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);
67 static gboolean gst_wavparse_sink_activate_pull (GstPad * sinkpad,
69 static gboolean gst_wavparse_send_event (GstElement * element,
71 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
72 GstStateChange transition);
74 static const GstQueryType *gst_wavparse_get_query_types (GstPad * pad);
75 static gboolean gst_wavparse_pad_query (GstPad * pad, GstQuery * query);
76 static gboolean gst_wavparse_pad_convert (GstPad * pad,
78 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
80 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
81 static gboolean gst_wavparse_sink_event (GstPad * pad, GstEvent * event);
82 static void gst_wavparse_loop (GstPad * pad);
83 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
85 static GstStaticPadTemplate sink_template_factory =
86 GST_STATIC_PAD_TEMPLATE ("sink",
89 GST_STATIC_CAPS ("audio/x-wav")
93 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
95 #define gst_wavparse_parent_class parent_class
96 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
100 gst_wavparse_class_init (GstWavParseClass * klass)
102 GstElementClass *gstelement_class;
103 GObjectClass *object_class;
104 GstPadTemplate *src_template;
106 gstelement_class = (GstElementClass *) klass;
107 object_class = (GObjectClass *) klass;
109 parent_class = g_type_class_peek_parent (klass);
111 object_class->dispose = gst_wavparse_dispose;
113 gstelement_class->change_state = gst_wavparse_change_state;
114 gstelement_class->send_event = gst_wavparse_send_event;
117 gst_element_class_add_pad_template (gstelement_class,
118 gst_static_pad_template_get (&sink_template_factory));
120 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
121 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
122 gst_element_class_add_pad_template (gstelement_class, src_template);
124 gst_element_class_set_details_simple (gstelement_class, "WAV audio demuxer",
125 "Codec/Demuxer/Audio",
126 "Parse a .wav file into raw audio",
127 "Erik Walthinsen <omega@cse.ogi.edu>");
131 gst_wavparse_reset (GstWavParse * wav)
133 wav->state = GST_WAVPARSE_START;
135 /* These will all be set correctly in the fmt chunk */
149 wav->got_fmt = FALSE;
153 gst_event_unref (wav->seek_event);
154 wav->seek_event = NULL;
156 gst_adapter_clear (wav->adapter);
157 g_object_unref (wav->adapter);
161 gst_tag_list_free (wav->tags);
164 gst_caps_unref (wav->caps);
166 if (wav->start_segment)
167 gst_event_unref (wav->start_segment);
168 wav->start_segment = NULL;
172 gst_wavparse_dispose (GObject * object)
174 GstWavParse *wav = GST_WAVPARSE (object);
176 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
177 gst_wavparse_reset (wav);
179 G_OBJECT_CLASS (parent_class)->dispose (object);
183 gst_wavparse_init (GstWavParse * wavparse)
185 gst_wavparse_reset (wavparse);
189 gst_pad_new_from_static_template (&sink_template_factory, "sink");
190 gst_pad_set_activate_function (wavparse->sinkpad,
191 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
192 gst_pad_set_activatepull_function (wavparse->sinkpad,
193 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
194 gst_pad_set_chain_function (wavparse->sinkpad,
195 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
196 gst_pad_set_event_function (wavparse->sinkpad,
197 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
198 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
202 gst_pad_new_from_template (gst_element_class_get_pad_template
203 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
204 gst_pad_use_fixed_caps (wavparse->srcpad);
205 gst_pad_set_query_type_function (wavparse->srcpad,
206 GST_DEBUG_FUNCPTR (gst_wavparse_get_query_types));
207 gst_pad_set_query_function (wavparse->srcpad,
208 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
209 gst_pad_set_event_function (wavparse->srcpad,
210 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
211 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
215 gst_wavparse_destroy_sourcepad (GstWavParse * wavparse)
217 if (wavparse->srcpad) {
218 gst_element_remove_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
219 wavparse->srcpad = NULL;
223 /* Compute (value * nom) % denom, avoiding overflow. This can be used
224 * to perform ceiling or rounding division together with
225 * gst_util_uint64_scale[_int]. */
226 #define uint64_scale_modulo(val, nom, denom) \
227 ((val % denom) * (nom % denom) % denom)
229 /* Like gst_util_uint64_scale, but performs ceiling division. */
231 uint64_ceiling_scale_int (guint64 val, gint num, gint denom)
233 guint64 result = gst_util_uint64_scale_int (val, num, denom);
235 if (uint64_scale_modulo (val, num, denom) == 0)
241 /* Like gst_util_uint64_scale, but performs ceiling division. */
243 uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom)
245 guint64 result = gst_util_uint64_scale (val, num, denom);
247 if (uint64_scale_modulo (val, num, denom) == 0)
254 /* FIXME: why is that not in use? */
257 gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
260 GstByteStream *bs = wavparse->bs;
261 gst_riff_chunk *temp_chunk, chunk;
263 struct _gst_riff_labl labl, *temp_labl;
264 struct _gst_riff_ltxt ltxt, *temp_ltxt;
265 struct _gst_riff_note note, *temp_note;
268 GstPropsEntry *entry;
272 props = wavparse->metadata->properties;
276 gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
277 if (got_bytes != sizeof (gst_riff_chunk)) {
280 temp_chunk = (gst_riff_chunk *) tempdata;
282 chunk.id = GUINT32_FROM_LE (temp_chunk->id);
283 chunk.size = GUINT32_FROM_LE (temp_chunk->size);
285 if (chunk.size == 0) {
286 gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
287 len -= sizeof (gst_riff_chunk);
292 case GST_RIFF_adtl_labl:
294 gst_bytestream_peek_bytes (bs, &tempdata,
295 sizeof (struct _gst_riff_labl));
296 if (got_bytes != sizeof (struct _gst_riff_labl)) {
300 temp_labl = (struct _gst_riff_labl *) tempdata;
301 labl.id = GUINT32_FROM_LE (temp_labl->id);
302 labl.size = GUINT32_FROM_LE (temp_labl->size);
303 labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
305 gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
306 len -= sizeof (struct _gst_riff_labl);
308 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
309 if (got_bytes != labl.size - 4) {
313 label_name = (char *) tempdata;
315 gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
316 len -= (((labl.size - 4) + 1) & ~1);
318 new_caps = gst_caps_new ("label",
319 "application/x-gst-metadata",
320 gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
321 "name", G_TYPE_STRING (label_name), NULL));
323 if (gst_props_get (props, "labels", &caps, NULL)) {
324 caps = g_list_append (caps, new_caps);
326 caps = g_list_append (NULL, new_caps);
328 entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
329 gst_props_add_entry (props, entry);
334 case GST_RIFF_adtl_ltxt:
336 gst_bytestream_peek_bytes (bs, &tempdata,
337 sizeof (struct _gst_riff_ltxt));
338 if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
342 temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
343 ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
344 ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
345 ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
346 ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
347 ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
348 ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
349 ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
350 ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
351 ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
353 gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
354 len -= sizeof (struct _gst_riff_ltxt);
356 if (ltxt.size - 20 > 0) {
357 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
358 if (got_bytes != ltxt.size - 20) {
362 gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
363 len -= (((ltxt.size - 20) + 1) & ~1);
365 label_name = (char *) tempdata;
