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
56 #include "gstwavparse.h"
57 #include "gst/riff/riff-ids.h"
58 #include "gst/riff/riff-media.h"
59 #include <gst/base/gsttypefindhelper.h>
60 #include <gst/gst-i18n-plugin.h>
62 GST_DEBUG_CATEGORY_STATIC (wavparse_debug);
63 #define GST_CAT_DEFAULT (wavparse_debug)
65 static void gst_wavparse_dispose (GObject * object);
67 static gboolean gst_wavparse_sink_activate (GstPad * sinkpad,
69 static gboolean gst_wavparse_sink_activate_mode (GstPad * sinkpad,
70 GstObject * parent, GstPadMode mode, gboolean active);
71 static gboolean gst_wavparse_send_event (GstElement * element,
73 static GstStateChangeReturn gst_wavparse_change_state (GstElement * element,
74 GstStateChange transition);
76 static gboolean gst_wavparse_pad_query (GstPad * pad, GstObject * parent,
78 static gboolean gst_wavparse_pad_convert (GstPad * pad, GstFormat src_format,
79 gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
81 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstObject * parent,
83 static gboolean gst_wavparse_sink_event (GstPad * pad, GstObject * parent,
85 static void gst_wavparse_loop (GstPad * pad);
86 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent,
89 static void gst_wavparse_set_property (GObject * object, guint prop_id,
90 const GValue * value, GParamSpec * pspec);
91 static void gst_wavparse_get_property (GObject * object, guint prop_id,
92 GValue * value, GParamSpec * pspec);
94 #define DEFAULT_IGNORE_LENGTH FALSE
102 static GstStaticPadTemplate sink_template_factory =
103 GST_STATIC_PAD_TEMPLATE ("sink",
106 GST_STATIC_CAPS ("audio/x-wav")
110 GST_DEBUG_CATEGORY_INIT (wavparse_debug, "wavparse", 0, "WAV parser");
112 #define gst_wavparse_parent_class parent_class
113 G_DEFINE_TYPE_WITH_CODE (GstWavParse, gst_wavparse, GST_TYPE_ELEMENT,
118 /* Offset Size Description Value
119 * 0x00 4 ID unique identification value
120 * 0x04 4 Position play order position
121 * 0x08 4 Data Chunk ID RIFF ID of corresponding data chunk
122 * 0x0c 4 Chunk Start Byte Offset of Data Chunk *
123 * 0x10 4 Block Start Byte Offset to sample of First Channel
124 * 0x14 4 Sample Offset Byte Offset to sample byte of First Channel
128 guint32 data_chunk_id;
131 guint32 sample_offset;
136 /* Offset Size Description Value
137 * 0x08 4 Cue Point ID 0 - 0xFFFFFFFF
140 guint32 cue_point_id;
145 gst_wavparse_class_init (GstWavParseClass * klass)
147 GstElementClass *gstelement_class;
148 GObjectClass *object_class;
149 GstPadTemplate *src_template;
151 gstelement_class = (GstElementClass *) klass;
152 object_class = (GObjectClass *) klass;
154 parent_class = g_type_class_peek_parent (klass);
156 object_class->dispose = gst_wavparse_dispose;
158 object_class->set_property = gst_wavparse_set_property;
159 object_class->get_property = gst_wavparse_get_property;
162 * GstWavParse:ignore-length
164 * This selects whether the length found in a data chunk
165 * should be ignored. This may be useful for streamed audio
166 * where the length is unknown until the end of streaming,
167 * and various software/hardware just puts some random value
168 * in there and hopes it doesn't break too much.
172 g_object_class_install_property (object_class, PROP_IGNORE_LENGTH,
173 g_param_spec_boolean ("ignore-length",
175 "Ignore length from the Wave header",
176 DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
179 gstelement_class->change_state = gst_wavparse_change_state;
180 gstelement_class->send_event = gst_wavparse_send_event;
183 gst_element_class_add_pad_template (gstelement_class,
184 gst_static_pad_template_get (&sink_template_factory));
186 src_template = gst_pad_template_new ("src", GST_PAD_SRC,
187 GST_PAD_ALWAYS, gst_riff_create_audio_template_caps ());
188 gst_element_class_add_pad_template (gstelement_class, src_template);
190 gst_element_class_set_static_metadata (gstelement_class, "WAV audio demuxer",
191 "Codec/Demuxer/Audio",
192 "Parse a .wav file into raw audio",
193 "Erik Walthinsen <omega@cse.ogi.edu>");
197 gst_wavparse_reset (GstWavParse * wav)
199 wav->state = GST_WAVPARSE_START;
201 /* These will all be set correctly in the fmt chunk */
215 wav->got_fmt = FALSE;
219 gst_event_unref (wav->seek_event);
220 wav->seek_event = NULL;
222 gst_adapter_clear (wav->adapter);
223 g_object_unref (wav->adapter);
227 gst_tag_list_free (wav->tags);
230 gst_toc_unref (wav->toc);
233 g_list_free_full (wav->cues, g_free);
236 g_list_free_full (wav->labls, g_free);
239 gst_caps_unref (wav->caps);
241 if (wav->start_segment)
242 gst_event_unref (wav->start_segment);
243 wav->start_segment = NULL;
247 gst_wavparse_dispose (GObject * object)
249 GstWavParse *wav = GST_WAVPARSE (object);
251 GST_DEBUG_OBJECT (wav, "WAV: Dispose");
252 gst_wavparse_reset (wav);
254 G_OBJECT_CLASS (parent_class)->dispose (object);
258 gst_wavparse_init (GstWavParse * wavparse)
260 gst_wavparse_reset (wavparse);
264 gst_pad_new_from_static_template (&sink_template_factory, "sink");
265 gst_pad_set_activate_function (wavparse->sinkpad,
266 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate));
267 gst_pad_set_activatemode_function (wavparse->sinkpad,
268 GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_mode));
269 gst_pad_set_chain_function (wavparse->sinkpad,
270 GST_DEBUG_FUNCPTR (gst_wavparse_chain));
271 gst_pad_set_event_function (wavparse->sinkpad,
272 GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
273 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
277 gst_pad_new_from_template (gst_element_class_get_pad_template
278 (GST_ELEMENT_GET_CLASS (wavparse), "src"), "src");
279 gst_pad_use_fixed_caps (wavparse->srcpad);
280 gst_pad_set_query_function (wavparse->srcpad,
281 GST_DEBUG_FUNCPTR (gst_wavparse_pad_query));
282 gst_pad_set_event_function (wavparse->srcpad,
283 GST_DEBUG_FUNCPTR (gst_wavparse_srcpad_event));
284 gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->srcpad);
287 /* FIXME: why is that not in use? */
290 gst_wavparse_parse_adtl (GstWavParse * wavparse, int len)
293 GstByteStream *bs = wavparse->bs;
294 gst_riff_chunk *temp_chunk, chunk;
296 struct _gst_riff_labl labl, *temp_labl;
297 struct _gst_riff_ltxt ltxt, *temp_ltxt;
298 struct _gst_riff_note note, *temp_note;
301 GstPropsEntry *entry;
305 props = wavparse->metadata->properties;
309 gst_bytestream_peek_bytes (bs, &tempdata, sizeof (gst_riff_chunk));
310 if (got_bytes != sizeof (gst_riff_chunk)) {
313 temp_chunk = (gst_riff_chunk *) tempdata;
315 chunk.id = GUINT32_FROM_LE (temp_chunk->id);
316 chunk.size = GUINT32_FROM_LE (temp_chunk->size);
318 if (chunk.size == 0) {
319 gst_bytestream_flush (bs, sizeof (gst_riff_chunk));
320 len -= sizeof (gst_riff_chunk);
325 case GST_RIFF_adtl_labl:
327 gst_bytestream_peek_bytes (bs, &tempdata,
328 sizeof (struct _gst_riff_labl));
329 if (got_bytes != sizeof (struct _gst_riff_labl)) {
333 temp_labl = (struct _gst_riff_labl *) tempdata;
334 labl.id = GUINT32_FROM_LE (temp_labl->id);
335 labl.size = GUINT32_FROM_LE (temp_labl->size);
336 labl.identifier = GUINT32_FROM_LE (temp_labl->identifier);
338 gst_bytestream_flush (bs, sizeof (struct _gst_riff_labl));
339 len -= sizeof (struct _gst_riff_labl);
341 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, labl.size - 4);
342 if (got_bytes != labl.size - 4) {
346 label_name = (char *) tempdata;
348 gst_bytestream_flush (bs, ((labl.size - 4) + 1) & ~1);
349 len -= (((labl.size - 4) + 1) & ~1);
351 new_caps = gst_caps_new ("label",
352 "application/x-gst-metadata",
353 gst_props_new ("identifier", G_TYPE_INT (labl.identifier),
354 "name", G_TYPE_STRING (label_name), NULL));
356 if (gst_props_get (props, "labels", &caps, NULL)) {
357 caps = g_list_append (caps, new_caps);
359 caps = g_list_append (NULL, new_caps);
361 entry = gst_props_entry_new ("labels", GST_PROPS_GLIST (caps));
362 gst_props_add_entry (props, entry);
367 case GST_RIFF_adtl_ltxt:
369 gst_bytestream_peek_bytes (bs, &tempdata,
370 sizeof (struct _gst_riff_ltxt));
371 if (got_bytes != sizeof (struct _gst_riff_ltxt)) {
375 temp_ltxt = (struct _gst_riff_ltxt *) tempdata;
376 ltxt.id = GUINT32_FROM_LE (temp_ltxt->id);
377 ltxt.size = GUINT32_FROM_LE (temp_ltxt->size);
378 ltxt.identifier = GUINT32_FROM_LE (temp_ltxt->identifier);
379 ltxt.length = GUINT32_FROM_LE (temp_ltxt->length);
380 ltxt.purpose = GUINT32_FROM_LE (temp_ltxt->purpose);
381 ltxt.country = GUINT16_FROM_LE (temp_ltxt->country);
382 ltxt.language = GUINT16_FROM_LE (temp_ltxt->language);
383 ltxt.dialect = GUINT16_FROM_LE (temp_ltxt->dialect);
384 ltxt.codepage = GUINT16_FROM_LE (temp_ltxt->codepage);
