diff -r 000000000000 -r 0e761a78d257 gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosink.c
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosink.c	Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,1556 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstbaseaudiosink.c: 
+ *
+ * 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:gstbaseaudiosink
+ * @short_description: Base class for audio sinks
+ * @see_also: #GstAudioSink, #GstRingBuffer.
+ *
+ * This is the base class for audio sinks. Subclasses need to implement the
+ * ::create_ringbuffer vmethod. This base class will then take care of
+ * writing samples to the ringbuffer, synchronisation, clipping and flushing.
+ *
+ * Last reviewed on 2006-09-27 (0.10.12)
+ */
+
+#include <string.h>
+
+#include "gstbaseaudiosink.h"
+
+#ifdef __SYMBIAN32__
+#include <glib_global.h>
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug);
+#define GST_CAT_DEFAULT gst_base_audio_sink_debug
+
+#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate))
+
+struct _GstBaseAudioSinkPrivate
+{
+  /* upstream latency */
+  GstClockTime us_latency;
+  /* the clock slaving algorithm in use */
+  GstBaseAudioSinkSlaveMethod slave_method;
+  /* running average of clock skew */
+  GstClockTimeDiff avg_skew;
+  /* the number of samples we aligned last time */
+  gint64 last_align;
+};
+
+/* BaseAudioSink signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+/* we tollerate half a second diff before we start resyncing. This
+ * should be enough to compensate for various rounding errors in the timestamp
+ * and sample offset position. 
+ * This is an emergency resync fallback since buffers marked as DISCONT will
+ * always lock to the correct timestamp immediatly and buffers not marked as
+ * DISCONT are contiguous by definition.
+ */
+#define DIFF_TOLERANCE  2
+
+/* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
+#define DEFAULT_BUFFER_TIME     ((200 * GST_MSECOND) / GST_USECOND)
+#define DEFAULT_LATENCY_TIME    ((10 * GST_MSECOND) / GST_USECOND)
+#define DEFAULT_PROVIDE_CLOCK   TRUE
+#define DEFAULT_SLAVE_METHOD    GST_BASE_AUDIO_SINK_SLAVE_SKEW
+
+enum
+{
+  PROP_0,
+  PROP_BUFFER_TIME,
+  PROP_LATENCY_TIME,
+  PROP_PROVIDE_CLOCK,
+  PROP_SLAVE_METHOD
+};
+
+#define GST_TYPE_SLAVE_METHOD (slave_method_get_type ())
+
+static GType
+slave_method_get_type (void)
+{
+  static GType slave_method_type = 0;
+  static const GEnumValue slave_method[] = {
+    {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "Resampling slaving", "resample"},
+    {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "Skew slaving", "skew"},
+    {GST_BASE_AUDIO_SINK_SLAVE_NONE, "No slaving", "none"},
+    {0, NULL, NULL},
+  };
+
+  if (!slave_method_type) {
+    slave_method_type =
+        g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
+  }
+  return slave_method_type;
+}
+
+
+#define _do_init(bla) \
+    GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
+
+GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_base_audio_sink, GstBaseSink,
+    GST_TYPE_BASE_SINK, _do_init);
+
+static void gst_base_audio_sink_dispose (GObject * object);
+
+static void gst_base_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_base_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
+    basesink);
+static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
+    element, GstStateChange transition);
+static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
+    gboolean active);
+static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
+    query);
+
+static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
+static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
+    GstBaseAudioSink * sink);
+static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
+    guint len, gpointer user_data);
+
+static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink,
+    GstBuffer * buffer);
+static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink,
+    GstBuffer * buffer);
+static gboolean gst_base_audio_sink_event (GstBaseSink * bsink,
+    GstEvent * event);
+static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
+    GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
+static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
+    GstCaps * caps);
+static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
+
+/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
+
+static void
+gst_base_audio_sink_base_init (gpointer g_class)
+{
+}
+
+static void
+gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSinkClass *gstbasesink_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesink_class = (GstBaseSinkClass *) klass;
+
+  g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate));
+
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_set_property);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_property);
