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gstreamer/ext/wavpack/gstwavpackparse.c

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/* GStreamer wavpack plugin
* Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* Copyright (c) 2006 Tim-Philipp Müller <tim centricular net>
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstwavpackparse.c: wavpack file parser
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-wavpackparse
*
* WavpackParse takes raw, unframed Wavpack streams and splits them into
* single Wavpack chunks with information like bit depth and the position
* in the stream.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! fakesink
* ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <gst/gst.h>
#include <gst/gst-i18n-plugin.h>
#include <math.h>
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackparse.h"
#include "gstwavpackstreamreader.h"
#include "gstwavpackcommon.h"
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug);
#define GST_CAT_DEFAULT gst_wavpack_parse_debug
static inline GstWavpackParseIndexEntry *
gst_wavpack_parse_index_entry_new (void)
{
return g_slice_new (GstWavpackParseIndexEntry);
}
static inline void
gst_wavpack_parse_index_entry_free (GstWavpackParseIndexEntry * entry)
{
g_slice_free (GstWavpackParseIndexEntry, entry);
}
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"framed = (boolean) false; "
"audio/x-wavpack-correction, " "framed = (boolean) false")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) [ 1, 32 ], "
"channels = (int) [ 1, 8 ], "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
static GstStaticPadTemplate wvc_src_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
);
static gboolean gst_wavpack_parse_sink_activate (GstPad * sinkpad);
static gboolean
gst_wavpack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active);
static void gst_wavpack_parse_loop (GstElement * element);
static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement *
element, GstStateChange transition);
static void gst_wavpack_parse_reset (GstWavpackParse * parse);
static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wvparse);
static GstBuffer *gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse,
gint64 offset, guint size, GstFlowReturn * flow);
static GstFlowReturn gst_wavpack_parse_chain (GstPad * pad, GstBuffer * buf);
GST_BOILERPLATE (GstWavpackParse, gst_wavpack_parse, GstElement,
GST_TYPE_ELEMENT);
static void
gst_wavpack_parse_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &src_factory);
gst_element_class_add_static_pad_template (element_class, &wvc_src_factory);
gst_element_class_add_static_pad_template (element_class, &sink_factory);
gst_element_class_set_details_simple (element_class, "Wavpack parser",
"Codec/Demuxer/Audio",
"Parses Wavpack files",
"Arwed v. Merkatz <v.merkatz@gmx.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
}
static void
gst_wavpack_parse_finalize (GObject * object)
{
gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object));
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->finalize = gst_wavpack_parse_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state);
}
static GstWavpackParseIndexEntry *
gst_wavpack_parse_index_get_last_entry (GstWavpackParse * wvparse)
{
g_assert (wvparse->entries != NULL);
return wvparse->entries->data;
}
static GstWavpackParseIndexEntry *
gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse,
gint64 sample_offset)
{
gint i;
GSList *node;
if (wvparse->entries == NULL)
return NULL;
for (node = wvparse->entries, i = 0; node; node = node->next, i++) {
GstWavpackParseIndexEntry *entry;
entry = node->data;
GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @"
" byte %" G_GINT64_FORMAT, i, entry->sample_offset, entry->byte_offset);
if (entry->sample_offset <= sample_offset &&
sample_offset < entry->sample_offset_end) {
GST_LOG_OBJECT (wvparse, "found match");
return entry;
}
/* as the list is sorted and we first look at the latest entry
* we can abort searching for an entry if the sample we want is
* after the latest one */
if (sample_offset >= entry->sample_offset_end)
break;
}
GST_LOG_OBJECT (wvparse, "no match in index");
return NULL;
}
static void
gst_wavpack_parse_index_append_entry (GstWavpackParse * wvparse,
gint64 byte_offset, gint64 sample_offset, gint64 num_samples)
{
GstWavpackParseIndexEntry *entry;
/* do we have this one already? */
if (wvparse->entries) {
entry = gst_wavpack_parse_index_get_last_entry (wvparse);
