1 /* GStreamer wavpack plugin
2 * (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
3 * (c) 2006 Tim-Philipp Müller <tim centricular net>
5 * gstwavpackparse.c: wavpack file parser
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
20 * Boston, MA 02111-1307, USA.
28 #include <wavpack/wavpack.h>
29 #include "gstwavpackparse.h"
30 #include "gstwavpackcommon.h"
32 GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug);
33 #define GST_CAT_DEFAULT gst_wavpack_parse_debug
35 static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
38 GST_STATIC_CAPS ("audio/x-wavpack, "
39 "framed = (boolean) false; "
40 "audio/x-wavpack-correction, " "framed = (boolean) false")
43 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
46 GST_STATIC_CAPS ("audio/x-wavpack, "
47 "width = (int) { 8, 16, 24, 32 }, "
48 "channels = (int) { 1, 2 }, "
49 "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
52 static GstStaticPadTemplate wvc_src_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
55 GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
58 static gboolean gst_wavepack_parse_sink_activate (GstPad * sinkpad);
60 gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active);
62 static void gst_wavpack_parse_loop (GstElement * element);
63 static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement *
64 element, GstStateChange transition);
65 static void gst_wavpack_parse_reset (GstWavpackParse * wavpackparse);
66 static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wvparse);
67 static GstBuffer *gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse,
68 gint64 offset, guint size, GstFlowReturn * flow);
70 GST_BOILERPLATE (GstWavpackParse, gst_wavpack_parse, GstElement,
73 static void gst_wavpack_parse_base_init (gpointer klass)
75 static GstElementDetails plugin_details = {
76 "Wavpack file parser",
77 "Codec/Demuxer/Audio",
78 "Parses Wavpack files",
79 "Arwed v. Merkatz <v.merkatz@gmx.net>"
81 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
83 gst_element_class_add_pad_template (element_class,
84 gst_static_pad_template_get (&src_factory));
85 gst_element_class_add_pad_template (element_class,
86 gst_static_pad_template_get (&wvc_src_factory));
87 gst_element_class_add_pad_template (element_class,
88 gst_static_pad_template_get (&sink_factory));
89 gst_element_class_set_details (element_class, &plugin_details);
93 gst_wavpack_parse_dispose (GObject * object)
95 gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object));
96 G_OBJECT_CLASS (parent_class)->dispose (object);
100 gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
102 GObjectClass *gobject_class;
103 GstElementClass *gstelement_class;
105 gobject_class = (GObjectClass *) klass;
106 gstelement_class = (GstElementClass *) klass;
108 gobject_class->dispose = gst_wavpack_parse_dispose;
109 gstelement_class->change_state =
110 GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state);
113 static GstWavpackParseIndexEntry *
114 gst_wavpack_parse_index_get_last_entry (GstWavpackParse * wvparse)
118 g_assert (wvparse->entries != NULL);
119 g_assert (wvparse->entries->len > 0);
121 last = wvparse->entries->len - 1;
122 return &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, last);
125 static GstWavpackParseIndexEntry *
126 gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse,
127 gint64 sample_offset)
131 if (wvparse->entries == NULL || wvparse->entries->len == 0)
134 for (i = wvparse->entries->len - 1; i >= 0; --i) {
135 GstWavpackParseIndexEntry *entry;
137 entry = &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, i);
139 GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @"
140 " byte %" G_GINT64_FORMAT, entry->sample_offset, entry->byte_offset);