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_base_audio_sink_dispose);
+
+  g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
+      g_param_spec_int64 ("buffer-time", "Buffer Time",
+          "Size of audio buffer in microseconds", 1,
+          G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
+      g_param_spec_int64 ("latency-time", "Latency Time",
+          "Audio latency in microseconds", 1,
+          G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
+      g_param_spec_boolean ("provide-clock", "Provide Clock",
+          "Provide a clock to be used as the global pipeline clock",
+          DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
+      g_param_spec_enum ("slave-method", "Slave Method",
+          "Algorithm to use to match the rate of the masterclock",
+          GST_TYPE_SLAVE_METHOD, DEFAULT_SLAVE_METHOD, G_PARAM_READWRITE));
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
+  gstelement_class->provide_clock =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
+  gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
+
+  gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
+  gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
+  gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render);
+  gstbasesink_class->get_times =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
+  gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
+  gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate);
+  gstbasesink_class->async_play =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
+  gstbasesink_class->activate_pull =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull);
+
+  /* ref class from a thread-safe context to work around missing bit of
+   * thread-safety in GObject */
+  g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
+}
+
+static void
+gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
+    GstBaseAudioSinkClass * g_class)
+{
+  baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
+
+  baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
+  baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
+  baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
+  baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD;
+
+  baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
+      (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
+
+  GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
+  /* FIXME, enable pull mode when segments, latency, state changes, negotiation
+   * and clock slaving are figured out */
+  GST_BASE_SINK (baseaudiosink)->can_activate_pull = FALSE;
+}
+
+static void
+gst_base_audio_sink_dispose (GObject * object)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (object);
+
+  if (sink->provided_clock)
+    gst_object_unref (sink->provided_clock);
+  sink->provided_clock = NULL;
+
+  if (sink->ringbuffer) {
+    gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
+    sink->ringbuffer = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static GstClock *
+gst_base_audio_sink_provide_clock (GstElement * elem)
+{
+  GstBaseAudioSink *sink;
+  GstClock *clock;
+
+  sink = GST_BASE_AUDIO_SINK (elem);
+
+  /* we have no ringbuffer (must be NULL state) */
+  if (sink->ringbuffer == NULL)
+    goto wrong_state;
+
+  if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
+    goto wrong_state;
+
+  GST_OBJECT_LOCK (sink);
+  if (!sink->provide_clock)
+    goto clock_disabled;
+
+  clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
+  GST_OBJECT_UNLOCK (sink);
+
+  return clock;
+
+  /* ERRORS */
+wrong_state:
+  {
+    GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
+    return NULL;
+  }
+clock_disabled:
+  {
+    GST_DEBUG_OBJECT (sink, "clock provide disabled");
+    GST_OBJECT_UNLOCK (sink);
+    return NULL;
+  }
+}
+
+static gboolean
+gst_base_audio_sink_query (GstElement * element, GstQuery * query)
+{
+  gboolean res = FALSE;
+
+  GstBaseAudioSink *basesink = GST_BASE_AUDIO_SINK (element);
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:
+    {
+      gboolean live, us_live;
+      GstClockTime min_l, max_l;
+
+      GST_DEBUG_OBJECT (basesink, "latency query");
+
+      if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) {
+        GST_DEBUG_OBJECT (basesink,
+            "we are not yet negotiated, can't report latency yet");
+        res = FALSE;
+        goto done;
+      }
+
+      /* ask parent first, it will do an upstream query for us. */
+      if ((res =
+              gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
+                  &us_live, &min_l, &max_l))) {
+        GstClockTime min_latency, max_latency;
+
+        /* we and upstream are both live, adjust the min_latency */
+        if (live && us_live) {
