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/pbutils/descriptions.h>
58 #include <gst/gst-i18n-plugin.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")
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,
120 /* Offset Size Description Value
121 * 0x00 4 ID unique identification value
122 * 0x04 4 Position play order position
123 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
124 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
125 * 0x10 4 Block Start Byte Offset to sample of First Channel
126 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
130 guint32 data_chunk_id;
133 guint32 sample_offset;
138 /* Offset Size Description Value
139 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
142 guint32 cue_point_id;
144 } GstWavParseLabl, GstWavParseNote;
147 gst_wavparse_class_init (GstWavParseClass * klass)
149 GstElementClass *gstelement_class;
150 GObjectClass *object_class;
151 GstPadTemplate *src_template;
153 gstelement_class = (GstElementClass *) klass;
154 object_class = (GObjectClass *) klass;
156 parent_class = g_type_class_peek_parent (klass);
158 object_class->dispose = gst_wavparse_dispose;
160 object_class->set_property = gst_wavparse_set_property;
161 object_class->get_property = gst_wavparse_get_property;
164 * GstWavParse:ignore-length:
166 * This selects whether the length found in a data chunk
167 * should be ignored. This may be useful for streamed audio
168 * where the length is unknown until the end of streaming,
169 * and various software/hardware just puts some random value
170 * in there and hopes it doesn't break too much.
172 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
173 g_param_spec_boolean ("ignore-length",
175 "Ignore length from the Wave header",
176 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
179 gstelement_class->change_state = gst_wavparse_change_state;
180 gstelement_class->send_event = gst_wavparse_send_event;
183 gst_element_class_add_static_pad_template (gstelement_class,
184 &sink_template_factory);
186 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
187 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
188 gst_element_class_add_pad_template (gstelement_class, src_template);
190 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
191 "Codec/Demuxer/Audio",
192 "Parse a .wav file into raw audio",
193 "Erik Walthinsen <omega@cse.ogi.edu>");
197 gst_wavparse_notes_free (GstWavParseNote * note)
205 gst_wavparse_labls_free (GstWavParseLabl * labl)
213 gst_wavparse_reset (GstWavParse * wav)
215 wav->state = GST_WAVPARSE_START;
217 /* These will all be set correctly in the fmt chunk */
231 wav->got_fmt = FALSE;
235 gst_event_unref (wav->seek_event);
236 wav->seek_event = NULL;
238 gst_adapter_clear (wav->adapter);
239 g_object_unref (wav->adapter);
243 gst_tag_list_unref (wav->tags);
246 gst_toc_unref (wav->toc);
249 g_list_free_full (wav->cues, g_free);
252 g_list_free_full (wav->labls, (GDestroyNotify) gst_wavparse_labls_free);
255 g_list_free_full (wav->notes, (GDestroyNotify) gst_wavparse_notes_free);
258 gst_caps_unref (wav->caps);
260 if (wav->start_segment)
261 gst_event_unref (wav->start_segment);
262 wav->start_segment = NULL;
266 gst_wavparse_dispose (GObject * object)
268 GstWavParse *wav = GST_WAVPARSE (object);
270 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
271 gst_wavparse_reset (wav);
273 G_OBJECT_CLASS (parent_class)->dispose (object);
277 gst_wavparse_init (GstWavParse * wavparse)
279 gst_wavparse_reset (wavparse);
283 gst_pad_new_from_static_template (&sink_template_factory, "sink");
284 gst_pad_set_activate_function (wavparse->sinkpad,
285 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
286 gst_pad_set_activatemode_function (wavparse->sinkpad,
287 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
288 gst_pad_set_chain_function (wavparse->sinkpad,
289 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
290 gst_pad_set_event_function (wavparse->sinkpad,
291 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
292 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
296 gst_pad_new_from_template (gst_element_class_get_pad_template
297 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
298 gst_pad_use_fixed_caps (wavparse->srcpad);
299 gst_pad_set_query_function (wavparse->srcpad,
300 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
301 gst_pad_set_event_function (wavparse->srcpad,
302 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
303 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
307 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
311 if (!gst_riff_parse_file_header (element, buf, &doctype))
314 if (doctype != GST_RIFF_RIFF_WAVE)
322 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
323 ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
329 gst_wavparse_stream_init (GstWavParse * wav)
332 GstBuffer *buf = NULL;
334 if ((res = gst_pad_pull_range (wav->sinkpad,
335 wav->offset, 12, &buf)) != GST_FLOW_OK)
337 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
338 return GST_FLOW_ERROR;
346 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
348 /* -1 always maps to -1 */
354 /* 0 always maps to 0 */
361 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
363 } else if (wav->fact) {
364 guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
365 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
372 /* This function is used to perform seeks on the element.
374 * It also works when event is NULL, in which case it will just
375 * start from the last configured segment. This technique is
376 * used when activating the element and to perform the seek in
380 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
384 GstFormat format, bformat;
386 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
387 gint64 cur, stop, upstream_size;
390 GstSegment seeksegment = { 0, };
395 GST_DEBUG_OBJECT (wav, "doing seek with event");
397 gst_event_parse_seek (event, &rate, &format, &flags,
398 &cur_type, &cur, &stop_type, &stop);
399 seqnum = gst_event_get_seqnum (event);
401 /* no negative rates yet */
405 if (format != wav->segment.format) {
406 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
407 gst_format_get_name (format),
408 gst_format_get_name (wav->segment.format));
410 if (cur_type != GST_SEEK_TYPE_NONE)
412 gst_pad_query_convert (wav->srcpad, format, cur,
413 wav->segment.format, &cur);
414 if (res && stop_type != GST_SEEK_TYPE_NONE)
416 gst_pad_query_convert (wav->srcpad, format, stop,
417 wav->segment.format, &stop);
421 format = wav->segment.format;
424 GST_DEBUG_OBJECT (wav, "doing seek without event");
427 cur_type = GST_SEEK_TYPE_SET;
428 stop_type = GST_SEEK_TYPE_SET;
431 /* in push mode, we must delegate to upstream */
432 if (wav->streaming) {
433 gboolean res = FALSE;
435 /* if streaming not yet started; only prepare initial newsegment */
436 if (!event || wav->state != GST_WAVPARSE_DATA) {
437 if (wav->start_segment)
438 gst_event_unref (wav->start_segment);
439 wav->start_segment = gst_event_new_segment (&wav->segment);
442 /* convert seek positions to byte positions in data sections */
443 if (format == GST_FORMAT_TIME) {
444 /* should not fail */
445 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
447 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
450 /* mind sample boundary and header */
452 cur -= (cur % wav->bytes_per_sample);
453 cur += wav->datastart;
456 stop -= (stop % wav->bytes_per_sample);
457 stop += wav->datastart;
459 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
460 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
462 /* BYTE seek event */
463 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
465 gst_event_set_seqnum (event, seqnum);
466 res = gst_pad_push_event (wav->sinkpad, event);
472 flush = flags & GST_SEEK_FLAG_FLUSH;
474 /* now we need to make sure the streaming thread is stopped. We do this by
475 * either sending a FLUSH_START event downstream which will cause the
476 * streaming thread to stop with a WRONG_STATE.
