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1170 NULL, &codec_name, NULL);
1173 gst_buffer_unref (extra);
1176 goto unknown_format;
1178 /* If we got raw audio from upstream, we remove the codec_data field,
1179 * which may have been added if the wav header included an extended
1180 * chunk. We want to keep it for non raw audio.
1182 s = gst_caps_get_structure (caps, 0);
1183 if (s && gst_structure_has_name (s, "audio/x-raw")) {
1184 gst_structure_remove_field (s, "codec_data");
1187 /* do more sanity checks of header fields
1188 * (these can be sanitized by gst_riff_create_audio_caps()
1190 wav->format = header->format;
1191 wav->rate = header->rate;
1192 wav->channels = header->channels;
1193 wav->blockalign = header->blockalign;
1194 wav->depth = header->bits_per_sample;
1195 wav->av_bps = header->av_bps;
1201 /* do format specific handling */
1202 switch (wav->format) {
1203 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1204 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1206 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1207 * bitrate inside the mpeg stream */
1208 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1212 case GST_RIFF_WAVE_FORMAT_PCM:
1213 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1214 goto invalid_blockalign;
1217 if (wav->av_bps > wav->blockalign * wav->rate)
1219 /* use the configured bps */
1220 wav->bps = wav->av_bps;
1224 wav->width = (wav->blockalign * 8) / wav->channels;
1225 wav->bytes_per_sample = wav->channels * wav->width / 8;
1227 if (wav->bytes_per_sample <= 0)
1228 goto no_bytes_per_sample;
1230 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1231 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1232 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1233 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1234 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1235 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1236 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1238 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1239 * formats). This will make the element output a BYTE format segment and
1240 * will not timestamp the outgoing buffers.
1242 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1244 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1246 /* create pad later so we can sniff the first few bytes
1247 * of the real data and correct our caps if necessary */
1248 gst_caps_replace (&wav->caps, caps);
1249 gst_caps_replace (&caps, NULL);
1251 wav->got_fmt = TRUE;
1253 if (wav->tags == NULL)
1254 wav->tags = gst_tag_list_new_empty ();
1257 GstCaps *templ_caps = gst_pad_get_pad_template_caps (wav->sinkpad);
1258 gst_pb_utils_add_codec_description_to_tag_list (wav->tags,
1259 GST_TAG_CONTAINER_FORMAT, templ_caps);
1260 gst_caps_unref (templ_caps);
1263 /* If bps is nonzero, then we do have a valid bitrate that can be
1264 * announced in a tag list. */
1266 guint bitrate = wav->bps * 8;
1267 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1268 GST_TAG_BITRATE, bitrate, NULL);
1272 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1273 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1275 g_free (codec_name);
1281 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1282 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1284 /* loop headers until we get data */
1286 if (wav->streaming) {
1287 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1294 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1295 &buf)) != GST_FLOW_OK)
1296 goto header_read_error;
1297 gst_buffer_map (buf, &map, GST_MAP_READ);
1298 tag = GST_READ_UINT32_LE (map.data);
1299 size = GST_READ_UINT32_LE (map.data + 4);
1300 gst_buffer_unmap (buf, &map);
1303 GST_INFO_OBJECT (wav,
1304 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
1305 G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
1307 /* Maximum valid size is INT_MAX */
1308 if (size & 0x80000000) {
1309 GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
1313 /* Clip to upstream size if known */
1314 if (upstream_size > 0 && size + wav->offset > upstream_size) {
1315 GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
1316 g_assert (upstream_size >= wav->offset);
1317 size = upstream_size - wav->offset;
1320 /* wav is a st00pid format, we don't know for sure where data starts.