386 gst_bytestream_flush (bs, sizeof (struct _gst_riff_ltxt));
387 len -= sizeof (struct _gst_riff_ltxt);
389 if (ltxt.size - 20 > 0) {
390 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, ltxt.size - 20);
391 if (got_bytes != ltxt.size - 20) {
395 gst_bytestream_flush (bs, ((ltxt.size - 20) + 1) & ~1);
396 len -= (((ltxt.size - 20) + 1) & ~1);
398 label_name = (char *) tempdata;
403 new_caps = gst_caps_new ("ltxt",
404 "application/x-gst-metadata",
405 gst_props_new ("identifier", G_TYPE_INT (ltxt.identifier),
406 "name", G_TYPE_STRING (label_name),
407 "length", G_TYPE_INT (ltxt.length), NULL));
409 if (gst_props_get (props, "ltxts", &caps, NULL)) {
410 caps = g_list_append (caps, new_caps);
412 caps = g_list_append (NULL, new_caps);
414 entry = gst_props_entry_new ("ltxts", GST_PROPS_GLIST (caps));
415 gst_props_add_entry (props, entry);
420 case GST_RIFF_adtl_note:
422 gst_bytestream_peek_bytes (bs, &tempdata,
423 sizeof (struct _gst_riff_note));
424 if (got_bytes != sizeof (struct _gst_riff_note)) {
428 temp_note = (struct _gst_riff_note *) tempdata;
429 note.id = GUINT32_FROM_LE (temp_note->id);
430 note.size = GUINT32_FROM_LE (temp_note->size);
431 note.identifier = GUINT32_FROM_LE (temp_note->identifier);
433 gst_bytestream_flush (bs, sizeof (struct _gst_riff_note));
434 len -= sizeof (struct _gst_riff_note);
436 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, note.size - 4);
437 if (got_bytes != note.size - 4) {
441 gst_bytestream_flush (bs, ((note.size - 4) + 1) & ~1);
442 len -= (((note.size - 4) + 1) & ~1);
444 label_name = (char *) tempdata;
446 new_caps = gst_caps_new ("note",
447 "application/x-gst-metadata",
448 gst_props_new ("identifier", G_TYPE_INT (note.identifier),
449 "name", G_TYPE_STRING (label_name), NULL));
451 if (gst_props_get (props, "notes", &caps, NULL)) {
452 caps = g_list_append (caps, new_caps);
454 caps = g_list_append (NULL, new_caps);
456 entry = gst_props_entry_new ("notes", GST_PROPS_GLIST (caps));
457 gst_props_add_entry (props, entry);
463 g_print ("Unknown chunk: %" GST_FOURCC_FORMAT "\n",
464 GST_FOURCC_ARGS (chunk.id));
469 g_object_notify (G_OBJECT (wavparse), "metadata");
473 gst_wavparse_parse_cues (GstWavParse * wavparse, int len)
476 GstByteStream *bs = wavparse->bs;
477 struct _gst_riff_cue *temp_cue, cue;
478 struct _gst_riff_cuepoints *points;
482 GstPropsEntry *entry;
488 gst_bytestream_peek_bytes (bs, &tempdata,
489 sizeof (struct _gst_riff_cue));
490 temp_cue = (struct _gst_riff_cue *) tempdata;
492 /* fixup for our big endian friends */
493 cue.id = GUINT32_FROM_LE (temp_cue->id);
494 cue.size = GUINT32_FROM_LE (temp_cue->size);
495 cue.cuepoints = GUINT32_FROM_LE (temp_cue->cuepoints);
497 gst_bytestream_flush (bs, sizeof (struct _gst_riff_cue));
498 if (got_bytes != sizeof (struct _gst_riff_cue)) {
502 len -= sizeof (struct _gst_riff_cue);
504 /* -4 because cue.size contains the cuepoints size
505 and we've already flushed that out of the system */
506 required = cue.size - 4;
507 got_bytes = gst_bytestream_peek_bytes (bs, &tempdata, required);
508 gst_bytestream_flush (bs, ((required) + 1) & ~1);
509 if (got_bytes != required) {
513 len -= (((cue.size - 4) + 1) & ~1);
515 /* now we have an array of struct _gst_riff_cuepoints in tempdata */
516 points = (struct _gst_riff_cuepoints *) tempdata;
518 for (i = 0; i < cue.cuepoints; i++) {
521 caps = gst_caps_new ("cues",
522 "application/x-gst-metadata",
523 gst_props_new ("identifier", G_TYPE_INT (points[i].identifier),
524 "position", G_TYPE_INT (points[i].offset), NULL));
525 cues = g_list_append (cues, caps);
528 entry = gst_props_entry_new ("cues", GST_PROPS_GLIST (cues));
529 gst_props_add_entry (wavparse->metadata->properties, entry);
532 g_object_notify (G_OBJECT (wavparse), "metadata");
535 /* Read 'fmt ' header */
537 gst_wavparse_fmt (GstWavParse * wav)
539 gst_riff_strf_auds *header = NULL;
542 if (!gst_riff_read_strf_auds (wav, &header))
545 wav->format = header->format;
546 wav->rate = header->rate;
547 wav->channels = header->channels;
548 if (wav->channels == 0)
551 wav->blockalign = header->blockalign;
552 wav->width = (header->blockalign * 8) / header->channels;
553 wav->depth = header->size;
554 wav->bps = header->av_bps;
558 /* Note: gst_riff_create_audio_caps might need to fix values in
559 * the header header depending on the format, so call it first */
560 /* FIXME: Need to handle the channel reorder map */
561 caps = gst_riff_create_audio_caps (header->format, NULL, header, NULL, NULL);
567 gst_wavparse_create_sourcepad (wav);
568 gst_pad_use_fixed_caps (wav->srcpad);
569 gst_pad_set_active (wav->srcpad, TRUE);
570 gst_pad_set_caps (wav->srcpad, caps);
571 gst_caps_free (caps);
572 gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad);
573 gst_element_no_more_pads (GST_ELEMENT_CAST (wav));
575 GST_DEBUG ("frequency %u, channels %u", wav->rate, wav->channels);
582 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
583 ("No FMT tag found"));
588 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
589 ("Stream claims to contain zero channels - invalid data"));
595 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
596 ("Stream claims to bitrate of <= zero - invalid data"));
602 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), (NULL));
608 gst_wavparse_other (GstWavParse * wav)
612 if (!gst_riff_peek_head (wav, &tag, &length, NULL)) {
613 GST_WARNING_OBJECT (wav, "could not peek head");
616 GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %u", tag,
617 (const gchar *) &tag, length);
620 case GST_RIFF_TAG_LIST:
621 if (!(tag = gst_riff_peek_list (wav))) {
622 GST_WARNING_OBJECT (wav, "could not peek list");
627 case GST_RIFF_LIST_INFO:
628 if (!gst_riff_read_list (wav, &tag) || !gst_riff_read_info (wav)) {
629 GST_WARNING_OBJECT (wav, "could not read list");
634 case GST_RIFF_LIST_adtl:
635 if (!gst_riff_read_skip (wav)) {
636 GST_WARNING_OBJECT (wav, "could not read skip");
642 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag,
644 if (!gst_riff_read_skip (wav)) {
645 GST_WARNING_OBJECT (wav, "could not read skip");
653 case GST_RIFF_TAG_data:
654 if (!gst_bytestream_flush (wav->bs, 8)) {
655 GST_WARNING_OBJECT (wav, "could not flush 8 bytes");
659 GST_DEBUG_OBJECT (wav, "switching to data mode");
660 wav->state = GST_WAVPARSE_DATA;
661 wav->datastart = gst_bytestream_tell (wav->bs);
665 /* length is 0, data probably stretches to the end
667 GST_DEBUG_OBJECT (wav, "length is 0 trying to find length");
668 /* get length of file */
669 file_length = gst_bytestream_length (wav->bs);
670 if (file_length == -1) {
671 GST_DEBUG_OBJECT (wav,
672 "could not get file length, assuming data to eof");
673 /* could not get length, assuming till eof */
674 length = G_MAXUINT32;
676 if (file_length > G_MAXUINT32) {
677 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
678 ", clipping to 32 bits", file_length);
679 /* could not get length, assuming till eof */
680 length = G_MAXUINT32;
682 GST_DEBUG_OBJECT (wav, "file length %" G_GUINT64_FORMAT
683 ", datalength %u", file_length, length);
684 /* substract offset of datastart from length */
685 length = file_length - wav->datastart;
686 GST_DEBUG_OBJECT (wav, "datalength %u", length);
689 wav->datasize = (guint64) length;
690 GST_DEBUG_OBJECT (wav, "datasize = %ld", length)
693 case GST_RIFF_TAG_cue:
694 if (!gst_riff_read_skip (wav)) {
695 GST_WARNING_OBJECT (wav, "could not read skip");
701 GST_DEBUG_OBJECT (wav, "skipping tag (%08x) %4.4s", tag, (gchar *) & tag);
702 if (!gst_riff_read_skip (wav))
713 gst_wavparse_parse_file_header (GstElement * element, GstBuffer * buf)
717 if (!gst_riff_parse_file_header (element, buf, &doctype))
720 if (doctype != GST_RIFF_RIFF_WAVE)
728 GST_ELEMENT_ERROR (element, STREAM, WRONG_TYPE, (NULL),
729 ("File is not a WAVE file: %" GST_FOURCC_FORMAT,
730 GST_FOURCC_ARGS (doctype)));
736 gst_wavparse_stream_init (GstWavParse * wav)
739 GstBuffer *buf = NULL;
741 if ((res = gst_pad_pull_range (wav->sinkpad,
742 wav->offset, 12, &buf)) != GST_FLOW_OK)
744 else if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), buf))
745 return GST_FLOW_ERROR;
753 gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
755 /* -1 always maps to -1 */
761 /* 0 always maps to 0 */
768 *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND);
770 } else if (wav->fact) {
772 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
773 *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND);
780 /* This function is used to perform seeks on the element.