142 if (entry->sample_offset <= sample_offset &&
143 sample_offset < entry->sample_offset_end) {
144 GST_LOG_OBJECT (wvparse, "found match");
148 GST_LOG_OBJECT (wvparse, "no match in index");
153 gst_wavpack_parse_index_append_entry (GstWavpackParse * wvparse,
154 gint64 byte_offset, gint64 sample_offset, gint64 num_samples)
156 GstWavpackParseIndexEntry entry;
158 if (wvparse->entries == NULL) {
159 wvparse->entries = g_array_new (FALSE, TRUE,
160 sizeof (GstWavpackParseIndexEntry));
162 /* do we have this one already? */
163 entry = *gst_wavpack_parse_index_get_last_entry (wvparse);
164 if (entry.byte_offset >= byte_offset)
168 GST_LOG_OBJECT (wvparse, "Adding index entry %8" G_GINT64_FORMAT " - %"
169 GST_TIME_FORMAT " @ offset 0x%08" G_GINT64_MODIFIER "x", sample_offset,
170 GST_TIME_ARGS (gst_util_uint64_scale_int (sample_offset,
171 GST_SECOND, wvparse->samplerate)), byte_offset);
173 entry.byte_offset = byte_offset;
174 entry.sample_offset = sample_offset;
175 entry.sample_offset_end = sample_offset + num_samples;
176 g_array_append_val (wvparse->entries, entry);
180 gst_wavpack_parse_reset (GstWavpackParse * wavpackparse)
182 wavpackparse->total_samples = 0;
183 wavpackparse->samplerate = 0;
184 wavpackparse->channels = 0;
186 gst_segment_init (&wavpackparse->segment, GST_FORMAT_UNDEFINED);
188 wavpackparse->current_offset = 0;
189 wavpackparse->need_newsegment = TRUE;
190 wavpackparse->upstream_length = -1;
192 if (wavpackparse->entries) {
193 g_array_free (wavpackparse->entries, TRUE);
194 wavpackparse->entries = NULL;
197 if (wavpackparse->srcpad != NULL) {
200 GST_DEBUG_OBJECT (wavpackparse, "Removing src pad");
201 res = gst_element_remove_pad (GST_ELEMENT (wavpackparse),
202 wavpackparse->srcpad);
203 g_return_if_fail (res != FALSE);
204 gst_object_unref (wavpackparse->srcpad);
205 wavpackparse->srcpad = NULL;
210 gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query)
212 GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
214 gboolean ret = FALSE;
216 switch (GST_QUERY_TYPE (query)) {
217 case GST_QUERY_POSITION:{
221 GST_OBJECT_LOCK (wavpackparse);
222 cur = wavpackparse->segment.last_stop;
223 len = wavpackparse->total_samples;
224 rate = wavpackparse->samplerate;
225 GST_OBJECT_UNLOCK (wavpackparse);
227 if (len <= 0 || rate == 0) {
228 GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet");
232 gst_query_parse_position (query, &format, NULL);
235 case GST_FORMAT_TIME:
236 cur = gst_util_uint64_scale_int (cur, GST_SECOND, rate);
237 gst_query_set_position (query, GST_FORMAT_TIME, cur);
240 case GST_FORMAT_DEFAULT:
241 gst_query_set_position (query, GST_FORMAT_DEFAULT, cur);
245 GST_DEBUG_OBJECT (wavpackparse, "cannot handle position query in "
246 "%s format", gst_format_get_name (format));
251 case GST_QUERY_DURATION:{
255 GST_OBJECT_LOCK (wavpackparse);
256 rate = wavpackparse->samplerate;
257 len = wavpackparse->total_samples;
258 GST_OBJECT_UNLOCK (wavpackparse);
260 if (len <= 0 || rate == 0) {
261 GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet");
265 gst_query_parse_duration (query, &format, NULL);
268 case GST_FORMAT_TIME:
269 len = gst_util_uint64_scale_int (len, GST_SECOND, rate);
270 gst_query_set_duration (query, GST_FORMAT_TIME, len);
273 case GST_FORMAT_DEFAULT:
274 gst_query_set_duration (query, GST_FORMAT_DEFAULT, len);
278 GST_DEBUG_OBJECT (wavpackparse, "cannot handle duration query in "
279 "%s format", gst_format_get_name (format));
285 ret = gst_pad_query_default (pad, query);
290 gst_object_unref (wavpackparse);
295 /* returns TRUE on success, with byte_offset set to the offset of the
296 * wavpack chunk containing the sample requested. start_sample will be
297 * set to the first sample in the chunk starting at byte_offset.
