1 /* -*- Mode: C; tab-width: 2; indent-tabs-mode: t; c-basic-offset: 2 -*- */
3 * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
4 * Copyright (C) <2006> Nokia Corporation, Stefan Kost <stefan.kost@nokia.com>.
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-wavparse
25 * Parse a .wav file into raw or compressed audio.
27 * Wavparse supports both push and pull mode operations, making it possible to
28 * stream from a network source.
31 * <title>Example launch line</title>
33 * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink
34 * ]| Read a wav file and output to the soundcard using the ALSA element. The
35 * wav file is assumed to contain raw uncompressed samples.
37 * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
38 * ]| Stream data from a network url.
41 * Last reviewed on 2007-02-14 (0.10.6)
46 * http://replaygain.hydrogenaudio.org/file_format_wav.html
56 #include "gstwavparse.h"
57 #include "gst/riff/riff-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;
142 } GstWavParseLabl, GstWavParseNote;
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_unref (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 = g_memdup (data + 4, size - 4);
1224 wav->labls = g_list_append (wav->labls, labl);
1230 * gst_wavparse_note_chunk:
1231 * @wav GstWavParse object
1232 * @data holder for data
1233 * @size holder for data size
1235 * Parse note from @data to wav->notes.
1237 * Returns: %TRUE when note chunk is available
1240 gst_wavparse_note_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
1242 GstWavParseNote *note;
1247 note = g_new0 (GstWavParseNote, 1);
1251 note->cue_point_id = GST_READ_UINT32_LE (data);
1252 note->text = g_memdup (data + 4, size - 4);
1254 wav->notes = g_list_append (wav->notes, note);
1260 * gst_wavparse_adtl_chunk:
1261 * @wav GstWavParse object
1262 * @data holder for data
1263 * @size holder for data size
1265 * Parse adtl from @data.
1267 * Returns: %TRUE when adtl chunk is available
1270 gst_wavparse_adtl_chunk (GstWavParse * wav, const guint8 * data, guint32 size)
1272 guint32 ltag, lsize, offset = 0;
1275 ltag = GST_READ_UINT32_LE (data + offset);
1276 lsize = GST_READ_UINT32_LE (data + offset + 4);
1278 case GST_RIFF_TAG_labl:
1279 gst_wavparse_labl_chunk (wav, data + offset, size);
1281 case GST_RIFF_TAG_note:
1282 gst_wavparse_note_chunk (wav, data + offset, size);
1287 offset += 8 + GST_ROUND_UP_2 (lsize);
1288 size -= 8 + GST_ROUND_UP_2 (lsize);
1295 gst_wavparse_get_tags_toc_entry (GstToc * toc, gchar * id)
1297 GstTagList *tags = NULL;
1298 GstTocEntry *entry = NULL;
1300 entry = gst_toc_find_entry (toc, id);
1301 if (entry != NULL) {
1302 tags = gst_toc_entry_get_tags (entry);
1304 tags = gst_tag_list_new_empty ();
1305 gst_toc_entry_set_tags (entry, tags);
1313 * gst_wavparse_create_toc:
1314 * @wav GstWavParse object
1316 * Create TOC from wav->cues and wav->labls.
1319 gst_wavparse_create_toc (GstWavParse * wav)
1324 GstWavParseCue *cue;
1325 GstWavParseLabl *labl;
1326 GstWavParseNote *note;
1329 GstTocEntry *entry = NULL, *cur_subentry = NULL, *prev_subentry = NULL;
1331 GST_OBJECT_LOCK (wav);
1333 GST_OBJECT_UNLOCK (wav);
1334 GST_WARNING_OBJECT (wav, "found another TOC");
1339 GST_OBJECT_UNLOCK (wav);
1343 /* FIXME: send CURRENT scope toc too */
1344 toc = gst_toc_new (GST_TOC_SCOPE_GLOBAL);
1346 /* add cue edition */
1347 entry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_EDITION, "cue");
1348 gst_toc_entry_set_start_stop_times (entry, 0, wav->duration);
1349 gst_toc_append_entry (toc, entry);
1351 /* add tracks in cue edition */
1352 list = g_list_first (wav->cues);
1355 prev_subentry = cur_subentry;
1356 /* previous track stop time = current track start time */
1357 if (prev_subentry != NULL) {
1358 gst_toc_entry_get_start_stop_times (prev_subentry, &start, NULL);
1359 stop = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1360 gst_toc_entry_set_start_stop_times (prev_subentry, start, stop);
1362 id = g_strdup_printf ("%08x", cue->id);
1363 cur_subentry = gst_toc_entry_new (GST_TOC_ENTRY_TYPE_TRACK, id);
1365 start = gst_util_uint64_scale_round (cue->position, GST_SECOND, wav->rate);
1366 stop = wav->duration;
1367 gst_toc_entry_set_start_stop_times (cur_subentry, start, stop);
1368 gst_toc_entry_append_sub_entry (entry, cur_subentry);
1369 list = g_list_next (list);
1372 /* add tags in tracks */
1373 list = g_list_first (wav->labls);
1376 id = g_strdup_printf ("%08x", labl->cue_point_id);
1377 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1380 gst_tag_list_add (tags, GST_TAG_MERGE_APPEND, GST_TAG_TITLE, labl->text,
1383 list = g_list_next (list);
1385 list = g_list_first (wav->notes);
1388 id = g_strdup_printf ("%08x", note->cue_point_id);
1389 tags = gst_wavparse_get_tags_toc_entry (toc, id);
1392 gst_tag_list_add (tags, GST_TAG_MERGE_PREPEND, GST_TAG_COMMENT,
1395 list = g_list_next (list);
1398 /* send data as TOC */
1401 /* send TOC event */
1403 GST_OBJECT_UNLOCK (wav);
1404 gst_pad_push_event (wav->srcpad, gst_event_new_toc (wav->toc, FALSE));
