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.
41 * Last reviewed on 2007-02-14 (0.10.6)
46 * http://replaygain.hydrogenaudio.org/file_format_wav.html
56 #include "gstwavparse.h"
57 #include "gst/riff/riff-media.h"
58 #include <gst/base/gsttypefindhelper.h>
59 #include <gst/gst-i18n-plugin.h>
61 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
62 #define GST_CAT_DEFAULT (wavparse_debug)
64 #define GST_BWF_TAG_iXML GST_MAKE_FOURCC ('i','X','M','L')
65 #define GST_BWF_TAG_qlty GST_MAKE_FOURCC ('q','l','t','y')
66 #define GST_BWF_TAG_mext GST_MAKE_FOURCC ('m','e','x','t')
67 #define GST_BWF_TAG_levl GST_MAKE_FOURCC ('l','e','v','l')
68 #define GST_BWF_TAG_link GST_MAKE_FOURCC ('l','i','n','k')
69 #define GST_BWF_TAG_axml GST_MAKE_FOURCC ('a','x','m','l')
71 static void gst_wavparse_dispose (GObject * object);
73 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
75 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
76 GstObject * parent, GstPadMode mode, gboolean active);
77 static gboolean gst_wavparse_send_event (GstElement * element,
79 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
80 GstStateChange transition);
82 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
84 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
85 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
87 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
89 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
91 static void gst_wavparse_loop (GstPad * pad);
92 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
95 static void gst_wavparse_set_property (GObject * object, guint prop_id,
96 const GValue * value, GParamSpec * pspec);
97 static void gst_wavparse_get_property (GObject * object, guint prop_id,
98 GValue * value, GParamSpec * pspec);
100 #define DEFAULT_IGNORE_LENGTH FALSE
108 static GstStaticPadTemplate sink_template_factory =
109 GST_STATIC_PAD_TEMPLATE ("sink",
112 GST_STATIC_CAPS ("audio/x-wav")
116 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
118 #define gst_wavparse_parent_class parent_class
119 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
124 /* Offset Size Description Value
125 * 0x00 4 ID unique identification value
126 * 0x04 4 Position play order position
127 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
128 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
129 * 0x10 4 Block Start Byte Offset to sample of First Channel
130 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
134 guint32 data_chunk_id;
137 guint32 sample_offset;
142 /* Offset Size Description Value
143 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
146 guint32 cue_point_id;
148 } GstWavParseLabl, GstWavParseNote;
151 gst_wavparse_class_init (GstWavParseClass * klass)
153 GstElementClass *gstelement_class;
154 GObjectClass *object_class;
155 GstPadTemplate *src_template;
157 gstelement_class = (GstElementClass *) klass;
158 object_class = (GObjectClass *) klass;
160 parent_class = g_type_class_peek_parent (klass);
162 object_class->dispose = gst_wavparse_dispose;
164 object_class->set_property = gst_wavparse_set_property;
165 object_class->get_property = gst_wavparse_get_property;
168 * GstWavParse:ignore-length:
170 * This selects whether the length found in a data chunk
171 * should be ignored. This may be useful for streamed audio
172 * where the length is unknown until the end of streaming,
173 * and various software/hardware just puts some random value
174 * in there and hopes it doesn't break too much.
176 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
177 g_param_spec_boolean ("ignore-length",
179 "Ignore length from the Wave header",
180 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
183 gstelement_class->change_state = gst_wavparse_change_state;
184 gstelement_class->send_event = gst_wavparse_send_event;
187 gst_element_class_add_pad_template (gstelement_class,
188 gst_static_pad_template_get (&sink_template_factory));
190 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
191 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
192 gst_element_class_add_pad_template (gstelement_class, src_template);
194 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
195 "Codec/Demuxer/Audio",
196 "Parse a .wav file into raw audio",
197 "Erik Walthinsen <omega@cse.ogi.edu>");
201 gst_wavparse_reset (GstWavParse * wav)
203 wav->state = GST_WAVPARSE_START;
205 /* These will all be set correctly in the fmt chunk */
219 wav->got_fmt = FALSE;
223 gst_event_unref (wav->seek_event);
224 wav->seek_event = NULL;
226 gst_adapter_clear (wav->adapter);
227 g_object_unref (wav->adapter);
231 gst_tag_list_unref (wav->tags);
234 gst_toc_unref (wav->toc);
237 g_list_free_full (wav->cues, g_free);
240 g_list_free_full (wav->labls, g_free);
243 gst_caps_unref (wav->caps);
245 if (wav->start_segment)
246 gst_event_unref (wav->start_segment);
247 wav->start_segment = NULL;
251 gst_wavparse_dispose (GObject * object)
253 GstWavParse *wav = GST_WAVPARSE (object);
255 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
256 gst_wavparse_reset (wav);
258 G_OBJECT_CLASS (parent_class)->dispose (object);
262 gst_wavparse_init (GstWavParse * wavparse)
264 gst_wavparse_reset (wavparse);
268 gst_pad_new_from_static_template (&sink_template_factory, "sink");
269 gst_pad_set_activate_function (wavparse->sinkpad,
270 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
271 gst_pad_set_activatemode_function (wavparse->sinkpad,
272 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
273 gst_pad_set_chain_function (wavparse->sinkpad,
274 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
275 gst_pad_set_event_function (wavparse->sinkpad,
276 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
277 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
281 gst_pad_new_from_template (gst_element_class_get_pad_template
282 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
283 gst_pad_use_fixed_caps (wavparse->srcpad);
284 gst_pad_set_query_function (wavparse->srcpad,
285 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
286 gst_pad_set_event_function (wavparse->srcpad,
287 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
288 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
292 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
296 if (!gst_riff_parse_file_header (element, buf, &doctype))
299 if (doctype != GST_RIFF_RIFF_WAVE)
307 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
308 ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
309 GST_FOURCC_ARGS (doctype)));
315 gst_wavparse_stream_init (GstWavParse * wav)
318 GstBuffer *buf = NULL;
320 if ((res = gst_pad_pull_range (wav->sinkpad,
321 wav->offset, 12, &buf)) != GST_FLOW_OK)
323 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
324 return GST_FLOW_ERROR;
332 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
334 /* -1 always maps to -1 */
340 /* 0 always maps to 0 */
347 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
349 } else if (wav->fact) {
351 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
352 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
359 /* This function is used to perform seeks on the element.