1321 * So we have to go bit by bit until we find the 'data' header
1324 case GST_RIFF_TAG_data:{
1327 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1329 if (wav->ignore_length) {
1330 GST_DEBUG_OBJECT (wav, "Ignoring length");
1333 if (wav->streaming) {
1334 gst_adapter_flush (wav->adapter, 8);
1337 gst_buffer_unref (buf);
1340 wav->datastart = wav->offset;
1341 /* use size from ds64 chunk if available */
1342 if (size64 == -1 && wav->datasize > 0) {
1343 GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
1344 size64 = wav->datasize;
1346 wav->chunk_size = size64;
1348 /* If size is zero, then the data chunk probably actually extends to
1349 the end of the file */
1350 if (size64 == 0 && upstream_size) {
1351 size64 = upstream_size - wav->datastart;
1353 /* Or the file might be truncated */
1354 else if (upstream_size) {
1355 size64 = MIN (size64, (upstream_size - wav->datastart));
1357 wav->datasize = size64;
1358 wav->dataleft = size64;
1359 wav->end_offset = size64 + wav->datastart;
1360 if (!wav->streaming) {
1361 /* We will continue parsing tags 'till end */
1362 wav->offset += size64;
1364 GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
1367 case GST_RIFF_TAG_fact:{
1368 if (wav->fact == 0 &&
1369 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1370 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1371 const guint data_size = 4;
1373 GST_INFO_OBJECT (wav, "Have fact chunk");
1374 if (size < data_size) {
1375 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1376 /* need more data */
1379 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1383 /* number of samples (for compressed formats) */
1384 if (wav->streaming) {
1385 const guint8 *data = NULL;
1387 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1390 gst_adapter_flush (wav->adapter, 8);
1391 data = gst_adapter_map (wav->adapter, data_size);
1392 wav->fact = GST_READ_UINT32_LE (data);
1393 gst_adapter_unmap (wav->adapter);
1394 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1396 gst_buffer_unref (buf);
1399 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1400 data_size, &buf)) != GST_FLOW_OK)
1401 goto header_read_error;
1402 gst_buffer_extract (buf, 0, &wav->fact, 4);
1403 wav->fact = GUINT32_FROM_LE (wav->fact);
1404 gst_buffer_unref (buf);
1406 GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
1407 wav->offset += 8 + GST_ROUND_UP_2 (size);
1410 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1411 /* need more data */
1417 case GST_RIFF_TAG_acid:{
1418 const gst_riff_acid *acid = NULL;
1419 const guint data_size = sizeof (gst_riff_acid);
1422 GST_INFO_OBJECT (wav, "Have acid chunk");
1423 if (size < data_size) {
1424 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1425 /* need more data */
1428 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1432 if (wav->streaming) {
1433 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1436 gst_adapter_flush (wav->adapter, 8);
1437 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1439 tempo = acid->tempo;
1440 gst_adapter_unmap (wav->adapter);
1443 gst_buffer_unref (buf);
1446 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1447 size, &buf)) != GST_FLOW_OK)
1448 goto header_read_error;
1449 gst_buffer_map (buf, &map, GST_MAP_READ);
1450 acid = (const gst_riff_acid *) map.data;
1451 tempo = acid->tempo;
1452 gst_buffer_unmap (buf, &map);
1454 /* send data as tags */
1456 wav->tags = gst_tag_list_new_empty ();
1457 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1458 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1460 size = GST_ROUND_UP_2 (size);
1461 if (wav->streaming) {
1462 gst_adapter_flush (wav->adapter, size);
1464 gst_buffer_unref (buf);
1466 wav->offset += 8 + size;
1469 /* FIXME: all list tags after data are ignored in streaming mode */
1470 case GST_RIFF_TAG_LIST:{
1473 if (wav->streaming) {
1474 const guint8 *data = NULL;
1476 if (gst_adapter_available (wav->adapter) < 12) {
1479 data = gst_adapter_map (wav->adapter, 12);
1480 ltag = GST_READ_UINT32_LE (data + 8);
1481 gst_adapter_unmap (wav->adapter);
1483 gst_buffer_unref (buf);
1486 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1487 &buf)) != GST_FLOW_OK)
1488 goto header_read_error;
1489 gst_buffer_extract (buf, 8, <ag, 4);
1490 ltag = GUINT32_FROM_LE (ltag);
1493 case GST_RIFF_LIST_INFO:{
1494 const gint data_size = size - 4;
1497 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1498 if (wav->streaming) {
1499 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1502 gst_adapter_flush (wav->adapter, 12);
1504 if (data_size > 0) {
1505 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1507 gst_adapter_flush (wav->adapter, 1);
1511 gst_buffer_unref (buf);
1513 if (data_size > 0) {
1515 gst_pad_pull_range (wav->sinkpad, wav->offset,
1516 data_size, &buf)) != GST_FLOW_OK)
1517 goto header_read_error;
1520 if (data_size > 0) {
1522 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1524 GstTagList *old = wav->tags;
1526 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1528 gst_tag_list_unref (old);
1529 gst_tag_list_unref (new);
1531 gst_buffer_unref (buf);
1532 wav->offset += GST_ROUND_UP_2 (data_size);
1536 case GST_RIFF_LIST_adtl:{
1537 const gint data_size = size - 4;
1539 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1540 if (wav->streaming) {
1541 const guint8 *data = NULL;
1543 gst_adapter_flush (wav->adapter, 12);
1545 data = gst_adapter_map (wav->adapter, data_size);
1546 gst_wavparse_adtl_chunk (wav, data, data_size);
1547 gst_adapter_unmap (wav->adapter);
1551 gst_buffer_unref (buf);
1555 gst_pad_pull_range (wav->sinkpad, wav->offset,
1556 data_size, &buf)) != GST_FLOW_OK)
1557 goto header_read_error;
1558 gst_buffer_map (buf, &map, GST_MAP_READ);
1559 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1561 gst_buffer_unmap (buf, &map);
1562 gst_buffer_unref (buf);
1564 wav->offset += GST_ROUND_UP_2 (data_size);
1568 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1569 GST_FOURCC_ARGS (ltag));
