ext/wavpack/: Add wavpack encoder element (#343131).

Original commit message from CVS:
Patch by: Sebastian Dröge  <mail at slomosnail de>
* ext/wavpack/Makefile.am:
* ext/wavpack/gstwavpack.c: (plugin_init):
* ext/wavpack/gstwavpackcommon.h:
* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_mode_get_type),
(gst_wavpack_enc_correction_mode_get_type),
(gst_wavpack_enc_joint_stereo_mode_get_type),
(gst_wavpack_enc_base_init), (gst_wavpack_enc_class_init),
(gst_wavpack_enc_init), (gst_wavpack_enc_dispose),
(gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config),
(gst_wavpack_enc_format_samples), (gst_wavpack_enc_push_block),
(gst_wavpack_enc_chain), (gst_wavpack_enc_rewrite_first_block),
(gst_wavpack_enc_sink_event), (gst_wavpack_enc_change_state),
(gst_wavpack_enc_set_property), (gst_wavpack_enc_get_property),
(gst_wavpack_enc_plugin_init):
* ext/wavpack/gstwavpackenc.h:
* ext/wavpack/md5.c:
* ext/wavpack/md5.h:
Add wavpack encoder element (#343131).

1 files changed, 1002 insertions, 0 deletions

diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c
new file mode 100644
index 00000000..a7d6f748
--- /dev/null
+++ b/ext/wavpack/gstwavpackenc.c
@@ -0,0 +1,1002 @@
+/* GStreamer Wavpack encoder plugin
+ * Copyright (c) 2006 Sebastian Dröge <mail@slomosnail.de>
+ *
+ * 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.
+ */
+
+/*
+ * TODO: - add multichannel handling. channel_mask is:
+ *                  front left
+ *                  front right
+ *                  center
+ *                  LFE
+ *                  back left
+ *                  back right
+ *                  front left center
+ *                  front right center
+ *                  back left
+ *                  back center
+ *                  side left
+ *                  side right
+ *                  ...
+ *        - 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"
+#include "md5.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_dispose (GObject * object);
+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_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) 32, "
+        "endianness = (int) LITTLE_ENDIAN, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
+        "audio/x-raw-int, "
+        "width = (int) 24, "
+        "depth = (int) 24, "
+        "endianness = (int) LITTLE_ENDIAN, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
+        "audio/x-raw-int, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "endianness = (int) LITTLE_ENDIAN, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE;"
+        "audio/x-raw-int, "
+        "width = (int) 8, "
+        "depth = (int) 8, "
+        "endianness = (int) LITTLE_ENDIAN, "
+        "channels = (int) [ 1, 2 ], "
+        "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) { 8, 16, 24, 32 }, "
+        "channels = (int) [ 1, 2 ], "
+        "rate = (int) [ 6000, 192000 ], " "framed = (boolean) FALSE")
+    );
+
+static GstStaticPadTemplate wvcsrc_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) FALSE")
+    );
+
+#define DEFAULT_MODE 1
+#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[] = {
+      {0, "Fast Compression", "0"},
+      {1, "Default", "1"},
+      {2, "High Compression", "2"},
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncMode", values);
+  }
+  return qtype;
+}
+
+#define DEFAULT_CORRECTION_MODE 0
+#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[] = {
+      {0, "Create no correction file (default)", "0"},
+      {1, "Create correction file", "1"},
+      {2, "Create optimized correction file", "2"},
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncCorrectionMode", values);
+  }
+  return qtype;
+}
+
+#define DEFAULT_JS_MODE 0