361 * It also works when event is NULL, in which case it will just
362 * start from the last configured segment. This technique is
363 * used when activating the element and to perform the seek in
367 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
371 GstFormat format, bformat;
373 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
374 gint64 cur, stop, upstream_size;
377 GstSegment seeksegment = { 0, };
381 GST_DEBUG_OBJECT (wav, "doing seek with event");
383 gst_event_parse_seek (event, &rate, &format, &flags,
384 &cur_type, &cur, &stop_type, &stop);
386 /* no negative rates yet */
390 if (format != wav->segment.format) {
391 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
392 gst_format_get_name (format),
393 gst_format_get_name (wav->segment.format));
395 if (cur_type != GST_SEEK_TYPE_NONE)
397 gst_pad_query_convert (wav->srcpad, format, cur,
398 wav->segment.format, &cur);
399 if (res && stop_type != GST_SEEK_TYPE_NONE)
401 gst_pad_query_convert (wav->srcpad, format, stop,
402 wav->segment.format, &stop);
406 format = wav->segment.format;
409 GST_DEBUG_OBJECT (wav, "doing seek without event");
412 cur_type = GST_SEEK_TYPE_SET;
413 stop_type = GST_SEEK_TYPE_SET;
416 /* in push mode, we must delegate to upstream */
417 if (wav->streaming) {
418 gboolean res = FALSE;
420 /* if streaming not yet started; only prepare initial newsegment */
421 if (!event || wav->state != GST_WAVPARSE_DATA) {
422 if (wav->start_segment)
423 gst_event_unref (wav->start_segment);
424 wav->start_segment = gst_event_new_segment (&wav->segment);
427 /* convert seek positions to byte positions in data sections */
428 if (format == GST_FORMAT_TIME) {
429 /* should not fail */
430 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
432 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
435 /* mind sample boundary and header */
437 cur -= (cur % wav->bytes_per_sample);
438 cur += wav->datastart;
441 stop -= (stop % wav->bytes_per_sample);
442 stop += wav->datastart;
444 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
445 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
447 /* BYTE seek event */
448 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
450 res = gst_pad_push_event (wav->sinkpad, event);
456 flush = flags & GST_SEEK_FLAG_FLUSH;
458 /* now we need to make sure the streaming thread is stopped. We do this by
459 * either sending a FLUSH_START event downstream which will cause the
460 * streaming thread to stop with a WRONG_STATE.
461 * For a non-flushing seek we simply pause the task, which will happen as soon
462 * as it completes one iteration (and thus might block when the sink is
463 * blocking in preroll). */
465 GST_DEBUG_OBJECT (wav, "sending flush start");
466 gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
468 gst_pad_pause_task (wav->sinkpad);
471 /* we should now be able to grab the streaming thread because we stopped it
472 * with the above flush/pause code */
473 GST_PAD_STREAM_LOCK (wav->sinkpad);
475 /* save current position */
476 last_stop = wav->segment.position;
478 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
480 /* copy segment, we need this because we still need the old
481 * segment when we close the current segment. */
482 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
484 /* configure the seek parameters in the seeksegment. We will then have the
485 * right values in the segment to perform the seek */
487 GST_DEBUG_OBJECT (wav, "configuring seek");
488 gst_segment_do_seek (&seeksegment, rate, format, flags,
489 cur_type, cur, stop_type, stop, &update);
492 /* figure out the last position we need to play. If it's configured (stop !=
493 * -1), use that, else we play until the total duration of the file */
494 if ((stop = seeksegment.stop) == -1)
495 stop = seeksegment.duration;
497 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
498 if ((cur_type != GST_SEEK_TYPE_NONE)) {
499 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
500 * we can just copy the last_stop. If not, we use the bps to convert TIME to
502 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
503 (gint64 *) & wav->offset))
504 wav->offset = seeksegment.position;
505 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
506 wav->offset -= (wav->offset % wav->bytes_per_sample);
507 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
508 wav->offset += wav->datastart;
509 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
511 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
515 if (stop_type != GST_SEEK_TYPE_NONE) {
516 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
517 wav->end_offset = stop;
518 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
519 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
520 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
521 wav->end_offset += wav->datastart;
522 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
524 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
528 /* make sure filesize is not exceeded due to rounding errors or so,
529 * same precaution as in _stream_headers */
530 bformat = GST_FORMAT_BYTES;
531 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
532 wav->end_offset = MIN (wav->end_offset, upstream_size);
534 /* this is the range of bytes we will use for playback */
535 wav->offset = MIN (wav->offset, wav->end_offset);
536 wav->dataleft = wav->end_offset - wav->offset;
538 GST_DEBUG_OBJECT (wav,
539 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
540 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
541 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
543 /* prepare for streaming again */
545 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
546 GST_DEBUG_OBJECT (wav, "sending flush stop");
547 gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
550 /* now we did the seek and can activate the new segment values */
551 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
553 /* if we're doing a segment seek, post a SEGMENT_START message */
554 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
555 gst_element_post_message (GST_ELEMENT_CAST (wav),
556 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
557 wav->segment.format, wav->segment.position));
560 /* now create the newsegment */
561 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
562 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
564 /* store the newsegment event so it can be sent from the streaming thread. */
565 if (wav->start_segment)
566 gst_event_unref (wav->start_segment);
567 wav->start_segment = gst_event_new_segment (&wav->segment);
569 /* mark discont if we are going to stream from another position. */
570 if (last_stop != wav->segment.position) {
571 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
575 /* and start the streaming task again */
576 if (!wav->streaming) {
577 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
581 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
588 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
593 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
598 GST_DEBUG_OBJECT (wav,
599 "Could not determine byte position for desired time");
605 * gst_wavparse_peek_chunk_info:
606 * @wav Wavparse object
607 * @tag holder for tag
608 * @size holder for tag size
610 * Peek next chunk info (tag and size)
612 * Returns: %TRUE when the chunk info (header) is available
615 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
617 const guint8 *data = NULL;
619 if (gst_adapter_available (wav->adapter) < 8)
622 data = gst_adapter_map (wav->adapter, 8);
623 *tag = GST_READ_UINT32_LE (data);
624 *size = GST_READ_UINT32_LE (data + 4);
625 gst_adapter_unmap (wav->adapter);
627 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
628 GST_FOURCC_ARGS (*tag));
634 * gst_wavparse_peek_chunk:
635 * @wav Wavparse object
636 * @tag holder for tag
637 * @size holder for tag size
639 * Peek enough data for one full chunk
641 * Returns: %TRUE when the full chunk is available
644 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
646 guint32 peek_size = 0;
649 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
652 /* size 0 -> empty data buffer would surprise most callers,
653 * large size -> do not bother trying to squeeze that into adapter,
654 * so we throw poor man's exception, which can be caught if caller really
655 * wants to handle 0 size chunk */
656 if (!(*size) || (*size) >= (1 << 30)) {
657 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
658 *size, GST_FOURCC_ARGS (*tag));
659 /* chain should give up */
660 wav->abort_buffering = TRUE;
663 peek_size = (*size + 1) & ~1;
664 available = gst_adapter_available (wav->adapter);
666 if (available >= (8 + peek_size)) {
669 GST_LOG ("but only %u bytes available now", available);
675 * gst_wavparse_calculate_duration:
676 * @wav: wavparse object
678 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
681 * Returns: %TRUE if duration is available.
684 gst_wavparse_calculate_duration (GstWavParse * wav)
686 if (wav->duration > 0)
690 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
692 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
694 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
695 GST_TIME_ARGS (wav->duration));
697 } else if (wav->fact) {
699 gst_util_uint64_scale_int_ceil (GST_SECOND, wav->fact, wav->rate);
700 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
701 GST_TIME_ARGS (wav->duration));
708 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
713 if (wav->streaming) {
714 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
717 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
718 GST_FOURCC_ARGS (tag));
719 flush = 8 + ((size + 1) & ~1);
720 wav->offset += flush;
721 if (wav->streaming) {
722 gst_adapter_flush (wav->adapter, flush);
724 gst_buffer_unref (buf);
731 * gst_wavparse_cue_chunk:
732 * @wav GstWavParse object
733 * @data holder for data
734 * @size holder for data size
736 * Parse cue chunk from @data to wav->cues.