782 * It also works when event is NULL, in which case it will just
783 * start from the last configured segment. This technique is
784 * used when activating the element and to perform the seek in
788 gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
792 GstFormat format, bformat;
794 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
795 gint64 cur, stop, upstream_size;
798 GstSegment seeksegment = { 0, };
802 GST_DEBUG_OBJECT (wav, "doing seek with event");
804 gst_event_parse_seek (event, &rate, &format, &flags,
805 &cur_type, &cur, &stop_type, &stop);
807 /* no negative rates yet */
811 if (format != wav->segment.format) {
812 GST_INFO_OBJECT (wav, "converting seek-event from %s to %s",
813 gst_format_get_name (format),
814 gst_format_get_name (wav->segment.format));
816 if (cur_type != GST_SEEK_TYPE_NONE)
818 gst_pad_query_convert (wav->srcpad, format, cur,
819 wav->segment.format, &cur);
820 if (res && stop_type != GST_SEEK_TYPE_NONE)
822 gst_pad_query_convert (wav->srcpad, format, stop,
823 wav->segment.format, &stop);
827 format = wav->segment.format;
830 GST_DEBUG_OBJECT (wav, "doing seek without event");
833 cur_type = GST_SEEK_TYPE_SET;
834 stop_type = GST_SEEK_TYPE_SET;
837 /* in push mode, we must delegate to upstream */
838 if (wav->streaming) {
839 gboolean res = FALSE;
841 /* if streaming not yet started; only prepare initial newsegment */
842 if (!event || wav->state != GST_WAVPARSE_DATA) {
843 if (wav->start_segment)
844 gst_event_unref (wav->start_segment);
845 wav->start_segment = gst_event_new_segment (&wav->segment);
848 /* convert seek positions to byte positions in data sections */
849 if (format == GST_FORMAT_TIME) {
850 /* should not fail */
851 if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
853 if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
856 /* mind sample boundary and header */
858 cur -= (cur % wav->bytes_per_sample);
859 cur += wav->datastart;
862 stop -= (stop % wav->bytes_per_sample);
863 stop += wav->datastart;
865 GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
866 "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
868 /* BYTE seek event */
869 event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
871 res = gst_pad_push_event (wav->sinkpad, event);
877 flush = flags & GST_SEEK_FLAG_FLUSH;
879 /* now we need to make sure the streaming thread is stopped. We do this by
880 * either sending a FLUSH_START event downstream which will cause the
881 * streaming thread to stop with a WRONG_STATE.
882 * For a non-flushing seek we simply pause the task, which will happen as soon
883 * as it completes one iteration (and thus might block when the sink is
884 * blocking in preroll). */
886 GST_DEBUG_OBJECT (wav, "sending flush start");
887 gst_pad_push_event (wav->srcpad, gst_event_new_flush_start ());
889 gst_pad_pause_task (wav->sinkpad);
892 /* we should now be able to grab the streaming thread because we stopped it
893 * with the above flush/pause code */
894 GST_PAD_STREAM_LOCK (wav->sinkpad);
896 /* save current position */
897 last_stop = wav->segment.position;
899 GST_DEBUG_OBJECT (wav, "stopped streaming at %" G_GINT64_FORMAT, last_stop);
901 /* copy segment, we need this because we still need the old
902 * segment when we close the current segment. */
903 memcpy (&seeksegment, &wav->segment, sizeof (GstSegment));
905 /* configure the seek parameters in the seeksegment. We will then have the
906 * right values in the segment to perform the seek */
908 GST_DEBUG_OBJECT (wav, "configuring seek");
909 gst_segment_do_seek (&seeksegment, rate, format, flags,
910 cur_type, cur, stop_type, stop, &update);
913 /* figure out the last position we need to play. If it's configured (stop !=
914 * -1), use that, else we play until the total duration of the file */
915 if ((stop = seeksegment.stop) == -1)
916 stop = seeksegment.duration;
918 GST_DEBUG_OBJECT (wav, "cur_type =%d", cur_type);
919 if ((cur_type != GST_SEEK_TYPE_NONE)) {
920 /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
921 * we can just copy the last_stop. If not, we use the bps to convert TIME to
923 if (!gst_wavparse_time_to_bytepos (wav, seeksegment.position,
924 (gint64 *) & wav->offset))
925 wav->offset = seeksegment.position;
926 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
927 wav->offset -= (wav->offset % wav->bytes_per_sample);
928 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
929 wav->offset += wav->datastart;
930 GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
932 GST_LOG_OBJECT (wav, "continue from offset=%" G_GUINT64_FORMAT,
936 if (stop_type != GST_SEEK_TYPE_NONE) {
937 if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
938 wav->end_offset = stop;
939 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
940 wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
941 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
942 wav->end_offset += wav->datastart;
943 GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
945 GST_LOG_OBJECT (wav, "continue to end_offset=%" G_GUINT64_FORMAT,
949 /* make sure filesize is not exceeded due to rounding errors or so,
950 * same precaution as in _stream_headers */
951 bformat = GST_FORMAT_BYTES;
952 if (gst_pad_peer_query_duration (wav->sinkpad, bformat, &upstream_size))
953 wav->end_offset = MIN (wav->end_offset, upstream_size);
955 /* this is the range of bytes we will use for playback */
956 wav->offset = MIN (wav->offset, wav->end_offset);
957 wav->dataleft = wav->end_offset - wav->offset;
959 GST_DEBUG_OBJECT (wav,
960 "seek: rate %lf, offset %" G_GUINT64_FORMAT ", end %" G_GUINT64_FORMAT
961 ", segment %" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT, rate, wav->offset,
962 wav->end_offset, GST_TIME_ARGS (seeksegment.start), GST_TIME_ARGS (stop));
964 /* prepare for streaming again */
966 /* if we sent a FLUSH_START, we now send a FLUSH_STOP */
967 GST_DEBUG_OBJECT (wav, "sending flush stop");
968 gst_pad_push_event (wav->srcpad, gst_event_new_flush_stop (TRUE));
971 /* now we did the seek and can activate the new segment values */
972 memcpy (&wav->segment, &seeksegment, sizeof (GstSegment));
974 /* if we're doing a segment seek, post a SEGMENT_START message */
975 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