+#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[] = {
+      {0, "auto (default)", "0"},
+      {1, "left/right", "1"},
+      {2, "mid/side", "2"},
+      {0, NULL, NULL}
+    };
+
+    qtype = g_enum_register_static ("GstWavpackEncJSMode", values);
+  }
+  return qtype;
+}
+
+GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstElement, GST_TYPE_ELEMENT);
+
+static void
+gst_wavpack_enc_base_init (gpointer klass)
+{
+  static GstElementDetails element_details = {
+    "Wavpack audio encoder",
+    "Codec/Encoder/Audio",
+    "Encodes audio with the Wavpack lossless/lossy audio codec",
+    "Sebastian Dröge <mail@slomosnail.de>"
+  };
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  /* add pad templates */
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&wvcsrc_factory));
+
+  /* set element details */
+  gst_element_class_set_details (element_class, &element_details);
+}
+
+
+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);
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_wavpack_enc_dispose);
+
+  /* set property handlers */
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_property);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (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, DEFAULT_MODE, G_PARAM_READWRITE));
+  g_object_class_install_property (gobject_class, ARG_BITRATE,
+      g_param_spec_double ("bitrate", "Bitrate",
+          "Try to encode with this average bitrate. "
+          "This enables lossy encoding! (0 .. 2.0 == disabled, 2.0 .. 23.9 == bits/sample, 24.0 .. 9600 == kbit/second)",
+          0.0, 9600.0, 0.0, G_PARAM_READWRITE));
+  g_object_class_install_property (gobject_class, ARG_CORRECTION_MODE,
+      g_param_spec_enum ("correction_mode", "Correction file mode",
+          "Use this mode for correction file creation. Only works in lossy mode!",
+          GST_TYPE_WAVPACK_ENC_CORRECTION_MODE, DEFAULT_CORRECTION_MODE,
+          G_PARAM_READWRITE));
+  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_object_class_install_property (gobject_class, ARG_EXTRA_PROCESSING,
+      g_param_spec_boolean ("extra_processing", "Extra processing",
+          "Extra encode processing.", FALSE, G_PARAM_READWRITE));
+  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,
+          DEFAULT_JS_MODE, G_PARAM_READWRITE));
+}
+
+static void
+gst_wavpack_enc_init (GstWavpackEnc * wavpack_enc, GstWavpackEncClass * gclass)
+{
+  GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpack_enc);
+
+  /* setup sink pad, add handlers */
+  wavpack_enc->sinkpad =
+      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
+          "sink"), "sink");
+  gst_pad_set_setcaps_function (wavpack_enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_set_caps));
+  gst_pad_set_chain_function (wavpack_enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_chain));
+  gst_pad_set_event_function (wavpack_enc->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event));
+  gst_element_add_pad (GST_ELEMENT (wavpack_enc),
+      GST_DEBUG_FUNCPTR (wavpack_enc->sinkpad));
+
+  /* setup src pad */
+  wavpack_enc->srcpad =
+      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
+          "src"), "src");
+  gst_element_add_pad (GST_ELEMENT (wavpack_enc),
+      GST_DEBUG_FUNCPTR (wavpack_enc->srcpad));
+
+  /* initialize object attributes */
+  wavpack_enc->wp_config = NULL;
+  wavpack_enc->wp_context = NULL;
+  wavpack_enc->first_block = NULL;
+  wavpack_enc->first_block_size = 0;
+  wavpack_enc->md5_context = NULL;
+  wavpack_enc->samplerate = 0;
+  wavpack_enc->width = 0;
+  wavpack_enc->channels = 0;
+
+  wavpack_enc->wv_id = (write_id *) g_malloc0 (sizeof (write_id));
+  wavpack_enc->wv_id->correction = FALSE;