738 * Returns: %TRUE when cue chunk is available
741 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
748 GST_WARNING_OBJECT (wav, "found another cue's");
752 ncues = GST_READ_UINT32_LE (data);
754 if (size < 4 + ncues * 24) {
755 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
761 for (i = 0; i < ncues; i++) {
762 cue = g_new0 (GstWavParseCue, 1);
763 cue->id = GST_READ_UINT32_LE (data);
764 cue->position = GST_READ_UINT32_LE (data + 4);
765 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
766 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
767 cue->block_start = GST_READ_UINT32_LE (data + 16);
768 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
769 cues = g_list_append (cues, cue);
779 * gst_wavparse_labl_chunk:
780 * @wav GstWavParse object
781 * @data holder for data
782 * @size holder for data size
784 * Parse labl from @data to wav->labls.
786 * Returns: %TRUE when labl chunk is available
789 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
791 GstWavParseLabl *labl;
796 labl = g_new0 (GstWavParseLabl, 1);
800 labl->cue_point_id = GST_READ_UINT32_LE (data);
801 labl->text = g_memdup (data + 4, size - 4);
803 wav->labls = g_list_append (wav->labls, labl);
809 * gst_wavparse_note_chunk:
810 * @wav GstWavParse object
811 * @data holder for data
812 * @size holder for data size
814 * Parse note from @data to wav->notes.
816 * Returns: %TRUE when note chunk is available
819 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
821 GstWavParseNote *note;
826 note = g_new0 (GstWavParseNote, 1);
830 note->cue_point_id = GST_READ_UINT32_LE (data);
831 note->text = g_memdup (data + 4, size - 4);
833 wav->notes = g_list_append (wav->notes, note);
839 * gst_wavparse_smpl_chunk:
840 * @wav GstWavParse object
841 * @data holder for data
842 * @size holder for data size
844 * Parse smpl chunk from @data.
846 * Returns: %TRUE when cue chunk is available
849 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
854 manufacturer_id = GST_READ_UINT32_LE (data);
855 product_id = GST_READ_UINT32_LE (data + 4);
856 sample_period = GST_READ_UINT32_LE (data + 8);
858 note_number = GST_READ_UINT32_LE (data + 12);
860 pitch_fraction = GST_READ_UINT32_LE (data + 16);
861 SMPTE_format = GST_READ_UINT32_LE (data + 20);
862 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
863 num_sample_loops = GST_READ_UINT32_LE (data + 28);
864 List of Sample Loops, 24 bytes each
868 wav->tags = gst_tag_list_new_empty ();
869 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
870 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
875 * gst_wavparse_adtl_chunk:
876 * @wav GstWavParse object
877 * @data holder for data
878 * @size holder for data size
880 * Parse adtl from @data.
882 * Returns: %TRUE when adtl chunk is available
885 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
887 guint32 ltag, lsize, offset = 0;
890 ltag = GST_READ_UINT32_LE (data + offset);
891 lsize = GST_READ_UINT32_LE (data + offset + 4);
893 case GST_RIFF_TAG_labl:
894 gst_wavparse_labl_chunk (wav, data + offset, size);
896 case GST_RIFF_TAG_note:
897 gst_wavparse_note_chunk (wav, data + offset, size);
900 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
901 GST_FOURCC_ARGS (ltag));
902 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
905 offset += 8 + GST_ROUND_UP_2 (lsize);
906 size -= 8 + GST_ROUND_UP_2 (lsize);
913 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
915 GstTagList *tags = NULL;
916 GstTocEntry *entry = NULL;
918 entry = gst_toc_find_entry (toc, id);
920 tags = gst_toc_entry_get_tags (entry);
922 tags = gst_tag_list_new_empty ();
923 gst_toc_entry_set_tags (entry, tags);
931 * gst_wavparse_create_toc:
932 * @wav GstWavParse object
934 * Create TOC from wav->cues and wav->labls.
937 gst_wavparse_create_toc (GstWavParse * wav)
943 GstWavParseLabl *labl;
944 GstWavParseNote *note;
947 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
949 GST_OBJECT_LOCK (wav);
951 GST_OBJECT_UNLOCK (wav);
952 GST_WARNING_OBJECT (wav, "found another TOC");
957 GST_OBJECT_UNLOCK (wav);
961 /* FIXME: send CURRENT scope toc too */
962 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
964 /* add cue edition */
965 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
966 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
967 gst_toc_append_entry (toc, entry);
969 /* add tracks in cue edition */
973 prev_subentry = cur_subentry;
974 /* previous track stop time = current track start time */
975 if (prev_subentry != NULL) {
976 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
977 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
978 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
980 id = g_strdup_printf ("%08x", cue->id);
981 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
983 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
984 stop = wav->duration;
985 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
986 gst_toc_entry_append_sub_entry (entry, cur_subentry);
987 list = g_list_next (list);
990 /* add tags in tracks */
994 id = g_strdup_printf ("%08x", labl->cue_point_id);
995 tags = gst_wavparse_get_tags_toc_entry (toc, id);
998 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1001 list = g_list_next (list);
1006 id = g_strdup_printf ("%08x", note->cue_point_id);
1007 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1010 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1013 list = g_list_next (list);
1016 /* send data as TOC */
1019 /* send TOC event */
1021 GST_OBJECT_UNLOCK (wav);
1022 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1028 #define MAX_BUFFER_SIZE 4096
1030 static GstFlowReturn
1031 gst_wavparse_stream_headers (GstWavParse * wav)
1033 GstFlowReturn res = GST_FLOW_OK;
1034 GstBuffer *buf = NULL;
1035 gst_riff_strf_auds *header = NULL;
1037 gboolean gotdata = FALSE;
1038 GstCaps *caps = NULL;
1039 gchar *codec_name = NULL;
1041 gint64 upstream_size = 0;
1043 /* search for "_fmt" chunk, which should be first */
1044 while (!wav->got_fmt) {
1047 /* The header starts with a 'fmt ' tag */
1048 if (wav->streaming) {
1049 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1052 gst_adapter_flush (wav->adapter, 8);
1056 buf = gst_adapter_take_buffer (wav->adapter, size);
1058 gst_adapter_flush (wav->adapter, 1);
1059 wav->offset += GST_ROUND_UP_2 (size);
1061 buf = gst_buffer_new ();
1064 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1065 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1069 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1070 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1071 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1072 tag == GST_RIFF_TAG_id3 || tag == GST_RIFF_TAG_IDVX ||
1073 tag == GST_BWF_TAG_iXML || tag == GST_BWF_TAG_qlty ||
1074 tag == GST_BWF_TAG_mext || tag == GST_BWF_TAG_levl ||
1075 tag == GST_BWF_TAG_link || tag == GST_BWF_TAG_axml) {
1076 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1077 GST_FOURCC_ARGS (tag));
1078 gst_buffer_unref (buf);
1083 if (tag != GST_RIFF_TAG_fmt)
1086 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1088 goto parse_header_error;
1090 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1092 /* do sanity checks of header fields */
1093 if (header->channels == 0)
1095 if (header->rate == 0)
1098 GST_DEBUG_OBJECT (wav, "creating the caps");
1100 /* Note: gst_riff_create_audio_caps might need to fix values in
1101 * the header header depending on the format, so call it first */
1102 /* FIXME: Need to handle the channel reorder map */
1103 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1104 NULL, &codec_name, NULL);
1107 gst_buffer_unref (extra);
1110 goto unknown_format;
1112 /* do more sanity checks of header fields
1113 * (these can be sanitized by gst_riff_create_audio_caps()
1115 wav->format = header->format;
1116 wav->rate = header->rate;
1117 wav->channels = header->channels;
1118 wav->blockalign = header->blockalign;
1119 wav->depth = header->bits_per_sample;
1120 wav->av_bps = header->av_bps;
1126 /* do format specific handling */
1127 switch (wav->format) {
1128 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1129 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1131 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1132 * bitrate inside the mpeg stream */
1133 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1137 case GST_RIFF_WAVE_FORMAT_PCM:
1138 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1139 goto invalid_blockalign;
1142 if (wav->av_bps > wav->blockalign * wav->rate)
1144 /* use the configured bps */
1145 wav->bps = wav->av_bps;
1149 wav->width = (wav->blockalign * 8) / wav->channels;
1150 wav->bytes_per_sample = wav->channels * wav->width / 8;
1152 if (wav->bytes_per_sample <= 0)
1153 goto no_bytes_per_sample;
1155 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1156 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1157 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1158 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1159 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1160 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1161 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1163 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1164 * formats). This will make the element output a BYTE format segment and
1165 * will not timestamp the outgoing buffers.
