Merge branch 'master' into 0.11

Conflicts:
	ext/wavpack/gstwavpackenc.c
	tests/check/elements/audioiirfilter.c
	tests/examples/v4l2/probe.c

ext/annodex/gstannodex.c

@@ -21,6 +21,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

ext/annodex/gstcmmlparser.c

@@ -21,6 +21,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <stdarg.h>
 #include <gst/gst.h>

ext/annodex/gstcmmltag.c

@@ -21,6 +21,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

ext/pulse/pulseprobe.c

@@ -21,6 +21,10 @@
  *  USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

ext/wavpack/gstwavpackdec.c

@@ -55,8 +55,6 @@
 #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
 
@@ -73,22 +71,34 @@ 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, "
+        "width = (int) 8, depth = (int) 8, "
-        "depth = (int) [ 1, 32 ], "
         "channels = (int) [ 1, 8 ], "
         "rate = (int) [ 6000, 192000 ], "
-        "endianness = (int) BYTE_ORDER, " "signed = (boolean) true")
+        "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
+        "audio/x-raw-int, "
+        "width = (int) 16, depth = (int) 16, "
+        "channels = (int) [ 1, 8 ], "
+        "rate = (int) [ 6000, 192000 ], "
+        "endianness = (int) BYTE_ORDER, " "signed = (boolean) true; "
+        "audio/x-raw-int, "
+        "width = (int) 32, depth = (int) 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_start (GstAudioDecoder * dec);
-static gboolean gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps);
+static gboolean gst_wavpack_dec_stop (GstAudioDecoder * dec);
-static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
+static gboolean gst_wavpack_dec_set_format (GstAudioDecoder * dec,
+    GstCaps * caps);
+static GstFlowReturn gst_wavpack_dec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * buffer);
+
 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);
+GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstAudioDecoder,
+    GST_TYPE_AUDIO_DECODER);
 
 static void
 gst_wavpack_dec_base_init (gpointer klass)
@@ -110,11 +120,14 @@ static void
 gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
-  GstElementClass *gstelement_class = (GstElementClass *) klass;
+  GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) (klass);
 
-  gstelement_class->change_state =
-      GST_DEBUG_FUNCPTR (gst_wavpack_dec_change_state);
   gobject_class->finalize = gst_wavpack_dec_finalize;
+
+  base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_dec_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_dec_stop);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_dec_set_format);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_dec_handle_frame);
 }
 
 static void
@@ -123,33 +136,15 @@ 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 ();
 
@@ -168,26 +163,79 @@ gst_wavpack_dec_finalize (GObject * object)
 }
 
 static gboolean
-gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
+gst_wavpack_dec_start (GstAudioDecoder * dec)
 {
-  GstWavpackDec *dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
+  GST_DEBUG_OBJECT (dec, "start");
+
+  /* never mind a few errors */
+  gst_audio_decoder_set_max_errors (dec, 16);
+  /* don't bother us with flushing */
+  gst_audio_decoder_set_drainable (dec, FALSE);
+
+  return TRUE;
+}
+
+static gboolean
+gst_wavpack_dec_stop (GstAudioDecoder * dec)
+{
+  GstWavpackDec *wpdec = GST_WAVPACK_DEC (dec);
+
+  GST_DEBUG_OBJECT (dec, "stop");
+
+  if (wpdec->context) {
+    WavpackCloseFile (wpdec->context);
+    wpdec->context = NULL;
+  }
+
+  gst_wavpack_dec_reset (wpdec);
+
+  return TRUE;
+}
+
+static void
+gst_wavpack_dec_negotiate (GstWavpackDec * dec)
+{
+  GstCaps *caps;
+
+  /* arrange for 1, 2 or 4-byte width == depth output */
+  if (dec->depth == 24)
+    dec->width = 32;
+  else
+    dec->width = dec->depth;
+
+  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->width,
+      "width", G_TYPE_INT, dec->width,
+      "endianness", G_TYPE_INT, G_BYTE_ORDER,
+      "signed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  /* Only set the channel layout for more than two channels
+   * otherwise things break unfortunately */
+  if (dec->channel_mask != 0 && dec->channels > 2)
+    if (!gst_wavpack_set_channel_layout (caps, dec->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 (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
+  gst_caps_unref (caps);
+}
+
+static gboolean
+gst_wavpack_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
+{
+  GstWavpackDec *dec = GST_WAVPACK_DEC (bdec);
   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")) {
@@ -204,19 +252,13 @@ gst_wavpack_dec_sink_set_caps (GstPad * pad, GstCaps * caps)
         g_free (pos);
     }
 
-    GST_DEBUG_OBJECT (dec, "setting caps %" GST_PTR_FORMAT, caps);
+    gst_wavpack_dec_negotiate (dec);
-
-    /* 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;
 }
 
@@ -227,28 +269,26 @@ gst_wavpack_dec_post_tags (GstWavpackDec * dec)
   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) &&
+  if (gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
-      gst_pad_query_peer_duration (dec->sinkpad, &format_time, &duration) &&
+          &format_bytes, &size) &&
-      size > 0 && duration > 0) {
+      gst_pad_query_peer_duration (GST_AUDIO_DECODER_SINK_PAD (dec),
+          &format_time, &duration) && size > 0 && duration > 0) {
     guint64 bitrate;
 
+    list = gst_tag_list_new ();
+
     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_element_post_message (GST_ELEMENT (dec),
-      gst_message_new_tag (GST_OBJECT (dec), list));
+        gst_message_new_tag (GST_OBJECT (dec), list));
+  }
 }
 
 static GstFlowReturn
-gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
+gst_wavpack_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
 {
   GstWavpackDec *dec;
   GstBuffer *outbuf = NULL;
@@ -256,8 +296,13 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
   WavpackHeader wph;
   int32_t decoded, unpacked_size;
   gboolean format_changed;
+  gint width, depth, i, max;
+  gint32 *dec_data = NULL;
+  guint8 *out_data;
 
-  dec = GST_WAVPACK_DEC (GST_PAD_PARENT (pad));
+  dec = GST_WAVPACK_DEC (bdec);
+
+  g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
 
   /* check input, we only accept framed input with complete chunks */
   if (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader))
@@ -285,46 +330,31 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
     dec->context = WavpackOpenFileInputEx (dec->stream_reader,
         &dec->wv_id, NULL, error_msg, OPEN_STREAMING, 0);
 