477 * For a non-flushing seek we simply pause the task, which will happen as soon
478 * as it completes one iteration (and thus might block when the sink is
479 * blocking in preroll). */
482 GST_DEBUG_OBJECT (wav, "sending flush start");
484 fevent = gst_event_new_flush_start ();
485 gst_event_set_seqnum (fevent, seqnum);
486 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
487 gst_pad_push_event (wav->srcpad, fevent);
489 gst_pad_pause_task (wav->sinkpad);
492 /* we should now be able to grab the streaming thread because we stopped it
493 * with the above flush/pause code */
494 GST_PAD_STREAM_LOCK (wav->sinkpad);
496 /* save current position */
497 last_stop = wav->segment.position;
499 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
501 /* copy segment, we need this because we still need the old
502 * segment when we close the current segment. */
503 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
505 /* configure the seek parameters in the seeksegment. We will then have the
506 * right values in the segment to perform the seek */
508 GST_DEBUG_OBJECT (wav, "configuring seek");
509 gst_segment_do_seek (&seeksegment, rate, format, flags,
510 cur_type, cur, stop_type, stop, &update);
513 /* figure out the last position we need to play. If it's configured (stop !=
514 * -1), use that, else we play until the total duration of the file */
515 if ((stop = seeksegment.stop) == -1)
516 stop = seeksegment.duration;
518 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
519 if ((cur_type != GST_SEEK_TYPE_NONE)) {
520 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
521 * we can just copy the last_stop. If not, we use the bps to convert TIME to
523 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
524 (gint64 *) & wav->offset))
525 wav->offset = seeksegment.position;
526 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
527 wav->offset -= (wav->offset % wav->bytes_per_sample);
528 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
529 wav->offset += wav->datastart;
530 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
532 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
536 if (stop_type != GST_SEEK_TYPE_NONE) {
537 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
538 wav->end_offset = stop;
539 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
540 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
541 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
542 wav->end_offset += wav->datastart;
543 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
545 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
549 /* make sure filesize is not exceeded due to rounding errors or so,
550 * same precaution as in _stream_headers */
551 bformat = GST_FORMAT_BYTES;
552 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
553 wav->end_offset = MIN (wav->end_offset, upstream_size);
555 if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
556 wav->end_offset = wav->datastart + wav->datasize;
558 /* this is the range of bytes we will use for playback */
559 wav->offset = MIN (wav->offset, wav->end_offset);
560 wav->dataleft = wav->end_offset - wav->offset;
562 GST_DEBUG_OBJECT (wav,
563 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
564 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
565 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
567 /* prepare for streaming again */
571 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
572 GST_DEBUG_OBJECT (wav, "sending flush stop");
574 fevent = gst_event_new_flush_stop (TRUE);
575 gst_event_set_seqnum (fevent, seqnum);
576 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
577 gst_pad_push_event (wav->srcpad, fevent);
580 /* now we did the seek and can activate the new segment values */
581 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
583 /* if we're doing a segment seek, post a SEGMENT_START message */
584 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
585 gst_element_post_message (GST_ELEMENT_CAST (wav),
586 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
587 wav->segment.format, wav->segment.position));
590 /* now create the newsegment */
591 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
592 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
594 /* store the newsegment event so it can be sent from the streaming thread. */
595 if (wav->start_segment)
596 gst_event_unref (wav->start_segment);
597 wav->start_segment = gst_event_new_segment (&wav->segment);
598 gst_event_set_seqnum (wav->start_segment, seqnum);
600 /* mark discont if we are going to stream from another position. */
601 if (last_stop != wav->segment.position) {
602 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
606 /* and start the streaming task again */
607 if (!wav->streaming) {
608 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
612 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
619 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
624 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
629 GST_DEBUG_OBJECT (wav,
630 "Could not determine byte position for desired time");
636 * gst_wavparse_peek_chunk_info:
637 * @wav Wavparse object
638 * @tag holder for tag
639 * @size holder for tag size
641 * Peek next chunk info (tag and size)
643 * Returns: %TRUE when the chunk info (header) is available
646 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
648 const guint8 *data = NULL;
650 if (gst_adapter_available (wav->adapter) < 8)
653 data = gst_adapter_map (wav->adapter, 8);
654 *tag = GST_READ_UINT32_LE (data);
655 *size = GST_READ_UINT32_LE (data + 4);
656 gst_adapter_unmap (wav->adapter);
658 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
659 GST_FOURCC_ARGS (*tag));
665 * gst_wavparse_peek_chunk:
666 * @wav Wavparse object
667 * @tag holder for tag
668 * @size holder for tag size
670 * Peek enough data for one full chunk
672 * Returns: %TRUE when the full chunk is available
675 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
677 guint32 peek_size = 0;
680 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
683 /* size 0 -> empty data buffer would surprise most callers,
684 * large size -> do not bother trying to squeeze that into adapter,
685 * so we throw poor man's exception, which can be caught if caller really
686 * wants to handle 0 size chunk */
687 if (!(*size) || (*size) >= (1 << 30)) {
688 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
689 *size, GST_FOURCC_ARGS (*tag));
690 /* chain should give up */
691 wav->abort_buffering = TRUE;
694 peek_size = (*size + 1) & ~1;
695 available = gst_adapter_available (wav->adapter);
697 if (available >= (8 + peek_size)) {
700 GST_LOG ("but only %u bytes available now", available);
706 * gst_wavparse_calculate_duration:
707 * @wav: wavparse object
709 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
712 * Returns: %TRUE if duration is available.
715 gst_wavparse_calculate_duration (GstWavParse * wav)
717 if (wav->duration > 0)
721 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
723 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
725 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
726 GST_TIME_ARGS (wav->duration));
728 } else if (wav->fact) {
730 gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
731 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
732 GST_TIME_ARGS (wav->duration));
739 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
744 if (wav->streaming) {
745 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
748 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
749 GST_FOURCC_ARGS (tag));
750 flush = 8 + ((size + 1) & ~1);
751 wav->offset += flush;
752 if (wav->streaming) {
753 gst_adapter_flush (wav->adapter, flush);
755 gst_buffer_unref (buf);
762 * gst_wavparse_cue_chunk:
763 * @wav GstWavParse object
764 * @data holder for data
765 * @size holder for data size
767 * Parse cue chunk from @data to wav->cues.
769 * Returns: %TRUE when cue chunk is available
772 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
779 GST_WARNING_OBJECT (wav, "found another cue's");
783 ncues = GST_READ_UINT32_LE (data);
785 if (size < 4 + ncues * 24) {
786 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
792 for (i = 0; i < ncues; i++) {
793 cue = g_new0 (GstWavParseCue, 1);
794 cue->id = GST_READ_UINT32_LE (data);
795 cue->position = GST_READ_UINT32_LE (data + 4);
796 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
797 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
798 cue->block_start = GST_READ_UINT32_LE (data + 16);
799 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
800 cues = g_list_append (cues, cue);
810 * gst_wavparse_labl_chunk:
811 * @wav GstWavParse object
812 * @data holder for data
813 * @size holder for data size
815 * Parse labl from @data to wav->labls.
817 * Returns: %TRUE when labl chunk is available
820 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
822 GstWavParseLabl *labl;
827 labl = g_new0 (GstWavParseLabl, 1);
831 labl->cue_point_id = GST_READ_UINT32_LE (data);
832 labl->text = g_memdup (data + 4, size - 4);
834 wav->labls = g_list_append (wav->labls, labl);
840 * gst_wavparse_note_chunk:
841 * @wav GstWavParse object
842 * @data holder for data
843 * @size holder for data size
845 * Parse note from @data to wav->notes.