if (entry->byte_offset >= byte_offset
|| entry->sample_offset >= sample_offset)
return;
}
GST_LOG_OBJECT (wvparse, "Adding index entry %8" G_GINT64_FORMAT " - %"
GST_TIME_FORMAT " @ offset 0x%08" G_GINT64_MODIFIER "x", sample_offset,
GST_TIME_ARGS (gst_util_uint64_scale_int (sample_offset,
GST_SECOND, wvparse->samplerate)), byte_offset);
entry = gst_wavpack_parse_index_entry_new ();
entry->byte_offset = byte_offset;
entry->sample_offset = sample_offset;
entry->sample_offset_end = sample_offset + num_samples;
wvparse->entries = g_slist_prepend (wvparse->entries, entry);
}
static void
gst_wavpack_parse_reset (GstWavpackParse * parse)
{
parse->total_samples = G_GINT64_CONSTANT (-1);
parse->samplerate = 0;
parse->channels = 0;
gst_segment_init (&parse->segment, GST_FORMAT_UNDEFINED);
parse->next_block_index = 0;
parse->current_offset = 0;
parse->need_newsegment = TRUE;
parse->discont = TRUE;
parse->upstream_length = -1;
if (parse->entries) {
g_slist_foreach (parse->entries, (GFunc) gst_wavpack_parse_index_entry_free,
NULL);
g_slist_free (parse->entries);
parse->entries = NULL;
}
if (parse->adapter) {
gst_adapter_clear (parse->adapter);
g_object_unref (parse->adapter);
parse->adapter = NULL;
}
if (parse->srcpad != NULL) {
gboolean res;
GST_DEBUG_OBJECT (parse, "Removing src pad");
res = gst_element_remove_pad (GST_ELEMENT (parse), parse->srcpad);
g_return_if_fail (res != FALSE);
gst_object_unref (parse->srcpad);
parse->srcpad = NULL;
}
g_list_foreach (parse->queued_events, (GFunc) gst_mini_object_unref, NULL);
g_list_free (parse->queued_events);
parse->queued_events = NULL;
if (parse->pending_buffer)
gst_buffer_unref (parse->pending_buffer);
parse->pending_buffer = NULL;
}
static const GstQueryType *
gst_wavpack_parse_get_src_query_types (GstPad * pad)
{
static const GstQueryType types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_SEEKING,
0
};
return types;
}
static gboolean
gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query)
{
GstWavpackParse *parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
GstFormat format;
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:{
gint64 cur;
guint rate;
GST_OBJECT_LOCK (parse);
cur = parse->segment.last_stop;
rate = parse->samplerate;
GST_OBJECT_UNLOCK (parse);
if (rate == 0) {
GST_DEBUG_OBJECT (parse, "haven't read header yet");
break;
}
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
cur = gst_util_uint64_scale_int (cur, GST_SECOND, rate);
gst_query_set_position (query, GST_FORMAT_TIME, cur);
ret = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, GST_FORMAT_DEFAULT, cur);
ret = TRUE;
break;
default:
GST_DEBUG_OBJECT (parse, "cannot handle position query in "
"%s format. Forwarding upstream.", gst_format_get_name (format));
ret = gst_pad_query_default (pad, query);
break;
}
break;
}
case GST_QUERY_DURATION:{
gint64 len;
guint rate;
GST_OBJECT_LOCK (parse);
rate = parse->samplerate;
len = parse->total_samples;
GST_OBJECT_UNLOCK (parse);
if (rate == 0) {
GST_DEBUG_OBJECT (parse, "haven't read header yet");
break;
}
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
if (len != G_GINT64_CONSTANT (-1))
len = gst_util_uint64_scale_int (len, GST_SECOND, rate);
gst_query_set_duration (query, GST_FORMAT_TIME, len);
ret = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_duration (query, GST_FORMAT_DEFAULT, len);
ret = TRUE;
break;
default:
GST_DEBUG_OBJECT (parse, "cannot handle duration query in "
"%s format. Forwarding upstream.", gst_format_get_name (format));
ret = gst_pad_query_default (pad, query);
break;
}
break;
}
case GST_QUERY_SEEKING:{
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
if (format == GST_FORMAT_TIME || format == GST_FORMAT_DEFAULT) {
gboolean seekable;
gint64 duration = -1;
/* only fails if we didn't read the headers yet and can't say
* anything about our seeking capabilities */
if (!gst_pad_query_duration (pad, &format, &duration))
break;
/* can't seek in streaming mode yet */
GST_OBJECT_LOCK (parse);
seekable = (parse->adapter == NULL);
GST_OBJECT_UNLOCK (parse);
gst_query_set_seeking (query, format, seekable, 0, duration);
ret = TRUE;
}
break;
}
default:{
ret = gst_pad_query_default (pad, query);
break;
}
}
gst_object_unref (parse);
return ret;
}
/* returns TRUE on success, with byte_offset set to the offset of the
* wavpack chunk containing the sample requested. start_sample will be
* set to the first sample in the chunk starting at byte_offset.