+    /* expect this to work */
     if (!dec->context) {
-      GST_WARNING ("Couldn't decode buffer: %s", error_msg);
+      GST_WARNING_OBJECT (dec, "Couldn't decode buffer: %s", error_msg);
-      dec->error_count++;
+      goto context_failed;
-      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)) ||
+      (dec->depth != WavpackGetBytesPerSample (dec->context) * 8) ||
 #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) {
+  if (!GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) || format_changed) {
-    GstCaps *caps;
     gint channel_mask;
 
     dec->sample_rate = WavpackGetSampleRate (dec->context);
     dec->channels = WavpackGetNumChannels (dec->context);
-    dec->depth = WavpackGetBitsPerSample (dec->context);
+    dec->depth = WavpackGetBytesPerSample (dec->context) * 8;
-
-    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;
@@ -336,17 +366,7 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
 
     dec->channel_mask = channel_mask;
 
-    /* Only set the channel layout for more than two channels
+    gst_wavpack_dec_negotiate (dec);
-     * 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 */
@@ -354,59 +374,82 @@ gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
   }
 
   /* alloc output buffer */
-  unpacked_size = 4 * wph.block_samples * dec->channels;
+  unpacked_size = (dec->width / 8) * wph.block_samples * dec->channels;
-  ret = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET (buf),
+  ret = gst_pad_alloc_buffer (GST_AUDIO_DECODER_SRC_PAD (dec),
-      unpacked_size, GST_PAD_CAPS (dec->srcpad), &outbuf);
+      GST_BUFFER_OFFSET (buf), unpacked_size,
+      GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
 
   if (ret != GST_FLOW_OK)
     goto out;
 
-  gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
+  dec_data = g_malloc (4 * wph.block_samples * dec->channels);
-
+  out_data = GST_BUFFER_DATA (outbuf);
-  /* 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,
+  decoded = WavpackUnpackSamples (dec->context, dec_data, wph.block_samples);
-      (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,
+  width = dec->width;
-              dec->sample_rate, 4 * dec->channels))) {
+  depth = dec->depth;
-    GST_LOG_OBJECT (dec, "pushing buffer with time %" GST_TIME_FORMAT,
+  max = dec->channels * wph.block_samples;
-        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
+  if (width == 8) {
-    ret = gst_pad_push (dec->srcpad, outbuf);
+    gint8 *outbuffer = (gint8 *) out_data;
+
+    for (i = 0; i < max; i--) {
+      *outbuffer++ = (gint8) (dec_data[i]);
+    }
+  } else if (width == 16) {
+    gint16 *outbuffer = (gint16 *) out_data;
+
+    for (i = 0; i < max; i++) {
+      *outbuffer++ = (gint8) (dec_data[i]);
+    }
+  } else if (dec->width == 32) {
+    gint32 *outbuffer = (gint32 *) out_data;
+
+    if (width != depth) {
+      for (i = 0; i < max; i++) {
+        *outbuffer++ = (gint32) (dec_data[i] << (width - depth));
+      }
+    } else {
+      for (i = 0; i < max; i++) {
+        *outbuffer++ = (gint32) dec_data[i];
+      }
+    }
+  } else {
+    g_assert_not_reached ();
   }
 
+  g_free (dec_data);
+
+  ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
 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;
   }
+context_failed:
+  {
+    GST_AUDIO_DECODER_ERROR (bdec, 1, LIBRARY, INIT, (NULL),
+        ("error creating Wavpack context"), ret);
+    goto out;
+  }
 decode_error:
   {
     const gchar *reason = "unknown";
@@ -420,88 +463,17 @@ decode_error:
     } else {
       reason = "couldn't create decoder context";
     }
-    GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
+    GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
-        ("Failed to decode wavpack stream: %s", reason));
+        ("decoding error: %s", reason), ret);
+    g_free (dec_data);
     if (outbuf)
       gst_buffer_unref (outbuf);
-    gst_buffer_unref (buf);
+    if (ret == GST_FLOW_OK)
-    return GST_FLOW_ERROR;
+      gst_audio_decoder_finish_frame (bdec, NULL, 1);
+    return ret;
   }
 }
 
-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)
 {

ext/wavpack/gstwavpackdec.h

@@ -24,6 +24,7 @@
 #define __GST_WAVPACK_DEC_H__
 
 #include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
 
 #include <wavpack/wavpack.h>
 
@@ -45,31 +46,25 @@ typedef struct _GstWavpackDecClass GstWavpackDecClass;
 
 struct _GstWavpackDec
 {
-  GstElement element;
+  GstAudioDecoder 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 width;
   gint channels;
   gint channel_mask;
-
-  gint error_count;
 };
 
 struct _GstWavpackDecClass
 {
-  GstElementClass parent;
+  GstAudioDecoderClass parent;
 };
 
 GType gst_wavpack_dec_get_type (void);

ext/wavpack/gstwavpackenc.c

@@ -55,12 +55,18 @@
 #include "gstwavpackenc.h"
 #include "gstwavpackcommon.h"
 
-static GstFlowReturn gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_wavpack_enc_start (GstAudioEncoder * enc);
-static gboolean gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps);
+static gboolean gst_wavpack_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_wavpack_enc_set_format (GstAudioEncoder * enc,
+    GstAudioInfo * info);
+static GstFlowReturn gst_wavpack_enc_handle_frame (GstAudioEncoder * enc,
+    GstBuffer * in_buf);
+static gboolean gst_wavpack_enc_sink_event (GstAudioEncoder * enc,
+    GstEvent * event);
+
 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 GstFlowReturn gst_wavpack_enc_drain (GstWavpackEnc * enc);
-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,
@@ -86,7 +92,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
         "width = (int) 32, "
-        "depth = (int) [ 1, 32], "
+        "depth = (int) { 24, 32 }, "
         "endianness = (int) BYTE_ORDER, "
         "channels = (int) [ 1, 8 ], "
         "rate = (int) [ 6000, 192000 ]," "signed = (boolean) TRUE")
@@ -97,7 +103,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-wavpack, "
         "width = (int) [ 1, 32 ], "
-        "channels = (int) [ 1, 2 ], "
+        "channels = (int) [ 1, 8 ], "
         "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE")
     );
 
@@ -196,21 +202,8 @@ gst_wavpack_enc_joint_stereo_mode_get_type (void)
   return qtype;
 }
 
-static void
+GST_BOILERPLATE (GstWavpackEnc, gst_wavpack_enc, GstAudioEncoder,
-_do_init (GType object_type)
+    GST_TYPE_AUDIO_ENCODER);
-{
-  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)
@@ -232,23 +225,24 @@ gst_wavpack_enc_base_init (gpointer klass)
       "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;
+  GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) (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;
 
