1 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
3 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
4 * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-wavparse
25 * Parse a .wav file into raw or compressed audio.
27 * Wavparse supports both push and pull mode operations, making it possible to
28 * stream from a network source.
31 * <title>Example launch line</title>
33 * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
34 * ]| Read a wav file and output to the soundcard using the ALSA element. The
35 * wav file is assumed to contain raw uncompressed samples.
37 * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
38 * ]| Stream data from a network url.
44 * http://replaygain.hydrogenaudio.org/file_format_wav.html
54 #include "gstwavparse.h"
55 #include "gst/riff/riff-media.h"
56 #include <gst/base/gsttypefindhelper.h>
57 #include <gst/gst-i18n-plugin.h>
59 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
60 #define GST_CAT_DEFAULT (wavparse_debug)
62 #define GST_RIFF_TAG_Fake GST_MAKE_FOURCC ('F','a','k','e')
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 /* Data size chunk of RF64,
72 * see http://tech.ebu.ch/docs/tech/tech3306-2009.pdf */
73 #define GST_RS64_TAG_DS64 GST_MAKE_FOURCC ('d','s','6','4')
75 static void gst_wavparse_dispose (GObject * object);
77 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
79 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
80 GstObject * parent, GstPadMode mode, gboolean active);
81 static gboolean gst_wavparse_send_event (GstElement * element,
83 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
84 GstStateChange transition);
86 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
88 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
89 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
91 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
93 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
95 static void gst_wavparse_loop (GstPad * pad);
96 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
99 static void gst_wavparse_set_property (GObject * object, guint prop_id,
100 const GValue * value, GParamSpec * pspec);
101 static void gst_wavparse_get_property (GObject * object, guint prop_id,
102 GValue * value, GParamSpec * pspec);
104 #define DEFAULT_IGNORE_LENGTH FALSE
112 static GstStaticPadTemplate sink_template_factory =
113 GST_STATIC_PAD_TEMPLATE ("sink",
116 GST_STATIC_CAPS ("audio/x-wav")
120 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
122 #define gst_wavparse_parent_class parent_class
123 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
128 /* Offset Size Description Value
129 * 0x00 4 ID unique identification value
130 * 0x04 4 Position play order position
131 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
132 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
133 * 0x10 4 Block Start Byte Offset to sample of First Channel
134 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
138 guint32 data_chunk_id;
141 guint32 sample_offset;
146 /* Offset Size Description Value
147 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
150 guint32 cue_point_id;
152 } GstWavParseLabl, GstWavParseNote;
155 gst_wavparse_class_init (GstWavParseClass * klass)
157 GstElementClass *gstelement_class;
158 GObjectClass *object_class;
159 GstPadTemplate *src_template;
161 gstelement_class = (GstElementClass *) klass;
162 object_class = (GObjectClass *) klass;
164 parent_class = g_type_class_peek_parent (klass);
166 object_class->dispose = gst_wavparse_dispose;
168 object_class->set_property = gst_wavparse_set_property;
169 object_class->get_property = gst_wavparse_get_property;
172 * GstWavParse:ignore-length:
174 * This selects whether the length found in a data chunk
175 * should be ignored. This may be useful for streamed audio
176 * where the length is unknown until the end of streaming,
177 * and various software/hardware just puts some random value
178 * in there and hopes it doesn't break too much.
180 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
181 g_param_spec_boolean ("ignore-length",
183 "Ignore length from the Wave header",
184 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
187 gstelement_class->change_state = gst_wavparse_change_state;
188 gstelement_class->send_event = gst_wavparse_send_event;
191 gst_element_class_add_pad_template (gstelement_class,
192 gst_static_pad_template_get (&sink_template_factory));
194 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
195 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
196 gst_element_class_add_pad_template (gstelement_class, src_template);
198 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
199 "Codec/Demuxer/Audio",
200 "Parse a .wav file into raw audio",
201 "Erik Walthinsen <omega@cse.ogi.edu>");
205 gst_wavparse_reset (GstWavParse * wav)
207 wav->state = GST_WAVPARSE_START;
209 /* These will all be set correctly in the fmt chunk */
223 wav->got_fmt = FALSE;
227 gst_event_unref (wav->seek_event);
228 wav->seek_event = NULL;
230 gst_adapter_clear (wav->adapter);
231 g_object_unref (wav->adapter);
235 gst_tag_list_unref (wav->tags);
238 gst_toc_unref (wav->toc);
241 g_list_free_full (wav->cues, g_free);
244 g_list_free_full (wav->labls, g_free);
247 gst_caps_unref (wav->caps);
249 if (wav->start_segment)
250 gst_event_unref (wav->start_segment);
251 wav->start_segment = NULL;
255 gst_wavparse_dispose (GObject * object)
257 GstWavParse *wav = GST_WAVPARSE (object);
259 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
260 gst_wavparse_reset (wav);
262 G_OBJECT_CLASS (parent_class)->dispose (object);
266 gst_wavparse_init (GstWavParse * wavparse)
268 gst_wavparse_reset (wavparse);
272 gst_pad_new_from_static_template (&sink_template_factory, "sink");
273 gst_pad_set_activate_function (wavparse->sinkpad,
274 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
275 gst_pad_set_activatemode_function (wavparse->sinkpad,
276 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
277 gst_pad_set_chain_function (wavparse->sinkpad,
278 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
279 gst_pad_set_event_function (wavparse->sinkpad,
280 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
281 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
285 gst_pad_new_from_template (gst_element_class_get_pad_template
286 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
287 gst_pad_use_fixed_caps (wavparse->srcpad);
288 gst_pad_set_query_function (wavparse->srcpad,
289 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
290 gst_pad_set_event_function (wavparse->srcpad,
291 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
292 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
296 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
300 if (!gst_riff_parse_file_header (element, buf, &doctype))
303 if (doctype != GST_RIFF_RIFF_WAVE)
311 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
312 ("File is not a WAVE file: 0x%" G_GINT32_MODIFIER "x", doctype));
318 gst_wavparse_stream_init (GstWavParse * wav)
321 GstBuffer *buf = NULL;
323 if ((res = gst_pad_pull_range (wav->sinkpad,
324 wav->offset, 12, &buf)) != GST_FLOW_OK)
326 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
327 return GST_FLOW_ERROR;
335 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
337 /* -1 always maps to -1 */
343 /* 0 always maps to 0 */
350 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
352 } else if (wav->fact) {
353 guint64 bps = gst_util_uint64_scale (wav->datasize, wav->rate, wav->fact);
354 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
361 /* This function is used to perform seeks on the element.
363 * It also works when event is NULL, in which case it will just
364 * start from the last configured segment. This technique is
365 * used when activating the element and to perform the seek in
369 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
373 GstFormat format, bformat;
375 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
376 gint64 cur, stop, upstream_size;
379 GstSegment seeksegment = { 0, };
384 GST_DEBUG_OBJECT (wav, "doing seek with event");
386 gst_event_parse_seek (event, &rate, &format, &flags,
387 &cur_type, &cur, &stop_type, &stop);
388 seqnum = gst_event_get_seqnum (event);
390 /* no negative rates yet */
394 if (format != wav->segment.format) {
395 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
396 gst_format_get_name (format),
397 gst_format_get_name (wav->segment.format));
399 if (cur_type != GST_SEEK_TYPE_NONE)
401 gst_pad_query_convert (wav->srcpad, format, cur,
402 wav->segment.format, &cur);
403 if (res && stop_type != GST_SEEK_TYPE_NONE)
405 gst_pad_query_convert (wav->srcpad, format, stop,
406 wav->segment.format, &stop);
410 format = wav->segment.format;
413 GST_DEBUG_OBJECT (wav, "doing seek without event");
416 cur_type = GST_SEEK_TYPE_SET;
417 stop_type = GST_SEEK_TYPE_SET;
420 /* in push mode, we must delegate to upstream */
421 if (wav->streaming) {
422 gboolean res = FALSE;
424 /* if streaming not yet started; only prepare initial newsegment */
425 if (!event || wav->state != GST_WAVPARSE_DATA) {
426 if (wav->start_segment)
427 gst_event_unref (wav->start_segment);
428 wav->start_segment = gst_event_new_segment (&wav->segment);
431 /* convert seek positions to byte positions in data sections */
432 if (format == GST_FORMAT_TIME) {
433 /* should not fail */
434 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
436 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
439 /* mind sample boundary and header */
441 cur -= (cur % wav->bytes_per_sample);
442 cur += wav->datastart;
445 stop -= (stop % wav->bytes_per_sample);
446 stop += wav->datastart;
448 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
449 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
451 /* BYTE seek event */
452 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
454 gst_event_set_seqnum (event, seqnum);
455 res = gst_pad_push_event (wav->sinkpad, event);
461 flush = flags & GST_SEEK_FLAG_FLUSH;
463 /* now we need to make sure the streaming thread is stopped. We do this by
464 * either sending a FLUSH_START event downstream which will cause the
465 * streaming thread to stop with a WRONG_STATE.
