1 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
3 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
4 * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-wavparse
25 * Parse a .wav file into raw or compressed audio.
27 * Wavparse supports both push and pull mode operations, making it possible to
28 * stream from a network source.
31 * <title>Example launch line</title>
33 * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
34 * ]| Read a wav file and output to the soundcard using the ALSA element. The
35 * wav file is assumed to contain raw uncompressed samples.
37 * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
38 * ]| Stream data from a network url.
44 * http://replaygain.hydrogenaudio.org/file_format_wav.html
54 #include "gstwavparse.h"
55 #include "gst/riff/riff-media.h"
56 #include <gst/base/gsttypefindhelper.h>
57 #include <gst/gst-i18n-plugin.h>
59 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
60 #define GST_CAT_DEFAULT (wavparse_debug)
62 #define GST_BWF_TAG_iXML GST_MAKE_FOURCC ('i','X','M','L')
63 #define GST_BWF_TAG_qlty GST_MAKE_FOURCC ('q','l','t','y')
64 #define GST_BWF_TAG_mext GST_MAKE_FOURCC ('m','e','x','t')
65 #define GST_BWF_TAG_levl GST_MAKE_FOURCC ('l','e','v','l')
66 #define GST_BWF_TAG_link GST_MAKE_FOURCC ('l','i','n','k')
67 #define GST_BWF_TAG_axml GST_MAKE_FOURCC ('a','x','m','l')
69 static void gst_wavparse_dispose (GObject * object);
71 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
73 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
74 GstObject * parent, GstPadMode mode, gboolean active);
75 static gboolean gst_wavparse_send_event (GstElement * element,
77 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
78 GstStateChange transition);
80 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
82 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
83 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
85 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
87 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
89 static void gst_wavparse_loop (GstPad * pad);
90 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
93 static void gst_wavparse_set_property (GObject * object, guint prop_id,
94 const GValue * value, GParamSpec * pspec);
95 static void gst_wavparse_get_property (GObject * object, guint prop_id,
96 GValue * value, GParamSpec * pspec);
98 #define DEFAULT_IGNORE_LENGTH FALSE
106 static GstStaticPadTemplate sink_template_factory =
107 GST_STATIC_PAD_TEMPLATE ("sink",
110 GST_STATIC_CAPS ("audio/x-wav")
114 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
116 #define gst_wavparse_parent_class parent_class
117 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
122 /* Offset Size Description Value
123 * 0x00 4 ID unique identification value
124 * 0x04 4 Position play order position
125 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
126 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
127 * 0x10 4 Block Start Byte Offset to sample of First Channel
128 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
132 guint32 data_chunk_id;
135 guint32 sample_offset;
140 /* Offset Size Description Value
141 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
144 guint32 cue_point_id;
146 } GstWavParseLabl, GstWavParseNote;
149 gst_wavparse_class_init (GstWavParseClass * klass)
151 GstElementClass *gstelement_class;
152 GObjectClass *object_class;
153 GstPadTemplate *src_template;
155 gstelement_class = (GstElementClass *) klass;
156 object_class = (GObjectClass *) klass;
158 parent_class = g_type_class_peek_parent (klass);
160 object_class->dispose = gst_wavparse_dispose;
162 object_class->set_property = gst_wavparse_set_property;
163 object_class->get_property = gst_wavparse_get_property;
166 * GstWavParse:ignore-length:
168 * This selects whether the length found in a data chunk
169 * should be ignored. This may be useful for streamed audio
170 * where the length is unknown until the end of streaming,
171 * and various software/hardware just puts some random value
172 * in there and hopes it doesn't break too much.
174 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
175 g_param_spec_boolean ("ignore-length",
177 "Ignore length from the Wave header",
178 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
181 gstelement_class->change_state = gst_wavparse_change_state;
182 gstelement_class->send_event = gst_wavparse_send_event;
185 gst_element_class_add_pad_template (gstelement_class,
186 gst_static_pad_template_get (&sink_template_factory));
188 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
189 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
190 gst_element_class_add_pad_template (gstelement_class, src_template);
192 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
193 "Codec/Demuxer/Audio",
194 "Parse a .wav file into raw audio",
195 "Erik Walthinsen <omega@cse.ogi.edu>");
199 gst_wavparse_reset (GstWavParse * wav)
201 wav->state = GST_WAVPARSE_START;
203 /* These will all be set correctly in the fmt chunk */
217 wav->got_fmt = FALSE;
221 gst_event_unref (wav->seek_event);
222 wav->seek_event = NULL;
224 gst_adapter_clear (wav->adapter);
225 g_object_unref (wav->adapter);
229 gst_tag_list_unref (wav->tags);
232 gst_toc_unref (wav->toc);
235 g_list_free_full (wav->cues, g_free);
238 g_list_free_full (wav->labls, g_free);
241 gst_caps_unref (wav->caps);
243 if (wav->start_segment)
244 gst_event_unref (wav->start_segment);
245 wav->start_segment = NULL;
249 gst_wavparse_dispose (GObject * object)
251 GstWavParse *wav = GST_WAVPARSE (object);
253 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
254 gst_wavparse_reset (wav);
256 G_OBJECT_CLASS (parent_class)->dispose (object);
260 gst_wavparse_init (GstWavParse * wavparse)
262 gst_wavparse_reset (wavparse);
266 gst_pad_new_from_static_template (&sink_template_factory, "sink");
267 gst_pad_set_activate_function (wavparse->sinkpad,
268 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
269 gst_pad_set_activatemode_function (wavparse->sinkpad,
270 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
271 gst_pad_set_chain_function (wavparse->sinkpad,
272 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
273 gst_pad_set_event_function (wavparse->sinkpad,
274 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
275 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
279 gst_pad_new_from_template (gst_element_class_get_pad_template
280 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
281 gst_pad_use_fixed_caps (wavparse->srcpad);
282 gst_pad_set_query_function (wavparse->srcpad,
283 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
284 gst_pad_set_event_function (wavparse->srcpad,
285 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
286 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
290 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
294 if (!gst_riff_parse_file_header (element, buf, &doctype))
297 if (doctype != GST_RIFF_RIFF_WAVE)
305 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
306 ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
307 GST_FOURCC_ARGS (doctype)));
313 gst_wavparse_stream_init (GstWavParse * wav)
316 GstBuffer *buf = NULL;
318 if ((res = gst_pad_pull_range (wav->sinkpad,
319 wav->offset, 12, &buf)) != GST_FLOW_OK)
321 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
322 return GST_FLOW_ERROR;
330 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
332 /* -1 always maps to -1 */
338 /* 0 always maps to 0 */
345 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
347 } else if (wav->fact) {
349 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
350 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
357 /* This function is used to perform seeks on the element.
