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., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, 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 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 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
55 #include "gstwavparse.h"
56 #include "gst/riff/riff-ids.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 static void gst_wavparse_dispose (GObject * object);
66 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
68 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
69 GstObject * parent, GstPadMode mode, gboolean active);
70 static gboolean gst_wavparse_send_event (GstElement * element,
72 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
73 GstStateChange transition);
75 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
77 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
78 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
80 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
82 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
84 static void gst_wavparse_loop (GstPad * pad);
85 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
88 static GstStaticPadTemplate sink_template_factory =
89 GST_STATIC_PAD_TEMPLATE ("sink",
92 GST_STATIC_CAPS ("audio/x-wav")
96 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
98 #define gst_wavparse_parent_class parent_class
99 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
103 gst_wavparse_class_init (GstWavParseClass * klass)
105 GstElementClass *gstelement_class;
106 GObjectClass *object_class;
107 GstPadTemplate *src_template;
109 gstelement_class = (GstElementClass *) klass;
110 object_class = (GObjectClass *) klass;
112 parent_class = g_type_class_peek_parent (klass);
114 object_class->dispose = gst_wavparse_dispose;
116 gstelement_class->change_state = gst_wavparse_change_state;
117 gstelement_class->send_event = gst_wavparse_send_event;
120 gst_element_class_add_pad_template (gstelement_class,
121 gst_static_pad_template_get (&sink_template_factory));
123 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
124 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
125 gst_element_class_add_pad_template (gstelement_class, src_template);
127 gst_element_class_set_details_simple (gstelement_class, "WAV audio demuxer",
128 "Codec/Demuxer/Audio",
129 "Parse a .wav file into raw audio",
130 "Erik Walthinsen <omega@cse.ogi.edu>");
134 gst_wavparse_reset (GstWavParse * wav)
136 wav->state = GST_WAVPARSE_START;
138 /* These will all be set correctly in the fmt chunk */
152 wav->got_fmt = FALSE;
156 gst_event_unref (wav->seek_event);
157 wav->seek_event = NULL;
159 gst_adapter_clear (wav->adapter);
160 g_object_unref (wav->adapter);
164 gst_tag_list_free (wav->tags);
167 gst_caps_unref (wav->caps);
169 if (wav->start_segment)
170 gst_event_unref (wav->start_segment);
171 wav->start_segment = NULL;
175 gst_wavparse_dispose (GObject * object)
177 GstWavParse *wav = GST_WAVPARSE (object);
179 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
180 gst_wavparse_reset (wav);
182 G_OBJECT_CLASS (parent_class)->dispose (object);
186 gst_wavparse_init (GstWavParse * wavparse)
188 gst_wavparse_reset (wavparse);
192 gst_pad_new_from_static_template (&sink_template_factory, "sink");
193 gst_pad_set_activate_function (wavparse->sinkpad,
194 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
195 gst_pad_set_activatemode_function (wavparse->sinkpad,
196 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
197 gst_pad_set_chain_function (wavparse->sinkpad,
198 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
199 gst_pad_set_event_function (wavparse->sinkpad,
200 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
201 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
205 gst_pad_new_from_template (gst_element_class_get_pad_template
206 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
207 gst_pad_use_fixed_caps (wavparse->srcpad);
208 gst_pad_set_query_function (wavparse->srcpad,
209 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
210 gst_pad_set_event_function (wavparse->srcpad,
211 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
212 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
215 /* Compute (value * nom) % denom, avoiding overflow. This can be used
216 * to perform ceiling or rounding division together with
217 * gst_util_uint64_scale[_int]. */
218 #define uint64_scale_modulo(val, nom, denom) \
219 ((val % denom) * (nom % denom) % denom)
221 /* Like gst_util_uint64_scale, but performs ceiling division. */
223 uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
225 guint64 result = gst_util_uint64_scale_int (val, num, denom);
227 if (uint64_scale_modulo (val, num, denom) == 0)
233 /* Like gst_util_uint64_scale, but performs ceiling division. */
235 uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom)
237 guint64 result = gst_util_uint64_scale (val, num, denom);
239 if (uint64_scale_modulo (val, num, denom) == 0)
246 /* FIXME: why is that not in use? */
249 gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
252 GstByteStream *bs = wavparse->bs;
253 gst_riff_chunk *temp_chunk, chunk;
255 struct _gst_riff_labl labl, *temp_labl;
256 struct _gst_riff_ltxt ltxt, *temp_ltxt;
257 struct _gst_riff_note note, *temp_note;
260 GstPropsEntry *entry;
264 props = wavparse->metadata->properties;
268 gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
269 if (got_bytes != sizeof (gst_riff_chunk)) {
272 temp_chunk = (gst_riff_chunk *) tempdata;
274 chunk.id = GUINT32_FROM_LE (temp_chunk->id);
275 chunk.size = GUINT32_FROM_LE (temp_chunk->size);
277 if (chunk.size == 0) {
278 gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
279 len -= sizeof (gst_riff_chunk);
284 case GST_RIFF_adtl_labl:
286 gst_bytestream_peek_bytes (bs, &tempdata,
287 sizeof (struct _gst_riff_labl));
288 if (got_bytes != sizeof (struct _gst_riff_labl)) {
292 temp_labl = (struct _gst_riff_labl *) tempdata;
293 labl.id = GUINT32_FROM_LE (temp_labl->id);
294 labl.size = GUINT32_FROM_LE (temp_labl->size);
295 labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
297 gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
298 len -= sizeof (struct _gst_riff_labl);
300 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
301 if (got_bytes != labl.size - 4) {
305 label_name = (char *) tempdata;
307 gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
308 len -= (((labl.size - 4) + 1) & ~1);
310 new_caps = gst_caps_new ("label",
311 "application/x-gst-metadata",
312 gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
313 "name", G_TYPE_STRING (label_name), NULL));
315 if (gst_props_get (props, "labels", &caps, NULL)) {
316 caps = g_list_append (caps, new_caps);
318 caps = g_list_append (NULL, new_caps);
320 entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
321 gst_props_add_entry (props, entry);
326 case GST_RIFF_adtl_ltxt:
328 gst_bytestream_peek_bytes (bs, &tempdata,
329 sizeof (struct _gst_riff_ltxt));
330 if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
334 temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
335 ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
336 ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
337 ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
338 ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
339 ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
340 ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
341 ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
342 ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
343 ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
345 gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
346 len -= sizeof (struct _gst_riff_ltxt);
348 if (ltxt.size - 20 > 0) {
349 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
350 if (got_bytes != ltxt.size - 20) {
354 gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
355 len -= (((ltxt.size - 20) + 1) & ~1);