466 * For a non-flushing seek we simply pause the task, which will happen as soon
467 * as it completes one iteration (and thus might block when the sink is
468 * blocking in preroll). */
471 GST_DEBUG_OBJECT (wav, "sending flush start");
473 fevent = gst_event_new_flush_start ();
474 gst_event_set_seqnum (fevent, seqnum);
475 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
476 gst_pad_push_event (wav->srcpad, fevent);
478 gst_pad_pause_task (wav->sinkpad);
481 /* we should now be able to grab the streaming thread because we stopped it
482 * with the above flush/pause code */
483 GST_PAD_STREAM_LOCK (wav->sinkpad);
485 /* save current position */
486 last_stop = wav->segment.position;
488 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
490 /* copy segment, we need this because we still need the old
491 * segment when we close the current segment. */
492 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
494 /* configure the seek parameters in the seeksegment. We will then have the
495 * right values in the segment to perform the seek */
497 GST_DEBUG_OBJECT (wav, "configuring seek");
498 gst_segment_do_seek (&seeksegment, rate, format, flags,
499 cur_type, cur, stop_type, stop, &update);
502 /* figure out the last position we need to play. If it's configured (stop !=
503 * -1), use that, else we play until the total duration of the file */
504 if ((stop = seeksegment.stop) == -1)
505 stop = seeksegment.duration;
507 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
508 if ((cur_type != GST_SEEK_TYPE_NONE)) {
509 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
510 * we can just copy the last_stop. If not, we use the bps to convert TIME to
512 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
513 (gint64 *) & wav->offset))
514 wav->offset = seeksegment.position;
515 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
516 wav->offset -= (wav->offset % wav->bytes_per_sample);
517 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
518 wav->offset += wav->datastart;
519 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
521 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
525 if (stop_type != GST_SEEK_TYPE_NONE) {
526 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
527 wav->end_offset = stop;
528 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
529 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
530 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
531 wav->end_offset += wav->datastart;
532 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
534 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
538 /* make sure filesize is not exceeded due to rounding errors or so,
539 * same precaution as in _stream_headers */
540 bformat = GST_FORMAT_BYTES;
541 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
542 wav->end_offset = MIN (wav->end_offset, upstream_size);
544 if (wav->datasize > 0 && wav->end_offset > wav->datastart + wav->datasize)
545 wav->end_offset = wav->datastart + wav->datasize;
547 /* this is the range of bytes we will use for playback */
548 wav->offset = MIN (wav->offset, wav->end_offset);
549 wav->dataleft = wav->end_offset - wav->offset;
551 GST_DEBUG_OBJECT (wav,
552 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
553 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
554 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
556 /* prepare for streaming again */
560 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
561 GST_DEBUG_OBJECT (wav, "sending flush stop");
563 fevent = gst_event_new_flush_stop (TRUE);
564 gst_event_set_seqnum (fevent, seqnum);
565 gst_pad_push_event (wav->sinkpad, gst_event_ref (fevent));
566 gst_pad_push_event (wav->srcpad, fevent);
569 /* now we did the seek and can activate the new segment values */
570 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
572 /* if we're doing a segment seek, post a SEGMENT_START message */
573 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
574 gst_element_post_message (GST_ELEMENT_CAST (wav),
575 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
576 wav->segment.format, wav->segment.position));
579 /* now create the newsegment */
580 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
581 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
583 /* store the newsegment event so it can be sent from the streaming thread. */
584 if (wav->start_segment)
585 gst_event_unref (wav->start_segment);
586 wav->start_segment = gst_event_new_segment (&wav->segment);
587 gst_event_set_seqnum (wav->start_segment, seqnum);
589 /* mark discont if we are going to stream from another position. */
590 if (last_stop != wav->segment.position) {
591 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
595 /* and start the streaming task again */
596 if (!wav->streaming) {
597 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
601 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
608 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
613 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
618 GST_DEBUG_OBJECT (wav,
619 "Could not determine byte position for desired time");
625 * gst_wavparse_peek_chunk_info:
626 * @wav Wavparse object
627 * @tag holder for tag
628 * @size holder for tag size
630 * Peek next chunk info (tag and size)
632 * Returns: %TRUE when the chunk info (header) is available
635 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
637 const guint8 *data = NULL;
639 if (gst_adapter_available (wav->adapter) < 8)
642 data = gst_adapter_map (wav->adapter, 8);
643 *tag = GST_READ_UINT32_LE (data);
644 *size = GST_READ_UINT32_LE (data + 4);
645 gst_adapter_unmap (wav->adapter);
647 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
648 GST_FOURCC_ARGS (*tag));
654 * gst_wavparse_peek_chunk:
655 * @wav Wavparse object
656 * @tag holder for tag
657 * @size holder for tag size
659 * Peek enough data for one full chunk
661 * Returns: %TRUE when the full chunk is available
664 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
666 guint32 peek_size = 0;
669 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
672 /* size 0 -> empty data buffer would surprise most callers,
673 * large size -> do not bother trying to squeeze that into adapter,
674 * so we throw poor man's exception, which can be caught if caller really
675 * wants to handle 0 size chunk */
676 if (!(*size) || (*size) >= (1 << 30)) {
677 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
678 *size, GST_FOURCC_ARGS (*tag));
679 /* chain should give up */
680 wav->abort_buffering = TRUE;
683 peek_size = (*size + 1) & ~1;
684 available = gst_adapter_available (wav->adapter);
686 if (available >= (8 + peek_size)) {
689 GST_LOG ("but only %u bytes available now", available);
695 * gst_wavparse_calculate_duration:
696 * @wav: wavparse object
698 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
701 * Returns: %TRUE if duration is available.
704 gst_wavparse_calculate_duration (GstWavParse * wav)
706 if (wav->duration > 0)
710 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
712 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
714 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
715 GST_TIME_ARGS (wav->duration));
717 } else if (wav->fact) {
719 gst_util_uint64_scale_ceil (GST_SECOND, wav->fact, wav->rate);
720 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
721 GST_TIME_ARGS (wav->duration));
728 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
733 if (wav->streaming) {
734 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
737 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
738 GST_FOURCC_ARGS (tag));
739 flush = 8 + ((size + 1) & ~1);
740 wav->offset += flush;
741 if (wav->streaming) {
742 gst_adapter_flush (wav->adapter, flush);
744 gst_buffer_unref (buf);
751 * gst_wavparse_cue_chunk:
752 * @wav GstWavParse object
753 * @data holder for data
754 * @size holder for data size
756 * Parse cue chunk from @data to wav->cues.
758 * Returns: %TRUE when cue chunk is available
761 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
768 GST_WARNING_OBJECT (wav, "found another cue's");
772 ncues = GST_READ_UINT32_LE (data);
774 if (size < 4 + ncues * 24) {
775 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
781 for (i = 0; i < ncues; i++) {
782 cue = g_new0 (GstWavParseCue, 1);
783 cue->id = GST_READ_UINT32_LE (data);
784 cue->position = GST_READ_UINT32_LE (data + 4);
785 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
786 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
787 cue->block_start = GST_READ_UINT32_LE (data + 16);
788 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
789 cues = g_list_append (cues, cue);
799 * gst_wavparse_labl_chunk:
800 * @wav GstWavParse object
801 * @data holder for data
802 * @size holder for data size
804 * Parse labl from @data to wav->labls.