359 * It also works when event is NULL, in which case it will just
360 * start from the last configured segment. This technique is
361 * used when activating the element and to perform the seek in
365 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
369 GstFormat format, bformat;
371 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
372 gint64 cur, stop, upstream_size;
375 GstSegment seeksegment = { 0, };
379 GST_DEBUG_OBJECT (wav, "doing seek with event");
381 gst_event_parse_seek (event, &rate, &format, &flags,
382 &cur_type, &cur, &stop_type, &stop);
384 /* no negative rates yet */
388 if (format != wav->segment.format) {
389 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
390 gst_format_get_name (format),
391 gst_format_get_name (wav->segment.format));
393 if (cur_type != GST_SEEK_TYPE_NONE)
395 gst_pad_query_convert (wav->srcpad, format, cur,
396 wav->segment.format, &cur);
397 if (res && stop_type != GST_SEEK_TYPE_NONE)
399 gst_pad_query_convert (wav->srcpad, format, stop,
400 wav->segment.format, &stop);
404 format = wav->segment.format;
407 GST_DEBUG_OBJECT (wav, "doing seek without event");
410 cur_type = GST_SEEK_TYPE_SET;
411 stop_type = GST_SEEK_TYPE_SET;
414 /* in push mode, we must delegate to upstream */
415 if (wav->streaming) {
416 gboolean res = FALSE;
418 /* if streaming not yet started; only prepare initial newsegment */
419 if (!event || wav->state != GST_WAVPARSE_DATA) {
420 if (wav->start_segment)
421 gst_event_unref (wav->start_segment);
422 wav->start_segment = gst_event_new_segment (&wav->segment);
425 /* convert seek positions to byte positions in data sections */
426 if (format == GST_FORMAT_TIME) {
427 /* should not fail */
428 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
430 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
433 /* mind sample boundary and header */
435 cur -= (cur % wav->bytes_per_sample);
436 cur += wav->datastart;
439 stop -= (stop % wav->bytes_per_sample);
440 stop += wav->datastart;
442 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
443 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
445 /* BYTE seek event */
446 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
448 res = gst_pad_push_event (wav->sinkpad, event);
454 flush = flags & GST_SEEK_FLAG_FLUSH;
456 /* now we need to make sure the streaming thread is stopped. We do this by
457 * either sending a FLUSH_START event downstream which will cause the
458 * streaming thread to stop with a WRONG_STATE.
459 * For a non-flushing seek we simply pause the task, which will happen as soon
460 * as it completes one iteration (and thus might block when the sink is
461 * blocking in preroll). */
463 GST_DEBUG_OBJECT (wav, "sending flush start");
464 gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
466 gst_pad_pause_task (wav->sinkpad);
469 /* we should now be able to grab the streaming thread because we stopped it
470 * with the above flush/pause code */
471 GST_PAD_STREAM_LOCK (wav->sinkpad);
473 /* save current position */
474 last_stop = wav->segment.position;
476 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
478 /* copy segment, we need this because we still need the old
479 * segment when we close the current segment. */
480 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
482 /* configure the seek parameters in the seeksegment. We will then have the
483 * right values in the segment to perform the seek */
485 GST_DEBUG_OBJECT (wav, "configuring seek");
486 gst_segment_do_seek (&seeksegment, rate, format, flags,
487 cur_type, cur, stop_type, stop, &update);
490 /* figure out the last position we need to play. If it's configured (stop !=
491 * -1), use that, else we play until the total duration of the file */
492 if ((stop = seeksegment.stop) == -1)
493 stop = seeksegment.duration;
495 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
496 if ((cur_type != GST_SEEK_TYPE_NONE)) {
497 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
498 * we can just copy the last_stop. If not, we use the bps to convert TIME to
500 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
501 (gint64 *) & wav->offset))
502 wav->offset = seeksegment.position;
503 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
504 wav->offset -= (wav->offset % wav->bytes_per_sample);
505 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
506 wav->offset += wav->datastart;
507 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
509 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
513 if (stop_type != GST_SEEK_TYPE_NONE) {
514 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
515 wav->end_offset = stop;
516 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
517 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
518 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
519 wav->end_offset += wav->datastart;
520 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
522 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
526 /* make sure filesize is not exceeded due to rounding errors or so,
527 * same precaution as in _stream_headers */
528 bformat = GST_FORMAT_BYTES;
529 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
530 wav->end_offset = MIN (wav->end_offset, upstream_size);
532 /* this is the range of bytes we will use for playback */
533 wav->offset = MIN (wav->offset, wav->end_offset);
534 wav->dataleft = wav->end_offset - wav->offset;
536 GST_DEBUG_OBJECT (wav,
537 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
538 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
539 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
541 /* prepare for streaming again */
543 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
544 GST_DEBUG_OBJECT (wav, "sending flush stop");
545 gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
548 /* now we did the seek and can activate the new segment values */
549 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
551 /* if we're doing a segment seek, post a SEGMENT_START message */
552 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
553 gst_element_post_message (GST_ELEMENT_CAST (wav),
554 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
555 wav->segment.format, wav->segment.position));
558 /* now create the newsegment */
559 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
560 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
562 /* store the newsegment event so it can be sent from the streaming thread. */
563 if (wav->start_segment)
564 gst_event_unref (wav->start_segment);
565 wav->start_segment = gst_event_new_segment (&wav->segment);
567 /* mark discont if we are going to stream from another position. */
568 if (last_stop != wav->segment.position) {
569 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
573 /* and start the streaming task again */
574 if (!wav->streaming) {
575 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
579 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
586 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
591 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
596 GST_DEBUG_OBJECT (wav,
597 "Could not determine byte position for desired time");
603 * gst_wavparse_peek_chunk_info:
604 * @wav Wavparse object
605 * @tag holder for tag
606 * @size holder for tag size
608 * Peek next chunk info (tag and size)
610 * Returns: %TRUE when the chunk info (header) is available
613 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
615 const guint8 *data = NULL;
617 if (gst_adapter_available (wav->adapter) < 8)
620 data = gst_adapter_map (wav->adapter, 8);
621 *tag = GST_READ_UINT32_LE (data);
622 *size = GST_READ_UINT32_LE (data + 4);
623 gst_adapter_unmap (wav->adapter);
625 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
626 GST_FOURCC_ARGS (*tag));
632 * gst_wavparse_peek_chunk:
633 * @wav Wavparse object
634 * @tag holder for tag
635 * @size holder for tag size
637 * Peek enough data for one full chunk
639 * Returns: %TRUE when the full chunk is available
642 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
644 guint32 peek_size = 0;
647 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
650 /* size 0 -> empty data buffer would surprise most callers,
651 * large size -> do not bother trying to squeeze that into adapter,
652 * so we throw poor man's exception, which can be caught if caller really
653 * wants to handle 0 size chunk */
654 if (!(*size) || (*size) >= (1 << 30)) {
655 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
656 *size, GST_FOURCC_ARGS (*tag));
657 /* chain should give up */
658 wav->abort_buffering = TRUE;
661 peek_size = (*size + 1) & ~1;
662 available = gst_adapter_available (wav->adapter);
664 if (available >= (8 + peek_size)) {
667 GST_LOG ("but only %u bytes available now", available);
673 * gst_wavparse_calculate_duration:
674 * @wav: wavparse object
676 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
679 * Returns: %TRUE if duration is available.