1410 #define MAX_BUFFER_SIZE 4096
1412 static GstFlowReturn
1413 gst_wavparse_stream_headers (GstWavParse * wav)
1415 GstFlowReturn res = GST_FLOW_OK;
1416 GstBuffer *buf = NULL;
1417 gst_riff_strf_auds *header = NULL;
1419 gboolean gotdata = FALSE;
1420 GstCaps *caps = NULL;
1421 gchar *codec_name = NULL;
1423 gint64 upstream_size = 0;
1425 /* search for "_fmt" chunk, which should be first */
1426 while (!wav->got_fmt) {
1429 /* The header starts with a 'fmt ' tag */
1430 if (wav->streaming) {
1431 if (!gst_wavparse_peek_chunk (wav, &tag, &size))
1434 gst_adapter_flush (wav->adapter, 8);
1438 buf = gst_adapter_take_buffer (wav->adapter, size);
1440 gst_adapter_flush (wav->adapter, 1);
1441 wav->offset += GST_ROUND_UP_2 (size);
1443 buf = gst_buffer_new ();
1446 if ((res = gst_riff_read_chunk (GST_ELEMENT_CAST (wav), wav->sinkpad,
1447 &wav->offset, &tag, &buf)) != GST_FLOW_OK)
1451 if (tag == GST_RIFF_TAG_JUNK || tag == GST_RIFF_TAG_JUNQ ||
1452 tag == GST_RIFF_TAG_bext || tag == GST_RIFF_TAG_BEXT ||
1453 tag == GST_RIFF_TAG_LIST || tag == GST_RIFF_TAG_ID32 ||
1454 tag == GST_RIFF_TAG_IDVX) {
1455 GST_DEBUG_OBJECT (wav, "skipping %" GST_FOURCC_FORMAT " chunk",
1456 GST_FOURCC_ARGS (tag));
1457 gst_buffer_unref (buf);
1462 if (tag != GST_RIFF_TAG_fmt)
1465 if (!(gst_riff_parse_strf_auds (GST_ELEMENT_CAST (wav), buf, &header,
1467 goto parse_header_error;
1469 buf = NULL; /* parse_strf_auds() took ownership of buffer */
1471 /* do sanity checks of header fields */
1472 if (header->channels == 0)
1474 if (header->rate == 0)
1477 GST_DEBUG_OBJECT (wav, "creating the caps");
1479 /* Note: gst_riff_create_audio_caps might need to fix values in
1480 * the header header depending on the format, so call it first */
1481 /* FIXME: Need to handle the channel reorder map */
1482 caps = gst_riff_create_audio_caps (header->format, NULL, header, extra,
1483 NULL, &codec_name, NULL);
1486 gst_buffer_unref (extra);
1489 goto unknown_format;
1491 /* do more sanity checks of header fields
1492 * (these can be sanitized by gst_riff_create_audio_caps()
1494 wav->format = header->format;
1495 wav->rate = header->rate;
1496 wav->channels = header->channels;
1497 wav->blockalign = header->blockalign;
1498 wav->depth = header->bits_per_sample;
1499 wav->av_bps = header->av_bps;
1505 /* do format specific handling */
1506 switch (wav->format) {
1507 case GST_RIFF_WAVE_FORMAT_MPEGL12:
1508 case GST_RIFF_WAVE_FORMAT_MPEGL3:
1510 /* Note: workaround for mp2/mp3 embedded in wav, that relies on the
1511 * bitrate inside the mpeg stream */
1512 GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps);
1516 case GST_RIFF_WAVE_FORMAT_PCM:
1517 if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8))
1518 goto invalid_blockalign;
1521 if (wav->av_bps > wav->blockalign * wav->rate)
1523 /* use the configured bps */
1524 wav->bps = wav->av_bps;
1528 wav->width = (wav->blockalign * 8) / wav->channels;
1529 wav->bytes_per_sample = wav->channels * wav->width / 8;
1531 if (wav->bytes_per_sample <= 0)
1532 goto no_bytes_per_sample;
1534 GST_DEBUG_OBJECT (wav, "blockalign = %u", (guint) wav->blockalign);
1535 GST_DEBUG_OBJECT (wav, "width = %u", (guint) wav->width);
1536 GST_DEBUG_OBJECT (wav, "depth = %u", (guint) wav->depth);
1537 GST_DEBUG_OBJECT (wav, "av_bps = %u", (guint) wav->av_bps);
1538 GST_DEBUG_OBJECT (wav, "frequency = %u", (guint) wav->rate);
1539 GST_DEBUG_OBJECT (wav, "channels = %u", (guint) wav->channels);
1540 GST_DEBUG_OBJECT (wav, "bytes_per_sample = %u", wav->bytes_per_sample);
1542 /* bps can be 0 when we don't have a valid bitrate (mostly for compressed
1543 * formats). This will make the element output a BYTE format segment and
1544 * will not timestamp the outgoing buffers.
1546 GST_DEBUG_OBJECT (wav, "bps = %u", (guint) wav->bps);
1548 GST_DEBUG_OBJECT (wav, "caps = %" GST_PTR_FORMAT, caps);
1550 /* create pad later so we can sniff the first few bytes
1551 * of the real data and correct our caps if necessary */
1552 gst_caps_replace (&wav->caps, caps);
1553 gst_caps_replace (&caps, NULL);
1555 wav->got_fmt = TRUE;
1558 wav->tags = gst_tag_list_new_empty ();
1560 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1561 GST_TAG_AUDIO_CODEC, codec_name, NULL);
1563 g_free (codec_name);
1569 gst_pad_peer_query_duration (wav->sinkpad, GST_FORMAT_BYTES, &upstream_size);
1570 GST_DEBUG_OBJECT (wav, "upstream size %" G_GUINT64_FORMAT, upstream_size);
1572 /* loop headers until we get data */
1574 if (wav->streaming) {
1575 if (!gst_wavparse_peek_chunk_info (wav, &tag, &size))
1582 gst_pad_pull_range (wav->sinkpad, wav->offset, 8,
1583 &buf)) != GST_FLOW_OK)
1584 goto header_read_error;
1585 gst_buffer_map (buf, &map, GST_MAP_READ);
1586 tag = GST_READ_UINT32_LE (map.data);
1587 size = GST_READ_UINT32_LE (map.data + 4);
1588 gst_buffer_unmap (buf, &map);
1591 GST_INFO_OBJECT (wav,
1592 "Got TAG: %" GST_FOURCC_FORMAT ", offset %" G_GUINT64_FORMAT,
1593 GST_FOURCC_ARGS (tag), wav->offset);
1595 /* wav is a st00pid format, we don't know for sure where data starts.