* Scanning from the last known header offset to the wanted position
* when seeking forward isn't very clever, but seems fast enough in
* practice and has the nice side effect of populating our index
* table */
static gboolean
gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse,
gint64 sample, gint64 * byte_offset, gint64 * start_sample)
{
GstWavpackParseIndexEntry *entry;
GstFlowReturn ret;
gint64 off = 0;
/* first, check if we have to scan at all */
entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample);
if (entry) {
*byte_offset = entry->byte_offset;
*start_sample = entry->sample_offset;
GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT
" @ offset %" G_GINT64_FORMAT, entry->sample_offset,
entry->byte_offset);
return TRUE;
}
GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ...");
/* if we have an index, we can start scanning from the last known offset
* in there, after all we know our wanted sample is not in the index */
if (parse->entries) {
GstWavpackParseIndexEntry *entry;
entry = gst_wavpack_parse_index_get_last_entry (parse);
off = entry->byte_offset;
}
/* now scan forward until we find the chunk we're looking for or hit EOS */
do {
WavpackHeader header;
GstBuffer *buf;
buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader),
&ret);
if (buf == NULL)
break;
gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
gst_buffer_unref (buf);
if (header.flags & INITIAL_BLOCK)
gst_wavpack_parse_index_append_entry (parse, off, header.block_index,
header.block_samples);
else
continue;
if (header.block_index <= sample &&
sample < (header.block_index + header.block_samples)) {
*byte_offset = off;
*start_sample = header.block_index;
return TRUE;
}
off += header.ckSize + 8;
} while (1);
GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)",
gst_flow_get_name (ret), off);
return FALSE;
}
static gboolean
gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update)
{
GstSegment *s = &wvparse->segment;
gboolean ret;
gint64 stop_time = -1;
gint64 start_time = 0;
gint64 cur_pos_time;
gint64 diff;
/* segment is in DEFAULT format, but we want to send a TIME newsegment */
start_time = gst_util_uint64_scale_int (s->start, GST_SECOND,
wvparse->samplerate);
if (s->stop != -1) {
stop_time = gst_util_uint64_scale_int (s->stop, GST_SECOND,
wvparse->samplerate);
}
GST_DEBUG_OBJECT (wvparse, "sending newsegment from %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT, GST_TIME_ARGS (start_time),
GST_TIME_ARGS (stop_time));
/* after a seek, s->last_stop will point to a chunk boundary, ie. from
* which sample we will start sending data again, while s->start will
* point to the sample we actually want to seek to and want to start
* playing right after the seek. Adjust clock-time for the difference
* so we start playing from start_time */
cur_pos_time = gst_util_uint64_scale_int (s->last_stop, GST_SECOND,
wvparse->samplerate);
diff = start_time - cur_pos_time;
ret = gst_pad_push_event (wvparse->srcpad,
gst_event_new_new_segment (update, s->rate, GST_FORMAT_TIME,
start_time, stop_time, start_time - diff));
return ret;
}
static gboolean
gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse,
GstEvent * event)
{
GstSeekFlags seek_flags;
GstSeekType start_type;
GstSeekType stop_type;
GstSegment segment;
GstFormat format;
gboolean only_update;
gboolean flush, ret;
gdouble speed;
gint64 stop;
gint64 start; /* sample we want to seek to */
gint64 byte_offset; /* byte offset the chunk we seek to starts at */
gint64 chunk_start; /* first sample in chunk we seek to */
guint rate;
gint64 last_stop;
if (wvparse->adapter) {
GST_DEBUG_OBJECT (wvparse, "seeking in streaming mode not implemented yet");
return FALSE;
}
gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type,
&start, &stop_type, &stop);
if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) {
GST_DEBUG ("seeking is only supported in TIME or DEFAULT format");
return FALSE;
}
if (speed < 0.0) {
GST_DEBUG ("only forward playback supported, rate %f not allowed", speed);
return FALSE;
}
GST_OBJECT_LOCK (wvparse);
rate = wvparse->samplerate;
if (rate == 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG ("haven't read header yet");
return FALSE;
}
/* figure out the last position we need to play. If it's configured (stop !=
* -1), use that, else we play until the total duration of the file */
if (stop == -1)
stop = wvparse->segment.duration;
/* convert from time to samples if necessary */
if (format == GST_FORMAT_TIME) {
if (start_type != GST_SEEK_TYPE_NONE)
start = gst_util_uint64_scale_int (start, rate, GST_SECOND);
if (stop_type != GST_SEEK_TYPE_NONE)
stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND);
}
if (start < 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start);
return FALSE;
}
flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0);
/* operate on segment copy until we know the seek worked */
segment = wvparse->segment;
gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT,
seek_flags, start_type, start, stop_type, stop, &only_update);
#if 0
if (only_update) {
wvparse->segment = segment;
gst_wavpack_parse_send_newsegment (wvparse, TRUE);
goto done;
}
#endif
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_pause_task (wvparse->sinkpad);
}
GST_PAD_STREAM_LOCK (wvparse->sinkpad);
/* Save current position */
last_stop = wvparse->segment.last_stop;
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ());
}
GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %"
G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate),
start);
ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start,
&byte_offset, &chunk_start);
if (ret) {
GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset);
wvparse->current_offset = byte_offset;
/* we want to send a newsegment event with the actual seek position
* as start, even though our first buffer might start before the
* configured segment. We leave it up to the decoder or sink to crop
* the output buffers accordingly */
wvparse->segment = segment;
wvparse->segment.last_stop = chunk_start;
wvparse->need_newsegment = TRUE;
wvparse->discont = (last_stop != chunk_start) ? TRUE : FALSE;
/* if we're doing a segment seek, post a SEGMENT_START message */
if (wvparse->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (wvparse),
gst_message_new_segment_start (GST_OBJECT_CAST (wvparse),
wvparse->segment.format, wvparse->segment.last_stop));
}
} else {
GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to");
}
GST_PAD_STREAM_UNLOCK (wvparse->sinkpad);
GST_OBJECT_UNLOCK (wvparse);
gst_pad_start_task (wvparse->sinkpad,
(GstTaskFunction) gst_wavpack_parse_loop, wvparse);
return ret;
}
static gboolean
gst_wavpack_parse_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackParse *parse;
gboolean ret = TRUE;
parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:{
if (parse->adapter) {
gst_adapter_clear (parse->adapter);
}
if (parse->pending_buffer) {
gst_buffer_unref (parse->pending_buffer);
parse->pending_buffer = NULL;
parse->pending_offset = 0;
}
ret = gst_pad_push_event (parse->srcpad, event);
break;
}
case GST_EVENT_NEWSEGMENT:{
parse->need_newsegment = TRUE;
gst_event_unref (event);
ret = TRUE;
break;
}
case GST_EVENT_EOS:{
if (parse->adapter) {
/* remove all bytes that are left in the adapter after EOS. They can't
* be a complete Wavpack block and we can't do anything with them */
gst_adapter_clear (parse->adapter);
}
if (parse->pending_buffer) {