682 gst_wavparse_calculate_duration (GstWavParse * wav)
684 if (wav->duration > 0)
688 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
690 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
692 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
693 GST_TIME_ARGS (wav->duration));
695 } else if (wav->fact) {
697 gst_util_uint64_scale_int_ceil (GST_SECOND, wav->fact, wav->rate);
698 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
699 GST_TIME_ARGS (wav->duration));
706 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
711 if (wav->streaming) {
712 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
715 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
716 GST_FOURCC_ARGS (tag));
717 flush = 8 + ((size + 1) & ~1);
718 wav->offset += flush;
719 if (wav->streaming) {
720 gst_adapter_flush (wav->adapter, flush);
722 gst_buffer_unref (buf);
729 * gst_wavparse_cue_chunk:
730 * @wav GstWavParse object
731 * @data holder for data
732 * @size holder for data size
734 * Parse cue chunk from @data to wav->cues.
736 * Returns: %TRUE when cue chunk is available
739 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
746 GST_WARNING_OBJECT (wav, "found another cue's");
750 ncues = GST_READ_UINT32_LE (data);
752 if (size < 4 + ncues * 24) {
753 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
759 for (i = 0; i < ncues; i++) {
760 cue = g_new0 (GstWavParseCue, 1);
761 cue->id = GST_READ_UINT32_LE (data);
762 cue->position = GST_READ_UINT32_LE (data + 4);
763 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
764 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
765 cue->block_start = GST_READ_UINT32_LE (data + 16);
766 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
767 cues = g_list_append (cues, cue);
777 * gst_wavparse_labl_chunk:
778 * @wav GstWavParse object
779 * @data holder for data
780 * @size holder for data size
782 * Parse labl from @data to wav->labls.
784 * Returns: %TRUE when labl chunk is available
787 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
789 GstWavParseLabl *labl;
794 labl = g_new0 (GstWavParseLabl, 1);
798 labl->cue_point_id = GST_READ_UINT32_LE (data);
799 labl->text = g_memdup (data + 4, size - 4);
801 wav->labls = g_list_append (wav->labls, labl);
807 * gst_wavparse_note_chunk:
808 * @wav GstWavParse object
809 * @data holder for data
810 * @size holder for data size
812 * Parse note from @data to wav->notes.
814 * Returns: %TRUE when note chunk is available
817 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
819 GstWavParseNote *note;
824 note = g_new0 (GstWavParseNote, 1);
828 note->cue_point_id = GST_READ_UINT32_LE (data);
829 note->text = g_memdup (data + 4, size - 4);
831 wav->notes = g_list_append (wav->notes, note);
837 * gst_wavparse_smpl_chunk:
838 * @wav GstWavParse object
839 * @data holder for data
840 * @size holder for data size
842 * Parse smpl chunk from @data.
844 * Returns: %TRUE when cue chunk is available
847 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
852 manufacturer_id = GST_READ_UINT32_LE (data);
853 product_id = GST_READ_UINT32_LE (data + 4);
854 sample_period = GST_READ_UINT32_LE (data + 8);
856 note_number = GST_READ_UINT32_LE (data + 12);
858 pitch_fraction = GST_READ_UINT32_LE (data + 16);
859 SMPTE_format = GST_READ_UINT32_LE (data + 20);
860 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
861 num_sample_loops = GST_READ_UINT32_LE (data + 28);
862 List of Sample Loops, 24 bytes each
866 wav->tags = gst_tag_list_new_empty ();
867 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
868 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
873 * gst_wavparse_adtl_chunk:
874 * @wav GstWavParse object
875 * @data holder for data
876 * @size holder for data size
878 * Parse adtl from @data.
880 * Returns: %TRUE when adtl chunk is available
883 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
885 guint32 ltag, lsize, offset = 0;
888 ltag = GST_READ_UINT32_LE (data + offset);
889 lsize = GST_READ_UINT32_LE (data + offset + 4);
891 case GST_RIFF_TAG_labl:
892 gst_wavparse_labl_chunk (wav, data + offset, size);
894 case GST_RIFF_TAG_note:
895 gst_wavparse_note_chunk (wav, data + offset, size);
898 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
899 GST_FOURCC_ARGS (ltag));
900 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
903 offset += 8 + GST_ROUND_UP_2 (lsize);
904 size -= 8 + GST_ROUND_UP_2 (lsize);
911 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
913 GstTagList *tags = NULL;
914 GstTocEntry *entry = NULL;
916 entry = gst_toc_find_entry (toc, id);
918 tags = gst_toc_entry_get_tags (entry);
920 tags = gst_tag_list_new_empty ();
921 gst_toc_entry_set_tags (entry, tags);
929 * gst_wavparse_create_toc:
930 * @wav GstWavParse object
932 * Create TOC from wav->cues and wav->labls.
935 gst_wavparse_create_toc (GstWavParse * wav)
941 GstWavParseLabl *labl;
942 GstWavParseNote *note;
945 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
947 GST_OBJECT_LOCK (wav);
949 GST_OBJECT_UNLOCK (wav);
950 GST_WARNING_OBJECT (wav, "found another TOC");
955 GST_OBJECT_UNLOCK (wav);
959 /* FIXME: send CURRENT scope toc too */
960 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
962 /* add cue edition */
963 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
964 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
965 gst_toc_append_entry (toc, entry);
967 /* add tracks in cue edition */
971 prev_subentry = cur_subentry;
972 /* previous track stop time = current track start time */
973 if (prev_subentry != NULL) {
974 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
975 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
976 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
978 id = g_strdup_printf ("%08x", cue->id);
979 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
981 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
982 stop = wav->duration;
983 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
984 gst_toc_entry_append_sub_entry (entry, cur_subentry);
985 list = g_list_next (list);
988 /* add tags in tracks */
992 id = g_strdup_printf ("%08x", labl->cue_point_id);
993 tags = gst_wavparse_get_tags_toc_entry (toc, id);
996 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
999 list = g_list_next (list);
1004 id = g_strdup_printf ("%08x", note->cue_point_id);
1005 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1008 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1011 list = g_list_next (list);
1014 /* send data as TOC */
1017 /* send TOC event */
1019 GST_OBJECT_UNLOCK (wav);
1020 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1026 #define MAX_BUFFER_SIZE 4096
1028 static GstFlowReturn
1029 gst_wavparse_stream_headers (GstWavParse * wav)
1031 GstFlowReturn res = GST_FLOW_OK;
1032 GstBuffer *buf = NULL;
1033 gst_riff_strf_auds *header = NULL;
1035 gboolean gotdata = FALSE;
1036 GstCaps *caps = NULL;
1037 gchar *codec_name = NULL;
1039 gint64 upstream_size = 0;
1041 /* search for "_fmt" chunk, which should be first */
1042 while (!wav->got_fmt) {
1045 /* The header starts with a 'fmt ' tag */
1046 if (wav->streaming) {
1047 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1050 gst_adapter_flush (wav->adapter, 8);
1054 buf = gst_adapter_take_buffer (wav->adapter, size);
1056 gst_adapter_flush (wav->adapter, 1);
1057 wav->offset += GST_ROUND_UP_2 (size);
1059 buf = gst_buffer_new ();
1062 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1063 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1067 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1068 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1069 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1070 tag == GST_RIFF_TAG_id3 || tag == GST_RIFF_TAG_IDVX ||
1071 tag == GST_BWF_TAG_iXML || tag == GST_BWF_TAG_qlty ||
1072 tag == GST_BWF_TAG_mext || tag == GST_BWF_TAG_levl ||
1073 tag == GST_BWF_TAG_link || tag == GST_BWF_TAG_axml) {
1074 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1075 GST_FOURCC_ARGS (tag));
1076 gst_buffer_unref (buf);
1081 if (tag != GST_RIFF_TAG_fmt)
1084 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1086 goto parse_header_error;
1088 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1090 /* do sanity checks of header fields */
1091 if (header->channels == 0)
1093 if (header->rate == 0)
1096 GST_DEBUG_OBJECT (wav, "creating the caps");
1098 /* Note: gst_riff_create_audio_caps might need to fix values in
1099 * the header header depending on the format, so call it first */
1100 /* FIXME: Need to handle the channel reorder map */
1101 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1102 NULL, &codec_name, NULL);
1105 gst_buffer_unref (extra);
1108 goto unknown_format;
1110 /* do more sanity checks of header fields
1111 * (these can be sanitized by gst_riff_create_audio_caps()
1113 wav->format = header->format;
1114 wav->rate = header->rate;
1115 wav->channels = header->channels;
1116 wav->blockalign = header->blockalign;
1117 wav->depth = header->bits_per_sample;
1118 wav->av_bps = header->av_bps;
1124 /* do format specific handling */
1125 switch (wav->format) {
1126 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1127 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1129 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1130 * bitrate inside the mpeg stream */
1131 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1135 case GST_RIFF_WAVE_FORMAT_PCM:
1136 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1137 goto invalid_blockalign;
1140 if (wav->av_bps > wav->blockalign * wav->rate)
1142 /* use the configured bps */
1143 wav->bps = wav->av_bps;
1147 wav->width = (wav->blockalign * 8) / wav->channels;
1148 wav->bytes_per_sample = wav->channels * wav->width / 8;
1150 if (wav->bytes_per_sample <= 0)
1151 goto no_bytes_per_sample;
1153 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1154 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1155 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1156 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1157 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1158 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1159 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1161 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1162 * formats). This will make the element output a BYTE format segment and
1163 * will not timestamp the outgoing buffers.