357 label_name = (char *) tempdata;
362 new_caps = gst_caps_new ("ltxt",
363 "application/x-gst-metadata",
364 gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
365 "name", G_TYPE_STRING (label_name),
366 "length", G_TYPE_INT (ltxt.length), NULL));
368 if (gst_props_get (props, "ltxts", &caps, NULL)) {
369 caps = g_list_append (caps, new_caps);
371 caps = g_list_append (NULL, new_caps);
373 entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
374 gst_props_add_entry (props, entry);
379 case GST_RIFF_adtl_note:
381 gst_bytestream_peek_bytes (bs, &tempdata,
382 sizeof (struct _gst_riff_note));
383 if (got_bytes != sizeof (struct _gst_riff_note)) {
387 temp_note = (struct _gst_riff_note *) tempdata;
388 note.id = GUINT32_FROM_LE (temp_note->id);
389 note.size = GUINT32_FROM_LE (temp_note->size);
390 note.identifier = GUINT32_FROM_LE (temp_note->identifier);
392 gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
393 len -= sizeof (struct _gst_riff_note);
395 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
396 if (got_bytes != note.size - 4) {
400 gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
401 len -= (((note.size - 4) + 1) & ~1);
403 label_name = (char *) tempdata;
405 new_caps = gst_caps_new ("note",
406 "application/x-gst-metadata",
407 gst_props_new ("identifier", G_TYPE_INT (note.identifier),
408 "name", G_TYPE_STRING (label_name), NULL));
410 if (gst_props_get (props, "notes", &caps, NULL)) {
411 caps = g_list_append (caps, new_caps);
413 caps = g_list_append (NULL, new_caps);
415 entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
416 gst_props_add_entry (props, entry);
422 g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
423 GST_FOURCC_ARGS (chunk.id));
428 g_object_notify (G_OBJECT (wavparse), "metadata");
432 gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
435 GstByteStream *bs = wavparse->bs;
436 struct _gst_riff_cue *temp_cue, cue;
437 struct _gst_riff_cuepoints *points;
441 GstPropsEntry *entry;
447 gst_bytestream_peek_bytes (bs, &tempdata,
448 sizeof (struct _gst_riff_cue));
449 temp_cue = (struct _gst_riff_cue *) tempdata;
451 /* fixup for our big endian friends */
452 cue.id = GUINT32_FROM_LE (temp_cue->id);
453 cue.size = GUINT32_FROM_LE (temp_cue->size);
454 cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
456 gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
457 if (got_bytes != sizeof (struct _gst_riff_cue)) {
461 len -= sizeof (struct _gst_riff_cue);
463 /* -4 because cue.size contains the cuepoints size
464 and we've already flushed that out of the system */
465 required = cue.size - 4;
466 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
467 gst_bytestream_flush (bs, ((required) + 1) & ~1);
468 if (got_bytes != required) {
472 len -= (((cue.size - 4) + 1) & ~1);
474 /* now we have an array of struct _gst_riff_cuepoints in tempdata */
475 points = (struct _gst_riff_cuepoints *) tempdata;
477 for (i = 0; i < cue.cuepoints; i++) {
480 caps = gst_caps_new ("cues",
481 "application/x-gst-metadata",
482 gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
483 "position", G_TYPE_INT (points[i].offset), NULL));
484 cues = g_list_append (cues, caps);
487 entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
488 gst_props_add_entry (wavparse->metadata->properties, entry);
491 g_object_notify (G_OBJECT (wavparse), "metadata");
494 /* Read 'fmt ' header */
496 gst_wavparse_fmt (GstWavParse * wav)
498 gst_riff_strf_auds *header = NULL;
501 if (!gst_riff_read_strf_auds (wav, &header))
504 wav->format = header->format;
505 wav->rate = header->rate;
506 wav->channels = header->channels;
507 if (wav->channels == 0)
510 wav->blockalign = header->blockalign;
511 wav->width = (header->blockalign * 8) / header->channels;
512 wav->depth = header->size;
513 wav->bps = header->av_bps;
517 /* Note: gst_riff_create_audio_caps might need to fix values in
518 * the header header depending on the format, so call it first */
519 caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
525 gst_wavparse_create_sourcepad (wav);
526 gst_pad_use_fixed_caps (wav->srcpad);
527 gst_pad_set_active (wav->srcpad, TRUE);
528 gst_pad_set_caps (wav->srcpad, caps);
529 gst_caps_free (caps);
530 gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
531 gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
533 GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
540 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
541 ("No FMT tag found"));
546 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
547 ("Stream claims to contain zero channels - invalid data"));
553 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
554 ("Stream claims to bitrate of <= zero - invalid data"));
560 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
566 gst_wavparse_other (GstWavParse * wav)
570 if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
571 GST_WARNING_OBJECT (wav, "could not peek head");
574 GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
575 (gchar *) & tag, length);
578 case GST_RIFF_TAG_LIST:
579 if (!(tag = gst_riff_peek_list (wav))) {
580 GST_WARNING_OBJECT (wav, "could not peek list");
585 case GST_RIFF_LIST_INFO:
586 if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
587 GST_WARNING_OBJECT (wav, "could not read list");
592 case GST_RIFF_LIST_adtl:
593 if (!gst_riff_read_skip (wav)) {
594 GST_WARNING_OBJECT (wav, "could not read skip");
600 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
602 if (!gst_riff_read_skip (wav)) {
603 GST_WARNING_OBJECT (wav, "could not read skip");
611 case GST_RIFF_TAG_data:
612 if (!gst_bytestream_flush (wav->bs, 8)) {
613 GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
617 GST_DEBUG_OBJECT (wav, "switching to data mode");
618 wav->state = GST_WAVPARSE_DATA;
619 wav->datastart = gst_bytestream_tell (wav->bs);
623 /* length is 0, data probably stretches to the end
625 GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
626 /* get length of file */
627 file_length = gst_bytestream_length (wav->bs);
628 if (file_length == -1) {
629 GST_DEBUG_OBJECT (wav,
630 "could not get file length, assuming data to eof");
631 /* could not get length, assuming till eof */
632 length = G_MAXUINT32;
634 if (file_length > G_MAXUINT32) {
635 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
636 ", clipping to 32 bits", file_length);
637 /* could not get length, assuming till eof */
638 length = G_MAXUINT32;
640 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
641 ", datalength %u", file_length, length);
642 /* substract offset of datastart from length */
643 length = file_length - wav->datastart;
644 GST_DEBUG_OBJECT (wav, "datalength %u", length);
647 wav->datasize = (guint64) length;
648 GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
651 case GST_RIFF_TAG_cue:
652 if (!gst_riff_read_skip (wav)) {
653 GST_WARNING_OBJECT (wav, "could not read skip");
659 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
660 if (!gst_riff_read_skip (wav))
671 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
675 if (!gst_riff_parse_file_header (element, buf, &doctype))
678 if (doctype != GST_RIFF_RIFF_WAVE)
686 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
687 ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
688 GST_FOURCC_ARGS (doctype)));
694 gst_wavparse_stream_init (GstWavParse * wav)
697 GstBuffer *buf = NULL;
699 if ((res = gst_pad_pull_range (wav->sinkpad,
700 wav->offset, 12, &buf)) != GST_FLOW_OK)
702 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
703 return GST_FLOW_ERROR;
711 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
713 /* -1 always maps to -1 */
719 /* 0 always maps to 0 */
726 *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND);
728 } else if (wav->fact) {
730 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
731 *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND);
738 /* This function is used to perform seeks on the element.