1570 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1571 /* need more data */
1577 case GST_RIFF_TAG_cue:{
1578 const guint data_size = size;
1580 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1581 if (wav->streaming) {
1582 const guint8 *data = NULL;
1584 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1587 gst_adapter_flush (wav->adapter, 8);
1589 data = gst_adapter_map (wav->adapter, data_size);
1590 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1591 goto header_read_error;
1593 gst_adapter_unmap (wav->adapter);
1598 gst_buffer_unref (buf);
1601 gst_pad_pull_range (wav->sinkpad, wav->offset,
1602 data_size, &buf)) != GST_FLOW_OK)
1603 goto header_read_error;
1604 gst_buffer_map (buf, &map, GST_MAP_READ);
1605 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1607 goto header_read_error;
1609 gst_buffer_unmap (buf, &map);
1611 size = GST_ROUND_UP_2 (size);
1612 if (wav->streaming) {
1613 gst_adapter_flush (wav->adapter, size);
1615 gst_buffer_unref (buf);
1617 size = GST_ROUND_UP_2 (size);
1618 wav->offset += size;
1621 case GST_RIFF_TAG_smpl:{
1622 const gint data_size = size;
1624 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1625 if (wav->streaming) {
1626 const guint8 *data = NULL;
1628 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1631 gst_adapter_flush (wav->adapter, 8);
1633 data = gst_adapter_map (wav->adapter, data_size);
1634 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1635 goto header_read_error;
1637 gst_adapter_unmap (wav->adapter);
1642 gst_buffer_unref (buf);
1645 gst_pad_pull_range (wav->sinkpad, wav->offset,
1646 data_size, &buf)) != GST_FLOW_OK)
1647 goto header_read_error;
1648 gst_buffer_map (buf, &map, GST_MAP_READ);
1649 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1651 goto header_read_error;
1653 gst_buffer_unmap (buf, &map);
1655 size = GST_ROUND_UP_2 (size);
1656 if (wav->streaming) {
1657 gst_adapter_flush (wav->adapter, size);
1659 gst_buffer_unref (buf);
1661 size = GST_ROUND_UP_2 (size);
1662 wav->offset += size;
1666 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1667 GST_FOURCC_ARGS (tag));
1668 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1669 /* need more data */
1674 if (upstream_size && (wav->offset >= upstream_size)) {
1675 /* Now we are gone through the whole file */
1680 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1682 if (wav->bps <= 0 && wav->fact) {
1684 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1686 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1687 (guint64) wav->fact);
1688 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1693 if (gst_wavparse_calculate_duration (wav)) {
1694 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1695 if (!wav->ignore_length)
1696 wav->segment.duration = wav->duration;
1698 gst_wavparse_create_toc (wav);
1700 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1701 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1702 if (!wav->ignore_length)
1703 wav->segment.duration = wav->datasize;
1706 /* now we have all the info to perform a pending seek if any, if no
1707 * event, this will still do the right thing and it will also send
1708 * the right newsegment event downstream. */
1709 gst_wavparse_perform_seek (wav, wav->seek_event);
1710 /* remove pending event */
1711 gst_event_replace (&wav->seek_event, NULL);
1713 /* we just started, we are discont */
1714 wav->discont = TRUE;
1716 wav->state = GST_WAVPARSE_DATA;
1718 /* determine reasonable max buffer size,
1719 * that is, buffers not too small either size or time wise
1720 * so we do not end up with too many of them */
1722 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1723 wav->max_buf_size = upstream_size;
1725 wav->max_buf_size = 0;
1726 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1727 if (wav->blockalign > 0)
1728 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1730 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1737 g_free (codec_name);
1740 gst_caps_unref (caps);
1745 res = GST_FLOW_ERROR;
1750 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1751 ("Couldn't parse audio header"));
1756 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1757 ("Stream claims to contain no channels - invalid data"));
1762 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1763 ("Stream with sample_rate == 0 - invalid data"));
1768 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1769 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1770 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1775 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1776 ("Stream claims av_bps = %u, which is more than %u - invalid data",
1777 wav->av_bps, wav->blockalign * wav->rate));
1780 no_bytes_per_sample:
1782 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1783 ("Could not calculate bytes per sample - invalid data"));
1788 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1789 ("No caps found for format 0x%x, %u channels, %u Hz",
1790 wav->format, wav->channels, wav->rate));
1795 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1796 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1802 * Read WAV file tag when streaming
1804 static GstFlowReturn
1805 gst_wavparse_parse_stream_init (GstWavParse * wav)
1807 if (gst_adapter_available (wav->adapter) >= 12) {
1810 /* _take flushes the data */
1811 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1813 GST_DEBUG ("Parsing wav header");
1814 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1815 return GST_FLOW_ERROR;
1818 /* Go to next state */
1819 wav->state = GST_WAVPARSE_HEADER;
1824 /* handle an event sent directly to the element.
1826 * This event can be sent either in the READY state or the
1827 * >READY state. The only event of interest really is the seek
1830 * In the READY state we can only store the event and try to
1831 * respect it when going to PAUSED. We assume we are in the
1832 * READY state when our parsing state != GST_WAVPARSE_DATA.
