tizen 2.3 releasetizen_2.3_releasesubmit/tizen_2.3/20150202.062707tizen_2.3

13 files changed, 4651 insertions, 0 deletions

diff --git a/ext/wavpack/Makefile.am b/ext/wavpack/Makefile.am
new file mode 100644
index 0000000..70d2431
--- /dev/null
+++ b/ext/wavpack/Makefile.am
@@ -0,0 +1,24 @@
+plugin_LTLIBRARIES = libgstwavpack.la
+
+libgstwavpack_la_SOURCES = \
+				gstwavpack.c \
+				gstwavpackcommon.c \
+				gstwavpackparse.c \
+				gstwavpackdec.c \
+				gstwavpackenc.c \
+				gstwavpackstreamreader.c
+
+libgstwavpack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \
+				$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(WAVPACK_CFLAGS)
+libgstwavpack_la_LIBADD =  $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) \
+				$(GST_BASE_LIBS) $(GST_LIBS) $(WAVPACK_LIBS)
+libgstwavpack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstwavpack_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = \
+		gstwavpackparse.h \
+		gstwavpackdec.h \
+		gstwavpackenc.h \
+		gstwavpackcommon.h \
+		gstwavpackstreamreader.h
+
diff --git a/ext/wavpack/Makefile.in b/ext/wavpack/Makefile.in
new file mode 100644
index 0000000..4eb1dd1
--- /dev/null
+++ b/ext/wavpack/Makefile.in
@@ -0,0 +1,868 @@
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+# @configure_input@
+
+# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
+# 2003, 2004, 2005, 2006, 2007, 2008, 2009, 2010, 2011 Free Software
+# Foundation, Inc.
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
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+
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+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-am
+all-am: Makefile $(LTLIBRARIES) $(HEADERS)
+installdirs:
+	for dir in "$(DESTDIR)$(plugindir)"; do \
+	  test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+	done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-pluginLTLIBRARIES \
+	mostlyclean-am
+
+distclean: distclean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-pluginLTLIBRARIES
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-pluginLTLIBRARIES
+
+.MAKE: install-am install-strip
+
+.PHONY: CTAGS GTAGS all all-am check check-am clean clean-generic \
+	clean-libtool clean-pluginLTLIBRARIES ctags distclean \
+	distclean-compile distclean-generic distclean-libtool \
+	distclean-tags distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-pluginLTLIBRARIES \
+	install-ps install-ps-am install-strip installcheck \
+	installcheck-am installdirs maintainer-clean \
+	maintainer-clean-generic mostlyclean mostlyclean-compile \
+	mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
+	tags uninstall uninstall-am uninstall-pluginLTLIBRARIES
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/ext/wavpack/gstwavpack.c b/ext/wavpack/gstwavpack.c
new file mode 100644
index 0000000..b01f443
--- /dev/null
+++ b/ext/wavpack/gstwavpack.c
@@ -0,0 +1,56 @@
+/* GStreamer wavpack plugin
+ * (c) 2004 Arwed v. Merkatz <v.merkatz@gmx.net>
+ *
+ * gstwavpack.c: plugin loader
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+
+#include "gstwavpackparse.h"
+#include "gstwavpackdec.h"
+#include "gstwavpackenc.h"
+
+/* debug category for common code */
+GST_DEBUG_CATEGORY (wavpack_debug);
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  GST_DEBUG_CATEGORY_INIT (wavpack_debug, "wavpack", 0, "Wavpack elements");
+
+#ifdef ENABLE_NLS
+  GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
+      LOCALEDIR);
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+  bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif
+
+  return (gst_wavpack_parse_plugin_init (plugin)
+      && gst_wavpack_dec_plugin_init (plugin)
+      && gst_wavpack_enc_plugin_init (plugin));
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "wavpack",
+    "Wavpack lossless/lossy audio format handling",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/ext/wavpack/gstwavpackcommon.c b/ext/wavpack/gstwavpackcommon.c
new file mode 100644
index 0000000..252b64c
--- /dev/null
+++ b/ext/wavpack/gstwavpackcommon.c
@@ -0,0 +1,282 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * Copyright (c) 1998 - 2005 Conifer Software
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackcommon.c: common helper functions
+ * 
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstwavpackcommon.h"
+#include <string.h>
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+
+GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
+#define GST_CAT_DEFAULT wavpack_debug
+
+gboolean
+gst_wavpack_read_header (WavpackHeader * header, guint8 * buf)
+{
+  g_memmove (header, buf, sizeof (WavpackHeader));
+
+#ifndef WAVPACK_OLD_API
+  WavpackLittleEndianToNative (header, (char *) WavpackHeaderFormat);
+#else
+  little_endian_to_native (header, WavpackHeaderFormat);
+#endif
+
+  return (memcmp (header->ckID, "wvpk", 4) == 0);
+}
+
+/* inspired by the original one in wavpack */
+gboolean
+gst_wavpack_read_metadata (GstWavpackMetadata * wpmd, guint8 * header_data,
+    guint8 ** p_data)
+{
+  WavpackHeader hdr;
+  guint8 *end;
+
+  gst_wavpack_read_header (&hdr, header_data);
+  end = header_data + hdr.ckSize + 8;
+
+  if (end - *p_data < 2)
+    return FALSE;
+
+  wpmd->id = GST_READ_UINT8 (*p_data);
+  wpmd->byte_length = 2 * (guint) GST_READ_UINT8 (*p_data + 1);
+
+  *p_data += 2;
+
+  if ((wpmd->id & ID_LARGE) == ID_LARGE) {
+    guint extra;
+
+    wpmd->id &= ~ID_LARGE;
+
+    if (end - *p_data < 2)
+      return FALSE;
+
+    extra = GST_READ_UINT16_LE (*p_data);
+    wpmd->byte_length += (extra << 9);
+    *p_data += 2;
+  }
+
+  if ((wpmd->id & ID_ODD_SIZE) == ID_ODD_SIZE) {
+    wpmd->id &= ~ID_ODD_SIZE;
+    --wpmd->byte_length;
+  }
+
+  if (wpmd->byte_length > 0) {
+    if (end - *p_data < wpmd->byte_length + (wpmd->byte_length & 1)) {
+      wpmd->data = NULL;
+      return FALSE;
+    }
+
+    wpmd->data = *p_data;
+    *p_data += wpmd->byte_length + (wpmd->byte_length & 1);
+  } else {
+    wpmd->data = NULL;
+  }
+
+  return TRUE;
+}
+
+gint
+gst_wavpack_get_default_channel_mask (gint nchannels)
+{
+  gint channel_mask = 0;
+
+  /* Set the default channel mask for the given number of channels.