806 * Returns: %TRUE when labl chunk is available
809 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
811 GstWavParseLabl *labl;
816 labl = g_new0 (GstWavParseLabl, 1);
820 labl->cue_point_id = GST_READ_UINT32_LE (data);
821 labl->text = g_memdup (data + 4, size - 4);
823 wav->labls = g_list_append (wav->labls, labl);
829 * gst_wavparse_note_chunk:
830 * @wav GstWavParse object
831 * @data holder for data
832 * @size holder for data size
834 * Parse note from @data to wav->notes.
836 * Returns: %TRUE when note chunk is available
839 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
841 GstWavParseNote *note;
846 note = g_new0 (GstWavParseNote, 1);
850 note->cue_point_id = GST_READ_UINT32_LE (data);
851 note->text = g_memdup (data + 4, size - 4);
853 wav->notes = g_list_append (wav->notes, note);
859 * gst_wavparse_smpl_chunk:
860 * @wav GstWavParse object
861 * @data holder for data
862 * @size holder for data size
864 * Parse smpl chunk from @data.
866 * Returns: %TRUE when cue chunk is available
869 gst_wavparse_smpl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
874 manufacturer_id = GST_READ_UINT32_LE (data);
875 product_id = GST_READ_UINT32_LE (data + 4);
876 sample_period = GST_READ_UINT32_LE (data + 8);
878 note_number = GST_READ_UINT32_LE (data + 12);
880 pitch_fraction = GST_READ_UINT32_LE (data + 16);
881 SMPTE_format = GST_READ_UINT32_LE (data + 20);
882 SMPTE_offset = GST_READ_UINT32_LE (data + 24);
883 num_sample_loops = GST_READ_UINT32_LE (data + 28);
884 List of Sample Loops, 24 bytes each
888 wav->tags = gst_tag_list_new_empty ();
889 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
890 GST_TAG_MIDI_BASE_NOTE, (guint) note_number, NULL);
895 * gst_wavparse_adtl_chunk:
896 * @wav GstWavParse object
897 * @data holder for data
898 * @size holder for data size
900 * Parse adtl from @data.
902 * Returns: %TRUE when adtl chunk is available
905 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
907 guint32 ltag, lsize, offset = 0;
910 ltag = GST_READ_UINT32_LE (data + offset);
911 lsize = GST_READ_UINT32_LE (data + offset + 4);
913 if (lsize + 8 > size) {
914 GST_WARNING_OBJECT (wav, "Invalid adtl size: %u + 8 > %u", lsize, size);
919 case GST_RIFF_TAG_labl:
920 gst_wavparse_labl_chunk (wav, data + offset, size);
922 case GST_RIFF_TAG_note:
923 gst_wavparse_note_chunk (wav, data + offset, size);
926 GST_WARNING_OBJECT (wav, "Unknowm adtl %" GST_FOURCC_FORMAT,
927 GST_FOURCC_ARGS (ltag));
928 GST_MEMDUMP_OBJECT (wav, "Unknowm adtl", &data[offset], lsize);
931 offset += 8 + GST_ROUND_UP_2 (lsize);
932 size -= 8 + GST_ROUND_UP_2 (lsize);
939 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
941 GstTagList *tags = NULL;
942 GstTocEntry *entry = NULL;
944 entry = gst_toc_find_entry (toc, id);
946 tags = gst_toc_entry_get_tags (entry);
948 tags = gst_tag_list_new_empty ();
949 gst_toc_entry_set_tags (entry, tags);
957 * gst_wavparse_create_toc:
958 * @wav GstWavParse object
960 * Create TOC from wav->cues and wav->labls.
963 gst_wavparse_create_toc (GstWavParse * wav)
969 GstWavParseLabl *labl;
970 GstWavParseNote *note;
973 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
975 GST_OBJECT_LOCK (wav);
977 GST_OBJECT_UNLOCK (wav);
978 GST_WARNING_OBJECT (wav, "found another TOC");
983 GST_OBJECT_UNLOCK (wav);
987 /* FIXME: send CURRENT scope toc too */
988 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
990 /* add cue edition */
991 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
992 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
993 gst_toc_append_entry (toc, entry);
995 /* add tracks in cue edition */
999 prev_subentry = cur_subentry;
1000 /* previous track stop time = current track start time */
1001 if (prev_subentry != NULL) {
1002 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1003 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1004 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1006 id = g_strdup_printf ("%08x", cue->id);
1007 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1009 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1010 stop = wav->duration;
1011 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1012 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1013 list = g_list_next (list);
1016 /* add tags in tracks */
1020 id = g_strdup_printf ("%08x", labl->cue_point_id);
1021 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1024 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1027 list = g_list_next (list);
1032 id = g_strdup_printf ("%08x", note->cue_point_id);
1033 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1036 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1039 list = g_list_next (list);
1042 /* send data as TOC */
1045 /* send TOC event */
1047 GST_OBJECT_UNLOCK (wav);
1048 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1054 #define MAX_BUFFER_SIZE 4096
1057 parse_ds64 (GstWavParse * wav, GstBuffer * buf)
1060 guint32 dataSizeLow, dataSizeHigh;
1061 guint32 sampleCountLow, sampleCountHigh;
1063 gst_buffer_map (buf, &map, GST_MAP_READ);
1064 dataSizeLow = GST_READ_UINT32_LE (map.data + 2 * 4);
1065 dataSizeHigh = GST_READ_UINT32_LE (map.data + 3 * 4);
1066 sampleCountLow = GST_READ_UINT32_LE (map.data + 4 * 4);
1067 sampleCountHigh = GST_READ_UINT32_LE (map.data + 5 * 4);
1068 gst_buffer_unmap (buf, &map);
1069 if (dataSizeHigh != 0xFFFFFFFF && dataSizeLow != 0xFFFFFFFF) {
1070 wav->datasize = ((guint64) dataSizeHigh << 32) | dataSizeLow;
1072 if (sampleCountHigh != 0xFFFFFFFF && sampleCountLow != 0xFFFFFFFF) {
1073 wav->fact = ((guint64) sampleCountHigh << 32) | sampleCountLow;
1076 GST_DEBUG_OBJECT (wav, "Got 'ds64' TAG, datasize : %" G_GINT64_FORMAT
1077 " fact: %" G_GINT64_FORMAT, wav->datasize, wav->fact);
1081 static GstFlowReturn
1082 gst_wavparse_stream_headers (GstWavParse * wav)
1084 GstFlowReturn res = GST_FLOW_OK;
1085 GstBuffer *buf = NULL;
1086 gst_riff_strf_auds *header = NULL;
1088 gboolean gotdata = FALSE;
1089 GstCaps *caps = NULL;
1090 gchar *codec_name = NULL;
1091 gint64 upstream_size = 0;
1094 /* search for "_fmt" chunk, which should be first */
1095 while (!wav->got_fmt) {
1098 /* The header starts with a 'fmt ' tag */
1099 if (wav->streaming) {
1100 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1103 gst_adapter_flush (wav->adapter, 8);
1107 buf = gst_adapter_take_buffer (wav->adapter, size);
1109 gst_adapter_flush (wav->adapter, 1);
1110 wav->offset += GST_ROUND_UP_2 (size);
1112 buf = gst_buffer_new ();
1115 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1116 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1120 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1121 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1122 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1123 tag == GST_RIFF_TAG_id3 || tag == GST_RIFF_TAG_IDVX ||
1124 tag == GST_BWF_TAG_iXML || tag == GST_BWF_TAG_qlty ||
1125 tag == GST_BWF_TAG_mext || tag == GST_BWF_TAG_levl ||
1126 tag == GST_BWF_TAG_link || tag == GST_BWF_TAG_axml ||
1127 tag == GST_RIFF_TAG_Fake) {
1128 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1129 GST_FOURCC_ARGS (tag));
1130 gst_buffer_unref (buf);
1135 if (tag == GST_RS64_TAG_DS64) {
1136 if (!parse_ds64 (wav, buf))
1142 if (tag != GST_RIFF_TAG_fmt)
1145 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1147 goto parse_header_error;
1149 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1151 /* do sanity checks of header fields */
1152 if (header->channels == 0)
1154 if (header->rate == 0)
1157 GST_DEBUG_OBJECT (wav, "creating the caps");
1159 /* Note: gst_riff_create_audio_caps might need to fix values in
1160 * the header header depending on the format, so call it first */
1161 /* FIXME: Need to handle the channel reorder map */
1162 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1163 NULL, &codec_name, NULL);
1166 gst_buffer_unref (extra);
1169 goto unknown_format;
1171 /* If we got raw audio from upstream, we remove the codec_data field,
1172 * which may have been added if the wav header included an extended
1173 * chunk. We want to keep it for non raw audio.