740 * It also works when event is NULL, in which case it will just
741 * start from the last configured segment. This technique is
742 * used when activating the element and to perform the seek in
746 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
750 GstFormat format, bformat;
752 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
753 gint64 cur, stop, upstream_size;
756 GstSegment seeksegment = { 0, };
760 GST_DEBUG_OBJECT (wav, "doing seek with event");
762 gst_event_parse_seek (event, &rate, &format, &flags,
763 &cur_type, &cur, &stop_type, &stop);
765 /* no negative rates yet */
769 if (format != wav->segment.format) {
770 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
771 gst_format_get_name (format),
772 gst_format_get_name (wav->segment.format));
774 if (cur_type != GST_SEEK_TYPE_NONE)
776 gst_pad_query_convert (wav->srcpad, format, cur,
777 wav->segment.format, &cur);
778 if (res && stop_type != GST_SEEK_TYPE_NONE)
780 gst_pad_query_convert (wav->srcpad, format, stop,
781 wav->segment.format, &stop);
785 format = wav->segment.format;
788 GST_DEBUG_OBJECT (wav, "doing seek without event");
791 cur_type = GST_SEEK_TYPE_SET;
792 stop_type = GST_SEEK_TYPE_SET;
795 /* in push mode, we must delegate to upstream */
796 if (wav->streaming) {
797 gboolean res = FALSE;
799 /* if streaming not yet started; only prepare initial newsegment */
800 if (!event || wav->state != GST_WAVPARSE_DATA) {
801 if (wav->start_segment)
802 gst_event_unref (wav->start_segment);
804 /* wav->start_segment =
805 gst_event_new_new_segment (FALSE, wav->segment.rate,
806 wav->segment.format, wav->segment.last_stop, wav->segment.duration,
807 wav->segment.last_stop);*/
810 /* convert seek positions to byte positions in data sections */
811 if (format == GST_FORMAT_TIME) {
812 /* should not fail */
813 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
815 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
818 /* mind sample boundary and header */
820 cur -= (cur % wav->bytes_per_sample);
821 cur += wav->datastart;
824 stop -= (stop % wav->bytes_per_sample);
825 stop += wav->datastart;
827 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
828 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
830 /* BYTE seek event */
831 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
833 res = gst_pad_push_event (wav->sinkpad, event);
839 flush = flags & GST_SEEK_FLAG_FLUSH;
841 /* now we need to make sure the streaming thread is stopped. We do this by
842 * either sending a FLUSH_START event downstream which will cause the
843 * streaming thread to stop with a WRONG_STATE.
844 * For a non-flushing seek we simply pause the task, which will happen as soon
845 * as it completes one iteration (and thus might block when the sink is
846 * blocking in preroll). */
848 GST_DEBUG_OBJECT (wav, "sending flush start");
849 gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
851 gst_pad_pause_task (wav->sinkpad);
854 /* we should now be able to grab the streaming thread because we stopped it
855 * with the above flush/pause code */
856 GST_PAD_STREAM_LOCK (wav->sinkpad);
858 /* save current position */
859 last_stop = wav->segment.position;
861 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
863 /* copy segment, we need this because we still need the old
864 * segment when we close the current segment. */
865 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
867 /* configure the seek parameters in the seeksegment. We will then have the
868 * right values in the segment to perform the seek */
870 GST_DEBUG_OBJECT (wav, "configuring seek");
871 gst_segment_do_seek (&seeksegment, rate, format, flags,
872 cur_type, cur, stop_type, stop, &update);
875 /* figure out the last position we need to play. If it's configured (stop !=
876 * -1), use that, else we play until the total duration of the file */
877 if ((stop = seeksegment.stop) == -1)
878 stop = seeksegment.duration;
880 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
881 if ((cur_type != GST_SEEK_TYPE_NONE)) {
882 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
883 * we can just copy the last_stop. If not, we use the bps to convert TIME to
885 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
886 (gint64 *) & wav->offset))
887 wav->offset = seeksegment.position;
888 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
889 wav->offset -= (wav->offset % wav->bytes_per_sample);
890 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
891 wav->offset += wav->datastart;
892 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
894 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
898 if (stop_type != GST_SEEK_TYPE_NONE) {
899 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
900 wav->end_offset = stop;
901 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
902 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
903 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
904 wav->end_offset += wav->datastart;
905 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
907 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
911 /* make sure filesize is not exceeded due to rounding errors or so,
912 * same precaution as in _stream_headers */
913 bformat = GST_FORMAT_BYTES;
914 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
915 wav->end_offset = MIN (wav->end_offset, upstream_size);
917 /* this is the range of bytes we will use for playback */
918 wav->offset = MIN (wav->offset, wav->end_offset);
919 wav->dataleft = wav->end_offset - wav->offset;
921 GST_DEBUG_OBJECT (wav,
922 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
923 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
924 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
926 /* prepare for streaming again */
928 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
929 GST_DEBUG_OBJECT (wav, "sending flush stop");
930 gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
933 /* now we did the seek and can activate the new segment values */
934 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
936 /* if we're doing a segment seek, post a SEGMENT_START message */
937 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
938 gst_element_post_message (GST_ELEMENT_CAST (wav),
939 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
940 wav->segment.format, wav->segment.position));
943 /* now create the newsegment */
944 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
945 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
947 /* store the newsegment event so it can be sent from the streaming thread. */
948 if (wav->start_segment)
949 gst_event_unref (wav->start_segment);
950 wav->start_segment = gst_event_new_segment (&wav->segment);
952 /* mark discont if we are going to stream from another position. */
953 if (last_stop != wav->segment.position) {
954 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
958 /* and start the streaming task again */
959 if (!wav->streaming) {
960 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
964 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
971 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
976 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
981 GST_DEBUG_OBJECT (wav,
982 "Could not determine byte position for desired time");
988 * gst_wavparse_peek_chunk_info:
989 * @wav Wavparse object
990 * @tag holder for tag
991 * @size holder for tag size
993 * Peek next chunk info (tag and size)
995 * Returns: %TRUE when the chunk info (header) is available
998 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
1000 const guint8 *data = NULL;
1002 if (gst_adapter_available (wav->adapter) < 8)
1005 data = gst_adapter_map (wav->adapter, 8);
1006 *tag = GST_READ_UINT32_LE (data);
1007 *size = GST_READ_UINT32_LE (data + 4);
1008 gst_adapter_unmap (wav->adapter);
1010 GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
1011 GST_FOURCC_ARGS (*tag));
1017 * gst_wavparse_peek_chunk:
1018 * @wav Wavparse object
1019 * @tag holder for tag
1020 * @size holder for tag size
1022 * Peek enough data for one full chunk
1024 * Returns: %TRUE when the full chunk is available
1027 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
1029 guint32 peek_size = 0;
1032 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
1035 /* size 0 -> empty data buffer would surprise most callers,
1036 * large size -> do not bother trying to squeeze that into adapter,
1037 * so we throw poor man's exception, which can be caught if caller really
1038 * wants to handle 0 size chunk */
1039 if (!(*size) || (*size) >= (1 << 30)) {
1040 GST_INFO ("Invalid/unexpected chunk size %d for tag %" GST_FOURCC_FORMAT,
1041 *size, GST_FOURCC_ARGS (*tag));
1042 /* chain should give up */
1043 wav->abort_buffering = TRUE;
1046 peek_size = (*size + 1) & ~1;
1047 available = gst_adapter_available (wav->adapter);
1049 if (available >= (8 + peek_size)) {
1052 GST_LOG ("but only %u bytes available now", available);
1058 * gst_wavparse_calculate_duration:
1059 * @wav: wavparse object
1061 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
1064 * Returns: %TRUE if duration is available.