370 new_caps = gst_caps_new ("ltxt",
371 "application/x-gst-metadata",
372 gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
373 "name", G_TYPE_STRING (label_name),
374 "length", G_TYPE_INT (ltxt.length), NULL));
376 if (gst_props_get (props, "ltxts", &caps, NULL)) {
377 caps = g_list_append (caps, new_caps);
379 caps = g_list_append (NULL, new_caps);
381 entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
382 gst_props_add_entry (props, entry);
387 case GST_RIFF_adtl_note:
389 gst_bytestream_peek_bytes (bs, &tempdata,
390 sizeof (struct _gst_riff_note));
391 if (got_bytes != sizeof (struct _gst_riff_note)) {
395 temp_note = (struct _gst_riff_note *) tempdata;
396 note.id = GUINT32_FROM_LE (temp_note->id);
397 note.size = GUINT32_FROM_LE (temp_note->size);
398 note.identifier = GUINT32_FROM_LE (temp_note->identifier);
400 gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
401 len -= sizeof (struct _gst_riff_note);
403 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
404 if (got_bytes != note.size - 4) {
408 gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
409 len -= (((note.size - 4) + 1) & ~1);
411 label_name = (char *) tempdata;
413 new_caps = gst_caps_new ("note",
414 "application/x-gst-metadata",
415 gst_props_new ("identifier", G_TYPE_INT (note.identifier),
416 "name", G_TYPE_STRING (label_name), NULL));
418 if (gst_props_get (props, "notes", &caps, NULL)) {
419 caps = g_list_append (caps, new_caps);
421 caps = g_list_append (NULL, new_caps);
423 entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
424 gst_props_add_entry (props, entry);
430 g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
431 GST_FOURCC_ARGS (chunk.id));
436 g_object_notify (G_OBJECT (wavparse), "metadata");
440 gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
443 GstByteStream *bs = wavparse->bs;
444 struct _gst_riff_cue *temp_cue, cue;
445 struct _gst_riff_cuepoints *points;
449 GstPropsEntry *entry;
455 gst_bytestream_peek_bytes (bs, &tempdata,
456 sizeof (struct _gst_riff_cue));
457 temp_cue = (struct _gst_riff_cue *) tempdata;
459 /* fixup for our big endian friends */
460 cue.id = GUINT32_FROM_LE (temp_cue->id);
461 cue.size = GUINT32_FROM_LE (temp_cue->size);
462 cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
464 gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
465 if (got_bytes != sizeof (struct _gst_riff_cue)) {
469 len -= sizeof (struct _gst_riff_cue);
471 /* -4 because cue.size contains the cuepoints size
472 and we've already flushed that out of the system */
473 required = cue.size - 4;
474 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
475 gst_bytestream_flush (bs, ((required) + 1) & ~1);
476 if (got_bytes != required) {
480 len -= (((cue.size - 4) + 1) & ~1);
482 /* now we have an array of struct _gst_riff_cuepoints in tempdata */
483 points = (struct _gst_riff_cuepoints *) tempdata;
485 for (i = 0; i < cue.cuepoints; i++) {
488 caps = gst_caps_new ("cues",
489 "application/x-gst-metadata",
490 gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
491 "position", G_TYPE_INT (points[i].offset), NULL));
492 cues = g_list_append (cues, caps);
495 entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
496 gst_props_add_entry (wavparse->metadata->properties, entry);
499 g_object_notify (G_OBJECT (wavparse), "metadata");
502 /* Read 'fmt ' header */
504 gst_wavparse_fmt (GstWavParse * wav)
506 gst_riff_strf_auds *header = NULL;
509 if (!gst_riff_read_strf_auds (wav, &header))
512 wav->format = header->format;
513 wav->rate = header->rate;
514 wav->channels = header->channels;
515 if (wav->channels == 0)
518 wav->blockalign = header->blockalign;
519 wav->width = (header->blockalign * 8) / header->channels;
520 wav->depth = header->size;
521 wav->bps = header->av_bps;
525 /* Note: gst_riff_create_audio_caps might need to fix values in
526 * the header header depending on the format, so call it first */
527 caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL);
533 gst_wavparse_create_sourcepad (wav);
534 gst_pad_use_fixed_caps (wav->srcpad);
535 gst_pad_set_active (wav->srcpad, TRUE);
536 gst_pad_set_caps (wav->srcpad, caps);
537 gst_caps_free (caps);
538 gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
539 gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
541 GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels);
548 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
549 ("No FMT tag found"));
554 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
555 ("Stream claims to contain zero channels - invalid data"));
561 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
562 ("Stream claims to bitrate of <= zero - invalid data"));
568 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
574 gst_wavparse_other (GstWavParse * wav)
578 if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
579 GST_WARNING_OBJECT (wav, "could not peek head");
582 GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag,
583 (gchar *) & tag, length);
586 case GST_RIFF_TAG_LIST:
587 if (!(tag = gst_riff_peek_list (wav))) {
588 GST_WARNING_OBJECT (wav, "could not peek list");
593 case GST_RIFF_LIST_INFO:
594 if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
595 GST_WARNING_OBJECT (wav, "could not read list");
600 case GST_RIFF_LIST_adtl:
601 if (!gst_riff_read_skip (wav)) {
602 GST_WARNING_OBJECT (wav, "could not read skip");
608 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
610 if (!gst_riff_read_skip (wav)) {
611 GST_WARNING_OBJECT (wav, "could not read skip");
619 case GST_RIFF_TAG_data:
620 if (!gst_bytestream_flush (wav->bs, 8)) {
621 GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
625 GST_DEBUG_OBJECT (wav, "switching to data mode");
626 wav->state = GST_WAVPARSE_DATA;
627 wav->datastart = gst_bytestream_tell (wav->bs);
631 /* length is 0, data probably stretches to the end
633 GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
634 /* get length of file */
635 file_length = gst_bytestream_length (wav->bs);
636 if (file_length == -1) {
637 GST_DEBUG_OBJECT (wav,
638 "could not get file length, assuming data to eof");
639 /* could not get length, assuming till eof */
640 length = G_MAXUINT32;
642 if (file_length > G_MAXUINT32) {
643 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
644 ", clipping to 32 bits", file_length);
645 /* could not get length, assuming till eof */
646 length = G_MAXUINT32;
648 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
649 ", datalength %u", file_length, length);
650 /* substract offset of datastart from length */
651 length = file_length - wav->datastart;
652 GST_DEBUG_OBJECT (wav, "datalength %u", length);
655 wav->datasize = (guint64) length;
656 GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
659 case GST_RIFF_TAG_cue:
660 if (!gst_riff_read_skip (wav)) {
661 GST_WARNING_OBJECT (wav, "could not read skip");
667 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
668 if (!gst_riff_read_skip (wav))
679 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
683 if (!gst_riff_parse_file_header (element, buf, &doctype))
686 if (doctype != GST_RIFF_RIFF_WAVE)
694 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
695 ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
696 GST_FOURCC_ARGS (doctype)));
702 gst_wavparse_stream_init (GstWavParse * wav)
705 GstBuffer *buf = NULL;
707 if ((res = gst_pad_pull_range (wav->sinkpad,
708 wav->offset, 12, &buf)) != GST_FLOW_OK)
710 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
711 return GST_FLOW_ERROR;
719 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
721 /* -1 always maps to -1 */
727 /* 0 always maps to 0 */
734 *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND);
736 } else if (wav->fact) {
738 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
739 *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND);
746 /* This function is used to perform seeks on the element.