+  base_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_enc_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_enc_stop);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_wavpack_enc_set_format);
+  base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_wavpack_enc_handle_frame);
+  base_class->event = GST_DEBUG_FUNCPTR (gst_wavpack_enc_sink_event);
+
   /* install all properties */
   g_object_class_install_property (gobject_class, ARG_MODE,
       g_param_spec_enum ("mode", "Encoding mode",
@@ -306,6 +300,9 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
     g_checksum_free (enc->md5_context);
     enc->md5_context = NULL;
   }
+  if (enc->pending_segment)
+    gst_event_unref (enc->pending_segment);
+  enc->pending_segment = NULL;
 
   if (enc->pending_buffer) {
     gst_buffer_unref (enc->pending_buffer);
@@ -332,18 +329,7 @@ gst_wavpack_enc_reset (GstWavpackEnc * enc)
 static void
 gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
 {
-  enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
+  GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
-  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;
@@ -367,37 +353,51 @@ gst_wavpack_enc_init (GstWavpackEnc * enc, GstWavpackEncClass * gclass)
   enc->md5 = FALSE;
   enc->extra_processing = 0;
   enc->joint_stereo_mode = GST_WAVPACK_JS_MODE_AUTO;
+
+  /* require perfect ts */
+  gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+}
+
+
+static gboolean
+gst_wavpack_enc_start (GstAudioEncoder * enc)
+{
+  GST_DEBUG_OBJECT (enc, "start");
+
+  return TRUE;
 }
 
 static gboolean
-gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
+gst_wavpack_enc_stop (GstAudioEncoder * enc)
 {
-  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  GstWavpackEnc *wpenc = GST_WAVPACK_ENC (enc);
-  GstStructure *structure = gst_caps_get_structure (caps, 0);
+
+  GST_DEBUG_OBJECT (enc, "stop");
+  gst_wavpack_enc_reset (wpenc);
+
+  return TRUE;
+}
+
+static gboolean
+gst_wavpack_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
   GstAudioChannelPosition *pos;
+  GstCaps *caps;
 
-  if (!gst_structure_get_int (structure, "channels", &enc->channels) ||
+  /* we may be configured again, but that change should have cleanup context */
-      !gst_structure_get_int (structure, "rate", &enc->samplerate) ||
+  g_assert (enc->wp_context == NULL);
-      !gst_structure_get_int (structure, "depth", &enc->depth)) {
+
-    GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+  enc->channels = GST_AUDIO_INFO_CHANNELS (info);
-        ("got invalid caps: %" GST_PTR_FORMAT, caps));
+  enc->depth = GST_AUDIO_INFO_DEPTH (info);
-    gst_object_unref (enc);
+  enc->samplerate = GST_AUDIO_INFO_RATE (info);
-    return FALSE;
+
-  }
+  pos = info->position;
+  g_assert (pos);
 
-  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);
+    goto invalid_channels;
-    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 =
@@ -405,7 +405,6 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
   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",
@@ -416,18 +415,28 @@ gst_wavpack_enc_sink_set_caps (GstPad * pad, GstCaps * caps)
   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)) {
+  if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps))
-    GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+    goto setting_src_caps_failed;
-        ("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);
+
+  /* no special feedback to base class; should provide all available samples */
+
   return TRUE;
+
+  /* ERRORS */
+setting_src_caps_failed:
+  {
+    GST_DEBUG_OBJECT (enc,
+        "Couldn't set caps on source pad: %" GST_PTR_FORMAT, caps);
+    gst_caps_unref (caps);
+    return FALSE;
+  }
+invalid_channels:
+  {
+    GST_DEBUG_OBJECT (enc, "input has invalid channel layout");
+    return FALSE;
+  }
 }
 
 static void
@@ -549,21 +558,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
   GstBuffer *buffer;
   GstPad *pad;
   guchar *block = (guchar *) data;
+  gint samples = 0;
 
-  pad = (wid->correction) ? enc->wvcsrcpad : enc->srcpad;
+  pad = (wid->correction) ? enc->wvcsrcpad : GST_AUDIO_ENCODER_SRC_PAD (enc);
   flow =
       (wid->correction) ? &enc->wvcsrcpad_last_return : &enc->
       srcpad_last_return;
 
-  *flow = gst_pad_alloc_buffer_and_set_caps (pad, GST_BUFFER_OFFSET_NONE,
+  buffer = gst_buffer_new_and_alloc (count);
-      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) {
@@ -599,12 +601,14 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
       enc->pending_buffer = NULL;
       enc->pending_offset = 0;
 
-      /* if it's the first wavpack block, send a NEW_SEGMENT event */
+      /* only send segment on correction pad,
-      if (wph.block_index == 0) {
+       * regular pad is handled normally by baseclass */
-        gst_pad_push_event (pad,
+      if (wid->correction && enc->pending_segment) {
-            gst_event_new_new_segment (FALSE,
+        gst_pad_push_event (pad, enc->pending_segment);
-                1.0, GST_FORMAT_TIME, 0, GST_BUFFER_OFFSET_NONE, 0));
+        enc->pending_segment = NULL;
+      }
 
+      if (wph.block_index == 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) {
@@ -614,29 +618,23 @@ gst_wavpack_enc_push_block (void *id, void *data, int32_t count)
         }
       }
     }
-
+    samples = wph.block_samples;
-    /* 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 */
+  if (wid->correction || wid->passthrough) {
-  GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
+    /* push the buffer and forward errors */
-      GST_BUFFER_SIZE (buffer));
+    GST_DEBUG_OBJECT (enc, "pushing buffer with %d bytes",
-  *flow = gst_pad_push (pad, buffer);
+        GST_BUFFER_SIZE (buffer));
+    *flow = gst_pad_push (pad, buffer);
+  } else {
+    GST_DEBUG_OBJECT (enc, "handing frame of %d bytes",
+        GST_BUFFER_SIZE (buffer));
+    *flow = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), buffer,
+        samples);
+  }
 
   if (*flow != GST_FLOW_OK) {
     GST_WARNING_OBJECT (enc, "flow on %s:%s = %s",
@@ -666,18 +664,25 @@ gst_wavpack_enc_fix_channel_order (GstWavpackEnc * enc, gint32 * data,
 }
 
 static GstFlowReturn
-gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
+gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
 {
-  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
-  uint32_t sample_count = GST_BUFFER_SIZE (buf) / 4;
+  uint32_t sample_count;
   GstFlowReturn ret;
 
+  /* base class ensures configuration */
+  g_return_val_if_fail (enc->depth != 0, GST_FLOW_NOT_NEGOTIATED);
+
   /* 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);
+  if (G_UNLIKELY (!buf))
+    return gst_wavpack_enc_drain (enc);
+
+  sample_count = GST_BUFFER_SIZE (buf) / 4;
+  GST_DEBUG_OBJECT (enc, "got %u raw samples", sample_count);
 
