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 /* Data size chunk of RF64,
63 * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
64 #define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
66 static void gst_wavparse_dispose (GObject * object);
68 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
70 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
71 GstObject * parent, GstPadMode mode, gboolean active);
72 static gboolean gst_wavparse_send_event (GstElement * element,
74 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
75 GstStateChange transition);
77 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
79 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
80 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
82 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
84 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
86 static void gst_wavparse_loop (GstPad * pad);
87 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
90 static void gst_wavparse_set_property (GObject * object, guint prop_id,
91 const GValue * value, GParamSpec * pspec);
92 static void gst_wavparse_get_property (GObject * object, guint prop_id,
93 GValue * value, GParamSpec * pspec);
95 #define DEFAULT_IGNORE_LENGTH FALSE
103 static GstStaticPadTemplate sink_template_factory =
104 GST_STATIC_PAD_TEMPLATE ("sink",
107 GST_STATIC_CAPS ("audio/x-wav")
111 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
113 #define gst_wavparse_parent_class parent_class
114 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
119 /* Offset Size Description Value
120 * 0x00 4 ID unique identification value
121 * 0x04 4 Position play order position
122 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
123 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
124 * 0x10 4 Block Start Byte Offset to sample of First Channel
125 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
129 guint32 data_chunk_id;
132 guint32 sample_offset;
137 /* Offset Size Description Value
138 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
141 guint32 cue_point_id;
143 } GstWavParseLabl, GstWavParseNote;
146 gst_wavparse_class_init (GstWavParseClass * klass)
148 GstElementClass *gstelement_class;
149 GObjectClass *object_class;
150 GstPadTemplate *src_template;
152 gstelement_class = (GstElementClass *) klass;
153 object_class = (GObjectClass *) klass;
155 parent_class = g_type_class_peek_parent (klass);
157 object_class->dispose = gst_wavparse_dispose;
159 object_class->set_property = gst_wavparse_set_property;
160 object_class->get_property = gst_wavparse_get_property;
163 * GstWavParse:ignore-length:
165 * This selects whether the length found in a data chunk
166 * should be ignored. This may be useful for streamed audio
167 * where the length is unknown until the end of streaming,
168 * and various software/hardware just puts some random value
169 * in there and hopes it doesn't break too much.
171 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
172 g_param_spec_boolean ("ignore-length",
174 "Ignore length from the Wave header",
175 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
178 gstelement_class->change_state = gst_wavparse_change_state;
179 gstelement_class->send_event = gst_wavparse_send_event;
182 gst_element_class_add_static_pad_template (gstelement_class,
183 &sink_template_factory);
185 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
186 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
187 gst_element_class_add_pad_template (gstelement_class, src_template);
189 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
190 "Codec/Demuxer/Audio",
191 "Parse a .wav file into raw audio",
192 "Erik Walthinsen <omega@cse.ogi.edu>");
196 gst_wavparse_reset (GstWavParse * wav)
198 wav->state = GST_WAVPARSE_START;
200 /* These will all be set correctly in the fmt chunk */
214 wav->got_fmt = FALSE;
218 gst_event_unref (wav->seek_event);
219 wav->seek_event = NULL;
221 gst_adapter_clear (wav->adapter);
222 g_object_unref (wav->adapter);
226 gst_tag_list_unref (wav->tags);
229 gst_toc_unref (wav->toc);
232 g_list_free_full (wav->cues, g_free);
235 g_list_free_full (wav->labls, g_free);
238 gst_caps_unref (wav->caps);
240 if (wav->start_segment)
241 gst_event_unref (wav->start_segment);
242 wav->start_segment = NULL;
246 gst_wavparse_dispose (GObject * object)
248 GstWavParse *wav = GST_WAVPARSE (object);
250 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
251 gst_wavparse_reset (wav);
253 G_OBJECT_CLASS (parent_class)->dispose (object);
257 gst_wavparse_init (GstWavParse * wavparse)
259 gst_wavparse_reset (wavparse);
263 gst_pad_new_from_static_template (&sink_template_factory, "sink");
264 gst_pad_set_activate_function (wavparse->sinkpad,
265 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
266 gst_pad_set_activatemode_function (wavparse->sinkpad,
267 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
268 gst_pad_set_chain_function (wavparse->sinkpad,
269 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
270 gst_pad_set_event_function (wavparse->sinkpad,
271 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
272 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
276 gst_pad_new_from_template (gst_element_class_get_pad_template
277 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
278 gst_pad_use_fixed_caps (wavparse->srcpad);
279 gst_pad_set_query_function (wavparse->srcpad,
280 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
281 gst_pad_set_event_function (wavparse->srcpad,
282 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
283 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
287 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
291 if (!gst_riff_parse_file_header (element, buf, &doctype))
294 if (doctype != GST_RIFF_RIFF_WAVE)
302 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
303 ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
309 gst_wavparse_stream_init (GstWavParse * wav)
312 GstBuffer *buf = NULL;
314 if ((res = gst_pad_pull_range (wav->sinkpad,
315 wav->offset, 12, &buf)) != GST_FLOW_OK)
317 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
318 return GST_FLOW_ERROR;
326 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
328 /* -1 always maps to -1 */
334 /* 0 always maps to 0 */
341 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
343 } else if (wav->fact) {
344 guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
345 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
352 /* This function is used to perform seeks on the element.
354 * It also works when event is NULL, in which case it will just
355 * start from the last configured segment. This technique is
356 * used when activating the element and to perform the seek in
360 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
364 GstFormat format, bformat;
366 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
367 gint64 cur, stop, upstream_size;
370 GstSegment seeksegment = { 0, };
375 GST_DEBUG_OBJECT (wav, "doing seek with event");
377 gst_event_parse_seek (event, &rate, &format, &flags,
378 &cur_type, &cur, &stop_type, &stop);
379 seqnum = gst_event_get_seqnum (event);
381 /* no negative rates yet */
385 if (format != wav->segment.format) {
386 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
387 gst_format_get_name (format),
388 gst_format_get_name (wav->segment.format));
390 if (cur_type != GST_SEEK_TYPE_NONE)
392 gst_pad_query_convert (wav->srcpad, format, cur,
393 wav->segment.format, &cur);
394 if (res && stop_type != GST_SEEK_TYPE_NONE)
396 gst_pad_query_convert (wav->srcpad, format, stop,
397 wav->segment.format, &stop);
401 format = wav->segment.format;
404 GST_DEBUG_OBJECT (wav, "doing seek without event");
407 cur_type = GST_SEEK_TYPE_SET;
408 stop_type = GST_SEEK_TYPE_SET;
411 /* in push mode, we must delegate to upstream */
412 if (wav->streaming) {
413 gboolean res = FALSE;
415 /* if streaming not yet started; only prepare initial newsegment */
416 if (!event || wav->state != GST_WAVPARSE_DATA) {
417 if (wav->start_segment)
418 gst_event_unref (wav->start_segment);
419 wav->start_segment = gst_event_new_segment (&wav->segment);
422 /* convert seek positions to byte positions in data sections */
423 if (format == GST_FORMAT_TIME) {
424 /* should not fail */
425 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
427 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
430 /* mind sample boundary and header */
432 cur -= (cur % wav->bytes_per_sample);
433 cur += wav->datastart;
436 stop -= (stop % wav->bytes_per_sample);
437 stop += wav->datastart;
439 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
440 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
442 /* BYTE seek event */
443 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
445 gst_event_set_seqnum (event, seqnum);
446 res = gst_pad_push_event (wav->sinkpad, event);
452 flush = flags & GST_SEEK_FLAG_FLUSH;
454 /* now we need to make sure the streaming thread is stopped. We do this by
455 * either sending a FLUSH_START event downstream which will cause the
456 * streaming thread to stop with a WRONG_STATE.