1067 gst_wavparse_calculate_duration (GstWavParse * wav)
1069 if (wav->duration > 0)
1073 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
1075 uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps);
1076 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
1077 GST_TIME_ARGS (wav->duration));
1079 } else if (wav->fact) {
1080 wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate);
1081 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
1082 GST_TIME_ARGS (wav->duration));
1089 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
1094 if (wav->streaming) {
1095 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1098 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
1099 GST_FOURCC_ARGS (tag));
1100 flush = 8 + ((size + 1) & ~1);
1101 wav->offset += flush;
1102 if (wav->streaming) {
1103 gst_adapter_flush (wav->adapter, flush);
1105 gst_buffer_unref (buf);
1111 #define MAX_BUFFER_SIZE 4096
1113 static GstFlowReturn
1114 gst_wavparse_stream_headers (GstWavParse * wav)
1116 GstFlowReturn res = GST_FLOW_OK;
1117 GstBuffer *buf = NULL;
1118 gst_riff_strf_auds *header = NULL;
1120 gboolean gotdata = FALSE;
1121 GstCaps *caps = NULL;
1122 gchar *codec_name = NULL;
1124 gint64 upstream_size = 0;
1126 /* search for "_fmt" chunk, which should be first */
1127 while (!wav->got_fmt) {
1130 /* The header starts with a 'fmt ' tag */
1131 if (wav->streaming) {
1132 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1135 gst_adapter_flush (wav->adapter, 8);
1139 buf = gst_adapter_take_buffer (wav->adapter, size);
1141 gst_adapter_flush (wav->adapter, 1);
1142 wav->offset += GST_ROUND_UP_2 (size);
1144 buf = gst_buffer_new ();
1147 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1148 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1152 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1153 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1154 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1155 tag == GST_RIFF_TAG_IDVX) {
1156 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1157 GST_FOURCC_ARGS (tag));
1158 gst_buffer_unref (buf);
1163 if (tag != GST_RIFF_TAG_fmt)
1166 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1168 goto parse_header_error;
1170 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1172 /* do sanity checks of header fields */
1173 if (header->channels == 0)
1175 if (header->rate == 0)
1178 GST_DEBUG_OBJECT (wav, "creating the caps");
1180 /* Note: gst_riff_create_audio_caps might need to fix values in
1181 * the header header depending on the format, so call it first */
1182 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1186 gst_buffer_unref (extra);
1189 goto unknown_format;
1191 /* do more sanity checks of header fields
1192 * (these can be sanitized by gst_riff_create_audio_caps()
1194 wav->format = header->format;
1195 wav->rate = header->rate;
1196 wav->channels = header->channels;
1197 wav->blockalign = header->blockalign;
1198 wav->depth = header->size;
1199 wav->av_bps = header->av_bps;
1205 /* do format specific handling */
1206 switch (wav->format) {
1207 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1208 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1210 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1211 * bitrate inside the mpeg stream */
1212 GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
1216 case GST_RIFF_WAVE_FORMAT_PCM:
1217 if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0))
1218 goto invalid_blockalign;
1221 if (wav->av_bps > wav->blockalign * wav->rate)
1223 /* use the configured bps */
1224 wav->bps = wav->av_bps;
1228 wav->width = (wav->blockalign * 8) / wav->channels;
1229 wav->bytes_per_sample = wav->channels * wav->width / 8;
1231 if (wav->bytes_per_sample <= 0)
1232 goto no_bytes_per_sample;
1234 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1235 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1236 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1237 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1238 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1239 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1240 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1242 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1243 * formats). This will make the element output a BYTE format segment and
1244 * will not timestamp the outgoing buffers.
1246 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1248 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1250 /* create pad later so we can sniff the first few bytes
1251 * of the real data and correct our caps if necessary */
1252 gst_caps_replace (&wav->caps, caps);
1253 gst_caps_replace (&caps, NULL);
1255 wav->got_fmt = TRUE;
1258 wav->tags = gst_tag_list_new_empty ();
1260 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1261 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1263 g_free (codec_name);
1269 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1270 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1272 /* loop headers until we get data */
1274 if (wav->streaming) {
1275 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1281 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1282 &buf)) != GST_FLOW_OK)
1283 goto header_read_error;
1284 data = gst_buffer_map (buf, NULL, NULL, -1);
1285 tag = GST_READ_UINT32_LE (data);
1286 size = GST_READ_UINT32_LE (data + 4);
1287 gst_buffer_unmap (buf, data, -1);
1290 GST_INFO_OBJECT (wav,
1291 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1292 GST_FOURCC_ARGS (tag), wav->offset);
1294 /* wav is a st00pid format, we don't know for sure where data starts.