+   * It's the same as for WAVE_FORMAT_EXTENDED:
+   * http://www.microsoft.com/whdc/device/audio/multichaud.mspx
+   */
+  switch (nchannels) {
+    case 11:
+      channel_mask |= 0x00400;
+      channel_mask |= 0x00200;
+    case 9:
+      channel_mask |= 0x00100;
+    case 8:
+      channel_mask |= 0x00080;
+      channel_mask |= 0x00040;
+    case 6:
+      channel_mask |= 0x00020;
+      channel_mask |= 0x00010;
+    case 4:
+      channel_mask |= 0x00008;
+    case 3:
+      channel_mask |= 0x00004;
+    case 2:
+      channel_mask |= 0x00002;
+      channel_mask |= 0x00001;
+      break;
+    case 1:
+      /* For mono use front center */
+      channel_mask |= 0x00004;
+      break;
+  }
+
+  return channel_mask;
+}
+
+static const struct
+{
+  const guint32 ms_mask;
+  const GstAudioChannelPosition gst_pos;
+} layout_mapping[] = {
+  {
+  0x00001, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT}, {
+  0x00002, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}, {
+  0x00004, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER}, {
+  0x00008, GST_AUDIO_CHANNEL_POSITION_LFE}, {
+  0x00010, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT}, {
+  0x00020, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT}, {
+  0x00040, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER}, {
+  0x00080, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER}, {
+  0x00100, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER}, {
+  0x00200, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT}, {
+  0x00400, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT}, {
+  0x00800, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_CENTER       */
+  {
+  0x01000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_LEFT   */
+  {
+  0x02000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_CENTER */
+  {
+  0x04000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_FRONT_RIGHT  */
+  {
+  0x08000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_LEFT    */
+  {
+  0x10000, GST_AUDIO_CHANNEL_POSITION_INVALID}, /* TOP_BACK_CENTER  */
+  {
+  0x20000, GST_AUDIO_CHANNEL_POSITION_INVALID}  /* TOP_BACK_RIGHT   */
+};
+
+#define MAX_CHANNEL_POSITIONS G_N_ELEMENTS (layout_mapping)
+
+gboolean
+gst_wavpack_set_channel_layout (GstCaps * caps, gint layout)
+{
+  GstAudioChannelPosition pos[MAX_CHANNEL_POSITIONS];
+  GstStructure *s;
+  gint num_channels, i, p;
+
+  s = gst_caps_get_structure (caps, 0);
+  if (!gst_structure_get_int (s, "channels", &num_channels))
+    g_return_val_if_reached (FALSE);
+
+  if (num_channels == 1 && layout == 0x00004) {
+    pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO;
+    gst_audio_set_channel_positions (s, pos);
+    return TRUE;
+  }
+
+  p = 0;
+  for (i = 0; i < MAX_CHANNEL_POSITIONS; ++i) {
+    if ((layout & layout_mapping[i].ms_mask) != 0) {
+      if (p >= num_channels) {
+        GST_WARNING ("More bits set in the channel layout map than there "
+            "are channels! Broken file");
+        return FALSE;
+      }
+      if (layout_mapping[i].gst_pos == GST_AUDIO_CHANNEL_POSITION_INVALID) {
+        GST_WARNING ("Unsupported channel position (mask 0x%08x) in channel "
+            "layout map - ignoring those channels", layout_mapping[i].ms_mask);
+        /* what to do? just ignore it and let downstream deal with a channel
+         * layout that has INVALID positions in it for now ... */
+      }
+      pos[p] = layout_mapping[i].gst_pos;
+      ++p;
+    }
+  }
+
+  if (p != num_channels) {
+    GST_WARNING ("Only %d bits set in the channel layout map, but there are "
+        "supposed to be %d channels! Broken file", p, num_channels);
+    return FALSE;
+  }
+
+  gst_audio_set_channel_positions (s, pos);
+  return TRUE;
+}
+
+GstAudioChannelPosition *
+gst_wavpack_get_default_channel_positions (gint nchannels)
+{
+  GstAudioChannelPosition *pos = g_new (GstAudioChannelPosition, nchannels);
+  gint i;
+
+  if (nchannels == 1) {
+    pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+    return pos;
+  }
+
+  for (i = 0; i < nchannels; i++)
+    pos[i] = layout_mapping[i].gst_pos;
+
+  return pos;
+}
+
+gint
+gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition * pos,
+    gint nchannels)
+{
+  gint channel_mask = 0;
+  gint i, j;
+
+  if (nchannels == 1 && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_MONO) {
+    channel_mask = 0x00000004;
+    return channel_mask;
+  }
+
+  /* FIXME: not exactly efficient but otherwise we need an inverse
+   * mapping table too */
+  for (i = 0; i < nchannels; i++) {
+    for (j = 0; j < MAX_CHANNEL_POSITIONS; j++) {
+      if (pos[i] == layout_mapping[j].gst_pos) {
+        channel_mask |= layout_mapping[j].ms_mask;
+        break;
+      }
+    }
+  }
+
+  return channel_mask;
+}
+
+gboolean
+gst_wavpack_set_channel_mapping (GstAudioChannelPosition * pos, gint nchannels,
+    gint8 * channel_mapping)
+{
+  gint i, j;
+  gboolean ret = TRUE;
+
+  for (i = 0; i < nchannels; i++) {
+    for (j = 0; j < MAX_CHANNEL_POSITIONS; j++) {
+      if (pos[i] == layout_mapping[j].gst_pos) {
+        channel_mapping[i] = j;
+        ret &= (i == j);
+        break;
+      }
+    }
+  }
+
+  return !ret;
+}
diff --git a/ext/wavpack/gstwavpackcommon.h b/ext/wavpack/gstwavpackcommon.h
new file mode 100644
index 0000000..6a9e516
--- /dev/null
+++ b/ext/wavpack/gstwavpackcommon.h
@@ -0,0 +1,75 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackcommon.h: common helper functions
+ *
+ * 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.
+ */
+
+#ifndef __GST_WAVPACK_COMMON_H__
+#define __GST_WAVPACK_COMMON_H__
+
+#include <gst/gst.h>
+#include <gst/audio/multichannel.h>
+#include <wavpack/wavpack.h>
+
+typedef struct
+{
+  guint32 byte_length;
+  guint8 *data;
+  guint8 id;
+} GstWavpackMetadata;
+
+#define ID_UNIQUE               0x3f
+#define ID_OPTIONAL_DATA        0x20
+#define ID_ODD_SIZE             0x40
+#define ID_LARGE                0x80
+
+#define ID_DUMMY                0x0
+#define ID_ENCODER_INFO         0x1
+#define ID_DECORR_TERMS         0x2
+#define ID_DECORR_WEIGHTS       0x3
+#define ID_DECORR_SAMPLES       0x4
+#define ID_ENTROPY_VARS         0x5
+#define ID_HYBRID_PROFILE       0x6
+#define ID_SHAPING_WEIGHTS      0x7
+#define ID_FLOAT_INFO           0x8
+#define ID_INT32_INFO           0x9
+#define ID_WV_BITSTREAM         0xa
+#define ID_WVC_BITSTREAM        0xb
+#define ID_WVX_BITSTREAM        0xc
+#define ID_CHANNEL_INFO         0xd
+
+#define ID_RIFF_HEADER          (ID_OPTIONAL_DATA | 0x1)
+#define ID_RIFF_TRAILER         (ID_OPTIONAL_DATA | 0x2)
+#define ID_REPLAY_GAIN          (ID_OPTIONAL_DATA | 0x3)
+#define ID_CUESHEET             (ID_OPTIONAL_DATA | 0x4)
+#define ID_CONFIG_BLOCK         (ID_OPTIONAL_DATA | 0x5)
+#define ID_MD5_CHECKSUM         (ID_OPTIONAL_DATA | 0x6)
+#define ID_SAMPLE_RATE          (ID_OPTIONAL_DATA | 0x7)
+
+
+gboolean gst_wavpack_read_header (WavpackHeader * header, guint8 * buf);
+gboolean gst_wavpack_read_metadata (GstWavpackMetadata * meta,
+    guint8 * header_data, guint8 ** p_data);
+gint gst_wavpack_get_default_channel_mask (gint nchannels);
+gboolean gst_wavpack_set_channel_layout (GstCaps * caps, gint layout);
+GstAudioChannelPosition *gst_wavpack_get_default_channel_positions (gint nchannels);
+gint gst_wavpack_get_channel_mask_from_positions (GstAudioChannelPosition *pos, gint nchannels);
+gboolean gst_wavpack_set_channel_mapping (GstAudioChannelPosition *pos, gint nchannels, gint8 *channel_mapping);
+
+#endif
diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c
new file mode 100644
index 0000000..4ba242d
--- /dev/null
+++ b/ext/wavpack/gstwavpackdec.c
@@ -0,0 +1,512 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * Copyright (c) 2006 Edward Hervey <bilboed@gmail.com>
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackdec.c: raw Wavpack bitstream decoder
+ *
+ * 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-wavpackdec
+ *
+ * WavpackDec decodes framed (for example by the WavpackParse element)
+ * Wavpack streams and decodes them to raw audio.