457 * For a non-flushing seek we simply pause the task, which will happen as soon
458 * as it completes one iteration (and thus might block when the sink is
459 * blocking in preroll). */
462 GST_DEBUG_OBJECT (wav, "sending flush start");
464 fevent = gst_event_new_flush_start ();
465 gst_event_set_seqnum (fevent, seqnum);
466 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
467 gst_pad_push_event (wav->srcpad, fevent);
469 gst_pad_pause_task (wav->sinkpad);
472 /* we should now be able to grab the streaming thread because we stopped it
473 * with the above flush/pause code */
474 GST_PAD_STREAM_LOCK (wav->sinkpad);
476 /* save current position */
477 last_stop = wav->segment.position;
479 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
481 /* copy segment, we need this because we still need the old
482 * segment when we close the current segment. */
483 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
485 /* configure the seek parameters in the seeksegment. We will then have the
486 * right values in the segment to perform the seek */
488 GST_DEBUG_OBJECT (wav, "configuring seek");
489 gst_segment_do_seek (&seeksegment, rate, format, flags,
490 cur_type, cur, stop_type, stop, &update);
493 /* figure out the last position we need to play. If it's configured (stop !=
494 * -1), use that, else we play until the total duration of the file */
495 if ((stop = seeksegment.stop) == -1)
496 stop = seeksegment.duration;
498 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
499 if ((cur_type != GST_SEEK_TYPE_NONE)) {
500 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
501 * we can just copy the last_stop. If not, we use the bps to convert TIME to
503 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
504 (gint64 *) & wav->offset))
505 wav->offset = seeksegment.position;
506 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
507 wav->offset -= (wav->offset % wav->bytes_per_sample);
508 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
509 wav->offset += wav->datastart;
510 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
512 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
516 if (stop_type != GST_SEEK_TYPE_NONE) {
517 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
518 wav->end_offset = stop;
519 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
520 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
521 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
522 wav->end_offset += wav->datastart;
523 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
525 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
529 /* make sure filesize is not exceeded due to rounding errors or so,
530 * same precaution as in _stream_headers */
531 bformat = GST_FORMAT_BYTES;
532 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
533 wav->end_offset = MIN (wav->end_offset, upstream_size);
535 if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
536 wav->end_offset = wav->datastart + wav->datasize;
538 /* this is the range of bytes we will use for playback */
539 wav->offset = MIN (wav->offset, wav->end_offset);
540 wav->dataleft = wav->end_offset - wav->offset;
542 GST_DEBUG_OBJECT (wav,
543 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
544 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
545 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
547 /* prepare for streaming again */
551 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
552 GST_DEBUG_OBJECT (wav, "sending flush stop");
554 fevent = gst_event_new_flush_stop (TRUE);
555 gst_event_set_seqnum (fevent, seqnum);
556 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
557 gst_pad_push_event (wav->srcpad, fevent);
560 /* now we did the seek and can activate the new segment values */
561 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
563 /* if we're doing a segment seek, post a SEGMENT_START message */
564 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
565 gst_element_post_message (GST_ELEMENT_CAST (wav),
566 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
567 wav->segment.format, wav->segment.position));
570 /* now create the newsegment */
571 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
572 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
574 /* store the newsegment event so it can be sent from the streaming thread. */
575 if (wav->start_segment)
576 gst_event_unref (wav->start_segment);
577 wav->start_segment = gst_event_new_segment (&wav->segment);
578 gst_event_set_seqnum (wav->start_segment, seqnum);
580 /* mark discont if we are going to stream from another position. */
581 if (last_stop != wav->segment.position) {
582 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
586 /* and start the streaming task again */
587 if (!wav->streaming) {
588 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
592 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
599 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
604 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
609 GST_DEBUG_OBJECT (wav,
610 "Could not determine byte position for desired time");
616 * gst_wavparse_peek_chunk_info:
617 * @wav Wavparse object
618 * @tag holder for tag
619 * @size holder for tag size
621 * Peek next chunk info (tag and size)
623 * Returns: %TRUE when the chunk info (header) is available
626 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
628 const guint8 *data = NULL;
630 if (gst_adapter_available (wav->adapter) < 8)
633 data = gst_adapter_map (wav->adapter, 8);
634 *tag = GST_READ_UINT32_LE (data);
635 *size = GST_READ_UINT32_LE (data + 4);
636 gst_adapter_unmap (wav->adapter);
638 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
639 GST_FOURCC_ARGS (*tag));
645 * gst_wavparse_peek_chunk:
646 * @wav Wavparse object
647 * @tag holder for tag
648 * @size holder for tag size
650 * Peek enough data for one full chunk
652 * Returns: %TRUE when the full chunk is available
655 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
657 guint32 peek_size = 0;
660 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
663 /* size 0 -> empty data buffer would surprise most callers,
664 * large size -> do not bother trying to squeeze that into adapter,
665 * so we throw poor man's exception, which can be caught if caller really
666 * wants to handle 0 size chunk */
667 if (!(*size) || (*size) >= (1 << 30)) {
668 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
669 *size, GST_FOURCC_ARGS (*tag));
670 /* chain should give up */
671 wav->abort_buffering = TRUE;
674 peek_size = (*size + 1) & ~1;
675 available = gst_adapter_available (wav->adapter);
677 if (available >= (8 + peek_size)) {
680 GST_LOG ("but only %u bytes available now", available);
686 * gst_wavparse_calculate_duration:
687 * @wav: wavparse object
689 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
692 * Returns: %TRUE if duration is available.
695 gst_wavparse_calculate_duration (GstWavParse * wav)
697 if (wav->duration > 0)
701 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
703 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
705 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
706 GST_TIME_ARGS (wav->duration));
708 } else if (wav->fact) {
710 gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
711 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
712 GST_TIME_ARGS (wav->duration));
719 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
724 if (wav->streaming) {
725 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
728 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
729 GST_FOURCC_ARGS (tag));
730 flush = 8 + ((size + 1) & ~1);
731 wav->offset += flush;
732 if (wav->streaming) {
733 gst_adapter_flush (wav->adapter, flush);
735 gst_buffer_unref (buf);
742 * gst_wavparse_cue_chunk:
743 * @wav GstWavParse object
744 * @data holder for data
745 * @size holder for data size
747 * Parse cue chunk from @data to wav->cues.
749 * Returns: %TRUE when cue chunk is available
752 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
759 GST_WARNING_OBJECT (wav, "found another cue's");
763 ncues = GST_READ_UINT32_LE (data);
765 if (size < 4 + ncues * 24) {
766 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
772 for (i = 0; i < ncues; i++) {
773 cue = g_new0 (GstWavParseCue, 1);
774 cue->id = GST_READ_UINT32_LE (data);
775 cue->position = GST_READ_UINT32_LE (data + 4);
776 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
777 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
778 cue->block_start = GST_READ_UINT32_LE (data + 16);
779 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
780 cues = g_list_append (cues, cue);
790 * gst_wavparse_labl_chunk:
791 * @wav GstWavParse object
792 * @data holder for data
793 * @size holder for data size
795 * Parse labl from @data to wav->labls.