1295 * So we have to go bit by bit until we find the 'data' header
1298 case GST_RIFF_TAG_data:{
1299 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
1300 if (wav->streaming) {
1301 gst_adapter_flush (wav->adapter, 8);
1304 gst_buffer_unref (buf);
1307 wav->datastart = wav->offset;
1308 /* If size is zero, then the data chunk probably actually extends to
1309 the end of the file */
1310 if (size == 0 && upstream_size) {
1311 size = upstream_size - wav->datastart;
1313 /* Or the file might be truncated */
1314 else if (upstream_size) {
1315 size = MIN (size, (upstream_size - wav->datastart));
1317 wav->datasize = (guint64) size;
1318 wav->dataleft = (guint64) size;
1319 wav->end_offset = size + wav->datastart;
1320 if (!wav->streaming) {
1321 /* We will continue parsing tags 'till end */
1322 wav->offset += size;
1324 GST_DEBUG_OBJECT (wav, "datasize = %d", size);
1327 case GST_RIFF_TAG_fact:{
1328 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1329 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1330 const guint data_size = 4;
1332 GST_INFO_OBJECT (wav, "Have fact chunk");
1333 if (size < data_size) {
1334 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1335 /* need more data */
1338 GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
1342 /* number of samples (for compressed formats) */
1343 if (wav->streaming) {
1344 const guint8 *data = NULL;
1346 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1349 gst_adapter_flush (wav->adapter, 8);
1350 data = gst_adapter_map (wav->adapter, data_size);
1351 wav->fact = GST_READ_UINT32_LE (data);
1352 gst_adapter_unmap (wav->adapter);
1353 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1355 gst_buffer_unref (buf);
1357 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1358 data_size, &buf)) != GST_FLOW_OK)
1359 goto header_read_error;
1360 gst_buffer_extract (buf, 0, &wav->fact, 4);
1361 wav->fact = GUINT32_FROM_LE (wav->fact);
1362 gst_buffer_unref (buf);
1364 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1365 wav->offset += 8 + GST_ROUND_UP_2 (size);
1368 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1369 /* need more data */
1375 case GST_RIFF_TAG_acid:{
1376 const gst_riff_acid *acid = NULL;
1377 const guint data_size = sizeof (gst_riff_acid);
1380 GST_INFO_OBJECT (wav, "Have acid chunk");
1381 if (size < data_size) {
1382 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1383 /* need more data */
1386 GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
1390 if (wav->streaming) {
1391 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1394 gst_adapter_flush (wav->adapter, 8);
1395 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1397 tempo = acid->tempo;
1398 gst_adapter_unmap (wav->adapter);
1400 gst_buffer_unref (buf);
1402 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1403 size, &buf)) != GST_FLOW_OK)
1404 goto header_read_error;
1405 acid = (const gst_riff_acid *) gst_buffer_map (buf, NULL, NULL,
1407 tempo = acid->tempo;
1408 gst_buffer_unmap (buf, (guint8 *) acid, -1);
1410 /* send data as tags */
1412 wav->tags = gst_tag_list_new_empty ();
1413 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1414 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1416 size = GST_ROUND_UP_2 (size);
1417 if (wav->streaming) {
1418 gst_adapter_flush (wav->adapter, size);
1420 gst_buffer_unref (buf);
1422 wav->offset += 8 + size;
1425 /* FIXME: all list tags after data are ignored in streaming mode */
1426 case GST_RIFF_TAG_LIST:{
1429 if (wav->streaming) {
1430 const guint8 *data = NULL;
1432 if (gst_adapter_available (wav->adapter) < 12) {
1435 data = gst_adapter_map (wav->adapter, 12);
1436 ltag = GST_READ_UINT32_LE (data + 8);
1437 gst_adapter_unmap (wav->adapter);
1439 gst_buffer_unref (buf);
1441 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1442 &buf)) != GST_FLOW_OK)
1443 goto header_read_error;
1444 gst_buffer_extract (buf, 8, <ag, 4);
1445 ltag = GUINT32_FROM_LE (ltag);
1448 case GST_RIFF_LIST_INFO:{
1449 const gint data_size = size - 4;
1452 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1453 if (wav->streaming) {
1454 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1457 gst_adapter_flush (wav->adapter, 12);
1459 if (data_size > 0) {
1460 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1462 gst_adapter_flush (wav->adapter, 1);
1466 gst_buffer_unref (buf);
1467 if (data_size > 0) {
1469 gst_pad_pull_range (wav->sinkpad, wav->offset,
1470 data_size, &buf)) != GST_FLOW_OK)
1471 goto header_read_error;
1474 if (data_size > 0) {
1476 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1478 GstTagList *old = wav->tags;
1480 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1482 gst_tag_list_free (old);
1483 gst_tag_list_free (new);
1485 gst_buffer_unref (buf);
1486 wav->offset += GST_ROUND_UP_2 (data_size);
1491 GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1492 GST_FOURCC_ARGS (ltag));
1493 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1494 /* need more data */
1501 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1502 /* need more data */
1507 if (upstream_size && (wav->offset >= upstream_size)) {
1508 /* Now we are gone through the whole file */
1513 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1515 if (wav->bps <= 0 && wav->fact) {
1517 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1519 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1520 (guint64) wav->fact);
1521 GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps);
1526 if (gst_wavparse_calculate_duration (wav)) {
1527 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1528 wav->segment.duration = wav->duration;
1530 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1531 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1532 wav->segment.duration = wav->datasize;
1535 /* now we have all the info to perform a pending seek if any, if no
1536 * event, this will still do the right thing and it will also send
1537 * the right newsegment event downstream. */
1538 gst_wavparse_perform_seek (wav, wav->seek_event);
1539 /* remove pending event */
1540 event_p = &wav->seek_event;
1541 gst_event_replace (event_p, NULL);
1543 /* we just started, we are discont */
1544 wav->discont = TRUE;
1546 wav->state = GST_WAVPARSE_DATA;
1548 /* determine reasonable max buffer size,
1549 * that is, buffers not too small either size or time wise
1550 * so we do not end up with too many of them */
1553 gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
1554 wav->max_buf_size = upstream_size;
1555 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1556 if (wav->blockalign > 0)
1557 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1559 GST_DEBUG_OBJECT (wav, "max buffer size %d", wav->max_buf_size);
1567 g_free (codec_name);
1571 gst_caps_unref (caps);
1576 res = GST_FLOW_ERROR;
1581 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1582 ("Invalid WAV header (no fmt at start): %"
1583 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1588 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1589 ("Couldn't parse audio header"));
1594 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1595 ("Stream claims to contain no channels - invalid data"));
1600 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1601 ("Stream with sample_rate == 0 - invalid data"));
1606 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1607 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1608 wav->blockalign, wav->channels * (guint) ceil (wav->depth / 8.0)));
1613 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1614 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1615 wav->av_bps, wav->blockalign * wav->rate));
1618 no_bytes_per_sample:
1620 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1621 ("Could not caluclate bytes per sample - invalid data"));
1626 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1627 ("No caps found for format 0x%x, %d channels, %d Hz",
1628 wav->format, wav->channels, wav->rate));
1633 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1634 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1640 * Read WAV file tag when streaming
1642 static GstFlowReturn
1643 gst_wavparse_parse_stream_init (GstWavParse * wav)
1645 if (gst_adapter_available (wav->adapter) >= 12) {
1648 /* _take flushes the data */
1649 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1651 GST_DEBUG ("Parsing wav header");
1652 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1653 return GST_FLOW_ERROR;
1656 /* Go to next state */
1657 wav->state = GST_WAVPARSE_HEADER;
1662 /* handle an event sent directly to the element.
1664 * This event can be sent either in the READY state or the
1665 * >READY state. The only event of interest really is the seek
1668 * In the READY state we can only store the event and try to
1669 * respect it when going to PAUSED. We assume we are in the
1670 * READY state when our parsing state != GST_WAVPARSE_DATA.