1167 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1169 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1171 /* create pad later so we can sniff the first few bytes
1172 * of the real data and correct our caps if necessary */
1173 gst_caps_replace (&wav->caps, caps);
1174 gst_caps_replace (&caps, NULL);
1176 wav->got_fmt = TRUE;
1179 wav->tags = gst_tag_list_new_empty ();
1181 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1182 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1184 g_free (codec_name);
1190 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1191 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1193 /* loop headers until we get data */
1195 if (wav->streaming) {
1196 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1203 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1204 &buf)) != GST_FLOW_OK)
1205 goto header_read_error;
1206 gst_buffer_map (buf, &map, GST_MAP_READ);
1207 tag = GST_READ_UINT32_LE (map.data);
1208 size = GST_READ_UINT32_LE (map.data + 4);
1209 gst_buffer_unmap (buf, &map);
1212 GST_INFO_OBJECT (wav,
1213 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1214 GST_FOURCC_ARGS (tag), wav->offset);
1216 /* wav is a st00pid format, we don't know for sure where data starts.
1217 * So we have to go bit by bit until we find the 'data' header
1220 case GST_RIFF_TAG_data:{
1221 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1222 if (wav->ignore_length) {
1223 GST_DEBUG_OBJECT (wav, "Ignoring length");
1226 if (wav->streaming) {
1227 gst_adapter_flush (wav->adapter, 8);
1230 gst_buffer_unref (buf);
1233 wav->datastart = wav->offset;
1234 /* If size is zero, then the data chunk probably actually extends to
1235 the end of the file */
1236 if (size == 0 && upstream_size) {
1237 size = upstream_size - wav->datastart;
1239 /* Or the file might be truncated */
1240 else if (upstream_size) {
1241 size = MIN (size, (upstream_size - wav->datastart));
1243 wav->datasize = (guint64) size;
1244 wav->dataleft = (guint64) size;
1245 wav->end_offset = size + wav->datastart;
1246 if (!wav->streaming) {
1247 /* We will continue parsing tags 'till end */
1248 wav->offset += size;
1250 GST_DEBUG_OBJECT (wav, "datasize = %u", size);
1253 case GST_RIFF_TAG_fact:{
1254 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1255 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1256 const guint data_size = 4;
1258 GST_INFO_OBJECT (wav, "Have fact chunk");
1259 if (size < data_size) {
1260 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1261 /* need more data */
1264 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1268 /* number of samples (for compressed formats) */
1269 if (wav->streaming) {
1270 const guint8 *data = NULL;
1272 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1275 gst_adapter_flush (wav->adapter, 8);
1276 data = gst_adapter_map (wav->adapter, data_size);
1277 wav->fact = GST_READ_UINT32_LE (data);
1278 gst_adapter_unmap (wav->adapter);
1279 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1281 gst_buffer_unref (buf);
1284 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1285 data_size, &buf)) != GST_FLOW_OK)
1286 goto header_read_error;
1287 gst_buffer_extract (buf, 0, &wav->fact, 4);
1288 wav->fact = GUINT32_FROM_LE (wav->fact);
1289 gst_buffer_unref (buf);
1291 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1292 wav->offset += 8 + GST_ROUND_UP_2 (size);
1295 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1296 /* need more data */
1302 case GST_RIFF_TAG_acid:{
1303 const gst_riff_acid *acid = NULL;
1304 const guint data_size = sizeof (gst_riff_acid);
1307 GST_INFO_OBJECT (wav, "Have acid chunk");
1308 if (size < data_size) {
1309 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1310 /* need more data */
1313 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1317 if (wav->streaming) {
1318 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1321 gst_adapter_flush (wav->adapter, 8);
1322 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1324 tempo = acid->tempo;
1325 gst_adapter_unmap (wav->adapter);
1328 gst_buffer_unref (buf);
1331 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1332 size, &buf)) != GST_FLOW_OK)
1333 goto header_read_error;
1334 gst_buffer_map (buf, &map, GST_MAP_READ);
1335 acid = (const gst_riff_acid *) map.data;
1336 tempo = acid->tempo;
1337 gst_buffer_unmap (buf, &map);
1339 /* send data as tags */
1341 wav->tags = gst_tag_list_new_empty ();
1342 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1343 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1345 size = GST_ROUND_UP_2 (size);
1346 if (wav->streaming) {
1347 gst_adapter_flush (wav->adapter, size);
1349 gst_buffer_unref (buf);
1351 wav->offset += 8 + size;
1354 /* FIXME: all list tags after data are ignored in streaming mode */
1355 case GST_RIFF_TAG_LIST:{
1358 if (wav->streaming) {
1359 const guint8 *data = NULL;
1361 if (gst_adapter_available (wav->adapter) < 12) {
1364 data = gst_adapter_map (wav->adapter, 12);
1365 ltag = GST_READ_UINT32_LE (data + 8);
1366 gst_adapter_unmap (wav->adapter);
1368 gst_buffer_unref (buf);
1371 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1372 &buf)) != GST_FLOW_OK)
1373 goto header_read_error;
1374 gst_buffer_extract (buf, 8, <ag, 4);
1375 ltag = GUINT32_FROM_LE (ltag);
1378 case GST_RIFF_LIST_INFO:{
1379 const gint data_size = size - 4;
1382 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1383 if (wav->streaming) {
1384 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1387 gst_adapter_flush (wav->adapter, 12);
1389 if (data_size > 0) {
1390 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1392 gst_adapter_flush (wav->adapter, 1);
1396 gst_buffer_unref (buf);
1398 if (data_size > 0) {
1400 gst_pad_pull_range (wav->sinkpad, wav->offset,
1401 data_size, &buf)) != GST_FLOW_OK)
1402 goto header_read_error;
1405 if (data_size > 0) {
1407 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1409 GstTagList *old = wav->tags;
1411 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1413 gst_tag_list_unref (old);
1414 gst_tag_list_unref (new);
1416 gst_buffer_unref (buf);
1417 wav->offset += GST_ROUND_UP_2 (data_size);
1421 case GST_RIFF_LIST_adtl:{
1422 const gint data_size = size;
1424 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1425 if (wav->streaming) {
1426 const guint8 *data = NULL;
1428 gst_adapter_flush (wav->adapter, 12);
1429 data = gst_adapter_map (wav->adapter, data_size);
1430 gst_wavparse_adtl_chunk (wav, data, data_size);