797 * Returns: %TRUE when labl chunk is available
800 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
802 GstWavParseLabl *labl;
807 labl = g_new0 (GstWavParseLabl, 1);
811 labl->cue_point_id = GST_READ_UINT32_LE (data);
812 labl->text = g_memdup (data + 4, size - 4);
814 wav->labls = g_list_append (wav->labls, labl);
820 * gst_wavparse_note_chunk:
821 * @wav GstWavParse object
822 * @data holder for data
823 * @size holder for data size
825 * Parse note from @data to wav->notes.
827 * Returns: %TRUE when note chunk is available
830 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
832 GstWavParseNote *note;
837 note = g_new0 (GstWavParseNote, 1);
841 note->cue_point_id = GST_READ_UINT32_LE (data);
842 note->text = g_memdup (data + 4, size - 4);
844 wav->notes = g_list_append (wav->notes, note);
850 * gst_wavparse_smpl_chunk:
851 * @wav GstWavParse object
852 * @data holder for data
853 * @size holder for data size
855 * Parse smpl chunk from @data.
857 * Returns: %TRUE when cue chunk is available
860 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
865 manufacturer_id = GST_READ_UINT32_LE (data);
866 product_id = GST_READ_UINT32_LE (data + 4);
867 sample_period = GST_READ_UINT32_LE (data + 8);
869 note_number = GST_READ_UINT32_LE (data + 12);
871 pitch_fraction = GST_READ_UINT32_LE (data + 16);
872 SMPTE_format = GST_READ_UINT32_LE (data + 20);
873 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
874 num_sample_loops = GST_READ_UINT32_LE (data + 28);
875 List of Sample Loops, 24 bytes each
879 wav->tags = gst_tag_list_new_empty ();
880 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
881 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
886 * gst_wavparse_adtl_chunk:
887 * @wav GstWavParse object
888 * @data holder for data
889 * @size holder for data size
891 * Parse adtl from @data.
893 * Returns: %TRUE when adtl chunk is available
896 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
898 guint32 ltag, lsize, offset = 0;
901 ltag = GST_READ_UINT32_LE (data + offset);
902 lsize = GST_READ_UINT32_LE (data + offset + 4);
904 if (lsize + 8 > size) {
905 GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
910 case GST_RIFF_TAG_labl:
911 gst_wavparse_labl_chunk (wav, data + offset, size);
913 case GST_RIFF_TAG_note:
914 gst_wavparse_note_chunk (wav, data + offset, size);
917 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
918 GST_FOURCC_ARGS (ltag));
919 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
922 offset += 8 + GST_ROUND_UP_2 (lsize);
923 size -= 8 + GST_ROUND_UP_2 (lsize);
930 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
932 GstTagList *tags = NULL;
933 GstTocEntry *entry = NULL;
935 entry = gst_toc_find_entry (toc, id);
937 tags = gst_toc_entry_get_tags (entry);
939 tags = gst_tag_list_new_empty ();
940 gst_toc_entry_set_tags (entry, tags);
948 * gst_wavparse_create_toc:
949 * @wav GstWavParse object
951 * Create TOC from wav->cues and wav->labls.
954 gst_wavparse_create_toc (GstWavParse * wav)
960 GstWavParseLabl *labl;
961 GstWavParseNote *note;
964 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
966 GST_OBJECT_LOCK (wav);
968 GST_OBJECT_UNLOCK (wav);
969 GST_WARNING_OBJECT (wav, "found another TOC");
974 GST_OBJECT_UNLOCK (wav);
978 /* FIXME: send CURRENT scope toc too */
979 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
981 /* add cue edition */
982 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
983 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
984 gst_toc_append_entry (toc, entry);
986 /* add tracks in cue edition */
990 prev_subentry = cur_subentry;
991 /* previous track stop time = current track start time */
992 if (prev_subentry != NULL) {
993 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
994 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
995 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
997 id = g_strdup_printf ("%08x", cue->id);
998 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1000 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1001 stop = wav->duration;
1002 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1003 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1004 list = g_list_next (list);
1007 /* add tags in tracks */
1011 id = g_strdup_printf ("%08x", labl->cue_point_id);
1012 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1015 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1018 list = g_list_next (list);
1023 id = g_strdup_printf ("%08x", note->cue_point_id);
1024 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1027 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1030 list = g_list_next (list);
1033 /* send data as TOC */
1036 /* send TOC event */
1038 GST_OBJECT_UNLOCK (wav);
1039 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1045 #define MAX_BUFFER_SIZE 4096
1048 parse_ds64 (GstWavParse * wav, GstBuffer * buf)
1051 guint32 dataSizeLow, dataSizeHigh;
1052 guint32 sampleCountLow, sampleCountHigh;
1054 gst_buffer_map (buf, &map, GST_MAP_READ);
1055 dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
1056 dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
1057 sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
1058 sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
1059 gst_buffer_unmap (buf, &map);
1060 if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
1061 wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
1063 if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
1064 wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
1067 GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
1068 " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
1072 static GstFlowReturn
1073 gst_wavparse_stream_headers (GstWavParse * wav)
1075 GstFlowReturn res = GST_FLOW_OK;
1076 GstBuffer *buf = NULL;
1077 gst_riff_strf_auds *header = NULL;
1079 gboolean gotdata = FALSE;
1080 GstCaps *caps = NULL;
1081 gchar *codec_name = NULL;
1082 gint64 upstream_size = 0;
1085 /* search for "_fmt" chunk, which must be before "data" */
1086 while (!wav->got_fmt) {
1089 if (wav->streaming) {
1090 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1093 gst_adapter_flush (wav->adapter, 8);
1097 buf = gst_adapter_take_buffer (wav->adapter, size);
1099 gst_adapter_flush (wav->adapter, 1);
1100 wav->offset += GST_ROUND_UP_2 (size);
1102 buf = gst_buffer_new ();
1105 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1106 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1110 if (tag == GST_RS64_TAG_DS64) {
1111 if (!parse_ds64 (wav, buf))
1117 if (tag != GST_RIFF_TAG_fmt) {
1118 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1119 GST_FOURCC_ARGS (tag));
1120 gst_buffer_unref (buf);
1125 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1127 goto parse_header_error;
1129 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1131 /* do sanity checks of header fields */
1132 if (header->channels == 0)
1134 if (header->rate == 0)
1137 GST_DEBUG_OBJECT (wav, "creating the caps");
1139 /* Note: gst_riff_create_audio_caps might need to fix values in
1140 * the header header depending on the format, so call it first */
1141 /* FIXME: Need to handle the channel reorder map */
1142 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1143 NULL, &codec_name, NULL);
1146 gst_buffer_unref (extra);
1149 goto unknown_format;
1151 /* If we got raw audio from upstream, we remove the codec_data field,
1152 * which may have been added if the wav header included an extended
1153 * chunk. We want to keep it for non raw audio.