748 * It also works when event is NULL, in which case it will just
749 * start from the last configured segment. This technique is
750 * used when activating the element and to perform the seek in
754 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
758 GstFormat format, bformat;
760 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
761 gint64 cur, stop, upstream_size;
764 GstSegment seeksegment = { 0, };
768 GST_DEBUG_OBJECT (wav, "doing seek with event");
770 gst_event_parse_seek (event, &rate, &format, &flags,
771 &cur_type, &cur, &stop_type, &stop);
773 /* no negative rates yet */
777 if (format != wav->segment.format) {
778 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
779 gst_format_get_name (format),
780 gst_format_get_name (wav->segment.format));
782 if (cur_type != GST_SEEK_TYPE_NONE)
784 gst_pad_query_convert (wav->srcpad, format, cur,
785 &wav->segment.format, &cur);
786 if (res && stop_type != GST_SEEK_TYPE_NONE)
788 gst_pad_query_convert (wav->srcpad, format, stop,
789 &wav->segment.format, &stop);
793 format = wav->segment.format;
796 GST_DEBUG_OBJECT (wav, "doing seek without event");
799 cur_type = GST_SEEK_TYPE_SET;
800 stop_type = GST_SEEK_TYPE_SET;
803 /* in push mode, we must delegate to upstream */
804 if (wav->streaming) {
805 gboolean res = FALSE;
807 /* if streaming not yet started; only prepare initial newsegment */
808 if (!event || wav->state != GST_WAVPARSE_DATA) {
809 if (wav->start_segment)
810 gst_event_unref (wav->start_segment);
812 /* wav->start_segment =
813 gst_event_new_new_segment (FALSE, wav->segment.rate,
814 wav->segment.format, wav->segment.last_stop, wav->segment.duration,
815 wav->segment.last_stop);*/
818 /* convert seek positions to byte positions in data sections */
819 if (format == GST_FORMAT_TIME) {
820 /* should not fail */
821 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
823 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
826 /* mind sample boundary and header */
828 cur -= (cur % wav->bytes_per_sample);
829 cur += wav->datastart;
832 stop -= (stop % wav->bytes_per_sample);
833 stop += wav->datastart;
835 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
836 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
838 /* BYTE seek event */
839 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
841 res = gst_pad_push_event (wav->sinkpad, event);
847 flush = flags & GST_SEEK_FLAG_FLUSH;
849 /* now we need to make sure the streaming thread is stopped. We do this by
850 * either sending a FLUSH_START event downstream which will cause the
851 * streaming thread to stop with a WRONG_STATE.
852 * For a non-flushing seek we simply pause the task, which will happen as soon
853 * as it completes one iteration (and thus might block when the sink is
854 * blocking in preroll). */
857 GST_DEBUG_OBJECT (wav, "sending flush start");
858 gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
861 gst_pad_pause_task (wav->sinkpad);
864 /* we should now be able to grab the streaming thread because we stopped it
865 * with the above flush/pause code */
866 GST_PAD_STREAM_LOCK (wav->sinkpad);
868 /* save current position */
869 last_stop = wav->segment.position;
871 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
873 /* copy segment, we need this because we still need the old
874 * segment when we close the current segment. */
875 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
877 /* configure the seek parameters in the seeksegment. We will then have the
878 * right values in the segment to perform the seek */
880 GST_DEBUG_OBJECT (wav, "configuring seek");
881 gst_segment_do_seek (&seeksegment, rate, format, flags,
882 cur_type, cur, stop_type, stop, &update);
885 /* figure out the last position we need to play. If it's configured (stop !=
886 * -1), use that, else we play until the total duration of the file */
887 if ((stop = seeksegment.stop) == -1)
888 stop = seeksegment.duration;
890 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
891 if ((cur_type != GST_SEEK_TYPE_NONE)) {
892 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
893 * we can just copy the last_stop. If not, we use the bps to convert TIME to
895 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
896 (gint64 *) & wav->offset))
897 wav->offset = seeksegment.position;
898 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
899 wav->offset -= (wav->offset % wav->bytes_per_sample);
900 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
901 wav->offset += wav->datastart;
902 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
904 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
908 if (stop_type != GST_SEEK_TYPE_NONE) {
909 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
910 wav->end_offset = stop;
911 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
912 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
913 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
914 wav->end_offset += wav->datastart;
915 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
917 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
921 /* make sure filesize is not exceeded due to rounding errors or so,
922 * same precaution as in _stream_headers */
923 bformat = GST_FORMAT_BYTES;
924 if (gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size))
925 wav->end_offset = MIN (wav->end_offset, upstream_size);
927 /* this is the range of bytes we will use for playback */
928 wav->offset = MIN (wav->offset, wav->end_offset);
929 wav->dataleft = wav->end_offset - wav->offset;
931 GST_DEBUG_OBJECT (wav,
932 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
933 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
934 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
936 /* prepare for streaming again */
939 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
940 GST_DEBUG_OBJECT (wav, "sending flush stop");
941 gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
945 /* now we did the seek and can activate the new segment values */
946 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
948 /* if we're doing a segment seek, post a SEGMENT_START message */
949 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
950 gst_element_post_message (GST_ELEMENT_CAST (wav),
951 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
952 wav->segment.format, wav->segment.position));
955 /* now create the newsegment */
956 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
957 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
959 /* store the newsegment event so it can be sent from the streaming thread. */
960 if (wav->start_segment)
961 gst_event_unref (wav->start_segment);
962 wav->start_segment = gst_event_new_segment (&wav->segment);
964 /* mark discont if we are going to stream from another position. */
965 if (last_stop != wav->segment.position) {
966 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
970 /* and start the streaming task again */
971 if (!wav->streaming) {
972 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
976 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
983 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
988 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
993 GST_DEBUG_OBJECT (wav,
994 "Could not determine byte position for desired time");
1000 * gst_wavparse_peek_chunk_info:
1001 * @wav Wavparse object
1002 * @tag holder for tag
1003 * @size holder for tag size
1005 * Peek next chunk info (tag and size)
1007 * Returns: %TRUE when the chunk info (header) is available
1010 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
1012 const guint8 *data = NULL;
1014 if (gst_adapter_available (wav->adapter) < 8)
1017 data = gst_adapter_map (wav->adapter, 8);
1018 *tag = GST_READ_UINT32_LE (data);
1019 *size = GST_READ_UINT32_LE (data + 4);
1020 gst_adapter_unmap (wav->adapter, 0);
1022 GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size,
1023 GST_FOURCC_ARGS (*tag));
1029 * gst_wavparse_peek_chunk:
1030 * @wav Wavparse object
1031 * @tag holder for tag
1032 * @size holder for tag size
1034 * Peek enough data for one full chunk
1036 * Returns: %TRUE when the full chunk is available
1039 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
1041 guint32 peek_size = 0;
1044 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
1047 /* size 0 -> empty data buffer would surprise most callers,
1048 * large size -> do not bother trying to squeeze that into adapter,
1049 * so we throw poor man's exception, which can be caught if caller really
1050 * wants to handle 0 size chunk */
1051 if (!(*size) || (*size) >= (1 << 30)) {
1052 GST_INFO ("Invalid/unexpected chunk size %d for tag %" GST_FOURCC_FORMAT,
1053 *size, GST_FOURCC_ARGS (*tag));
1054 /* chain should give up */
1055 wav->abort_buffering = TRUE;
1058 peek_size = (*size + 1) & ~1;
1059 available = gst_adapter_available (wav->adapter);
1061 if (available >= (8 + peek_size)) {
1064 GST_LOG ("but only %u bytes available now", available);
1070 * gst_wavparse_calculate_duration:
1071 * @wav: wavparse object
1073 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
1076 * Returns: %TRUE if duration is available.