976 gst_element_post_message (GST_ELEMENT_CAST (wav),
977 gst_message_new_segment_start (GST_OBJECT_CAST (wav),
978 wav->segment.format, wav->segment.position));
981 /* now create the newsegment */
982 GST_DEBUG_OBJECT (wav, "Creating newsegment from %" G_GINT64_FORMAT
983 " to %" G_GINT64_FORMAT, wav->segment.position, stop);
985 /* store the newsegment event so it can be sent from the streaming thread. */
986 if (wav->start_segment)
987 gst_event_unref (wav->start_segment);
988 wav->start_segment = gst_event_new_segment (&wav->segment);
990 /* mark discont if we are going to stream from another position. */
991 if (last_stop != wav->segment.position) {
992 GST_DEBUG_OBJECT (wav, "mark DISCONT, we did a seek to another position");
996 /* and start the streaming task again */
997 if (!wav->streaming) {
998 gst_pad_start_task (wav->sinkpad, (GstTaskFunction) gst_wavparse_loop,
1002 GST_PAD_STREAM_UNLOCK (wav->sinkpad);
1009 GST_DEBUG_OBJECT (wav, "negative playback rates are not supported yet.");
1014 GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
1019 GST_DEBUG_OBJECT (wav,
1020 "Could not determine byte position for desired time");
1026 * gst_wavparse_peek_chunk_info:
1027 * @wav Wavparse object
1028 * @tag holder for tag
1029 * @size holder for tag size
1031 * Peek next chunk info (tag and size)
1033 * Returns: %TRUE when the chunk info (header) is available
1036 gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size)
1038 const guint8 *data = NULL;
1040 if (gst_adapter_available (wav->adapter) < 8)
1043 data = gst_adapter_map (wav->adapter, 8);
1044 *tag = GST_READ_UINT32_LE (data);
1045 *size = GST_READ_UINT32_LE (data + 4);
1046 gst_adapter_unmap (wav->adapter);
1048 GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size,
1049 GST_FOURCC_ARGS (*tag));
1055 * gst_wavparse_peek_chunk:
1056 * @wav Wavparse object
1057 * @tag holder for tag
1058 * @size holder for tag size
1060 * Peek enough data for one full chunk
1062 * Returns: %TRUE when the full chunk is available
1065 gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size)
1067 guint32 peek_size = 0;
1070 if (!gst_wavparse_peek_chunk_info (wav, tag, size))
1073 /* size 0 -> empty data buffer would surprise most callers,
1074 * large size -> do not bother trying to squeeze that into adapter,
1075 * so we throw poor man's exception, which can be caught if caller really
1076 * wants to handle 0 size chunk */
1077 if (!(*size) || (*size) >= (1 << 30)) {
1078 GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT,
1079 *size, GST_FOURCC_ARGS (*tag));
1080 /* chain should give up */
1081 wav->abort_buffering = TRUE;
1084 peek_size = (*size + 1) & ~1;
1085 available = gst_adapter_available (wav->adapter);
1087 if (available >= (8 + peek_size)) {
1090 GST_LOG ("but only %u bytes available now", available);
1096 * gst_wavparse_calculate_duration:
1097 * @wav: wavparse object
1099 * Calculate duration on demand and store in @wav. Prefer bps, but use fact as a
1102 * Returns: %TRUE if duration is available.
1105 gst_wavparse_calculate_duration (GstWavParse * wav)
1107 if (wav->duration > 0)
1111 GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize);
1113 gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND,
1114 (guint64) wav->bps);
1115 GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT,
1116 GST_TIME_ARGS (wav->duration));
1118 } else if (wav->fact) {
1120 gst_util_uint64_scale_int_ceil (GST_SECOND, wav->fact, wav->rate);
1121 GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT,
1122 GST_TIME_ARGS (wav->duration));
1129 gst_waveparse_ignore_chunk (GstWavParse * wav, GstBuffer * buf, guint32 tag,
1134 if (wav->streaming) {
1135 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1138 GST_DEBUG_OBJECT (wav, "Ignoring tag %" GST_FOURCC_FORMAT,
1139 GST_FOURCC_ARGS (tag));
1140 flush = 8 + ((size + 1) & ~1);
1141 wav->offset += flush;
1142 if (wav->streaming) {
1143 gst_adapter_flush (wav->adapter, flush);
1145 gst_buffer_unref (buf);
1152 * gst_wavparse_cue_chunk:
1153 * @wav GstWavParse object
1154 * @data holder for data
1155 * @size holder for data size
1157 * Parse cue chunk from @data to wav->cues.
1159 * Returns: %TRUE when cue chunk is available
1162 gst_wavparse_cue_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
1166 GstWavParseCue *cue;
1169 GST_WARNING_OBJECT (wav, "found another cue's");
1173 ncues = GST_READ_UINT32_LE (data);
1175 if (size < 4 + ncues * 24) {
1176 GST_WARNING_OBJECT (wav, "broken file %d %d", size, ncues);
1182 for (i = 0; i < ncues; i++) {
1183 cue = g_new0 (GstWavParseCue, 1);
1184 cue->id = GST_READ_UINT32_LE (data);
1185 cue->position = GST_READ_UINT32_LE (data + 4);
1186 cue->data_chunk_id = GST_READ_UINT32_LE (data + 8);
1187 cue->chunk_start = GST_READ_UINT32_LE (data + 12);
1188 cue->block_start = GST_READ_UINT32_LE (data + 16);
1189 cue->sample_offset = GST_READ_UINT32_LE (data + 20);
1190 cues = g_list_append (cues, cue);
1200 * gst_wavparse_labl_chunk:
1201 * @wav GstWavParse object
1202 * @data holder for data
1203 * @size holder for data size
1205 * Parse labl from @data to wav->labls.
1207 * Returns: %TRUE when labl chunk is available
1210 gst_wavparse_labl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
1212 GstWavParseLabl *labl;
1217 labl = g_new0 (GstWavParseLabl, 1);
1221 labl->cue_point_id = GST_READ_UINT32_LE (data);
1222 labl->text = (gchar *) g_new0 (gchar *, size - 4 + 1);
1223 memcpy (labl->text, data + 4, size - 4);
1225 wav->labls = g_list_append (wav->labls, labl);
1231 * gst_wavparse_adtl_chunk:
1232 * @wav GstWavParse object
1233 * @data holder for data
1234 * @size holder for data size
1236 * Parse adtl from @data.
1238 * Returns: %TRUE when adtl chunk is available
1241 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
1243 guint32 ltag, lsize, offset = 0;
1246 ltag = GST_READ_UINT32_LE (data + offset);
1247 lsize = GST_READ_UINT32_LE (data + offset + 4);
1249 case GST_RIFF_TAG_labl:
1250 gst_wavparse_labl_chunk (wav, data + offset, size);
1254 offset += 8 + GST_ROUND_UP_2 (lsize);
1255 size -= 8 + GST_ROUND_UP_2 (lsize);
1262 * gst_wavparse_create_toc:
1263 * @wav GstWavParse object
1265 * Create TOC from wav->cues and wav->labls.