1155 s = gst_caps_get_structure (caps, 0);
1156 if (s && gst_structure_has_name (s, "audio/x-raw")) {
1157 gst_structure_remove_field (s, "codec_data");
1160 /* do more sanity checks of header fields
1161 * (these can be sanitized by gst_riff_create_audio_caps()
1163 wav->format = header->format;
1164 wav->rate = header->rate;
1165 wav->channels = header->channels;
1166 wav->blockalign = header->blockalign;
1167 wav->depth = header->bits_per_sample;
1168 wav->av_bps = header->av_bps;
1174 /* do format specific handling */
1175 switch (wav->format) {
1176 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1177 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1179 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1180 * bitrate inside the mpeg stream */
1181 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1185 case GST_RIFF_WAVE_FORMAT_PCM:
1186 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1187 goto invalid_blockalign;
1190 if (wav->av_bps > wav->blockalign * wav->rate)
1192 /* use the configured bps */
1193 wav->bps = wav->av_bps;
1197 wav->width = (wav->blockalign * 8) / wav->channels;
1198 wav->bytes_per_sample = wav->channels * wav->width / 8;
1200 if (wav->bytes_per_sample <= 0)
1201 goto no_bytes_per_sample;
1203 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1204 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1205 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1206 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1207 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1208 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1209 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1211 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1212 * formats). This will make the element output a BYTE format segment and
1213 * will not timestamp the outgoing buffers.
1215 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1217 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1219 /* create pad later so we can sniff the first few bytes
1220 * of the real data and correct our caps if necessary */
1221 gst_caps_replace (&wav->caps, caps);
1222 gst_caps_replace (&caps, NULL);
1224 wav->got_fmt = TRUE;
1227 wav->tags = gst_tag_list_new_empty ();
1229 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1230 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1232 g_free (codec_name);
1238 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1239 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1241 /* loop headers until we get data */
1243 if (wav->streaming) {
1244 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1251 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1252 &buf)) != GST_FLOW_OK)
1253 goto header_read_error;
1254 gst_buffer_map (buf, &map, GST_MAP_READ);
1255 tag = GST_READ_UINT32_LE (map.data);
1256 size = GST_READ_UINT32_LE (map.data + 4);
1257 gst_buffer_unmap (buf, &map);
1260 GST_INFO_OBJECT (wav,
1261 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
1262 G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
1264 /* Maximum valid size is INT_MAX */
1265 if (size & 0x80000000) {
1266 GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
1270 /* Clip to upstream size if known */
1271 if (wav->datasize > 0 && size + wav->offset > wav->datasize) {
1272 GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
1273 size = wav->datasize - wav->offset;
1276 /* wav is a st00pid format, we don't know for sure where data starts.
1277 * So we have to go bit by bit until we find the 'data' header
1280 case GST_RIFF_TAG_data:{
1283 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1285 if (wav->ignore_length) {
1286 GST_DEBUG_OBJECT (wav, "Ignoring length");
1289 if (wav->streaming) {
1290 gst_adapter_flush (wav->adapter, 8);
1293 gst_buffer_unref (buf);
1296 wav->datastart = wav->offset;
1297 /* use size from ds64 chunk if available */
1298 if (size64 == -1 && wav->datasize > 0) {
1299 GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
1300 size64 = wav->datasize;
1302 /* If size is zero, then the data chunk probably actually extends to
1303 the end of the file */
1304 if (size64 == 0 && upstream_size) {
1305 size64 = upstream_size - wav->datastart;
1307 /* Or the file might be truncated */
1308 else if (upstream_size) {
1309 size64 = MIN (size64, (upstream_size - wav->datastart));
1311 wav->datasize = size64;
1312 wav->dataleft = size64;
1313 wav->end_offset = size64 + wav->datastart;
1314 if (!wav->streaming) {
1315 /* We will continue parsing tags 'till end */
1316 wav->offset += size64;
1318 GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
1321 case GST_RIFF_TAG_fact:{
1322 if (wav->fact == 0 &&
1323 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1324 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1325 const guint data_size = 4;
1327 GST_INFO_OBJECT (wav, "Have fact chunk");
1328 if (size < data_size) {
1329 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1330 /* need more data */
1333 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1337 /* number of samples (for compressed formats) */
1338 if (wav->streaming) {
1339 const guint8 *data = NULL;
1341 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1344 gst_adapter_flush (wav->adapter, 8);
1345 data = gst_adapter_map (wav->adapter, data_size);
1346 wav->fact = GST_READ_UINT32_LE (data);
1347 gst_adapter_unmap (wav->adapter);
1348 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1350 gst_buffer_unref (buf);
1353 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1354 data_size, &buf)) != GST_FLOW_OK)
1355 goto header_read_error;
1356 gst_buffer_extract (buf, 0, &wav->fact, 4);
1357 wav->fact = GUINT32_FROM_LE (wav->fact);
1358 gst_buffer_unref (buf);
1360 GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
1361 wav->offset += 8 + GST_ROUND_UP_2 (size);
1364 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1365 /* need more data */
1371 case GST_RIFF_TAG_acid:{
1372 const gst_riff_acid *acid = NULL;
1373 const guint data_size = sizeof (gst_riff_acid);
1376 GST_INFO_OBJECT (wav, "Have acid chunk");
1377 if (size < data_size) {
1378 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1379 /* need more data */
1382 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1386 if (wav->streaming) {
1387 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1390 gst_adapter_flush (wav->adapter, 8);
1391 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1393 tempo = acid->tempo;
1394 gst_adapter_unmap (wav->adapter);
1397 gst_buffer_unref (buf);
1400 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1401 size, &buf)) != GST_FLOW_OK)
1402 goto header_read_error;
1403 gst_buffer_map (buf, &map, GST_MAP_READ);
1404 acid = (const gst_riff_acid *) map.data;
1405 tempo = acid->tempo;
1406 gst_buffer_unmap (buf, &map);
1408 /* send data as tags */
1410 wav->tags = gst_tag_list_new_empty ();
1411 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1412 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1414 size = GST_ROUND_UP_2 (size);
1415 if (wav->streaming) {
1416 gst_adapter_flush (wav->adapter, size);
1418 gst_buffer_unref (buf);
1420 wav->offset += 8 + size;
1423 /* FIXME: all list tags after data are ignored in streaming mode */
1424 case GST_RIFF_TAG_LIST:{
1427 if (wav->streaming) {
1428 const guint8 *data = NULL;
1430 if (gst_adapter_available (wav->adapter) < 12) {
1433 data = gst_adapter_map (wav->adapter, 12);
1434 ltag = GST_READ_UINT32_LE (data + 8);
1435 gst_adapter_unmap (wav->adapter);
1437 gst_buffer_unref (buf);
1440 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1441 &buf)) != GST_FLOW_OK)
1442 goto header_read_error;
1443 gst_buffer_extract (buf, 8, <ag, 4);
1444 ltag = GUINT32_FROM_LE (ltag);
1447 case GST_RIFF_LIST_INFO:{
1448 const gint data_size = size - 4;
1451 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1452 if (wav->streaming) {
1453 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1456 gst_adapter_flush (wav->adapter, 12);
1458 if (data_size > 0) {
1459 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1461 gst_adapter_flush (wav->adapter, 1);
1465 gst_buffer_unref (buf);
1467 if (data_size > 0) {