1079 gst_wavparse_calculate_duration (GstWavParse * wav)
1081 if (wav->duration > 0)
1085 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
1087 uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps);
1088 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
1089 GST_TIME_ARGS (wav->duration));
1091 } else if (wav->fact) {
1092 wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate);
1093 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
1094 GST_TIME_ARGS (wav->duration));
1101 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
1106 if (wav->streaming) {
1107 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1110 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
1111 GST_FOURCC_ARGS (tag));
1112 flush = 8 + ((size + 1) & ~1);
1113 wav->offset += flush;
1114 if (wav->streaming) {
1115 gst_adapter_flush (wav->adapter, flush);
1117 gst_buffer_unref (buf);
1123 #define MAX_BUFFER_SIZE 4096
1125 static GstFlowReturn
1126 gst_wavparse_stream_headers (GstWavParse * wav)
1128 GstFlowReturn res = GST_FLOW_OK;
1129 GstBuffer *buf = NULL;
1130 gst_riff_strf_auds *header = NULL;
1132 gboolean gotdata = FALSE;
1133 GstCaps *caps = NULL;
1134 gchar *codec_name = NULL;
1137 gint64 upstream_size = 0;
1139 /* search for "_fmt" chunk, which should be first */
1140 while (!wav->got_fmt) {
1143 /* The header starts with a 'fmt ' tag */
1144 if (wav->streaming) {
1145 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1148 gst_adapter_flush (wav->adapter, 8);
1152 buf = gst_adapter_take_buffer (wav->adapter, size);
1154 gst_adapter_flush (wav->adapter, 1);
1155 wav->offset += GST_ROUND_UP_2 (size);
1157 buf = gst_buffer_new ();
1160 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1161 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1165 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1166 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1167 tag == GST_RIFF_TAG_LIST) {
1168 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1169 GST_FOURCC_ARGS (tag));
1170 gst_buffer_unref (buf);
1175 if (tag != GST_RIFF_TAG_fmt)
1178 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1180 goto parse_header_error;
1182 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1184 /* do sanity checks of header fields */
1185 if (header->channels == 0)
1187 if (header->rate == 0)
1190 GST_DEBUG_OBJECT (wav, "creating the caps");
1192 /* Note: gst_riff_create_audio_caps might need to fix values in
1193 * the header header depending on the format, so call it first */
1194 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1198 gst_buffer_unref (extra);
1201 goto unknown_format;
1203 /* do more sanity checks of header fields
1204 * (these can be sanitized by gst_riff_create_audio_caps()
1206 wav->format = header->format;
1207 wav->rate = header->rate;
1208 wav->channels = header->channels;
1209 wav->blockalign = header->blockalign;
1210 wav->depth = header->size;
1211 wav->av_bps = header->av_bps;
1217 /* do format specific handling */
1218 switch (wav->format) {
1219 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1220 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1222 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1223 * bitrate inside the mpeg stream */
1224 GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps);
1228 case GST_RIFF_WAVE_FORMAT_PCM:
1229 if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0))
1230 goto invalid_blockalign;
1233 if (wav->av_bps > wav->blockalign * wav->rate)
1235 /* use the configured bps */
1236 wav->bps = wav->av_bps;
1240 wav->width = (wav->blockalign * 8) / wav->channels;
1241 wav->bytes_per_sample = wav->channels * wav->width / 8;
1243 if (wav->bytes_per_sample <= 0)
1244 goto no_bytes_per_sample;
1246 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1247 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1248 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1249 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1250 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1251 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1252 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1254 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1255 * formats). This will make the element output a BYTE format segment and
1256 * will not timestamp the outgoing buffers.
1258 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1260 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1262 /* create pad later so we can sniff the first few bytes
1263 * of the real data and correct our caps if necessary */
1264 gst_caps_replace (&wav->caps, caps);
1265 gst_caps_replace (&caps, NULL);
1267 wav->got_fmt = TRUE;
1270 wav->tags = gst_tag_list_new ();
1272 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1273 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1275 g_free (codec_name);
1281 bformat = GST_FORMAT_BYTES;
1282 gst_pad_query_peer_duration (wav->sinkpad, &bformat, &upstream_size);
1283 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1285 /* loop headers until we get data */
1287 if (wav->streaming) {
1288 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1294 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1295 &buf)) != GST_FLOW_OK)
1296 goto header_read_error;
1297 data = gst_buffer_map (buf, NULL, NULL, -1);
1298 tag = GST_READ_UINT32_LE (data);
1299 size = GST_READ_UINT32_LE (data + 4);
1300 gst_buffer_unmap (buf, data, -1);
1303 GST_INFO_OBJECT (wav,
1304 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1305 GST_FOURCC_ARGS (tag), wav->offset);
1307 /* wav is a st00pid format, we don't know for sure where data starts.