1268 gst_wavparse_create_toc (GstWavParse * wav)
1273 GstWavParseCue *cue;
1274 GstWavParseLabl *labl;
1277 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
1279 GST_OBJECT_LOCK (wav);
1281 GST_OBJECT_UNLOCK (wav);
1282 GST_WARNING_OBJECT (wav, "found another TOC");
1286 toc = gst_toc_new ();
1288 /* add cue edition */
1289 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
1290 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
1291 gst_toc_append_entry (toc, entry);
1293 /* add chapters in cue edition */
1294 list = g_list_first (wav->cues);
1295 while (list != NULL) {
1297 prev_subentry = cur_subentry;
1298 /* previous chapter stop time = current chapter start time */
1299 if (prev_subentry != NULL) {
1300 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1301 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1302 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1304 id = g_strdup_printf ("%08x", cue->id);
1305 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_CHAPTER, id);
1307 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1308 stop = wav->duration;
1309 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1310 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1311 list = g_list_next (list);
1314 /* add tags in chapters */
1315 list = g_list_first (wav->labls);
1316 while (list != NULL) {
1318 id = g_strdup_printf ("%08x", labl->cue_point_id);
1319 cur_subentry = gst_toc_find_entry (toc, id);
1321 if (cur_subentry != NULL) {
1322 tags = gst_tag_list_new_empty ();
1323 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1325 gst_toc_entry_set_tags (cur_subentry, tags);
1327 list = g_list_next (list);
1330 /* send data as TOC */
1333 /* send TOC event */
1335 GST_OBJECT_UNLOCK (wav);
1336 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1342 #define MAX_BUFFER_SIZE 4096
1344 static GstFlowReturn
1345 gst_wavparse_stream_headers (GstWavParse * wav)
1347 GstFlowReturn res = GST_FLOW_OK;
1348 GstBuffer *buf = NULL;
1349 gst_riff_strf_auds *header = NULL;
1351 gboolean gotdata = FALSE;
1352 GstCaps *caps = NULL;
1353 gchar *codec_name = NULL;
1355 gint64 upstream_size = 0;
1357 /* search for "_fmt" chunk, which should be first */
1358 while (!wav->got_fmt) {
1361 /* The header starts with a 'fmt ' tag */
1362 if (wav->streaming) {
1363 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1366 gst_adapter_flush (wav->adapter, 8);
1370 buf = gst_adapter_take_buffer (wav->adapter, size);
1372 gst_adapter_flush (wav->adapter, 1);
1373 wav->offset += GST_ROUND_UP_2 (size);
1375 buf = gst_buffer_new ();
1378 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1379 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1383 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1384 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1385 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1386 tag == GST_RIFF_TAG_IDVX) {
1387 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1388 GST_FOURCC_ARGS (tag));
1389 gst_buffer_unref (buf);
1394 if (tag != GST_RIFF_TAG_fmt)
1397 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1399 goto parse_header_error;
1401 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1403 /* do sanity checks of header fields */
1404 if (header->channels == 0)
1406 if (header->rate == 0)
1409 GST_DEBUG_OBJECT (wav, "creating the caps");
1411 /* Note: gst_riff_create_audio_caps might need to fix values in
1412 * the header header depending on the format, so call it first */
1413 /* FIXME: Need to handle the channel reorder map */
1414 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1415 NULL, &codec_name, NULL);
1418 gst_buffer_unref (extra);
1421 goto unknown_format;
1423 /* do more sanity checks of header fields
1424 * (these can be sanitized by gst_riff_create_audio_caps()
1426 wav->format = header->format;
1427 wav->rate = header->rate;
1428 wav->channels = header->channels;
1429 wav->blockalign = header->blockalign;
1430 wav->depth = header->bits_per_sample;
1431 wav->av_bps = header->av_bps;
1437 /* do format specific handling */
1438 switch (wav->format) {
1439 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1440 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1442 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1443 * bitrate inside the mpeg stream */
1444 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1448 case GST_RIFF_WAVE_FORMAT_PCM:
1449 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1450 goto invalid_blockalign;
1453 if (wav->av_bps > wav->blockalign * wav->rate)
1455 /* use the configured bps */
1456 wav->bps = wav->av_bps;
1460 wav->width = (wav->blockalign * 8) / wav->channels;
1461 wav->bytes_per_sample = wav->channels * wav->width / 8;
1463 if (wav->bytes_per_sample <= 0)
1464 goto no_bytes_per_sample;
1466 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1467 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1468 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1469 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1470 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1471 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1472 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1474 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1475 * formats). This will make the element output a BYTE format segment and
1476 * will not timestamp the outgoing buffers.
1478 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1480 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1482 /* create pad later so we can sniff the first few bytes
1483 * of the real data and correct our caps if necessary */
1484 gst_caps_replace (&wav->caps, caps);
1485 gst_caps_replace (&caps, NULL);
1487 wav->got_fmt = TRUE;
1490 wav->tags = gst_tag_list_new_empty ();
1492 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1493 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1495 g_free (codec_name);
1501 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1502 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1504 /* loop headers until we get data */
1506 if (wav->streaming) {
1507 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1514 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1515 &buf)) != GST_FLOW_OK)
1516 goto header_read_error;
1517 gst_buffer_map (buf, &map, GST_MAP_READ);
1518 tag = GST_READ_UINT32_LE (map.data);
1519 size = GST_READ_UINT32_LE (map.data + 4);
1520 gst_buffer_unmap (buf, &map);
1523 GST_INFO_OBJECT (wav,
1524 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1525 GST_FOURCC_ARGS (tag), wav->offset);
1527 /* wav is a st00pid format, we don't know for sure where data starts.
1528 * So we have to go bit by bit until we find the 'data' header
1531 case GST_RIFF_TAG_data:{
1532 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1533 if (wav->ignore_length) {
1534 GST_DEBUG_OBJECT (wav, "Ignoring length");
1537 if (wav->streaming) {
1538 gst_adapter_flush (wav->adapter, 8);
1541 gst_buffer_unref (buf);
1544 wav->datastart = wav->offset;
1545 /* If size is zero, then the data chunk probably actually extends to
1546 the end of the file */
1547 if (size == 0 && upstream_size) {
1548 size = upstream_size - wav->datastart;
1550 /* Or the file might be truncated */
1551 else if (upstream_size) {
1552 size = MIN (size, (upstream_size - wav->datastart));
1554 wav->datasize = (guint64) size;
1555 wav->dataleft = (guint64) size;
1556 wav->end_offset = size + wav->datastart;
1557 if (!wav->streaming) {
1558 /* We will continue parsing tags 'till end */
1559 wav->offset += size;
1561 GST_DEBUG_OBJECT (wav, "datasize = %u", size);
1564 case GST_RIFF_TAG_fact:{
1565 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1566 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1567 const guint data_size = 4;
1569 GST_INFO_OBJECT (wav, "Have fact chunk");
1570 if (size < data_size) {
1571 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1572 /* need more data */
1575 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1579 /* number of samples (for compressed formats) */
1580 if (wav->streaming) {
1581 const guint8 *data = NULL;
1583 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1586 gst_adapter_flush (wav->adapter, 8);
1587 data = gst_adapter_map (wav->adapter, data_size);
1588 wav->fact = GST_READ_UINT32_LE (data);
1589 gst_adapter_unmap (wav->adapter);
1590 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1592 gst_buffer_unref (buf);
1595 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1596 data_size, &buf)) != GST_FLOW_OK)
1597 goto header_read_error;
1598 gst_buffer_extract (buf, 0, &wav->fact, 4);
1599 wav->fact = GUINT32_FROM_LE (wav->fact);
1600 gst_buffer_unref (buf);
1602 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1603 wav->offset += 8 + GST_ROUND_UP_2 (size);
1606 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1607 /* need more data */
1613 case GST_RIFF_TAG_acid:{
1614 const gst_riff_acid *acid = NULL;
1615 const guint data_size = sizeof (gst_riff_acid);
1618 GST_INFO_OBJECT (wav, "Have acid chunk");
1619 if (size < data_size) {
1620 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1621 /* need more data */
1624 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1628 if (wav->streaming) {
1629 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1632 gst_adapter_flush (wav->adapter, 8);
1633 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1635 tempo = acid->tempo;
1636 gst_adapter_unmap (wav->adapter);
1639 gst_buffer_unref (buf);
1642 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1643 size, &buf)) != GST_FLOW_OK)
1644 goto header_read_error;
1645 gst_buffer_map (buf, &map, GST_MAP_READ);
1646 acid = (const gst_riff_acid *) map.data;
1647 tempo = acid->tempo;
1648 gst_buffer_unmap (buf, &map);
1650 /* send data as tags */
1652 wav->tags = gst_tag_list_new_empty ();
1653 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1654 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1656 size = GST_ROUND_UP_2 (size);
1657 if (wav->streaming) {
1658 gst_adapter_flush (wav->adapter, size);
1660 gst_buffer_unref (buf);
1662 wav->offset += 8 + size;
1665 /* FIXME: all list tags after data are ignored in streaming mode */
1666 case GST_RIFF_TAG_LIST:{
1669 if (wav->streaming) {
1670 const guint8 *data = NULL;
1672 if (gst_adapter_available (wav->adapter) < 12) {