298 * Scanning from the last known header offset to the wanted position
299 * when seeking forward isn't very clever, but seems fast enough in
300 * practice and has the nice side effect of populating our index
303 gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse,
304 gint64 sample, gint64 * byte_offset, gint64 * start_sample)
306 GstWavpackParseIndexEntry *entry;
310 /* first, check if we have to scan at all */
311 entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample);
313 *byte_offset = entry->byte_offset;
314 *start_sample = entry->sample_offset;
315 GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT
316 " @ offset %" G_GINT64_FORMAT, entry->sample_offset,
321 GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ...");
323 /* if we have an index, we can start scanning from the last known offset
324 * in there, after all we know our wanted sample is not in the index */
325 if (parse->entries && parse->entries->len > 0) {
326 GstWavpackParseIndexEntry *entry;
328 entry = gst_wavpack_parse_index_get_last_entry (parse);
329 off = entry->byte_offset;
332 /* now scan forward until we find the chunk we're looking for or hit EOS */
334 WavpackHeader header = { {0,}
338 buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader),
344 gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
345 gst_buffer_unref (buf);
347 gst_wavpack_parse_index_append_entry (parse, off, header.block_index,
348 header.block_samples);
350 if (header.block_index <= sample &&
351 sample < (header.block_index + header.block_samples)) {
353 *start_sample = header.block_index;
357 off += header.ckSize + 8;
360 GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)",
361 gst_flow_get_name (ret), off);
367 gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update)
369 GstSegment *s = &wvparse->segment;
371 gint64 stop_time = -1;
372 gint64 start_time = 0;
376 /* segment is in DEFAULT format, but we want to send a TIME newsegment */
377 start_time = gst_util_uint64_scale_int (s->start, GST_SECOND,
378 wvparse->samplerate);
381 stop_time = gst_util_uint64_scale_int (s->stop, GST_SECOND,
382 wvparse->samplerate);
385 GST_DEBUG_OBJECT (wvparse, "sending newsegment from %" GST_TIME_FORMAT
386 " to %" GST_TIME_FORMAT, GST_TIME_ARGS (start_time),
387 GST_TIME_ARGS (stop_time));
389 /* after a seek, s->last_stop will point to a chunk boundary, ie. from
390 * which sample we will start sending data again, while s->start will
391 * point to the sample we actually want to seek to and want to start
392 * playing right after the seek. Adjust clock-time for the difference
393 * so we start playing from start_time */
394 cur_pos_time = gst_util_uint64_scale_int (s->last_stop, GST_SECOND,
395 wvparse->samplerate);
396 diff = start_time - cur_pos_time;
398 ret = gst_pad_push_event (wvparse->srcpad,
399 gst_event_new_new_segment (update, s->rate, GST_FORMAT_TIME,
400 start_time, stop_time, start_time - diff));
406 gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse,
409 GstSeekFlags seek_flags;
410 GstSeekType start_type;
411 GstSeekType stop_type;
414 gboolean only_update;
418 gint64 start; /* sample we want to seek to */
419 gint64 byte_offset; /* byte offset the chunk we seek to starts at */
420 gint64 chunk_start; /* first sample in chunk we seek to */
423 gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type,
424 &start, &stop_type, &stop);
426 if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) {
427 GST_DEBUG ("seeking is only supported in TIME or DEFAULT format");
432 GST_DEBUG ("only forward playback supported, rate %f not allowed", speed);
436 GST_OBJECT_LOCK (wvparse);
438 rate = wvparse->samplerate;
440 GST_OBJECT_UNLOCK (wvparse);
441 GST_DEBUG ("haven't read header yet");
445 /* convert from time to samples if necessary */
446 if (format == GST_FORMAT_TIME) {
447 if (start_type != GST_SEEK_TYPE_NONE)
448 start = gst_util_uint64_scale_int (start, rate, GST_SECOND);
449 if (stop_type != GST_SEEK_TYPE_NONE)
450 stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND);
453 flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0);
456 GST_OBJECT_UNLOCK (wvparse);
457 GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start);
461 /* operate on segment copy until we know the seek worked */
462 segment = wvparse->segment;
464 gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT,
465 seek_flags, start_type, start, stop_type, stop, &only_update);
469 wvparse->segment = segment;
470 gst_wavpack_parse_send_newsegment (wvparse, TRUE);
475 gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ());
478 gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ());