1308 * So we have to go bit by bit until we find the 'data' header
1311 case GST_RIFF_TAG_data:{
1312 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size);
1313 if (wav->streaming) {
1314 gst_adapter_flush (wav->adapter, 8);
1317 gst_buffer_unref (buf);
1320 wav->datastart = wav->offset;
1321 /* If size is zero, then the data chunk probably actually extends to
1322 the end of the file */
1323 if (size == 0 && upstream_size) {
1324 size = upstream_size - wav->datastart;
1326 /* Or the file might be truncated */
1327 else if (upstream_size) {
1328 size = MIN (size, (upstream_size - wav->datastart));
1330 wav->datasize = (guint64) size;
1331 wav->dataleft = (guint64) size;
1332 wav->end_offset = size + wav->datastart;
1333 if (!wav->streaming) {
1334 /* We will continue parsing tags 'till end */
1335 wav->offset += size;
1337 GST_DEBUG_OBJECT (wav, "datasize = %d", size);
1340 case GST_RIFF_TAG_fact:{
1341 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1342 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1343 const guint data_size = 4;
1345 GST_INFO_OBJECT (wav, "Have fact chunk");
1346 if (size < data_size) {
1347 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1348 /* need more data */
1351 GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
1355 /* number of samples (for compressed formats) */
1356 if (wav->streaming) {
1357 const guint8 *data = NULL;
1359 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1362 gst_adapter_flush (wav->adapter, 8);
1363 data = gst_adapter_map (wav->adapter, data_size);
1364 wav->fact = GST_READ_UINT32_LE (data);
1365 gst_adapter_unmap (wav->adapter, GST_ROUND_UP_2 (size));
1367 gst_buffer_unref (buf);
1369 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1370 data_size, &buf)) != GST_FLOW_OK)
1371 goto header_read_error;
1372 gst_buffer_extract (buf, 0, &wav->fact, 4);
1373 wav->fact = GUINT32_FROM_LE (wav->fact);
1374 gst_buffer_unref (buf);
1376 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1377 wav->offset += 8 + GST_ROUND_UP_2 (size);
1380 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1381 /* need more data */
1387 case GST_RIFF_TAG_acid:{
1388 const gst_riff_acid *acid = NULL;
1389 const guint data_size = sizeof (gst_riff_acid);
1392 GST_INFO_OBJECT (wav, "Have acid chunk");
1393 if (size < data_size) {
1394 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1395 /* need more data */
1398 GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk",
1402 if (wav->streaming) {
1403 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1406 gst_adapter_flush (wav->adapter, 8);
1407 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1409 tempo = acid->tempo;
1410 gst_adapter_unmap (wav->adapter, 0);
1412 gst_buffer_unref (buf);
1414 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1415 size, &buf)) != GST_FLOW_OK)
1416 goto header_read_error;
1417 acid = (const gst_riff_acid *) gst_buffer_map (buf, NULL, NULL,
1419 tempo = acid->tempo;
1420 gst_buffer_unmap (buf, (guint8 *) acid, -1);
1422 /* send data as tags */
1424 wav->tags = gst_tag_list_new ();
1425 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1426 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1428 size = GST_ROUND_UP_2 (size);
1429 if (wav->streaming) {
1430 gst_adapter_flush (wav->adapter, size);
1432 gst_buffer_unref (buf);
1434 wav->offset += 8 + size;
1437 /* FIXME: all list tags after data are ignored in streaming mode */
1438 case GST_RIFF_TAG_LIST:{
1441 if (wav->streaming) {
1442 const guint8 *data = NULL;
1444 if (gst_adapter_available (wav->adapter) < 12) {
1447 data = gst_adapter_map (wav->adapter, 12);
1448 ltag = GST_READ_UINT32_LE (data + 8);
1449 gst_adapter_unmap (wav->adapter, 0);
1451 gst_buffer_unref (buf);
1453 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1454 &buf)) != GST_FLOW_OK)
1455 goto header_read_error;
1456 gst_buffer_extract (buf, 8, <ag, 4);
1457 ltag = GUINT32_FROM_LE (ltag);
1460 case GST_RIFF_LIST_INFO:{
1461 const gint data_size = size - 4;
1464 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1465 if (wav->streaming) {
1466 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1469 gst_adapter_flush (wav->adapter, 12);
1471 if (data_size > 0) {
1472 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1474 gst_adapter_flush (wav->adapter, 1);
1478 gst_buffer_unref (buf);
1479 if (data_size > 0) {
1481 gst_pad_pull_range (wav->sinkpad, wav->offset,
1482 data_size, &buf)) != GST_FLOW_OK)
1483 goto header_read_error;
1486 if (data_size > 0) {
1488 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1490 GstTagList *old = wav->tags;
1492 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1494 gst_tag_list_free (old);
1495 gst_tag_list_free (new);
1497 gst_buffer_unref (buf);
1498 wav->offset += GST_ROUND_UP_2 (data_size);
1503 GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1504 GST_FOURCC_ARGS (ltag));
1505 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1506 /* need more data */
1513 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1514 /* need more data */
1519 if (upstream_size && (wav->offset >= upstream_size)) {
1520 /* Now we are gone through the whole file */
1525 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1527 if (wav->bps <= 0 && wav->fact) {
1529 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1531 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1532 (guint64) wav->fact);
1533 GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps);
1538 if (gst_wavparse_calculate_duration (wav)) {
1539 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1540 wav->segment.duration = wav->duration;
1542 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1543 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1544 wav->segment.duration = wav->datasize;
1547 /* now we have all the info to perform a pending seek if any, if no
1548 * event, this will still do the right thing and it will also send
1549 * the right newsegment event downstream. */
1550 gst_wavparse_perform_seek (wav, wav->seek_event);
1551 /* remove pending event */
1552 event_p = &wav->seek_event;
1553 gst_event_replace (event_p, NULL);
1555 /* we just started, we are discont */
1556 wav->discont = TRUE;
1558 wav->state = GST_WAVPARSE_DATA;
1560 /* determine reasonable max buffer size,
1561 * that is, buffers not too small either size or time wise
1562 * so we do not end up with too many of them */
1565 gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
1566 wav->max_buf_size = upstream_size;
1567 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1568 if (wav->blockalign > 0)
1569 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1571 GST_DEBUG_OBJECT (wav, "max buffer size %d", wav->max_buf_size);
1579 g_free (codec_name);
1583 gst_caps_unref (caps);
1588 res = GST_FLOW_ERROR;
1593 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1594 ("Invalid WAV header (no fmt at start): %"
1595 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1600 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1601 ("Couldn't parse audio header"));
1606 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1607 ("Stream claims to contain no channels - invalid data"));
1612 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1613 ("Stream with sample_rate == 0 - invalid data"));
1618 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1619 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1620 wav->blockalign, wav->channels * (guint) ceil (wav->depth / 8.0)));
1625 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1626 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1627 wav->av_bps, wav->blockalign * wav->rate));
1630 no_bytes_per_sample:
1632 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1633 ("Could not caluclate bytes per sample - invalid data"));
1638 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1639 ("No caps found for format 0x%x, %d channels, %d Hz",
1640 wav->format, wav->channels, wav->rate));
1645 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1646 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1652 * Read WAV file tag when streaming
1654 static GstFlowReturn
1655 gst_wavparse_parse_stream_init (GstWavParse * wav)
1657 if (gst_adapter_available (wav->adapter) >= 12) {
1660 /* _take flushes the data */
1661 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1663 GST_DEBUG ("Parsing wav header");
1664 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1665 return GST_FLOW_ERROR;
1668 /* Go to next state */
1669 wav->state = GST_WAVPARSE_HEADER;
1674 /* handle an event sent directly to the element.
1676 * This event can be sent either in the READY state or the
1677 * >READY state. The only event of interest really is the seek
1680 * In the READY state we can only store the event and try to
1681 * respect it when going to PAUSED. We assume we are in the
1682 * READY state when our parsing state != GST_WAVPARSE_DATA.