1596 * So we have to go bit by bit until we find the 'data' header
1599 case GST_RIFF_TAG_data:{
1600 GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size);
1601 if (wav->ignore_length) {
1602 GST_DEBUG_OBJECT (wav, "Ignoring length");
1605 if (wav->streaming) {
1606 gst_adapter_flush (wav->adapter, 8);
1609 gst_buffer_unref (buf);
1612 wav->datastart = wav->offset;
1613 /* If size is zero, then the data chunk probably actually extends to
1614 the end of the file */
1615 if (size == 0 && upstream_size) {
1616 size = upstream_size - wav->datastart;
1618 /* Or the file might be truncated */
1619 else if (upstream_size) {
1620 size = MIN (size, (upstream_size - wav->datastart));
1622 wav->datasize = (guint64) size;
1623 wav->dataleft = (guint64) size;
1624 wav->end_offset = size + wav->datastart;
1625 if (!wav->streaming) {
1626 /* We will continue parsing tags 'till end */
1627 wav->offset += size;
1629 GST_DEBUG_OBJECT (wav, "datasize = %u", size);
1632 case GST_RIFF_TAG_fact:{
1633 if (wav->format != GST_RIFF_WAVE_FORMAT_MPEGL12 &&
1634 wav->format != GST_RIFF_WAVE_FORMAT_MPEGL3) {
1635 const guint data_size = 4;
1637 GST_INFO_OBJECT (wav, "Have fact chunk");
1638 if (size < data_size) {
1639 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1640 /* need more data */
1643 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1647 /* number of samples (for compressed formats) */
1648 if (wav->streaming) {
1649 const guint8 *data = NULL;
1651 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1654 gst_adapter_flush (wav->adapter, 8);
1655 data = gst_adapter_map (wav->adapter, data_size);
1656 wav->fact = GST_READ_UINT32_LE (data);
1657 gst_adapter_unmap (wav->adapter);
1658 gst_adapter_flush (wav->adapter, GST_ROUND_UP_2 (size));
1660 gst_buffer_unref (buf);
1663 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1664 data_size, &buf)) != GST_FLOW_OK)
1665 goto header_read_error;
1666 gst_buffer_extract (buf, 0, &wav->fact, 4);
1667 wav->fact = GUINT32_FROM_LE (wav->fact);
1668 gst_buffer_unref (buf);
1670 GST_DEBUG_OBJECT (wav, "have fact %u", wav->fact);
1671 wav->offset += 8 + GST_ROUND_UP_2 (size);
1674 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1675 /* need more data */
1681 case GST_RIFF_TAG_acid:{
1682 const gst_riff_acid *acid = NULL;
1683 const guint data_size = sizeof (gst_riff_acid);
1686 GST_INFO_OBJECT (wav, "Have acid chunk");
1687 if (size < data_size) {
1688 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size)) {
1689 /* need more data */
1692 GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk",
1696 if (wav->streaming) {
1697 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1700 gst_adapter_flush (wav->adapter, 8);
1701 acid = (const gst_riff_acid *) gst_adapter_map (wav->adapter,
1703 tempo = acid->tempo;
1704 gst_adapter_unmap (wav->adapter);
1707 gst_buffer_unref (buf);
1710 gst_pad_pull_range (wav->sinkpad, wav->offset + 8,
1711 size, &buf)) != GST_FLOW_OK)
1712 goto header_read_error;
1713 gst_buffer_map (buf, &map, GST_MAP_READ);
1714 acid = (const gst_riff_acid *) map.data;
1715 tempo = acid->tempo;
1716 gst_buffer_unmap (buf, &map);
1718 /* send data as tags */
1720 wav->tags = gst_tag_list_new_empty ();
1721 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
1722 GST_TAG_BEATS_PER_MINUTE, tempo, NULL);
1724 size = GST_ROUND_UP_2 (size);
1725 if (wav->streaming) {
1726 gst_adapter_flush (wav->adapter, size);
1728 gst_buffer_unref (buf);
1730 wav->offset += 8 + size;
1733 /* FIXME: all list tags after data are ignored in streaming mode */
1734 case GST_RIFF_TAG_LIST:{
1737 if (wav->streaming) {
1738 const guint8 *data = NULL;
1740 if (gst_adapter_available (wav->adapter) < 12) {
1743 data = gst_adapter_map (wav->adapter, 12);
1744 ltag = GST_READ_UINT32_LE (data + 8);
1745 gst_adapter_unmap (wav->adapter);
1747 gst_buffer_unref (buf);
1750 gst_pad_pull_range (wav->sinkpad, wav->offset, 12,
1751 &buf)) != GST_FLOW_OK)
1752 goto header_read_error;
1753 gst_buffer_extract (buf, 8, <ag, 4);
1754 ltag = GUINT32_FROM_LE (ltag);
1757 case GST_RIFF_LIST_INFO:{
1758 const gint data_size = size - 4;
1761 GST_INFO_OBJECT (wav, "Have LIST chunk INFO size %u", data_size);
1762 if (wav->streaming) {
1763 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1766 gst_adapter_flush (wav->adapter, 12);
1768 if (data_size > 0) {
1769 buf = gst_adapter_take_buffer (wav->adapter, data_size);
1771 gst_adapter_flush (wav->adapter, 1);
1775 gst_buffer_unref (buf);
1777 if (data_size > 0) {
1779 gst_pad_pull_range (wav->sinkpad, wav->offset,
1780 data_size, &buf)) != GST_FLOW_OK)
1781 goto header_read_error;
1784 if (data_size > 0) {
1786 gst_riff_parse_info (GST_ELEMENT (wav), buf, &new);
1788 GstTagList *old = wav->tags;
1790 gst_tag_list_merge (old, new, GST_TAG_MERGE_REPLACE);
1792 gst_tag_list_unref (old);
1793 gst_tag_list_unref (new);
1795 gst_buffer_unref (buf);
1796 wav->offset += GST_ROUND_UP_2 (data_size);
1800 case GST_RIFF_LIST_adtl:{
1801 const gint data_size = size;
1803 GST_INFO_OBJECT (wav, "Have 'adtl' LIST, size %u", data_size);
1804 if (wav->streaming) {
1805 const guint8 *data = NULL;
1807 gst_adapter_flush (wav->adapter, 12);
1808 data = gst_adapter_map (wav->adapter, data_size);
1809 gst_wavparse_adtl_chunk (wav, data, data_size);
1810 gst_adapter_unmap (wav->adapter);
1814 gst_buffer_unref (buf);
1817 gst_pad_pull_range (wav->sinkpad, wav->offset + 12,
1818 data_size, &buf)) != GST_FLOW_OK)
1819 goto header_read_error;