1175 s = gst_caps_get_structure (caps, 0);
1176 if (s && gst_structure_has_name (s, "audio/x-raw")) {
1177 gst_structure_remove_field (s, "codec_data");
1180 /* do more sanity checks of header fields
1181 * (these can be sanitized by gst_riff_create_audio_caps()
1183 wav->format = header->format;
1184 wav->rate = header->rate;
1185 wav->channels = header->channels;
1186 wav->blockalign = header->blockalign;
1187 wav->depth = header->bits_per_sample;
1188 wav->av_bps = header->av_bps;
1194 /* do format specific handling */
1195 switch (wav->format) {
1196 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1197 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1199 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1200 * bitrate inside the mpeg stream */
1201 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1205 case GST_RIFF_WAVE_FORMAT_PCM:
1206 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1207 goto invalid_blockalign;
1210 if (wav->av_bps > wav->blockalign * wav->rate)
1212 /* use the configured bps */
1213 wav->bps = wav->av_bps;
1217 wav->width = (wav->blockalign * 8) / wav->channels;
1218 wav->bytes_per_sample = wav->channels * wav->width / 8;
1220 if (wav->bytes_per_sample <= 0)
1221 goto no_bytes_per_sample;
1223 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1224 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1225 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1226 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1227 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1228 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1229 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1231 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1232 * formats). This will make the element output a BYTE format segment and
1233 * will not timestamp the outgoing buffers.
1235 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1237 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1239 /* create pad later so we can sniff the first few bytes
1240 * of the real data and correct our caps if necessary */
1241 gst_caps_replace (&wav->caps, caps);
1242 gst_caps_replace (&caps, NULL);
1244 wav->got_fmt = TRUE;
1247 wav->tags = gst_tag_list_new_empty ();
1249 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1250 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1252 g_free (codec_name);
1258 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1259 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1261 /* loop headers until we get data */
1263 if (wav->streaming) {
1264 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1271 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1272 &buf)) != GST_FLOW_OK)
1273 goto header_read_error;
1274 gst_buffer_map (buf, &map, GST_MAP_READ);
1275 tag = GST_READ_UINT32_LE (map.data);
1276 size = GST_READ_UINT32_LE (map.data + 4);
1277 gst_buffer_unmap (buf, &map);
1280 GST_INFO_OBJECT (wav,
1281 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT ", size %"
1282 G_GUINT32_FORMAT, GST_FOURCC_ARGS (tag), wav->offset, size);
1284 /* Maximum valid size is INT_MAX */
1285 if (size & 0x80000000) {
1286 GST_WARNING_OBJECT (wav, "Invalid size, clipping to 0x7fffffff");
1290 /* Clip to upstream size if known */
1291 if (wav->datasize > 0 && size + wav->offset > wav->datasize) {
1292 GST_WARNING_OBJECT (wav, "Clipping chunk size to file size");
1293 size = wav->datasize - wav->offset;
1296 /* wav is a st00pid format, we don't know for sure where data starts.
1297 * So we have to go bit by bit until we find the 'data' header
1300 case GST_RIFF_TAG_data:{
1303 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1305 if (wav->ignore_length) {
1306 GST_DEBUG_OBJECT (wav, "Ignoring length");
1309 if (wav->streaming) {
1310 gst_adapter_flush (wav->adapter, 8);
1313 gst_buffer_unref (buf);
1316 wav->datastart = wav->offset;
1317 /* use size from ds64 chunk if available */
1318 if (size64 == -1 && wav->datasize > 0) {
1319 GST_DEBUG_OBJECT (wav, "Using ds64 datasize");
1320 size64 = wav->datasize;
1322 /* If size is zero, then the data chunk probably actually extends to
1323 the end of the file */
1324 if (size64 == 0 && upstream_size) {
1325 size64 = upstream_size - wav->datastart;
1327 /* Or the file might be truncated */
1328 else if (upstream_size) {
1329 size64 = MIN (size64, (upstream_size - wav->datastart));
1331 wav->datasize = size64;
1332 wav->dataleft = size64;
1333 wav->end_offset = size64 + wav->datastart;
1334 if (!wav->streaming) {
1335 /* We will continue parsing tags 'till end */
1336 wav->offset += size64;
1338 GST_DEBUG_OBJECT (wav, "datasize = %" G_GUINT64_FORMAT, size64);
1341 case GST_RIFF_TAG_fact:{
1342 if (wav->fact == 0 &&
1343 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1344 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1345 const guint data_size = 4;
1347 GST_INFO_OBJECT (wav, "Have fact chunk");
1348 if (size < data_size) {
1349 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1350 /* need more data */
1353 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1357 /* number of samples (for compressed formats) */
1358 if (wav->streaming) {
1359 const guint8 *data = NULL;
1361 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1364 gst_adapter_flush (wav->adapter, 8);
1365 data = gst_adapter_map (wav->adapter, data_size);
1366 wav->fact = GST_READ_UINT32_LE (data);
1367 gst_adapter_unmap (wav->adapter);
1368 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1370 gst_buffer_unref (buf);
1373 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1374 data_size, &buf)) != GST_FLOW_OK)
1375 goto header_read_error;
1376 gst_buffer_extract (buf, 0, &wav->fact, 4);
1377 wav->fact = GUINT32_FROM_LE (wav->fact);
1378 gst_buffer_unref (buf);
1380 GST_DEBUG_OBJECT (wav, "have fact %" G_GUINT64_FORMAT, wav->fact);
1381 wav->offset += 8 + GST_ROUND_UP_2 (size);
1384 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1385 /* need more data */
1391 case GST_RIFF_TAG_acid:{
1392 const gst_riff_acid *acid = NULL;
1393 const guint data_size = sizeof (gst_riff_acid);
1396 GST_INFO_OBJECT (wav, "Have acid chunk");
1397 if (size < data_size) {
1398 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1399 /* need more data */
1402 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1406 if (wav->streaming) {
1407 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1410 gst_adapter_flush (wav->adapter, 8);
1411 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1413 tempo = acid->tempo;
1414 gst_adapter_unmap (wav->adapter);
1417 gst_buffer_unref (buf);
1420 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1421 size, &buf)) != GST_FLOW_OK)
1422 goto header_read_error;
1423 gst_buffer_map (buf, &map, GST_MAP_READ);
1424 acid = (const gst_riff_acid *) map.data;
1425 tempo = acid->tempo;
1426 gst_buffer_unmap (buf, &map);
1428 /* send data as tags */
1430 wav->tags = gst_tag_list_new_empty ();
1431 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1432 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1434 size = GST_ROUND_UP_2 (size);
1435 if (wav->streaming) {
1436 gst_adapter_flush (wav->adapter, size);
1438 gst_buffer_unref (buf);
1440 wav->offset += 8 + size;
1443 /* FIXME: all list tags after data are ignored in streaming mode */
1444 case GST_RIFF_TAG_LIST:{
1447 if (wav->streaming) {
1448 const guint8 *data = NULL;
1450 if (gst_adapter_available (wav->adapter) < 12) {
1453 data = gst_adapter_map (wav->adapter, 12);
1454 ltag = GST_READ_UINT32_LE (data + 8);
1455 gst_adapter_unmap (wav->adapter);
1457 gst_buffer_unref (buf);
1460 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1461 &buf)) != GST_FLOW_OK)
1462 goto header_read_error;
1463 gst_buffer_extract (buf, 8, <ag, 4);
1464 ltag = GUINT32_FROM_LE (ltag);
1467 case GST_RIFF_LIST_INFO:{