847 * Returns: %TRUE when note chunk is available
850 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
852 GstWavParseNote *note;
857 note = g_new0 (GstWavParseNote, 1);
861 note->cue_point_id = GST_READ_UINT32_LE (data);
862 note->text = g_memdup (data + 4, size - 4);
864 wav->notes = g_list_append (wav->notes, note);
870 * gst_wavparse_smpl_chunk:
871 * @wav GstWavParse object
872 * @data holder for data
873 * @size holder for data size
875 * Parse smpl chunk from @data.
877 * Returns: %TRUE when cue chunk is available
880 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
885 manufacturer_id = GST_READ_UINT32_LE (data);
886 product_id = GST_READ_UINT32_LE (data + 4);
887 sample_period = GST_READ_UINT32_LE (data + 8);
889 note_number = GST_READ_UINT32_LE (data + 12);
891 pitch_fraction = GST_READ_UINT32_LE (data + 16);
892 SMPTE_format = GST_READ_UINT32_LE (data + 20);
893 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
894 num_sample_loops = GST_READ_UINT32_LE (data + 28);
895 List of Sample Loops, 24 bytes each
899 wav->tags = gst_tag_list_new_empty ();
900 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
901 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
906 * gst_wavparse_adtl_chunk:
907 * @wav GstWavParse object
908 * @data holder for data
909 * @size holder for data size
911 * Parse adtl from @data.
913 * Returns: %TRUE when adtl chunk is available
916 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
918 guint32 ltag, lsize, offset = 0;
921 ltag = GST_READ_UINT32_LE (data + offset);
922 lsize = GST_READ_UINT32_LE (data + offset + 4);
924 if (lsize + 8 > size) {
925 GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
930 case GST_RIFF_TAG_labl:
931 gst_wavparse_labl_chunk (wav, data + offset, size);
933 case GST_RIFF_TAG_note:
934 gst_wavparse_note_chunk (wav, data + offset, size);
937 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
938 GST_FOURCC_ARGS (ltag));
939 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
942 offset += 8 + GST_ROUND_UP_2 (lsize);
943 size -= 8 + GST_ROUND_UP_2 (lsize);
950 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
952 GstTagList *tags = NULL;
953 GstTocEntry *entry = NULL;
955 entry = gst_toc_find_entry (toc, id);
957 tags = gst_toc_entry_get_tags (entry);
959 tags = gst_tag_list_new_empty ();
960 gst_toc_entry_set_tags (entry, tags);
968 * gst_wavparse_create_toc:
969 * @wav GstWavParse object
971 * Create TOC from wav->cues and wav->labls.
974 gst_wavparse_create_toc (GstWavParse * wav)
980 GstWavParseLabl *labl;
981 GstWavParseNote *note;
984 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
986 GST_OBJECT_LOCK (wav);
988 GST_OBJECT_UNLOCK (wav);
989 GST_WARNING_OBJECT (wav, "found another TOC");
994 GST_OBJECT_UNLOCK (wav);
998 /* FIXME: send CURRENT scope toc too */
999 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
1001 /* add cue edition */
1002 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
1003 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
1004 gst_toc_append_entry (toc, entry);
1006 /* add tracks in cue edition */
1010 prev_subentry = cur_subentry;
1011 /* previous track stop time = current track start time */
1012 if (prev_subentry != NULL) {
1013 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1014 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1015 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1017 id = g_strdup_printf ("%08x", cue->id);
1018 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1020 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1021 stop = wav->duration;
1022 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1023 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1024 list = g_list_next (list);
1027 /* add tags in tracks */
1031 id = g_strdup_printf ("%08x", labl->cue_point_id);
1032 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1035 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1038 list = g_list_next (list);
1043 id = g_strdup_printf ("%08x", note->cue_point_id);
1044 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1047 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1050 list = g_list_next (list);
1053 /* send data as TOC */
1056 /* send TOC event */
1058 GST_OBJECT_UNLOCK (wav);
1059 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1065 #define MAX_BUFFER_SIZE 4096
1068 parse_ds64 (GstWavParse * wav, GstBuffer * buf)
1071 guint32 dataSizeLow, dataSizeHigh;
1072 guint32 sampleCountLow, sampleCountHigh;
1074 gst_buffer_map (buf, &map, GST_MAP_READ);
1075 dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
1076 dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
1077 sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
1078 sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
1079 gst_buffer_unmap (buf, &map);
1080 if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
1081 wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
1083 if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
1084 wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
1087 GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
1088 " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
1092 static GstFlowReturn
1093 gst_wavparse_stream_headers (GstWavParse * wav)
1095 GstFlowReturn res = GST_FLOW_OK;
1096 GstBuffer *buf = NULL;
1097 gst_riff_strf_auds *header = NULL;
1099 gboolean gotdata = FALSE;
1100 GstCaps *caps = NULL;
1101 gchar *codec_name = NULL;
1102 gint64 upstream_size = 0;
1105 /* search for "_fmt" chunk, which must be before "data" */
1106 while (!wav->got_fmt) {
1109 if (wav->streaming) {
1110 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1113 gst_adapter_flush (wav->adapter, 8);
1117 buf = gst_adapter_take_buffer (wav->adapter, size);
1119 gst_adapter_flush (wav->adapter, 1);
1120 wav->offset += GST_ROUND_UP_2 (size);
1122 buf = gst_buffer_new ();
1125 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1126 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1130 if (tag == GST_RS64_TAG_DS64) {
1131 if (!parse_ds64 (wav, buf))
1137 if (tag != GST_RIFF_TAG_fmt) {
1138 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1139 GST_FOURCC_ARGS (tag));
1140 gst_buffer_unref (buf);
1145 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1147 goto parse_header_error;
1149 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1151 /* do sanity checks of header fields */
1152 if (header->channels == 0)
1154 if (header->rate == 0)
1157 GST_DEBUG_OBJECT (wav, "creating the caps");
1159 /* Note: gst_riff_create_audio_caps might need to fix values in
1160 * the header header depending on the format, so call it first */
1161 /* FIXME: Need to handle the channel reorder map */
1162 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1163 NULL, &codec_name, NULL);
1166 gst_buffer_unref (extra);
1169 goto unknown_format;
1171 /* If we got raw audio from upstream, we remove the codec_data field,
1172 * which may have been added if the wav header included an extended
1173 * chunk. We want to keep it for non raw audio.
1175 s = gst_caps_get_structure (caps, 0);
1176 if (s && gst_structure_has_name (s, "audio/x-raw")) {
1177 gst_structure_remove_field (s, "codec_data");
1180 /* do more sanity checks of header fields
1181 * (these can be sanitized by gst_riff_create_audio_caps()
1183 wav->format = header->format;
1184 wav->rate = header->rate;
1185 wav->channels = header->channels;
1186 wav->blockalign = header->blockalign;
1187 wav->depth = header->bits_per_sample;
1188 wav->av_bps = header->av_bps;
1194 /* do format specific handling */
1195 switch (wav->format) {
1196 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1197 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1199 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1200 * bitrate inside the mpeg stream */
1201 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1205 case GST_RIFF_WAVE_FORMAT_PCM:
1206 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1207 goto invalid_blockalign;
1210 if (wav->av_bps > wav->blockalign * wav->rate)
1212 /* use the configured bps */
1213 wav->bps = wav->av_bps;
1217 wav->width = (wav->blockalign * 8) / wav->channels;
1218 wav->bytes_per_sample = wav->channels * wav->width / 8;
1220 if (wav->bytes_per_sample <= 0)
1221 goto no_bytes_per_sample;
1223 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1224 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1225 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1226 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1227 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1228 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1229 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1231 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1232 * formats). This will make the element output a BYTE format segment and
1233 * will not timestamp the outgoing buffers.