1165 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1167 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1169 /* create pad later so we can sniff the first few bytes
1170 * of the real data and correct our caps if necessary */
1171 gst_caps_replace (&wav->caps, caps);
1172 gst_caps_replace (&caps, NULL);
1174 wav->got_fmt = TRUE;
1177 wav->tags = gst_tag_list_new_empty ();
1179 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1180 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1182 g_free (codec_name);
1188 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1189 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1191 /* loop headers until we get data */
1193 if (wav->streaming) {
1194 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1201 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1202 &buf)) != GST_FLOW_OK)
1203 goto header_read_error;
1204 gst_buffer_map (buf, &map, GST_MAP_READ);
1205 tag = GST_READ_UINT32_LE (map.data);
1206 size = GST_READ_UINT32_LE (map.data + 4);
1207 gst_buffer_unmap (buf, &map);
1210 GST_INFO_OBJECT (wav,
1211 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1212 GST_FOURCC_ARGS (tag), wav->offset);
1214 /* wav is a st00pid format, we don't know for sure where data starts.
1215 * So we have to go bit by bit until we find the 'data' header
1218 case GST_RIFF_TAG_data:{
1219 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1220 if (wav->ignore_length) {
1221 GST_DEBUG_OBJECT (wav, "Ignoring length");
1224 if (wav->streaming) {
1225 gst_adapter_flush (wav->adapter, 8);
1228 gst_buffer_unref (buf);
1231 wav->datastart = wav->offset;
1232 /* If size is zero, then the data chunk probably actually extends to
1233 the end of the file */
1234 if (size == 0 && upstream_size) {
1235 size = upstream_size - wav->datastart;
1237 /* Or the file might be truncated */
1238 else if (upstream_size) {
1239 size = MIN (size, (upstream_size - wav->datastart));
1241 wav->datasize = (guint64) size;
1242 wav->dataleft = (guint64) size;
1243 wav->end_offset = size + wav->datastart;
1244 if (!wav->streaming) {
1245 /* We will continue parsing tags 'till end */
1246 wav->offset += size;
1248 GST_DEBUG_OBJECT (wav, "datasize = %u", size);
1251 case GST_RIFF_TAG_fact:{
1252 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1253 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1254 const guint data_size = 4;
1256 GST_INFO_OBJECT (wav, "Have fact chunk");
1257 if (size < data_size) {
1258 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1259 /* need more data */
1262 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1266 /* number of samples (for compressed formats) */
1267 if (wav->streaming) {
1268 const guint8 *data = NULL;
1270 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1273 gst_adapter_flush (wav->adapter, 8);
1274 data = gst_adapter_map (wav->adapter, data_size);
1275 wav->fact = GST_READ_UINT32_LE (data);
1276 gst_adapter_unmap (wav->adapter);
1277 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1279 gst_buffer_unref (buf);
1282 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1283 data_size, &buf)) != GST_FLOW_OK)
1284 goto header_read_error;
1285 gst_buffer_extract (buf, 0, &wav->fact, 4);
1286 wav->fact = GUINT32_FROM_LE (wav->fact);
1287 gst_buffer_unref (buf);
1289 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1290 wav->offset += 8 + GST_ROUND_UP_2 (size);
1293 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1294 /* need more data */
1300 case GST_RIFF_TAG_acid:{
1301 const gst_riff_acid *acid = NULL;
1302 const guint data_size = sizeof (gst_riff_acid);
1305 GST_INFO_OBJECT (wav, "Have acid chunk");
1306 if (size < data_size) {
1307 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1308 /* need more data */
1311 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1315 if (wav->streaming) {
1316 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1319 gst_adapter_flush (wav->adapter, 8);
1320 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1322 tempo = acid->tempo;
1323 gst_adapter_unmap (wav->adapter);
1326 gst_buffer_unref (buf);
1329 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1330 size, &buf)) != GST_FLOW_OK)
1331 goto header_read_error;
1332 gst_buffer_map (buf, &map, GST_MAP_READ);
1333 acid = (const gst_riff_acid *) map.data;
1334 tempo = acid->tempo;
1335 gst_buffer_unmap (buf, &map);
1337 /* send data as tags */
1339 wav->tags = gst_tag_list_new_empty ();
1340 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1341 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1343 size = GST_ROUND_UP_2 (size);
1344 if (wav->streaming) {
1345 gst_adapter_flush (wav->adapter, size);
1347 gst_buffer_unref (buf);
1349 wav->offset += 8 + size;
1352 /* FIXME: all list tags after data are ignored in streaming mode */
1353 case GST_RIFF_TAG_LIST:{
1356 if (wav->streaming) {
1357 const guint8 *data = NULL;
1359 if (gst_adapter_available (wav->adapter) < 12) {
1362 data = gst_adapter_map (wav->adapter, 12);
1363 ltag = GST_READ_UINT32_LE (data + 8);
1364 gst_adapter_unmap (wav->adapter);
1366 gst_buffer_unref (buf);
1369 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1370 &buf)) != GST_FLOW_OK)
1371 goto header_read_error;
1372 gst_buffer_extract (buf, 8, <ag, 4);
1373 ltag = GUINT32_FROM_LE (ltag);
1376 case GST_RIFF_LIST_INFO:{
1377 const gint data_size = size - 4;
1380 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1381 if (wav->streaming) {
1382 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1385 gst_adapter_flush (wav->adapter, 12);
1387 if (data_size > 0) {
1388 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1390 gst_adapter_flush (wav->adapter, 1);
1394 gst_buffer_unref (buf);
1396 if (data_size > 0) {
1398 gst_pad_pull_range (wav->sinkpad, wav->offset,
1399 data_size, &buf)) != GST_FLOW_OK)
1400 goto header_read_error;
1403 if (data_size > 0) {
1405 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1407 GstTagList *old = wav->tags;
1409 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1411 gst_tag_list_unref (old);
1412 gst_tag_list_unref (new);
1414 gst_buffer_unref (buf);
1415 wav->offset += GST_ROUND_UP_2 (data_size);
1419 case GST_RIFF_LIST_adtl:{
1420 const gint data_size = size;
1422 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1423 if (wav->streaming) {
1424 const guint8 *data = NULL;
1426 gst_adapter_flush (wav->adapter, 12);
1427 data = gst_adapter_map (wav->adapter, data_size);
1428 gst_wavparse_adtl_chunk (wav, data, data_size);
1429 gst_adapter_unmap (wav->adapter);