1820 gst_buffer_map (buf, &map, GST_MAP_READ);
1821 gst_wavparse_adtl_chunk (wav, (const guint8 *) map.data,
1823 gst_buffer_unmap (buf, &map);
1827 GST_INFO_OBJECT (wav, "Ignoring LIST chunk %" GST_FOURCC_FORMAT,
1828 GST_FOURCC_ARGS (ltag));
1829 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1830 /* need more data */
1836 case GST_RIFF_TAG_cue:{
1837 const guint data_size = size;
1839 GST_DEBUG_OBJECT (wav, "Have 'cue' TAG, size : %u", data_size);
1840 if (wav->streaming) {
1841 const guint8 *data = NULL;
1843 if (!gst_wavparse_peek_chunk (wav, &tag, &size)) {
1846 gst_adapter_flush (wav->adapter, 8);
1848 data = gst_adapter_map (wav->adapter, data_size);
1849 if (!gst_wavparse_cue_chunk (wav, data, data_size)) {
1850 goto header_read_error;
1852 gst_adapter_unmap (wav->adapter);
1857 gst_buffer_unref (buf);
1860 gst_pad_pull_range (wav->sinkpad, wav->offset,
1861 data_size, &buf)) != GST_FLOW_OK)
1862 goto header_read_error;
1863 gst_buffer_map (buf, &map, GST_MAP_READ);
1864 if (!gst_wavparse_cue_chunk (wav, (const guint8 *) map.data,
1866 goto header_read_error;
1868 gst_buffer_unmap (buf, &map);
1870 size = GST_ROUND_UP_2 (size);
1871 if (wav->streaming) {
1872 gst_adapter_flush (wav->adapter, size);
1874 gst_buffer_unref (buf);
1876 size = GST_ROUND_UP_2 (size);
1877 wav->offset += size;
1881 if (!gst_waveparse_ignore_chunk (wav, buf, tag, size))
1882 /* need more data */
1887 if (upstream_size && (wav->offset >= upstream_size)) {
1888 /* Now we are gone through the whole file */
1893 GST_DEBUG_OBJECT (wav, "Finished parsing headers");
1895 if (wav->bps <= 0 && wav->fact) {
1897 /* not a good idea, as for embedded mp2/mp3 we set bps to 0 earlier */
1899 (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize,
1900 (guint64) wav->fact);
1901 GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps);
1906 if (gst_wavparse_calculate_duration (wav)) {
1907 gst_segment_init (&wav->segment, GST_FORMAT_TIME);
1908 if (!wav->ignore_length)
1909 wav->segment.duration = wav->duration;
1911 gst_wavparse_create_toc (wav);
1913 /* no bitrate, let downstream peer do the math, we'll feed it bytes. */
1914 gst_segment_init (&wav->segment, GST_FORMAT_BYTES);
1915 if (!wav->ignore_length)
1916 wav->segment.duration = wav->datasize;
1919 /* now we have all the info to perform a pending seek if any, if no
1920 * event, this will still do the right thing and it will also send
1921 * the right newsegment event downstream. */
1922 gst_wavparse_perform_seek (wav, wav->seek_event);
1923 /* remove pending event */
1924 event_p = &wav->seek_event;
1925 gst_event_replace (event_p, NULL);
1927 /* we just started, we are discont */
1928 wav->discont = TRUE;
1930 wav->state = GST_WAVPARSE_DATA;
1932 /* determine reasonable max buffer size,
1933 * that is, buffers not too small either size or time wise
1934 * so we do not end up with too many of them */
1937 gst_wavparse_time_to_bytepos (wav, 40 * GST_MSECOND, &upstream_size);
1938 wav->max_buf_size = upstream_size;
1939 wav->max_buf_size = MAX (wav->max_buf_size, MAX_BUFFER_SIZE);
1940 if (wav->blockalign > 0)
1941 wav->max_buf_size -= (wav->max_buf_size % wav->blockalign);
1943 GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size);
1951 g_free (codec_name);
1955 gst_caps_unref (caps);
1960 res = GST_FLOW_ERROR;
1965 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
1966 ("Invalid WAV header (no fmt at start): %"
1967 GST_FOURCC_FORMAT, GST_FOURCC_ARGS (tag)));
1972 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
1973 ("Couldn't parse audio header"));
1978 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1979 ("Stream claims to contain no channels - invalid data"));
1984 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1985 ("Stream with sample_rate == 0 - invalid data"));
1990 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1991 ("Stream claims blockalign = %u, which is more than %u - invalid data",
1992 wav->blockalign, wav->channels * ((wav->depth + 7) / 8)));
1997 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
1998 ("Stream claims av_bsp = %u, which is more than %u - invalid data",
1999 wav->av_bps, wav->blockalign * wav->rate));
2002 no_bytes_per_sample:
2004 GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL),
2005 ("Could not caluclate bytes per sample - invalid data"));
2010 GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL),
2011 ("No caps found for format 0x%x, %u channels, %u Hz",
2012 wav->format, wav->channels, wav->rate));
2017 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL),
2018 ("Couldn't read in header %d (%s)", res, gst_flow_get_name (res)));
2024 * Read WAV file tag when streaming
2026 static GstFlowReturn
2027 gst_wavparse_parse_stream_init (GstWavParse * wav)
2029 if (gst_adapter_available (wav->adapter) >= 12) {
2032 /* _take flushes the data */
2033 tmp = gst_adapter_take_buffer (wav->adapter, 12);
2035 GST_DEBUG ("Parsing wav header");
2036 if (!gst_wavparse_parse_file_header (GST_ELEMENT_CAST (wav), tmp))
2037 return GST_FLOW_ERROR;
2040 /* Go to next state */
2041 wav->state = GST_WAVPARSE_HEADER;
2046 /* handle an event sent directly to the element.
2048 * This event can be sent either in the READY state or the
2049 * >READY state. The only event of interest really is the seek
2052 * In the READY state we can only store the event and try to
2053 * respect it when going to PAUSED. We assume we are in the
2054 * READY state when our parsing state != GST_WAVPARSE_DATA.