   /* check if we already have a valid WavpackContext, otherwise make one */
   if (!enc->wp_context) {
@@ -685,13 +690,8 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
     enc->wp_context =
         WavpackOpenFileOutput (gst_wavpack_enc_push_block, &enc->wv_id,
         (enc->correction_mode > 0) ? &enc->wvc_id : NULL);
-    if (!enc->wp_context) {
+    if (!enc->wp_context)
-      GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+      goto context_failed;
-          ("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);
@@ -701,76 +701,12 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
     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);
+      goto config_failed;
-      gst_buffer_unref (buf);
-      return GST_FLOW_ERROR;
     }
-    GST_DEBUG ("setup of encoding context successfull");
+    GST_DEBUG_OBJECT (enc, "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),
@@ -787,7 +723,7 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * 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");
+    GST_DEBUG_OBJECT (enc, "encoding samples successful");
     ret = GST_FLOW_OK;
   } else {
     if ((enc->srcpad_last_return == GST_FLOW_RESEND) ||
@@ -803,15 +739,35 @@ gst_wavpack_enc_chain (GstPad * pad, GstBuffer * buf)
         (enc->wvcsrcpad_last_return == GST_FLOW_FLUSHING)) {
       ret = GST_FLOW_FLUSHING;
     } else {
-      GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
+      goto encoding_failed;
-          ("encoding samples failed"));
-      ret = GST_FLOW_ERROR;
     }
   }
 
-  gst_buffer_unref (buf);
+exit:
-  gst_object_unref (enc);
   return ret;
+
+  /* ERRORS */
+encoding_failed:
+  {
+    GST_ELEMENT_ERROR (enc, LIBRARY, ENCODE, (NULL),
+        ("encoding samples failed"));
+    ret = GST_FLOW_ERROR;
+    goto exit;
+  }
+config_failed:
+  {
+    GST_ELEMENT_ERROR (enc, LIBRARY, SETTINGS, (NULL),
+        ("error setting up wavpack encoding context"));
+    ret = GST_FLOW_ERROR;
+    goto exit;
+  }
+context_failed:
+  {
+    GST_ELEMENT_ERROR (enc, LIBRARY, INIT, (NULL),
+        ("error creating Wavpack context"));
+    ret = GST_FLOW_ERROR;
+    goto exit;
+  }
 }
 
 static void
@@ -828,7 +784,7 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
   WavpackUpdateNumSamples (enc->wp_context, enc->first_block);
 
   /* try to seek to the beginning of the output */
-  ret = gst_pad_push_event (enc->srcpad, event);
+  ret = gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc), event);
   if (ret) {
     /* try to rewrite the first block */
     GST_DEBUG_OBJECT (enc, "rewriting first block ...");
@@ -836,111 +792,84 @@ gst_wavpack_enc_rewrite_first_block (GstWavpackEnc * enc)
     ret = gst_wavpack_enc_push_block (&enc->wv_id,
         enc->first_block, enc->first_block_size);
     enc->wv_id.passthrough = FALSE;
+    g_free (enc->first_block);
+    enc->first_block = NULL;
   } else {
     GST_WARNING_OBJECT (enc, "rewriting of first block failed. "
         "Seeking to first block failed!");
   }
 }
 
-static gboolean
+static GstFlowReturn
-gst_wavpack_enc_sink_event (GstPad * pad, GstEvent * event)
+gst_wavpack_enc_drain (GstWavpackEnc * enc)
 {
-  GstWavpackEnc *enc = GST_WAVPACK_ENC (gst_pad_get_parent (pad));
+  if (!enc->wp_context)
-  gboolean ret = TRUE;
+    return GST_FLOW_OK;
 
-  GST_DEBUG ("Received %s event on sinkpad", GST_EVENT_TYPE_NAME (event));
+  GST_DEBUG_OBJECT (enc, "draining");
+
+  /* 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);
+      WavpackFlushSamples (enc->wp_context);
+    } 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;
+  }
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_wavpack_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
+{
+  GstWavpackEnc *enc = GST_WAVPACK_ENC (benc);
+
+  GST_DEBUG_OBJECT (enc, "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
+      /* peek and hold NEWSEGMENT events for sending on correction pad */
-       * the first buffer to the pads */
+      if (enc->pending_segment)
-      gst_event_unref (event);
+        gst_event_unref (enc->pending_segment);
-      ret = TRUE;
+      enc->pending_segment = gst_event_ref (event);
-      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;
+  /* baseclass handles rest */
+  return FALSE;
 }
 
 static void

ext/wavpack/gstwavpackenc.h

@@ -23,6 +23,7 @@
 #define __GST_WAVPACK_ENC_H__
 
 #include <gst/gst.h>
+#include <gst/audio/gstaudioencoder.h>
 
 #include <wavpack/wavpack.h>
 
@@ -50,10 +51,9 @@ typedef struct
 
 struct _GstWavpackEnc
 {
-  GstElement element;
+  GstAudioEncoder element;
 
   /*< private > */
-  GstPad *sinkpad, *srcpad;
   GstPad *wvcsrcpad;
 
   GstFlowReturn srcpad_last_return;
@@ -86,6 +86,7 @@ struct _GstWavpackEnc
 
   GstBuffer *pending_buffer;
   gint32 pending_offset;
+  GstEvent *pending_segment;
 
   GstClockTime timestamp_offset;
   GstClockTime next_ts;
@@ -93,7 +94,7 @@ struct _GstWavpackEnc
 
 struct _GstWavpackEncClass
 {
-  GstElementClass parent;
+  GstAudioEncoderClass parent;
 };
 
 GType gst_wavpack_enc_get_type (void);

gst/audiofx/audiofirfilter.c

@@ -45,6 +45,10 @@
  * </refsect2>
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

gst/audiofx/audioiirfilter.c

@@ -41,6 +41,10 @@
  * </refsect2>
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

gst/audioparsers/Makefile.am

@@ -3,7 +3,7 @@ plugin_LTLIBRARIES = libgstaudioparsers.la
 libgstaudioparsers_la_SOURCES = \
 	gstaacparse.c gstamrparse.c gstac3parse.c \
 	gstdcaparse.c gstflacparse.c gstmpegaudioparse.c \
-	plugin.c
+	gstwavpackparse.c plugin.c
 
 libgstaudioparsers_la_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
@@ -15,4 +15,4 @@ libgstaudioparsers_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstaudioparsers_la_LIBTOOLFLAGS = --tag=disable-static
 
 noinst_HEADERS = gstaacparse.h gstamrparse.h gstac3parse.h \
-	gstdcaparse.h gstflacparse.h gstmpegaudioparse.h
+	gstdcaparse.h gstflacparse.h gstmpegaudioparse.h gstwavpackparse.h