1235 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1237 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1239 /* create pad later so we can sniff the first few bytes
1240 * of the real data and correct our caps if necessary */
1241 gst_caps_replace (&wav->caps, caps);
1242 gst_caps_replace (&caps, NULL);
1244 wav->got_fmt = TRUE;
1246 if (wav->tags == NULL)
1247 wav->tags = gst_tag_list_new_empty ();
1250 GstCaps *templ_caps = gst_pad_get_pad_template_caps (wav->sinkpad);
1251 gst_pb_utils_add_codec_description_to_tag_list (wav->tags,
1252 GST_TAG_CONTAINER_FORMAT, templ_caps);
1253 gst_caps_unref (templ_caps);
1256 /* If bps is nonzero, then we do have a valid bitrate that can be
1257 * announced in a tag list. */
1259 guint bitrate = wav->bps * 8;
1260 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1261 GST_TAG_BITRATE, bitrate, NULL);
1265 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1266 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1268 g_free (codec_name);
1274 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1275 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1277 /* loop headers until we get data */
1279 if (wav->streaming) {
1280 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1287 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1288 &buf)) != GST_FLOW_OK)
1289 goto header_read_error;
1290 gst_buffer_map (buf, &map, GST_MAP_READ);
1291 tag = GST_READ_UINT32_LE (map.data);
1292 size = GST_READ_UINT32_LE (map.data + 4);
1293 gst_buffer_unmap (buf, &map);
1296 GST_INFO_OBJECT (wav,
1297 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
1298 G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
1300 /* Maximum valid size is INT_MAX */
1301 if (size & 0x80000000) {
1302 GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
1306 /* Clip to upstream size if known */
1307 if (upstream_size > 0 && size + wav->offset > upstream_size) {
1308 GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
1309 g_assert (upstream_size >= wav->offset);
1310 size = upstream_size - wav->offset;
1313 /* wav is a st00pid format, we don't know for sure where data starts.
1314 * So we have to go bit by bit until we find the 'data' header
1317 case GST_RIFF_TAG_data:{
1320 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1322 if (wav->ignore_length) {
1323 GST_DEBUG_OBJECT (wav, "Ignoring length");
1326 if (wav->streaming) {
1327 gst_adapter_flush (wav->adapter, 8);
1330 gst_buffer_unref (buf);
1333 wav->datastart = wav->offset;
1334 /* use size from ds64 chunk if available */
1335 if (size64 == -1 && wav->datasize > 0) {
1336 GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
1337 size64 = wav->datasize;
1339 /* If size is zero, then the data chunk probably actually extends to
1340 the end of the file */
1341 if (size64 == 0 && upstream_size) {
1342 size64 = upstream_size - wav->datastart;
1344 /* Or the file might be truncated */
1345 else if (upstream_size) {
1346 size64 = MIN (size64, (upstream_size - wav->datastart));
1348 wav->datasize = size64;
1349 wav->dataleft = size64;
1350 wav->end_offset = size64 + wav->datastart;
1351 if (!wav->streaming) {
1352 /* We will continue parsing tags 'till end */
1353 wav->offset += size64;
1355 GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
1358 case GST_RIFF_TAG_fact:{
1359 if (wav->fact == 0 &&
1360 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1361 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1362 const guint data_size = 4;
1364 GST_INFO_OBJECT (wav, "Have fact chunk");
1365 if (size < data_size) {
1366 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1367 /* need more data */
1370 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1374 /* number of samples (for compressed formats) */
1375 if (wav->streaming) {
1376 const guint8 *data = NULL;
1378 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1381 gst_adapter_flush (wav->adapter, 8);
1382 data = gst_adapter_map (wav->adapter, data_size);
1383 wav->fact = GST_READ_UINT32_LE (data);
1384 gst_adapter_unmap (wav->adapter);
1385 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1387 gst_buffer_unref (buf);
1390 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1391 data_size, &buf)) != GST_FLOW_OK)
1392 goto header_read_error;
1393 gst_buffer_extract (buf, 0, &wav->fact, 4);
1394 wav->fact = GUINT32_FROM_LE (wav->fact);
1395 gst_buffer_unref (buf);
1397 GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
1398 wav->offset += 8 + GST_ROUND_UP_2 (size);
1401 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1402 /* need more data */
1408 case GST_RIFF_TAG_acid:{
1409 const gst_riff_acid *acid = NULL;
1410 const guint data_size = sizeof (gst_riff_acid);
1413 GST_INFO_OBJECT (wav, "Have acid chunk");
1414 if (size < data_size) {
1415 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1416 /* need more data */
1419 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1423 if (wav->streaming) {
1424 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1427 gst_adapter_flush (wav->adapter, 8);
1428 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1430 tempo = acid->tempo;
1431 gst_adapter_unmap (wav->adapter);
1434 gst_buffer_unref (buf);
1437 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1438 size, &buf)) != GST_FLOW_OK)
1439 goto header_read_error;
1440 gst_buffer_map (buf, &map, GST_MAP_READ);
1441 acid = (const gst_riff_acid *) map.data;
1442 tempo = acid->tempo;
1443 gst_buffer_unmap (buf, &map);
1445 /* send data as tags */
1447 wav->tags = gst_tag_list_new_empty ();
1448 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1449 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1451 size = GST_ROUND_UP_2 (size);
1452 if (wav->streaming) {
1453 gst_adapter_flush (wav->adapter, size);
1455 gst_buffer_unref (buf);
1457 wav->offset += 8 + size;
1460 /* FIXME: all list tags after data are ignored in streaming mode */
1461 case GST_RIFF_TAG_LIST:{
1464 if (wav->streaming) {
1465 const guint8 *data = NULL;
1467 if (gst_adapter_available (wav->adapter) < 12) {
1470 data = gst_adapter_map (wav->adapter, 12);
1471 ltag = GST_READ_UINT32_LE (data + 8);
1472 gst_adapter_unmap (wav->adapter);
1474 gst_buffer_unref (buf);
1477 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1478 &buf)) != GST_FLOW_OK)
1479 goto header_read_error;
1480 gst_buffer_extract (buf, 8, <ag, 4);
1481 ltag = GUINT32_FROM_LE (ltag);
1484 case GST_RIFF_LIST_INFO:{
1485 const gint data_size = size - 4;
1488 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1489 if (wav->streaming) {
1490 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1493 gst_adapter_flush (wav->adapter, 12);
1495 if (data_size > 0) {
1496 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1498 gst_adapter_flush (wav->adapter, 1);
1502 gst_buffer_unref (buf);
1504 if (data_size > 0) {
1506 gst_pad_pull_range (wav->sinkpad, wav->offset,
1507 data_size, &buf)) != GST_FLOW_OK)
1508 goto header_read_error;
1511 if (data_size > 0) {
1513 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1515 GstTagList *old = wav->tags;
1517 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1519 gst_tag_list_unref (old);
1520 gst_tag_list_unref (new);
1522 gst_buffer_unref (buf);
1523 wav->offset += GST_ROUND_UP_2 (data_size);
1527 case GST_RIFF_LIST_adtl:{
1528 const gint data_size = size - 4;
1530 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1531 if (wav->streaming) {
1532 const guint8 *data = NULL;
1534 gst_adapter_flush (wav->adapter, 12);