1675 data = gst_adapter_map (wav->adapter, 12);
1676 ltag = GST_READ_UINT32_LE (data + 8);
1677 gst_adapter_unmap (wav->adapter);
1679 gst_buffer_unref (buf);
1682 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1683 &buf)) != GST_FLOW_OK)
1684 goto header_read_error;
1685 gst_buffer_extract (buf, 8, <ag, 4);
1686 ltag = GUINT32_FROM_LE (ltag);
1689 case GST_RIFF_LIST_INFO:{
1690 const gint data_size = size - 4;
1693 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1694 if (wav->streaming) {
1695 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1698 gst_adapter_flush (wav->adapter, 12);
1700 if (data_size > 0) {
1701 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1703 gst_adapter_flush (wav->adapter, 1);
1707 gst_buffer_unref (buf);
1709 if (data_size > 0) {
1711 gst_pad_pull_range (wav->sinkpad, wav->offset,
1712 data_size, &buf)) != GST_FLOW_OK)
1713 goto header_read_error;
1716 if (data_size > 0) {
1718 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1720 GstTagList *old = wav->tags;
1722 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1724 gst_tag_list_free (old);
1725 gst_tag_list_free (new);
1727 gst_buffer_unref (buf);
1728 wav->offset += GST_ROUND_UP_2 (data_size);
1732 case GST_RIFF_LIST_adtl:{
1733 const gint data_size = size;
1735 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1736 if (wav->streaming) {
1737 const guint8 *data = NULL;
1739 gst_adapter_flush (wav->adapter, 12);
1740 data = gst_adapter_map (wav->adapter, data_size);
1741 gst_wavparse_adtl_chunk (wav, data, data_size);
1742 gst_adapter_unmap (wav->adapter);
1746 gst_buffer_unref (buf);
1749 gst_pad_pull_range (wav->sinkpad, wav->offset + 12,
1750 data_size, &buf)) != GST_FLOW_OK)
1751 goto header_read_error;
1752 gst_buffer_map (buf, &map, GST_MAP_READ);
1753 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1755 gst_buffer_unmap (buf, &map);
1759 GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1760 GST_FOURCC_ARGS (ltag));
1761 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1762 /* need more data */
1768 case GST_RIFF_TAG_cue:{
1769 const guint data_size = size;
1771 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1772 if (wav->streaming) {
1773 const guint8 *data = NULL;
1775 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1778 gst_adapter_flush (wav->adapter, 8);
1780 data = gst_adapter_map (wav->adapter, data_size);
1781 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1782 goto header_read_error;
1784 gst_adapter_unmap (wav->adapter);
1789 gst_buffer_unref (buf);
1792 gst_pad_pull_range (wav->sinkpad, wav->offset,
1793 data_size, &buf)) != GST_FLOW_OK)
1794 goto header_read_error;
1795 gst_buffer_map (buf, &map, GST_MAP_READ);
1796 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1798 goto header_read_error;
1800 gst_buffer_unmap (buf, &map);
1802 size = GST_ROUND_UP_2 (size);
1803 if (wav->streaming) {
1804 gst_adapter_flush (wav->adapter, size);
1806 gst_buffer_unref (buf);
1808 size = GST_ROUND_UP_2 (size);
1809 wav->offset += size;
1813 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1814 /* need more data */
1819 if (upstream_size && (wav->offset >= upstream_size)) {
1820 /* Now we are gone through the whole file */
1825 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1827 if (wav->bps <= 0 && wav->fact) {
1829 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1831 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1832 (guint64) wav->fact);
1833 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1838 if (gst_wavparse_calculate_duration (wav)) {
1839 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1840 if (!wav->ignore_length)
1841 wav->segment.duration = wav->duration;
1843 gst_wavparse_create_toc (wav);
1845 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1846 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1847 if (!wav->ignore_length)
1848 wav->segment.duration = wav->datasize;
1851 /* now we have all the info to perform a pending seek if any, if no
1852 * event, this will still do the right thing and it will also send
1853 * the right newsegment event downstream. */
1854 gst_wavparse_perform_seek (wav, wav->seek_event);
1855 /* remove pending event */
1856 event_p = &wav->seek_event;
1857 gst_event_replace (event_p, NULL);
1859 /* we just started, we are discont */
1860 wav->discont = TRUE;
1862 wav->state = GST_WAVPARSE_DATA;
1864 /* determine reasonable max buffer size,
1865 * that is, buffers not too small either size or time wise
1866 * so we do not end up with too many of them */
1869 gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
1870 wav->max_buf_size = upstream_size;
1871 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1872 if (wav->blockalign > 0)
1873 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1875 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1883 g_free (codec_name);
1887 gst_caps_unref (caps);
1892 res = GST_FLOW_ERROR;
1897 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1898 ("Invalid WAV header (no fmt at start): %"
1899 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1904 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1905 ("Couldn't parse audio header"));
1910 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1911 ("Stream claims to contain no channels - invalid data"));
1916 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1917 ("Stream with sample_rate == 0 - invalid data"));
1922 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1923 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1924 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1929 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1930 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1931 wav->av_bps, wav->blockalign * wav->rate));
1934 no_bytes_per_sample:
1936 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1937 ("Could not caluclate bytes per sample - invalid data"));
1942 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1943 ("No caps found for format 0x%x, %u channels, %u Hz",
1944 wav->format, wav->channels, wav->rate));
1949 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1950 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
1956 * Read WAV file tag when streaming
1958 static GstFlowReturn
1959 gst_wavparse_parse_stream_init (GstWavParse * wav)
1961 if (gst_adapter_available (wav->adapter) >= 12) {
1964 /* _take flushes the data */
1965 tmp = gst_adapter_take_buffer (wav->adapter, 12);
1967 GST_DEBUG ("Parsing wav header");
1968 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
1969 return GST_FLOW_ERROR;
1972 /* Go to next state */
1973 wav->state = GST_WAVPARSE_HEADER;
1978 /* handle an event sent directly to the element.
1980 * This event can be sent either in the READY state or the
1981 * >READY state. The only event of interest really is the seek
1984 * In the READY state we can only store the event and try to
1985 * respect it when going to PAUSED. We assume we are in the
1986 * READY state when our parsing state != GST_WAVPARSE_DATA.
1988 * When we are steaming, we can simply perform the seek right
1992 gst_wavparse_send_event (GstElement * element, GstEvent * event)
1994 GstWavParse *wav = GST_WAVPARSE (element);
1995 gboolean res = FALSE;
1998 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
2000 switch (GST_EVENT_TYPE (event)) {
2001 case GST_EVENT_SEEK:
2002 if (wav->state == GST_WAVPARSE_DATA) {
2003 /* we can handle the seek directly when streaming data */
2004 res = gst_wavparse_perform_seek (wav, event);
2006 GST_DEBUG_OBJECT (wav, "queuing seek for later");
2008 event_p = &wav->seek_event;
2009 gst_event_replace (event_p, event);
2011 /* we always return true */
2018 gst_event_unref (event);
2023 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
2027 s = gst_caps_get_structure (caps, 0);
2028 if (!gst_structure_has_name (s, "audio/x-dts"))
2030 if (prob >= GST_TYPE_FIND_LIKELY)
2032 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
2033 if (prob < GST_TYPE_FIND_POSSIBLE)
2035 /* .. in which case we want at least a valid-looking rate and channels */
2036 if (!gst_structure_has_field (s, "channels"))
2038 /* and for extra assurance we could also check the rate from the DTS frame
2039 * against the one in the wav header, but for now let's not do that */
2040 return gst_structure_has_field (s, "rate");
2044 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
2048 GST_DEBUG_OBJECT (wav, "adding src pad");
2051 s = gst_caps_get_structure (wav->caps, 0);
2052 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
2053 GstTypeFindProbability prob;
2056 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
2057 if (tf_caps != NULL) {
2058 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
2059 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
2060 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
2061 gst_caps_unref (wav->caps);
2062 wav->caps = tf_caps;
2064 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
2065 GST_TAG_AUDIO_CODEC, "dts", NULL);
2067 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
2068 "marked as raw PCM audio, but ignoring for now", tf_caps);
2069 gst_caps_unref (tf_caps);
2075 gst_pad_set_caps (wav->srcpad, wav->caps);
2076 gst_caps_replace (&wav->caps, NULL);
2078 if (wav->start_segment) {
2079 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
2080 gst_pad_push_event (wav->srcpad, wav->start_segment);
2081 wav->start_segment = NULL;
2085 gst_pad_push_event (wav->srcpad, gst_event_new_tag ("GstParser",
2091 static GstFlowReturn
2092 gst_wavparse_stream_data (GstWavParse * wav)
2094 GstBuffer *buf = NULL;
2095 GstFlowReturn res = GST_FLOW_OK;
2096 guint64 desired, obtained;
2097 GstClockTime timestamp, next_timestamp, duration;
2098 guint64 pos, nextpos;
2101 GST_LOG_OBJECT (wav,
2102 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
2103 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
2105 /* Get the next n bytes and output them */
2106 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
2109 /* scale the amount of data by the segment rate so we get equal
2110 * amounts of data regardless of the playback rate */
2112 MIN (gst_guint64_to_gdouble (wav->dataleft),
2113 wav->max_buf_size * ABS (wav->segment.rate));
2115 if (desired >= wav->blockalign && wav->blockalign > 0)