gst/audioparsers/gstwavpackparse.c

@@ -0,0 +1,648 @@
+/* GStreamer Wavpack parser
+ * Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * Copyright (C) 2012 Nokia Corporation. All rights reserved.
+ *   Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * 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
+ * @short_description: Wavpack parser
+ * @see_also: #GstAmrParse, #GstAACParse
+ *
+ * This is an Wavpack parser.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
+ * ]|
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "gstwavpackparse.h"
+
+#include <gst/base/gstbytereader.h>
+#include <gst/audio/multichannel.h>
+
+GST_DEBUG_CATEGORY_STATIC (wavpack_parse_debug);
+#define GST_CAT_DEFAULT wavpack_parse_debug
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-wavpack, "
+        "width = (int) [ 1, 32 ], "
+        "channels = (int) [ 1, 8 ], "
+        "rate = (int) [ 6000, 192000 ], " "framed = (boolean) TRUE; "
+        "audio/x-wavpack-correction, " "framed = (boolean) TRUE")
+    );
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-wavpack"));
+
+static void gst_wavpack_parse_finalize (GObject * object);
+
+static gboolean gst_wavpack_parse_start (GstBaseParse * parse);
+static gboolean gst_wavpack_parse_stop (GstBaseParse * parse);
+static gboolean gst_wavpack_parse_check_valid_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame, guint * size, gint * skipsize);
+static GstFlowReturn gst_wavpack_parse_parse_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame);
+static GstCaps *gst_wavpack_parse_get_sink_caps (GstBaseParse * parse);
+
+/* FIXME remove when all properly renamed */
+typedef GstWavpackParse GstWavpackParse2;
+typedef GstWavpackParseClass GstWavpackParse2Class;
+
+GST_BOILERPLATE (GstWavpackParse2, gst_wavpack_parse, GstBaseParse,
+    GST_TYPE_BASE_PARSE);
+
+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, &sink_template);
+  gst_element_class_add_static_pad_template (element_class, &src_template);
+
+  gst_element_class_set_details_simple (element_class,
+      "Wavpack audio stream parser", "Codec/Parser/Audio",
+      "Wavpack parser", "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
+}
+
+static void
+gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
+{
+  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+
+  GST_DEBUG_CATEGORY_INIT (wavpack_parse_debug, "wavpackparse", 0,
+      "Wavpack audio stream parser");
+
+  object_class->finalize = gst_wavpack_parse_finalize;
+
+  parse_class->start = GST_DEBUG_FUNCPTR (gst_wavpack_parse_start);
+  parse_class->stop = GST_DEBUG_FUNCPTR (gst_wavpack_parse_stop);
+  parse_class->check_valid_frame =
+      GST_DEBUG_FUNCPTR (gst_wavpack_parse_check_valid_frame);
+  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_wavpack_parse_parse_frame);
+  parse_class->get_sink_caps =
+      GST_DEBUG_FUNCPTR (gst_wavpack_parse_get_sink_caps);
+}
+
+static void
+gst_wavpack_parse_reset (GstWavpackParse * wvparse)
+{
+  wvparse->channels = -1;
+  wvparse->channel_mask = 0;
+  wvparse->sample_rate = -1;
+  wvparse->width = -1;
+  wvparse->total_samples = 0;
+}
+
+static void
+gst_wavpack_parse_init (GstWavpackParse * wvparse, GstWavpackParseClass * klass)
+{
+  gst_wavpack_parse_reset (wvparse);
+}
+
+static void
+gst_wavpack_parse_finalize (GObject * object)
+{
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_wavpack_parse_start (GstBaseParse * parse)
+{
+  GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
+
+  GST_DEBUG_OBJECT (parse, "starting");
+
+  gst_wavpack_parse_reset (wvparse);
+
+  /* need header at least */
+  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (wvparse),
+      sizeof (WavpackHeader));
+
+  /* inform baseclass we can come up with ts, based on counters in packets */
+  gst_base_parse_set_has_timing_info (GST_BASE_PARSE_CAST (wvparse), TRUE);
+  gst_base_parse_set_syncable (GST_BASE_PARSE_CAST (wvparse), TRUE);
+
+  return TRUE;
+}
+
+static gboolean
+gst_wavpack_parse_stop (GstBaseParse * parse)
+{
+  GST_DEBUG_OBJECT (parse, "stopping");
+
+  return TRUE;
+}
+
+static 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)
+
+static 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;
+}
+
+static const guint32 sample_rates[] = {
+  6000, 8000, 9600, 11025, 12000, 16000, 22050,
+  24000, 32000, 44100, 48000, 64000, 88200, 96000, 192000
+};
+
+#define CHECK(call) { \
+  if (!call) \
+    goto read_failed; \
+}
+
+/* caller ensures properly sync'ed with enough data */
+static gboolean
+gst_wavpack_parse_frame_metadata (GstWavpackParse * parse, GstBuffer * buf,
+    gint skip, WavpackHeader * wph, WavpackInfo * wpi)
+{
+  GstByteReader br;
+  gint i;
+
+  g_return_val_if_fail (wph != NULL || wpi != NULL, FALSE);
+  g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader),
+      FALSE);
+
+  gst_byte_reader_init (&br, GST_BUFFER_DATA (buf) + skip, wph->ckSize + 8);
+  /* skip past header */
+  gst_byte_reader_skip_unchecked (&br, sizeof (WavpackHeader));
+
+  /* get some basics from header */
+  i = (wph->flags >> 23) & 0xF;
+  if (!wpi->rate)
+    wpi->rate = (i < G_N_ELEMENTS (sample_rates)) ? sample_rates[i] : 44100;
+  wpi->width = ((wph->flags & 0x3) + 1) * 8;
+  if (!wpi->channels)
+    wpi->channels = (wph->flags & 0x4) ? 1 : 2;
+  if (!wpi->channel_mask)
+    wpi->channel_mask = 5 - wpi->channels;
+
+  /* need to dig metadata blocks for some more */
+  while (gst_byte_reader_get_remaining (&br)) {
+    gint size = 0;
+    guint16 size2 = 0;
+    guint8 c, id;
+    const guint8 *data;
+    GstByteReader mbr;
+
+    CHECK (gst_byte_reader_get_uint8 (&br, &id));
+    CHECK (gst_byte_reader_get_uint8 (&br, &c));
+    if (id & ID_LARGE)
+      CHECK (gst_byte_reader_get_uint16_le (&br, &size2));
+    size = size2;
+    size <<= 8;
+    size += c;
+    size <<= 1;
+    if (id & ID_ODD_SIZE)
+      size--;
+
+    CHECK (gst_byte_reader_get_data (&br, size + (size & 1), &data));
+    gst_byte_reader_init (&mbr, data, size);
+
+    switch (id) {
+      case ID_WVC_BITSTREAM:
+        GST_LOG_OBJECT (parse, "correction bitstream");
+        wpi->correction = TRUE;
+        break;
+      case ID_WV_BITSTREAM:
+      case ID_WVX_BITSTREAM:
+        break;
+      case ID_SAMPLE_RATE:
+        if (size == 3) {
+          CHECK (gst_byte_reader_get_uint24_le (&mbr, &wpi->rate));
+          GST_LOG_OBJECT (parse, "updated with custom rate %d", wpi->rate);
+        } else {
+          GST_DEBUG_OBJECT (parse, "unexpected size for SAMPLE_RATE metadata");
+        }
+        break;
+      case ID_CHANNEL_INFO:
+      {
+        guint16 channels;
+        guint32 mask = 0;
+
+        if (size == 6) {
+          CHECK (gst_byte_reader_get_uint16_le (&mbr, &channels));
+          channels = channels & 0xFFF;
+          CHECK (gst_byte_reader_get_uint24_le (&mbr, &mask));
+        } else if (size) {
+          CHECK (gst_byte_reader_get_uint8 (&mbr, &c));
+          channels = c;
+          while (gst_byte_reader_get_uint8 (&mbr, &c))
+            mask |= (((guint32) c) << 8);
+        } else {
+          GST_DEBUG_OBJECT (parse, "unexpected size for CHANNEL_INFO metadata");
+          break;
+        }
+        wpi->channels = channels;
+        wpi->channel_mask = mask;
+        break;
+      }
+      default:
+        GST_LOG_OBJECT (parse, "unparsed ID 0x%x", id);
+        break;
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+read_failed:
+  {
+    GST_DEBUG_OBJECT (parse, "short read while parsing metadata");
+    /* let's look the other way anyway */
+    return TRUE;
+  }
+}
+
+/* caller ensures properly sync'ed with enough data */
+static gboolean
+gst_wavpack_parse_frame_header (GstWavpackParse * parse, GstBuffer * buf,
+    gint skip, WavpackHeader * _wph)
+{
+  GstByteReader br = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
+  WavpackHeader wph;
+