gst_buffer_unref (parse->pending_buffer);
parse->pending_buffer = NULL;
parse->pending_offset = 0;
}
ret = gst_pad_push_event (parse->srcpad, event);
break;
}
default:{
/* stream lock is recursive, should be fine for all events */
GST_PAD_STREAM_LOCK (pad);
if (parse->srcpad == NULL) {
parse->queued_events = g_list_append (parse->queued_events, event);
} else {
ret = gst_pad_push_event (parse->srcpad, event);
}
GST_PAD_STREAM_UNLOCK (pad);
}
}
gst_object_unref (parse);
return ret;
}
static gboolean
gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event)
{
GstWavpackParse *parse;
gboolean ret;
parse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
ret = gst_wavpack_parse_handle_seek_event (parse, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (parse);
return ret;
}
static void
gst_wavpack_parse_init (GstWavpackParse * parse, GstWavpackParseClass * gclass)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (parse);
GstPadTemplate *tmpl;
tmpl = gst_element_class_get_pad_template (klass, "sink");
parse->sinkpad = gst_pad_new_from_template (tmpl, "sink");
gst_pad_set_activate_function (parse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_activate));
gst_pad_set_activatepull_function (parse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_activate_pull));
gst_pad_set_event_function (parse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_sink_event));
gst_pad_set_chain_function (parse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_chain));
gst_element_add_pad (GST_ELEMENT (parse), parse->sinkpad);
parse->srcpad = NULL;
gst_wavpack_parse_reset (parse);
}
static gint64
gst_wavpack_parse_get_upstream_length (GstWavpackParse * parse)
{
gint64 length = -1;
GstFormat format = GST_FORMAT_BYTES;
if (!gst_pad_query_peer_duration (parse->sinkpad, &format, &length)) {
length = -1;
} else {
GST_DEBUG ("upstream length: %" G_GINT64_FORMAT, length);
}
return length;
}
static GstBuffer *
gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset,
guint size, GstFlowReturn * flow)
{
GstFlowReturn flow_ret;
GstBuffer *buf = NULL;
if (offset + size > wvparse->upstream_length) {
wvparse->upstream_length = gst_wavpack_parse_get_upstream_length (wvparse);
if (offset + size > wvparse->upstream_length) {
GST_DEBUG_OBJECT (wvparse, "EOS: %" G_GINT64_FORMAT " + %u > %"
G_GINT64_FORMAT, offset, size, wvparse->upstream_length);
flow_ret = GST_FLOW_UNEXPECTED;
goto done;
}
}
flow_ret = gst_pad_pull_range (wvparse->sinkpad, offset, size, &buf);
if (flow_ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (wvparse, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", offset, size, gst_flow_get_name (flow_ret));
buf = NULL;
goto done;
}
if (GST_BUFFER_SIZE (buf) < size) {
GST_DEBUG_OBJECT (wvparse, "Short read at offset %" G_GINT64_FORMAT
", got only %u of %u bytes", offset, GST_BUFFER_SIZE (buf), size);
gst_buffer_unref (buf);
buf = NULL;
flow_ret = GST_FLOW_UNEXPECTED;
}
done:
if (flow)
*flow = flow_ret;
return buf;
}
static gboolean
gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf,
WavpackHeader * header)
{
GstWavpackMetadata meta;
GstCaps *caps = NULL;
guchar *bufptr;
g_assert (wvparse->srcpad == NULL);
bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader);
while (gst_wavpack_read_metadata (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
switch (meta.id) {
case ID_WVC_BITSTREAM:{
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
wvparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc");
break;
}
case ID_WV_BITSTREAM:
case ID_WVX_BITSTREAM:{
WavpackStreamReader *stream_reader = gst_wavpack_stream_reader_new ();
WavpackContext *wpc;
gchar error_msg[80];
read_id rid;
gint channel_mask;