480 gst_pad_stop_task (wvparse->sinkpad);
483 GST_PAD_STREAM_LOCK (wvparse->sinkpad);
485 gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ());
488 gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ());
491 GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %"
492 G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate),
495 ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start,
496 &byte_offset, &chunk_start);
499 GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset);
500 wvparse->current_offset = byte_offset;
501 /* we want to send a newsegment event with the actual seek position
502 * as start, even though our first buffer might start before the
503 * configured segment. We leave it up to the decoder or sink to crop
504 * the output buffers accordingly */
505 wvparse->segment = segment;
506 wvparse->segment.last_stop = chunk_start;
507 gst_wavpack_parse_send_newsegment (wvparse, FALSE);
509 GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to");
512 GST_PAD_STREAM_UNLOCK (wvparse->sinkpad);
513 GST_OBJECT_UNLOCK (wvparse);
515 gst_pad_start_task (wvparse->sinkpad,
516 (GstTaskFunction) gst_wavpack_parse_loop, wvparse);
522 gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event)
524 GstWavpackParse *wavpackparse;
527 wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
529 switch (GST_EVENT_TYPE (event)) {
531 ret = gst_wavpack_parse_handle_seek_event (wavpackparse, event);
534 ret = gst_pad_event_default (pad, event);
538 gst_object_unref (wavpackparse);
543 gst_wavpack_parse_init (GstWavpackParse * wavpackparse,
544 GstWavpackParseClass * gclass)
546 GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackparse);
547 GstPadTemplate *tmpl;
549 tmpl = gst_element_class_get_pad_template (klass, "sink");
550 wavpackparse->sinkpad = gst_pad_new_from_template (tmpl, "sink");
552 gst_pad_set_activate_function (wavpackparse->sinkpad,
553 GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate));
555 gst_pad_set_activatepull_function (wavpackparse->sinkpad,
556 GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate_pull));
558 gst_element_add_pad (GST_ELEMENT (wavpackparse), wavpackparse->sinkpad);
560 wavpackparse->srcpad = NULL;
561 gst_wavpack_parse_reset (wavpackparse);
565 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wavpackparse)
570 peer = gst_pad_get_peer (wavpackparse->sinkpad);
572 GstFormat format = GST_FORMAT_BYTES;
574 if (!gst_pad_query_duration (peer, &format, &length)) {
577 GST_DEBUG ("upstream length: %" G_GINT64_FORMAT, length);
579 gst_object_unref (peer);
581 GST_DEBUG ("no peer!");
588 gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset,
589 guint size, GstFlowReturn * flow)
591 GstFlowReturn flow_ret;
592 GstBuffer *buf = NULL;
594 if (offset + size >= wvparse->upstream_length) {
595 wvparse->upstream_length = gst_wavpack_parse_get_upstream_length (wvparse);
596 if (offset + size >= wvparse->upstream_length) {
597 GST_DEBUG_OBJECT (wvparse, "EOS: %" G_GINT64_FORMAT " + %u > %"
598 G_GINT64_FORMAT, offset, size, wvparse->upstream_length);
599 flow_ret = GST_FLOW_UNEXPECTED;
604 flow_ret = gst_pad_pull_range (wvparse->sinkpad, offset, size, &buf);
606 if (flow_ret != GST_FLOW_OK) {
607 GST_DEBUG_OBJECT (wvparse, "pull_range (%" G_GINT64_FORMAT ", %u) "
608 "failed, flow: %s", offset, size, gst_flow_get_name (flow_ret));
612 if (GST_BUFFER_SIZE (buf) < size) {
613 GST_DEBUG_OBJECT (wvparse, "Short read at offset %" G_GINT64_FORMAT
614 ", got only %u of %u bytes", offset, GST_BUFFER_SIZE (buf), size);
615 gst_buffer_unref (buf);
617 flow_ret = GST_FLOW_UNEXPECTED;
627 gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf,
628 WavpackHeader * header)
630 WavpackMetadata meta;
631 GstCaps *caps = NULL;
634 g_assert (wvparse->srcpad == NULL);
636 bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader);
638 while (read_metadata_buff (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
640 case ID_WVC_BITSTREAM:{
641 caps = gst_caps_new_simple ("audio/x-wavpack-correction",
642 "framed", G_TYPE_BOOLEAN, TRUE, NULL);
644 gst_pad_new_from_template (gst_element_class_get_pad_template
645 (GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc");
648 case ID_RIFF_HEADER:{
651 /* skip RiffChunkHeader and ChunkHeader */
652 g_memmove (&wheader, meta.data + 20, sizeof (WaveHeader));
653 little_endian_to_native (&wheader, WaveHeaderFormat);
654 wvparse->samplerate = wheader.SampleRate;
655 wvparse->channels = wheader.NumChannels;
656 wvparse->total_samples = header->total_samples;
657 caps = gst_caps_new_simple ("audio/x-wavpack",
658 "width", G_TYPE_INT, wheader.BitsPerSample,
659 "channels", G_TYPE_INT, wvparse->channels,