2056 * When we are steaming, we can simply perform the seek right
2060 gst_wavparse_send_event (GstElement * element, GstEvent * event)
2062 GstWavParse *wav = GST_WAVPARSE (element);
2063 gboolean res = FALSE;
2066 GST_DEBUG_OBJECT (wav, "received event %s", GST_EVENT_TYPE_NAME (event));
2068 switch (GST_EVENT_TYPE (event)) {
2069 case GST_EVENT_SEEK:
2070 if (wav->state == GST_WAVPARSE_DATA) {
2071 /* we can handle the seek directly when streaming data */
2072 res = gst_wavparse_perform_seek (wav, event);
2074 GST_DEBUG_OBJECT (wav, "queuing seek for later");
2076 event_p = &wav->seek_event;
2077 gst_event_replace (event_p, event);
2079 /* we always return true */
2086 gst_event_unref (event);
2091 gst_wavparse_have_dts_caps (const GstCaps * caps, GstTypeFindProbability prob)
2095 s = gst_caps_get_structure (caps, 0);
2096 if (!gst_structure_has_name (s, "audio/x-dts"))
2098 if (prob >= GST_TYPE_FIND_LIKELY)
2100 /* DTS at non-0 offsets and without second sync may yield POSSIBLE .. */
2101 if (prob < GST_TYPE_FIND_POSSIBLE)
2103 /* .. in which case we want at least a valid-looking rate and channels */
2104 if (!gst_structure_has_field (s, "channels"))
2106 /* and for extra assurance we could also check the rate from the DTS frame
2107 * against the one in the wav header, but for now let's not do that */
2108 return gst_structure_has_field (s, "rate");
2112 gst_wavparse_add_src_pad (GstWavParse * wav, GstBuffer * buf)
2116 GST_DEBUG_OBJECT (wav, "adding src pad");
2118 g_assert (wav->caps != NULL);
2120 s = gst_caps_get_structure (wav->caps, 0);
2121 if (s && gst_structure_has_name (s, "audio/x-raw") && buf != NULL) {
2122 GstTypeFindProbability prob;
2125 tf_caps = gst_type_find_helper_for_buffer (GST_OBJECT (wav), buf, &prob);
2126 if (tf_caps != NULL) {
2127 GST_LOG ("typefind caps = %" GST_PTR_FORMAT ", P=%d", tf_caps, prob);
2128 if (gst_wavparse_have_dts_caps (tf_caps, prob)) {
2129 GST_INFO_OBJECT (wav, "Found DTS marker in file marked as raw PCM");
2130 gst_caps_unref (wav->caps);
2131 wav->caps = tf_caps;
2133 gst_tag_list_add (wav->tags, GST_TAG_MERGE_REPLACE,
2134 GST_TAG_AUDIO_CODEC, "dts", NULL);
2136 GST_DEBUG_OBJECT (wav, "found caps %" GST_PTR_FORMAT " for stream "
2137 "marked as raw PCM audio, but ignoring for now", tf_caps);
2138 gst_caps_unref (tf_caps);
2143 gst_pad_set_caps (wav->srcpad, wav->caps);
2144 gst_caps_replace (&wav->caps, NULL);
2146 if (wav->start_segment) {
2147 GST_DEBUG_OBJECT (wav, "Send start segment event on newpad");
2148 gst_pad_push_event (wav->srcpad, wav->start_segment);
2149 wav->start_segment = NULL;
2153 gst_pad_push_event (wav->srcpad, gst_event_new_tag (wav->tags));
2158 static GstFlowReturn
2159 gst_wavparse_stream_data (GstWavParse * wav)
2161 GstBuffer *buf = NULL;
2162 GstFlowReturn res = GST_FLOW_OK;
2163 guint64 desired, obtained;
2164 GstClockTime timestamp, next_timestamp, duration;
2165 guint64 pos, nextpos;
2168 GST_LOG_OBJECT (wav,
2169 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT " , dataleft: %"
2170 G_GINT64_FORMAT, wav->offset, wav->end_offset, wav->dataleft);
2172 /* Get the next n bytes and output them */
2173 if (wav->dataleft == 0 || wav->dataleft < wav->blockalign)
2176 /* scale the amount of data by the segment rate so we get equal
2177 * amounts of data regardless of the playback rate */
2179 MIN (gst_guint64_to_gdouble (wav->dataleft),
2180 wav->max_buf_size * ABS (wav->segment.rate));
2182 if (desired >= wav->blockalign && wav->blockalign > 0)
2183 desired -= (desired % wav->blockalign);
2185 GST_LOG_OBJECT (wav, "Fetching %" G_GINT64_FORMAT " bytes of data "
2186 "from the sinkpad", desired);
2188 if (wav->streaming) {
2189 guint avail = gst_adapter_available (wav->adapter);
2192 /* flush some bytes if evil upstream sends segment that starts
2193 * before data or does is not send sample aligned segment */
2194 if (G_LIKELY (wav->offset >= wav->datastart)) {
2195 extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
2197 extra = wav->datastart - wav->offset;
2200 if (G_UNLIKELY (extra)) {
2201 extra = wav->bytes_per_sample - extra;
2202 if (extra <= avail) {
2203 GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra);
2204 gst_adapter_flush (wav->adapter, extra);
2205 wav->offset += extra;
2206 wav->dataleft -= extra;
2207 goto iterate_adapter;
2209 GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail);
2210 gst_adapter_clear (wav->adapter);
2211 wav->offset += avail;
2212 wav->dataleft -= avail;
2217 if (avail < desired) {
2218 GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail);
2222 buf = gst_adapter_take_buffer (wav->adapter, desired);
2224 if ((res = gst_pad_pull_range (wav->sinkpad, wav->offset,
2225 desired, &buf)) != GST_FLOW_OK)
2228 /* we may get a short buffer at the end of the file */
2229 if (gst_buffer_get_size (buf) < desired) {
2230 gsize size = gst_buffer_get_size (buf);
2232 GST_LOG_OBJECT (wav, "Got only %" G_GSIZE_FORMAT " bytes of data", size);
2233 if (size >= wav->blockalign) {
2234 buf = gst_buffer_make_writable (buf);
2235 gst_buffer_resize (buf, 0, size - (size % wav->blockalign));
2237 gst_buffer_unref (buf);
2243 obtained = gst_buffer_get_size (buf);
2245 /* our positions in bytes */
2246 pos = wav->offset - wav->datastart;
2247 nextpos = pos + obtained;
2249 /* update offsets, does not overflow. */
2250 buf = gst_buffer_make_writable (buf);
2251 GST_BUFFER_OFFSET (buf) = pos / wav->bytes_per_sample;
2252 GST_BUFFER_OFFSET_END (buf) = nextpos / wav->bytes_per_sample;
2254 /* first chunk of data? create the source pad. We do this only here so
2255 * we can detect broken .wav files with dts disguised as raw PCM (sigh) */
2256 if (G_UNLIKELY (wav->first)) {
2258 /* this will also push the segment events */
2259 gst_wavparse_add_src_pad (wav, buf);
2261 /* If we have a pending start segment, send it now. */
2262 if (G_UNLIKELY (wav->start_segment != NULL)) {
2263 gst_pad_push_event (wav->srcpad, wav->start_segment);
2264 wav->start_segment = NULL;
2269 /* and timestamps if we have a bitrate, be careful for overflows */
2271 gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps);
2273 gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps);
2274 duration = next_timestamp - timestamp;