+ * <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 ! audioconvert ! audioresample ! autoaudiosink
+ * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
+ * tries to play it back using an automatically found audio sink.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.h>
+
+#include <math.h>
+#include <string.h>
+
+#include <wavpack/wavpack.h>
+#include "gstwavpackdec.h"
+#include "gstwavpackcommon.h"
+#include "gstwavpackstreamreader.h"
+
+
+#define WAVPACK_DEC_MAX_ERRORS 16
+
+GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
+#define GST_CAT_DEFAULT gst_wavpack_dec_debug
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-wavpack, "
+        "width = (int) [ 1, 32 ], "
+        "channels = (int) [ 1, 8 ], "
+        "rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
+    );
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "width = (int) 32, "
+        "depth = (int) [ 1, 32 ], "
+        "channels = (int) [ 1, 8 ], "
+        "rate = (int) [ 6000, 192000 ], "
+        "endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
+    );
+
+static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
+static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
+static void gst_wavpack_dec_finalize (GObject * object);
+static GstStateChangeReturn gst_wavpack_dec_change_state (GstElement * element,
+    GstStateChange transition);
+static void gst_wavpack_dec_post_tags (GstWavpackDec * dec);
+
+GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT);
+
+static void
+gst_wavpack_dec_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, &sink_factory);
+  gst_element_class_set_details_simple (element_class, "Wavpack audio decoder",
+      "Codec/Decoder/Audio",
+      "Decodes Wavpack audio data",
+      "Arwed v. Merkatz <v.merkatz@gmx.net>, "
+      "Sebastian Dröge <slomo@circular-chaos.org>");
+}
+
+static void
+gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
+  gobject_class->finalize = gst_wavpack_dec_finalize;
+}
+
+static void
+gst_wavpack_dec_reset (GstWavpackDec * dec)
+{
+  dec->wv_id.buffer = NULL;
+  dec->wv_id.position = dec->wv_id.length = 0;
+
+  dec->error_count = 0;
+
+  dec->channels = 0;
+  dec->channel_mask = 0;
+  dec->sample_rate = 0;
+  dec->depth = 0;
+
+  gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+  dec->next_block_index = 0;
+}
+
+static void
+gst_wavpack_dec_init (GstWavpackDec * dec, GstWavpackDecClass * gklass)
+{
+  dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
+  gst_pad_set_chain_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
+  gst_pad_set_setcaps_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_set_caps));
+  gst_pad_set_event_function (dec->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
+  gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
+
+  dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
+  gst_pad_use_fixed_caps (dec->srcpad);
+  gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
+
+  dec->context = NULL;
+  dec->stream_reader = gst_wavpack_stream_reader_new ();
+
+  gst_wavpack_dec_reset (dec);
+}
+
+static void
+gst_wavpack_dec_finalize (GObject * object)
+{
+  GstWavpackDec *dec = GST_WAVPACK_DEC (object);
+
+  g_free (dec->stream_reader);
+  dec->stream_reader = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
+{
+  GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
+  GstStructure *structure = gst_caps_get_structure (caps, 0);
+
+  /* Check if we can set the caps here already */
+  if (gst_structure_get_int (structure, "channels", &dec->channels) &&
+      gst_structure_get_int (structure, "rate", &dec->sample_rate) &&
+      gst_structure_get_int (structure, "width", &dec->depth)) {
+    GstCaps *caps;
+    GstAudioChannelPosition *pos;
+
+    caps = gst_caps_new_simple ("audio/x-raw-int",
+        "rate", G_TYPE_INT, dec->sample_rate,
+        "channels", G_TYPE_INT, dec->channels,
+        "depth", G_TYPE_INT, dec->depth,
+        "width", G_TYPE_INT, 32,
+        "endianness", G_TYPE_INT, G_BYTE_ORDER,
+        "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+    /* If we already have the channel layout set from upstream
+     * take this */
+    if (gst_structure_has_field (structure, "channel-positions")) {
+      pos = gst_audio_get_channel_positions (structure);
+      if (pos != NULL && dec->channels > 2) {
+        GstStructure *new_str = gst_caps_get_structure (caps, 0);
+
+        gst_audio_set_channel_positions (new_str, pos);
+        dec->channel_mask =
+            gst_wavpack_get_channel_mask_from_positions (pos, dec->channels);
+      }
+
+      if (pos != NULL)
+        g_free (pos);
+    }
+
+    GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
+
+    /* should always succeed */
+    gst_pad_set_caps (dec->srcpad, caps);
+    gst_caps_unref (caps);
+
+    /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
+     * is decoded or after the format has changed */
+    gst_wavpack_dec_post_tags (dec);
+  }
+
+  gst_object_unref (dec);
+
+  return TRUE;
+}
+
+static void
+gst_wavpack_dec_post_tags (GstWavpackDec * dec)
+{
+  GstTagList *list;
+  GstFormat format_time = GST_FORMAT_TIME, format_bytes = GST_FORMAT_BYTES;
+  gint64 duration, size;
+
+  list = gst_tag_list_new ();
+
+  gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_AUDIO_CODEC, "Wavpack", NULL);
+
+  /* try to estimate the average bitrate */
+  if (gst_pad_query_peer_duration (dec->sinkpad, &format_bytes, &size) &&
+      gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
+      size > 0 && duration > 0) {
+    guint64 bitrate;
+
+    bitrate = gst_util_uint64_scale (size, 8 * GST_SECOND, duration);
+    gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
+        (guint) bitrate, NULL);
+  }
+
+  gst_element_post_message (GST_ELEMENT (dec),
+      gst_message_new_tag (GST_OBJECT (dec), list));
+}
+
+static GstFlowReturn
+gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstWavpackDec *dec;
+  GstBuffer *outbuf = NULL;
+  GstFlowReturn ret = GST_FLOW_OK;
+  WavpackHeader wph;
+  int32_t decoded, unpacked_size;
+  gboolean format_changed;
+
+  dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
+
+  /* check input, we only accept framed input with complete chunks */
+  if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
+    goto input_not_framed;
+
+  if (!gst_wavpack_read_header (&wph, GST_BUFFER_DATA (buf)))
+    goto invalid_header;
+
+  if (GST_BUFFER_SIZE (buf) < wph.ckSize + 4 * 1 + 4)
+    goto input_not_framed;
+
+  if (!(wph.flags & INITIAL_BLOCK))
+    goto input_not_framed;
+
+  dec->wv_id.buffer = GST_BUFFER_DATA (buf);
+  dec->wv_id.length = GST_BUFFER_SIZE (buf);
+  dec->wv_id.position = 0;
+
+  /* create a new wavpack context if there is none yet but if there
+   * was already one (i.e. caps were set on the srcpad) check whether
+   * the new one has the same caps */
+  if (!dec->context) {
+    gchar error_msg[80];
+
+    dec->context = WavpackOpenFileInputEx (dec->stream_reader,
+        &dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
+
+    if (!dec->context) {
+      GST_WARNING ("Couldn't decode buffer: %s", error_msg);
+      dec->error_count++;
+      if (dec->error_count <= WAVPACK_DEC_MAX_ERRORS) {
+        goto out;               /* just return OK for now */
+      } else {
+        goto decode_error;
+      }
+    }
+  }
+
+  g_assert (dec->context != NULL);
+
+  dec->error_count = 0;
+
+  format_changed =
+      (dec->sample_rate != WavpackGetSampleRate (dec->context)) ||
+      (dec->channels != WavpackGetNumChannels (dec->context)) ||
+      (dec->depth != WavpackGetBitsPerSample (dec->context)) ||
+#ifdef WAVPACK_OLD_API
+      (dec->channel_mask != dec->context->config.channel_mask);
+#else
+      (dec->channel_mask != WavpackGetChannelMask (dec->context));
+#endif
+
+  if (!GST_PAD_CAPS (dec->srcpad) || format_changed) {
+    GstCaps *caps;
+    gint channel_mask;
+
+    dec->sample_rate = WavpackGetSampleRate (dec->context);
+    dec->channels = WavpackGetNumChannels (dec->context);
+    dec->depth = WavpackGetBitsPerSample (dec->context);
+
+    caps = gst_caps_new_simple ("audio/x-raw-int",
+        "rate", G_TYPE_INT, dec->sample_rate,
+        "channels", G_TYPE_INT, dec->channels,
+        "depth", G_TYPE_INT, dec->depth,
+        "width", G_TYPE_INT, 32,
+        "endianness", G_TYPE_INT, G_BYTE_ORDER,
+        "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+#ifdef WAVPACK_OLD_API
+    channel_mask = dec->context->config.channel_mask;
+#else
+    channel_mask = WavpackGetChannelMask (dec->context);
+#endif
+    if (channel_mask == 0)
+      channel_mask = gst_wavpack_get_default_channel_mask (dec->channels);
+
+    dec->channel_mask = channel_mask;
+
+    /* Only set the channel layout for more than two channels
+     * otherwise things break unfortunately */
+    if (channel_mask != 0 && dec->channels > 2)
+      if (!gst_wavpack_set_channel_layout (caps, channel_mask))
+        GST_WARNING_OBJECT (dec, "Failed to set channel layout");
+
+    GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
+
+    /* should always succeed */
+    gst_pad_set_caps (dec->srcpad, caps);
+    gst_caps_unref (caps);
+
+    /* send GST_TAG_AUDIO_CODEC and GST_TAG_BITRATE tags before something
+     * is decoded or after the format has changed */
+    gst_wavpack_dec_post_tags (dec);
+  }
+
+  /* alloc output buffer */
+  unpacked_size = 4 * wph.block_samples * dec->channels;
+  ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
+      unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
+
+  if (ret != GST_FLOW_OK)
+    goto out;
+
+  gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
+
+  /* If we got a DISCONT buffer forward the flag. Nothing else
+   * has to be done as libwavpack doesn't store state between
+   * Wavpack blocks */
+  if (GST_BUFFER_IS_DISCONT (buf) || dec->next_block_index != wph.block_index)
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+
+  dec->next_block_index = wph.block_index + wph.block_samples;
+
+  /* decode */
+  decoded = WavpackUnpackSamples (dec->context,
+      (int32_t *) GST_BUFFER_DATA (outbuf), wph.block_samples);
+  if (decoded != wph.block_samples)
+    goto decode_error;
+
+  if ((outbuf = gst_audio_buffer_clip (outbuf, &dec->segment,
+              dec->sample_rate, 4 * dec->channels))) {
+    GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
+    ret = gst_pad_push (dec->srcpad, outbuf);
+  }
+
+out:
+
+  if (G_UNLIKELY (ret != GST_FLOW_OK)) {
+    GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (ret));
+  }
+
+  gst_buffer_unref (buf);
+
+  return ret;
+
+/* ERRORS */
+input_not_framed:
+  {
+    GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Expected framed input"));
+    gst_buffer_unref (buf);
+    return GST_FLOW_ERROR;
+  }
+invalid_header:
+  {
+    GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL), ("Invalid wavpack header"));
+    gst_buffer_unref (buf);
+    return GST_FLOW_ERROR;
+  }
+decode_error:
+  {
+    const gchar *reason = "unknown";
+
+    if (dec->context) {
+#ifdef WAVPACK_OLD_API
+      reason = dec->context->error_message;
+#else
+      reason = WavpackGetErrorMessage (dec->context);
+#endif
+    } else {
+      reason = "couldn't create decoder context";
+    }
+    GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
+        ("Failed to decode wavpack stream: %s", reason));
+    if (outbuf)
+      gst_buffer_unref (outbuf);
+    gst_buffer_unref (buf);
+    return GST_FLOW_ERROR;
+  }
+}
+
+static gboolean
+gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
+
+  GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:{
+      GstFormat fmt;
+      gboolean is_update;
+      gint64 start, end, base;
+      gdouble rate;
+
+      gst_event_parse_new_segment (event, &is_update, &rate, &fmt, &start,
+          &end, &base);
+      if (fmt == GST_FORMAT_TIME) {
+        GST_DEBUG ("Got NEWSEGMENT event in GST_FORMAT_TIME, passing on (%"
+            GST_TIME_FORMAT " - %" GST_TIME_FORMAT ")", GST_TIME_ARGS (start),
+            GST_TIME_ARGS (end));
+        gst_segment_set_newsegment (&dec->segment, is_update, rate, fmt,
+            start, end, base);
+      } else {
+        gst_segment_init (&dec->segment, GST_FORMAT_TIME);
+      }
+      break;
+    }
+    default:
+      break;
+  }
+
+  gst_object_unref (dec);
+  return gst_pad_event_default (pad, event);
+}
+
+static GstStateChangeReturn
+gst_wavpack_dec_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstWavpackDec *dec = GST_WAVPACK_DEC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      if (dec->context) {
+        WavpackCloseFile (dec->context);
+        dec->context = NULL;
+      }
+
+      gst_wavpack_dec_reset (dec);
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+gboolean
+gst_wavpack_dec_plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "wavpackdec",
+          GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
+    return FALSE;
+  GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpack_dec", 0,
+      "Wavpack decoder");
+  return TRUE;
+}
diff --git a/ext/wavpack/gstwavpackdec.h b/ext/wavpack/gstwavpackdec.h
new file mode 100644
index 0000000..eb6e4c3
--- /dev/null
+++ b/ext/wavpack/gstwavpackdec.h
@@ -0,0 +1,80 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2004 Arwed v. Merkatz <v.merkatz@gmx.net>
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackdec.h: raw Wavpack bitstream decoder
+ *
+ * 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.