1684 * When we are steaming, we can simply perform the seek right
1688 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1690 GstWavParse *wav = GST_WAVPARSE (element);
1691 gboolean res = FALSE;
1694 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
1696 switch (GST_EVENT_TYPE (event)) {
1697 case GST_EVENT_SEEK:
1698 if (wav->state == GST_WAVPARSE_DATA) {
1699 /* we can handle the seek directly when streaming data */
1700 res = gst_wavparse_perform_seek (wav, event);
1702 GST_DEBUG_OBJECT (wav, "queuing seek for later");
1704 event_p = &wav->seek_event;
1705 gst_event_replace (event_p, event);
1707 /* we always return true */
1714 gst_event_unref (event);
1719 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
1723 s = gst_caps_get_structure (caps, 0);
1724 if (!gst_structure_has_name (s, "audio/x-dts"))
1726 if (prob >= GST_TYPE_FIND_LIKELY)
1728 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
1729 if (prob < GST_TYPE_FIND_POSSIBLE)
1731 /* .. in which case we want at least a valid-looking rate and channels */
1732 if (!gst_structure_has_field (s, "channels"))
1734 /* and for extra assurance we could also check the rate from the DTS frame
1735 * against the one in the wav header, but for now let's not do that */
1736 return gst_structure_has_field (s, "rate");
1740 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
1744 GST_DEBUG_OBJECT (wav, "adding src pad");
1747 s = gst_caps_get_structure (wav->caps, 0);
1748 if (s && gst_structure_has_name (s, "audio/x-raw-int") && buf != NULL) {
1749 GstTypeFindProbability prob;
1752 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
1753 if (tf_caps != NULL) {
1754 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
1755 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
1756 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
1757 gst_caps_unref (wav->caps);
1758 wav->caps = tf_caps;
1760 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1761 GST_TAG_AUDIO_CODEC, "dts", NULL);
1763 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
1764 "marked as raw PCM audio, but ignoring for now", tf_caps);
1765 gst_caps_unref (tf_caps);
1771 gst_pad_set_caps (wav->srcpad, wav->caps);
1772 gst_caps_replace (&wav->caps, NULL);
1774 if (wav->start_segment) {
1775 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
1776 gst_pad_push_event (wav->srcpad, wav->start_segment);
1777 wav->start_segment = NULL;
1781 gst_element_found_tags_for_pad (GST_ELEMENT_CAST (wav), wav->srcpad,
1787 static GstFlowReturn
1788 gst_wavparse_stream_data (GstWavParse * wav)
1790 GstBuffer *buf = NULL;
1791 GstFlowReturn res = GST_FLOW_OK;
1792 guint64 desired, obtained;
1793 GstClockTime timestamp, next_timestamp, duration;
1794 guint64 pos, nextpos;
1797 GST_LOG_OBJECT (wav,
1798 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
1799 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
1801 /* Get the next n bytes and output them */
1802 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
1805 /* scale the amount of data by the segment rate so we get equal
1806 * amounts of data regardless of the playback rate */
1808 MIN (gst_guint64_to_gdouble (wav->dataleft),
1809 wav->max_buf_size * ABS (wav->segment.rate));
1811 if (desired >= wav->blockalign && wav->blockalign > 0)
1812 desired -= (desired % wav->blockalign);
1814 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
1815 "from the sinkpad", desired);
1817 if (wav->streaming) {
1818 guint avail = gst_adapter_available (wav->adapter);
1821 /* flush some bytes if evil upstream sends segment that starts
1822 * before data or does is not send sample aligned segment */
1823 if (G_LIKELY (wav->offset >= wav->datastart)) {
1824 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
1826 extra = wav->datastart - wav->offset;
1829 if (G_UNLIKELY (extra)) {
1830 extra = wav->bytes_per_sample - extra;
1831 if (extra <= avail) {
1832 GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra);
1833 gst_adapter_flush (wav->adapter, extra);
1834 wav->offset += extra;
1835 wav->dataleft -= extra;
1836 goto iterate_adapter;
1838 GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail);
1839 gst_adapter_clear (wav->adapter);
1840 wav->offset += avail;
1841 wav->dataleft -= avail;
1846 if (avail < desired) {
1847 GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
1851 buf = gst_adapter_take_buffer (wav->adapter, desired);
1853 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
1854 desired, &buf)) != GST_FLOW_OK)
1857 /* we may get a short buffer at the end of the file */
1858 if (gst_buffer_get_size (buf) < desired) {
1859 gsize size = gst_buffer_get_size (buf);
1861 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
1862 if (size >= wav->blockalign) {
1863 buf = gst_buffer_make_writable (buf);
1864 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
1866 gst_buffer_unref (buf);
1872 obtained = gst_buffer_get_size (buf);
1874 /* our positions in bytes */
1875 pos = wav->offset - wav->datastart;
1876 nextpos = pos + obtained;
1878 /* update offsets, does not overflow. */
1879 buf = gst_buffer_make_writable (buf);
1880 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
1881 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
1883 /* first chunk of data? create the source pad. We do this only here so
1884 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
1885 if (G_UNLIKELY (wav->first)) {
1887 /* this will also push the segment events */
1888 gst_wavparse_add_src_pad (wav, buf);
1890 /* If we have a pending start segment, send it now. */
1891 if (G_UNLIKELY (wav->start_segment != NULL)) {
1892 gst_pad_push_event (wav->srcpad, wav->start_segment);
1893 wav->start_segment = NULL;
1898 /* and timestamps if we have a bitrate, be careful for overflows */
1899 timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps);
1901 uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps);
1902 duration = next_timestamp - timestamp;
1904 /* update current running segment position */
1905 if (G_LIKELY (next_timestamp >= wav->segment.start))
1906 wav->segment.position = next_timestamp;
1907 } else if (wav->fact) {
1909 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
1910 /* and timestamps if we have a bitrate, be careful for overflows */
1911 timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps);
1912 next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps);
1913 duration = next_timestamp - timestamp;
1915 /* no bitrate, all we know is that the first sample has timestamp 0, all
1916 * other positions and durations have unknown timestamp. */
1920 timestamp = GST_CLOCK_TIME_NONE;
1921 duration = GST_CLOCK_TIME_NONE;
1922 /* update current running segment position with byte offset */
1923 if (G_LIKELY (nextpos >= wav->segment.start))
1924 wav->segment.position = nextpos;
1926 if ((pos > 0) && wav->vbr) {
1927 /* don't set timestamps for VBR files if it's not the first buffer */
1928 timestamp = GST_CLOCK_TIME_NONE;
1929 duration = GST_CLOCK_TIME_NONE;
1932 GST_DEBUG_OBJECT (wav, "marking DISCONT");
1933 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
1934 wav->discont = FALSE;
1937 GST_BUFFER_TIMESTAMP (buf) = timestamp;
1938 GST_BUFFER_DURATION (buf) = duration;
1940 GST_LOG_OBJECT (wav,
1941 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
1942 ", size:%u", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration),
1943 gst_buffer_get_size (buf));
1945 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
1948 if (obtained < wav->dataleft) {
1949 wav->offset += obtained;
1950 wav->dataleft -= obtained;