1536 data = gst_adapter_map (wav->adapter, data_size);
1537 gst_wavparse_adtl_chunk (wav, data, data_size);
1538 gst_adapter_unmap (wav->adapter);
1542 gst_buffer_unref (buf);
1546 gst_pad_pull_range (wav->sinkpad, wav->offset,
1547 data_size, &buf)) != GST_FLOW_OK)
1548 goto header_read_error;
1549 gst_buffer_map (buf, &map, GST_MAP_READ);
1550 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1552 gst_buffer_unmap (buf, &map);
1554 wav->offset += GST_ROUND_UP_2 (data_size);
1558 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1559 GST_FOURCC_ARGS (ltag));
1560 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1561 /* need more data */
1567 case GST_RIFF_TAG_cue:{
1568 const guint data_size = size;
1570 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1571 if (wav->streaming) {
1572 const guint8 *data = NULL;
1574 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1577 gst_adapter_flush (wav->adapter, 8);
1579 data = gst_adapter_map (wav->adapter, data_size);
1580 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1581 goto header_read_error;
1583 gst_adapter_unmap (wav->adapter);
1588 gst_buffer_unref (buf);
1591 gst_pad_pull_range (wav->sinkpad, wav->offset,
1592 data_size, &buf)) != GST_FLOW_OK)
1593 goto header_read_error;
1594 gst_buffer_map (buf, &map, GST_MAP_READ);
1595 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1597 goto header_read_error;
1599 gst_buffer_unmap (buf, &map);
1601 size = GST_ROUND_UP_2 (size);
1602 if (wav->streaming) {
1603 gst_adapter_flush (wav->adapter, size);
1605 gst_buffer_unref (buf);
1607 size = GST_ROUND_UP_2 (size);
1608 wav->offset += size;
1611 case GST_RIFF_TAG_smpl:{
1612 const gint data_size = size;
1614 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1615 if (wav->streaming) {
1616 const guint8 *data = NULL;
1618 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1621 gst_adapter_flush (wav->adapter, 8);
1623 data = gst_adapter_map (wav->adapter, data_size);
1624 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1625 goto header_read_error;
1627 gst_adapter_unmap (wav->adapter);
1632 gst_buffer_unref (buf);
1635 gst_pad_pull_range (wav->sinkpad, wav->offset,
1636 data_size, &buf)) != GST_FLOW_OK)
1637 goto header_read_error;
1638 gst_buffer_map (buf, &map, GST_MAP_READ);
1639 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1641 goto header_read_error;
1643 gst_buffer_unmap (buf, &map);
1645 size = GST_ROUND_UP_2 (size);
1646 if (wav->streaming) {
1647 gst_adapter_flush (wav->adapter, size);
1649 gst_buffer_unref (buf);
1651 size = GST_ROUND_UP_2 (size);
1652 wav->offset += size;
1656 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1657 GST_FOURCC_ARGS (tag));
1658 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1659 /* need more data */
1664 if (upstream_size && (wav->offset >= upstream_size)) {
1665 /* Now we are gone through the whole file */
1670 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1672 if (wav->bps <= 0 && wav->fact) {
1674 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1676 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1677 (guint64) wav->fact);
1678 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1683 if (gst_wavparse_calculate_duration (wav)) {
1684 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1685 if (!wav->ignore_length)
1686 wav->segment.duration = wav->duration;
1688 gst_wavparse_create_toc (wav);
1690 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1691 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1692 if (!wav->ignore_length)
1693 wav->segment.duration = wav->datasize;
1696 /* now we have all the info to perform a pending seek if any, if no
1697 * event, this will still do the right thing and it will also send
1698 * the right newsegment event downstream. */
1699 gst_wavparse_perform_seek (wav, wav->seek_event);
1700 /* remove pending event */
1701 gst_event_replace (&wav->seek_event, NULL);
1703 /* we just started, we are discont */
1704 wav->discont = TRUE;
1706 wav->state = GST_WAVPARSE_DATA;
1708 /* determine reasonable max buffer size,
1709 * that is, buffers not too small either size or time wise
1710 * so we do not end up with too many of them */
1712 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1713 wav->max_buf_size = upstream_size;
1715 wav->max_buf_size = 0;
1716 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1717 if (wav->blockalign > 0)
1718 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1720 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1727 g_free (codec_name);
1730 gst_caps_unref (caps);
1735 res = GST_FLOW_ERROR;
1740 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1741 ("Couldn't parse audio header"));
1746 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1747 ("Stream claims to contain no channels - invalid data"));
1752 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1753 ("Stream with sample_rate == 0 - invalid data"));
1758 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1759 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1760 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1765 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1766 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1767 wav->av_bps, wav->blockalign * wav->rate));
1770 no_bytes_per_sample:
1772 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1773 ("Could not caluclate bytes per sample - invalid data"));
1778 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1779 ("No caps found for format 0x%x, %u channels, %u Hz",
1780 wav->format, wav->channels, wav->rate));
1785 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1786 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1792 * Read WAV file tag when streaming
1794 static GstFlowReturn
1795 gst_wavparse_parse_stream_init (GstWavParse * wav)
1797 if (gst_adapter_available (wav->adapter) >= 12) {
1800 /* _take flushes the data */
1801 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1803 GST_DEBUG ("Parsing wav header");
1804 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1805 return GST_FLOW_ERROR;
1808 /* Go to next state */
1809 wav->state = GST_WAVPARSE_HEADER;
1814 /* handle an event sent directly to the element.
1816 * This event can be sent either in the READY state or the
1817 * >READY state. The only event of interest really is the seek
1820 * In the READY state we can only store the event and try to
1821 * respect it when going to PAUSED. We assume we are in the
1822 * READY state when our parsing state != GST_WAVPARSE_DATA.
1824 * When we are steaming, we can simply perform the seek right
1828 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1830 GstWavParse *wav = GST_WAVPARSE (element);
1831 gboolean res = FALSE;
1833 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1835 switch (GST_EVENT_TYPE (event)) {
1836 case GST_EVENT_SEEK:
1837 if (wav->state == GST_WAVPARSE_DATA) {
1838 /* we can handle the seek directly when streaming data */
1839 res = gst_wavparse_perform_seek (wav, event);
1841 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1843 gst_event_replace (&wav->seek_event, event);
1845 /* we always return true */
1852 gst_event_unref (event);
1857 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1861 s = gst_caps_get_structure (caps, 0);
1862 if (!gst_structure_has_name (s, "audio/x-dts"))
1864 /* typefind behavior for DTS:
1865 * MAXIMUM: multiple frame syncs detected, certainly DTS
1866 * LIKELY: single frame sync at offset 0. Maybe DTS?