1468 const gint data_size = size - 4;
1471 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1472 if (wav->streaming) {
1473 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1476 gst_adapter_flush (wav->adapter, 12);
1478 if (data_size > 0) {
1479 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1481 gst_adapter_flush (wav->adapter, 1);
1485 gst_buffer_unref (buf);
1487 if (data_size > 0) {
1489 gst_pad_pull_range (wav->sinkpad, wav->offset,
1490 data_size, &buf)) != GST_FLOW_OK)
1491 goto header_read_error;
1494 if (data_size > 0) {
1496 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1498 GstTagList *old = wav->tags;
1500 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1502 gst_tag_list_unref (old);
1503 gst_tag_list_unref (new);
1505 gst_buffer_unref (buf);
1506 wav->offset += GST_ROUND_UP_2 (data_size);
1510 case GST_RIFF_LIST_adtl:{
1511 const gint data_size = size - 4;
1513 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1514 if (wav->streaming) {
1515 const guint8 *data = NULL;
1517 gst_adapter_flush (wav->adapter, 12);
1519 data = gst_adapter_map (wav->adapter, data_size);
1520 gst_wavparse_adtl_chunk (wav, data, data_size);
1521 gst_adapter_unmap (wav->adapter);
1525 gst_buffer_unref (buf);
1529 gst_pad_pull_range (wav->sinkpad, wav->offset,
1530 data_size, &buf)) != GST_FLOW_OK)
1531 goto header_read_error;
1532 gst_buffer_map (buf, &map, GST_MAP_READ);
1533 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1535 gst_buffer_unmap (buf, &map);
1537 wav->offset += GST_ROUND_UP_2 (data_size);
1541 GST_WARNING_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1542 GST_FOURCC_ARGS (ltag));
1543 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1544 /* need more data */
1550 case GST_RIFF_TAG_cue:{
1551 const guint data_size = size;
1553 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1554 if (wav->streaming) {
1555 const guint8 *data = NULL;
1557 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1560 gst_adapter_flush (wav->adapter, 8);
1562 data = gst_adapter_map (wav->adapter, data_size);
1563 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1564 goto header_read_error;
1566 gst_adapter_unmap (wav->adapter);
1571 gst_buffer_unref (buf);
1574 gst_pad_pull_range (wav->sinkpad, wav->offset,
1575 data_size, &buf)) != GST_FLOW_OK)
1576 goto header_read_error;
1577 gst_buffer_map (buf, &map, GST_MAP_READ);
1578 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1580 goto header_read_error;
1582 gst_buffer_unmap (buf, &map);
1584 size = GST_ROUND_UP_2 (size);
1585 if (wav->streaming) {
1586 gst_adapter_flush (wav->adapter, size);
1588 gst_buffer_unref (buf);
1590 size = GST_ROUND_UP_2 (size);
1591 wav->offset += size;
1594 case GST_RIFF_TAG_smpl:{
1595 const gint data_size = size;
1597 GST_DEBUG_OBJECT (wav, "Have 'smpl' TAG, size : %u", data_size);
1598 if (wav->streaming) {
1599 const guint8 *data = NULL;
1601 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1604 gst_adapter_flush (wav->adapter, 8);
1606 data = gst_adapter_map (wav->adapter, data_size);
1607 if (!gst_wavparse_smpl_chunk (wav, data, data_size)) {
1608 goto header_read_error;
1610 gst_adapter_unmap (wav->adapter);
1615 gst_buffer_unref (buf);
1618 gst_pad_pull_range (wav->sinkpad, wav->offset,
1619 data_size, &buf)) != GST_FLOW_OK)
1620 goto header_read_error;
1621 gst_buffer_map (buf, &map, GST_MAP_READ);
1622 if (!gst_wavparse_smpl_chunk (wav, (const guint8 *) map.data,
1624 goto header_read_error;
1626 gst_buffer_unmap (buf, &map);
1628 size = GST_ROUND_UP_2 (size);
1629 if (wav->streaming) {
1630 gst_adapter_flush (wav->adapter, size);
1632 gst_buffer_unref (buf);
1634 size = GST_ROUND_UP_2 (size);
1635 wav->offset += size;
1639 GST_WARNING_OBJECT (wav, "Ignoring chunk %" GST_FOURCC_FORMAT,
1640 GST_FOURCC_ARGS (tag));
1641 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1642 /* need more data */
1647 if (upstream_size && (wav->offset >= upstream_size)) {
1648 /* Now we are gone through the whole file */
1653 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1655 if (wav->bps <= 0 && wav->fact) {
1657 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1659 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1660 (guint64) wav->fact);
1661 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1666 if (gst_wavparse_calculate_duration (wav)) {
1667 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1668 if (!wav->ignore_length)
1669 wav->segment.duration = wav->duration;
1671 gst_wavparse_create_toc (wav);
1673 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1674 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1675 if (!wav->ignore_length)
1676 wav->segment.duration = wav->datasize;
1679 /* now we have all the info to perform a pending seek if any, if no
1680 * event, this will still do the right thing and it will also send
1681 * the right newsegment event downstream. */
1682 gst_wavparse_perform_seek (wav, wav->seek_event);
1683 /* remove pending event */
1684 gst_event_replace (&wav->seek_event, NULL);
1686 /* we just started, we are discont */
1687 wav->discont = TRUE;
1689 wav->state = GST_WAVPARSE_DATA;
1691 /* determine reasonable max buffer size,
1692 * that is, buffers not too small either size or time wise
1693 * so we do not end up with too many of them */
1695 if (gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size))
1696 wav->max_buf_size = upstream_size;
1698 wav->max_buf_size = 0;
1699 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1700 if (wav->blockalign > 0)
1701 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1703 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1710 g_free (codec_name);
1713 gst_caps_unref (caps);
1718 res = GST_FLOW_ERROR;
1723 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1724 ("Invalid WAV header (no fmt at start): %"
1725 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1730 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1731 ("Couldn't parse audio header"));
1736 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1737 ("Stream claims to contain no channels - invalid data"));
1742 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1743 ("Stream with sample_rate == 0 - invalid data"));
1748 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1749 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1750 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1755 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1756 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1757 wav->av_bps, wav->blockalign * wav->rate));
1760 no_bytes_per_sample:
1762 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1763 ("Could not caluclate bytes per sample - invalid data"));
1768 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1769 ("No caps found for format 0x%x, %u channels, %u Hz",
1770 wav->format, wav->channels, wav->rate));
1775 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1776 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1782 * Read WAV file tag when streaming
1784 static GstFlowReturn
1785 gst_wavparse_parse_stream_init (GstWavParse * wav)
1787 if (gst_adapter_available (wav->adapter) >= 12) {
1790 /* _take flushes the data */
1791 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1793 GST_DEBUG ("Parsing wav header");
1794 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1795 return GST_FLOW_ERROR;
1798 /* Go to next state */
1799 wav->state = GST_WAVPARSE_HEADER;
1804 /* handle an event sent directly to the element.
1806 * This event can be sent either in the READY state or the
1807 * >READY state. The only event of interest really is the seek
1810 * In the READY state we can only store the event and try to
1811 * respect it when going to PAUSED. We assume we are in the
1812 * READY state when our parsing state != GST_WAVPARSE_DATA.