rid.buffer = GST_BUFFER_DATA (buf);
rid.length = GST_BUFFER_SIZE (buf);
rid.position = 0;
wpc =
WavpackOpenFileInputEx (stream_reader, &rid, NULL, error_msg, 0, 0);
if (!wpc)
return FALSE;
wvparse->samplerate = WavpackGetSampleRate (wpc);
wvparse->channels = WavpackGetNumChannels (wpc);
wvparse->total_samples =
(header->total_samples ==
0xffffffff) ? G_GINT64_CONSTANT (-1) : header->total_samples;
caps = gst_caps_new_simple ("audio/x-wavpack",
"width", G_TYPE_INT, WavpackGetBitsPerSample (wpc),
"channels", G_TYPE_INT, wvparse->channels,
"rate", G_TYPE_INT, wvparse->samplerate,
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
#ifdef WAVPACK_OLD_API
channel_mask = wpc->config.channel_mask;
#else
channel_mask = WavpackGetChannelMask (wpc);
#endif
if (channel_mask == 0)
channel_mask =
gst_wavpack_get_default_channel_mask (wvparse->channels);
if (channel_mask != 0) {
if (!gst_wavpack_set_channel_layout (caps, channel_mask)) {
GST_WARNING_OBJECT (wvparse, "Failed to set channel layout");
gst_caps_unref (caps);
caps = NULL;
WavpackCloseFile (wpc);
g_free (stream_reader);
break;
}
}
wvparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wvparse), "src"), "src");
WavpackCloseFile (wpc);
g_free (stream_reader);
break;
}
default:{
GST_LOG_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id);
break;
}
}
if (caps != NULL)
break;
}
if (caps == NULL || wvparse->srcpad == NULL)
return FALSE;
GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps);
gst_pad_set_query_function (wvparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query));
gst_pad_set_query_type_function (wvparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_src_query_types));
gst_pad_set_event_function (wvparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event));
gst_pad_set_caps (wvparse->srcpad, caps);
gst_caps_unref (caps);
gst_pad_use_fixed_caps (wvparse->srcpad);
gst_object_ref (wvparse->srcpad);
gst_pad_set_active (wvparse->srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT (wvparse), wvparse->srcpad);
gst_element_no_more_pads (GST_ELEMENT (wvparse));
return TRUE;
}
static GstFlowReturn
gst_wavpack_parse_push_buffer (GstWavpackParse * wvparse, GstBuffer * buf,
WavpackHeader * header)
{
GstFlowReturn ret;
wvparse->current_offset += header->ckSize + 8;
wvparse->segment.last_stop = header->block_index;
if (wvparse->need_newsegment) {
if (gst_wavpack_parse_send_newsegment (wvparse, FALSE))
wvparse->need_newsegment = FALSE;
}
/* send any queued events */
if (wvparse->queued_events) {
GList *l;
for (l = wvparse->queued_events; l != NULL; l = l->next) {
gst_pad_push_event (wvparse->srcpad, GST_EVENT (l->data));
}
g_list_free (wvparse->queued_events);
wvparse->queued_events = NULL;
}
if (wvparse->pending_buffer == NULL) {
wvparse->pending_buffer = buf;
wvparse->pending_offset = header->block_index;
} else if (wvparse->pending_offset == header->block_index) {
wvparse->pending_buffer = gst_buffer_join (wvparse->pending_buffer, buf);
} else {
GST_ERROR ("Got incomplete block, dropping");
gst_buffer_unref (wvparse->pending_buffer);
wvparse->pending_buffer = buf;
wvparse->pending_offset = header->block_index;
}
if (!(header->flags & FINAL_BLOCK))
return GST_FLOW_OK;
buf = wvparse->pending_buffer;
wvparse->pending_buffer = NULL;
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header->block_index,
GST_SECOND, wvparse->samplerate);
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header->block_samples,
GST_SECOND, wvparse->samplerate);
GST_BUFFER_OFFSET (buf) = header->block_index;
GST_BUFFER_OFFSET_END (buf) = header->block_index + header->block_samples;
if (wvparse->discont || wvparse->next_block_index != header->block_index) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