1672 * When we are steaming, we can simply perform the seek right
1676 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1678 GstWavParse *wav = GST_WAVPARSE (element);
1679 gboolean res = FALSE;
1682 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1684 switch (GST_EVENT_TYPE (event)) {
1685 case GST_EVENT_SEEK:
1686 if (wav->state == GST_WAVPARSE_DATA) {
1687 /* we can handle the seek directly when streaming data */
1688 res = gst_wavparse_perform_seek (wav, event);
1690 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1692 event_p = &wav->seek_event;
1693 gst_event_replace (event_p, event);
1695 /* we always return true */
1702 gst_event_unref (event);
1707 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1711 s = gst_caps_get_structure (caps, 0);
1712 if (!gst_structure_has_name (s, "audio/x-dts"))
1714 if (prob >= GST_TYPE_FIND_LIKELY)
1716 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
1717 if (prob < GST_TYPE_FIND_POSSIBLE)
1719 /* .. in which case we want at least a valid-looking rate and channels */
1720 if (!gst_structure_has_field (s, "channels"))
1722 /* and for extra assurance we could also check the rate from the DTS frame
1723 * against the one in the wav header, but for now let's not do that */
1724 return gst_structure_has_field (s, "rate");
1728 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1732 GST_DEBUG_OBJECT (wav, "adding src pad");
1735 s = gst_caps_get_structure (wav->caps, 0);
1736 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
1737 GstTypeFindProbability prob;
1740 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1741 if (tf_caps != NULL) {
1742 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1743 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1744 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1745 gst_caps_unref (wav->caps);
1746 wav->caps = tf_caps;
1748 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1749 GST_TAG_AUDIO_CODEC, "dts", NULL);
1751 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1752 "marked as raw PCM audio, but ignoring for now", tf_caps);
1753 gst_caps_unref (tf_caps);
1759 gst_pad_set_caps (wav->srcpad, wav->caps);
1760 gst_caps_replace (&wav->caps, NULL);
1762 if (wav->start_segment) {
1763 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1764 gst_pad_push_event (wav->srcpad, wav->start_segment);
1765 wav->start_segment = NULL;
1769 gst_pad_push_event (wav->srcpad, gst_event_new_tag (wav->tags));
1774 static GstFlowReturn
1775 gst_wavparse_stream_data (GstWavParse * wav)
1777 GstBuffer *buf = NULL;
1778 GstFlowReturn res = GST_FLOW_OK;
1779 guint64 desired, obtained;
1780 GstClockTime timestamp, next_timestamp, duration;
1781 guint64 pos, nextpos;
1784 GST_LOG_OBJECT (wav,
1785 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1786 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1788 /* Get the next n bytes and output them */
1789 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1792 /* scale the amount of data by the segment rate so we get equal
1793 * amounts of data regardless of the playback rate */
1795 MIN (gst_guint64_to_gdouble (wav->dataleft),
1796 wav->max_buf_size * ABS (wav->segment.rate));
1798 if (desired >= wav->blockalign && wav->blockalign > 0)
1799 desired -= (desired % wav->blockalign);
1801 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1802 "from the sinkpad", desired);
1804 if (wav->streaming) {
1805 guint avail = gst_adapter_available (wav->adapter);
1808 /* flush some bytes if evil upstream sends segment that starts
1809 * before data or does is not send sample aligned segment */
1810 if (G_LIKELY (wav->offset >= wav->datastart)) {
1811 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1813 extra = wav->datastart - wav->offset;
1816 if (G_UNLIKELY (extra)) {
1817 extra = wav->bytes_per_sample - extra;
1818 if (extra <= avail) {
1819 GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra);
1820 gst_adapter_flush (wav->adapter, extra);
1821 wav->offset += extra;
1822 wav->dataleft -= extra;
1823 goto iterate_adapter;
1825 GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail);
1826 gst_adapter_clear (wav->adapter);
1827 wav->offset += avail;
1828 wav->dataleft -= avail;
1833 if (avail < desired) {
1834 GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
1838 buf = gst_adapter_take_buffer (wav->adapter, desired);
1840 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
1841 desired, &buf)) != GST_FLOW_OK)
1844 /* we may get a short buffer at the end of the file */
1845 if (gst_buffer_get_size (buf) < desired) {
1846 gsize size = gst_buffer_get_size (buf);
1848 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
1849 if (size >= wav->blockalign) {
1850 buf = gst_buffer_make_writable (buf);
1851 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
1853 gst_buffer_unref (buf);
1859 obtained = gst_buffer_get_size (buf);
1861 /* our positions in bytes */
1862 pos = wav->offset - wav->datastart;
1863 nextpos = pos + obtained;
1865 /* update offsets, does not overflow. */
1866 buf = gst_buffer_make_writable (buf);
1867 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
1868 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
1870 /* first chunk of data? create the source pad. We do this only here so
1871 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
1872 if (G_UNLIKELY (wav->first)) {
1874 /* this will also push the segment events */
1875 gst_wavparse_add_src_pad (wav, buf);
1877 /* If we have a pending start segment, send it now. */
1878 if (G_UNLIKELY (wav->start_segment != NULL)) {
1879 gst_pad_push_event (wav->srcpad, wav->start_segment);
1880 wav->start_segment = NULL;
1885 /* and timestamps if we have a bitrate, be careful for overflows */
1886 timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps);
1888 uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps);
1889 duration = next_timestamp - timestamp;
1891 /* update current running segment position */
1892 if (G_LIKELY (next_timestamp >= wav->segment.start))
1893 wav->segment.position = next_timestamp;
1894 } else if (wav->fact) {
1896 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
1897 /* and timestamps if we have a bitrate, be careful for overflows */
1898 timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps);
1899 next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps);
1900 duration = next_timestamp - timestamp;
1902 /* no bitrate, all we know is that the first sample has timestamp 0, all
1903 * other positions and durations have unknown timestamp. */
1907 timestamp = GST_CLOCK_TIME_NONE;
1908 duration = GST_CLOCK_TIME_NONE;
1909 /* update current running segment position with byte offset */
1910 if (G_LIKELY (nextpos >= wav->segment.start))
1911 wav->segment.position = nextpos;
1913 if ((pos > 0) && wav->vbr) {
1914 /* don't set timestamps for VBR files if it's not the first buffer */
1915 timestamp = GST_CLOCK_TIME_NONE;
1916 duration = GST_CLOCK_TIME_NONE;
1919 GST_DEBUG_OBJECT (wav, "marking DISCONT");
1920 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
1921 wav->discont = FALSE;
1924 GST_BUFFER_TIMESTAMP (buf) = timestamp;
1925 GST_BUFFER_DURATION (buf) = duration;
1927 GST_LOG_OBJECT (wav,
1928 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
1929 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
1930 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
1932 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