1814 * When we are steaming, we can simply perform the seek right
1818 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1820 GstWavParse *wav = GST_WAVPARSE (element);
1821 gboolean res = FALSE;
1823 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1825 switch (GST_EVENT_TYPE (event)) {
1826 case GST_EVENT_SEEK:
1827 if (wav->state == GST_WAVPARSE_DATA) {
1828 /* we can handle the seek directly when streaming data */
1829 res = gst_wavparse_perform_seek (wav, event);
1831 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1833 gst_event_replace (&wav->seek_event, event);
1835 /* we always return true */
1842 gst_event_unref (event);
1847 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1851 s = gst_caps_get_structure (caps, 0);
1852 if (!gst_structure_has_name (s, "audio/x-dts"))
1854 /* typefind behavior for DTS:
1855 * MAXIMUM: multiple frame syncs detected, certainly DTS
1856 * LIKELY: single frame sync at offset 0. Maybe DTS?
1857 * POSSIBLE: single frame sync, not at offset 0. Highly unlikely
1859 if (prob > GST_TYPE_FIND_LIKELY)
1861 if (prob <= GST_TYPE_FIND_POSSIBLE)
1863 /* for maybe, check for at least a valid-looking rate and channels */
1864 if (!gst_structure_has_field (s, "channels"))
1866 /* and for extra assurance we could also check the rate from the DTS frame
1867 * against the one in the wav header, but for now let's not do that */
1868 return gst_structure_has_field (s, "rate");
1872 gst_wavparse_get_upstream_tags (GstWavParse * wav, GstTagScope scope)
1874 GstTagList *tags = NULL;
1879 while ((ev = gst_pad_get_sticky_event (wav->sinkpad, GST_EVENT_TAG, i++))) {
1880 gst_event_parse_tag (ev, &tags);
1881 if (tags != NULL && gst_tag_list_get_scope (tags) == scope) {
1882 tags = gst_tag_list_copy (tags);
1883 gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
1884 gst_event_unref (ev);
1888 gst_event_unref (ev);
1894 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1897 GstTagList *tags, *utags;
1899 GST_DEBUG_OBJECT (wav, "adding src pad");
1901 g_assert (wav->caps != NULL);
1903 s = gst_caps_get_structure (wav->caps, 0);
1904 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1905 GstTypeFindProbability prob;
1908 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1909 if (tf_caps != NULL) {
1910 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1911 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1912 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1913 gst_caps_unref (wav->caps);
1914 wav->caps = tf_caps;
1916 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1917 GST_TAG_AUDIO_CODEC, "dts", NULL);
1919 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1920 "marked as raw PCM audio, but ignoring for now", tf_caps);
1921 gst_caps_unref (tf_caps);
1926 gst_pad_set_caps (wav->srcpad, wav->caps);
1927 gst_caps_replace (&wav->caps, NULL);
1929 if (wav->start_segment) {
1930 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1931 gst_pad_push_event (wav->srcpad, wav->start_segment);
1932 wav->start_segment = NULL;
1935 /* upstream tags, e.g. from id3/ape tag before the wav file; assume for now
1936 * that there'll be only one scope/type of tag list from upstream, if any */
1937 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_GLOBAL);
1939 utags = gst_wavparse_get_upstream_tags (wav, GST_TAG_SCOPE_STREAM);
1941 /* if there's a tag upstream it's probably been added to override the
1942 * tags from inside the wav header, so keep upstream tags if in doubt */
1943 tags = gst_tag_list_merge (utags, wav->tags, GST_TAG_MERGE_KEEP);
1945 if (wav->tags != NULL) {
1946 gst_tag_list_unref (wav->tags);
1951 gst_tag_list_unref (utags);
1953 /* send tags downstream, if any */
1955 gst_pad_push_event (wav->srcpad, gst_event_new_tag (tags));
1958 static GstFlowReturn
1959 gst_wavparse_stream_data (GstWavParse * wav)
1961 GstBuffer *buf = NULL;
1962 GstFlowReturn res = GST_FLOW_OK;
1963 guint64 desired, obtained;
1964 GstClockTime timestamp, next_timestamp, duration;
1965 guint64 pos, nextpos;
1968 GST_LOG_OBJECT (wav,
1969 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1970 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1972 /* Get the next n bytes and output them */
1973 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1976 /* scale the amount of data by the segment rate so we get equal
1977 * amounts of data regardless of the playback rate */
1979 MIN (gst_guint64_to_gdouble (wav->dataleft),
1980 wav->max_buf_size * ABS (wav->segment.rate));
1982 if (desired >= wav->blockalign && wav->blockalign > 0)
1983 desired -= (desired % wav->blockalign);
1985 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1986 "from the sinkpad", desired);
1988 if (wav->streaming) {
1989 guint avail = gst_adapter_available (wav->adapter);
1992 /* flush some bytes if evil upstream sends segment that starts
1993 * before data or does is not send sample aligned segment */
1994 if (G_LIKELY (wav->offset >= wav->datastart)) {
1995 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1997 extra = wav->datastart - wav->offset;
2000 if (G_UNLIKELY (extra)) {
2001 extra = wav->bytes_per_sample - extra;
2002 if (extra <= avail) {
2003 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2004 gst_adapter_flush (wav->adapter, extra);
2005 wav->offset += extra;
2006 wav->dataleft -= extra;
2007 goto iterate_adapter;
2009 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2010 gst_adapter_clear (wav->adapter);
2011 wav->offset += avail;
2012 wav->dataleft -= avail;
2017 if (avail < desired) {
2018 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2022 buf = gst_adapter_take_buffer (wav->adapter, desired);
2024 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2025 desired, &buf)) != GST_FLOW_OK)
2028 /* we may get a short buffer at the end of the file */
2029 if (gst_buffer_get_size (buf) < desired) {
2030 gsize size = gst_buffer_get_size (buf);
2032 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2033 if (size >= wav->blockalign) {
2034 if (wav->blockalign > 0) {
2035 buf = gst_buffer_make_writable (buf);
2036 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2039 gst_buffer_unref (buf);
2045 obtained = gst_buffer_get_size (buf);
2047 /* our positions in bytes */
2048 pos = wav->offset - wav->datastart;
2049 nextpos = pos + obtained;
2051 /* update offsets, does not overflow. */
2052 buf = gst_buffer_make_writable (buf);
2053 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2054 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2056 /* first chunk of data? create the source pad. We do this only here so
2057 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2058 if (G_UNLIKELY (wav->first)) {
2060 /* this will also push the segment events */
2061 gst_wavparse_add_src_pad (wav, buf);
2063 /* If we have a pending start segment, send it now. */
2064 if (G_UNLIKELY (wav->start_segment != NULL)) {
2065 gst_pad_push_event (wav->srcpad, wav->start_segment);
2066 wav->start_segment = NULL;
2071 /* and timestamps if we have a bitrate, be careful for overflows */
2073 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2075 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2076 duration = next_timestamp - timestamp;
2078 /* update current running segment position */
2079 if (G_LIKELY (next_timestamp >= wav->segment.start))
2080 wav->segment.position = next_timestamp;
2081 } else if (wav->fact) {
2083 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2084 /* and timestamps if we have a bitrate, be careful for overflows */
2085 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2086 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2087 duration = next_timestamp - timestamp;
2089 /* no bitrate, all we know is that the first sample has timestamp 0, all
2090 * other positions and durations have unknown timestamp. */
2094 timestamp = GST_CLOCK_TIME_NONE;
2095 duration = GST_CLOCK_TIME_NONE;
2096 /* update current running segment position with byte offset */
2097 if (G_LIKELY (nextpos >= wav->segment.start))
2098 wav->segment.position = nextpos;
2100 if ((pos > 0) && wav->vbr) {
2101 /* don't set timestamps for VBR files if it's not the first buffer */
2102 timestamp = GST_CLOCK_TIME_NONE;
2103 duration = GST_CLOCK_TIME_NONE;
2106 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2107 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2108 wav->discont = FALSE;
2111 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2112 GST_BUFFER_DURATION (buf) = duration;
2114 GST_LOG_OBJECT (wav,