2116 desired -= (desired % wav->blockalign);
2118 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
2119 "from the sinkpad", desired);
2121 if (wav->streaming) {
2122 guint avail = gst_adapter_available (wav->adapter);
2125 /* flush some bytes if evil upstream sends segment that starts
2126 * before data or does is not send sample aligned segment */
2127 if (G_LIKELY (wav->offset >= wav->datastart)) {
2128 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
2130 extra = wav->datastart - wav->offset;
2133 if (G_UNLIKELY (extra)) {
2134 extra = wav->bytes_per_sample - extra;
2135 if (extra <= avail) {
2136 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2137 gst_adapter_flush (wav->adapter, extra);
2138 wav->offset += extra;
2139 wav->dataleft -= extra;
2140 goto iterate_adapter;
2142 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2143 gst_adapter_clear (wav->adapter);
2144 wav->offset += avail;
2145 wav->dataleft -= avail;
2150 if (avail < desired) {
2151 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2155 buf = gst_adapter_take_buffer (wav->adapter, desired);
2157 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2158 desired, &buf)) != GST_FLOW_OK)
2161 /* we may get a short buffer at the end of the file */
2162 if (gst_buffer_get_size (buf) < desired) {
2163 gsize size = gst_buffer_get_size (buf);
2165 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2166 if (size >= wav->blockalign) {
2167 buf = gst_buffer_make_writable (buf);
2168 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2170 gst_buffer_unref (buf);
2176 obtained = gst_buffer_get_size (buf);
2178 /* our positions in bytes */
2179 pos = wav->offset - wav->datastart;
2180 nextpos = pos + obtained;
2182 /* update offsets, does not overflow. */
2183 buf = gst_buffer_make_writable (buf);
2184 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2185 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2187 /* first chunk of data? create the source pad. We do this only here so
2188 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2189 if (G_UNLIKELY (wav->first)) {
2191 /* this will also push the segment events */
2192 gst_wavparse_add_src_pad (wav, buf);
2194 /* If we have a pending start segment, send it now. */
2195 if (G_UNLIKELY (wav->start_segment != NULL)) {
2196 gst_pad_push_event (wav->srcpad, wav->start_segment);
2197 wav->start_segment = NULL;
2202 /* and timestamps if we have a bitrate, be careful for overflows */
2204 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2206 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2207 duration = next_timestamp - timestamp;
2209 /* update current running segment position */
2210 if (G_LIKELY (next_timestamp >= wav->segment.start))
2211 wav->segment.position = next_timestamp;
2212 } else if (wav->fact) {
2214 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2215 /* and timestamps if we have a bitrate, be careful for overflows */
2216 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2217 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2218 duration = next_timestamp - timestamp;
2220 /* no bitrate, all we know is that the first sample has timestamp 0, all
2221 * other positions and durations have unknown timestamp. */
2225 timestamp = GST_CLOCK_TIME_NONE;
2226 duration = GST_CLOCK_TIME_NONE;
2227 /* update current running segment position with byte offset */
2228 if (G_LIKELY (nextpos >= wav->segment.start))
2229 wav->segment.position = nextpos;
2231 if ((pos > 0) && wav->vbr) {
2232 /* don't set timestamps for VBR files if it's not the first buffer */
2233 timestamp = GST_CLOCK_TIME_NONE;
2234 duration = GST_CLOCK_TIME_NONE;
2237 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2238 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2239 wav->discont = FALSE;
2242 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2243 GST_BUFFER_DURATION (buf) = duration;
2245 GST_LOG_OBJECT (wav,
2246 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2247 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2248 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2250 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2253 if (obtained < wav->dataleft) {
2254 wav->offset += obtained;
2255 wav->dataleft -= obtained;
2257 wav->offset += wav->dataleft;
2261 /* Iterate until need more data, so adapter size won't grow */
2262 if (wav->streaming) {
2263 GST_LOG_OBJECT (wav,
2264 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2266 goto iterate_adapter;
2273 GST_DEBUG_OBJECT (wav, "found EOS");
2274 return GST_FLOW_EOS;
2278 /* check if we got EOS */
2279 if (res == GST_FLOW_EOS)
2282 GST_WARNING_OBJECT (wav,
2283 "Error getting %" G_GINT64_FORMAT " bytes from the "
2284 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2289 GST_INFO_OBJECT (wav,
2290 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2291 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2292 gst_pad_is_linked (wav->srcpad));
2298 gst_wavparse_loop (GstPad * pad)
2301 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2303 GST_LOG_OBJECT (wav, "process data");
2305 switch (wav->state) {
2306 case GST_WAVPARSE_START:
2307 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2308 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2311 wav->state = GST_WAVPARSE_HEADER;
2314 case GST_WAVPARSE_HEADER:
2315 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2316 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2319 wav->state = GST_WAVPARSE_DATA;
2320 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2323 case GST_WAVPARSE_DATA:
2324 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2328 g_assert_not_reached ();
2335 const gchar *reason = gst_flow_get_name (ret);
2337 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2338 gst_pad_pause_task (pad);
2340 if (ret == GST_FLOW_EOS) {
2341 /* handle end-of-stream/segment */
2342 /* so align our position with the end of it, if there is one
2343 * this ensures a subsequent will arrive at correct base/acc time */
2344 if (wav->segment.format == GST_FORMAT_TIME) {
2345 if (wav->segment.rate > 0.0 &&
2346 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2347 wav->segment.position = wav->segment.stop;
2348 else if (wav->segment.rate < 0.0)
2349 wav->segment.position = wav->segment.start;
2351 /* add pad before we perform EOS */
2352 if (G_UNLIKELY (wav->first)) {
2354 gst_wavparse_add_src_pad (wav, NULL);
2357 if (wav->state == GST_WAVPARSE_START)
2358 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2359 ("No valid input found before end of stream"), (NULL));
2361 /* perform EOS logic */
2362 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2365 if ((stop = wav->segment.stop) == -1)
2366 stop = wav->segment.duration;
2368 gst_element_post_message (GST_ELEMENT_CAST (wav),
2369 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2370 wav->segment.format, stop));
2371 gst_pad_push_event (wav->srcpad,
2372 gst_event_new_segment_done (wav->segment.format, stop));
2374 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2376 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2377 /* for fatal errors we post an error message, post the error
2378 * first so the app knows about the error first. */
2379 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2380 (_("Internal data flow error.")),
2381 ("streaming task paused, reason %s (%d)", reason, ret));
2382 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2388 static GstFlowReturn
2389 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2392 GstWavParse *wav = GST_WAVPARSE (parent);
2394 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2395 gst_buffer_get_size (buf));
2397 gst_adapter_push (wav->adapter, buf);
2399 switch (wav->state) {
2400 case GST_WAVPARSE_START:
2401 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2402 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2405 if (wav->state != GST_WAVPARSE_HEADER)
2408 /* otherwise fall-through */
2409 case GST_WAVPARSE_HEADER:
2410 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2411 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2414 if (!wav->got_fmt || wav->datastart == 0)
2417 wav->state = GST_WAVPARSE_DATA;
2418 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2421 case GST_WAVPARSE_DATA:
2422 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2423 wav->discont = TRUE;
2424 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2428 g_return_val_if_reached (GST_FLOW_ERROR);
2431 if (G_UNLIKELY (wav->abort_buffering)) {
2432 wav->abort_buffering = FALSE;
2433 ret = GST_FLOW_ERROR;
2434 /* sort of demux/parse error */
2435 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2441 static GstFlowReturn
2442 gst_wavparse_flush_data (GstWavParse * wav)
2444 GstFlowReturn ret = GST_FLOW_OK;
2447 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2449 wav->end_offset = wav->offset + av;
2450 ret = gst_wavparse_stream_data (wav);
2457 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2459 GstWavParse *wav = GST_WAVPARSE (parent);
2460 gboolean ret = TRUE;
2462 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2464 switch (GST_EVENT_TYPE (event)) {
2465 case GST_EVENT_CAPS:
2467 /* discard, we'll come up with proper src caps */
2468 gst_event_unref (event);
2471 case GST_EVENT_SEGMENT:
2473 gint64 start, stop, offset = 0, end_offset = -1;
2476 /* some debug output */
2477 gst_event_copy_segment (event, &segment);
2478 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2481 if (wav->state != GST_WAVPARSE_DATA) {
2482 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2486 /* now we are either committed to TIME or BYTE format,
2487 * and we only expect a BYTE segment, e.g. following a seek */
2488 if (segment.format == GST_FORMAT_BYTES) {
2489 /* handle (un)signed issues */
2490 start = segment.start;
2491 stop = segment.stop;
2494 start -= wav->datastart;
2495 start = MAX (start, 0);
2499 segment.stop -= wav->datastart;
2500 segment.stop = MAX (stop, 0);
2502 if (wav->segment.format == GST_FORMAT_TIME) {
2503 guint64 bps = wav->bps;
2505 /* operating in format TIME, so we can convert */
2506 if (!bps && wav->fact)
2508 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2512 gst_util_uint64_scale_ceil (start, GST_SECOND,
2513 (guint64) wav->bps);
2516 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2517 (guint64) wav->bps);