+ */
+
+#ifndef __GST_WAVPACK_DEC_H__
+#define __GST_WAVPACK_DEC_H__
+
+#include <gst/gst.h>
+
+#include <wavpack/wavpack.h>
+
+#include "gstwavpackstreamreader.h"
+
+G_BEGIN_DECLS
+#define GST_TYPE_WAVPACK_DEC \
+  (gst_wavpack_dec_get_type())
+#define GST_WAVPACK_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_DEC,GstWavpackDec))
+#define GST_WAVPACK_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_DEC,GstWavpackDecClass))
+#define GST_IS_WAVPACK_DEC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_DEC))
+#define GST_IS_WAVPACK_DEC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_DEC))
+typedef struct _GstWavpackDec GstWavpackDec;
+typedef struct _GstWavpackDecClass GstWavpackDecClass;
+
+struct _GstWavpackDec
+{
+  GstElement element;
+
+  /*< private > */
+  GstPad *sinkpad;
+  GstPad *srcpad;
+
+  WavpackContext *context;
+  WavpackStreamReader *stream_reader;
+
+  read_id wv_id;
+
+  GstSegment segment;           /* used for clipping, TIME format */
+  guint32 next_block_index;
+
+  gint sample_rate;
+  gint depth;
+  gint channels;
+  gint channel_mask;
+
+  gint error_count;
+};
+
+struct _GstWavpackDecClass
+{
+  GstElementClass parent;
+};
+
+GType gst_wavpack_dec_get_type (void);
+
+gboolean gst_wavpack_dec_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+#endif /* __GST_WAVPACK_DEC_H__ */
diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c
new file mode 100644
index 0000000..a22dd23
--- /dev/null
+++ b/ext/wavpack/gstwavpackenc.c
@@ -0,0 +1,1049 @@
+/* GStreamer Wavpack encoder plugin
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackdec.c: Wavpack audio encoder
+ *
+ * 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-wavpackenc
+ *
+ * WavpackEnc encodes raw audio into a framed Wavpack 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 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
+ * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
+ * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
+ * |[
+ * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
+ * lossless encoding (the file output will be fairly large).
+ * |[
+ * gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
+ * lossy encoding at a certain bitrate (the file will be fairly small).
+ * </refsect2>
+ */
+
+/*
+ * TODO: - add 32 bit float mode. CONFIG_FLOAT_DATA
+ */
+
+#include <string.h>
+#include <gst/gst.h>
+#include <glib/gprintf.h>
+
+#include <wavpack/wavpack.h>
+#include "gstwavpackenc.h"
+#include "gstwavpackcommon.h"
+
+static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
+static int gst_wavpack_enc_push_block (void *id, void *data, int32_t count);
+static gboolean gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event);
+static GstStateChangeReturn gst_wavpack_enc_change_state (GstElement * element,
+    GstStateChange transition);
+static void gst_wavpack_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_wavpack_enc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+enum
+{
+  ARG_0,
+  ARG_MODE,
+  ARG_BITRATE,
+  ARG_BITSPERSAMPLE,
+  ARG_CORRECTION_MODE,
+  ARG_MD5,
+  ARG_EXTRA_PROCESSING,
+  ARG_JOINT_STEREO_MODE
+};
+
+GST_DEBUG_CATEGORY_STATIC (gst_wavpack_enc_debug);
+#define GST_CAT_DEFAULT gst_wavpack_enc_debug
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "width = (int) 32, "
+        "depth = (int) [ 1, 32], "
+        "endianness = (int) BYTE_ORDER, "
+        "channels = (int) [ 1, 8 ], "
+        "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
+    );
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-wavpack, "
+        "width = (int) [ 1, 32 ], "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
+    );
+
+static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) TRUE")
+    );
+
+enum
+{
+  GST_WAVPACK_ENC_MODE_VERY_FAST = 0,
+  GST_WAVPACK_ENC_MODE_FAST,
+  GST_WAVPACK_ENC_MODE_DEFAULT,
+  GST_WAVPACK_ENC_MODE_HIGH,
+  GST_WAVPACK_ENC_MODE_VERY_HIGH
+};
+
+#define GST_TYPE_WAVPACK_ENC_MODE (gst_wavpack_enc_mode_get_type ())
+static GType
+gst_wavpack_enc_mode_get_type (void)
+{
+  static GType qtype = 0;
+
+  if (qtype == 0) {
+    static const GEnumValue values[] = {
+#if 0
+      /* Very Fast Compression is not supported yet, but will be supported
+       * in future wavpack versions */
+      {GST_WAVPACK_ENC_MODE_VERY_FAST, "Very Fast Compression", "veryfast"},
+#endif
+      {GST_WAVPACK_ENC_MODE_FAST, "Fast Compression", "fast"},
+      {GST_WAVPACK_ENC_MODE_DEFAULT, "Normal Compression", "normal"},
+      {GST_WAVPACK_ENC_MODE_HIGH, "High Compression", "high"},
+#ifndef WAVPACK_OLD_API
+      {GST_WAVPACK_ENC_MODE_VERY_HIGH, "Very High Compression", "veryhigh"},
+#endif
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncMode", values);
+  }
+  return qtype;
+}
+
+enum
+{
+  GST_WAVPACK_CORRECTION_MODE_OFF = 0,
+  GST_WAVPACK_CORRECTION_MODE_ON,
+  GST_WAVPACK_CORRECTION_MODE_OPTIMIZED
+};
+
+#define GST_TYPE_WAVPACK_ENC_CORRECTION_MODE (gst_wavpack_enc_correction_mode_get_type ())
+static GType
+gst_wavpack_enc_correction_mode_get_type (void)
+{
+  static GType qtype = 0;
+
+  if (qtype == 0) {
+    static const GEnumValue values[] = {
+      {GST_WAVPACK_CORRECTION_MODE_OFF, "Create no correction file", "off"},
+      {GST_WAVPACK_CORRECTION_MODE_ON, "Create correction file", "on"},
+      {GST_WAVPACK_CORRECTION_MODE_OPTIMIZED,
+          "Create optimized correction file", "optimized"},
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
+  }
+  return qtype;
+}
+
+enum
+{
+  GST_WAVPACK_JS_MODE_AUTO = 0,
+  GST_WAVPACK_JS_MODE_LEFT_RIGHT,
+  GST_WAVPACK_JS_MODE_MID_SIDE
+};
+
+#define GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE (gst_wavpack_enc_joint_stereo_mode_get_type ())
+static GType
+gst_wavpack_enc_joint_stereo_mode_get_type (void)
+{
+  static GType qtype = 0;
+
+  if (qtype == 0) {
+    static const GEnumValue values[] = {
+      {GST_WAVPACK_JS_MODE_AUTO, "auto", "auto"},
+      {GST_WAVPACK_JS_MODE_LEFT_RIGHT, "left/right", "leftright"},
+      {GST_WAVPACK_JS_MODE_MID_SIDE, "mid/side", "midside"},
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
+  }
+  return qtype;
+}
+
+static void
+_do_init (GType object_type)
+{
+  const GInterfaceInfo preset_interface_info = {
+    NULL,                       /* interface_init */
+    NULL,                       /* interface_finalize */
+    NULL                        /* interface_data */
+  };
+
+  g_type_add_interface_static (object_type, GST_TYPE_PRESET,
+      &preset_interface_info);
+}
+
+GST_BOILERPLATE_FULL (GstWavpackEnc, gst_wavpack_enc, GstElement,
+    GST_TYPE_ELEMENT, _do_init);
+
+static void
+gst_wavpack_enc_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  /* add pad templates */
+  gst_element_class_add_static_pad_template (element_class, &sink_factory);
+  gst_element_class_add_static_pad_template (element_class, &src_factory);
+  gst_element_class_add_static_pad_template (element_class,
+      &wvcsrc_factory);
+
+  /* set element details */
+  gst_element_class_set_details_simple (element_class, "Wavpack audio encoder",
+      "Codec/Encoder/Audio",
+      "Encodes audio with the Wavpack lossless/lossy audio codec",
+      "Sebastian Dröge <slomo@circular-chaos.org>");
+}
+
+
+static void
+gst_wavpack_enc_class_init (GstWavpackEncClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  /* set state change handler */
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_change_state);
+
+  /* set property handlers */
+  gobject_class->set_property = gst_wavpack_enc_set_property;
+  gobject_class->get_property = gst_wavpack_enc_get_property;
+
+  /* install all properties */
+  g_object_class_install_property (gobject_class, ARG_MODE,
+      g_param_spec_enum ("mode", "Encoding mode",
+          "Speed versus compression tradeoff.",
+          GST_TYPE_WAVPACK_ENC_MODE, GST_WAVPACK_ENC_MODE_DEFAULT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_BITRATE,
+      g_param_spec_uint ("bitrate", "Bitrate",
+          "Try to encode with this average bitrate (bits/sec). "
+          "This enables lossy encoding, values smaller than 24000 disable it again.",
+          0, 9600000, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_BITSPERSAMPLE,
+      g_param_spec_double ("bits-per-sample", "Bits per sample",
+          "Try to encode with this amount of bits per sample. "