1834 * When we are steaming, we can simply perform the seek right
1838 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1840 GstWavParse *wav = GST_WAVPARSE (element);
1841 gboolean res = FALSE;
1843 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1845 switch (GST_EVENT_TYPE (event)) {
1846 case GST_EVENT_SEEK:
1847 if (wav->state == GST_WAVPARSE_DATA) {
1848 /* we can handle the seek directly when streaming data */
1849 res = gst_wavparse_perform_seek (wav, event);
1851 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1853 gst_event_replace (&wav->seek_event, event);
1855 /* we always return true */
1862 gst_event_unref (event);
1867 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1871 s = gst_caps_get_structure (caps, 0);
1872 if (!gst_structure_has_name (s, "audio/x-dts"))
1874 /* typefind behavior for DTS:
1875 * MAXIMUM: multiple frame syncs detected, certainly DTS
1876 * LIKELY: single frame sync at offset 0. Maybe DTS?
1877 * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
1879 if (prob > GST_TYPE_FIND_LIKELY)
1881 if (prob <= GST_TYPE_FIND_POSSIBLE)
1883 /* for maybe, check for at least a valid-looking rate and channels */
1884 if (!gst_structure_has_field (s, "channels"))
1886 /* and for extra assurance we could also check the rate from the DTS frame
1887 * against the one in the wav header, but for now let's not do that */
1888 return gst_structure_has_field (s, "rate");
1892 gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
1894 GstTagList *tags = NULL;
1899 while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
1900 gst_event_parse_tag (ev, &tags);
1901 if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
1902 tags = gst_tag_list_copy (tags);
1903 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1904 gst_event_unref (ev);
1908 gst_event_unref (ev);
1914 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1917 GstTagList *tags, *utags;
1919 GST_DEBUG_OBJECT (wav, "adding src pad");
1921 g_assert (wav->caps != NULL);
1923 s = gst_caps_get_structure (wav->caps, 0);
1924 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL
1925 && (GST_BUFFER_OFFSET (buf) == 0 || !GST_BUFFER_OFFSET_IS_VALID (buf))) {
1926 GstTypeFindProbability prob;
1927 GstCaps *tf_caps, *dts_caps;
1929 dts_caps = gst_caps_from_string ("audio/x-dts");
1931 gst_type_find_helper_for_buffer_with_caps (GST_OBJECT (wav), buf,
1933 if (tf_caps != NULL) {
1934 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1935 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1936 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1937 gst_caps_unref (wav->caps);
1938 wav->caps = tf_caps;
1940 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1941 GST_TAG_AUDIO_CODEC, "dts", NULL);
1943 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1944 "marked as raw PCM audio, but ignoring for now", tf_caps);
1945 gst_caps_unref (tf_caps);
1948 gst_caps_unref (dts_caps);
1951 gst_pad_set_caps (wav->srcpad, wav->caps);
1953 if (wav->start_segment) {
1954 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1955 gst_pad_push_event (wav->srcpad, wav->start_segment);
1956 wav->start_segment = NULL;
1959 /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
1960 * that there'll be only one scope/type of tag list from upstream, if any */
1961 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
1963 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
1965 /* if there's a tag upstream it's probably been added to override the
1966 * tags from inside the wav header, so keep upstream tags if in doubt */
1967 tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
1969 if (wav->tags != NULL) {
1970 gst_tag_list_unref (wav->tags);
1975 gst_tag_list_unref (utags);
1977 /* send tags downstream, if any */
1979 gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
1982 static GstFlowReturn
1983 gst_wavparse_stream_data (GstWavParse * wav, gboolean flushing)
1985 GstBuffer *buf = NULL;
1986 GstFlowReturn res = GST_FLOW_OK;
1987 guint64 desired, obtained;
1988 GstClockTime timestamp, next_timestamp, duration;
1989 guint64 pos, nextpos;
1992 GST_LOG_OBJECT (wav,
1993 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1994 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1996 if ((wav->dataleft == 0 || wav->dataleft < wav->blockalign)) {
1997 /* In case chunk size is not declared in the beginning get size from the
1998 * file size directly */
1999 if (wav->chunk_size == 0) {
2000 gint64 upstream_size = 0;
2002 /* Get the size of the file */
2003 if (!gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES,
2007 if (upstream_size < wav->offset + wav->datastart)
2010 /* If file has updated since the beginning continue reading the file */
2011 wav->dataleft = upstream_size - wav->offset - wav->datastart;
2012 wav->end_offset = upstream_size;
2014 /* Get the next n bytes and output them, if we can */
2015 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
2022 /* scale the amount of data by the segment rate so we get equal
2023 * amounts of data regardless of the playback rate */
2025 MIN (gst_guint64_to_gdouble (wav->dataleft),
2026 wav->max_buf_size * ABS (wav->segment.rate));
2028 if (desired >= wav->blockalign && wav->blockalign > 0)
2029 desired -= (desired % wav->blockalign);
2031 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
2032 "from the sinkpad", desired);
2034 if (wav->streaming) {
2035 guint avail = gst_adapter_available (wav->adapter);
2038 /* flush some bytes if evil upstream sends segment that starts