1433 gst_buffer_unref (buf);
1436 gst_pad_pull_range (wav->sinkpad, wav->offset + 12,
1437 data_size, &buf)) != GST_FLOW_OK)
1438 goto header_read_error;
1439 gst_buffer_map (buf, &map, GST_MAP_READ);
1440 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1442 gst_buffer_unmap (buf, &map);
1447 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1448 GST_FOURCC_ARGS (ltag));
1449 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1450 /* need more data */
1456 case GST_RIFF_TAG_cue:{
1457 const guint data_size = size;
1459 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1460 if (wav->streaming) {
1461 const guint8 *data = NULL;
1463 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1466 gst_adapter_flush (wav->adapter, 8);
1468 data = gst_adapter_map (wav->adapter, data_size);
1469 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1470 goto header_read_error;
1472 gst_adapter_unmap (wav->adapter);
1477 gst_buffer_unref (buf);
1480 gst_pad_pull_range (wav->sinkpad, wav->offset,
1481 data_size, &buf)) != GST_FLOW_OK)
1482 goto header_read_error;
1483 gst_buffer_map (buf, &map, GST_MAP_READ);
1484 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1486 goto header_read_error;
1488 gst_buffer_unmap (buf, &map);
1490 size = GST_ROUND_UP_2 (size);
1491 if (wav->streaming) {
1492 gst_adapter_flush (wav->adapter, size);
1494 gst_buffer_unref (buf);
1496 size = GST_ROUND_UP_2 (size);
1497 wav->offset += size;
1500 case GST_RIFF_TAG_smpl:{
1501 const gint data_size = size;
1503 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1504 if (wav->streaming) {
1505 const guint8 *data = NULL;
1507 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1510 gst_adapter_flush (wav->adapter, 8);
1512 data = gst_adapter_map (wav->adapter, data_size);
1513 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1514 goto header_read_error;
1516 gst_adapter_unmap (wav->adapter);
1521 gst_buffer_unref (buf);
1524 gst_pad_pull_range (wav->sinkpad, wav->offset,
1525 data_size, &buf)) != GST_FLOW_OK)
1526 goto header_read_error;
1527 gst_buffer_map (buf, &map, GST_MAP_READ);
1528 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1530 goto header_read_error;
1532 gst_buffer_unmap (buf, &map);
1534 size = GST_ROUND_UP_2 (size);
1535 if (wav->streaming) {
1536 gst_adapter_flush (wav->adapter, size);
1538 gst_buffer_unref (buf);
1540 size = GST_ROUND_UP_2 (size);
1541 wav->offset += size;
1545 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1546 GST_FOURCC_ARGS (tag));
1547 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1548 /* need more data */
1553 if (upstream_size && (wav->offset >= upstream_size)) {
1554 /* Now we are gone through the whole file */
1559 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1561 if (wav->bps <= 0 && wav->fact) {
1563 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1565 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1566 (guint64) wav->fact);
1567 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1572 if (gst_wavparse_calculate_duration (wav)) {
1573 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1574 if (!wav->ignore_length)
1575 wav->segment.duration = wav->duration;
1577 gst_wavparse_create_toc (wav);
1579 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1580 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1581 if (!wav->ignore_length)
1582 wav->segment.duration = wav->datasize;
1585 /* now we have all the info to perform a pending seek if any, if no
1586 * event, this will still do the right thing and it will also send
1587 * the right newsegment event downstream. */
1588 gst_wavparse_perform_seek (wav, wav->seek_event);
1589 /* remove pending event */
1590 event_p = &wav->seek_event;
1591 gst_event_replace (event_p, NULL);
1593 /* we just started, we are discont */
1594 wav->discont = TRUE;
1596 wav->state = GST_WAVPARSE_DATA;
1598 /* determine reasonable max buffer size,
1599 * that is, buffers not too small either size or time wise
1600 * so we do not end up with too many of them */
1602 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1603 wav->max_buf_size = upstream_size;
1605 wav->max_buf_size = 0;
1606 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1607 if (wav->blockalign > 0)
1608 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1610 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1618 g_free (codec_name);
1622 gst_caps_unref (caps);
1627 res = GST_FLOW_ERROR;
1632 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1633 ("Invalid WAV header (no fmt at start): %"
1634 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1639 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1640 ("Couldn't parse audio header"));
1645 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1646 ("Stream claims to contain no channels - invalid data"));
1651 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1652 ("Stream with sample_rate == 0 - invalid data"));
1657 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1658 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1659 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1664 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1665 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1666 wav->av_bps, wav->blockalign * wav->rate));
1669 no_bytes_per_sample:
1671 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1672 ("Could not caluclate bytes per sample - invalid data"));
1677 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1678 ("No caps found for format 0x%x, %u channels, %u Hz",
1679 wav->format, wav->channels, wav->rate));
1684 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1685 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1691 * Read WAV file tag when streaming
1693 static GstFlowReturn
1694 gst_wavparse_parse_stream_init (GstWavParse * wav)
1696 if (gst_adapter_available (wav->adapter) >= 12) {
1699 /* _take flushes the data */
1700 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1702 GST_DEBUG ("Parsing wav header");
1703 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1704 return GST_FLOW_ERROR;
1707 /* Go to next state */
1708 wav->state = GST_WAVPARSE_HEADER;
1713 /* handle an event sent directly to the element.
1715 * This event can be sent either in the READY state or the
1716 * >READY state. The only event of interest really is the seek
1719 * In the READY state we can only store the event and try to
1720 * respect it when going to PAUSED. We assume we are in the
1721 * READY state when our parsing state != GST_WAVPARSE_DATA.