1469 gst_pad_pull_range (wav->sinkpad, wav->offset,
1470 data_size, &buf)) != GST_FLOW_OK)
1471 goto header_read_error;
1474 if (data_size > 0) {
1476 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1478 GstTagList *old = wav->tags;
1480 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1482 gst_tag_list_unref (old);
1483 gst_tag_list_unref (new);
1485 gst_buffer_unref (buf);
1486 wav->offset += GST_ROUND_UP_2 (data_size);
1490 case GST_RIFF_LIST_adtl:{
1491 const gint data_size = size - 4;
1493 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1494 if (wav->streaming) {
1495 const guint8 *data = NULL;
1497 gst_adapter_flush (wav->adapter, 12);
1499 data = gst_adapter_map (wav->adapter, data_size);
1500 gst_wavparse_adtl_chunk (wav, data, data_size);
1501 gst_adapter_unmap (wav->adapter);
1505 gst_buffer_unref (buf);
1509 gst_pad_pull_range (wav->sinkpad, wav->offset,
1510 data_size, &buf)) != GST_FLOW_OK)
1511 goto header_read_error;
1512 gst_buffer_map (buf, &map, GST_MAP_READ);
1513 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1515 gst_buffer_unmap (buf, &map);
1517 wav->offset += GST_ROUND_UP_2 (data_size);
1521 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1522 GST_FOURCC_ARGS (ltag));
1523 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1524 /* need more data */
1530 case GST_RIFF_TAG_cue:{
1531 const guint data_size = size;
1533 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1534 if (wav->streaming) {
1535 const guint8 *data = NULL;
1537 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1540 gst_adapter_flush (wav->adapter, 8);
1542 data = gst_adapter_map (wav->adapter, data_size);
1543 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1544 goto header_read_error;
1546 gst_adapter_unmap (wav->adapter);
1551 gst_buffer_unref (buf);
1554 gst_pad_pull_range (wav->sinkpad, wav->offset,
1555 data_size, &buf)) != GST_FLOW_OK)
1556 goto header_read_error;
1557 gst_buffer_map (buf, &map, GST_MAP_READ);
1558 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1560 goto header_read_error;
1562 gst_buffer_unmap (buf, &map);
1564 size = GST_ROUND_UP_2 (size);
1565 if (wav->streaming) {
1566 gst_adapter_flush (wav->adapter, size);
1568 gst_buffer_unref (buf);
1570 size = GST_ROUND_UP_2 (size);
1571 wav->offset += size;
1574 case GST_RIFF_TAG_smpl:{
1575 const gint data_size = size;
1577 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1578 if (wav->streaming) {
1579 const guint8 *data = NULL;
1581 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1584 gst_adapter_flush (wav->adapter, 8);
1586 data = gst_adapter_map (wav->adapter, data_size);
1587 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1588 goto header_read_error;
1590 gst_adapter_unmap (wav->adapter);
1595 gst_buffer_unref (buf);
1598 gst_pad_pull_range (wav->sinkpad, wav->offset,
1599 data_size, &buf)) != GST_FLOW_OK)
1600 goto header_read_error;
1601 gst_buffer_map (buf, &map, GST_MAP_READ);
1602 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1604 goto header_read_error;
1606 gst_buffer_unmap (buf, &map);
1608 size = GST_ROUND_UP_2 (size);
1609 if (wav->streaming) {
1610 gst_adapter_flush (wav->adapter, size);
1612 gst_buffer_unref (buf);
1614 size = GST_ROUND_UP_2 (size);
1615 wav->offset += size;
1619 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1620 GST_FOURCC_ARGS (tag));
1621 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1622 /* need more data */
1627 if (upstream_size && (wav->offset >= upstream_size)) {
1628 /* Now we are gone through the whole file */
1633 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1635 if (wav->bps <= 0 && wav->fact) {
1637 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1639 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1640 (guint64) wav->fact);
1641 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1646 if (gst_wavparse_calculate_duration (wav)) {
1647 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1648 if (!wav->ignore_length)
1649 wav->segment.duration = wav->duration;
1651 gst_wavparse_create_toc (wav);
1653 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1654 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1655 if (!wav->ignore_length)
1656 wav->segment.duration = wav->datasize;
1659 /* now we have all the info to perform a pending seek if any, if no
1660 * event, this will still do the right thing and it will also send
1661 * the right newsegment event downstream. */
1662 gst_wavparse_perform_seek (wav, wav->seek_event);
1663 /* remove pending event */
1664 gst_event_replace (&wav->seek_event, NULL);
1666 /* we just started, we are discont */
1667 wav->discont = TRUE;
1669 wav->state = GST_WAVPARSE_DATA;
1671 /* determine reasonable max buffer size,
1672 * that is, buffers not too small either size or time wise
1673 * so we do not end up with too many of them */
1675 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1676 wav->max_buf_size = upstream_size;
1678 wav->max_buf_size = 0;
1679 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1680 if (wav->blockalign > 0)
1681 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1683 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1690 g_free (codec_name);
1693 gst_caps_unref (caps);
1698 res = GST_FLOW_ERROR;
1703 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1704 ("Couldn't parse audio header"));
1709 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1710 ("Stream claims to contain no channels - invalid data"));
1715 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1716 ("Stream with sample_rate == 0 - invalid data"));
1721 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1722 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1723 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1728 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1729 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1730 wav->av_bps, wav->blockalign * wav->rate));
1733 no_bytes_per_sample:
1735 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1736 ("Could not caluclate bytes per sample - invalid data"));
1741 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1742 ("No caps found for format 0x%x, %u channels, %u Hz",
1743 wav->format, wav->channels, wav->rate));
1748 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1749 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1755 * Read WAV file tag when streaming
1757 static GstFlowReturn
1758 gst_wavparse_parse_stream_init (GstWavParse * wav)
1760 if (gst_adapter_available (wav->adapter) >= 12) {
1763 /* _take flushes the data */
1764 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1766 GST_DEBUG ("Parsing wav header");
1767 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1768 return GST_FLOW_ERROR;
1771 /* Go to next state */
1772 wav->state = GST_WAVPARSE_HEADER;
1777 /* handle an event sent directly to the element.
1779 * This event can be sent either in the READY state or the
1780 * >READY state. The only event of interest really is the seek
1783 * In the READY state we can only store the event and try to
1784 * respect it when going to PAUSED. We assume we are in the
1785 * READY state when our parsing state != GST_WAVPARSE_DATA.
1787 * When we are steaming, we can simply perform the seek right
1791 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1793 GstWavParse *wav = GST_WAVPARSE (element);
1794 gboolean res = FALSE;
1796 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1798 switch (GST_EVENT_TYPE (event)) {
1799 case GST_EVENT_SEEK:
1800 if (wav->state == GST_WAVPARSE_DATA) {
1801 /* we can handle the seek directly when streaming data */
1802 res = gst_wavparse_perform_seek (wav, event);
1804 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1806 gst_event_replace (&wav->seek_event, event);
1808 /* we always return true */
1815 gst_event_unref (event);
1820 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1824 s = gst_caps_get_structure (caps, 0);
1825 if (!gst_structure_has_name (s, "audio/x-dts"))
1827 /* typefind behavior for DTS:
1828 * MAXIMUM: multiple frame syncs detected, certainly DTS
1829 * LIKELY: single frame sync at offset 0. Maybe DTS?