660 "rate", G_TYPE_INT, wvparse->samplerate,
661 "framed", G_TYPE_BOOLEAN, TRUE, NULL);
663 gst_pad_new_from_template (gst_element_class_get_pad_template
664 (GST_ELEMENT_GET_CLASS (wvparse), "src"), "src");
668 GST_WARNING_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id);
676 if (caps == NULL || wvparse->srcpad == NULL)
679 GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps);
681 gst_pad_set_query_function (wvparse->srcpad,
682 GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query));
683 gst_pad_set_event_function (wvparse->srcpad,
684 GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event));
686 gst_pad_set_caps (wvparse->srcpad, caps);
687 gst_pad_use_fixed_caps (wvparse->srcpad);
689 gst_object_ref (wvparse->srcpad);
690 gst_element_add_pad (GST_ELEMENT (wvparse), wvparse->srcpad);
691 gst_element_no_more_pads (GST_ELEMENT (wvparse));
697 gst_wavpack_parse_loop (GstElement * element)
699 GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (element);
700 GstFlowReturn flow_ret;
701 WavpackHeader header = { {0,}, 0, };
702 GstBuffer *buf = NULL;
704 GST_LOG_OBJECT (wavpackparse, "Current offset: %" G_GINT64_FORMAT,
705 wavpackparse->current_offset);
707 buf = gst_wavpack_parse_pull_buffer (wavpackparse,
708 wavpackparse->current_offset, sizeof (WavpackHeader), &flow_ret);
710 if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) {
712 } else if (buf == NULL) {
716 gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
717 gst_buffer_unref (buf);
719 GST_LOG_OBJECT (wavpackparse, "Read header at offset %" G_GINT64_FORMAT
720 ": chunk size = %u+8", wavpackparse->current_offset, header.ckSize);
722 buf = gst_wavpack_parse_pull_buffer (wavpackparse,
723 wavpackparse->current_offset, header.ckSize + 8, &flow_ret);
725 if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) {
727 } else if (buf == NULL) {
731 if (wavpackparse->srcpad == NULL) {
732 if (!gst_wavpack_parse_create_src_pad (wavpackparse, buf, &header)) {
733 GST_ELEMENT_ERROR (wavpackparse, STREAM, DECODE, (NULL), (NULL));
738 gst_wavpack_parse_index_append_entry (wavpackparse,
739 wavpackparse->current_offset, header.block_index, header.block_samples);
741 wavpackparse->current_offset += header.ckSize + 8;
743 wavpackparse->segment.last_stop = header.block_index;
745 if (wavpackparse->need_newsegment) {
746 if (gst_wavpack_parse_send_newsegment (wavpackparse, FALSE))
747 wavpackparse->need_newsegment = FALSE;
750 GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header.block_index,
751 GST_SECOND, wavpackparse->samplerate);
752 GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header.block_samples,
753 GST_SECOND, wavpackparse->samplerate);
754 GST_BUFFER_OFFSET (buf) = header.block_index;
755 gst_buffer_set_caps (buf, GST_PAD_CAPS (wavpackparse->srcpad));
757 GST_LOG_OBJECT (wavpackparse, "Pushing buffer with time %" GST_TIME_FORMAT,
758 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
760 flow_ret = gst_pad_push (wavpackparse->srcpad, buf);
761 if (flow_ret != GST_FLOW_OK) {
762 GST_DEBUG_OBJECT (wavpackparse, "Push failed, flow: %s",
763 gst_flow_get_name (flow_ret));
771 GST_DEBUG_OBJECT (wavpackparse, "sending EOS");
772 if (wavpackparse->srcpad) {
773 gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ());
775 /* fall through and pause task */
779 GST_DEBUG_OBJECT (wavpackparse, "Pausing task");
780 gst_pad_pause_task (wavpackparse->sinkpad);
785 static GstStateChangeReturn
786 gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition)
788 GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element);
789 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
791 switch (transition) {
792 case GST_STATE_CHANGE_READY_TO_PAUSED:
793 gst_segment_init (&wvparse->segment, GST_FORMAT_DEFAULT);
794 wvparse->segment.last_stop = 0;
799 if (GST_ELEMENT_CLASS (parent_class)->change_state)
800 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
802 switch (transition) {
803 case GST_STATE_CHANGE_PAUSED_TO_READY:
804 gst_wavpack_parse_reset (wvparse);
815 gst_wavepack_parse_sink_activate (GstPad * sinkpad)
817 if (gst_pad_check_pull_range (sinkpad)) {
818 return gst_pad_activate_pull (sinkpad, TRUE);
825 gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active)
830 result = gst_pad_start_task (sinkpad,
831 (GstTaskFunction) gst_wavpack_parse_loop, GST_PAD_PARENT (sinkpad));
833 result = gst_pad_stop_task (sinkpad);
840 gst_wavpack_parse_plugin_init (GstPlugin * plugin)
842 if (!gst_element_register (plugin, "wavpackparse",
843 GST_RANK_PRIMARY, GST_TYPE_WAVPACK_PARSE)) {
847 GST_DEBUG_CATEGORY_INIT (gst_wavpack_parse_debug, "wavpackparse", 0,
848 "wavpack file parser");