1723 * When we are steaming, we can simply perform the seek right
1727 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1729 GstWavParse *wav = GST_WAVPARSE (element);
1730 gboolean res = FALSE;
1733 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1735 switch (GST_EVENT_TYPE (event)) {
1736 case GST_EVENT_SEEK:
1737 if (wav->state == GST_WAVPARSE_DATA) {
1738 /* we can handle the seek directly when streaming data */
1739 res = gst_wavparse_perform_seek (wav, event);
1741 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1743 event_p = &wav->seek_event;
1744 gst_event_replace (event_p, event);
1746 /* we always return true */
1753 gst_event_unref (event);
1758 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1762 s = gst_caps_get_structure (caps, 0);
1763 if (!gst_structure_has_name (s, "audio/x-dts"))
1765 if (prob >= GST_TYPE_FIND_LIKELY)
1767 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
1768 if (prob < GST_TYPE_FIND_POSSIBLE)
1770 /* .. in which case we want at least a valid-looking rate and channels */
1771 if (!gst_structure_has_field (s, "channels"))
1773 /* and for extra assurance we could also check the rate from the DTS frame
1774 * against the one in the wav header, but for now let's not do that */
1775 return gst_structure_has_field (s, "rate");
1779 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1783 GST_DEBUG_OBJECT (wav, "adding src pad");
1785 g_assert (wav->caps != NULL);
1787 s = gst_caps_get_structure (wav->caps, 0);
1788 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1789 GstTypeFindProbability prob;
1792 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1793 if (tf_caps != NULL) {
1794 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1795 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1796 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1797 gst_caps_unref (wav->caps);
1798 wav->caps = tf_caps;
1800 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1801 GST_TAG_AUDIO_CODEC, "dts", NULL);
1803 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1804 "marked as raw PCM audio, but ignoring for now", tf_caps);
1805 gst_caps_unref (tf_caps);
1810 gst_pad_set_caps (wav->srcpad, wav->caps);
1811 gst_caps_replace (&wav->caps, NULL);
1813 if (wav->start_segment) {
1814 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1815 gst_pad_push_event (wav->srcpad, wav->start_segment);
1816 wav->start_segment = NULL;
1820 gst_pad_push_event (wav->srcpad, gst_event_new_tag (wav->tags));
1825 static GstFlowReturn
1826 gst_wavparse_stream_data (GstWavParse * wav)
1828 GstBuffer *buf = NULL;
1829 GstFlowReturn res = GST_FLOW_OK;
1830 guint64 desired, obtained;
1831 GstClockTime timestamp, next_timestamp, duration;
1832 guint64 pos, nextpos;
1835 GST_LOG_OBJECT (wav,
1836 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1837 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1839 /* Get the next n bytes and output them */
1840 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1843 /* scale the amount of data by the segment rate so we get equal
1844 * amounts of data regardless of the playback rate */
1846 MIN (gst_guint64_to_gdouble (wav->dataleft),
1847 wav->max_buf_size * ABS (wav->segment.rate));
1849 if (desired >= wav->blockalign && wav->blockalign > 0)
1850 desired -= (desired % wav->blockalign);
1852 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1853 "from the sinkpad", desired);
1855 if (wav->streaming) {
1856 guint avail = gst_adapter_available (wav->adapter);
1859 /* flush some bytes if evil upstream sends segment that starts
1860 * before data or does is not send sample aligned segment */
1861 if (G_LIKELY (wav->offset >= wav->datastart)) {
1862 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1864 extra = wav->datastart - wav->offset;
1867 if (G_UNLIKELY (extra)) {
1868 extra = wav->bytes_per_sample - extra;
1869 if (extra <= avail) {
1870 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
1871 gst_adapter_flush (wav->adapter, extra);
1872 wav->offset += extra;
1873 wav->dataleft -= extra;
1874 goto iterate_adapter;
1876 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
1877 gst_adapter_clear (wav->adapter);
1878 wav->offset += avail;
1879 wav->dataleft -= avail;
1884 if (avail < desired) {
1885 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
1889 buf = gst_adapter_take_buffer (wav->adapter, desired);
1891 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
1892 desired, &buf)) != GST_FLOW_OK)
1895 /* we may get a short buffer at the end of the file */
1896 if (gst_buffer_get_size (buf) < desired) {
1897 gsize size = gst_buffer_get_size (buf);
1899 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
1900 if (size >= wav->blockalign) {
1901 buf = gst_buffer_make_writable (buf);
1902 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
1904 gst_buffer_unref (buf);
1910 obtained = gst_buffer_get_size (buf);
1912 /* our positions in bytes */
1913 pos = wav->offset - wav->datastart;
1914 nextpos = pos + obtained;
1916 /* update offsets, does not overflow. */
1917 buf = gst_buffer_make_writable (buf);
1918 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
1919 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
1921 /* first chunk of data? create the source pad. We do this only here so
1922 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
1923 if (G_UNLIKELY (wav->first)) {
1925 /* this will also push the segment events */
1926 gst_wavparse_add_src_pad (wav, buf);
1928 /* If we have a pending start segment, send it now. */
1929 if (G_UNLIKELY (wav->start_segment != NULL)) {
1930 gst_pad_push_event (wav->srcpad, wav->start_segment);
1931 wav->start_segment = NULL;
1936 /* and timestamps if we have a bitrate, be careful for overflows */
1938 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
1940 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
1941 duration = next_timestamp - timestamp;
1943 /* update current running segment position */
1944 if (G_LIKELY (next_timestamp >= wav->segment.start))
1945 wav->segment.position = next_timestamp;
1946 } else if (wav->fact) {
1948 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
1949 /* and timestamps if we have a bitrate, be careful for overflows */
1950 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
1951 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
1952 duration = next_timestamp - timestamp;
1954 /* no bitrate, all we know is that the first sample has timestamp 0, all
1955 * other positions and durations have unknown timestamp. */
1959 timestamp = GST_CLOCK_TIME_NONE;
1960 duration = GST_CLOCK_TIME_NONE;
1961 /* update current running segment position with byte offset */
1962 if (G_LIKELY (nextpos >= wav->segment.start))
1963 wav->segment.position = nextpos;
1965 if ((pos > 0) && wav->vbr) {
1966 /* don't set timestamps for VBR files if it's not the first buffer */
1967 timestamp = GST_CLOCK_TIME_NONE;
1968 duration = GST_CLOCK_TIME_NONE;
1971 GST_DEBUG_OBJECT (wav, "marking DISCONT");
1972 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
1973 wav->discont = FALSE;
1976 GST_BUFFER_TIMESTAMP (buf) = timestamp;
1977 GST_BUFFER_DURATION (buf) = duration;
1979 GST_LOG_OBJECT (wav,
1980 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
1981 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
1982 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
1984 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
1987 if (obtained < wav->dataleft) {
1988 wav->offset += obtained;
1989 wav->dataleft -= obtained;
1991 wav->offset += wav->dataleft;
1995 /* Iterate until need more data, so adapter size won't grow */
1996 if (wav->streaming) {
1997 GST_LOG_OBJECT (wav,
1998 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2000 goto iterate_adapter;
2007 GST_DEBUG_OBJECT (wav, "found EOS");
2008 return GST_FLOW_EOS;
2012 /* check if we got EOS */
2013 if (res == GST_FLOW_EOS)
2016 GST_WARNING_OBJECT (wav,
2017 "Error getting %" G_GINT64_FORMAT " bytes from the "
2018 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2023 GST_INFO_OBJECT (wav,