2039 * before data or does is not send sample aligned segment */
2040 if (G_LIKELY (wav->offset >= wav->datastart)) {
2041 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
2043 extra = wav->datastart - wav->offset;
2046 if (G_UNLIKELY (extra)) {
2047 extra = wav->bytes_per_sample - extra;
2048 if (extra <= avail) {
2049 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2050 gst_adapter_flush (wav->adapter, extra);
2051 wav->offset += extra;
2052 wav->dataleft -= extra;
2053 goto iterate_adapter;
2055 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2056 gst_adapter_clear (wav->adapter);
2057 wav->offset += avail;
2058 wav->dataleft -= avail;
2063 if (avail < desired) {
2064 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2066 /* If we are at the end of the stream, we need to flush whatever we have left */
2067 if (avail > 0 && flushing) {
2068 if (avail >= wav->blockalign && wav->blockalign > 0) {
2069 avail -= (avail % wav->blockalign);
2070 buf = gst_adapter_take_buffer (wav->adapter, avail);
2078 buf = gst_adapter_take_buffer (wav->adapter, desired);
2081 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2082 desired, &buf)) != GST_FLOW_OK)
2085 /* we may get a short buffer at the end of the file */
2086 if (gst_buffer_get_size (buf) < desired) {
2087 gsize size = gst_buffer_get_size (buf);
2089 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2090 if (size >= wav->blockalign) {
2091 if (wav->blockalign > 0) {
2092 buf = gst_buffer_make_writable (buf);
2093 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2096 gst_buffer_unref (buf);
2102 obtained = gst_buffer_get_size (buf);
2104 /* our positions in bytes */
2105 pos = wav->offset - wav->datastart;
2106 nextpos = pos + obtained;
2108 /* update offsets, does not overflow. */
2109 buf = gst_buffer_make_writable (buf);
2110 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2111 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2113 /* first chunk of data? create the source pad. We do this only here so
2114 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2115 if (G_UNLIKELY (wav->first)) {
2117 /* this will also push the segment events */
2118 gst_wavparse_add_src_pad (wav, buf);
2120 /* If we have a pending start segment, send it now. */
2121 if (G_UNLIKELY (wav->start_segment != NULL)) {
2122 gst_pad_push_event (wav->srcpad, wav->start_segment);
2123 wav->start_segment = NULL;
2128 /* and timestamps if we have a bitrate, be careful for overflows */
2130 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2132 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2133 duration = next_timestamp - timestamp;
2135 /* update current running segment position */
2136 if (G_LIKELY (next_timestamp >= wav->segment.start))
2137 wav->segment.position = next_timestamp;
2138 } else if (wav->fact) {
2140 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2141 /* and timestamps if we have a bitrate, be careful for overflows */
2142 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2143 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2144 duration = next_timestamp - timestamp;
2146 /* no bitrate, all we know is that the first sample has timestamp 0, all
2147 * other positions and durations have unknown timestamp. */
2151 timestamp = GST_CLOCK_TIME_NONE;
2152 duration = GST_CLOCK_TIME_NONE;
2153 /* update current running segment position with byte offset */
2154 if (G_LIKELY (nextpos >= wav->segment.start))
2155 wav->segment.position = nextpos;
2157 if ((pos > 0) && wav->vbr) {
2158 /* don't set timestamps for VBR files if it's not the first buffer */
2159 timestamp = GST_CLOCK_TIME_NONE;
2160 duration = GST_CLOCK_TIME_NONE;
2163 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2164 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2165 wav->discont = FALSE;
2167 GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
2170 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2171 GST_BUFFER_DURATION (buf) = duration;
2173 GST_LOG_OBJECT (wav,
2174 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2175 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2176 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2178 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2181 if (obtained < wav->dataleft) {
2182 wav->offset += obtained;
2183 wav->dataleft -= obtained;
2185 wav->offset += wav->dataleft;
2189 /* Iterate until need more data, so adapter size won't grow */
2190 if (wav->streaming) {
2191 GST_LOG_OBJECT (wav,
2192 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2194 goto iterate_adapter;
2201 GST_DEBUG_OBJECT (wav, "found EOS");
2202 return GST_FLOW_EOS;
2206 /* check if we got EOS */
2207 if (res == GST_FLOW_EOS)
2210 GST_WARNING_OBJECT (wav,
2211 "Error getting %" G_GINT64_FORMAT " bytes from the "
2212 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2217 GST_INFO_OBJECT (wav,
2218 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2219 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2220 gst_pad_is_linked (wav->srcpad));
2226 gst_wavparse_loop (GstPad * pad)
2229 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2233 GST_LOG_OBJECT (wav, "process data");
2235 switch (wav->state) {
2236 case GST_WAVPARSE_START:
2237 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2238 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2242 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2243 event = gst_event_new_stream_start (stream_id);
2244 gst_event_set_group_id (event, gst_util_group_id_next ());
2245 gst_pad_push_event (wav->srcpad, event);