+          "This enables lossy encoding, values smaller than 2.0 disable it again.",
+          0.0, 24.0, 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
+      g_param_spec_enum ("correction-mode", "Correction stream mode",
+          "Use this mode for the correction stream. Only works in lossy mode!",
+          GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, GST_WAVPACK_CORRECTION_MODE_OFF,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_MD5,
+      g_param_spec_boolean ("md5", "MD5",
+          "Store MD5 hash of raw samples within the file.", FALSE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
+      g_param_spec_uint ("extra-processing", "Extra processing",
+          "Use better but slower filters for better compression/quality.",
+          0, 6, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, ARG_JOINT_STEREO_MODE,
+      g_param_spec_enum ("joint-stereo-mode", "Joint-Stereo mode",
+          "Use this joint-stereo mode.", GST_TYPE_WAVPACK_ENC_JOINT_STEREO_MODE,
+          GST_WAVPACK_JS_MODE_AUTO,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_wavpack_enc_reset (GstWavpackEnc * enc)
+{
+  /* close and free everything stream related if we already did something */
+  if (enc->wp_context) {
+    WavpackCloseFile (enc->wp_context);
+    enc->wp_context = NULL;
+  }
+  if (enc->wp_config) {
+    g_free (enc->wp_config);
+    enc->wp_config = NULL;
+  }
+  if (enc->first_block) {
+    g_free (enc->first_block);
+    enc->first_block = NULL;
+  }
+  enc->first_block_size = 0;
+  if (enc->md5_context) {
+    g_checksum_free (enc->md5_context);
+    enc->md5_context = NULL;
+  }
+
+  if (enc->pending_buffer) {
+    gst_buffer_unref (enc->pending_buffer);
+    enc->pending_buffer = NULL;
+    enc->pending_offset = 0;
+  }
+
+  /* reset the last returns to GST_FLOW_OK. This is only set to something else
+   * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
+   * so not valid anymore */
+  enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
+
+  /* reset stream information */
+  enc->samplerate = 0;
+  enc->depth = 0;
+  enc->channels = 0;
+  enc->channel_mask = 0;
+  enc->need_channel_remap = FALSE;
+
+  enc->timestamp_offset = GST_CLOCK_TIME_NONE;
+  enc->next_ts = GST_CLOCK_TIME_NONE;
+}
+
+static void
+gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
+{
+  enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
+  gst_pad_set_setcaps_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
+  gst_pad_set_chain_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
+  gst_pad_set_event_function (enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
+  gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
+
+  /* setup src pad */
+  enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
+  gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
+
+  /* initialize object attributes */
+  enc->wp_config = NULL;
+  enc->wp_context = NULL;
+  enc->first_block = NULL;
+  enc->md5_context = NULL;
+  gst_wavpack_enc_reset (enc);
+
+  enc->wv_id.correction = FALSE;
+  enc->wv_id.wavpack_enc = enc;
+  enc->wv_id.passthrough = FALSE;
+  enc->wvc_id.correction = TRUE;
+  enc->wvc_id.wavpack_enc = enc;
+  enc->wvc_id.passthrough = FALSE;
+
+  /* set default values of params */
+  enc->mode = GST_WAVPACK_ENC_MODE_DEFAULT;
+  enc->bitrate = 0;
+  enc->bps = 0.0;
+  enc->correction_mode = GST_WAVPACK_CORRECTION_MODE_OFF;
+  enc->md5 = FALSE;
+  enc->extra_processing = 0;
+  enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
+}
+
+static gboolean
+gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  GstStructure *structure = gst_caps_get_structure (caps, 0);
+  GstAudioChannelPosition *pos;
+
+  if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
+      !gst_structure_get_int (structure, "rate", &enc->samplerate) ||
+      !gst_structure_get_int (structure, "depth", &enc->depth)) {
+    GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+        ("got invalid caps: %" GST_PTR_FORMAT, caps));
+    gst_object_unref (enc);
+    return FALSE;
+  }
+
+  pos = gst_audio_get_channel_positions (structure);
+  /* If one channel is NONE they'll be all undefined */
+  if (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE) {
+    g_free (pos);
+    pos = NULL;
+  }
+
+  if (pos == NULL) {
+    GST_ELEMENT_ERROR (enc, STREAM, FORMAT, (NULL),
+        ("input has no valid channel layout"));
+
+    gst_object_unref (enc);
+    return FALSE;
+  }
+
+  enc->channel_mask =
+      gst_wavpack_get_channel_mask_from_positions (pos, enc->channels);
+  enc->need_channel_remap =
+      gst_wavpack_set_channel_mapping (pos, enc->channels,
+      enc->channel_mapping);
+  g_free (pos);
+
+  /* set fixed src pad caps now that we know what we will get */
+  caps = gst_caps_new_simple ("audio/x-wavpack",
+      "channels", G_TYPE_INT, enc->channels,
+      "rate", G_TYPE_INT, enc->samplerate,
+      "width", G_TYPE_INT, enc->depth, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  if (!gst_wavpack_set_channel_layout (caps, enc->channel_mask))
+    GST_WARNING_OBJECT (enc, "setting channel layout failed");
+
+  if (!gst_pad_set_caps (enc->srcpad, caps)) {
+    GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+        ("setting caps failed: %" GST_PTR_FORMAT, caps));
+    gst_caps_unref (caps);
+    gst_object_unref (enc);
+    return FALSE;
+  }
+  gst_pad_use_fixed_caps (enc->srcpad);
+
+  gst_caps_unref (caps);
+  gst_object_unref (enc);
+  return TRUE;
+}
+
+static void
+gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
+{
+  enc->wp_config = g_new0 (WavpackConfig, 1);
+  /* set general stream informations in the WavpackConfig */
+  enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
+  enc->wp_config->bits_per_sample = enc->depth;
+  enc->wp_config->num_channels = enc->channels;
+  enc->wp_config->channel_mask = enc->channel_mask;
+  enc->wp_config->sample_rate = enc->samplerate;
+
+  /*
+   * Set parameters in WavpackConfig
+   */
+
+  /* Encoding mode */
+  switch (enc->mode) {
+#if 0
+    case GST_WAVPACK_ENC_MODE_VERY_FAST:
+      enc->wp_config->flags |= CONFIG_VERY_FAST_FLAG;
+      enc->wp_config->flags |= CONFIG_FAST_FLAG;
+      break;
+#endif
+    case GST_WAVPACK_ENC_MODE_FAST:
+      enc->wp_config->flags |= CONFIG_FAST_FLAG;
+      break;
+    case GST_WAVPACK_ENC_MODE_DEFAULT:
+      break;
+    case GST_WAVPACK_ENC_MODE_HIGH:
+      enc->wp_config->flags |= CONFIG_HIGH_FLAG;
+      break;
+#ifndef WAVPACK_OLD_API
+    case GST_WAVPACK_ENC_MODE_VERY_HIGH:
+      enc->wp_config->flags |= CONFIG_HIGH_FLAG;
+      enc->wp_config->flags |= CONFIG_VERY_HIGH_FLAG;
+      break;
+#endif
+  }
+
+  /* Bitrate, enables lossy mode */
+  if (enc->bitrate) {
+    enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
+    enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
+    enc->wp_config->bitrate = enc->bitrate / 1000.0;
+  } else if (enc->bps) {
+    enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
+    enc->wp_config->bitrate = enc->bps;
+  }
+
+  /* Correction Mode, only in lossy mode */
+  if (enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
+    if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
+      GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
+          "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+      enc->wvcsrcpad =
+          gst_pad_new_from_static_template (&wvcsrc_factory, "wvcsrc");
+
+      /* try to add correction src pad, don't set correction mode on failure */
+      GST_DEBUG_OBJECT (enc, "Adding correction pad with caps %"
+          GST_PTR_FORMAT, caps);
+      if (!gst_pad_set_caps (enc->wvcsrcpad, caps)) {
+        enc->correction_mode = 0;
+        GST_WARNING_OBJECT (enc, "setting correction caps failed");
+      } else {
+        gst_pad_use_fixed_caps (enc->wvcsrcpad);
+        gst_pad_set_active (enc->wvcsrcpad, TRUE);
+        gst_element_add_pad (GST_ELEMENT (enc), enc->wvcsrcpad);
+        enc->wp_config->flags |= CONFIG_CREATE_WVC;
+        if (enc->correction_mode == GST_WAVPACK_CORRECTION_MODE_OPTIMIZED) {
+          enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
+        }
+      }
+      gst_caps_unref (caps);
+    }
+  } else {
+    if (enc->correction_mode > GST_WAVPACK_CORRECTION_MODE_OFF) {
+      enc->correction_mode = 0;
+      GST_WARNING_OBJECT (enc, "setting correction mode only has "