+  g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= skip + sizeof (WavpackHeader),
+      FALSE);
+
+  /* marker */
+  gst_byte_reader_skip_unchecked (&br, skip + 4);
+
+  /* read */
+  gst_byte_reader_get_uint32_le (&br, &wph.ckSize);
+  gst_byte_reader_get_uint16_le (&br, &wph.version);
+  gst_byte_reader_get_uint8 (&br, &wph.track_no);
+  gst_byte_reader_get_uint8 (&br, &wph.index_no);
+  gst_byte_reader_get_uint32_le (&br, &wph.total_samples);
+  gst_byte_reader_get_uint32_le (&br, &wph.block_index);
+  gst_byte_reader_get_uint32_le (&br, &wph.block_samples);
+  gst_byte_reader_get_uint32_le (&br, &wph.flags);
+  gst_byte_reader_get_uint32_le (&br, &wph.crc);
+
+  /* dump */
+  GST_LOG_OBJECT (parse, "size %d", wph.ckSize);
+  GST_LOG_OBJECT (parse, "version 0x%x", wph.version);
+  GST_LOG_OBJECT (parse, "total samples %d", wph.total_samples);
+  GST_LOG_OBJECT (parse, "block index %d", wph.block_index);
+  GST_LOG_OBJECT (parse, "block samples %d", wph.block_samples);
+  GST_LOG_OBJECT (parse, "flags 0x%x", wph.flags);
+  GST_LOG_OBJECT (parse, "crc 0x%x", wph.flags);
+
+  if (!parse->total_samples && wph.block_index == 0 && wph.total_samples != -1) {
+    GST_DEBUG_OBJECT (parse, "determined duration of %u samples",
+        wph.total_samples);
+    parse->total_samples = wph.total_samples;
+  }
+
+  if (_wph)
+    *_wph = wph;
+
+  return TRUE;
+}
+
+static gboolean
+gst_wavpack_parse_check_valid_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
+{
+  GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
+  GstBuffer *buf = frame->buffer;
+  GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
+  gint off;
+  gboolean lost_sync, draining, final;
+  guint frmsize = 0;
+  WavpackHeader wph;
+  WavpackInfo wpi = { 0, };
+
+  if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < sizeof (WavpackHeader)))
+    return FALSE;
+
+  /* scan for 'wvpk' marker */
+  off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffffffff, 0x7776706b,
+      0, GST_BUFFER_SIZE (buf));
+
+  GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
+
+  /* didn't find anything that looks like a sync word, skip */
+  if (off < 0) {
+    *skipsize = GST_BUFFER_SIZE (buf) - 3;
+    goto skip;
+  }
+
+  /* possible frame header, but not at offset 0? skip bytes before sync */
+  if (off > 0) {
+    *skipsize = off;
+    goto skip;
+  }
+
+  /* make sure the values in the frame header look sane */
+  gst_wavpack_parse_frame_header (wvparse, buf, 0, &wph);
+  frmsize = wph.ckSize + 8;
+
+  /* need the entire frame for parsing */
+  if (gst_byte_reader_get_remaining (&reader) < frmsize)
+    goto more;
+
+  /* got a frame, now we can dig for some more metadata */
+  GST_LOG_OBJECT (parse, "got frame");
+  gst_wavpack_parse_frame_metadata (wvparse, buf, 0, &wph, &wpi);
+
+  lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
+  draining = GST_BASE_PARSE_DRAINING (parse);
+
+  while (!(final = (wph.flags & FLAG_FINAL_BLOCK)) || (lost_sync && !draining)) {
+    guint32 word = 0;
+
+    GST_LOG_OBJECT (wvparse, "checking next frame syncword; "
+        "lost_sync: %d, draining: %d, final: %d", lost_sync, draining, final);
+
+    if (!gst_byte_reader_skip (&reader, wph.ckSize + 8) ||
+        !gst_byte_reader_peek_uint32_be (&reader, &word)) {
+      GST_DEBUG_OBJECT (wvparse, "... but not sufficient data");
+      frmsize += 4;
+      goto more;
+    } else {
+      if (word != 0x7776706b) {
+        GST_DEBUG_OBJECT (wvparse, "0x%x not OK", word);
+        *skipsize = off + 2;
+        goto skip;
+      }
+      /* need to parse each frame/block for metadata if several ones */
+      if (!final) {
+        gint av;
+
+        GST_LOG_OBJECT (wvparse, "checking frame at offset %d (0x%x)",
+            frmsize, frmsize);
+        av = gst_byte_reader_get_remaining (&reader);
+        if (av < sizeof (WavpackHeader)) {
+          frmsize += sizeof (WavpackHeader);
+          goto more;
+        }
+        gst_wavpack_parse_frame_header (wvparse, buf, frmsize, &wph);
+        off = frmsize;
+        frmsize += wph.ckSize + 8;
+        if (av < wph.ckSize + 8)
+          goto more;
+        gst_wavpack_parse_frame_metadata (wvparse, buf, off, &wph, &wpi);
+        /* could also check for matching block_index and block_samples ?? */
+      }
+    }
+
+    /* resynced if we make it here */
+    lost_sync = FALSE;
+  }
+
+  /* found frame (up to final), record gathered metadata */
+  wvparse->wpi = wpi;
+  wvparse->wph = wph;
+
+  *framesize = frmsize;
+  gst_base_parse_set_min_frame_size (parse, sizeof (WavpackHeader));
+
+  return TRUE;
+
+skip:
+  GST_LOG_OBJECT (wvparse, "skipping %d", *skipsize);
+  return FALSE;
+
+more:
+  GST_LOG_OBJECT (wvparse, "need at least %u", frmsize);
+  gst_base_parse_set_min_frame_size (parse, frmsize);
+  *skipsize = 0;
+  return FALSE;
+}
+
+static GstFlowReturn
+gst_wavpack_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
+{
+  GstWavpackParse *wvparse = GST_WAVPACK_PARSE (parse);
+  GstBuffer *buf = frame->buffer;
+  guint rate, chans, width, mask;
+
+  /* re-use previously parsed data */
+  rate = wvparse->wpi.rate;
+  width = wvparse->wpi.width;
+  chans = wvparse->wpi.channels;
+  mask = wvparse->wpi.channel_mask;
+
+  GST_LOG_OBJECT (parse, "rate: %u, width: %u, chans: %u", rate, width, chans);
+
+  GST_BUFFER_TIMESTAMP (buf) =
+      gst_util_uint64_scale_int (wvparse->wph.block_index, GST_SECOND, rate);
+  GST_BUFFER_DURATION (buf) =
+      gst_util_uint64_scale_int (wvparse->wph.block_index +
+      wvparse->wph.block_samples, GST_SECOND, rate) -
+      GST_BUFFER_TIMESTAMP (buf);
+
+  if (G_UNLIKELY (wvparse->sample_rate != rate || wvparse->channels != chans
+          || wvparse->width != width || wvparse->channel_mask != mask)) {
+    GstCaps *caps;
+
+    if (wvparse->wpi.correction) {
+      caps = gst_caps_new_simple ("audio/x-wavpack-correction",
+          "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+    } else {
+      caps = gst_caps_new_simple ("audio/x-wavpack",
+          "channels", G_TYPE_INT, chans,
+          "rate", G_TYPE_INT, rate,
+          "width", G_TYPE_INT, width, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+      if (!mask)
+        mask = gst_wavpack_get_default_channel_mask (wvparse->channels);
+      if (mask != 0) {
+        if (!gst_wavpack_set_channel_layout (caps, mask)) {
+          GST_WARNING_OBJECT (wvparse, "Failed to set channel layout");
+        }
+      }
+    }
+
+    gst_buffer_set_caps (buf, caps);
+    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
+    gst_caps_unref (caps);
+
+    wvparse->sample_rate = rate;
+    wvparse->channels = chans;
+    wvparse->width = width;
+    wvparse->channel_mask = mask;
+
+    if (wvparse->total_samples) {
+      GST_DEBUG_OBJECT (wvparse, "setting duration");
+      gst_base_parse_set_duration (GST_BASE_PARSE (wvparse),
+          GST_FORMAT_TIME, gst_util_uint64_scale_int (wvparse->total_samples,
+              GST_SECOND, wvparse->sample_rate), 0);
+    }
+  }
+
+  return GST_FLOW_OK;
+}
+
+static GstCaps *
+gst_wavpack_parse_get_sink_caps (GstBaseParse * parse)
+{
+  GstCaps *peercaps;
+  GstCaps *res;
+
+  peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
+  if (peercaps) {
+    guint i, n;
+
+    /* Remove the framed field */
+    peercaps = gst_caps_make_writable (peercaps);
+    n = gst_caps_get_size (peercaps);
+    for (i = 0; i < n; i++) {
+      GstStructure *s = gst_caps_get_structure (peercaps, i);
+
+      gst_structure_remove_field (s, "framed");
+    }
+
+    res =
+        gst_caps_intersect_full (peercaps,
+        gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
+        GST_CAPS_INTERSECT_FIRST);
+    gst_caps_unref (peercaps);
+  } else {
+    res =
+        gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
+            (parse)));
+  }
+
+  return res;
+}