2248 wav->state = GST_WAVPARSE_HEADER;
2251 case GST_WAVPARSE_HEADER:
2252 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2253 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2256 wav->state = GST_WAVPARSE_DATA;
2257 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2260 case GST_WAVPARSE_DATA:
2261 if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
2265 g_assert_not_reached ();
2272 const gchar *reason = gst_flow_get_name (ret);
2274 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2275 gst_pad_pause_task (pad);
2277 if (ret == GST_FLOW_EOS) {
2278 /* handle end-of-stream/segment */
2279 /* so align our position with the end of it, if there is one
2280 * this ensures a subsequent will arrive at correct base/acc time */
2281 if (wav->segment.format == GST_FORMAT_TIME) {
2282 if (wav->segment.rate > 0.0 &&
2283 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2284 wav->segment.position = wav->segment.stop;
2285 else if (wav->segment.rate < 0.0)
2286 wav->segment.position = wav->segment.start;
2288 if (wav->state == GST_WAVPARSE_START || !wav->caps) {
2289 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2290 ("No valid input found before end of stream"));
2291 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2293 /* add pad before we perform EOS */
2294 if (G_UNLIKELY (wav->first)) {
2296 gst_wavparse_add_src_pad (wav, NULL);
2298 /* If we have a pending start segment, send it now. Can happen if a seek
2299 * causes an immediate EOS */
2300 if (G_UNLIKELY (wav->start_segment != NULL)) {
2301 gst_pad_push_event (wav->srcpad, wav->start_segment);
2302 wav->start_segment = NULL;
2306 /* perform EOS logic */
2307 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2310 if ((stop = wav->segment.stop) == -1)
2311 stop = wav->segment.duration;
2313 gst_element_post_message (GST_ELEMENT_CAST (wav),
2314 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2315 wav->segment.format, stop));
2316 gst_pad_push_event (wav->srcpad,
2317 gst_event_new_segment_done (wav->segment.format, stop));
2319 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2322 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2323 /* for fatal errors we post an error message, post the error
2324 * first so the app knows about the error first. */
2325 GST_ELEMENT_FLOW_ERROR (wav, ret);
2326 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2332 static GstFlowReturn
2333 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2336 GstWavParse *wav = GST_WAVPARSE (parent);
2338 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2339 gst_buffer_get_size (buf));
2341 /* Hold a reference to the buffer, as we access buffer properties in the
2342 `GST_WAVPARSE_DATA` case below and `gst_adapter_push` steals a reference
2344 gst_buffer_ref (buf);
2346 gst_adapter_push (wav->adapter, buf);
2348 switch (wav->state) {
2349 case GST_WAVPARSE_START:
2350 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2351 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2354 if (wav->state != GST_WAVPARSE_HEADER)
2357 /* otherwise fall-through */
2358 case GST_WAVPARSE_HEADER:
2359 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2360 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2363 if (!wav->got_fmt || wav->datastart == 0)
2366 wav->state = GST_WAVPARSE_DATA;
2367 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2370 case GST_WAVPARSE_DATA:
2371 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2372 wav->discont = TRUE;
2373 if ((ret = gst_wavparse_stream_data (wav, FALSE)) != GST_FLOW_OK)
2377 g_assert_not_reached ();
2380 if (G_UNLIKELY (wav->abort_buffering)) {
2381 wav->abort_buffering = FALSE;
2382 ret = GST_FLOW_ERROR;
2383 /* sort of demux/parse error */
2384 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2387 gst_buffer_unref (buf);
2392 static GstFlowReturn
2393 gst_wavparse_flush_data (GstWavParse * wav)
2395 GstFlowReturn ret = GST_FLOW_OK;
2398 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2399 ret = gst_wavparse_stream_data (wav, TRUE);
2406 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2408 GstWavParse *wav = GST_WAVPARSE (parent);
2409 gboolean ret = TRUE;
2411 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2413 switch (GST_EVENT_TYPE (event)) {
2414 case GST_EVENT_CAPS:
2416 /* discard, we'll come up with proper src caps */
2417 gst_event_unref (event);
2420 case GST_EVENT_SEGMENT:
2422 gint64 start, stop, offset = 0, end_offset = -1;
2425 /* some debug output */
2426 gst_event_copy_segment (event, &segment);
2427 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2430 if (wav->state != GST_WAVPARSE_DATA) {
2431 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2435 /* now we are either committed to TIME or BYTE format,
2436 * and we only expect a BYTE segment, e.g. following a seek */
2437 if (segment.format == GST_FORMAT_BYTES) {
2438 /* handle (un)signed issues */
2439 start = segment.start;
2440 stop = segment.stop;
2443 start -= wav->datastart;
2444 start = MAX (start, 0);
2448 stop -= wav->datastart;
2449 stop = MAX (stop, 0);
2451 if (wav->segment.format == GST_FORMAT_TIME) {
2452 guint64 bps = wav->bps;
2454 /* operating in format TIME, so we can convert */
2455 if (!bps && wav->fact)
2457 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2461 gst_util_uint64_scale_ceil (start, GST_SECOND,
2462 (guint64) wav->bps);
2465 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2466 (guint64) wav->bps);
2470 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2474 segment.start = start;
2475 segment.stop = stop;