1431 gst_adapter_unmap (wav->adapter);
1435 gst_buffer_unref (buf);
1438 gst_pad_pull_range (wav->sinkpad, wav->offset + 12,
1439 data_size, &buf)) != GST_FLOW_OK)
1440 goto header_read_error;
1441 gst_buffer_map (buf, &map, GST_MAP_READ);
1442 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1444 gst_buffer_unmap (buf, &map);
1449 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1450 GST_FOURCC_ARGS (ltag));
1451 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1452 /* need more data */
1458 case GST_RIFF_TAG_cue:{
1459 const guint data_size = size;
1461 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1462 if (wav->streaming) {
1463 const guint8 *data = NULL;
1465 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1468 gst_adapter_flush (wav->adapter, 8);
1470 data = gst_adapter_map (wav->adapter, data_size);
1471 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1472 goto header_read_error;
1474 gst_adapter_unmap (wav->adapter);
1479 gst_buffer_unref (buf);
1482 gst_pad_pull_range (wav->sinkpad, wav->offset,
1483 data_size, &buf)) != GST_FLOW_OK)
1484 goto header_read_error;
1485 gst_buffer_map (buf, &map, GST_MAP_READ);
1486 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1488 goto header_read_error;
1490 gst_buffer_unmap (buf, &map);
1492 size = GST_ROUND_UP_2 (size);
1493 if (wav->streaming) {
1494 gst_adapter_flush (wav->adapter, size);
1496 gst_buffer_unref (buf);
1498 size = GST_ROUND_UP_2 (size);
1499 wav->offset += size;
1502 case GST_RIFF_TAG_smpl:{
1503 const gint data_size = size;
1505 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1506 if (wav->streaming) {
1507 const guint8 *data = NULL;
1509 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1512 gst_adapter_flush (wav->adapter, 8);
1514 data = gst_adapter_map (wav->adapter, data_size);
1515 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1516 goto header_read_error;
1518 gst_adapter_unmap (wav->adapter);
1523 gst_buffer_unref (buf);
1526 gst_pad_pull_range (wav->sinkpad, wav->offset,
1527 data_size, &buf)) != GST_FLOW_OK)
1528 goto header_read_error;
1529 gst_buffer_map (buf, &map, GST_MAP_READ);
1530 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1532 goto header_read_error;
1534 gst_buffer_unmap (buf, &map);
1536 size = GST_ROUND_UP_2 (size);
1537 if (wav->streaming) {
1538 gst_adapter_flush (wav->adapter, size);
1540 gst_buffer_unref (buf);
1542 size = GST_ROUND_UP_2 (size);
1543 wav->offset += size;
1547 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1548 GST_FOURCC_ARGS (tag));
1549 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1550 /* need more data */
1555 if (upstream_size && (wav->offset >= upstream_size)) {
1556 /* Now we are gone through the whole file */
1561 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1563 if (wav->bps <= 0 && wav->fact) {
1565 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1567 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1568 (guint64) wav->fact);
1569 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1574 if (gst_wavparse_calculate_duration (wav)) {
1575 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1576 if (!wav->ignore_length)
1577 wav->segment.duration = wav->duration;
1579 gst_wavparse_create_toc (wav);
1581 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1582 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1583 if (!wav->ignore_length)
1584 wav->segment.duration = wav->datasize;
1587 /* now we have all the info to perform a pending seek if any, if no
1588 * event, this will still do the right thing and it will also send
1589 * the right newsegment event downstream. */
1590 gst_wavparse_perform_seek (wav, wav->seek_event);
1591 /* remove pending event */
1592 event_p = &wav->seek_event;
1593 gst_event_replace (event_p, NULL);
1595 /* we just started, we are discont */
1596 wav->discont = TRUE;
1598 wav->state = GST_WAVPARSE_DATA;
1600 /* determine reasonable max buffer size,
1601 * that is, buffers not too small either size or time wise
1602 * so we do not end up with too many of them */
1605 gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
1606 wav->max_buf_size = upstream_size;
1607 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1608 if (wav->blockalign > 0)
1609 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1611 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1619 g_free (codec_name);
1623 gst_caps_unref (caps);
1628 res = GST_FLOW_ERROR;
1633 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1634 ("Invalid WAV header (no fmt at start): %"
1635 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1640 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1641 ("Couldn't parse audio header"));
1646 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1647 ("Stream claims to contain no channels - invalid data"));
1652 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1653 ("Stream with sample_rate == 0 - invalid data"));
1658 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1659 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1660 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1665 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1666 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1667 wav->av_bps, wav->blockalign * wav->rate));
1670 no_bytes_per_sample:
1672 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1673 ("Could not caluclate bytes per sample - invalid data"));
1678 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1679 ("No caps found for format 0x%x, %u channels, %u Hz",
1680 wav->format, wav->channels, wav->rate));
1685 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1686 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1692 * Read WAV file tag when streaming
1694 static GstFlowReturn
1695 gst_wavparse_parse_stream_init (GstWavParse * wav)
1697 if (gst_adapter_available (wav->adapter) >= 12) {
1700 /* _take flushes the data */
1701 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1703 GST_DEBUG ("Parsing wav header");
1704 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1705 return GST_FLOW_ERROR;
1708 /* Go to next state */
1709 wav->state = GST_WAVPARSE_HEADER;
1714 /* handle an event sent directly to the element.
1716 * This event can be sent either in the READY state or the
1717 * >READY state. The only event of interest really is the seek
1720 * In the READY state we can only store the event and try to
1721 * respect it when going to PAUSED. We assume we are in the
1722 * READY state when our parsing state != GST_WAVPARSE_DATA.