gst/audioparsers/gstwavpackparse.h

@@ -0,0 +1,134 @@
+/* GStreamer Wavpack parser
+ * Copyright (C) 2012 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+ * Copyright (C) 2012 Nokia Corporation. All rights reserved.
+ *   Contact: Stefan Kost <stefan.kost@nokia.com>
+ *
+ * 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/gstbaseparse.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))
+
+
+#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)
+
+#define FLAG_FINAL_BLOCK        (1 << 12)
+
+typedef struct {
+  char ckID [4];             /* "wvpk" */
+  guint32 ckSize;            /* size of entire block (minus 8, of course) */
+  guint16 version;           /* 0x402 to 0x410 are currently valid for decode */
+  guchar track_no;           /* track number (0 if not used, like now) */
+  guchar index_no;           /* track sub-index (0 if not used, like now) */
+  guint32 total_samples;     /* total samples for entire file, but this is
+                              * only valid if block_index == 0 and a value of
+                              * -1 indicates unknown length */
+  guint32 block_index;       /* index of first sample in block relative to
+                              * beginning of file (normally this would start
+                              * at 0 for the first block) */
+  guint32 block_samples;     /* number of samples in this block (0 = no audio) */
+  guint32 flags;             /* various flags for id and decoding */
+  guint32 crc;               /* crc for actual decoded data */
+} WavpackHeader;
+
+typedef struct {
+  gboolean correction;
+  guint rate;
+  guint width;
+  guint channels;
+  guint channel_mask;
+} WavpackInfo;
+
+typedef struct _GstWavpackParse GstWavpackParse;
+typedef struct _GstWavpackParseClass GstWavpackParseClass;
+
+/**
+ * GstWavpackParse:
+ *
+ * The opaque GstWavpackParse object
+ */
+struct _GstWavpackParse {
+  GstBaseParse baseparse;
+
+  /*< private >*/
+  gint          sample_rate;
+  gint          channels;
+  gint          width;
+  gint          channel_mask;
+
+  guint         total_samples;
+
+  WavpackHeader wph;
+  WavpackInfo   wpi;
+};
+
+/**
+ * GstWavpackParseClass:
+ * @parent_class: Element parent class.
+ *
+ * The opaque GstWavpackParseClass data structure.
+ */
+struct _GstWavpackParseClass {
+  GstBaseParseClass baseparse_class;
+};
+
+GType gst_wavpack_parse_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_WAVPACK_PARSE_H__ */