2477 /* accept upstream's notion of segment and distribute along */
2478 segment.format = wav->segment.format;
2479 segment.time = segment.position = segment.start;
2480 segment.duration = wav->segment.duration;
2481 segment.base = gst_segment_to_running_time (&wav->segment,
2482 GST_FORMAT_TIME, wav->segment.position);
2484 gst_segment_copy_into (&segment, &wav->segment);
2486 /* also store the newsegment event for the streaming thread */
2487 if (wav->start_segment)
2488 gst_event_unref (wav->start_segment);
2489 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2490 wav->start_segment = gst_event_new_segment (&segment);
2492 /* stream leftover data in current segment */
2493 gst_wavparse_flush_data (wav);
2494 /* and set up streaming thread for next one */
2495 wav->offset = offset;
2496 wav->end_offset = end_offset;
2498 if (wav->datasize > 0 && (wav->end_offset == -1
2499 || wav->end_offset > wav->datastart + wav->datasize))
2500 wav->end_offset = wav->datastart + wav->datasize;
2502 if (wav->end_offset != -1) {
2503 wav->dataleft = wav->end_offset - wav->offset;
2505 /* infinity; upstream will EOS when done */
2506 wav->dataleft = G_MAXUINT64;
2509 gst_event_unref (event);
2514 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2515 ("No valid input found before end of stream"));
2517 switch (wav->state) {
2518 case GST_WAVPARSE_START:
2519 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2520 ("No valid input found before end of stream"));
2522 case GST_WAVPARSE_HEADER:
2523 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
2524 ("No audio data chunk found before end of stream"));
2526 case GST_WAVPARSE_DATA:
2527 /* add pad if needed so EOS is seen downstream */
2528 if (G_UNLIKELY (wav->first)) {
2530 gst_wavparse_add_src_pad (wav, NULL);
2532 /* stream leftover data in current segment */
2533 gst_wavparse_flush_data (wav);
2536 g_assert_not_reached ();
2540 case GST_EVENT_FLUSH_STOP:
2545 gst_adapter_clear (wav->adapter);
2546 wav->discont = TRUE;
2547 dur = wav->segment.duration;
2548 gst_segment_init (&wav->segment, wav->segment.format);
2549 wav->segment.duration = dur;
2553 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2561 /* convert and query stuff */
2562 static const GstFormat *
2563 gst_wavparse_get_formats (GstPad * pad)
2565 static const GstFormat formats[] = {
2568 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2577 gst_wavparse_pad_convert (GstPad * pad,
2578 GstFormat src_format, gint64 src_value,
2579 GstFormat * dest_format, gint64 * dest_value)
2581 GstWavParse *wavparse;
2582 gboolean res = TRUE;
2584 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2586 if (*dest_format == src_format) {
2587 *dest_value = src_value;
2591 if ((wavparse->bps == 0) && !wavparse->fact)
2594 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2595 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2597 switch (src_format) {
2598 case GST_FORMAT_BYTES:
2599 switch (*dest_format) {
2600 case GST_FORMAT_DEFAULT:
2601 *dest_value = src_value / wavparse->bytes_per_sample;
2602 /* make sure we end up on a sample boundary */
2603 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2605 case GST_FORMAT_TIME:
2606 /* src_value + datastart = offset */
2607 GST_INFO_OBJECT (wavparse,
2608 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2610 if (wavparse->bps > 0)
2611 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2612 (guint64) wavparse->bps);
2613 else if (wavparse->fact) {
2614 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2615 wavparse->rate, wavparse->fact);
2618 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2629 case GST_FORMAT_DEFAULT:
2630 switch (*dest_format) {
2631 case GST_FORMAT_BYTES:
2632 *dest_value = src_value * wavparse->bytes_per_sample;
2634 case GST_FORMAT_TIME:
2635 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2636 (guint64) wavparse->rate);
2644 case GST_FORMAT_TIME:
2645 switch (*dest_format) {
2646 case GST_FORMAT_BYTES:
2647 if (wavparse->bps > 0)
2648 *dest_value = gst_util_uint64_scale (src_value,
2649 (guint64) wavparse->bps, GST_SECOND);
2651 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2652 wavparse->rate, wavparse->fact);
2654 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2656 /* make sure we end up on a sample boundary */
2657 *dest_value -= *dest_value % wavparse->blockalign;
2659 case GST_FORMAT_DEFAULT:
2660 *dest_value = gst_util_uint64_scale (src_value,
2661 (guint64) wavparse->rate, GST_SECOND);
2680 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2686 /* handle queries for location and length in requested format */
2688 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2690 gboolean res = TRUE;
2691 GstWavParse *wav = GST_WAVPARSE (parent);
2693 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2695 if (wav->state != GST_WAVPARSE_DATA) {
2696 return gst_pad_query_default (pad, parent, query);
2699 switch (GST_QUERY_TYPE (query)) {
2700 case GST_QUERY_POSITION:
2706 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2707 curb = wav->offset - wav->datastart;
2708 gst_query_parse_position (query, &format, NULL);
2709 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2712 case GST_FORMAT_BYTES:
2713 format = GST_FORMAT_BYTES;
2717 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2722 gst_query_set_position (query, format, cur);
2725 case GST_QUERY_DURATION:
2727 gint64 duration = 0;
2730 if (wav->ignore_length) {
2735 gst_query_parse_duration (query, &format, NULL);
2738 case GST_FORMAT_BYTES:{