+          "any effect if a bitrate is provided.");
+    }
+  }
+  gst_element_no_more_pads (GST_ELEMENT (enc));
+
+  /* MD5, setup MD5 context */
+  if ((enc->md5) && !(enc->md5_context)) {
+    enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
+    enc->md5_context = g_checksum_new (G_CHECKSUM_MD5);
+  }
+
+  /* Extra encode processing */
+  if (enc->extra_processing) {
+    enc->wp_config->flags |= CONFIG_EXTRA_MODE;
+    enc->wp_config->xmode = enc->extra_processing;
+  }
+
+  /* Joint stereo mode */
+  switch (enc->joint_stereo_mode) {
+    case GST_WAVPACK_JS_MODE_AUTO:
+      break;
+    case GST_WAVPACK_JS_MODE_LEFT_RIGHT:
+      enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
+      enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
+      break;
+    case GST_WAVPACK_JS_MODE_MID_SIDE:
+      enc->wp_config->flags |= (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
+      break;
+  }
+}
+
+static int
+gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
+{
+  GstWavpackEncWriteID *wid = (GstWavpackEncWriteID *) id;
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (wid->wavpack_enc);
+  GstFlowReturn *flow;
+  GstBuffer *buffer;
+  GstPad *pad;
+  guchar *block = (guchar *) data;
+
+  pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
+  flow =
+      (wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
+      srcpad_last_return;
+
+  *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
+      count, GST_PAD_CAPS (pad), &buffer);
+
+  if (*flow != GST_FLOW_OK) {
+    GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
+        GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
+    return FALSE;
+  }
+
+  g_memmove (GST_BUFFER_DATA (buffer), block, count);
+
+  if (count > sizeof (WavpackHeader) && memcmp (block, "wvpk", 4) == 0) {
+    /* if it's a Wavpack block set buffer timestamp and duration, etc */
+    WavpackHeader wph;
+
+    GST_LOG_OBJECT (enc, "got %d bytes of encoded wavpack %sdata",
+        count, (wid->correction) ? "correction " : "");
+
+    gst_wavpack_read_header (&wph, block);
+
+    /* Only set when pushing the first buffer again, in that case
+     * we don't want to delay the buffer or push newsegment events
+     */
+    if (!wid->passthrough) {
+      /* Only push complete blocks */
+      if (enc->pending_buffer == NULL) {
+        enc->pending_buffer = buffer;
+        enc->pending_offset = wph.block_index;
+      } else if (enc->pending_offset == wph.block_index) {
+        enc->pending_buffer = gst_buffer_join (enc->pending_buffer, buffer);
+      } else {
+        GST_ERROR ("Got incomplete block, dropping");
+        gst_buffer_unref (enc->pending_buffer);
+        enc->pending_buffer = buffer;
+        enc->pending_offset = wph.block_index;
+      }
+
+      if (!(wph.flags & FINAL_BLOCK))
+        return TRUE;
+
+      buffer = enc->pending_buffer;
+      enc->pending_buffer = NULL;
+      enc->pending_offset = 0;
+
+      /* if it's the first wavpack block, send a NEW_SEGMENT event */
+      if (wph.block_index == 0) {
+        gst_pad_push_event (pad,
+            gst_event_new_new_segment (FALSE,
+                1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
+
+        /* save header for later reference, so we can re-send it later on
+         * EOS with fixed up values for total sample count etc. */
+        if (enc->first_block == NULL && !wid->correction) {
+          enc->first_block =
+              g_memdup (GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
+          enc->first_block_size = GST_BUFFER_SIZE (buffer);
+        }
+      }
+    }
+
+    /* set buffer timestamp, duration, offset, offset_end from
+     * the wavpack header */
+    GST_BUFFER_TIMESTAMP (buffer) = enc->timestamp_offset +
+        gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
+        enc->samplerate);
+    GST_BUFFER_DURATION (buffer) =
+        gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
+        enc->samplerate);
+    GST_BUFFER_OFFSET (buffer) = wph.block_index;
+    GST_BUFFER_OFFSET_END (buffer) = wph.block_index + wph.block_samples;
+  } else {
+    /* if it's something else set no timestamp and duration on the buffer */
+    GST_DEBUG_OBJECT (enc, "got %d bytes of unknown data", count);
+
+    GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
+    GST_BUFFER_DURATION (buffer) = GST_CLOCK_TIME_NONE;
+  }
+
+  /* push the buffer and forward errors */
+  GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
+      GST_BUFFER_SIZE (buffer));
+  *flow = gst_pad_push (pad, buffer);
+
+  if (*flow != GST_FLOW_OK) {
+    GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
+        GST_DEBUG_PAD_NAME (pad), gst_flow_get_name (*flow));
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+static void
+gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
+    gint nsamples)
+{
+  gint i, j;
+  gint32 tmp[8];
+
+  for (i = 0; i < nsamples / enc->channels; i++) {
+    for (j = 0; j < enc->channels; j++) {
+      tmp[enc->channel_mapping[j]] = data[j];
+    }
+    for (j = 0; j < enc->channels; j++) {
+      data[j] = tmp[j];
+    }
+    data += enc->channels;
+  }
+}
+
+static GstFlowReturn
+gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
+  GstFlowReturn ret;
+
+  /* reset the last returns to GST_FLOW_OK. This is only set to something else
+   * while WavpackPackSamples() or more specific gst_wavpack_enc_push_block()
+   * so not valid anymore */
+  enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
+
+  GST_DEBUG ("got %u raw samples", sample_count);
+
+  /* check if we already have a valid WavpackContext, otherwise make one */
+  if (!enc->wp_context) {
+    /* create raw context */
+    enc->wp_context =
+        WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
+        (enc->correction_mode > 0) ? &enc->wvc_id : NULL);
+    if (!enc->wp_context) {
+      GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+          ("error creating Wavpack context"));
+      gst_object_unref (enc);
+      gst_buffer_unref (buf);
+      return GST_FLOW_ERROR;
+    }
+
+    /* set the WavpackConfig according to our parameters */
+    gst_wavpack_enc_set_wp_config (enc);
+
+    /* set the configuration to the context now that we know everything
+     * and initialize the encoder */
+    if (!WavpackSetConfiguration (enc->wp_context,
+            enc->wp_config, (uint32_t) (-1))
+        || !WavpackPackInit (enc->wp_context)) {
+      GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
+          ("error setting up wavpack encoding context"));
+      WavpackCloseFile (enc->wp_context);
+      gst_object_unref (enc);
+      gst_buffer_unref (buf);
+      return GST_FLOW_ERROR;
+    }
+    GST_DEBUG ("setup of encoding context successfull");
+  }
+
+  /* Save the timestamp of the first buffer. This will be later
+   * used as offset for all following buffers */
+  if (enc->timestamp_offset == GST_CLOCK_TIME_NONE) {
+    if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+      enc->timestamp_offset = GST_BUFFER_TIMESTAMP (buf);
+      enc->next_ts = GST_BUFFER_TIMESTAMP (buf);
+    } else {
+      enc->timestamp_offset = 0;
+      enc->next_ts = 0;
+    }
+  }
+
+  /* Check if we have a continous stream, if not drop some samples or the buffer or
+   * insert some silence samples */
+  if (enc->next_ts != GST_CLOCK_TIME_NONE &&
+      GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
+    guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
+    guint64 diff_bytes;
+
+    GST_WARNING_OBJECT (enc, "Buffer is older than previous "
+        "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
+        "), cannot handle. Clipping buffer.",
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+        GST_TIME_ARGS (enc->next_ts));
+
+    diff_bytes =
+        GST_CLOCK_TIME_TO_FRAMES (diff, enc->samplerate) * enc->channels * 2;
+    if (diff_bytes >= GST_BUFFER_SIZE (buf)) {
+      gst_buffer_unref (buf);
+      return GST_FLOW_OK;
+    }
+    buf = gst_buffer_make_metadata_writable (buf);
+    GST_BUFFER_DATA (buf) += diff_bytes;
+    GST_BUFFER_SIZE (buf) -= diff_bytes;
+
+    GST_BUFFER_TIMESTAMP (buf) += diff;
+    if (GST_BUFFER_DURATION_IS_VALID (buf))
+      GST_BUFFER_DURATION (buf) -= diff;
+  }
+
+  /* Allow a diff of at most 5 ms */
+  if (enc->next_ts != GST_CLOCK_TIME_NONE