1830 * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
1832 if (prob > GST_TYPE_FIND_LIKELY)
1834 if (prob <= GST_TYPE_FIND_POSSIBLE)
1836 /* for maybe, check for at least a valid-looking rate and channels */
1837 if (!gst_structure_has_field (s, "channels"))
1839 /* and for extra assurance we could also check the rate from the DTS frame
1840 * against the one in the wav header, but for now let's not do that */
1841 return gst_structure_has_field (s, "rate");
1845 gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
1847 GstTagList *tags = NULL;
1852 while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
1853 gst_event_parse_tag (ev, &tags);
1854 if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
1855 tags = gst_tag_list_copy (tags);
1856 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1857 gst_event_unref (ev);
1861 gst_event_unref (ev);
1867 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1870 GstTagList *tags, *utags;
1872 GST_DEBUG_OBJECT (wav, "adding src pad");
1874 g_assert (wav->caps != NULL);
1876 s = gst_caps_get_structure (wav->caps, 0);
1877 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1878 GstTypeFindProbability prob;
1881 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1882 if (tf_caps != NULL) {
1883 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1884 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1885 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1886 gst_caps_unref (wav->caps);
1887 wav->caps = tf_caps;
1889 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1890 GST_TAG_AUDIO_CODEC, "dts", NULL);
1892 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1893 "marked as raw PCM audio, but ignoring for now", tf_caps);
1894 gst_caps_unref (tf_caps);
1899 gst_pad_set_caps (wav->srcpad, wav->caps);
1900 gst_caps_replace (&wav->caps, NULL);
1902 if (wav->start_segment) {
1903 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1904 gst_pad_push_event (wav->srcpad, wav->start_segment);
1905 wav->start_segment = NULL;
1908 /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
1909 * that there'll be only one scope/type of tag list from upstream, if any */
1910 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
1912 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
1914 /* if there's a tag upstream it's probably been added to override the
1915 * tags from inside the wav header, so keep upstream tags if in doubt */
1916 tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
1918 if (wav->tags != NULL) {
1919 gst_tag_list_unref (wav->tags);
1924 gst_tag_list_unref (utags);
1926 /* send tags downstream, if any */
1928 gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
1931 static GstFlowReturn
1932 gst_wavparse_stream_data (GstWavParse * wav)
1934 GstBuffer *buf = NULL;
1935 GstFlowReturn res = GST_FLOW_OK;
1936 guint64 desired, obtained;
1937 GstClockTime timestamp, next_timestamp, duration;
1938 guint64 pos, nextpos;
1941 GST_LOG_OBJECT (wav,
1942 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1943 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1945 /* Get the next n bytes and output them */
1946 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1949 /* scale the amount of data by the segment rate so we get equal
1950 * amounts of data regardless of the playback rate */
1952 MIN (gst_guint64_to_gdouble (wav->dataleft),
1953 wav->max_buf_size * ABS (wav->segment.rate));
1955 if (desired >= wav->blockalign && wav->blockalign > 0)
1956 desired -= (desired % wav->blockalign);
1958 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1959 "from the sinkpad", desired);
1961 if (wav->streaming) {
1962 guint avail = gst_adapter_available (wav->adapter);
1965 /* flush some bytes if evil upstream sends segment that starts
1966 * before data or does is not send sample aligned segment */
1967 if (G_LIKELY (wav->offset >= wav->datastart)) {
1968 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1970 extra = wav->datastart - wav->offset;
1973 if (G_UNLIKELY (extra)) {
1974 extra = wav->bytes_per_sample - extra;
1975 if (extra <= avail) {
1976 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
1977 gst_adapter_flush (wav->adapter, extra);
1978 wav->offset += extra;
1979 wav->dataleft -= extra;
1980 goto iterate_adapter;
1982 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
1983 gst_adapter_clear (wav->adapter);
1984 wav->offset += avail;
1985 wav->dataleft -= avail;
1990 if (avail < desired) {
1991 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
1995 buf = gst_adapter_take_buffer (wav->adapter, desired);
1997 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
1998 desired, &buf)) != GST_FLOW_OK)
2001 /* we may get a short buffer at the end of the file */
2002 if (gst_buffer_get_size (buf) < desired) {
2003 gsize size = gst_buffer_get_size (buf);
2005 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2006 if (size >= wav->blockalign) {
2007 if (wav->blockalign > 0) {
2008 buf = gst_buffer_make_writable (buf);
2009 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2012 gst_buffer_unref (buf);
2018 obtained = gst_buffer_get_size (buf);
2020 /* our positions in bytes */
2021 pos = wav->offset - wav->datastart;
2022 nextpos = pos + obtained;
2024 /* update offsets, does not overflow. */
2025 buf = gst_buffer_make_writable (buf);
2026 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2027 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2029 /* first chunk of data? create the source pad. We do this only here so
2030 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2031 if (G_UNLIKELY (wav->first)) {
2033 /* this will also push the segment events */
2034 gst_wavparse_add_src_pad (wav, buf);
2036 /* If we have a pending start segment, send it now. */
2037 if (G_UNLIKELY (wav->start_segment != NULL)) {
2038 gst_pad_push_event (wav->srcpad, wav->start_segment);
2039 wav->start_segment = NULL;
2044 /* and timestamps if we have a bitrate, be careful for overflows */
2046 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2048 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2049 duration = next_timestamp - timestamp;
2051 /* update current running segment position */
2052 if (G_LIKELY (next_timestamp >= wav->segment.start))
2053 wav->segment.position = next_timestamp;
2054 } else if (wav->fact) {
2056 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2057 /* and timestamps if we have a bitrate, be careful for overflows */
2058 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2059 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2060 duration = next_timestamp - timestamp;
2062 /* no bitrate, all we know is that the first sample has timestamp 0, all
2063 * other positions and durations have unknown timestamp. */
2067 timestamp = GST_CLOCK_TIME_NONE;
2068 duration = GST_CLOCK_TIME_NONE;
2069 /* update current running segment position with byte offset */
2070 if (G_LIKELY (nextpos >= wav->segment.start))
2071 wav->segment.position = nextpos;
2073 if ((pos > 0) && wav->vbr) {
2074 /* don't set timestamps for VBR files if it's not the first buffer */
2075 timestamp = GST_CLOCK_TIME_NONE;
2076 duration = GST_CLOCK_TIME_NONE;
2079 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2080 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2081 wav->discont = FALSE;
2084 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2085 GST_BUFFER_DURATION (buf) = duration;
2087 GST_LOG_OBJECT (wav,
2088 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2089 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2090 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2092 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2095 if (obtained < wav->dataleft) {
2096 wav->offset += obtained;
2097 wav->dataleft -= obtained;
2099 wav->offset += wav->dataleft;
2103 /* Iterate until need more data, so adapter size won't grow */
2104 if (wav->streaming) {
2105 GST_LOG_OBJECT (wav,
2106 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2108 goto iterate_adapter;
2115 GST_DEBUG_OBJECT (wav, "found EOS");
2116 return GST_FLOW_EOS;
2120 /* check if we got EOS */
2121 if (res == GST_FLOW_EOS)
2124 GST_WARNING_OBJECT (wav,