+  wavpack_enc->wv_id->wavpack_enc = wavpack_enc;
+  wavpack_enc->wvc_id = (write_id *) g_malloc0 (sizeof (write_id));
+  wavpack_enc->wvc_id->correction = TRUE;
+  wavpack_enc->wvc_id->wavpack_enc = wavpack_enc;
+
+  /* set default values of params */
+  wavpack_enc->mode = 1;
+  wavpack_enc->bitrate = 0.0;
+  wavpack_enc->correction_mode = 0;
+  wavpack_enc->md5 = FALSE;
+  wavpack_enc->extra_processing = FALSE;
+  wavpack_enc->joint_stereo_mode = 0;
+}
+
+static void
+gst_wavpack_enc_dispose (GObject * object)
+{
+  GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (object);
+
+  /* free the blockout helpers */
+  g_free (wavpack_enc->wv_id);
+  g_free (wavpack_enc->wvc_id);
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static gboolean
+gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
+{
+  GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  GstStructure *structure = gst_caps_get_structure (caps, 0);
+  int depth = 0;
+
+  /* check caps and put relevant parts into our object attributes */
+  if ((!gst_structure_get_int (structure, "channels", &wavpack_enc->channels))
+      || (!gst_structure_get_int (structure, "rate", &wavpack_enc->samplerate))
+      || (!gst_structure_get_int (structure, "width", &wavpack_enc->width))
+      || (!(gst_structure_get_int (structure, "depth", &depth))
+          || depth != wavpack_enc->width)) {
+    GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
+        ("got invalid caps: %", GST_PTR_FORMAT, caps));
+    gst_object_unref (wavpack_enc);
+    return FALSE;
+  }
+
+  /* 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, wavpack_enc->channels,
+      "rate", G_TYPE_INT, wavpack_enc->samplerate,
+      "width", G_TYPE_INT, wavpack_enc->width,
+      "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  if (!gst_pad_set_caps (wavpack_enc->srcpad, caps)) {
+    GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
+        ("setting caps failed: %", GST_PTR_FORMAT, caps));
+    gst_caps_unref (caps);
+    gst_object_unref (wavpack_enc);
+    return FALSE;
+  }
+  gst_pad_use_fixed_caps (wavpack_enc->srcpad);
+
+  gst_caps_unref (caps);
+  gst_object_unref (wavpack_enc);
+  return TRUE;
+}
+
+static void
+gst_wavpack_enc_set_wp_config (GstWavpackEnc * wavpack_enc)
+{
+  wavpack_enc->wp_config = (WavpackConfig *) g_malloc0 (sizeof (WavpackConfig));
+  /* set general stream informations in the WavpackConfig */
+  wavpack_enc->wp_config->bytes_per_sample = (wavpack_enc->width + 7) >> 3;
+  wavpack_enc->wp_config->bits_per_sample = wavpack_enc->width;
+  wavpack_enc->wp_config->num_channels = wavpack_enc->channels;
+
+  /* TODO: handle more than 2 channels correctly! */
+  if (wavpack_enc->channels == 1) {
+    wavpack_enc->wp_config->channel_mask = 0x4;
+  } else if (wavpack_enc->channels == 2) {
+    wavpack_enc->wp_config->channel_mask = 0x2 | 0x1;
+  }
+  wavpack_enc->wp_config->sample_rate = wavpack_enc->samplerate;
+
+  /*
+   * Set parameters in WavpackConfig
+   */
+
+  /* Encoding mode */
+  switch (wavpack_enc->mode) {
+    case 0:
+      wavpack_enc->wp_config->flags |= CONFIG_FAST_FLAG;
+      break;
+    case 1:                    /* default */
+      break;
+    case 2:
+      wavpack_enc->wp_config->flags |= CONFIG_HIGH_FLAG;
+      break;
+  }
+
+  /* Bitrate, enables lossy mode */
+  if (wavpack_enc->bitrate > 2.0) {
+    wavpack_enc->wp_config->flags |= CONFIG_HYBRID_FLAG;
+    wavpack_enc->wp_config->bitrate = wavpack_enc->bitrate;
+    if (wavpack_enc->bitrate >= 24.0)
+      wavpack_enc->wp_config->flags |= CONFIG_BITRATE_KBPS;
+  }
+
+  /* Correction Mode, only in lossy mode */
+  if (wavpack_enc->wp_config->flags & CONFIG_HYBRID_FLAG) {