+      && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
+    if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
+        GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > 5 * GST_MSECOND) {
+      GST_WARNING_OBJECT (enc,
+          "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
+          GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, 5 * GST_MSECOND);
+
+      WavpackFlushSamples (enc->wp_context);
+      enc->timestamp_offset += (GST_BUFFER_TIMESTAMP (buf) - enc->next_ts);
+    }
+  }
+
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
+      && GST_BUFFER_DURATION_IS_VALID (buf))
+    enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+  else
+    enc->next_ts = GST_CLOCK_TIME_NONE;
+
+  if (enc->need_channel_remap) {
+    buf = gst_buffer_make_writable (buf);
+    gst_wavpack_enc_fix_channel_order (enc, (gint32 *) GST_BUFFER_DATA (buf),
+        sample_count);
+  }
+
+  /* if we want to append the MD5 sum to the stream update it here
+   * with the current raw samples */
+  if (enc->md5) {
+    g_checksum_update (enc->md5_context, GST_BUFFER_DATA (buf),
+        GST_BUFFER_SIZE (buf));
+  }
+
+  /* encode and handle return values from encoding */
+  if (WavpackPackSamples (enc->wp_context, (int32_t *) GST_BUFFER_DATA (buf),
+          sample_count / enc->channels)) {
+    GST_DEBUG ("encoding samples successful");
+    ret = GST_FLOW_OK;
+  } else {
+    if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
+        (enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
+      ret = GST_FLOW_RESEND;
+    } else if ((enc->srcpad_last_return == GST_FLOW_OK) ||
+        (enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
+      ret = GST_FLOW_OK;
+    } else if ((enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
+        (enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
+      ret = GST_FLOW_NOT_LINKED;
+    } else if ((enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
+        (enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
+      ret = GST_FLOW_WRONG_STATE;
+    } else {
+      GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
+          ("encoding samples failed"));
+      ret = GST_FLOW_ERROR;
+    }
+  }
+
+  gst_buffer_unref (buf);
+  gst_object_unref (enc);
+  return ret;
+}
+
+static void
+gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
+{
+  GstEvent *event = gst_event_new_new_segment (TRUE, 1.0, GST_FORMAT_BYTES,
+      0, GST_BUFFER_OFFSET_NONE, 0);
+  gboolean ret;
+
+  g_return_if_fail (enc);
+  g_return_if_fail (enc->first_block);
+
+  /* update the sample count in the first block */
+  WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
+
+  /* try to seek to the beginning of the output */
+  ret = gst_pad_push_event (enc->srcpad, event);
+  if (ret) {
+    /* try to rewrite the first block */
+    GST_DEBUG_OBJECT (enc, "rewriting first block ...");
+    enc->wv_id.passthrough = TRUE;
+    ret = gst_wavpack_enc_push_block (&enc->wv_id,
+        enc->first_block, enc->first_block_size);
+    enc->wv_id.passthrough = FALSE;
+  } else {
+    GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
+        "Seeking to first block failed!");
+  }
+}
+
+static gboolean
+gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  gboolean ret = TRUE;
+
+  GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      /* Encode all remaining samples and flush them to the src pads */
+      WavpackFlushSamples (enc->wp_context);
+
+      /* Drop all remaining data, this is no complete block otherwise
+       * it would've been pushed already */
+      if (enc->pending_buffer) {
+        gst_buffer_unref (enc->pending_buffer);
+        enc->pending_buffer = NULL;
+        enc->pending_offset = 0;
+      }
+
+      /* write the MD5 sum if we have to write one */
+      if ((enc->md5) && (enc->md5_context)) {
+        guint8 md5_digest[16];
+        gsize digest_len = sizeof (md5_digest);
+
+        g_checksum_get_digest (enc->md5_context, md5_digest, &digest_len);
+        if (digest_len == sizeof (md5_digest))
+          WavpackStoreMD5Sum (enc->wp_context, md5_digest);
+        else
+          GST_WARNING_OBJECT (enc, "Calculating MD5 digest failed");
+      }
+
+      /* Try to rewrite the first frame with the correct sample number */
+      if (enc->first_block)
+        gst_wavpack_enc_rewrite_first_block (enc);
+
+      /* close the context if not already happened */
+      if (enc->wp_context) {
+        WavpackCloseFile (enc->wp_context);
+        enc->wp_context = NULL;
+      }
+
+      ret = gst_pad_event_default (pad, event);
+      break;
+    case GST_EVENT_NEWSEGMENT:
+      if (enc->wp_context) {
+        GST_WARNING_OBJECT (enc, "got NEWSEGMENT after encoding "
+            "already started");
+      }
+      /* drop NEWSEGMENT events, we create our own when pushing
+       * the first buffer to the pads */
+      gst_event_unref (event);
+      ret = TRUE;
+      break;
+    default:
+      ret = gst_pad_event_default (pad, event);
+      break;
+  }
+
+  gst_object_unref (enc);
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      /* set the last returned GstFlowReturns of the two pads to GST_FLOW_OK
+       * as they're only set to something else in WavpackPackSamples() or more
+       * specific gst_wavpack_enc_push_block() and nothing happened there yet */
+      enc->srcpad_last_return = enc->wvcsrcpad_last_return = GST_FLOW_OK;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_wavpack_enc_reset (enc);
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+static void
+gst_wavpack_enc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
+
+  switch (prop_id) {
+    case ARG_MODE:
+      enc->mode = g_value_get_enum (value);
+      break;
+    case ARG_BITRATE:{
+      guint val = g_value_get_uint (value);
+
+      if ((val >= 24000) && (val <= 9600000)) {
+        enc->bitrate = val;
+        enc->bps = 0.0;
+      } else {
+        enc->bitrate = 0;
+        enc->bps = 0.0;
+      }
+      break;
+    }
+    case ARG_BITSPERSAMPLE:{
+      gdouble val = g_value_get_double (value);
+
+      if ((val >= 2.0) && (val <= 24.0)) {
+        enc->bps = val;
+        enc->bitrate = 0;
+      } else {
+        enc->bps = 0.0;
+        enc->bitrate = 0;
+      }
+      break;
+    }
+    case ARG_CORRECTION_MODE:
+      enc->correction_mode = g_value_get_enum (value);
+      break;
+    case ARG_MD5:
+      enc->md5 = g_value_get_boolean (value);
+      break;
+    case ARG_EXTRA_PROCESSING:
+      enc->extra_processing = g_value_get_uint (value);
+      break;
+    case ARG_JOINT_STEREO_MODE:
+      enc->joint_stereo_mode = g_value_get_enum (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_wavpack_enc_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (object);
+
+  switch (prop_id) {
+    case ARG_MODE:
+      g_value_set_enum (value, enc->mode);
+      break;
+    case ARG_BITRATE:
+      if (enc->bps == 0.0) {
+        g_value_set_uint (value, enc->bitrate);
+      } else {
+        g_value_set_uint (value, 0);
+      }
+      break;
+    case ARG_BITSPERSAMPLE:
+      if (enc->bitrate == 0) {
+        g_value_set_double (value, enc->bps);
+      } else {
+        g_value_set_double (value, 0.0);
+      }
+      break;
+    case ARG_CORRECTION_MODE:
+      g_value_set_enum (value, enc->correction_mode);
+      break;
+    case ARG_MD5:
+      g_value_set_boolean (value, enc->md5);
+      break;
+    case ARG_EXTRA_PROCESSING:
+      g_value_set_uint (value, enc->extra_processing);
+      break;
+    case ARG_JOINT_STEREO_MODE:
+      g_value_set_enum (value, enc->joint_stereo_mode);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+gboolean
+gst_wavpack_enc_plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "wavpackenc",
+          GST_RANK_NONE, GST_TYPE_WAVPACK_ENC))
+    return FALSE;
+
+  GST_DEBUG_CATEGORY_INIT (gst_wavpack_enc_debug, "wavpack_enc", 0,
+      "Wavpack encoder");
+
+  return TRUE;
+}
diff --git a/ext/wavpack/gstwavpackenc.h b/ext/wavpack/gstwavpackenc.h
new file mode 100644
index 0000000..d2df844
--- /dev/null
+++ b/ext/wavpack/gstwavpackenc.h
@@ -0,0 +1,104 @@
+/* GStreamer Wavpack encoder plugin
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackenc.h: Wavpack audio encoder
+ *
+ * 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.