wvparse->discont = FALSE;
}
wvparse->next_block_index = header->block_index + header->block_samples;
gst_buffer_set_caps (buf, GST_PAD_CAPS (wvparse->srcpad));
GST_LOG_OBJECT (wvparse, "Pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
ret = gst_pad_push (wvparse->srcpad, buf);
wvparse->segment.last_stop = wvparse->next_block_index;
return ret;
}
static guint8 *
gst_wavpack_parse_find_marker (guint8 * buf, guint size)
{
gint i;
guint8 *ret = NULL;
if (G_UNLIKELY (size < 4))
return NULL;
for (i = 0; i < size - 4; i++) {
if (memcmp (buf + i, "wvpk", 4) == 0) {
ret = buf + i;
break;
}
}
return ret;
}
static GstFlowReturn
gst_wavpack_parse_resync_loop (GstWavpackParse * parse, WavpackHeader * header)
{
GstFlowReturn flow_ret = GST_FLOW_UNEXPECTED;
GstBuffer *buf = NULL;
/* loop until we have a frame header or reach the end of the stream */
while (1) {
guint8 *data, *marker;
guint len, size;
if (buf) {
gst_buffer_unref (buf);
buf = NULL;
}
if (parse->upstream_length == 0 ||
parse->upstream_length <= parse->current_offset) {
parse->upstream_length = gst_wavpack_parse_get_upstream_length (parse);
if (parse->upstream_length == 0 ||
parse->upstream_length <= parse->current_offset) {
break;
}
}
len = MIN (parse->upstream_length - parse->current_offset, 2048);
GST_LOG_OBJECT (parse, "offset: %" G_GINT64_FORMAT, parse->current_offset);
buf = gst_wavpack_parse_pull_buffer (parse, parse->current_offset,
len, &flow_ret);
/* whatever the problem is, there's nothing more for us to do for now */
if (flow_ret != GST_FLOW_OK)
break;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* not enough data for a header? */
if (size < sizeof (WavpackHeader))
break;
/* got a header right where we are at now? */
if (gst_wavpack_read_header (header, data))
break;
/* nope, let's see if we can find one */
marker = gst_wavpack_parse_find_marker (data + 1, size - 1);
if (marker) {
parse->current_offset += marker - data;
/* do one more loop iteration to make sure we pull enough
* data for a full header, we'll bail out then */
} else {
parse->current_offset += len - 4;
}
}
if (buf)
gst_buffer_unref (buf);
return flow_ret;
}
static void
gst_wavpack_parse_loop (GstElement * element)
{
GstWavpackParse *parse = GST_WAVPACK_PARSE (element);
GstFlowReturn flow_ret;
WavpackHeader header = { {0,}, 0, };
GstBuffer *buf = NULL;
flow_ret = gst_wavpack_parse_resync_loop (parse, &header);
if (flow_ret != GST_FLOW_OK)
goto pause;
GST_LOG_OBJECT (parse, "Read header at offset %" G_GINT64_FORMAT
": chunk size = %u+8", parse->current_offset, header.ckSize);
buf = gst_wavpack_parse_pull_buffer (parse, parse->current_offset,
header.ckSize + 8, &flow_ret);
if (flow_ret != GST_FLOW_OK)
goto pause;
if (parse->srcpad == NULL) {
if (!gst_wavpack_parse_create_src_pad (parse, buf, &header)) {
GST_ERROR_OBJECT (parse, "Failed to create src pad");
flow_ret = GST_FLOW_ERROR;
goto pause;
}
}
if (header.flags & INITIAL_BLOCK)
gst_wavpack_parse_index_append_entry (parse, parse->current_offset,
header.block_index, header.block_samples);
flow_ret = gst_wavpack_parse_push_buffer (parse, buf, &header);
if (flow_ret != GST_FLOW_OK)
goto pause;
return;
pause:
{
const gchar *reason = gst_flow_get_name (flow_ret);
GST_LOG_OBJECT (parse, "pausing task, reason %s", reason);
gst_pad_pause_task (parse->sinkpad);
if (flow_ret == GST_FLOW_UNEXPECTED && parse->srcpad) {
if (parse->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (parse, "Sending segment done");
if ((stop = parse->segment.stop) == -1)
stop = parse->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (parse),
gst_message_new_segment_done (GST_OBJECT_CAST (parse),
parse->segment.format, stop));
} else {