2125 "Error getting %" G_GINT64_FORMAT " bytes from the "
2126 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2131 GST_INFO_OBJECT (wav,
2132 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2133 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2134 gst_pad_is_linked (wav->srcpad));
2140 gst_wavparse_loop (GstPad * pad)
2143 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2147 GST_LOG_OBJECT (wav, "process data");
2149 switch (wav->state) {
2150 case GST_WAVPARSE_START:
2151 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2152 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2156 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2157 event = gst_event_new_stream_start (stream_id);
2158 gst_event_set_group_id (event, gst_util_group_id_next ());
2159 gst_pad_push_event (wav->srcpad, event);
2162 wav->state = GST_WAVPARSE_HEADER;
2165 case GST_WAVPARSE_HEADER:
2166 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2167 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2170 wav->state = GST_WAVPARSE_DATA;
2171 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2174 case GST_WAVPARSE_DATA:
2175 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2179 g_assert_not_reached ();
2186 const gchar *reason = gst_flow_get_name (ret);
2188 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2189 gst_pad_pause_task (pad);
2191 if (ret == GST_FLOW_EOS) {
2192 /* handle end-of-stream/segment */
2193 /* so align our position with the end of it, if there is one
2194 * this ensures a subsequent will arrive at correct base/acc time */
2195 if (wav->segment.format == GST_FORMAT_TIME) {
2196 if (wav->segment.rate > 0.0 &&
2197 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2198 wav->segment.position = wav->segment.stop;
2199 else if (wav->segment.rate < 0.0)
2200 wav->segment.position = wav->segment.start;
2202 if (wav->state == GST_WAVPARSE_START) {
2203 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2204 ("No valid input found before end of stream"));
2205 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2207 /* add pad before we perform EOS */
2208 if (G_UNLIKELY (wav->first)) {
2210 gst_wavparse_add_src_pad (wav, NULL);
2213 /* perform EOS logic */
2214 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2217 if ((stop = wav->segment.stop) == -1)
2218 stop = wav->segment.duration;
2220 gst_element_post_message (GST_ELEMENT_CAST (wav),
2221 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2222 wav->segment.format, stop));
2223 gst_pad_push_event (wav->srcpad,
2224 gst_event_new_segment_done (wav->segment.format, stop));
2226 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2229 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2230 /* for fatal errors we post an error message, post the error
2231 * first so the app knows about the error first. */
2232 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2233 (_("Internal data flow error.")),
2234 ("streaming task paused, reason %s (%d)", reason, ret));
2235 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2241 static GstFlowReturn
2242 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2245 GstWavParse *wav = GST_WAVPARSE (parent);
2247 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2248 gst_buffer_get_size (buf));
2250 gst_adapter_push (wav->adapter, buf);
2252 switch (wav->state) {
2253 case GST_WAVPARSE_START:
2254 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2255 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2258 if (wav->state != GST_WAVPARSE_HEADER)
2261 /* otherwise fall-through */
2262 case GST_WAVPARSE_HEADER:
2263 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2264 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2267 if (!wav->got_fmt || wav->datastart == 0)
2270 wav->state = GST_WAVPARSE_DATA;
2271 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2274 case GST_WAVPARSE_DATA:
2275 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2276 wav->discont = TRUE;
2277 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2281 g_return_val_if_reached (GST_FLOW_ERROR);
2284 if (G_UNLIKELY (wav->abort_buffering)) {
2285 wav->abort_buffering = FALSE;
2286 ret = GST_FLOW_ERROR;
2287 /* sort of demux/parse error */
2288 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2294 static GstFlowReturn
2295 gst_wavparse_flush_data (GstWavParse * wav)
2297 GstFlowReturn ret = GST_FLOW_OK;
2300 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2301 ret = gst_wavparse_stream_data (wav);
2308 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2310 GstWavParse *wav = GST_WAVPARSE (parent);
2311 gboolean ret = TRUE;
2313 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2315 switch (GST_EVENT_TYPE (event)) {
2316 case GST_EVENT_CAPS:
2318 /* discard, we'll come up with proper src caps */
2319 gst_event_unref (event);
2322 case GST_EVENT_SEGMENT:
2324 gint64 start, stop, offset = 0, end_offset = -1;
2327 /* some debug output */
2328 gst_event_copy_segment (event, &segment);
2329 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2332 if (wav->state != GST_WAVPARSE_DATA) {
2333 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2337 /* now we are either committed to TIME or BYTE format,
2338 * and we only expect a BYTE segment, e.g. following a seek */
2339 if (segment.format == GST_FORMAT_BYTES) {
2340 /* handle (un)signed issues */
2341 start = segment.start;
2342 stop = segment.stop;
2345 start -= wav->datastart;
2346 start = MAX (start, 0);
2350 stop -= wav->datastart;
2351 stop = MAX (stop, 0);
2353 if (wav->segment.format == GST_FORMAT_TIME) {
2354 guint64 bps = wav->bps;
2356 /* operating in format TIME, so we can convert */
2357 if (!bps && wav->fact)
2359 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2363 gst_util_uint64_scale_ceil (start, GST_SECOND,
2364 (guint64) wav->bps);
2367 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2368 (guint64) wav->bps);
2372 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2376 segment.start = start;
2377 segment.stop = stop;
2379 /* accept upstream's notion of segment and distribute along */
2380 segment.format = wav->segment.format;
2381 segment.time = segment.position = segment.start;
2382 segment.duration = wav->segment.duration;
2383 segment.base = gst_segment_to_running_time (&wav->segment,
2384 GST_FORMAT_TIME, wav->segment.position);
2386 gst_segment_copy_into (&segment, &wav->segment);
2388 /* also store the newsegment event for the streaming thread */
2389 if (wav->start_segment)
2390 gst_event_unref (wav->start_segment);
2391 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2392 wav->start_segment = gst_event_new_segment (&segment);
2394 /* stream leftover data in current segment */
2395 gst_wavparse_flush_data (wav);
2396 /* and set up streaming thread for next one */
2397 wav->offset = offset;
2398 wav->end_offset = end_offset;
2400 if (wav->datasize > 0 && (wav->end_offset == -1
2401 || wav->end_offset > wav->datastart + wav->datasize))
2402 wav->end_offset = wav->datastart + wav->datasize;
2404 if (wav->end_offset != -1) {
2405 wav->dataleft = wav->end_offset - wav->offset;
2407 /* infinity; upstream will EOS when done */
2408 wav->dataleft = G_MAXUINT64;
2411 gst_event_unref (event);
2415 if (wav->state == GST_WAVPARSE_START) {
2416 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2417 ("No valid input found before end of stream"));
2419 /* add pad if needed so EOS is seen downstream */
2420 if (G_UNLIKELY (wav->first)) {
2422 gst_wavparse_add_src_pad (wav, NULL);
2424 /* stream leftover data in current segment */
2425 gst_wavparse_flush_data (wav);
2430 case GST_EVENT_FLUSH_STOP:
2434 gst_adapter_clear (wav->adapter);
2435 wav->discont = TRUE;
2436 dur = wav->segment.duration;
2437 gst_segment_init (&wav->segment, wav->segment.format);
2438 wav->segment.duration = dur;
2442 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2450 /* convert and query stuff */
2451 static const GstFormat *
2452 gst_wavparse_get_formats (GstPad * pad)
2454 static const GstFormat formats[] = {
2457 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2466 gst_wavparse_pad_convert (GstPad * pad,
2467 GstFormat src_format, gint64 src_value,
2468 GstFormat * dest_format, gint64 * dest_value)