gst/audioparsers/plugin.c

@@ -27,6 +27,7 @@
 #include "gstdcaparse.h"
 #include "gstflacparse.h"
 #include "gstmpegaudioparse.h"
+#include "gstwavpackparse.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
@@ -45,6 +46,8 @@ plugin_init (GstPlugin * plugin)
       GST_RANK_PRIMARY + 1, GST_TYPE_FLAC_PARSE);
   ret &= gst_element_register (plugin, "mpegaudioparse",
       GST_RANK_PRIMARY + 2, GST_TYPE_MPEG_AUDIO_PARSE);
+  ret &= gst_element_register (plugin, "wavpackparse2",
+      GST_RANK_SECONDARY, GST_TYPE_WAVPACK_PARSE);
 
   return ret;
 }

gst/interleave/interleave.c

@@ -57,6 +57,10 @@
  * </refsect2>
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #  include "config.h"
 #endif

gst/rtpmanager/rtpsession.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 
 #include <gst/rtp/gstrtpbuffer.h>

gst/udp/gstdynudpsink.c

@@ -21,6 +21,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

gst/udp/gstmultiudpsink.c

@@ -29,6 +29,10 @@
  * It can be combined with rtp payload encoders to implement RTP streaming.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

sys/oss4/oss4-audio.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

sys/oss4/oss4-property-probe.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include <config.h>
 #endif

sys/v4l2/gstv4l2object.c

@@ -18,6 +18,10 @@
  * USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include <config.h>
 #endif

tests/check/elements/interleave.c

@@ -18,6 +18,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 # include "config.h"
 #endif

tests/check/elements/wavpackdec.c

@@ -52,7 +52,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
-        "width = (int) 32, "
+        "width = (int) 16, "
         "depth = (int) 16, "
         "channels = (int) 1, "
         "rate = (int) 44100, "
@@ -112,23 +112,20 @@ GST_START_TEST (test_decode_frame)
   memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame));
   ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
   GST_BUFFER_TIMESTAMP (inbuffer) = 0;
-  gst_buffer_ref (inbuffer);
 
   gst_element_set_bus (wavpackdec, bus);
 
   /* should decode the buffer without problems */
   fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_unref (inbuffer);
 
   outbuffer = GST_BUFFER (buffers->data);
 
   fail_if (outbuffer == NULL);
   /* uncompressed data should be 102400 bytes */
-  fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 102400);
+  fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), 51200);
 
-  /* and all 102400 bytes must be 0, i.e. silence */
+  /* and all bytes must be 0, i.e. silence */
-  for (i = 0; i < 102400; i++)
+  for (i = 0; i < 51200; i++)
     fail_unless_equals_int (GST_BUFFER_DATA (outbuffer)[i], 0);
 
   ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
@@ -166,14 +163,11 @@ GST_START_TEST (test_decode_frame_with_broken_header)
   GST_BUFFER_DATA (inbuffer)[2] = 'e';
   ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
   GST_BUFFER_TIMESTAMP (inbuffer) = 0;
-  gst_buffer_ref (inbuffer);
 
   gst_element_set_bus (wavpackdec, bus);
 
   /* should fail gracefully */
   fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_unref (inbuffer);
 
   fail_if ((message = gst_bus_pop (bus)) == NULL);
   fail_unless_message_error (message, STREAM, DECODE);
@@ -204,14 +198,11 @@ GST_START_TEST (test_decode_frame_with_incomplete_frame)
   memcpy (GST_BUFFER_DATA (inbuffer), test_frame, sizeof (test_frame) - 2);
   ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
   GST_BUFFER_TIMESTAMP (inbuffer) = 0;
-  gst_buffer_ref (inbuffer);
 
   gst_element_set_bus (wavpackdec, bus);
 
   /* should fail gracefully */
   fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_unref (inbuffer);
 
   fail_if ((message = gst_bus_pop (bus)) == NULL);
   fail_unless_message_error (message, STREAM, DECODE);

tests/check/elements/wavpackenc.c

@@ -32,14 +32,14 @@ static GstBus *bus;
 
 #define RAW_CAPS_STRING "audio/x-raw-int, " \
                         "width = (int) 32, " \
-                        "depth = (int) 16, " \
+                        "depth = (int) 32, " \
                         "channels = (int) 1, " \
                         "rate = (int) 44100, " \
                         "endianness = (int) BYTE_ORDER, " \
                         "signed = (boolean) true"
 
 #define WAVPACK_CAPS_STRING "audio/x-wavpack, " \
-                            "width = (int) 16, " \
+                            "width = (int) 32, " \
                             "channels = (int) 1, " \
                             "rate = (int) 44100, " \
                             "framed = (boolean) true"
@@ -48,7 +48,7 @@ static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-wavpack, "
-        "width = (int) 16, "
+        "width = (int) 32, "
         "channels = (int) 1, "
         "rate = (int) 44100, " "framed = (boolean) true"));
 
@@ -57,7 +57,7 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-raw-int, "
         "width = (int) 32, "
-        "depth = (int) 16, "
+        "depth = (int) 32, "
         "channels = (int) 1, "
         "rate = (int) 44100, "
         "endianness = (int) BYTE_ORDER, " "signed = (boolean) true"));
@@ -118,13 +118,10 @@ GST_START_TEST (test_encode_silence)
   gst_caps_unref (caps);
   GST_BUFFER_TIMESTAMP (inbuffer) = 0;
   ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_ref (inbuffer);
 
   gst_element_set_bus (wavpackenc, bus);
 
   fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
-  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
-  gst_buffer_unref (inbuffer);
 
   fail_if (gst_pad_push_event (mysrcpad, eos) != TRUE);
 
@@ -134,9 +131,7 @@ GST_START_TEST (test_encode_silence)
   fail_if (outbuffer == NULL);
 
   fail_unless_equals_int (GST_BUFFER_TIMESTAMP (outbuffer), 0);
-  fail_unless_equals_int (GST_BUFFER_OFFSET (outbuffer), 0);
   fail_unless_equals_int (GST_BUFFER_DURATION (outbuffer), 5668934);
-  fail_unless_equals_int (GST_BUFFER_OFFSET_END (outbuffer), 250);
 
   fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), "wvpk", 4) == 0,
       "Failed to encode to valid Wavpack frames");

tests/examples/audiofx/firfilter-example.c

@@ -21,6 +21,10 @@
  * by transforming the frequency response to the filter kernel.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <math.h>
 

tests/examples/audiofx/iirfilter-example.c

@@ -23,6 +23,10 @@
  * of the IIR filter that is used.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <math.h>
 

tests/examples/pulse/pulse.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <gst/gst.h>
 
 #if 0

tests/examples/rtp/server-alsasrc-PCMA.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #include <string.h>
 #include <math.h>
 

tests/icles/test-oss4.c

@@ -17,6 +17,10 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
 #ifdef HAVE_CONFIG_H
 #include <config.h>
 #endif