+    if (wavpack_enc->correction_mode > 0) {
+      wavpack_enc->wvcsrcpad =
+          gst_pad_new_from_template (gst_element_class_get_pad_template
+          (GST_ELEMENT_GET_CLASS (wavpack_enc), "wvcsrc"), "wvcsrc");
+
+      /* try to add correction src pad, don't set correction mode on failure */
+      if (gst_element_add_pad (GST_ELEMENT (wavpack_enc),
+              GST_DEBUG_FUNCPTR (wavpack_enc->wvcsrcpad))) {
+        GstCaps *caps = gst_caps_new_simple ("audio/x-wavpack-correction",
+            "framed", G_TYPE_BOOLEAN, FALSE, NULL);
+
+        gst_element_no_more_pads (GST_ELEMENT (wavpack_enc));
+
+        if (!gst_pad_set_caps (wavpack_enc->wvcsrcpad, caps)) {
+          wavpack_enc->correction_mode = 0;
+          GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL),
+              ("setting correction caps failed: %", GST_PTR_FORMAT, caps));
+        } else {
+          gst_pad_use_fixed_caps (wavpack_enc->wvcsrcpad);
+          wavpack_enc->wp_config->flags |= CONFIG_CREATE_WVC;
+          if (wavpack_enc->correction_mode == 2) {
+            wavpack_enc->wp_config->flags |= CONFIG_OPTIMIZE_WVC;
+          }
+        }
+        gst_caps_unref (caps);
+      } else {
+        wavpack_enc->correction_mode = 0;
+        GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, INIT, (NULL),
+            ("add correction pad failed. no correction file will be created."));
+      }
+    }
+  } else {
+    if (wavpack_enc->correction_mode > 0) {
+      wavpack_enc->correction_mode = 0;
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, SETTINGS, (NULL),
+          ("settings correction mode only has effect if a bitrate is provided."));
+    }
+  }
+
+  /* MD5, setup MD5 context */
+  if ((wavpack_enc->md5) && !(wavpack_enc->md5_context)) {
+    wavpack_enc->wp_config->flags |= CONFIG_MD5_CHECKSUM;
+    wavpack_enc->md5_context = (MD5_CTX *) g_malloc0 (sizeof (MD5_CTX));
+    MD5Init (wavpack_enc->md5_context);
+  }
+
+  /* Extra encode processing */
+  if (wavpack_enc->extra_processing) {
+    wavpack_enc->wp_config->flags |= CONFIG_EXTRA_MODE;
+  }
+
+  /* Joint stereo mode */
+  switch (wavpack_enc->joint_stereo_mode) {
+    case 0:                    /* default */
+      break;
+    case 1:
+      wavpack_enc->wp_config->flags |= CONFIG_JOINT_OVERRIDE;
+      wavpack_enc->wp_config->flags &= ~CONFIG_JOINT_STEREO;
+      break;
+    case 2:
+      wavpack_enc->wp_config->flags |=
+          (CONFIG_JOINT_OVERRIDE | CONFIG_JOINT_STEREO);
+      break;
+  }
+}
+
+static int32_t *
+gst_wavpack_enc_format_samples (const uchar * src_data, uint32_t sample_count,
+    guint width)
+{
+  int32_t *data = (int32_t *) g_malloc0 (sizeof (int32_t) * sample_count);
+
+  /* put all samples into an int32_t*, no matter what
+   * width we have and convert them from little endian
+   * to host byte order */
+
+  switch (width) {
+      int i;
+
+    case 8:
+      for (i = 0; i < sample_count; i++)
+        data[i] = (int32_t) (int8_t) src_data[i];
+      break;
+    case 16:
+      for (i = 0; i < sample_count; i++)
+        data[i] = (int32_t) src_data[2 * i]
+            | ((int32_t) (int8_t) src_data[2 * i + 1] << 8);
+      break;
+    case 24:
+      for (i = 0; i < sample_count; i++)
+        data[i] = (int32_t) src_data[3 * i]
+            | ((int32_t) src_data[3 * i + 1] << 8)
+            | ((int32_t) (int8_t) src_data[3 * i + 2] << 16);
+      break;
+    case 32:
+      for (i = 0; i < sample_count; i++)
+        data[i] = (int32_t) src_data[4 * i]
+            | ((int32_t) src_data[4 * i + 1] << 8)
+            | ((int32_t) src_data[4 * i + 2] << 16)
+            | ((int32_t) (int8_t) src_data[4 * i + 3] << 24);
+      break;
+  }
+
+  return data;
+}