1867 * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
1869 if (prob > GST_TYPE_FIND_LIKELY)
1871 if (prob <= GST_TYPE_FIND_POSSIBLE)
1873 /* for maybe, check for at least a valid-looking rate and channels */
1874 if (!gst_structure_has_field (s, "channels"))
1876 /* and for extra assurance we could also check the rate from the DTS frame
1877 * against the one in the wav header, but for now let's not do that */
1878 return gst_structure_has_field (s, "rate");
1882 gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
1884 GstTagList *tags = NULL;
1889 while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
1890 gst_event_parse_tag (ev, &tags);
1891 if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
1892 tags = gst_tag_list_copy (tags);
1893 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1894 gst_event_unref (ev);
1898 gst_event_unref (ev);
1904 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1907 GstTagList *tags, *utags;
1909 GST_DEBUG_OBJECT (wav, "adding src pad");
1911 g_assert (wav->caps != NULL);
1913 s = gst_caps_get_structure (wav->caps, 0);
1914 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1915 GstTypeFindProbability prob;
1918 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1919 if (tf_caps != NULL) {
1920 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1921 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1922 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1923 gst_caps_unref (wav->caps);
1924 wav->caps = tf_caps;
1926 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1927 GST_TAG_AUDIO_CODEC, "dts", NULL);
1929 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1930 "marked as raw PCM audio, but ignoring for now", tf_caps);
1931 gst_caps_unref (tf_caps);
1936 gst_pad_set_caps (wav->srcpad, wav->caps);
1938 if (wav->start_segment) {
1939 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1940 gst_pad_push_event (wav->srcpad, wav->start_segment);
1941 wav->start_segment = NULL;
1944 /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
1945 * that there'll be only one scope/type of tag list from upstream, if any */
1946 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
1948 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
1950 /* if there's a tag upstream it's probably been added to override the
1951 * tags from inside the wav header, so keep upstream tags if in doubt */
1952 tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
1954 if (wav->tags != NULL) {
1955 gst_tag_list_unref (wav->tags);
1960 gst_tag_list_unref (utags);
1962 /* send tags downstream, if any */
1964 gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
1967 static GstFlowReturn
1968 gst_wavparse_stream_data (GstWavParse * wav)
1970 GstBuffer *buf = NULL;
1971 GstFlowReturn res = GST_FLOW_OK;
1972 guint64 desired, obtained;
1973 GstClockTime timestamp, next_timestamp, duration;
1974 guint64 pos, nextpos;
1977 GST_LOG_OBJECT (wav,
1978 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1979 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1981 /* Get the next n bytes and output them */
1982 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1985 /* scale the amount of data by the segment rate so we get equal
1986 * amounts of data regardless of the playback rate */
1988 MIN (gst_guint64_to_gdouble (wav->dataleft),
1989 wav->max_buf_size * ABS (wav->segment.rate));
1991 if (desired >= wav->blockalign && wav->blockalign > 0)
1992 desired -= (desired % wav->blockalign);
1994 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1995 "from the sinkpad", desired);
1997 if (wav->streaming) {
1998 guint avail = gst_adapter_available (wav->adapter);
2001 /* flush some bytes if evil upstream sends segment that starts
2002 * before data or does is not send sample aligned segment */
2003 if (G_LIKELY (wav->offset >= wav->datastart)) {
2004 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
2006 extra = wav->datastart - wav->offset;
2009 if (G_UNLIKELY (extra)) {
2010 extra = wav->bytes_per_sample - extra;
2011 if (extra <= avail) {
2012 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2013 gst_adapter_flush (wav->adapter, extra);
2014 wav->offset += extra;
2015 wav->dataleft -= extra;
2016 goto iterate_adapter;
2018 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2019 gst_adapter_clear (wav->adapter);
2020 wav->offset += avail;
2021 wav->dataleft -= avail;
2026 if (avail < desired) {
2027 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2031 buf = gst_adapter_take_buffer (wav->adapter, desired);
2033 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2034 desired, &buf)) != GST_FLOW_OK)
2037 /* we may get a short buffer at the end of the file */
2038 if (gst_buffer_get_size (buf) < desired) {
2039 gsize size = gst_buffer_get_size (buf);
2041 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2042 if (size >= wav->blockalign) {
2043 if (wav->blockalign > 0) {
2044 buf = gst_buffer_make_writable (buf);
2045 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2048 gst_buffer_unref (buf);
2054 obtained = gst_buffer_get_size (buf);
2056 /* our positions in bytes */
2057 pos = wav->offset - wav->datastart;
2058 nextpos = pos + obtained;
2060 /* update offsets, does not overflow. */
2061 buf = gst_buffer_make_writable (buf);
2062 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2063 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2065 /* first chunk of data? create the source pad. We do this only here so
2066 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2067 if (G_UNLIKELY (wav->first)) {
2069 /* this will also push the segment events */
2070 gst_wavparse_add_src_pad (wav, buf);
2072 /* If we have a pending start segment, send it now. */
2073 if (G_UNLIKELY (wav->start_segment != NULL)) {
2074 gst_pad_push_event (wav->srcpad, wav->start_segment);
2075 wav->start_segment = NULL;
2080 /* and timestamps if we have a bitrate, be careful for overflows */
2082 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2084 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2085 duration = next_timestamp - timestamp;
2087 /* update current running segment position */
2088 if (G_LIKELY (next_timestamp >= wav->segment.start))
2089 wav->segment.position = next_timestamp;
2090 } else if (wav->fact) {
2092 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2093 /* and timestamps if we have a bitrate, be careful for overflows */
2094 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2095 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2096 duration = next_timestamp - timestamp;
2098 /* no bitrate, all we know is that the first sample has timestamp 0, all
2099 * other positions and durations have unknown timestamp. */
2103 timestamp = GST_CLOCK_TIME_NONE;
2104 duration = GST_CLOCK_TIME_NONE;
2105 /* update current running segment position with byte offset */
2106 if (G_LIKELY (nextpos >= wav->segment.start))
2107 wav->segment.position = nextpos;
2109 if ((pos > 0) && wav->vbr) {
2110 /* don't set timestamps for VBR files if it's not the first buffer */
2111 timestamp = GST_CLOCK_TIME_NONE;
2112 duration = GST_CLOCK_TIME_NONE;
2115 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2116 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2117 wav->discont = FALSE;
2120 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2121 GST_BUFFER_DURATION (buf) = duration;
2123 GST_LOG_OBJECT (wav,
2124 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2125 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2126 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2128 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2131 if (obtained < wav->dataleft) {
2132 wav->offset += obtained;
2133 wav->dataleft -= obtained;
2135 wav->offset += wav->dataleft;
2139 /* Iterate until need more data, so adapter size won't grow */
2140 if (wav->streaming) {
2141 GST_LOG_OBJECT (wav,
2142 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2144 goto iterate_adapter;
2151 GST_DEBUG_OBJECT (wav, "found EOS");
2152 return GST_FLOW_EOS;
2156 /* check if we got EOS */
2157 if (res == GST_FLOW_EOS)
2160 GST_WARNING_OBJECT (wav,