2470 GstWavParse *wavparse;
2471 gboolean res = TRUE;
2473 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2475 if (*dest_format == src_format) {
2476 *dest_value = src_value;
2480 if ((wavparse->bps == 0) && !wavparse->fact)
2483 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2484 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2486 switch (src_format) {
2487 case GST_FORMAT_BYTES:
2488 switch (*dest_format) {
2489 case GST_FORMAT_DEFAULT:
2490 *dest_value = src_value / wavparse->bytes_per_sample;
2491 /* make sure we end up on a sample boundary */
2492 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2494 case GST_FORMAT_TIME:
2495 /* src_value + datastart = offset */
2496 GST_INFO_OBJECT (wavparse,
2497 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2499 if (wavparse->bps > 0)
2500 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2501 (guint64) wavparse->bps);
2502 else if (wavparse->fact) {
2503 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2504 wavparse->rate, wavparse->fact);
2507 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2518 case GST_FORMAT_DEFAULT:
2519 switch (*dest_format) {
2520 case GST_FORMAT_BYTES:
2521 *dest_value = src_value * wavparse->bytes_per_sample;
2523 case GST_FORMAT_TIME:
2524 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2525 (guint64) wavparse->rate);
2533 case GST_FORMAT_TIME:
2534 switch (*dest_format) {
2535 case GST_FORMAT_BYTES:
2536 if (wavparse->bps > 0)
2537 *dest_value = gst_util_uint64_scale (src_value,
2538 (guint64) wavparse->bps, GST_SECOND);
2540 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2541 wavparse->rate, wavparse->fact);
2543 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2545 /* make sure we end up on a sample boundary */
2546 *dest_value -= *dest_value % wavparse->blockalign;
2548 case GST_FORMAT_DEFAULT:
2549 *dest_value = gst_util_uint64_scale (src_value,
2550 (guint64) wavparse->rate, GST_SECOND);
2569 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2575 /* handle queries for location and length in requested format */
2577 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2579 gboolean res = TRUE;
2580 GstWavParse *wav = GST_WAVPARSE (parent);
2582 /* only if we know */
2583 if (wav->state != GST_WAVPARSE_DATA) {
2587 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2589 switch (GST_QUERY_TYPE (query)) {
2590 case GST_QUERY_POSITION:
2596 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2597 curb = wav->offset - wav->datastart;
2598 gst_query_parse_position (query, &format, NULL);
2599 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2602 case GST_FORMAT_BYTES:
2603 format = GST_FORMAT_BYTES;
2607 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2612 gst_query_set_position (query, format, cur);
2615 case GST_QUERY_DURATION:
2617 gint64 duration = 0;
2620 if (wav->ignore_length) {
2625 gst_query_parse_duration (query, &format, NULL);
2628 case GST_FORMAT_BYTES:{
2629 format = GST_FORMAT_BYTES;
2630 duration = wav->datasize;
2633 case GST_FORMAT_TIME:
2634 if ((res = gst_wavparse_calculate_duration (wav))) {
2635 duration = wav->duration;
2643 gst_query_set_duration (query, format, duration);
2646 case GST_QUERY_CONVERT:
2648 gint64 srcvalue, dstvalue;
2649 GstFormat srcformat, dstformat;
2651 gst_query_parse_convert (query, &srcformat, &srcvalue,
2652 &dstformat, &dstvalue);
2653 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2654 &dstformat, &dstvalue);
2656 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2659 case GST_QUERY_SEEKING:{
2661 gboolean seekable = FALSE;
2663 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2664 if (fmt == wav->segment.format) {
2665 if (wav->streaming) {
2668 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2669 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2670 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2671 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2673 gst_query_unref (q);
2675 GST_LOG_OBJECT (wav, "looping => seekable");
2679 } else if (fmt == GST_FORMAT_TIME) {
2683 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2688 res = gst_pad_query_default (pad, parent, query);
2695 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2697 GstWavParse *wavparse = GST_WAVPARSE (parent);
2698 gboolean res = FALSE;
2700 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2702 switch (GST_EVENT_TYPE (event)) {
2703 case GST_EVENT_SEEK:
2704 /* can only handle events when we are in the data state */
2705 if (wavparse->state == GST_WAVPARSE_DATA) {
2706 res = gst_wavparse_perform_seek (wavparse, event);
2708 gst_event_unref (event);
2711 case GST_EVENT_TOC_SELECT:
2714 GstTocEntry *entry = NULL;
2715 GstEvent *seek_event;
2718 if (!wavparse->toc) {
2719 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2722 gst_event_parse_toc_select (event, &uid);
2724 GST_OBJECT_LOCK (wavparse);
2725 entry = gst_toc_find_entry (wavparse->toc, uid);
2726 if (entry == NULL) {
2727 GST_OBJECT_UNLOCK (wavparse);
2728 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2732 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2733 GST_OBJECT_UNLOCK (wavparse);
2734 seek_event = gst_event_new_seek (1.0,
2736 GST_SEEK_FLAG_FLUSH,
2737 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2738 res = gst_wavparse_perform_seek (wavparse, seek_event);
2739 gst_event_unref (seek_event);
2743 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2747 gst_event_unref (event);
2752 res = gst_pad_push_event (wavparse->sinkpad, event);
2759 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2761 GstWavParse *wav = GST_WAVPARSE (parent);
2766 gst_adapter_clear (wav->adapter);
2767 g_object_unref (wav->adapter);
2768 wav->adapter = NULL;
2771 query = gst_query_new_scheduling ();
2773 if (!gst_pad_peer_query (sinkpad, query)) {
2774 gst_query_unref (query);
2778 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2779 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2780 gst_query_unref (query);
2785 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2786 wav->streaming = FALSE;
2787 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2791 GST_DEBUG_OBJECT (sinkpad, "activating push");
2792 wav->streaming = TRUE;
2793 wav->adapter = gst_adapter_new ();
2794 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2800 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2801 GstPadMode mode, gboolean active)
2806 case GST_PAD_MODE_PUSH:
2809 case GST_PAD_MODE_PULL:
2811 /* if we have a scheduler we can start the task */
2812 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2815 res = gst_pad_stop_task (sinkpad);
2825 static GstStateChangeReturn
2826 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2828 GstStateChangeReturn ret;
2829 GstWavParse *wav = GST_WAVPARSE (element);
2831 switch (transition) {
2832 case GST_STATE_CHANGE_NULL_TO_READY:
2834 case GST_STATE_CHANGE_READY_TO_PAUSED:
2835 gst_wavparse_reset (wav);
2837 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2843 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2845 switch (transition) {
2846 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2848 case GST_STATE_CHANGE_PAUSED_TO_READY:
2849 gst_wavparse_reset (wav);
2851 case GST_STATE_CHANGE_READY_TO_NULL:
2860 gst_wavparse_set_property (GObject * object, guint prop_id,
2861 const GValue * value, GParamSpec * pspec)
2865 g_return_if_fail (GST_IS_WAVPARSE (object));
2866 self = GST_WAVPARSE (object);
2869 case PROP_IGNORE_LENGTH:
2870 self->ignore_length = g_value_get_boolean (value);
2873 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2879 gst_wavparse_get_property (GObject * object, guint prop_id,
2880 GValue * value, GParamSpec * pspec)
2884 g_return_if_fail (GST_IS_WAVPARSE (object));
2885 self = GST_WAVPARSE (object);
2888 case PROP_IGNORE_LENGTH:
2889 g_value_set_boolean (value, self->ignore_length);
2892 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2897 plugin_init (GstPlugin * plugin)
2901 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2905 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2908 "Parse a .wav file into raw audio",
2909 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)