2276 /* update current running segment position */
2277 if (G_LIKELY (next_timestamp >= wav->segment.start))
2278 wav->segment.position = next_timestamp;
2279 } else if (wav->fact) {
2281 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2282 /* and timestamps if we have a bitrate, be careful for overflows */
2283 timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps);
2284 next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps);
2285 duration = next_timestamp - timestamp;
2287 /* no bitrate, all we know is that the first sample has timestamp 0, all
2288 * other positions and durations have unknown timestamp. */
2292 timestamp = GST_CLOCK_TIME_NONE;
2293 duration = GST_CLOCK_TIME_NONE;
2294 /* update current running segment position with byte offset */
2295 if (G_LIKELY (nextpos >= wav->segment.start))
2296 wav->segment.position = nextpos;
2298 if ((pos > 0) && wav->vbr) {
2299 /* don't set timestamps for VBR files if it's not the first buffer */
2300 timestamp = GST_CLOCK_TIME_NONE;
2301 duration = GST_CLOCK_TIME_NONE;
2304 GST_DEBUG_OBJECT (wav, "marking DISCONT");
2305 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
2306 wav->discont = FALSE;
2309 GST_BUFFER_TIMESTAMP (buf) = timestamp;
2310 GST_BUFFER_DURATION (buf) = duration;
2312 GST_LOG_OBJECT (wav,
2313 "Got buffer. timestamp:%" GST_TIME_FORMAT " , duration:%" GST_TIME_FORMAT
2314 ", size:%" G_GSIZE_FORMAT, GST_TIME_ARGS (timestamp),
2315 GST_TIME_ARGS (duration), gst_buffer_get_size (buf));
2317 if ((res = gst_pad_push (wav->srcpad, buf)) != GST_FLOW_OK)
2320 if (obtained < wav->dataleft) {
2321 wav->offset += obtained;
2322 wav->dataleft -= obtained;
2324 wav->offset += wav->dataleft;
2328 /* Iterate until need more data, so adapter size won't grow */
2329 if (wav->streaming) {
2330 GST_LOG_OBJECT (wav,
2331 "offset: %" G_GINT64_FORMAT " , end: %" G_GINT64_FORMAT, wav->offset,
2333 goto iterate_adapter;
2340 GST_DEBUG_OBJECT (wav, "found EOS");
2341 return GST_FLOW_EOS;
2345 /* check if we got EOS */
2346 if (res == GST_FLOW_EOS)
2349 GST_WARNING_OBJECT (wav,
2350 "Error getting %" G_GINT64_FORMAT " bytes from the "
2351 "sinkpad (dataleft = %" G_GINT64_FORMAT ")", desired, wav->dataleft);
2356 GST_INFO_OBJECT (wav,
2357 "Error pushing on srcpad %s:%s, reason %s, is linked? = %d",
2358 GST_DEBUG_PAD_NAME (wav->srcpad), gst_flow_get_name (res),
2359 gst_pad_is_linked (wav->srcpad));
2365 gst_wavparse_loop (GstPad * pad)
2368 GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
2370 GST_LOG_OBJECT (wav, "process data");
2372 switch (wav->state) {
2373 case GST_WAVPARSE_START:
2374 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2375 if ((ret = gst_wavparse_stream_init (wav)) != GST_FLOW_OK)
2378 wav->state = GST_WAVPARSE_HEADER;
2381 case GST_WAVPARSE_HEADER:
2382 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2383 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2386 wav->state = GST_WAVPARSE_DATA;
2387 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2390 case GST_WAVPARSE_DATA:
2391 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2395 g_assert_not_reached ();
2402 const gchar *reason = gst_flow_get_name (ret);
2404 GST_DEBUG_OBJECT (wav, "pausing task, reason %s", reason);
2405 gst_pad_pause_task (pad);
2407 if (ret == GST_FLOW_EOS) {
2408 /* handle end-of-stream/segment */
2409 /* so align our position with the end of it, if there is one
2410 * this ensures a subsequent will arrive at correct base/acc time */
2411 if (wav->segment.format == GST_FORMAT_TIME) {
2412 if (wav->segment.rate > 0.0 &&
2413 GST_CLOCK_TIME_IS_VALID (wav->segment.stop))
2414 wav->segment.position = wav->segment.stop;
2415 else if (wav->segment.rate < 0.0)
2416 wav->segment.position = wav->segment.start;
2418 if (wav->state == GST_WAVPARSE_START) {
2419 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2420 ("No valid input found before end of stream"));
2421 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2423 /* add pad before we perform EOS */
2424 if (G_UNLIKELY (wav->first)) {
2426 gst_wavparse_add_src_pad (wav, NULL);
2429 /* perform EOS logic */
2430 if (wav->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2433 if ((stop = wav->segment.stop) == -1)
2434 stop = wav->segment.duration;
2436 gst_element_post_message (GST_ELEMENT_CAST (wav),
2437 gst_message_new_segment_done (GST_OBJECT_CAST (wav),
2438 wav->segment.format, stop));
2439 gst_pad_push_event (wav->srcpad,
2440 gst_event_new_segment_done (wav->segment.format, stop));
2442 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2445 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
2446 /* for fatal errors we post an error message, post the error
2447 * first so the app knows about the error first. */
2448 GST_ELEMENT_ERROR (wav, STREAM, FAILED,
2449 (_("Internal data flow error.")),
2450 ("streaming task paused, reason %s (%d)", reason, ret));
2451 gst_pad_push_event (wav->srcpad, gst_event_new_eos ());
2457 static GstFlowReturn
2458 gst_wavparse_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
2461 GstWavParse *wav = GST_WAVPARSE (parent);
2463 GST_LOG_OBJECT (wav, "adapter_push %" G_GSIZE_FORMAT " bytes",
2464 gst_buffer_get_size (buf));
2466 gst_adapter_push (wav->adapter, buf);
2468 switch (wav->state) {
2469 case GST_WAVPARSE_START:
2470 GST_INFO_OBJECT (wav, "GST_WAVPARSE_START");
2471 if ((ret = gst_wavparse_parse_stream_init (wav)) != GST_FLOW_OK)
2474 if (wav->state != GST_WAVPARSE_HEADER)
2477 /* otherwise fall-through */
2478 case GST_WAVPARSE_HEADER:
2479 GST_INFO_OBJECT (wav, "GST_WAVPARSE_HEADER");
2480 if ((ret = gst_wavparse_stream_headers (wav)) != GST_FLOW_OK)
2483 if (!wav->got_fmt || wav->datastart == 0)
2486 wav->state = GST_WAVPARSE_DATA;
2487 GST_INFO_OBJECT (wav, "GST_WAVPARSE_DATA");
2490 case GST_WAVPARSE_DATA:
2491 if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
2492 wav->discont = TRUE;
2493 if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
2497 g_return_val_if_reached (GST_FLOW_ERROR);
2500 if (G_UNLIKELY (wav->abort_buffering)) {
2501 wav->abort_buffering = FALSE;
2502 ret = GST_FLOW_ERROR;
2503 /* sort of demux/parse error */
2504 GST_ELEMENT_ERROR (wav, STREAM, DEMUX, (NULL), ("unhandled buffer size"));