2161 "Error getting %" G_GINT64_FORMAT " bytes from the "
2162 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2167 GST_INFO_OBJECT (wav,
2168 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2169 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2170 gst_pad_is_linked (wav->srcpad));
2176 gst_wavparse_loop (GstPad * pad)
2179 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2183 GST_LOG_OBJECT (wav, "process data");
2185 switch (wav->state) {
2186 case GST_WAVPARSE_START:
2187 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2188 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2192 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2193 event = gst_event_new_stream_start (stream_id);
2194 gst_event_set_group_id (event, gst_util_group_id_next ());
2195 gst_pad_push_event (wav->srcpad, event);
2198 wav->state = GST_WAVPARSE_HEADER;
2201 case GST_WAVPARSE_HEADER:
2202 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2203 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2206 wav->state = GST_WAVPARSE_DATA;
2207 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2210 case GST_WAVPARSE_DATA:
2211 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2215 g_assert_not_reached ();
2222 const gchar *reason = gst_flow_get_name (ret);
2224 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2225 gst_pad_pause_task (pad);
2227 if (ret == GST_FLOW_EOS) {
2228 /* handle end-of-stream/segment */
2229 /* so align our position with the end of it, if there is one
2230 * this ensures a subsequent will arrive at correct base/acc time */
2231 if (wav->segment.format == GST_FORMAT_TIME) {
2232 if (wav->segment.rate > 0.0 &&
2233 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2234 wav->segment.position = wav->segment.stop;
2235 else if (wav->segment.rate < 0.0)
2236 wav->segment.position = wav->segment.start;
2238 if (wav->state == GST_WAVPARSE_START || !wav->caps) {
2239 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2240 ("No valid input found before end of stream"));
2241 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2243 /* add pad before we perform EOS */
2244 if (G_UNLIKELY (wav->first)) {
2246 gst_wavparse_add_src_pad (wav, NULL);
2249 /* perform EOS logic */
2250 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2253 if ((stop = wav->segment.stop) == -1)
2254 stop = wav->segment.duration;
2256 gst_element_post_message (GST_ELEMENT_CAST (wav),
2257 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2258 wav->segment.format, stop));
2259 gst_pad_push_event (wav->srcpad,
2260 gst_event_new_segment_done (wav->segment.format, stop));
2262 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2265 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2266 /* for fatal errors we post an error message, post the error
2267 * first so the app knows about the error first. */
2268 GST_ELEMENT_FLOW_ERROR (wav, ret);
2269 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2275 static GstFlowReturn
2276 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2279 GstWavParse *wav = GST_WAVPARSE (parent);
2281 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2282 gst_buffer_get_size (buf));
2284 gst_adapter_push (wav->adapter, buf);
2286 switch (wav->state) {
2287 case GST_WAVPARSE_START:
2288 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2289 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2292 if (wav->state != GST_WAVPARSE_HEADER)
2295 /* otherwise fall-through */
2296 case GST_WAVPARSE_HEADER:
2297 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2298 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2301 if (!wav->got_fmt || wav->datastart == 0)
2304 wav->state = GST_WAVPARSE_DATA;
2305 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2308 case GST_WAVPARSE_DATA:
2309 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2310 wav->discont = TRUE;
2311 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2315 g_return_val_if_reached (GST_FLOW_ERROR);
2318 if (G_UNLIKELY (wav->abort_buffering)) {
2319 wav->abort_buffering = FALSE;
2320 ret = GST_FLOW_ERROR;
2321 /* sort of demux/parse error */
2322 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2328 static GstFlowReturn
2329 gst_wavparse_flush_data (GstWavParse * wav)
2331 GstFlowReturn ret = GST_FLOW_OK;
2334 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2335 ret = gst_wavparse_stream_data (wav);
2342 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2344 GstWavParse *wav = GST_WAVPARSE (parent);
2345 gboolean ret = TRUE;
2347 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2349 switch (GST_EVENT_TYPE (event)) {
2350 case GST_EVENT_CAPS:
2352 /* discard, we'll come up with proper src caps */
2353 gst_event_unref (event);
2356 case GST_EVENT_SEGMENT:
2358 gint64 start, stop, offset = 0, end_offset = -1;
2361 /* some debug output */
2362 gst_event_copy_segment (event, &segment);
2363 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2366 if (wav->state != GST_WAVPARSE_DATA) {
2367 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2371 /* now we are either committed to TIME or BYTE format,
2372 * and we only expect a BYTE segment, e.g. following a seek */
2373 if (segment.format == GST_FORMAT_BYTES) {
2374 /* handle (un)signed issues */
2375 start = segment.start;
2376 stop = segment.stop;
2379 start -= wav->datastart;
2380 start = MAX (start, 0);
2384 stop -= wav->datastart;
2385 stop = MAX (stop, 0);
2387 if (wav->segment.format == GST_FORMAT_TIME) {
2388 guint64 bps = wav->bps;
2390 /* operating in format TIME, so we can convert */
2391 if (!bps && wav->fact)
2393 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2397 gst_util_uint64_scale_ceil (start, GST_SECOND,
2398 (guint64) wav->bps);
2401 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2402 (guint64) wav->bps);
2406 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2410 segment.start = start;
2411 segment.stop = stop;
2413 /* accept upstream's notion of segment and distribute along */
2414 segment.format = wav->segment.format;
2415 segment.time = segment.position = segment.start;
2416 segment.duration = wav->segment.duration;
2417 segment.base = gst_segment_to_running_time (&wav->segment,
2418 GST_FORMAT_TIME, wav->segment.position);
2420 gst_segment_copy_into (&segment, &wav->segment);
2422 /* also store the newsegment event for the streaming thread */
2423 if (wav->start_segment)
2424 gst_event_unref (wav->start_segment);
2425 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2426 wav->start_segment = gst_event_new_segment (&segment);
2428 /* stream leftover data in current segment */
2429 gst_wavparse_flush_data (wav);
2430 /* and set up streaming thread for next one */
2431 wav->offset = offset;
2432 wav->end_offset = end_offset;
2434 if (wav->datasize > 0 && (wav->end_offset == -1
2435 || wav->end_offset > wav->datastart + wav->datasize))
2436 wav->end_offset = wav->datastart + wav->datasize;
2438 if (wav->end_offset != -1) {
2439 wav->dataleft = wav->end_offset - wav->offset;
2441 /* infinity; upstream will EOS when done */
2442 wav->dataleft = G_MAXUINT64;
2445 gst_event_unref (event);
2449 if (wav->state == GST_WAVPARSE_START || !wav->caps) {
2450 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2451 ("No valid input found before end of stream"));
2453 /* add pad if needed so EOS is seen downstream */
2454 if (G_UNLIKELY (wav->first)) {
2456 gst_wavparse_add_src_pad (wav, NULL);
2458 /* stream leftover data in current segment */
2459 gst_wavparse_flush_data (wav);
2464 case GST_EVENT_FLUSH_STOP:
2469 gst_adapter_clear (wav->adapter);
2470 wav->discont = TRUE;
2471 dur = wav->segment.duration;
2472 gst_segment_init (&wav->segment, wav->segment.format);
2473 wav->segment.duration = dur;
2477 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2485 /* convert and query stuff */
2486 static const GstFormat *
2487 gst_wavparse_get_formats (GstPad * pad)
2489 static const GstFormat formats[] = {
2492 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2501 gst_wavparse_pad_convert (GstPad * pad,
2502 GstFormat src_format, gint64 src_value,
2503 GstFormat * dest_format, gint64 * dest_value)
2505 GstWavParse *wavparse;