2115 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2116 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2117 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2119 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2122 if (obtained < wav->dataleft) {
2123 wav->offset += obtained;
2124 wav->dataleft -= obtained;
2126 wav->offset += wav->dataleft;
2130 /* Iterate until need more data, so adapter size won't grow */
2131 if (wav->streaming) {
2132 GST_LOG_OBJECT (wav,
2133 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2135 goto iterate_adapter;
2142 GST_DEBUG_OBJECT (wav, "found EOS");
2143 return GST_FLOW_EOS;
2147 /* check if we got EOS */
2148 if (res == GST_FLOW_EOS)
2151 GST_WARNING_OBJECT (wav,
2152 "Error getting %" G_GINT64_FORMAT " bytes from the "
2153 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2158 GST_INFO_OBJECT (wav,
2159 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2160 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2161 gst_pad_is_linked (wav->srcpad));
2167 gst_wavparse_loop (GstPad * pad)
2170 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2174 GST_LOG_OBJECT (wav, "process data");
2176 switch (wav->state) {
2177 case GST_WAVPARSE_START:
2178 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2179 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2183 gst_pad_create_stream_id (wav->srcpad, GST_ELEMENT_CAST (wav), NULL);
2184 event = gst_event_new_stream_start (stream_id);
2185 gst_event_set_group_id (event, gst_util_group_id_next ());
2186 gst_pad_push_event (wav->srcpad, event);
2189 wav->state = GST_WAVPARSE_HEADER;
2192 case GST_WAVPARSE_HEADER:
2193 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2194 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2197 wav->state = GST_WAVPARSE_DATA;
2198 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2201 case GST_WAVPARSE_DATA:
2202 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2206 g_assert_not_reached ();
2213 const gchar *reason = gst_flow_get_name (ret);
2215 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2216 gst_pad_pause_task (pad);
2218 if (ret == GST_FLOW_EOS) {
2219 /* handle end-of-stream/segment */
2220 /* so align our position with the end of it, if there is one
2221 * this ensures a subsequent will arrive at correct base/acc time */
2222 if (wav->segment.format == GST_FORMAT_TIME) {
2223 if (wav->segment.rate > 0.0 &&
2224 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2225 wav->segment.position = wav->segment.stop;
2226 else if (wav->segment.rate < 0.0)
2227 wav->segment.position = wav->segment.start;
2229 if (wav->state == GST_WAVPARSE_START) {
2230 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2231 ("No valid input found before end of stream"));
2232 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2234 /* add pad before we perform EOS */
2235 if (G_UNLIKELY (wav->first)) {
2237 gst_wavparse_add_src_pad (wav, NULL);
2240 /* perform EOS logic */
2241 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2244 if ((stop = wav->segment.stop) == -1)
2245 stop = wav->segment.duration;
2247 gst_element_post_message (GST_ELEMENT_CAST (wav),
2248 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2249 wav->segment.format, stop));
2250 gst_pad_push_event (wav->srcpad,
2251 gst_event_new_segment_done (wav->segment.format, stop));
2253 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2256 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2257 /* for fatal errors we post an error message, post the error
2258 * first so the app knows about the error first. */
2259 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2260 (_("Internal data flow error.")),
2261 ("streaming task paused, reason %s (%d)", reason, ret));
2262 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2268 static GstFlowReturn
2269 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2272 GstWavParse *wav = GST_WAVPARSE (parent);
2274 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2275 gst_buffer_get_size (buf));
2277 gst_adapter_push (wav->adapter, buf);
2279 switch (wav->state) {
2280 case GST_WAVPARSE_START:
2281 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2282 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2285 if (wav->state != GST_WAVPARSE_HEADER)
2288 /* otherwise fall-through */
2289 case GST_WAVPARSE_HEADER:
2290 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2291 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2294 if (!wav->got_fmt || wav->datastart == 0)
2297 wav->state = GST_WAVPARSE_DATA;
2298 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2301 case GST_WAVPARSE_DATA:
2302 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2303 wav->discont = TRUE;
2304 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2308 g_return_val_if_reached (GST_FLOW_ERROR);
2311 if (G_UNLIKELY (wav->abort_buffering)) {
2312 wav->abort_buffering = FALSE;
2313 ret = GST_FLOW_ERROR;
2314 /* sort of demux/parse error */
2315 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2321 static GstFlowReturn
2322 gst_wavparse_flush_data (GstWavParse * wav)
2324 GstFlowReturn ret = GST_FLOW_OK;
2327 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2328 ret = gst_wavparse_stream_data (wav);
2335 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2337 GstWavParse *wav = GST_WAVPARSE (parent);
2338 gboolean ret = TRUE;
2340 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2342 switch (GST_EVENT_TYPE (event)) {
2343 case GST_EVENT_CAPS:
2345 /* discard, we'll come up with proper src caps */
2346 gst_event_unref (event);
2349 case GST_EVENT_SEGMENT:
2351 gint64 start, stop, offset = 0, end_offset = -1;
2354 /* some debug output */
2355 gst_event_copy_segment (event, &segment);
2356 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2359 if (wav->state != GST_WAVPARSE_DATA) {
2360 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2364 /* now we are either committed to TIME or BYTE format,
2365 * and we only expect a BYTE segment, e.g. following a seek */
2366 if (segment.format == GST_FORMAT_BYTES) {
2367 /* handle (un)signed issues */
2368 start = segment.start;
2369 stop = segment.stop;
2372 start -= wav->datastart;
2373 start = MAX (start, 0);
2377 stop -= wav->datastart;
2378 stop = MAX (stop, 0);
2380 if (wav->segment.format == GST_FORMAT_TIME) {
2381 guint64 bps = wav->bps;
2383 /* operating in format TIME, so we can convert */
2384 if (!bps && wav->fact)
2386 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2390 gst_util_uint64_scale_ceil (start, GST_SECOND,
2391 (guint64) wav->bps);
2394 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2395 (guint64) wav->bps);
2399 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2403 segment.start = start;
2404 segment.stop = stop;
2406 /* accept upstream's notion of segment and distribute along */
2407 segment.format = wav->segment.format;
2408 segment.time = segment.position = segment.start;
2409 segment.duration = wav->segment.duration;
2410 segment.base = gst_segment_to_running_time (&wav->segment,
2411 GST_FORMAT_TIME, wav->segment.position);
2413 gst_segment_copy_into (&segment, &wav->segment);
2415 /* also store the newsegment event for the streaming thread */
2416 if (wav->start_segment)
2417 gst_event_unref (wav->start_segment);
2418 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2419 wav->start_segment = gst_event_new_segment (&segment);
2421 /* stream leftover data in current segment */
2422 gst_wavparse_flush_data (wav);
2423 /* and set up streaming thread for next one */
2424 wav->offset = offset;
2425 wav->end_offset = end_offset;
2427 if (wav->datasize > 0 && (wav->end_offset == -1
2428 || wav->end_offset > wav->datastart + wav->datasize))
2429 wav->end_offset = wav->datastart + wav->datasize;
2431 if (wav->end_offset != -1) {
2432 wav->dataleft = wav->end_offset - wav->offset;
2434 /* infinity; upstream will EOS when done */
2435 wav->dataleft = G_MAXUINT64;
2438 gst_event_unref (event);
2442 if (wav->state == GST_WAVPARSE_START) {
2443 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2444 ("No valid input found before end of stream"));
2446 /* add pad if needed so EOS is seen downstream */
2447 if (G_UNLIKELY (wav->first)) {
2449 gst_wavparse_add_src_pad (wav, NULL);
2451 /* stream leftover data in current segment */
2452 gst_wavparse_flush_data (wav);
2457 case GST_EVENT_FLUSH_STOP:
2461 gst_adapter_clear (wav->adapter);
2462 wav->discont = TRUE;
2463 dur = wav->segment.duration;
2464 gst_segment_init (&wav->segment, wav->segment.format);
2465 wav->segment.duration = dur;