2510 static GstFlowReturn
2511 gst_wavparse_flush_data (GstWavParse * wav)
2513 GstFlowReturn ret = GST_FLOW_OK;
2516 if ((av = gst_adapter_available (wav->adapter)) > 0) {
2518 wav->end_offset = wav->offset + av;
2519 ret = gst_wavparse_stream_data (wav);
2526 gst_wavparse_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
2528 GstWavParse *wav = GST_WAVPARSE (parent);
2529 gboolean ret = TRUE;
2531 GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
2533 switch (GST_EVENT_TYPE (event)) {
2534 case GST_EVENT_CAPS:
2536 /* discard, we'll come up with proper src caps */
2537 gst_event_unref (event);
2540 case GST_EVENT_SEGMENT:
2542 gint64 start, stop, offset = 0, end_offset = -1;
2545 /* some debug output */
2546 gst_event_copy_segment (event, &segment);
2547 GST_DEBUG_OBJECT (wav, "received newsegment %" GST_SEGMENT_FORMAT,
2550 if (wav->state != GST_WAVPARSE_DATA) {
2551 GST_DEBUG_OBJECT (wav, "still starting, eating event");
2555 /* now we are either committed to TIME or BYTE format,
2556 * and we only expect a BYTE segment, e.g. following a seek */
2557 if (segment.format == GST_FORMAT_BYTES) {
2558 /* handle (un)signed issues */
2559 start = segment.start;
2560 stop = segment.stop;
2563 start -= wav->datastart;
2564 start = MAX (start, 0);
2568 segment.stop -= wav->datastart;
2569 segment.stop = MAX (stop, 0);
2571 if (wav->segment.format == GST_FORMAT_TIME) {
2572 guint64 bps = wav->bps;
2574 /* operating in format TIME, so we can convert */
2575 if (!bps && wav->fact)
2577 gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
2581 gst_util_uint64_scale_ceil (start, GST_SECOND,
2582 (guint64) wav->bps);
2585 gst_util_uint64_scale_ceil (stop, GST_SECOND,
2586 (guint64) wav->bps);
2590 GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
2594 segment.start = start;
2595 segment.stop = stop;
2597 /* accept upstream's notion of segment and distribute along */
2598 segment.format = wav->segment.format;
2599 segment.time = segment.position = segment.start;
2600 segment.duration = wav->segment.duration;
2601 segment.base = gst_segment_to_running_time (&wav->segment,
2602 GST_FORMAT_TIME, wav->segment.position);
2604 gst_segment_copy_into (&segment, &wav->segment);
2606 /* also store the newsegment event for the streaming thread */
2607 if (wav->start_segment)
2608 gst_event_unref (wav->start_segment);
2609 GST_DEBUG_OBJECT (wav, "Storing newseg %" GST_SEGMENT_FORMAT, &segment);
2610 wav->start_segment = gst_event_new_segment (&segment);
2612 /* stream leftover data in current segment */
2613 gst_wavparse_flush_data (wav);
2614 /* and set up streaming thread for next one */
2615 wav->offset = offset;
2616 wav->end_offset = end_offset;
2617 if (wav->end_offset > 0) {
2618 wav->dataleft = wav->end_offset - wav->offset;
2620 /* infinity; upstream will EOS when done */
2621 wav->dataleft = G_MAXUINT64;
2624 gst_event_unref (event);
2628 if (wav->state == GST_WAVPARSE_START) {
2629 GST_ELEMENT_ERROR (wav, STREAM, WRONG_TYPE, (NULL),
2630 ("No valid input found before end of stream"));
2632 /* add pad if needed so EOS is seen downstream */
2633 if (G_UNLIKELY (wav->first)) {
2635 gst_wavparse_add_src_pad (wav, NULL);
2637 /* stream leftover data in current segment */
2638 gst_wavparse_flush_data (wav);
2643 case GST_EVENT_FLUSH_STOP:
2647 gst_adapter_clear (wav->adapter);
2648 wav->discont = TRUE;
2649 dur = wav->segment.duration;
2650 gst_segment_init (&wav->segment, wav->segment.format);
2651 wav->segment.duration = dur;
2655 ret = gst_pad_event_default (wav->sinkpad, parent, event);
2663 /* convert and query stuff */
2664 static const GstFormat *
2665 gst_wavparse_get_formats (GstPad * pad)
2667 static GstFormat formats[] = {
2670 GST_FORMAT_DEFAULT, /* a "frame", ie a set of samples per Hz */
2679 gst_wavparse_pad_convert (GstPad * pad,
2680 GstFormat src_format, gint64 src_value,
2681 GstFormat * dest_format, gint64 * dest_value)
2683 GstWavParse *wavparse;
2684 gboolean res = TRUE;
2686 wavparse = GST_WAVPARSE (GST_PAD_PARENT (pad));
2688 if (*dest_format == src_format) {
2689 *dest_value = src_value;
2693 if ((wavparse->bps == 0) && !wavparse->fact)
2696 GST_INFO_OBJECT (wavparse, "converting value from %s to %s",
2697 gst_format_get_name (src_format), gst_format_get_name (*dest_format));
2699 switch (src_format) {
2700 case GST_FORMAT_BYTES:
2701 switch (*dest_format) {
2702 case GST_FORMAT_DEFAULT:
2703 *dest_value = src_value / wavparse->bytes_per_sample;
2704 /* make sure we end up on a sample boundary */
2705 *dest_value -= *dest_value % wavparse->bytes_per_sample;
2707 case GST_FORMAT_TIME:
2708 /* src_value + datastart = offset */
2709 GST_INFO_OBJECT (wavparse,
2710 "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value,
2712 if (wavparse->bps > 0)
2713 *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND,
2714 (guint64) wavparse->bps);
2715 else if (wavparse->fact) {
2716 guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize,
2717 wavparse->rate, wavparse->fact);
2720 gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps);
2731 case GST_FORMAT_DEFAULT:
2732 switch (*dest_format) {
2733 case GST_FORMAT_BYTES:
2734 *dest_value = src_value * wavparse->bytes_per_sample;
2736 case GST_FORMAT_TIME:
2737 *dest_value = gst_util_uint64_scale (src_value, GST_SECOND,
2738 (guint64) wavparse->rate);
2746 case GST_FORMAT_TIME:
2747 switch (*dest_format) {
2748 case GST_FORMAT_BYTES:
2749 if (wavparse->bps > 0)
2750 *dest_value = gst_util_uint64_scale (src_value,
2751 (guint64) wavparse->bps, GST_SECOND);
2753 guint64 bps = gst_util_uint64_scale_int (wavparse->datasize,
2754 wavparse->rate, wavparse->fact);
2756 *dest_value = gst_util_uint64_scale (src_value, bps, GST_SECOND);
2758 /* make sure we end up on a sample boundary */
2759 *dest_value -= *dest_value % wavparse->blockalign;
2761 case GST_FORMAT_DEFAULT:
2762 *dest_value = gst_util_uint64_scale (src_value,
2763 (guint64) wavparse->rate, GST_SECOND);
2782 GST_DEBUG_OBJECT (wavparse, "bps 0 or no fact chunk, cannot convert");
2788 /* handle queries for location and length in requested format */
2790 gst_wavparse_pad_query (GstPad * pad, GstObject * parent, GstQuery * query)
2792 gboolean res = TRUE;
2793 GstWavParse *wav = GST_WAVPARSE (parent);