+
+static int
+gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
+{
+  write_id *wid = (write_id *) id;
+  GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (wid->wavpack_enc);
+  GstFlowReturn ret;
+  GstBuffer *buffer;
+  guchar *block = (guchar *) data;
+
+  if (wid->correction == FALSE) {
+    /* we got something that should be pushed to the (non-correction) src pad */
+
+    /* try to allocate a buffer, compatible with the pad, fail otherwise */
+    ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->srcpad,
+        GST_BUFFER_OFFSET_NONE, count, GST_PAD_CAPS (wavpack_enc->srcpad),
+        &buffer);
+    if (ret != GST_FLOW_OK) {
+      wavpack_enc->srcpad_last_return = ret;
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("Dropped one block (%d bytes) of encoded data while allocating buffer! Reason: '%s'\n",
+              count, gst_flow_get_name (ret)));
+      return FALSE;
+    }
+
+    g_memmove (GST_BUFFER_DATA (buffer), block, count);
+
+    if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p')
+        && (block[3] == 'k')) {
+      /* if it's a Wavpack block set buffer timestamp and duration, etc */
+      WavpackHeader wph;
+
+      GST_DEBUG ("got %d bytes of encoded wavpack data", count);
+      gst_wavpack_read_header (&wph, block);
+
+      /* if it's the first wavpack block save it for later reference
+       * i.e. sample count correction and send a NEW_SEGMENT event */
+      if (wph.block_index == 0) {
+        GstEvent *event = gst_event_new_new_segment (FALSE,
+            1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0);
+
+        gst_pad_push_event (wavpack_enc->srcpad, event);
+        wavpack_enc->first_block = g_malloc0 (count);
+        g_memmove (wavpack_enc->first_block, block, count);
+        wavpack_enc->first_block_size = count;
+      }
+
+      /* set buffer timestamp, duration, offset, offset_end from
+       * the wavpack header */
+      GST_BUFFER_TIMESTAMP (buffer) =
+          gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
+          wavpack_enc->samplerate);
+      GST_BUFFER_DURATION (buffer) =
+          gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
+          wavpack_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 ("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 */
+    ret = gst_pad_push (wavpack_enc->srcpad, buffer);
+    wavpack_enc->srcpad_last_return = ret;
+    if (ret == GST_FLOW_OK) {
+      return TRUE;
+    } else {
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("Dropped one block (%d bytes) of encoded data while pushing! Reason: '%s'\n",
+              count, gst_flow_get_name (ret)));
+      return FALSE;
+    }
+  } else if (wid->correction == TRUE) {
+    /* we got something that should be pushed to the correction src pad */
+
+    /* is the correction pad linked? */
+    if (!gst_pad_is_linked (wavpack_enc->wvcsrcpad)) {
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("Dropped one block (%d bytes) of encoded correction data because of unlinked pad",
+              count));
+      wavpack_enc->wvcsrcpad_last_return = GST_FLOW_NOT_LINKED;
+      return FALSE;
+    }
+
+    /* try to allocate a buffer, compatible with the pad, fail otherwise */
+    ret = gst_pad_alloc_buffer_and_set_caps (wavpack_enc->wvcsrcpad,
+        GST_BUFFER_OFFSET_NONE, count,
+        GST_PAD_CAPS (wavpack_enc->wvcsrcpad), &buffer);
+    if (ret != GST_FLOW_OK) {
+      wavpack_enc->wvcsrcpad_last_return = ret;
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("Dropped one block (%d bytes) of encoded correction data while allocating buffer! Reason: '%s'\n",
+              count, gst_flow_get_name (ret)));
+      return FALSE;
+    }
+