2506 gboolean res = TRUE;
2508 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2510 if (*dest_format == src_format) {
2511 *dest_value = src_value;
2515 if ((wavparse->bps == 0) && !wavparse->fact)
2518 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2519 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2521 switch (src_format) {
2522 case GST_FORMAT_BYTES:
2523 switch (*dest_format) {
2524 case GST_FORMAT_DEFAULT:
2525 *dest_value = src_value / wavparse->bytes_per_sample;
2526 /* make sure we end up on a sample boundary */
2527 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2529 case GST_FORMAT_TIME:
2530 /* src_value + datastart = offset */
2531 GST_INFO_OBJECT (wavparse,
2532 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2534 if (wavparse->bps > 0)
2535 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2536 (guint64) wavparse->bps);
2537 else if (wavparse->fact) {
2538 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2539 wavparse->rate, wavparse->fact);
2542 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2553 case GST_FORMAT_DEFAULT:
2554 switch (*dest_format) {
2555 case GST_FORMAT_BYTES:
2556 *dest_value = src_value * wavparse->bytes_per_sample;
2558 case GST_FORMAT_TIME:
2559 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2560 (guint64) wavparse->rate);
2568 case GST_FORMAT_TIME:
2569 switch (*dest_format) {
2570 case GST_FORMAT_BYTES:
2571 if (wavparse->bps > 0)
2572 *dest_value = gst_util_uint64_scale (src_value,
2573 (guint64) wavparse->bps, GST_SECOND);
2575 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2576 wavparse->rate, wavparse->fact);
2578 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2580 /* make sure we end up on a sample boundary */
2581 *dest_value -= *dest_value % wavparse->blockalign;
2583 case GST_FORMAT_DEFAULT:
2584 *dest_value = gst_util_uint64_scale (src_value,
2585 (guint64) wavparse->rate, GST_SECOND);
2604 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2610 /* handle queries for location and length in requested format */
2612 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2614 gboolean res = TRUE;
2615 GstWavParse *wav = GST_WAVPARSE (parent);
2617 /* only if we know */
2618 if (wav->state != GST_WAVPARSE_DATA) {
2622 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2624 switch (GST_QUERY_TYPE (query)) {
2625 case GST_QUERY_POSITION:
2631 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2632 curb = wav->offset - wav->datastart;
2633 gst_query_parse_position (query, &format, NULL);
2634 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2637 case GST_FORMAT_BYTES:
2638 format = GST_FORMAT_BYTES;
2642 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2647 gst_query_set_position (query, format, cur);
2650 case GST_QUERY_DURATION:
2652 gint64 duration = 0;
2655 if (wav->ignore_length) {
2660 gst_query_parse_duration (query, &format, NULL);
2663 case GST_FORMAT_BYTES:{
2664 format = GST_FORMAT_BYTES;
2665 duration = wav->datasize;
2668 case GST_FORMAT_TIME:
2669 if ((res = gst_wavparse_calculate_duration (wav))) {
2670 duration = wav->duration;
2678 gst_query_set_duration (query, format, duration);
2681 case GST_QUERY_CONVERT:
2683 gint64 srcvalue, dstvalue;
2684 GstFormat srcformat, dstformat;
2686 gst_query_parse_convert (query, &srcformat, &srcvalue,
2687 &dstformat, &dstvalue);
2688 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2689 &dstformat, &dstvalue);
2691 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2694 case GST_QUERY_SEEKING:{
2696 gboolean seekable = FALSE;
2698 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2699 if (fmt == wav->segment.format) {
2700 if (wav->streaming) {
2703 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2704 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2705 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2706 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2708 gst_query_unref (q);
2710 GST_LOG_OBJECT (wav, "looping => seekable");
2714 } else if (fmt == GST_FORMAT_TIME) {
2718 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2723 res = gst_pad_query_default (pad, parent, query);
2730 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2732 GstWavParse *wavparse = GST_WAVPARSE (parent);
2733 gboolean res = FALSE;
2735 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2737 switch (GST_EVENT_TYPE (event)) {
2738 case GST_EVENT_SEEK:
2739 /* can only handle events when we are in the data state */
2740 if (wavparse->state == GST_WAVPARSE_DATA) {
2741 res = gst_wavparse_perform_seek (wavparse, event);
2743 gst_event_unref (event);
2746 case GST_EVENT_TOC_SELECT:
2749 GstTocEntry *entry = NULL;
2750 GstEvent *seek_event;
2753 if (!wavparse->toc) {
2754 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2757 gst_event_parse_toc_select (event, &uid);
2759 GST_OBJECT_LOCK (wavparse);
2760 entry = gst_toc_find_entry (wavparse->toc, uid);
2761 if (entry == NULL) {
2762 GST_OBJECT_UNLOCK (wavparse);
2763 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2767 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2768 GST_OBJECT_UNLOCK (wavparse);
2769 seek_event = gst_event_new_seek (1.0,
2771 GST_SEEK_FLAG_FLUSH,
2772 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2773 res = gst_wavparse_perform_seek (wavparse, seek_event);
2774 gst_event_unref (seek_event);
2778 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2782 gst_event_unref (event);
2787 res = gst_pad_push_event (wavparse->sinkpad, event);
2794 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2796 GstWavParse *wav = GST_WAVPARSE (parent);
2801 gst_adapter_clear (wav->adapter);
2802 g_object_unref (wav->adapter);
2803 wav->adapter = NULL;
2806 query = gst_query_new_scheduling ();
2808 if (!gst_pad_peer_query (sinkpad, query)) {
2809 gst_query_unref (query);
2813 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2814 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2815 gst_query_unref (query);
2820 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2821 wav->streaming = FALSE;
2822 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2826 GST_DEBUG_OBJECT (sinkpad, "activating push");
2827 wav->streaming = TRUE;
2828 wav->adapter = gst_adapter_new ();
2829 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2835 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2836 GstPadMode mode, gboolean active)
2841 case GST_PAD_MODE_PUSH:
2844 case GST_PAD_MODE_PULL:
2846 /* if we have a scheduler we can start the task */
2847 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2850 res = gst_pad_stop_task (sinkpad);
2860 static GstStateChangeReturn
2861 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2863 GstStateChangeReturn ret;
2864 GstWavParse *wav = GST_WAVPARSE (element);
2866 switch (transition) {
2867 case GST_STATE_CHANGE_NULL_TO_READY:
2869 case GST_STATE_CHANGE_READY_TO_PAUSED:
2870 gst_wavparse_reset (wav);
2872 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2878 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2880 switch (transition) {
2881 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2883 case GST_STATE_CHANGE_PAUSED_TO_READY:
2884 gst_wavparse_reset (wav);
2886 case GST_STATE_CHANGE_READY_TO_NULL:
2895 gst_wavparse_set_property (GObject * object, guint prop_id,
2896 const GValue * value, GParamSpec * pspec)
2900 g_return_if_fail (GST_IS_WAVPARSE (object));
2901 self = GST_WAVPARSE (object);
2904 case PROP_IGNORE_LENGTH:
2905 self->ignore_length = g_value_get_boolean (value);
2908 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2914 gst_wavparse_get_property (GObject * object, guint prop_id,
2915 GValue * value, GParamSpec * pspec)
2919 g_return_if_fail (GST_IS_WAVPARSE (object));
2920 self = GST_WAVPARSE (object);
2923 case PROP_IGNORE_LENGTH:
2924 g_value_set_boolean (value, self->ignore_length);
2927 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2932 plugin_init (GstPlugin * plugin)
2936 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2940 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2943 "Parse a .wav file into raw audio",
2944 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)