1724 * When we are steaming, we can simply perform the seek right
1728 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1730 GstWavParse *wav = GST_WAVPARSE (element);
1731 gboolean res = FALSE;
1734 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1736 switch (GST_EVENT_TYPE (event)) {
1737 case GST_EVENT_SEEK:
1738 if (wav->state == GST_WAVPARSE_DATA) {
1739 /* we can handle the seek directly when streaming data */
1740 res = gst_wavparse_perform_seek (wav, event);
1742 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1744 event_p = &wav->seek_event;
1745 gst_event_replace (event_p, event);
1747 /* we always return true */
1754 gst_event_unref (event);
1759 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1763 s = gst_caps_get_structure (caps, 0);
1764 if (!gst_structure_has_name (s, "audio/x-dts"))
1766 if (prob >= GST_TYPE_FIND_LIKELY)
1768 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
1769 if (prob < GST_TYPE_FIND_POSSIBLE)
1771 /* .. in which case we want at least a valid-looking rate and channels */
1772 if (!gst_structure_has_field (s, "channels"))
1774 /* and for extra assurance we could also check the rate from the DTS frame
1775 * against the one in the wav header, but for now let's not do that */
1776 return gst_structure_has_field (s, "rate");
1780 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1784 GST_DEBUG_OBJECT (wav, "adding src pad");
1786 g_assert (wav->caps != NULL);
1788 s = gst_caps_get_structure (wav->caps, 0);
1789 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1790 GstTypeFindProbability prob;
1793 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1794 if (tf_caps != NULL) {
1795 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1796 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1797 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1798 gst_caps_unref (wav->caps);
1799 wav->caps = tf_caps;
1801 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1802 GST_TAG_AUDIO_CODEC, "dts", NULL);
1804 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1805 "marked as raw PCM audio, but ignoring for now", tf_caps);
1806 gst_caps_unref (tf_caps);
1811 gst_pad_set_caps (wav->srcpad, wav->caps);
1812 gst_caps_replace (&wav->caps, NULL);
1814 if (wav->start_segment) {
1815 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1816 gst_pad_push_event (wav->srcpad, wav->start_segment);
1817 wav->start_segment = NULL;
1821 gst_pad_push_event (wav->srcpad, gst_event_new_tag (wav->tags));
1826 static GstFlowReturn
1827 gst_wavparse_stream_data (GstWavParse * wav)
1829 GstBuffer *buf = NULL;
1830 GstFlowReturn res = GST_FLOW_OK;
1831 guint64 desired, obtained;
1832 GstClockTime timestamp, next_timestamp, duration;
1833 guint64 pos, nextpos;
1836 GST_LOG_OBJECT (wav,
1837 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1838 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1840 /* Get the next n bytes and output them */
1841 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1844 /* scale the amount of data by the segment rate so we get equal
1845 * amounts of data regardless of the playback rate */
1847 MIN (gst_guint64_to_gdouble (wav->dataleft),
1848 wav->max_buf_size * ABS (wav->segment.rate));
1850 if (desired >= wav->blockalign && wav->blockalign > 0)
1851 desired -= (desired % wav->blockalign);
1853 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1854 "from the sinkpad", desired);
1856 if (wav->streaming) {
1857 guint avail = gst_adapter_available (wav->adapter);
1860 /* flush some bytes if evil upstream sends segment that starts
1861 * before data or does is not send sample aligned segment */
1862 if (G_LIKELY (wav->offset >= wav->datastart)) {
1863 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1865 extra = wav->datastart - wav->offset;
1868 if (G_UNLIKELY (extra)) {
1869 extra = wav->bytes_per_sample - extra;
1870 if (extra <= avail) {
1871 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
1872 gst_adapter_flush (wav->adapter, extra);
1873 wav->offset += extra;
1874 wav->dataleft -= extra;
1875 goto iterate_adapter;
1877 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
1878 gst_adapter_clear (wav->adapter);
1879 wav->offset += avail;
1880 wav->dataleft -= avail;
1885 if (avail < desired) {
1886 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
1890 buf = gst_adapter_take_buffer (wav->adapter, desired);
1892 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
1893 desired, &buf)) != GST_FLOW_OK)
1896 /* we may get a short buffer at the end of the file */
1897 if (gst_buffer_get_size (buf) < desired) {
1898 gsize size = gst_buffer_get_size (buf);
1900 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
1901 if (size >= wav->blockalign) {
1902 buf = gst_buffer_make_writable (buf);
1903 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
1905 gst_buffer_unref (buf);
1911 obtained = gst_buffer_get_size (buf);
1913 /* our positions in bytes */
1914 pos = wav->offset - wav->datastart;
1915 nextpos = pos + obtained;
1917 /* update offsets, does not overflow. */
1918 buf = gst_buffer_make_writable (buf);
1919 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
1920 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
1922 /* first chunk of data? create the source pad. We do this only here so
1923 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
1924 if (G_UNLIKELY (wav->first)) {
1926 /* this will also push the segment events */
1927 gst_wavparse_add_src_pad (wav, buf);
1929 /* If we have a pending start segment, send it now. */
1930 if (G_UNLIKELY (wav->start_segment != NULL)) {
1931 gst_pad_push_event (wav->srcpad, wav->start_segment);
1932 wav->start_segment = NULL;
1937 /* and timestamps if we have a bitrate, be careful for overflows */
1939 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
1941 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
1942 duration = next_timestamp - timestamp;
1944 /* update current running segment position */
1945 if (G_LIKELY (next_timestamp >= wav->segment.start))
1946 wav->segment.position = next_timestamp;
1947 } else if (wav->fact) {
1949 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
1950 /* and timestamps if we have a bitrate, be careful for overflows */
1951 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
1952 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
1953 duration = next_timestamp - timestamp;
1955 /* no bitrate, all we know is that the first sample has timestamp 0, all
1956 * other positions and durations have unknown timestamp. */
1960 timestamp = GST_CLOCK_TIME_NONE;
1961 duration = GST_CLOCK_TIME_NONE;
1962 /* update current running segment position with byte offset */
1963 if (G_LIKELY (nextpos >= wav->segment.start))
1964 wav->segment.position = nextpos;
1966 if ((pos > 0) && wav->vbr) {
1967 /* don't set timestamps for VBR files if it's not the first buffer */
1968 timestamp = GST_CLOCK_TIME_NONE;
1969 duration = GST_CLOCK_TIME_NONE;
1972 GST_DEBUG_OBJECT (wav, "marking DISCONT");
1973 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
1974 wav->discont = FALSE;
1977 GST_BUFFER_TIMESTAMP (buf) = timestamp;
1978 GST_BUFFER_DURATION (buf) = duration;
1980 GST_LOG_OBJECT (wav,
1981 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
1982 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
1983 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
1985 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
1988 if (obtained < wav->dataleft) {
1989 wav->offset += obtained;
1990 wav->dataleft -= obtained;
1992 wav->offset += wav->dataleft;
1996 /* Iterate until need more data, so adapter size won't grow */
1997 if (wav->streaming) {
1998 GST_LOG_OBJECT (wav,
1999 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2001 goto iterate_adapter;
2008 GST_DEBUG_OBJECT (wav, "found EOS");
2009 return GST_FLOW_EOS;
2013 /* check if we got EOS */
2014 if (res == GST_FLOW_EOS)
2017 GST_WARNING_OBJECT (wav,
2018 "Error getting %" G_GINT64_FORMAT " bytes from the "
2019 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2024 GST_INFO_OBJECT (wav,