1952 wav->offset += wav->dataleft;
1956 /* Iterate until need more data, so adapter size won't grow */
1957 if (wav->streaming) {
1958 GST_LOG_OBJECT (wav,
1959 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
1961 goto iterate_adapter;
1968 GST_DEBUG_OBJECT (wav, "found EOS");
1969 return GST_FLOW_UNEXPECTED;
1973 /* check if we got EOS */
1974 if (res == GST_FLOW_UNEXPECTED)
1977 GST_WARNING_OBJECT (wav,
1978 "Error getting %" G_GINT64_FORMAT " bytes from the "
1979 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
1984 GST_INFO_OBJECT (wav,
1985 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
1986 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
1987 gst_pad_is_linked (wav->srcpad));
1993 gst_wavparse_loop (GstPad * pad)
1996 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
1998 GST_LOG_OBJECT (wav, "process data");
2000 switch (wav->state) {
2001 case GST_WAVPARSE_START:
2002 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2003 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2006 wav->state = GST_WAVPARSE_HEADER;
2009 case GST_WAVPARSE_HEADER:
2010 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2011 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2014 wav->state = GST_WAVPARSE_DATA;
2015 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2018 case GST_WAVPARSE_DATA:
2019 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2023 g_assert_not_reached ();
2030 const gchar *reason = gst_flow_get_name (ret);
2032 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2033 gst_pad_pause_task (pad);
2035 if (ret == GST_FLOW_UNEXPECTED) {
2036 /* handle end-of-stream/segment */
2037 /* so align our position with the end of it, if there is one
2038 * this ensures a subsequent will arrive at correct base/acc time */
2039 if (wav->segment.format == GST_FORMAT_TIME) {
2040 if (wav->segment.rate > 0.0 &&
2041 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2042 wav->segment.position = wav->segment.stop;
2043 else if (wav->segment.rate < 0.0)
2044 wav->segment.position = wav->segment.start;
2046 /* add pad before we perform EOS */
2047 if (G_UNLIKELY (wav->first)) {
2049 gst_wavparse_add_src_pad (wav, NULL);
2052 if (wav->state == GST_WAVPARSE_START)
2053 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2054 ("No valid input found before end of stream"), (NULL));
2056 /* perform EOS logic */
2057 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2060 if ((stop = wav->segment.stop) == -1)
2061 stop = wav->segment.duration;
2063 gst_element_post_message (GST_ELEMENT_CAST (wav),
2064 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2065 wav->segment.format, stop));
2067 if (wav->srcpad != NULL)
2068 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2070 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_UNEXPECTED) {
2071 /* for fatal errors we post an error message, post the error
2072 * first so the app knows about the error first. */
2073 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2074 (_("Internal data flow error.")),
2075 ("streaming task paused, reason %s (%d)", reason, ret));
2076 if (wav->srcpad != NULL)
2077 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2083 static GstFlowReturn
2084 gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
2087 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2089 GST_LOG_OBJECT (wav, "adapter_push %u bytes", gst_buffer_get_size (buf));
2091 gst_adapter_push (wav->adapter, buf);
2093 switch (wav->state) {
2094 case GST_WAVPARSE_START:
2095 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2096 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2099 if (wav->state != GST_WAVPARSE_HEADER)
2102 /* otherwise fall-through */
2103 case GST_WAVPARSE_HEADER:
2104 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2105 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2108 if (!wav->got_fmt || wav->datastart == 0)
2111 wav->state = GST_WAVPARSE_DATA;
2112 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2115 case GST_WAVPARSE_DATA:
2116 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2117 wav->discont = TRUE;
2118 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2122 g_return_val_if_reached (GST_FLOW_ERROR);
2125 if (G_UNLIKELY (wav->abort_buffering)) {
2126 wav->abort_buffering = FALSE;
2127 ret = GST_FLOW_ERROR;
2128 /* sort of demux/parse error */
2129 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2135 static GstFlowReturn
2136 gst_wavparse_flush_data (GstWavParse * wav)
2138 GstFlowReturn ret = GST_FLOW_OK;
2141 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2143 wav->end_offset = wav->offset + av;
2144 ret = gst_wavparse_stream_data (wav);
2151 gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
2153 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2154 gboolean ret = TRUE;
2156 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2158 switch (GST_EVENT_TYPE (event)) {
2159 case GST_EVENT_CAPS:
2161 /* discard, we'll come up with proper src caps */
2162 gst_event_unref (event);
2165 case GST_EVENT_SEGMENT:
2167 gint64 start, stop, offset = 0, end_offset = -1;
2170 /* some debug output */
2171 gst_event_copy_segment (event, &segment);
2172 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2175 if (wav->state != GST_WAVPARSE_DATA) {
2176 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2180 /* now we are either committed to TIME or BYTE format,
2181 * and we only expect a BYTE segment, e.g. following a seek */
2182 if (segment.format == GST_FORMAT_BYTES) {
2183 /* handle (un)signed issues */
2184 start = segment.start;
2185 stop = segment.stop;
2188 start -= wav->datastart;
2189 start = MAX (start, 0);
2193 segment.stop -= wav->datastart;
2194 segment.stop = MAX (stop, 0);
2196 if (wav->segment.format == GST_FORMAT_TIME) {
2197 guint64 bps = wav->bps;
2199 /* operating in format TIME, so we can convert */
2200 if (!bps && wav->fact)
2202 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2206 uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps);
2209 uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps);
2213 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2217 segment.start = start;
2218 segment.stop = stop;
2220 /* accept upstream's notion of segment and distribute along */
2221 segment.time = segment.start = segment.position;
2222 segment.duration = wav->segment.duration;
2223 segment.base = gst_segment_to_running_time (&wav->segment,
2224 GST_FORMAT_TIME, wav->segment.position);
2226 gst_segment_copy_into (&segment, &wav->segment);
2228 /* also store the newsegment event for the streaming thread */
2229 if (wav->start_segment)
2230 gst_event_unref (wav->start_segment);
2231 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2232 wav->start_segment = gst_event_new_segment (&segment);
2234 /* stream leftover data in current segment */
2235 gst_wavparse_flush_data (wav);
2236 /* and set up streaming thread for next one */
2237 wav->offset = offset;
2238 wav->end_offset = end_offset;
2239 if (wav->end_offset > 0) {
2240 wav->dataleft = wav->end_offset - wav->offset;
2242 /* infinity; upstream will EOS when done */
2243 wav->dataleft = G_MAXUINT64;
2246 gst_event_unref (event);
2250 /* add pad if needed so EOS is seen downstream */
2251 if (G_UNLIKELY (wav->first)) {
2253 gst_wavparse_add_src_pad (wav, NULL);
2255 /* stream leftover data in current segment */
2256 gst_wavparse_flush_data (wav);
2259 if (wav->state == GST_WAVPARSE_START)
2260 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2261 ("No valid input found before end of stream"), (NULL));