+    g_memmove (GST_BUFFER_DATA (buffer), block, count);
+
+    if ((block[0] == 'w') && (block[1] == 'v') && (block[2] == 'p')
+        && (block[3] == 'k')) {
+      /* if it's a Wavpack block set buffer timestamp and duration, etc */
+      WavpackHeader wph;
+
+      GST_DEBUG ("got %d bytes of encoded wavpack correction data", count);
+      gst_wavpack_read_header (&wph, block);
+
+      /* if it's the first wavpack block send a NEW_SEGMENT
+       * event */
+      if (wph.block_index == 0) {
+        GstEvent *event = gst_event_new_new_segment (FALSE,
+            1.0, GST_FORMAT_BYTES, 0, GST_BUFFER_OFFSET_NONE, 0);
+
+        gst_pad_push_event (wavpack_enc->wvcsrcpad, event);
+      }
+
+      /* set buffer timestamp, duration, offset, offset_end from
+       * the wavpack header */
+      GST_BUFFER_TIMESTAMP (buffer) =
+          gst_util_uint64_scale_int (GST_SECOND, wph.block_index,
+          wavpack_enc->samplerate);
+      GST_BUFFER_DURATION (buffer) =
+          gst_util_uint64_scale_int (GST_SECOND, wph.block_samples,
+          wavpack_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 ("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 */
+    ret = gst_pad_push (wavpack_enc->wvcsrcpad, buffer);
+    wavpack_enc->wvcsrcpad_last_return = ret;
+    if (ret == GST_FLOW_OK)
+      return TRUE;
+    else {
+      GST_ELEMENT_WARNING (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("Dropped one block (%d bytes) of encoded correction data while pushing! Reason: '%s'\n",
+              count, gst_flow_get_name (ret)));
+      return FALSE;
+    }
+  } else {
+    /* (correction != TRUE) && (correction != FALSE), wtf? ignore this */
+    g_assert_not_reached ();
+    return TRUE;
+  }
+}
+
+static GstFlowReturn
+gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstWavpackEnc *wavpack_enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  uint32_t sample_count =
+      GST_BUFFER_SIZE (buf) / ((wavpack_enc->width + 7) >> 3);
+  int32_t *data;
+  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 */
+  wavpack_enc->srcpad_last_return = wavpack_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 (!wavpack_enc->wp_context) {
+    gint64 duration;
+    GstFormat fmt = GST_FORMAT_DEFAULT;
+
+    /* create raw context */
+    wavpack_enc->wp_context =
+        WavpackOpenFileOutput (gst_wavpack_enc_push_block, wavpack_enc->wv_id,
+        (wavpack_enc->correction_mode > 0) ? wavpack_enc->wvc_id : NULL);
+    if (!wavpack_enc->wp_context) {
+      GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, INIT, (NULL),
+          ("error creating Wavpack context"));
+      gst_object_unref (wavpack_enc);
+      gst_buffer_unref (buf);
+      return GST_FLOW_ERROR;
+    }
+
+    /* set the WavpackConfig according to our parameters */
+    gst_wavpack_enc_set_wp_config (wavpack_enc);
+
+    /* try to get the duration (or an estimate) in samples from upstream */
+    if (gst_pad_query_peer_duration (pad, &fmt, &duration)) {
+      switch (fmt) {
+        case GST_FORMAT_DEFAULT:
+          break;
+        case GST_FORMAT_TIME:
+          duration =
+              gst_util_uint64_scale (wavpack_enc->samplerate,
+              duration, GST_SECOND);
+          break;
+        default:
+          duration = 0;
+          break;
+      }
+    } else {
+      duration = 0;
+    }
+
+    /* Wavpack doesn't support more than 2^32 samples unfortunately */
+    if (duration > G_GINT64_CONSTANT (1) << 32) {
+      GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, SETTINGS, (NULL),
+          ("more than 2^32 samples are not supported"));
+      WavpackCloseFile (wavpack_enc->wp_context);
+      gst_object_unref (wavpack_enc);