2025 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2026 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2027 gst_pad_is_linked (wav->srcpad));
2033 gst_wavparse_loop (GstPad * pad)
2036 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2040 GST_LOG_OBJECT (wav, "process data");
2042 switch (wav->state) {
2043 case GST_WAVPARSE_START:
2044 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2045 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2049 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2050 event = gst_event_new_stream_start (stream_id);
2051 gst_event_set_group_id (event, gst_util_group_id_next ());
2052 gst_pad_push_event (wav->srcpad, event);
2055 wav->state = GST_WAVPARSE_HEADER;
2058 case GST_WAVPARSE_HEADER:
2059 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2060 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2063 wav->state = GST_WAVPARSE_DATA;
2064 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2067 case GST_WAVPARSE_DATA:
2068 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2072 g_assert_not_reached ();
2079 const gchar *reason = gst_flow_get_name (ret);
2081 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2082 gst_pad_pause_task (pad);
2084 if (ret == GST_FLOW_EOS) {
2085 /* handle end-of-stream/segment */
2086 /* so align our position with the end of it, if there is one
2087 * this ensures a subsequent will arrive at correct base/acc time */
2088 if (wav->segment.format == GST_FORMAT_TIME) {
2089 if (wav->segment.rate > 0.0 &&
2090 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2091 wav->segment.position = wav->segment.stop;
2092 else if (wav->segment.rate < 0.0)
2093 wav->segment.position = wav->segment.start;
2095 if (wav->state == GST_WAVPARSE_START) {
2096 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2097 ("No valid input found before end of stream"));
2098 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2100 /* add pad before we perform EOS */
2101 if (G_UNLIKELY (wav->first)) {
2103 gst_wavparse_add_src_pad (wav, NULL);
2106 /* perform EOS logic */
2107 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2110 if ((stop = wav->segment.stop) == -1)
2111 stop = wav->segment.duration;
2113 gst_element_post_message (GST_ELEMENT_CAST (wav),
2114 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2115 wav->segment.format, stop));
2116 gst_pad_push_event (wav->srcpad,
2117 gst_event_new_segment_done (wav->segment.format, stop));
2119 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2122 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2123 /* for fatal errors we post an error message, post the error
2124 * first so the app knows about the error first. */
2125 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2126 (_("Internal data flow error.")),
2127 ("streaming task paused, reason %s (%d)", reason, ret));
2128 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2134 static GstFlowReturn
2135 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2138 GstWavParse *wav = GST_WAVPARSE (parent);
2140 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2141 gst_buffer_get_size (buf));
2143 gst_adapter_push (wav->adapter, buf);
2145 switch (wav->state) {
2146 case GST_WAVPARSE_START:
2147 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2148 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2151 if (wav->state != GST_WAVPARSE_HEADER)
2154 /* otherwise fall-through */
2155 case GST_WAVPARSE_HEADER:
2156 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2157 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2160 if (!wav->got_fmt || wav->datastart == 0)
2163 wav->state = GST_WAVPARSE_DATA;
2164 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2167 case GST_WAVPARSE_DATA:
2168 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2169 wav->discont = TRUE;
2170 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2174 g_return_val_if_reached (GST_FLOW_ERROR);
2177 if (G_UNLIKELY (wav->abort_buffering)) {
2178 wav->abort_buffering = FALSE;
2179 ret = GST_FLOW_ERROR;
2180 /* sort of demux/parse error */
2181 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2187 static GstFlowReturn
2188 gst_wavparse_flush_data (GstWavParse * wav)
2190 GstFlowReturn ret = GST_FLOW_OK;
2193 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2195 wav->end_offset = wav->offset + av;
2196 ret = gst_wavparse_stream_data (wav);
2203 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2205 GstWavParse *wav = GST_WAVPARSE (parent);
2206 gboolean ret = TRUE;
2208 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2210 switch (GST_EVENT_TYPE (event)) {
2211 case GST_EVENT_CAPS:
2213 /* discard, we'll come up with proper src caps */
2214 gst_event_unref (event);
2217 case GST_EVENT_SEGMENT:
2219 gint64 start, stop, offset = 0, end_offset = -1;
2222 /* some debug output */
2223 gst_event_copy_segment (event, &segment);
2224 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2227 if (wav->state != GST_WAVPARSE_DATA) {
2228 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2232 /* now we are either committed to TIME or BYTE format,
2233 * and we only expect a BYTE segment, e.g. following a seek */
2234 if (segment.format == GST_FORMAT_BYTES) {
2235 /* handle (un)signed issues */
2236 start = segment.start;
2237 stop = segment.stop;
2240 start -= wav->datastart;
2241 start = MAX (start, 0);
2245 segment.stop -= wav->datastart;
2246 segment.stop = MAX (stop, 0);
2248 if (wav->segment.format == GST_FORMAT_TIME) {
2249 guint64 bps = wav->bps;
2251 /* operating in format TIME, so we can convert */
2252 if (!bps && wav->fact)
2254 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2258 gst_util_uint64_scale_ceil (start, GST_SECOND,
2259 (guint64) wav->bps);
2262 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2263 (guint64) wav->bps);
2267 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2271 segment.start = start;
2272 segment.stop = stop;
2274 /* accept upstream's notion of segment and distribute along */
2275 segment.format = wav->segment.format;
2276 segment.time = segment.position = segment.start;
2277 segment.duration = wav->segment.duration;
2278 segment.base = gst_segment_to_running_time (&wav->segment,
2279 GST_FORMAT_TIME, wav->segment.position);
2281 gst_segment_copy_into (&segment, &wav->segment);
2283 /* also store the newsegment event for the streaming thread */
2284 if (wav->start_segment)
2285 gst_event_unref (wav->start_segment);
2286 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2287 wav->start_segment = gst_event_new_segment (&segment);
2289 /* stream leftover data in current segment */
2290 gst_wavparse_flush_data (wav);
2291 /* and set up streaming thread for next one */
2292 wav->offset = offset;
2293 wav->end_offset = end_offset;
2294 if (wav->end_offset > 0) {
2295 wav->dataleft = wav->end_offset - wav->offset;
2297 /* infinity; upstream will EOS when done */
2298 wav->dataleft = G_MAXUINT64;
2301 gst_event_unref (event);
2305 if (wav->state == GST_WAVPARSE_START) {
2306 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2307 ("No valid input found before end of stream"));
2309 /* add pad if needed so EOS is seen downstream */
2310 if (G_UNLIKELY (wav->first)) {
2312 gst_wavparse_add_src_pad (wav, NULL);
2314 /* stream leftover data in current segment */
2315 gst_wavparse_flush_data (wav);
2320 case GST_EVENT_FLUSH_STOP:
2324 gst_adapter_clear (wav->adapter);
2325 wav->discont = TRUE;
2326 dur = wav->segment.duration;
2327 gst_segment_init (&wav->segment, wav->segment.format);
2328 wav->segment.duration = dur;
2332 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2340 /* convert and query stuff */
2341 static const GstFormat *
2342 gst_wavparse_get_formats (GstPad * pad)
2344 static GstFormat formats[] = {
2347 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2356 gst_wavparse_pad_convert (GstPad * pad,
2357 GstFormat src_format, gint64 src_value,
2358 GstFormat * dest_format, gint64 * dest_value)
2360 GstWavParse *wavparse;
2361 gboolean res = TRUE;
2363 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2365 if (*dest_format == src_format) {