2521 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2525 segment.start = start;
2526 segment.stop = stop;
2528 /* accept upstream's notion of segment and distribute along */
2529 segment.format = wav->segment.format;
2530 segment.time = segment.position = segment.start;
2531 segment.duration = wav->segment.duration;
2532 segment.base = gst_segment_to_running_time (&wav->segment,
2533 GST_FORMAT_TIME, wav->segment.position);
2535 gst_segment_copy_into (&segment, &wav->segment);
2537 /* also store the newsegment event for the streaming thread */
2538 if (wav->start_segment)
2539 gst_event_unref (wav->start_segment);
2540 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2541 wav->start_segment = gst_event_new_segment (&segment);
2543 /* stream leftover data in current segment */
2544 gst_wavparse_flush_data (wav);
2545 /* and set up streaming thread for next one */
2546 wav->offset = offset;
2547 wav->end_offset = end_offset;
2548 if (wav->end_offset > 0) {
2549 wav->dataleft = wav->end_offset - wav->offset;
2551 /* infinity; upstream will EOS when done */
2552 wav->dataleft = G_MAXUINT64;
2555 gst_event_unref (event);
2559 /* add pad if needed so EOS is seen downstream */
2560 if (G_UNLIKELY (wav->first)) {
2562 gst_wavparse_add_src_pad (wav, NULL);
2564 /* stream leftover data in current segment */
2565 gst_wavparse_flush_data (wav);
2568 if (wav->state == GST_WAVPARSE_START)
2569 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE,
2570 ("No valid input found before end of stream"), (NULL));
2573 case GST_EVENT_FLUSH_STOP:
2577 gst_adapter_clear (wav->adapter);
2578 wav->discont = TRUE;
2579 dur = wav->segment.duration;
2580 gst_segment_init (&wav->segment, wav->segment.format);
2581 wav->segment.duration = dur;
2585 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2593 /* convert and query stuff */
2594 static const GstFormat *
2595 gst_wavparse_get_formats (GstPad * pad)
2597 static GstFormat formats[] = {
2600 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2609 gst_wavparse_pad_convert (GstPad * pad,
2610 GstFormat src_format, gint64 src_value,
2611 GstFormat * dest_format, gint64 * dest_value)
2613 GstWavParse *wavparse;
2614 gboolean res = TRUE;
2616 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2618 if (*dest_format == src_format) {
2619 *dest_value = src_value;
2623 if ((wavparse->bps == 0) && !wavparse->fact)
2626 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2627 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2629 switch (src_format) {
2630 case GST_FORMAT_BYTES:
2631 switch (*dest_format) {
2632 case GST_FORMAT_DEFAULT:
2633 *dest_value = src_value / wavparse->bytes_per_sample;
2634 /* make sure we end up on a sample boundary */
2635 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2637 case GST_FORMAT_TIME:
2638 /* src_value + datastart = offset */
2639 GST_INFO_OBJECT (wavparse,
2640 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2642 if (wavparse->bps > 0)
2643 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2644 (guint64) wavparse->bps);
2645 else if (wavparse->fact) {
2646 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2647 wavparse->rate, wavparse->fact);
2650 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2661 case GST_FORMAT_DEFAULT:
2662 switch (*dest_format) {
2663 case GST_FORMAT_BYTES:
2664 *dest_value = src_value * wavparse->bytes_per_sample;
2666 case GST_FORMAT_TIME:
2667 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2668 (guint64) wavparse->rate);
2676 case GST_FORMAT_TIME:
2677 switch (*dest_format) {
2678 case GST_FORMAT_BYTES:
2679 if (wavparse->bps > 0)
2680 *dest_value = gst_util_uint64_scale (src_value,
2681 (guint64) wavparse->bps, GST_SECOND);
2683 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2684 wavparse->rate, wavparse->fact);
2686 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2688 /* make sure we end up on a sample boundary */
2689 *dest_value -= *dest_value % wavparse->blockalign;
2691 case GST_FORMAT_DEFAULT:
2692 *dest_value = gst_util_uint64_scale (src_value,
2693 (guint64) wavparse->rate, GST_SECOND);
2712 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2718 /* handle queries for location and length in requested format */
2720 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2722 gboolean res = TRUE;
2723 GstWavParse *wav = GST_WAVPARSE (parent);
2725 /* only if we know */
2726 if (wav->state != GST_WAVPARSE_DATA) {
2730 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2732 switch (GST_QUERY_TYPE (query)) {
2733 case GST_QUERY_POSITION:
2739 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2740 curb = wav->offset - wav->datastart;
2741 gst_query_parse_position (query, &format, NULL);
2742 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2745 case GST_FORMAT_BYTES:
2746 format = GST_FORMAT_BYTES;
2750 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2755 gst_query_set_position (query, format, cur);
2758 case GST_QUERY_DURATION:
2760 gint64 duration = 0;
2763 if (wav->ignore_length) {
2768 gst_query_parse_duration (query, &format, NULL);
2771 case GST_FORMAT_BYTES:{
2772 format = GST_FORMAT_BYTES;
2773 duration = wav->datasize;
2776 case GST_FORMAT_TIME:
2777 if ((res = gst_wavparse_calculate_duration (wav))) {
2778 duration = wav->duration;
2786 gst_query_set_duration (query, format, duration);
2789 case GST_QUERY_CONVERT:
2791 gint64 srcvalue, dstvalue;
2792 GstFormat srcformat, dstformat;
2794 gst_query_parse_convert (query, &srcformat, &srcvalue,
2795 &dstformat, &dstvalue);
2796 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2797 &dstformat, &dstvalue);
2799 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2802 case GST_QUERY_SEEKING:{
2804 gboolean seekable = FALSE;
2806 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2807 if (fmt == wav->segment.format) {
2808 if (wav->streaming) {
2811 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2812 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2813 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2814 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2816 gst_query_unref (q);
2818 GST_LOG_OBJECT (wav, "looping => seekable");
2822 } else if (fmt == GST_FORMAT_TIME) {
2826 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2831 res = gst_pad_query_default (pad, parent, query);
2838 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2840 GstWavParse *wavparse = GST_WAVPARSE (parent);
2841 gboolean res = FALSE;
2843 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2845 switch (GST_EVENT_TYPE (event)) {
2846 case GST_EVENT_SEEK:
2847 /* can only handle events when we are in the data state */
2848 if (wavparse->state == GST_WAVPARSE_DATA) {
2849 res = gst_wavparse_perform_seek (wavparse, event);
2851 gst_event_unref (event);
2854 case GST_EVENT_TOC_SELECT:
2857 GstTocEntry *entry = NULL;
2858 GstEvent *seek_event;
2861 if (!wavparse->toc) {
2862 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2865 gst_event_parse_toc_select (event, &uid);
2867 GST_OBJECT_LOCK (wavparse);
2868 entry = gst_toc_find_entry (wavparse->toc, uid);
2869 if (entry == NULL) {
2870 GST_OBJECT_UNLOCK (wavparse);
2871 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2875 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2876 GST_OBJECT_UNLOCK (wavparse);
2877 seek_event = gst_event_new_seek (1.0,
2879 GST_SEEK_FLAG_FLUSH,
2880 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2881 res = gst_wavparse_perform_seek (wavparse, seek_event);
2882 gst_event_unref (seek_event);
2886 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2890 gst_event_unref (event);
2895 res = gst_pad_push_event (wavparse->sinkpad, event);
2902 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2904 GstWavParse *wav = GST_WAVPARSE (parent);
2909 gst_adapter_clear (wav->adapter);
2910 g_object_unref (wav->adapter);
2911 wav->adapter = NULL;
2914 query = gst_query_new_scheduling ();
2916 if (!gst_pad_peer_query (sinkpad, query)) {
2917 gst_query_unref (query);
2921 pull_mode = gst_query_has_scheduling_mode (query, GST_PAD_MODE_PULL);
2922 gst_query_unref (query);
2927 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2928 wav->streaming = FALSE;
2929 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
2933 GST_DEBUG_OBJECT (sinkpad, "activating push");
2934 wav->streaming = TRUE;
2935 wav->adapter = gst_adapter_new ();
2936 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
2942 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
2943 GstPadMode mode, gboolean active)
2948 case GST_PAD_MODE_PUSH:
2951 case GST_PAD_MODE_PULL:
2953 /* if we have a scheduler we can start the task */
2954 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
2957 res = gst_pad_stop_task (sinkpad);
2967 static GstStateChangeReturn
2968 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
2970 GstStateChangeReturn ret;
2971 GstWavParse *wav = GST_WAVPARSE (element);
2973 switch (transition) {
2974 case GST_STATE_CHANGE_NULL_TO_READY:
2976 case GST_STATE_CHANGE_READY_TO_PAUSED:
2977 gst_wavparse_reset (wav);
2979 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
2985 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2987 switch (transition) {
2988 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2990 case GST_STATE_CHANGE_PAUSED_TO_READY:
2991 gst_wavparse_reset (wav);
2993 case GST_STATE_CHANGE_READY_TO_NULL:
3002 gst_wavparse_set_property (GObject * object, guint prop_id,
3003 const GValue * value, GParamSpec * pspec)
3007 g_return_if_fail (GST_IS_WAVPARSE (object));
3008 self = GST_WAVPARSE (object);
3011 case PROP_IGNORE_LENGTH:
3012 self->ignore_length = g_value_get_boolean (value);
3015 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3021 gst_wavparse_get_property (GObject * object, guint prop_id,
3022 GValue * value, GParamSpec * pspec)
3026 g_return_if_fail (GST_IS_WAVPARSE (object));
3027 self = GST_WAVPARSE (object);
3030 case PROP_IGNORE_LENGTH:
3031 g_value_set_boolean (value, self->ignore_length);
3034 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3039 plugin_init (GstPlugin * plugin)
3043 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
3047 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
3050 "Parse a .wav file into raw audio",
3051 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)