1935 if (obtained < wav->dataleft) {
1936 wav->offset += obtained;
1937 wav->dataleft -= obtained;
1939 wav->offset += wav->dataleft;
1943 /* Iterate until need more data, so adapter size won't grow */
1944 if (wav->streaming) {
1945 GST_LOG_OBJECT (wav,
1946 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
1948 goto iterate_adapter;
1955 GST_DEBUG_OBJECT (wav, "found EOS");
1956 return GST_FLOW_UNEXPECTED;
1960 /* check if we got EOS */
1961 if (res == GST_FLOW_UNEXPECTED)
1964 GST_WARNING_OBJECT (wav,
1965 "Error getting %" G_GINT64_FORMAT " bytes from the "
1966 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
1971 GST_INFO_OBJECT (wav,
1972 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
1973 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
1974 gst_pad_is_linked (wav->srcpad));
1980 gst_wavparse_loop (GstPad * pad)
1983 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
1985 GST_LOG_OBJECT (wav, "process data");
1987 switch (wav->state) {
1988 case GST_WAVPARSE_START:
1989 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
1990 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
1993 wav->state = GST_WAVPARSE_HEADER;
1996 case GST_WAVPARSE_HEADER:
1997 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
1998 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2001 wav->state = GST_WAVPARSE_DATA;
2002 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2005 case GST_WAVPARSE_DATA:
2006 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2010 g_assert_not_reached ();
2017 const gchar *reason = gst_flow_get_name (ret);
2019 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2020 gst_pad_pause_task (pad);
2022 if (ret == GST_FLOW_UNEXPECTED) {
2023 /* handle end-of-stream/segment */
2024 /* so align our position with the end of it, if there is one
2025 * this ensures a subsequent will arrive at correct base/acc time */
2026 if (wav->segment.format == GST_FORMAT_TIME) {
2027 if (wav->segment.rate > 0.0 &&
2028 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2029 wav->segment.position = wav->segment.stop;
2030 else if (wav->segment.rate < 0.0)
2031 wav->segment.position = wav->segment.start;
2033 /* add pad before we perform EOS */
2034 if (G_UNLIKELY (wav->first)) {
2036 gst_wavparse_add_src_pad (wav, NULL);
2039 if (wav->state == GST_WAVPARSE_START)
2040 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2041 ("No valid input found before end of stream"), (NULL));
2043 /* perform EOS logic */
2044 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2047 if ((stop = wav->segment.stop) == -1)
2048 stop = wav->segment.duration;
2050 gst_element_post_message (GST_ELEMENT_CAST (wav),
2051 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2052 wav->segment.format, stop));
2054 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2056 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
2057 /* for fatal errors we post an error message, post the error
2058 * first so the app knows about the error first. */
2059 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2060 (_("Internal data flow error.")),
2061 ("streaming task paused, reason %s (%d)", reason, ret));
2062 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2068 static GstFlowReturn
2069 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2072 GstWavParse *wav = GST_WAVPARSE (parent);
2074 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2075 gst_buffer_get_size (buf));
2077 gst_adapter_push (wav->adapter, buf);
2079 switch (wav->state) {
2080 case GST_WAVPARSE_START:
2081 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2082 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2085 if (wav->state != GST_WAVPARSE_HEADER)
2088 /* otherwise fall-through */
2089 case GST_WAVPARSE_HEADER:
2090 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2091 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2094 if (!wav->got_fmt || wav->datastart == 0)
2097 wav->state = GST_WAVPARSE_DATA;
2098 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2101 case GST_WAVPARSE_DATA:
2102 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2103 wav->discont = TRUE;
2104 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2108 g_return_val_if_reached (GST_FLOW_ERROR);
2111 if (G_UNLIKELY (wav->abort_buffering)) {
2112 wav->abort_buffering = FALSE;
2113 ret = GST_FLOW_ERROR;
2114 /* sort of demux/parse error */
2115 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2121 static GstFlowReturn
2122 gst_wavparse_flush_data (GstWavParse * wav)
2124 GstFlowReturn ret = GST_FLOW_OK;
2127 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2129 wav->end_offset = wav->offset + av;
2130 ret = gst_wavparse_stream_data (wav);
2137 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2139 GstWavParse *wav = GST_WAVPARSE (parent);
2140 gboolean ret = TRUE;
2142 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2144 switch (GST_EVENT_TYPE (event)) {
2145 case GST_EVENT_CAPS:
2147 /* discard, we'll come up with proper src caps */
2148 gst_event_unref (event);
2151 case GST_EVENT_SEGMENT:
2153 gint64 start, stop, offset = 0, end_offset = -1;
2156 /* some debug output */
2157 gst_event_copy_segment (event, &segment);
2158 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2161 if (wav->state != GST_WAVPARSE_DATA) {
2162 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2166 /* now we are either committed to TIME or BYTE format,
2167 * and we only expect a BYTE segment, e.g. following a seek */
2168 if (segment.format == GST_FORMAT_BYTES) {
2169 /* handle (un)signed issues */
2170 start = segment.start;
2171 stop = segment.stop;
2174 start -= wav->datastart;
2175 start = MAX (start, 0);
2179 segment.stop -= wav->datastart;
2180 segment.stop = MAX (stop, 0);
2182 if (wav->segment.format == GST_FORMAT_TIME) {
2183 guint64 bps = wav->bps;
2185 /* operating in format TIME, so we can convert */
2186 if (!bps && wav->fact)
2188 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2192 uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps);
2195 uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps);
2199 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2203 segment.start = start;
2204 segment.stop = stop;
2206 /* accept upstream's notion of segment and distribute along */
2207 segment.time = segment.start = segment.position;
2208 segment.duration = wav->segment.duration;
2209 segment.base = gst_segment_to_running_time (&wav->segment,
2210 GST_FORMAT_TIME, wav->segment.position);
2212 gst_segment_copy_into (&segment, &wav->segment);
2214 /* also store the newsegment event for the streaming thread */
2215 if (wav->start_segment)
2216 gst_event_unref (wav->start_segment);
2217 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2218 wav->start_segment = gst_event_new_segment (&segment);
2220 /* stream leftover data in current segment */
2221 gst_wavparse_flush_data (wav);
2222 /* and set up streaming thread for next one */
2223 wav->offset = offset;
2224 wav->end_offset = end_offset;
2225 if (wav->end_offset > 0) {
2226 wav->dataleft = wav->end_offset - wav->offset;
2228 /* infinity; upstream will EOS when done */
2229 wav->dataleft = G_MAXUINT64;
2232 gst_event_unref (event);
2236 /* add pad if needed so EOS is seen downstream */
2237 if (G_UNLIKELY (wav->first)) {
2239 gst_wavparse_add_src_pad (wav, NULL);
2241 /* stream leftover data in current segment */