+      gst_buffer_unref (buf);
+      return GST_FLOW_ERROR;
+    }
+
+    /* set the configuration to the context now that we know everything
+     * and initialize the encoder */
+    if (!WavpackSetConfiguration (wavpack_enc->wp_context,
+            wavpack_enc->wp_config, (uint32_t) duration)
+        || !WavpackPackInit (wavpack_enc->wp_context)) {
+      GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, SETTINGS, (NULL),
+          ("error setting up wavpack encoding context"));
+      WavpackCloseFile (wavpack_enc->wp_context);
+      gst_object_unref (wavpack_enc);
+      gst_buffer_unref (buf);
+      return GST_FLOW_ERROR;
+    }
+    GST_DEBUG ("setup of encoding context successfull");
+  }
+
+  /* if we want to append the MD5 sum to the stream update it here
+   * with the current raw samples */
+  if (wavpack_enc->md5) {
+    MD5Update (wavpack_enc->md5_context, GST_BUFFER_DATA (buf),
+        GST_BUFFER_SIZE (buf));
+  }
+
+  /* put all samples into an int32_t*, no matter what
+   * width we have and convert them from little endian
+   * to host byte order */
+  data =
+      gst_wavpack_enc_format_samples (GST_BUFFER_DATA (buf), sample_count,
+      wavpack_enc->width);
+
+  gst_buffer_unref (buf);
+
+  /* encode and handle return values from encoding */
+  if (WavpackPackSamples (wavpack_enc->wp_context, data,
+          sample_count / wavpack_enc->channels)) {
+    GST_DEBUG ("encoding samples successfull");
+    ret = GST_FLOW_OK;
+  } else {
+    if ((wavpack_enc->srcpad_last_return == GST_FLOW_RESEND) ||
+        (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_RESEND)) {
+      ret = GST_FLOW_RESEND;
+    } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_OK) ||
+        (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_OK)) {
+      ret = GST_FLOW_OK;
+    } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_NOT_LINKED) &&
+        (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_NOT_LINKED)) {
+      ret = GST_FLOW_NOT_LINKED;
+    } else if ((wavpack_enc->srcpad_last_return == GST_FLOW_WRONG_STATE) &&
+        (wavpack_enc->wvcsrcpad_last_return == GST_FLOW_WRONG_STATE)) {
+      ret = GST_FLOW_WRONG_STATE;
+    } else {
+      GST_ELEMENT_ERROR (wavpack_enc, LIBRARY, ENCODE, (NULL),
+          ("encoding samples failed"));
+      ret = GST_FLOW_ERROR;
+    }
+  }
+
+  g_free (data);
+  gst_object_unref (wavpack_enc);
+  return ret;
+}
+
+static void
+gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * wavpack_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 (wavpack_enc);
+  g_return_if_fail (wavpack_enc->first_block);
+
+  /* update the sample count in the first block */
+  WavpackUpdateNumSamples (wavpack_enc->wp_context, wavpack_enc->first_block);
+
+  /* try to seek to the beginning of the output */
+  ret = gst_pad_push_event (wavpack_enc->srcpad, event);
+  if (ret) {
+    /* try to rewrite the first block */
+    ret = gst_wavpack_enc_push_block (wavpack_enc->wv_id,
+        wavpack_enc->first_block, wavpack_enc->first_block_size);
+    if (ret) {
+      GST_DEBUG ("rewriting of first block succeeded!");
+    } else {
+      GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, WRITE, (NULL),
+          ("rewriting of first block failed while pushing!"));
+    }
+  } else {
+    GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL),
+        ("rewriting of first block failed. Seeking to first block failed!"));
+  }
+}
+
+static gboolean
+gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstWavpackEnc *wavpack_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 (wavpack_enc->wp_context);
+
+      /* write the MD5 sum if we have to write one */
+      if ((wavpack_enc->md5) && (wavpack_enc->md5_context)) {