2795 /* only if we know */
2796 if (wav->state != GST_WAVPARSE_DATA) {
2800 GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
2802 switch (GST_QUERY_TYPE (query)) {
2803 case GST_QUERY_POSITION:
2809 /* this is not very precise, as we have pushed severla buffer upstream for prerolling */
2810 curb = wav->offset - wav->datastart;
2811 gst_query_parse_position (query, &format, NULL);
2812 GST_INFO_OBJECT (wav, "pos query at %" G_GINT64_FORMAT, curb);
2815 case GST_FORMAT_BYTES:
2816 format = GST_FORMAT_BYTES;
2820 res = gst_wavparse_pad_convert (pad, GST_FORMAT_BYTES, curb,
2825 gst_query_set_position (query, format, cur);
2828 case GST_QUERY_DURATION:
2830 gint64 duration = 0;
2833 if (wav->ignore_length) {
2838 gst_query_parse_duration (query, &format, NULL);
2841 case GST_FORMAT_BYTES:{
2842 format = GST_FORMAT_BYTES;
2843 duration = wav->datasize;
2846 case GST_FORMAT_TIME:
2847 if ((res = gst_wavparse_calculate_duration (wav))) {
2848 duration = wav->duration;
2856 gst_query_set_duration (query, format, duration);
2859 case GST_QUERY_CONVERT:
2861 gint64 srcvalue, dstvalue;
2862 GstFormat srcformat, dstformat;
2864 gst_query_parse_convert (query, &srcformat, &srcvalue,
2865 &dstformat, &dstvalue);
2866 res = gst_wavparse_pad_convert (pad, srcformat, srcvalue,
2867 &dstformat, &dstvalue);
2869 gst_query_set_convert (query, srcformat, srcvalue, dstformat, dstvalue);
2872 case GST_QUERY_SEEKING:{
2874 gboolean seekable = FALSE;
2876 gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
2877 if (fmt == wav->segment.format) {
2878 if (wav->streaming) {
2881 q = gst_query_new_seeking (GST_FORMAT_BYTES);
2882 if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
2883 gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
2884 GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
2886 gst_query_unref (q);
2888 GST_LOG_OBJECT (wav, "looping => seekable");
2892 } else if (fmt == GST_FORMAT_TIME) {
2896 gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
2901 res = gst_pad_query_default (pad, parent, query);
2908 gst_wavparse_srcpad_event (GstPad * pad, GstObject * parent, GstEvent * event)
2910 GstWavParse *wavparse = GST_WAVPARSE (parent);
2911 gboolean res = FALSE;
2913 GST_DEBUG_OBJECT (wavparse, "%s event", GST_EVENT_TYPE_NAME (event));
2915 switch (GST_EVENT_TYPE (event)) {
2916 case GST_EVENT_SEEK:
2917 /* can only handle events when we are in the data state */
2918 if (wavparse->state == GST_WAVPARSE_DATA) {
2919 res = gst_wavparse_perform_seek (wavparse, event);
2921 gst_event_unref (event);
2924 case GST_EVENT_TOC_SELECT:
2927 GstTocEntry *entry = NULL;
2928 GstEvent *seek_event;
2931 if (!wavparse->toc) {
2932 GST_DEBUG_OBJECT (wavparse, "no TOC to select");
2935 gst_event_parse_toc_select (event, &uid);
2937 GST_OBJECT_LOCK (wavparse);
2938 entry = gst_toc_find_entry (wavparse->toc, uid);
2939 if (entry == NULL) {
2940 GST_OBJECT_UNLOCK (wavparse);
2941 GST_WARNING_OBJECT (wavparse, "no TOC entry with given UID: %s",
2945 gst_toc_entry_get_start_stop_times (entry, &start_pos, NULL);
2946 GST_OBJECT_UNLOCK (wavparse);
2947 seek_event = gst_event_new_seek (1.0,
2949 GST_SEEK_FLAG_FLUSH,
2950 GST_SEEK_TYPE_SET, start_pos, GST_SEEK_TYPE_SET, -1);
2951 res = gst_wavparse_perform_seek (wavparse, seek_event);
2952 gst_event_unref (seek_event);
2956 GST_WARNING_OBJECT (wavparse, "received empty TOC select event");
2960 gst_event_unref (event);
2965 res = gst_pad_push_event (wavparse->sinkpad, event);
2972 gst_wavparse_sink_activate (GstPad * sinkpad, GstObject * parent)
2974 GstWavParse *wav = GST_WAVPARSE (parent);
2979 gst_adapter_clear (wav->adapter);
2980 g_object_unref (wav->adapter);
2981 wav->adapter = NULL;
2984 query = gst_query_new_scheduling ();
2986 if (!gst_pad_peer_query (sinkpad, query)) {
2987 gst_query_unref (query);
2991 pull_mode = gst_query_has_scheduling_mode_with_flags (query,
2992 GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
2993 gst_query_unref (query);
2998 GST_DEBUG_OBJECT (sinkpad, "activating pull");
2999 wav->streaming = FALSE;
3000 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
3004 GST_DEBUG_OBJECT (sinkpad, "activating push");
3005 wav->streaming = TRUE;
3006 wav->adapter = gst_adapter_new ();
3007 return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
3013 gst_wavparse_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
3014 GstPadMode mode, gboolean active)
3019 case GST_PAD_MODE_PUSH:
3022 case GST_PAD_MODE_PULL:
3024 /* if we have a scheduler we can start the task */
3025 res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_wavparse_loop,
3028 res = gst_pad_stop_task (sinkpad);
3038 static GstStateChangeReturn
3039 gst_wavparse_change_state (GstElement * element, GstStateChange transition)
3041 GstStateChangeReturn ret;
3042 GstWavParse *wav = GST_WAVPARSE (element);
3044 switch (transition) {
3045 case GST_STATE_CHANGE_NULL_TO_READY:
3047 case GST_STATE_CHANGE_READY_TO_PAUSED:
3048 gst_wavparse_reset (wav);
3050 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
3056 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
3058 switch (transition) {
3059 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
3061 case GST_STATE_CHANGE_PAUSED_TO_READY:
3062 gst_wavparse_reset (wav);
3064 case GST_STATE_CHANGE_READY_TO_NULL:
3073 gst_wavparse_set_property (GObject * object, guint prop_id,
3074 const GValue * value, GParamSpec * pspec)
3078 g_return_if_fail (GST_IS_WAVPARSE (object));
3079 self = GST_WAVPARSE (object);
3082 case PROP_IGNORE_LENGTH:
3083 self->ignore_length = g_value_get_boolean (value);
3086 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3092 gst_wavparse_get_property (GObject * object, guint prop_id,
3093 GValue * value, GParamSpec * pspec)
3097 g_return_if_fail (GST_IS_WAVPARSE (object));
3098 self = GST_WAVPARSE (object);
3101 case PROP_IGNORE_LENGTH:
3102 g_value_set_boolean (value, self->ignore_length);
3105 G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec);
3110 plugin_init (GstPlugin * plugin)
3114 return gst_element_register (plugin, "wavparse", GST_RANK_PRIMARY,
3118 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
3121 "Parse a .wav file into raw audio",
3122 plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)