2024 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2025 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2026 gst_pad_is_linked (wav->srcpad));
2032 gst_wavparse_loop (GstPad * pad)
2035 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2039 GST_LOG_OBJECT (wav, "process data");
2041 switch (wav->state) {
2042 case GST_WAVPARSE_START:
2043 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2044 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2048 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2049 event = gst_event_new_stream_start (stream_id);
2050 gst_event_set_group_id (event, gst_util_group_id_next ());
2051 gst_pad_push_event (wav->srcpad, event);
2054 wav->state = GST_WAVPARSE_HEADER;
2057 case GST_WAVPARSE_HEADER:
2058 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2059 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2062 wav->state = GST_WAVPARSE_DATA;
2063 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2066 case GST_WAVPARSE_DATA:
2067 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2071 g_assert_not_reached ();
2078 const gchar *reason = gst_flow_get_name (ret);
2080 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2081 gst_pad_pause_task (pad);
2083 if (ret == GST_FLOW_EOS) {
2084 /* handle end-of-stream/segment */
2085 /* so align our position with the end of it, if there is one
2086 * this ensures a subsequent will arrive at correct base/acc time */
2087 if (wav->segment.format == GST_FORMAT_TIME) {
2088 if (wav->segment.rate > 0.0 &&
2089 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2090 wav->segment.position = wav->segment.stop;
2091 else if (wav->segment.rate < 0.0)
2092 wav->segment.position = wav->segment.start;
2094 if (wav->state == GST_WAVPARSE_START) {
2095 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2096 ("No valid input found before end of stream"));
2097 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2099 /* add pad before we perform EOS */
2100 if (G_UNLIKELY (wav->first)) {
2102 gst_wavparse_add_src_pad (wav, NULL);
2105 /* perform EOS logic */
2106 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2109 if ((stop = wav->segment.stop) == -1)
2110 stop = wav->segment.duration;
2112 gst_element_post_message (GST_ELEMENT_CAST (wav),
2113 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2114 wav->segment.format, stop));
2115 gst_pad_push_event (wav->srcpad,
2116 gst_event_new_segment_done (wav->segment.format, stop));
2118 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2121 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2122 /* for fatal errors we post an error message, post the error
2123 * first so the app knows about the error first. */
2124 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2125 (_("Internal data flow error.")),
2126 ("streaming task paused, reason %s (%d)", reason, ret));
2127 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2133 static GstFlowReturn
2134 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2137 GstWavParse *wav = GST_WAVPARSE (parent);
2139 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2140 gst_buffer_get_size (buf));
2142 gst_adapter_push (wav->adapter, buf);
2144 switch (wav->state) {
2145 case GST_WAVPARSE_START:
2146 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2147 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2150 if (wav->state != GST_WAVPARSE_HEADER)
2153 /* otherwise fall-through */
2154 case GST_WAVPARSE_HEADER:
2155 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2156 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2159 if (!wav->got_fmt || wav->datastart == 0)
2162 wav->state = GST_WAVPARSE_DATA;
2163 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2166 case GST_WAVPARSE_DATA:
2167 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2168 wav->discont = TRUE;
2169 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2173 g_return_val_if_reached (GST_FLOW_ERROR);
2176 if (G_UNLIKELY (wav->abort_buffering)) {
2177 wav->abort_buffering = FALSE;
2178 ret = GST_FLOW_ERROR;
2179 /* sort of demux/parse error */
2180 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2186 static GstFlowReturn
2187 gst_wavparse_flush_data (GstWavParse * wav)
2189 GstFlowReturn ret = GST_FLOW_OK;
2192 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2194 wav->end_offset = wav->offset + av;
2195 ret = gst_wavparse_stream_data (wav);
2202 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2204 GstWavParse *wav = GST_WAVPARSE (parent);
2205 gboolean ret = TRUE;
2207 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2209 switch (GST_EVENT_TYPE (event)) {
2210 case GST_EVENT_CAPS:
2212 /* discard, we'll come up with proper src caps */
2213 gst_event_unref (event);
2216 case GST_EVENT_SEGMENT:
2218 gint64 start, stop, offset = 0, end_offset = -1;
2221 /* some debug output */
2222 gst_event_copy_segment (event, &segment);
2223 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2226 if (wav->state != GST_WAVPARSE_DATA) {
2227 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2231 /* now we are either committed to TIME or BYTE format,
2232 * and we only expect a BYTE segment, e.g. following a seek */
2233 if (segment.format == GST_FORMAT_BYTES) {
2234 /* handle (un)signed issues */
2235 start = segment.start;
2236 stop = segment.stop;
2239 start -= wav->datastart;
2240 start = MAX (start, 0);
2244 segment.stop -= wav->datastart;
2245 segment.stop = MAX (stop, 0);
2247 if (wav->segment.format == GST_FORMAT_TIME) {
2248 guint64 bps = wav->bps;
2250 /* operating in format TIME, so we can convert */
2251 if (!bps && wav->fact)
2253 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2257 gst_util_uint64_scale_ceil (start, GST_SECOND,
2258 (guint64) wav->bps);
2261 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2262 (guint64) wav->bps);
2266 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2270 segment.start = start;
2271 segment.stop = stop;
2273 /* accept upstream's notion of segment and distribute along */
2274 segment.format = wav->segment.format;
2275 segment.time = segment.position = segment.start;
2276 segment.duration = wav->segment.duration;
2277 segment.base = gst_segment_to_running_time (&wav->segment,
2278 GST_FORMAT_TIME, wav->segment.position);
2280 gst_segment_copy_into (&segment, &wav->segment);
2282 /* also store the newsegment event for the streaming thread */
2283 if (wav->start_segment)
2284 gst_event_unref (wav->start_segment);
2285 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2286 wav->start_segment = gst_event_new_segment (&segment);
2288 /* stream leftover data in current segment */
2289 gst_wavparse_flush_data (wav);
2290 /* and set up streaming thread for next one */
2291 wav->offset = offset;
2292 wav->end_offset = end_offset;
2293 if (wav->end_offset > 0) {
2294 wav->dataleft = wav->end_offset - wav->offset;
2296 /* infinity; upstream will EOS when done */
2297 wav->dataleft = G_MAXUINT64;
2300 gst_event_unref (event);
2304 if (wav->state == GST_WAVPARSE_START) {
2305 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2306 ("No valid input found before end of stream"));
2308 /* add pad if needed so EOS is seen downstream */
2309 if (G_UNLIKELY (wav->first)) {
2311 gst_wavparse_add_src_pad (wav, NULL);
2313 /* stream leftover data in current segment */
2314 gst_wavparse_flush_data (wav);
2319 case GST_EVENT_FLUSH_STOP:
2323 gst_adapter_clear (wav->adapter);
2324 wav->discont = TRUE;
2325 dur = wav->segment.duration;
2326 gst_segment_init (&wav->segment, wav->segment.format);
2327 wav->segment.duration = dur;
2331 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2339 /* convert and query stuff */
2340 static const GstFormat *
2341 gst_wavparse_get_formats (GstPad * pad)
2343 static GstFormat formats[] = {
2346 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2355 gst_wavparse_pad_convert (GstPad * pad,
2356 GstFormat src_format, gint64 src_value,
2357 GstFormat * dest_format, gint64 * dest_value)
2359 GstWavParse *wavparse;
2360 gboolean res = TRUE;
2362 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2364 if (*dest_format == src_format) {