2242 gst_wavparse_flush_data (wav);
2245 if (wav->state == GST_WAVPARSE_START)
2246 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2247 ("No valid input found before end of stream"), (NULL));
2250 case GST_EVENT_FLUSH_STOP:
2254 gst_adapter_clear (wav->adapter);
2255 wav->discont = TRUE;
2256 dur = wav->segment.duration;
2257 gst_segment_init (&wav->segment, wav->segment.format);
2258 wav->segment.duration = dur;
2262 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2270 /* convert and query stuff */
2271 static const GstFormat *
2272 gst_wavparse_get_formats (GstPad * pad)
2274 static GstFormat formats[] = {
2277 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2286 gst_wavparse_pad_convert (GstPad * pad,
2287 GstFormat src_format, gint64 src_value,
2288 GstFormat * dest_format, gint64 * dest_value)
2290 GstWavParse *wavparse;
2291 gboolean res = TRUE;
2293 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2295 if (*dest_format == src_format) {
2296 *dest_value = src_value;
2300 if ((wavparse->bps == 0) && !wavparse->fact)
2303 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2304 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2306 switch (src_format) {
2307 case GST_FORMAT_BYTES:
2308 switch (*dest_format) {
2309 case GST_FORMAT_DEFAULT:
2310 *dest_value = src_value / wavparse->bytes_per_sample;
2311 /* make sure we end up on a sample boundary */
2312 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2314 case GST_FORMAT_TIME:
2315 /* src_value + datastart = offset */
2316 GST_INFO_OBJECT (wavparse,
2317 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2319 if (wavparse->bps > 0)
2320 *dest_value = uint64_ceiling_scale (src_value, GST_SECOND,
2321 (guint64) wavparse->bps);
2322 else if (wavparse->fact) {
2323 guint64 bps = uint64_ceiling_scale_int (wavparse->datasize,
2324 wavparse->rate, wavparse->fact);
2326 *dest_value = uint64_ceiling_scale_int (src_value, GST_SECOND, bps);
2337 case GST_FORMAT_DEFAULT:
2338 switch (*dest_format) {
2339 case GST_FORMAT_BYTES:
2340 *dest_value = src_value * wavparse->bytes_per_sample;
2342 case GST_FORMAT_TIME:
2343 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2344 (guint64) wavparse->rate);
2352 case GST_FORMAT_TIME:
2353 switch (*dest_format) {
2354 case GST_FORMAT_BYTES:
2355 if (wavparse->bps > 0)
2356 *dest_value = gst_util_uint64_scale (src_value,
2357 (guint64) wavparse->bps, GST_SECOND);
2359 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2360 wavparse->rate, wavparse->fact);
2362 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2364 /* make sure we end up on a sample boundary */
2365 *dest_value -= *dest_value % wavparse->blockalign;
2367 case GST_FORMAT_DEFAULT:
2368 *dest_value = gst_util_uint64_scale (src_value,
2369 (guint64) wavparse->rate, GST_SECOND);
2388 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2394 /* handle queries for location and length in requested format */
2396 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2398 gboolean res = TRUE;
2399 GstWavParse *wav = GST_WAVPARSE (parent);
2401 /* only if we know */
2402 if (wav->state != GST_WAVPARSE_DATA) {
2406 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2408 switch (GST_QUERY_TYPE (query)) {
2409 case GST_QUERY_POSITION:
2415 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2416 curb = wav->offset - wav->datastart;
2417 gst_query_parse_position (query, &format, NULL);
2418 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2421 case GST_FORMAT_TIME:
2422 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2426 format = GST_FORMAT_BYTES;
2431 gst_query_set_position (query, format, cur);
2434 case GST_QUERY_DURATION:
2436 gint64 duration = 0;
2439 gst_query_parse_duration (query, &format, NULL);
2442 case GST_FORMAT_TIME:{
2443 if ((res = gst_wavparse_calculate_duration (wav))) {
2444 duration = wav->duration;
2449 format = GST_FORMAT_BYTES;
2450 duration = wav->datasize;
2453 gst_query_set_duration (query, format, duration);
2456 case GST_QUERY_CONVERT:
2458 gint64 srcvalue, dstvalue;
2459 GstFormat srcformat, dstformat;
2461 gst_query_parse_convert (query, &srcformat, &srcvalue,
2462 &dstformat, &dstvalue);
2463 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2464 &dstformat, &dstvalue);
2466 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2469 case GST_QUERY_SEEKING:{
2471 gboolean seekable = FALSE;
2473 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2474 if (fmt == wav->segment.format) {
2475 if (wav->streaming) {
2478 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2479 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2480 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2481 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2483 gst_query_unref (q);
2485 GST_LOG_OBJECT (wav, "looping => seekable");
2489 } else if (fmt == GST_FORMAT_TIME) {
2493 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2498 res = gst_pad_query_default (pad, parent, query);
2505 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2507 GstWavParse *wavparse = GST_WAVPARSE (parent);
2508 gboolean res = FALSE;
2510 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2512 switch (GST_EVENT_TYPE (event)) {
2513 case GST_EVENT_SEEK:
2514 /* can only handle events when we are in the data state */
2515 if (wavparse->state == GST_WAVPARSE_DATA) {
2516 res = gst_wavparse_perform_seek (wavparse, event);
2518 gst_event_unref (event);
2521 res = gst_pad_push_event (wavparse->sinkpad, event);
2528 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2530 GstWavParse *wav = GST_WAVPARSE (parent);
2535 gst_adapter_clear (wav->adapter);
2536 g_object_unref (wav->adapter);
2537 wav->adapter = NULL;
2540 query = gst_query_new_scheduling ();
2542 if (!gst_pad_peer_query (sinkpad, query)) {
2543 gst_query_unref (query);
2547 pull_mode = gst_query_has_scheduling_mode (query, GST_PAD_MODE_PULL);
2548 gst_query_unref (query);
2553 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2554 wav->streaming = FALSE;
2555 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2559 GST_DEBUG_OBJECT (sinkpad, "activating push");
2560 wav->streaming = TRUE;
2561 wav->adapter = gst_adapter_new ();
2562 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2568 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2569 GstPadMode mode, gboolean active)
2574 case GST_PAD_MODE_PUSH:
2577 case GST_PAD_MODE_PULL:
2579 /* if we have a scheduler we can start the task */
2580 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2583 res = gst_pad_stop_task (sinkpad);
2593 static GstStateChangeReturn
2594 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2596 GstStateChangeReturn ret;
2597 GstWavParse *wav = GST_WAVPARSE (element);
2599 switch (transition) {
2600 case GST_STATE_CHANGE_NULL_TO_READY:
2602 case GST_STATE_CHANGE_READY_TO_PAUSED:
2603 gst_wavparse_reset (wav);
2605 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2611 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2613 switch (transition) {
2614 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2616 case GST_STATE_CHANGE_PAUSED_TO_READY:
2617 gst_wavparse_reset (wav);
2619 case GST_STATE_CHANGE_READY_TO_NULL:
2628 plugin_init (GstPlugin * plugin)
2632 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2636 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2639 "Parse a .wav file into raw audio",
2640 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)