GST_LOG_OBJECT (parse, "Sending EOS, at end of stream");
gst_pad_push_event (parse->srcpad, gst_event_new_eos ());
}
} else if (flow_ret == GST_FLOW_NOT_LINKED
|| flow_ret < GST_FLOW_UNEXPECTED) {
GST_ELEMENT_ERROR (parse, STREAM, FAILED,
(_("Internal data stream error.")), ("stream stopped, reason %s",
reason));
if (parse->srcpad)
gst_pad_push_event (parse->srcpad, gst_event_new_eos ());
}
return;
}
}
static gboolean
gst_wavpack_parse_resync_adapter (GstAdapter * adapter)
{
const guint8 *buf, *marker;
guint avail = gst_adapter_available (adapter);
if (avail < 4)
return FALSE;
/* if the marker is at the beginning don't do the expensive search */
buf = gst_adapter_peek (adapter, 4);
if (memcmp (buf, "wvpk", 4) == 0)
return TRUE;
if (avail == 4)
return FALSE;
/* search for the marker in the complete content of the adapter */
buf = gst_adapter_peek (adapter, avail);
if (buf && (marker = gst_wavpack_parse_find_marker ((guint8 *) buf, avail))) {
gst_adapter_flush (adapter, marker - buf);
return TRUE;
}
/* flush everything except the last 4 bytes. they could contain
* the start of a new marker */
gst_adapter_flush (adapter, avail - 4);
return FALSE;
}
static GstFlowReturn
gst_wavpack_parse_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (GST_PAD_PARENT (pad));
GstFlowReturn ret = GST_FLOW_OK;
WavpackHeader wph;
const guint8 *tmp_buf;
if (!wvparse->adapter) {
wvparse->adapter = gst_adapter_new ();
}
if (GST_BUFFER_IS_DISCONT (buf)) {
gst_adapter_clear (wvparse->adapter);
wvparse->discont = TRUE;
}
gst_adapter_push (wvparse->adapter, buf);
if (gst_adapter_available (wvparse->adapter) < sizeof (WavpackHeader))
return ret;
if (!gst_wavpack_parse_resync_adapter (wvparse->adapter))
return ret;
tmp_buf = gst_adapter_peek (wvparse->adapter, sizeof (WavpackHeader));
gst_wavpack_read_header (&wph, (guint8 *) tmp_buf);
while (gst_adapter_available (wvparse->adapter) >= wph.ckSize + 4 * 1 + 4) {
GstBuffer *outbuf =
gst_adapter_take_buffer (wvparse->adapter, wph.ckSize + 4 * 1 + 4);
if (!outbuf)
return GST_FLOW_ERROR;
if (wvparse->srcpad == NULL) {
if (!gst_wavpack_parse_create_src_pad (wvparse, outbuf, &wph)) {
GST_ERROR_OBJECT (wvparse, "Failed to create src pad");
ret = GST_FLOW_ERROR;
break;
}
}
ret = gst_wavpack_parse_push_buffer (wvparse, outbuf, &wph);
if (ret != GST_FLOW_OK)
break;
if (gst_adapter_available (wvparse->adapter) >= sizeof (WavpackHeader)) {
tmp_buf = gst_adapter_peek (wvparse->adapter, sizeof (WavpackHeader));
if (!gst_wavpack_parse_resync_adapter (wvparse->adapter))
break;
gst_wavpack_read_header (&wph, (guint8 *) tmp_buf);
}
}
return ret;
}
static GstStateChangeReturn
gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&wvparse->segment, GST_FORMAT_DEFAULT);
wvparse->segment.last_stop = 0;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavpack_parse_reset (wvparse);
break;
default:
break;
}
return ret;
}
static gboolean
gst_wavpack_parse_sink_activate (GstPad * sinkpad)
{
if (gst_pad_check_pull_range (sinkpad)) {
return gst_pad_activate_pull (sinkpad, TRUE);
} else {
return gst_pad_activate_push (sinkpad, TRUE);
}
}
static gboolean
gst_wavpack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
gboolean result;
if (active) {
result = gst_pad_start_task (sinkpad,
(GstTaskFunction) gst_wavpack_parse_loop, GST_PAD_PARENT (sinkpad));
} else {
result = gst_pad_stop_task (sinkpad);
}
return result;
}
gboolean
gst_wavpack_parse_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackparse",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_PARSE)) {
return FALSE;
}
GST_DEBUG_CATEGORY_INIT (gst_wavpack_parse_debug, "wavpack_parse", 0,
"Wavpack file parser");
return TRUE;
}
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