2365 *dest_value = src_value;
2369 if ((wavparse->bps == 0) && !wavparse->fact)
2372 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2373 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2375 switch (src_format) {
2376 case GST_FORMAT_BYTES:
2377 switch (*dest_format) {
2378 case GST_FORMAT_DEFAULT:
2379 *dest_value = src_value / wavparse->bytes_per_sample;
2380 /* make sure we end up on a sample boundary */
2381 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2383 case GST_FORMAT_TIME:
2384 /* src_value + datastart = offset */
2385 GST_INFO_OBJECT (wavparse,
2386 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2388 if (wavparse->bps > 0)
2389 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2390 (guint64) wavparse->bps);
2391 else if (wavparse->fact) {
2392 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2393 wavparse->rate, wavparse->fact);
2396 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2407 case GST_FORMAT_DEFAULT:
2408 switch (*dest_format) {
2409 case GST_FORMAT_BYTES:
2410 *dest_value = src_value * wavparse->bytes_per_sample;
2412 case GST_FORMAT_TIME:
2413 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2414 (guint64) wavparse->rate);
2422 case GST_FORMAT_TIME:
2423 switch (*dest_format) {
2424 case GST_FORMAT_BYTES:
2425 if (wavparse->bps > 0)
2426 *dest_value = gst_util_uint64_scale (src_value,
2427 (guint64) wavparse->bps, GST_SECOND);
2429 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2430 wavparse->rate, wavparse->fact);
2432 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2434 /* make sure we end up on a sample boundary */
2435 *dest_value -= *dest_value % wavparse->blockalign;
2437 case GST_FORMAT_DEFAULT:
2438 *dest_value = gst_util_uint64_scale (src_value,
2439 (guint64) wavparse->rate, GST_SECOND);
2458 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2464 /* handle queries for location and length in requested format */
2466 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2468 gboolean res = TRUE;
2469 GstWavParse *wav = GST_WAVPARSE (parent);
2471 /* only if we know */
2472 if (wav->state != GST_WAVPARSE_DATA) {
2476 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2478 switch (GST_QUERY_TYPE (query)) {
2479 case GST_QUERY_POSITION:
2485 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2486 curb = wav->offset - wav->datastart;
2487 gst_query_parse_position (query, &format, NULL);
2488 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2491 case GST_FORMAT_BYTES:
2492 format = GST_FORMAT_BYTES;
2496 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2501 gst_query_set_position (query, format, cur);
2504 case GST_QUERY_DURATION:
2506 gint64 duration = 0;
2509 if (wav->ignore_length) {
2514 gst_query_parse_duration (query, &format, NULL);
2517 case GST_FORMAT_BYTES:{
2518 format = GST_FORMAT_BYTES;
2519 duration = wav->datasize;
2522 case GST_FORMAT_TIME:
2523 if ((res = gst_wavparse_calculate_duration (wav))) {
2524 duration = wav->duration;
2532 gst_query_set_duration (query, format, duration);
2535 case GST_QUERY_CONVERT:
2537 gint64 srcvalue, dstvalue;
2538 GstFormat srcformat, dstformat;
2540 gst_query_parse_convert (query, &srcformat, &srcvalue,
2541 &dstformat, &dstvalue);
2542 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2543 &dstformat, &dstvalue);
2545 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2548 case GST_QUERY_SEEKING:{
2550 gboolean seekable = FALSE;
2552 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2553 if (fmt == wav->segment.format) {
2554 if (wav->streaming) {
2557 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2558 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2559 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2560 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2562 gst_query_unref (q);
2564 GST_LOG_OBJECT (wav, "looping => seekable");
2568 } else if (fmt == GST_FORMAT_TIME) {
2572 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2577 res = gst_pad_query_default (pad, parent, query);
2584 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2586 GstWavParse *wavparse = GST_WAVPARSE (parent);
2587 gboolean res = FALSE;
2589 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2591 switch (GST_EVENT_TYPE (event)) {
2592 case GST_EVENT_SEEK:
2593 /* can only handle events when we are in the data state */
2594 if (wavparse->state == GST_WAVPARSE_DATA) {
2595 res = gst_wavparse_perform_seek (wavparse, event);
2597 gst_event_unref (event);
2600 case GST_EVENT_TOC_SELECT:
2603 GstTocEntry *entry = NULL;
2604 GstEvent *seek_event;
2607 if (!wavparse->toc) {
2608 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2611 gst_event_parse_toc_select (event, &uid);
2613 GST_OBJECT_LOCK (wavparse);
2614 entry = gst_toc_find_entry (wavparse->toc, uid);
2615 if (entry == NULL) {
2616 GST_OBJECT_UNLOCK (wavparse);
2617 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2621 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2622 GST_OBJECT_UNLOCK (wavparse);
2623 seek_event = gst_event_new_seek (1.0,
2625 GST_SEEK_FLAG_FLUSH,
2626 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2627 res = gst_wavparse_perform_seek (wavparse, seek_event);
2628 gst_event_unref (seek_event);
2632 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2636 gst_event_unref (event);
2641 res = gst_pad_push_event (wavparse->sinkpad, event);
2648 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2650 GstWavParse *wav = GST_WAVPARSE (parent);
2655 gst_adapter_clear (wav->adapter);
2656 g_object_unref (wav->adapter);
2657 wav->adapter = NULL;
2660 query = gst_query_new_scheduling ();
2662 if (!gst_pad_peer_query (sinkpad, query)) {
2663 gst_query_unref (query);
2667 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2668 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2669 gst_query_unref (query);
2674 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2675 wav->streaming = FALSE;
2676 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2680 GST_DEBUG_OBJECT (sinkpad, "activating push");
2681 wav->streaming = TRUE;
2682 wav->adapter = gst_adapter_new ();
2683 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2689 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2690 GstPadMode mode, gboolean active)
2695 case GST_PAD_MODE_PUSH:
2698 case GST_PAD_MODE_PULL:
2700 /* if we have a scheduler we can start the task */
2701 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2704 res = gst_pad_stop_task (sinkpad);
2714 static GstStateChangeReturn
2715 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2717 GstStateChangeReturn ret;
2718 GstWavParse *wav = GST_WAVPARSE (element);
2720 switch (transition) {
2721 case GST_STATE_CHANGE_NULL_TO_READY:
2723 case GST_STATE_CHANGE_READY_TO_PAUSED:
2724 gst_wavparse_reset (wav);
2726 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2732 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2734 switch (transition) {
2735 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2737 case GST_STATE_CHANGE_PAUSED_TO_READY:
2738 gst_wavparse_reset (wav);
2740 case GST_STATE_CHANGE_READY_TO_NULL:
2749 gst_wavparse_set_property (GObject * object, guint prop_id,
2750 const GValue * value, GParamSpec * pspec)
2754 g_return_if_fail (GST_IS_WAVPARSE (object));
2755 self = GST_WAVPARSE (object);
2758 case PROP_IGNORE_LENGTH:
2759 self->ignore_length = g_value_get_boolean (value);
2762 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2768 gst_wavparse_get_property (GObject * object, guint prop_id,
2769 GValue * value, GParamSpec * pspec)
2773 g_return_if_fail (GST_IS_WAVPARSE (object));
2774 self = GST_WAVPARSE (object);
2777 case PROP_IGNORE_LENGTH:
2778 g_value_set_boolean (value, self->ignore_length);
2781 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2786 plugin_init (GstPlugin * plugin)
2790 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2794 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2797 "Parse a .wav file into raw audio",
2798 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)