+ */
+
+#ifndef __GST_WAVPACK_ENC_H__
+#define __GST_WAVPACK_ENC_H__
+
+#include <gst/gst.h>
+
+#include <wavpack/wavpack.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_WAVPACK_ENC \
+  (gst_wavpack_enc_get_type())
+#define GST_WAVPACK_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_ENC,GstWavpackEnc))
+#define GST_WAVPACK_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_ENC,GstWavpackEnc))
+#define GST_IS_WAVPACK_ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_ENC))
+#define GST_IS_WAVPACK_ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_ENC))
+typedef struct _GstWavpackEnc GstWavpackEnc;
+typedef struct _GstWavpackEncClass GstWavpackEncClass;
+
+typedef struct
+{
+  gboolean correction;
+  GstWavpackEnc *wavpack_enc;
+  gboolean passthrough;
+} GstWavpackEncWriteID;
+
+
+struct _GstWavpackEnc
+{
+  GstElement element;
+
+  /*< private > */
+  GstPad *sinkpad, *srcpad;
+  GstPad *wvcsrcpad;
+
+  GstFlowReturn srcpad_last_return;
+  GstFlowReturn wvcsrcpad_last_return;
+
+  WavpackConfig *wp_config;
+  WavpackContext *wp_context;
+
+  gint samplerate;
+  gint channels;
+  gint channel_mask;
+  gint8 channel_mapping[8];
+  gboolean need_channel_remap;
+  gint depth;
+
+  GstWavpackEncWriteID wv_id;
+  GstWavpackEncWriteID wvc_id;
+
+  guint mode;
+  guint bitrate;
+  gdouble bps;
+  guint correction_mode;
+  gboolean md5;
+  GChecksum *md5_context;
+  guint extra_processing;
+  guint joint_stereo_mode;
+
+  void *first_block;
+  int32_t first_block_size;
+
+  GstBuffer *pending_buffer;
+  gint32 pending_offset;
+
+  GstClockTime timestamp_offset;
+  GstClockTime next_ts;
+};
+
+struct _GstWavpackEncClass
+{
+  GstElementClass parent;
+};
+
+GType gst_wavpack_enc_get_type (void);
+
+gboolean gst_wavpack_enc_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+#endif /* __GST_WAVPACK_ENC_H__ */
diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c
new file mode 100644
index 0000000..e1ff785
--- /dev/null
+++ b/ext/wavpack/gstwavpackparse.c
@@ -0,0 +1,1344 @@
+/* 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)
+{
+  int 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;
+}
diff --git a/ext/wavpack/gstwavpackparse.h b/ext/wavpack/gstwavpackparse.h
new file mode 100644
index 0000000..60504a7
--- /dev/null
+++ b/ext/wavpack/gstwavpackparse.h
@@ -0,0 +1,97 @@
+/* GStreamer wavpack plugin
+ * (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
+ *
+ * gstwavpackparse.h: 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.
+ */
+
+#ifndef __GST_WAVPACK_PARSE_H__
+#define __GST_WAVPACK_PARSE_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_WAVPACK_PARSE \
+  (gst_wavpack_parse_get_type())
+#define GST_WAVPACK_PARSE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WAVPACK_PARSE,GstWavpackParse))
+#define GST_WAVPACK_PARSE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_WAVPACK_PARSE,GstWavpackParseClass))
+#define GST_IS_WAVPACK_PARSE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WAVPACK_PARSE))
+#define GST_IS_WAVPACK_PARSE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_WAVPACK_PARSE))
+typedef struct _GstWavpackParse GstWavpackParse;
+typedef struct _GstWavpackParseClass GstWavpackParseClass;
+typedef struct _GstWavpackParseIndexEntry GstWavpackParseIndexEntry;
+
+struct _GstWavpackParseIndexEntry
+{
+  gint64 byte_offset;           /* byte offset of this chunk  */
+  gint64 sample_offset;         /* first sample in this chunk */
+  gint64 sample_offset_end;     /* first sample in next chunk  */
+};
+
+struct _GstWavpackParse
+{
+  GstElement element;
+
+  /*< private > */
+  GstPad *sinkpad;
+  GstPad *srcpad;
+
+  guint samplerate;
+  guint channels;
+  gint64 total_samples;
+
+  gboolean need_newsegment;
+  gboolean discont;
+
+  gint64 current_offset;        /* byte offset on sink pad */
+  gint64 upstream_length;       /* length of file in bytes */
+
+  GstSegment segment;           /* the currently configured segment, in
+                                 * samples/audio frames (DEFAULT format) */
+
+  GstBuffer *pending_buffer;
+  gint32 pending_offset;
+  guint32 next_block_index;
+
+  GstAdapter *adapter;          /* when operating chain-based, otherwise NULL */
+
+  /* List of GstWavpackParseIndexEntry structs, mapping known
+   * sample offsets to byte offsets. Is kept increasing without
+   * gaps (ie. append only and consecutive entries must always
+   * map to consecutive chunks in the file). */
+  GSList *entries;
+
+  /* Queued events (e.g. tag events we receive before we create the src pad) */
+  GList *queued_events;         /* STREAM_LOCK */
+};
+
+struct _GstWavpackParseClass
+{
+  GstElementClass parent;
+};
+
+GType gst_wavpack_parse_get_type (void);
+
+gboolean gst_wavpack_parse_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+#endif /* __GST_WAVPACK_PARSE_H__ */
diff --git a/ext/wavpack/gstwavpackstreamreader.c b/ext/wavpack/gstwavpackstreamreader.c
new file mode 100644
index 0000000..074a2e7
--- /dev/null
+++ b/ext/wavpack/gstwavpackstreamreader.c
@@ -0,0 +1,124 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackstreamreader.c: stream reader used for decoding
+ *
+ * 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.
+ */
+
+#include <string.h>
+#include <math.h>
+#include <gst/gst.h>
+
+#include "gstwavpackstreamreader.h"
+
+GST_DEBUG_CATEGORY_EXTERN (wavpack_debug);
+#define GST_CAT_DEFAULT wavpack_debug
+
+static int32_t
+gst_wavpack_stream_reader_read_bytes (void *id, void *data, int32_t bcount)
+{
+  read_id *rid = (read_id *) id;
+  uint32_t left = rid->length - rid->position;
+  uint32_t to_read = MIN (left, bcount);
+
+  GST_DEBUG ("Trying to read %d of %d bytes from position %d", bcount,
+      rid->length, rid->position);
+
+  if (to_read > 0) {
+    g_memmove (data, rid->buffer + rid->position, to_read);
+    rid->position += to_read;
+    return to_read;
+  } else {
+    GST_WARNING ("Couldn't read %d bytes", bcount);
+    return 0;
+  }
+}
+
+static uint32_t
+gst_wavpack_stream_reader_get_pos (void *id)
+{
+  GST_DEBUG ("Returning position %d", ((read_id *) id)->position);
+  return ((read_id *) id)->position;
+}
+
+static int
+gst_wavpack_stream_reader_set_pos_abs (void *id, uint32_t pos)
+{
+  GST_WARNING ("Should not be called: tried to set absolute position to %d",
+      pos);
+  return -1;
+}
+
+static int
+gst_wavpack_stream_reader_set_pos_rel (void *id, int32_t delta, int mode)
+{
+  GST_WARNING ("Should not be called: tried to set relative position to %d"
+      " with mode %d", delta, mode);
+  return -1;
+}
+
+static int
+gst_wavpack_stream_reader_push_back_byte (void *id, int c)
+{
+  read_id *rid = (read_id *) id;
+
+  GST_DEBUG ("Pushing back one byte: 0x%x", c);
+
+  rid->position -= 1;
+  if (rid->position < 0)
+    rid->position = 0;
+  return rid->position;
+}
+
+static uint32_t
+gst_wavpack_stream_reader_get_length (void *id)
+{
+  GST_DEBUG ("Returning length %d", ((read_id *) id)->length);
+
+  return ((read_id *) id)->length;
+}
+
+static int
+gst_wavpack_stream_reader_can_seek (void *id)
+{
+  GST_DEBUG ("Can't seek");
+  return FALSE;
+}
+
+static int32_t
+gst_wavpack_stream_reader_write_bytes (void *id, void *data, int32_t bcount)
+{
+  GST_WARNING ("Should not be called, tried to write %d bytes", bcount);
+  return 0;
+}
+
+WavpackStreamReader *
+gst_wavpack_stream_reader_new (void)
+{
+  WavpackStreamReader *stream_reader =
+      (WavpackStreamReader *) g_malloc0 (sizeof (WavpackStreamReader));
+  stream_reader->read_bytes = gst_wavpack_stream_reader_read_bytes;
+  stream_reader->get_pos = gst_wavpack_stream_reader_get_pos;
+  stream_reader->set_pos_abs = gst_wavpack_stream_reader_set_pos_abs;
+  stream_reader->set_pos_rel = gst_wavpack_stream_reader_set_pos_rel;
+  stream_reader->push_back_byte = gst_wavpack_stream_reader_push_back_byte;
+  stream_reader->get_length = gst_wavpack_stream_reader_get_length;
+  stream_reader->can_seek = gst_wavpack_stream_reader_can_seek;
+  stream_reader->write_bytes = gst_wavpack_stream_reader_write_bytes;
+
+  return stream_reader;
+}
diff --git a/ext/wavpack/gstwavpackstreamreader.h b/ext/wavpack/gstwavpackstreamreader.h
new file mode 100644
index 0000000..713b4c1
--- /dev/null
+++ b/ext/wavpack/gstwavpackstreamreader.h
@@ -0,0 +1,36 @@
+/* GStreamer Wavpack plugin
+ * Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * gstwavpackstreamreader.h: stream reader used for decoding
+ *
+ * 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.
+ */
+
+#ifndef __GST_WAVPACK_STREAM_READER_H__
+#define __GST_WAVPACK_STREAM_READER_H__
+
+#include <wavpack/wavpack.h>
+
+typedef struct
+{
+  guint8 *buffer;
+  uint32_t length;
+  uint32_t position;
+} read_id;
+
+WavpackStreamReader *gst_wavpack_stream_reader_new (void);
+
+#endif