2366 *dest_value = src_value;
2370 if ((wavparse->bps == 0) && !wavparse->fact)
2373 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2374 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2376 switch (src_format) {
2377 case GST_FORMAT_BYTES:
2378 switch (*dest_format) {
2379 case GST_FORMAT_DEFAULT:
2380 *dest_value = src_value / wavparse->bytes_per_sample;
2381 /* make sure we end up on a sample boundary */
2382 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2384 case GST_FORMAT_TIME:
2385 /* src_value + datastart = offset */
2386 GST_INFO_OBJECT (wavparse,
2387 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2389 if (wavparse->bps > 0)
2390 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2391 (guint64) wavparse->bps);
2392 else if (wavparse->fact) {
2393 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2394 wavparse->rate, wavparse->fact);
2397 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2408 case GST_FORMAT_DEFAULT:
2409 switch (*dest_format) {
2410 case GST_FORMAT_BYTES:
2411 *dest_value = src_value * wavparse->bytes_per_sample;
2413 case GST_FORMAT_TIME:
2414 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2415 (guint64) wavparse->rate);
2423 case GST_FORMAT_TIME:
2424 switch (*dest_format) {
2425 case GST_FORMAT_BYTES:
2426 if (wavparse->bps > 0)
2427 *dest_value = gst_util_uint64_scale (src_value,
2428 (guint64) wavparse->bps, GST_SECOND);
2430 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2431 wavparse->rate, wavparse->fact);
2433 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2435 /* make sure we end up on a sample boundary */
2436 *dest_value -= *dest_value % wavparse->blockalign;
2438 case GST_FORMAT_DEFAULT:
2439 *dest_value = gst_util_uint64_scale (src_value,
2440 (guint64) wavparse->rate, GST_SECOND);
2459 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2465 /* handle queries for location and length in requested format */
2467 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2469 gboolean res = TRUE;
2470 GstWavParse *wav = GST_WAVPARSE (parent);
2472 /* only if we know */
2473 if (wav->state != GST_WAVPARSE_DATA) {
2477 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2479 switch (GST_QUERY_TYPE (query)) {
2480 case GST_QUERY_POSITION:
2486 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2487 curb = wav->offset - wav->datastart;
2488 gst_query_parse_position (query, &format, NULL);
2489 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2492 case GST_FORMAT_BYTES:
2493 format = GST_FORMAT_BYTES;
2497 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2502 gst_query_set_position (query, format, cur);
2505 case GST_QUERY_DURATION:
2507 gint64 duration = 0;
2510 if (wav->ignore_length) {
2515 gst_query_parse_duration (query, &format, NULL);
2518 case GST_FORMAT_BYTES:{
2519 format = GST_FORMAT_BYTES;
2520 duration = wav->datasize;
2523 case GST_FORMAT_TIME:
2524 if ((res = gst_wavparse_calculate_duration (wav))) {
2525 duration = wav->duration;
2533 gst_query_set_duration (query, format, duration);
2536 case GST_QUERY_CONVERT:
2538 gint64 srcvalue, dstvalue;
2539 GstFormat srcformat, dstformat;
2541 gst_query_parse_convert (query, &srcformat, &srcvalue,
2542 &dstformat, &dstvalue);
2543 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2544 &dstformat, &dstvalue);
2546 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2549 case GST_QUERY_SEEKING:{
2551 gboolean seekable = FALSE;
2553 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2554 if (fmt == wav->segment.format) {
2555 if (wav->streaming) {
2558 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2559 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2560 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2561 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2563 gst_query_unref (q);
2565 GST_LOG_OBJECT (wav, "looping => seekable");
2569 } else if (fmt == GST_FORMAT_TIME) {
2573 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2578 res = gst_pad_query_default (pad, parent, query);
2585 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2587 GstWavParse *wavparse = GST_WAVPARSE (parent);
2588 gboolean res = FALSE;
2590 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2592 switch (GST_EVENT_TYPE (event)) {
2593 case GST_EVENT_SEEK:
2594 /* can only handle events when we are in the data state */
2595 if (wavparse->state == GST_WAVPARSE_DATA) {
2596 res = gst_wavparse_perform_seek (wavparse, event);
2598 gst_event_unref (event);
2601 case GST_EVENT_TOC_SELECT:
2604 GstTocEntry *entry = NULL;
2605 GstEvent *seek_event;
2608 if (!wavparse->toc) {
2609 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2612 gst_event_parse_toc_select (event, &uid);
2614 GST_OBJECT_LOCK (wavparse);
2615 entry = gst_toc_find_entry (wavparse->toc, uid);
2616 if (entry == NULL) {
2617 GST_OBJECT_UNLOCK (wavparse);
2618 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2622 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2623 GST_OBJECT_UNLOCK (wavparse);
2624 seek_event = gst_event_new_seek (1.0,
2626 GST_SEEK_FLAG_FLUSH,
2627 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2628 res = gst_wavparse_perform_seek (wavparse, seek_event);
2629 gst_event_unref (seek_event);
2633 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2637 gst_event_unref (event);
2642 res = gst_pad_push_event (wavparse->sinkpad, event);
2649 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2651 GstWavParse *wav = GST_WAVPARSE (parent);
2656 gst_adapter_clear (wav->adapter);
2657 g_object_unref (wav->adapter);
2658 wav->adapter = NULL;
2661 query = gst_query_new_scheduling ();
2663 if (!gst_pad_peer_query (sinkpad, query)) {
2664 gst_query_unref (query);
2668 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2669 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2670 gst_query_unref (query);
2675 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2676 wav->streaming = FALSE;
2677 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2681 GST_DEBUG_OBJECT (sinkpad, "activating push");
2682 wav->streaming = TRUE;
2683 wav->adapter = gst_adapter_new ();
2684 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2690 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2691 GstPadMode mode, gboolean active)
2696 case GST_PAD_MODE_PUSH:
2699 case GST_PAD_MODE_PULL:
2701 /* if we have a scheduler we can start the task */
2702 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2705 res = gst_pad_stop_task (sinkpad);
2715 static GstStateChangeReturn
2716 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2718 GstStateChangeReturn ret;
2719 GstWavParse *wav = GST_WAVPARSE (element);
2721 switch (transition) {
2722 case GST_STATE_CHANGE_NULL_TO_READY:
2724 case GST_STATE_CHANGE_READY_TO_PAUSED:
2725 gst_wavparse_reset (wav);
2727 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2733 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2735 switch (transition) {
2736 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2738 case GST_STATE_CHANGE_PAUSED_TO_READY:
2739 gst_wavparse_reset (wav);
2741 case GST_STATE_CHANGE_READY_TO_NULL:
2750 gst_wavparse_set_property (GObject * object, guint prop_id,
2751 const GValue * value, GParamSpec * pspec)
2755 g_return_if_fail (GST_IS_WAVPARSE (object));
2756 self = GST_WAVPARSE (object);
2759 case PROP_IGNORE_LENGTH:
2760 self->ignore_length = g_value_get_boolean (value);
2763 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2769 gst_wavparse_get_property (GObject * object, guint prop_id,
2770 GValue * value, GParamSpec * pspec)
2774 g_return_if_fail (GST_IS_WAVPARSE (object));
2775 self = GST_WAVPARSE (object);
2778 case PROP_IGNORE_LENGTH:
2779 g_value_set_boolean (value, self->ignore_length);
2782 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2787 plugin_init (GstPlugin * plugin)
2791 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2795 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2798 "Parse a .wav file into raw audio",
2799 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)