2739 format = GST_FORMAT_BYTES;
2740 duration = wav->datasize;
2743 case GST_FORMAT_TIME:
2744 if ((res = gst_wavparse_calculate_duration (wav))) {
2745 duration = wav->duration;
2753 gst_query_set_duration (query, format, duration);
2756 case GST_QUERY_CONVERT:
2758 gint64 srcvalue, dstvalue;
2759 GstFormat srcformat, dstformat;
2761 gst_query_parse_convert (query, &srcformat, &srcvalue,
2762 &dstformat, &dstvalue);
2763 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2764 &dstformat, &dstvalue);
2766 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2769 case GST_QUERY_SEEKING:{
2771 gboolean seekable = FALSE;
2773 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2774 if (fmt == wav->segment.format) {
2775 if (wav->streaming) {
2778 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2779 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2780 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2781 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2783 gst_query_unref (q);
2785 GST_LOG_OBJECT (wav, "looping => seekable");
2789 } else if (fmt == GST_FORMAT_TIME) {
2793 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2797 case GST_QUERY_SEGMENT:
2802 format = wav->segment.format;
2805 gst_segment_to_stream_time (&wav->segment, format,
2806 wav->segment.start);
2807 if ((stop = wav->segment.stop) == -1)
2808 stop = wav->segment.duration;
2810 stop = gst_segment_to_stream_time (&wav->segment, format, stop);
2812 gst_query_set_segment (query, wav->segment.rate, format, start, stop);
2817 res = gst_pad_query_default (pad, parent, query);
2824 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2826 GstWavParse *wavparse = GST_WAVPARSE (parent);
2827 gboolean res = FALSE;
2829 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2831 switch (GST_EVENT_TYPE (event)) {
2832 case GST_EVENT_SEEK:
2833 /* can only handle events when we are in the data state */
2834 if (wavparse->state == GST_WAVPARSE_DATA) {
2835 res = gst_wavparse_perform_seek (wavparse, event);
2837 gst_event_unref (event);
2840 case GST_EVENT_TOC_SELECT:
2843 GstTocEntry *entry = NULL;
2844 GstEvent *seek_event;
2847 if (!wavparse->toc) {
2848 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2851 gst_event_parse_toc_select (event, &uid);
2853 GST_OBJECT_LOCK (wavparse);
2854 entry = gst_toc_find_entry (wavparse->toc, uid);
2855 if (entry == NULL) {
2856 GST_OBJECT_UNLOCK (wavparse);
2857 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2861 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2862 GST_OBJECT_UNLOCK (wavparse);
2863 seek_event = gst_event_new_seek (1.0,
2865 GST_SEEK_FLAG_FLUSH,
2866 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2867 res = gst_wavparse_perform_seek (wavparse, seek_event);
2868 gst_event_unref (seek_event);
2872 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2876 gst_event_unref (event);
2881 res = gst_pad_push_event (wavparse->sinkpad, event);
2888 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2890 GstWavParse *wav = GST_WAVPARSE (parent);
2895 gst_adapter_clear (wav->adapter);
2896 g_object_unref (wav->adapter);
2897 wav->adapter = NULL;
2900 query = gst_query_new_scheduling ();
2902 if (!gst_pad_peer_query (sinkpad, query)) {
2903 gst_query_unref (query);
2907 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2908 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2909 gst_query_unref (query);
2914 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2915 wav->streaming = FALSE;
2916 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2920 GST_DEBUG_OBJECT (sinkpad, "activating push");
2921 wav->streaming = TRUE;
2922 wav->adapter = gst_adapter_new ();
2923 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2929 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2930 GstPadMode mode, gboolean active)
2935 case GST_PAD_MODE_PUSH:
2938 case GST_PAD_MODE_PULL:
2940 /* if we have a scheduler we can start the task */
2941 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2944 res = gst_pad_stop_task (sinkpad);
2954 static GstStateChangeReturn
2955 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2957 GstStateChangeReturn ret;
2958 GstWavParse *wav = GST_WAVPARSE (element);
2960 switch (transition) {
2961 case GST_STATE_CHANGE_NULL_TO_READY:
2963 case GST_STATE_CHANGE_READY_TO_PAUSED:
2965 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2971 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2973 switch (transition) {
2974 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2976 case GST_STATE_CHANGE_PAUSED_TO_READY:
2977 gst_wavparse_reset (wav);
2979 case GST_STATE_CHANGE_READY_TO_NULL:
2988 gst_wavparse_set_property (GObject * object, guint prop_id,
2989 const GValue * value, GParamSpec * pspec)
2993 g_return_if_fail (GST_IS_WAVPARSE (object));
2994 self = GST_WAVPARSE (object);
2997 case PROP_IGNORE_LENGTH:
2998 self->ignore_length = g_value_get_boolean (value);
3001 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3007 gst_wavparse_get_property (GObject * object, guint prop_id,
3008 GValue * value, GParamSpec * pspec)
3012 g_return_if_fail (GST_IS_WAVPARSE (object));
3013 self = GST_WAVPARSE (object);
3016 case PROP_IGNORE_LENGTH:
3017 g_value_set_boolean (value, self->ignore_length);
3020 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3025 plugin_init (GstPlugin * plugin)
3027 return GST_ELEMENT_REGISTER (wavparse, plugin);
3030 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
3033 "Parse a .wav file into raw audio",
3034 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)