2469 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2477 /* convert and query stuff */
2478 static const GstFormat *
2479 gst_wavparse_get_formats (GstPad * pad)
2481 static const GstFormat formats[] = {
2484 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2493 gst_wavparse_pad_convert (GstPad * pad,
2494 GstFormat src_format, gint64 src_value,
2495 GstFormat * dest_format, gint64 * dest_value)
2497 GstWavParse *wavparse;
2498 gboolean res = TRUE;
2500 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2502 if (*dest_format == src_format) {
2503 *dest_value = src_value;
2507 if ((wavparse->bps == 0) && !wavparse->fact)
2510 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2511 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2513 switch (src_format) {
2514 case GST_FORMAT_BYTES:
2515 switch (*dest_format) {
2516 case GST_FORMAT_DEFAULT:
2517 *dest_value = src_value / wavparse->bytes_per_sample;
2518 /* make sure we end up on a sample boundary */
2519 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2521 case GST_FORMAT_TIME:
2522 /* src_value + datastart = offset */
2523 GST_INFO_OBJECT (wavparse,
2524 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2526 if (wavparse->bps > 0)
2527 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2528 (guint64) wavparse->bps);
2529 else if (wavparse->fact) {
2530 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2531 wavparse->rate, wavparse->fact);
2534 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2545 case GST_FORMAT_DEFAULT:
2546 switch (*dest_format) {
2547 case GST_FORMAT_BYTES:
2548 *dest_value = src_value * wavparse->bytes_per_sample;
2550 case GST_FORMAT_TIME:
2551 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2552 (guint64) wavparse->rate);
2560 case GST_FORMAT_TIME:
2561 switch (*dest_format) {
2562 case GST_FORMAT_BYTES:
2563 if (wavparse->bps > 0)
2564 *dest_value = gst_util_uint64_scale (src_value,
2565 (guint64) wavparse->bps, GST_SECOND);
2567 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2568 wavparse->rate, wavparse->fact);
2570 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2572 /* make sure we end up on a sample boundary */
2573 *dest_value -= *dest_value % wavparse->blockalign;
2575 case GST_FORMAT_DEFAULT:
2576 *dest_value = gst_util_uint64_scale (src_value,
2577 (guint64) wavparse->rate, GST_SECOND);
2596 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2602 /* handle queries for location and length in requested format */
2604 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2606 gboolean res = TRUE;
2607 GstWavParse *wav = GST_WAVPARSE (parent);
2609 /* only if we know */
2610 if (wav->state != GST_WAVPARSE_DATA) {
2614 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2616 switch (GST_QUERY_TYPE (query)) {
2617 case GST_QUERY_POSITION:
2623 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2624 curb = wav->offset - wav->datastart;
2625 gst_query_parse_position (query, &format, NULL);
2626 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2629 case GST_FORMAT_BYTES:
2630 format = GST_FORMAT_BYTES;
2634 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2639 gst_query_set_position (query, format, cur);
2642 case GST_QUERY_DURATION:
2644 gint64 duration = 0;
2647 if (wav->ignore_length) {
2652 gst_query_parse_duration (query, &format, NULL);
2655 case GST_FORMAT_BYTES:{
2656 format = GST_FORMAT_BYTES;
2657 duration = wav->datasize;
2660 case GST_FORMAT_TIME:
2661 if ((res = gst_wavparse_calculate_duration (wav))) {
2662 duration = wav->duration;
2670 gst_query_set_duration (query, format, duration);
2673 case GST_QUERY_CONVERT:
2675 gint64 srcvalue, dstvalue;
2676 GstFormat srcformat, dstformat;
2678 gst_query_parse_convert (query, &srcformat, &srcvalue,
2679 &dstformat, &dstvalue);
2680 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2681 &dstformat, &dstvalue);
2683 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2686 case GST_QUERY_SEEKING:{
2688 gboolean seekable = FALSE;
2690 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2691 if (fmt == wav->segment.format) {
2692 if (wav->streaming) {
2695 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2696 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2697 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2698 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2700 gst_query_unref (q);
2702 GST_LOG_OBJECT (wav, "looping => seekable");
2706 } else if (fmt == GST_FORMAT_TIME) {
2710 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2715 res = gst_pad_query_default (pad, parent, query);
2722 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2724 GstWavParse *wavparse = GST_WAVPARSE (parent);
2725 gboolean res = FALSE;
2727 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2729 switch (GST_EVENT_TYPE (event)) {
2730 case GST_EVENT_SEEK:
2731 /* can only handle events when we are in the data state */
2732 if (wavparse->state == GST_WAVPARSE_DATA) {
2733 res = gst_wavparse_perform_seek (wavparse, event);
2735 gst_event_unref (event);
2738 case GST_EVENT_TOC_SELECT:
2741 GstTocEntry *entry = NULL;
2742 GstEvent *seek_event;
2745 if (!wavparse->toc) {
2746 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2749 gst_event_parse_toc_select (event, &uid);
2751 GST_OBJECT_LOCK (wavparse);
2752 entry = gst_toc_find_entry (wavparse->toc, uid);
2753 if (entry == NULL) {
2754 GST_OBJECT_UNLOCK (wavparse);
2755 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2759 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2760 GST_OBJECT_UNLOCK (wavparse);
2761 seek_event = gst_event_new_seek (1.0,
2763 GST_SEEK_FLAG_FLUSH,
2764 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2765 res = gst_wavparse_perform_seek (wavparse, seek_event);
2766 gst_event_unref (seek_event);
2770 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2774 gst_event_unref (event);
2779 res = gst_pad_push_event (wavparse->sinkpad, event);
2786 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2788 GstWavParse *wav = GST_WAVPARSE (parent);
2793 gst_adapter_clear (wav->adapter);
2794 g_object_unref (wav->adapter);
2795 wav->adapter = NULL;
2798 query = gst_query_new_scheduling ();
2800 if (!gst_pad_peer_query (sinkpad, query)) {
2801 gst_query_unref (query);
2805 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2806 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2807 gst_query_unref (query);
2812 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2813 wav->streaming = FALSE;
2814 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2818 GST_DEBUG_OBJECT (sinkpad, "activating push");
2819 wav->streaming = TRUE;
2820 wav->adapter = gst_adapter_new ();
2821 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2827 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2828 GstPadMode mode, gboolean active)
2833 case GST_PAD_MODE_PUSH:
2836 case GST_PAD_MODE_PULL:
2838 /* if we have a scheduler we can start the task */
2839 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2842 res = gst_pad_stop_task (sinkpad);
2852 static GstStateChangeReturn
2853 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2855 GstStateChangeReturn ret;
2856 GstWavParse *wav = GST_WAVPARSE (element);
2858 switch (transition) {
2859 case GST_STATE_CHANGE_NULL_TO_READY:
2861 case GST_STATE_CHANGE_READY_TO_PAUSED:
2862 gst_wavparse_reset (wav);
2864 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2870 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2872 switch (transition) {
2873 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2875 case GST_STATE_CHANGE_PAUSED_TO_READY:
2876 gst_wavparse_reset (wav);
2878 case GST_STATE_CHANGE_READY_TO_NULL:
2887 gst_wavparse_set_property (GObject * object, guint prop_id,
2888 const GValue * value, GParamSpec * pspec)
2892 g_return_if_fail (GST_IS_WAVPARSE (object));
2893 self = GST_WAVPARSE (object);
2896 case PROP_IGNORE_LENGTH:
2897 self->ignore_length = g_value_get_boolean (value);
2900 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2906 gst_wavparse_get_property (GObject * object, guint prop_id,
2907 GValue * value, GParamSpec * pspec)
2911 g_return_if_fail (GST_IS_WAVPARSE (object));
2912 self = GST_WAVPARSE (object);
2915 case PROP_IGNORE_LENGTH:
2916 g_value_set_boolean (value, self->ignore_length);
2919 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
2924 plugin_init (GstPlugin * plugin)
2928 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2932 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2935 "Parse a .wav file into raw audio",
2936 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)