+        guchar md5_digest[16];
+
+        MD5Final (md5_digest, wavpack_enc->md5_context);
+        WavpackStoreMD5Sum (wavpack_enc->wp_context, md5_digest);
+      }
+
+      /* Try to rewrite the first frame with the correct sample number if we
+       * had a wrong one at the start of encoding */
+      if ((wavpack_enc->first_block)
+          && (WavpackGetNumSamples (wavpack_enc->wp_context) !=
+              WavpackGetSampleIndex (wavpack_enc->wp_context)))
+        gst_wavpack_enc_rewrite_first_block (wavpack_enc);
+
+      /* close the context if not already happened */
+      if (wavpack_enc->wp_context) {
+        WavpackCloseFile (wavpack_enc->wp_context);
+        wavpack_enc->wp_context = NULL;
+      }
+
+      ret = gst_pad_event_default (pad, event);
+      break;
+    case GST_EVENT_NEWSEGMENT:
+      if (wavpack_enc->wp_context) {
+        GST_ELEMENT_WARNING (wavpack_enc, RESOURCE, SEEK, (NULL),
+            ("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 (wavpack_enc);
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_wavpack_enc_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstWavpackEnc *wavpack_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 */
+      wavpack_enc->srcpad_last_return = wavpack_enc->wvcsrcpad_last_return =
+          GST_FLOW_OK;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    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:
+      /* close and free everything stream related */
+      if (wavpack_enc->wp_context) {
+        WavpackCloseFile (wavpack_enc->wp_context);
+        wavpack_enc->wp_context = NULL;
+      }
+      if (wavpack_enc->wp_config) {
+        g_free (wavpack_enc->wp_config);
+        wavpack_enc->wp_config = NULL;
+      }
+      if (wavpack_enc->first_block) {
+        g_free (wavpack_enc->first_block);
+        wavpack_enc->first_block = NULL;
+        wavpack_enc->first_block_size = 0;
+      }
+      if (wavpack_enc->md5_context) {
+        g_free (wavpack_enc->md5_context);
+        wavpack_enc->md5_context = NULL;
+      }
+
+      /* 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 */
+      wavpack_enc->srcpad_last_return = wavpack_enc->wvcsrcpad_last_return =
+          GST_FLOW_OK;
+      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 *wavpack_enc = GST_WAVPACK_ENC (object);
+
+  switch (prop_id) {
+    case ARG_MODE:
+      wavpack_enc->mode = g_value_get_enum (value);
+      break;
+    case ARG_BITRATE:
+      wavpack_enc->bitrate = g_value_get_double (value);
+      break;
+    case ARG_CORRECTION_MODE:
+      wavpack_enc->correction_mode = g_value_get_enum (value);
+      break;
+    case ARG_MD5:
+      wavpack_enc->md5 = g_value_get_boolean (value);
+      break;
+    case ARG_EXTRA_PROCESSING:
+      wavpack_enc->extra_processing = g_value_get_boolean (value);
+      break;
+    case ARG_JOINT_STEREO_MODE:
+      wavpack_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 *wavpack_enc = GST_WAVPACK_ENC (object);
+
+  switch (prop_id) {
+    case ARG_MODE:
+      g_value_set_enum (value, wavpack_enc->mode);
+      break;
+    case ARG_BITRATE:
+      g_value_set_double (value, wavpack_enc->bitrate);
+      break;
+    case ARG_CORRECTION_MODE:
+      g_value_set_enum (value, wavpack_enc->correction_mode);
+      break;
+    case ARG_MD5:
+      g_value_set_boolean (value, wavpack_enc->md5);
+      break;
+    case ARG_EXTRA_PROCESSING:
+      g_value_set_boolean (value, wavpack_enc->extra_processing);
+      break;
+    case ARG_JOINT_STEREO_MODE:
+      g_value_set_enum (value, wavpack_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, "wavpackenc", 0,
+      "wavpack encoder");
+
+  return TRUE;
+}