2264 case GST_EVENT_FLUSH_STOP:
2268 gst_adapter_clear (wav->adapter);
2269 wav->discont = TRUE;
2270 dur = wav->segment.duration;
2271 gst_segment_init (&wav->segment, wav->segment.format);
2272 wav->segment.duration = dur;
2276 ret = gst_pad_event_default (wav->sinkpad, event);
2284 /* convert and query stuff */
2285 static const GstFormat *
2286 gst_wavparse_get_formats (GstPad * pad)
2288 static GstFormat formats[] = {
2291 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2300 gst_wavparse_pad_convert (GstPad * pad,
2301 GstFormat src_format, gint64 src_value,
2302 GstFormat * dest_format, gint64 * dest_value)
2304 GstWavParse *wavparse;
2305 gboolean res = TRUE;
2307 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2309 if (*dest_format == src_format) {
2310 *dest_value = src_value;
2314 if ((wavparse->bps == 0) && !wavparse->fact)
2317 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2318 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2320 switch (src_format) {
2321 case GST_FORMAT_BYTES:
2322 switch (*dest_format) {
2323 case GST_FORMAT_DEFAULT:
2324 *dest_value = src_value / wavparse->bytes_per_sample;
2325 /* make sure we end up on a sample boundary */
2326 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2328 case GST_FORMAT_TIME:
2329 /* src_value + datastart = offset */
2330 GST_INFO_OBJECT (wavparse,
2331 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2333 if (wavparse->bps > 0)
2334 *dest_value = uint64_ceiling_scale (src_value, GST_SECOND,
2335 (guint64) wavparse->bps);
2336 else if (wavparse->fact) {
2337 guint64 bps = uint64_ceiling_scale_int (wavparse->datasize,
2338 wavparse->rate, wavparse->fact);
2340 *dest_value = uint64_ceiling_scale_int (src_value, GST_SECOND, bps);
2351 case GST_FORMAT_DEFAULT:
2352 switch (*dest_format) {
2353 case GST_FORMAT_BYTES:
2354 *dest_value = src_value * wavparse->bytes_per_sample;
2356 case GST_FORMAT_TIME:
2357 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2358 (guint64) wavparse->rate);
2366 case GST_FORMAT_TIME:
2367 switch (*dest_format) {
2368 case GST_FORMAT_BYTES:
2369 if (wavparse->bps > 0)
2370 *dest_value = gst_util_uint64_scale (src_value,
2371 (guint64) wavparse->bps, GST_SECOND);
2373 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2374 wavparse->rate, wavparse->fact);
2376 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2378 /* make sure we end up on a sample boundary */
2379 *dest_value -= *dest_value % wavparse->blockalign;
2381 case GST_FORMAT_DEFAULT:
2382 *dest_value = gst_util_uint64_scale (src_value,
2383 (guint64) wavparse->rate, GST_SECOND);
2402 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2408 static const GstQueryType *
2409 gst_wavparse_get_query_types (GstPad * pad)
2411 static const GstQueryType types[] = {
2422 /* handle queries for location and length in requested format */
2424 gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
2426 gboolean res = TRUE;
2427 GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (pad));
2429 /* only if we know */
2430 if (wav->state != GST_WAVPARSE_DATA) {
2431 gst_object_unref (wav);
2435 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2437 switch (GST_QUERY_TYPE (query)) {
2438 case GST_QUERY_POSITION:
2444 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2445 curb = wav->offset - wav->datastart;
2446 gst_query_parse_position (query, &format, NULL);
2447 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2450 case GST_FORMAT_TIME:
2451 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2455 format = GST_FORMAT_BYTES;
2460 gst_query_set_position (query, format, cur);
2463 case GST_QUERY_DURATION:
2465 gint64 duration = 0;
2468 gst_query_parse_duration (query, &format, NULL);
2471 case GST_FORMAT_TIME:{
2472 if ((res = gst_wavparse_calculate_duration (wav))) {
2473 duration = wav->duration;
2478 format = GST_FORMAT_BYTES;
2479 duration = wav->datasize;
2482 gst_query_set_duration (query, format, duration);
2485 case GST_QUERY_CONVERT:
2487 gint64 srcvalue, dstvalue;
2488 GstFormat srcformat, dstformat;
2490 gst_query_parse_convert (query, &srcformat, &srcvalue,
2491 &dstformat, &dstvalue);
2492 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2493 &dstformat, &dstvalue);
2495 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2498 case GST_QUERY_SEEKING:{
2500 gboolean seekable = FALSE;
2502 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2503 if (fmt == wav->segment.format) {
2504 if (wav->streaming) {
2507 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2508 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2509 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2510 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2512 gst_query_unref (q);
2514 GST_LOG_OBJECT (wav, "looping => seekable");
2518 } else if (fmt == GST_FORMAT_TIME) {
2522 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2527 res = gst_pad_query_default (pad, query);
2530 gst_object_unref (wav);
2535 gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event)
2537 GstWavParse *wavparse = GST_WAVPARSE (gst_pad_get_parent (pad));
2538 gboolean res = FALSE;
2540 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2542 switch (GST_EVENT_TYPE (event)) {
2543 case GST_EVENT_SEEK:
2544 /* can only handle events when we are in the data state */
2545 if (wavparse->state == GST_WAVPARSE_DATA) {
2546 res = gst_wavparse_perform_seek (wavparse, event);
2548 gst_event_unref (event);
2551 res = gst_pad_push_event (wavparse->sinkpad, event);
2554 gst_object_unref (wavparse);
2559 gst_wavparse_sink_activate (GstPad * sinkpad)
2561 GstWavParse *wav = GST_WAVPARSE (gst_pad_get_parent (sinkpad));
2566 gst_adapter_clear (wav->adapter);
2567 g_object_unref (wav->adapter);
2568 wav->adapter = NULL;
2571 query = gst_query_new_scheduling ();
2573 if (!gst_pad_peer_query (sinkpad, query)) {
2574 gst_query_unref (query);
2578 gst_query_parse_scheduling (query, &pull_mode, NULL, NULL, NULL, NULL, NULL);
2579 gst_query_unref (query);
2584 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2585 wav->streaming = FALSE;
2586 gst_object_unref (wav);
2587 return gst_pad_activate_pull (sinkpad, TRUE);
2591 GST_DEBUG_OBJECT (sinkpad, "activating push");
2592 wav->streaming = TRUE;
2593 wav->adapter = gst_adapter_new ();
2594 gst_object_unref (wav);
2595 return gst_pad_activate_push (sinkpad, TRUE);
2601 gst_wavparse_sink_activate_pull (GstPad * sinkpad, gboolean active)
2604 /* if we have a scheduler we can start the task */
2605 return gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2608 return gst_pad_stop_task (sinkpad);
2612 static GstStateChangeReturn
2613 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2615 GstStateChangeReturn ret;
2616 GstWavParse *wav = GST_WAVPARSE (element);
2618 switch (transition) {
2619 case GST_STATE_CHANGE_NULL_TO_READY:
2621 case GST_STATE_CHANGE_READY_TO_PAUSED:
2622 gst_wavparse_reset (wav);
2624 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2630 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2632 switch (transition) {
2633 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2635 case GST_STATE_CHANGE_PAUSED_TO_READY:
2636 gst_wavparse_destroy_sourcepad (wav);
2637 gst_wavparse_reset (wav);
2639 case GST_STATE_CHANGE_READY_TO_NULL:
2648 plugin_init (GstPlugin * plugin)
2652 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
2656 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
2659 "Parse a .wav file into raw audio",
2660 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)