+          GstRingBufferSpec *spec;
+
+          spec = &basesink->ringbuffer->spec;
+
+          basesink->priv->us_latency = min_l;
+
+          min_latency =
+              gst_util_uint64_scale_int (spec->segtotal * spec->segsize,
+              GST_SECOND, spec->rate * spec->bytes_per_sample);
+
+          /* we cannot go lower than the buffer size and the min peer latency */
+          min_latency = min_latency + min_l;
+          /* the max latency is the max of the peer, we can delay an infinite
+           * amount of time. */
+          max_latency = min_latency + (max_l == -1 ? 0 : max_l);
+
+          GST_DEBUG_OBJECT (basesink,
+              "peer min %" GST_TIME_FORMAT ", our min latency: %"
+              GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
+              GST_TIME_ARGS (min_latency));
+        } else {
+          GST_DEBUG_OBJECT (basesink,
+              "peer or we are not live, don't care about latency");
+          min_latency = 0;
+          max_latency = -1;
+        }
+        gst_query_set_latency (query, live, min_latency, max_latency);
+      }
+      break;
+    }
+    default:
+      res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
+      break;
+  }
+
+done:
+  return res;
+}
+
+
+static GstClockTime
+gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
+{
+  guint64 raw, samples;
+  guint delay;
+  GstClockTime result, us_latency;
+
+  if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
+    return GST_CLOCK_TIME_NONE;
+
+  /* our processed samples are always increasing */
+  raw = samples = gst_ring_buffer_samples_done (sink->ringbuffer);
+
+  /* the number of samples not yet processed, this is still queued in the
+   * device (not played for playback). */
+  delay = gst_ring_buffer_delay (sink->ringbuffer);
+
+  if (G_LIKELY (samples >= delay))
+    samples -= delay;
+  else
+    samples = 0;
+
+  result = gst_util_uint64_scale_int (samples, GST_SECOND,
+      sink->ringbuffer->spec.rate);
+
+  /* latency before starting the clock */
+  us_latency = sink->priv->us_latency;
+
+  result += us_latency;
+
+  GST_DEBUG_OBJECT (sink,
+      "processed samples: raw %llu, delay %u, real %llu, time %"
+      GST_TIME_FORMAT ", upstream latency %" GST_TIME_FORMAT, raw, delay,
+      samples, GST_TIME_ARGS (result), GST_TIME_ARGS (us_latency));
+
+  return result;
+}
+
+/**
+ * gst_base_audio_sink_set_provide_clock:
+ * @sink: a #GstBaseAudioSink
+ * @provide: new state
+ *
+ * Controls whether @sink will provide a clock or not. If @provide is %TRUE, 
+ * gst_element_provide_clock() will return a clock that reflects the datarate
+ * of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
+ *
+ * Since: 0.10.16
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink,
+    gboolean provide)
+{
+  g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+
+  GST_OBJECT_LOCK (sink);
+  sink->provide_clock = provide;
+  GST_OBJECT_UNLOCK (sink);
+}
+
+/**
+ * gst_base_audio_sink_get_provide_clock:
+ * @sink: a #GstBaseAudioSink
+ *
+ * Queries whether @sink will provide a clock or not. See also
+ * gst_base_audio_sink_set_provide_clock.
+ *
+ * Returns: %TRUE if @sink will provide a clock.
+ *
+ * Since: 0.10.16
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+gboolean
+gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE);
+
+  GST_OBJECT_LOCK (sink);
+  result = sink->provide_clock;
+  GST_OBJECT_UNLOCK (sink);
+
+  return result;
+}
+
+/**
+ * gst_base_audio_sink_set_slave_method:
+ * @sink: a #GstBaseAudioSink
+ * @method: the new slave method
+ *
+ * Controls how clock slaving will be performed in @sink. 
+ *
+ * Since: 0.10.16
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+void
+gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink,
+    GstBaseAudioSinkSlaveMethod method)
+{
+  g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+
+  GST_OBJECT_LOCK (sink);
+  sink->priv->slave_method = method;
+  GST_OBJECT_UNLOCK (sink);
+}
+
+/**
+ * gst_base_audio_sink_get_slave_method:
+ * @sink: a #GstBaseAudioSink
+ *
+ * Get the current slave method used by @sink.
+ *
+ * Returns: The current slave method used by @sink.
+ *
+ * Since: 0.10.16
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstBaseAudioSinkSlaveMethod
+gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink)
+{
+  GstBaseAudioSinkSlaveMethod result;
+
+  g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+
+  GST_OBJECT_LOCK (sink);
+  result = sink->priv->slave_method;
+  GST_OBJECT_UNLOCK (sink);
+
+  return result;
+}
+
+static void
+gst_base_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_BUFFER_TIME:
+      sink->buffer_time = g_value_get_int64 (value);
+      break;
+    case PROP_LATENCY_TIME:
+      sink->latency_time = g_value_get_int64 (value);
+      break;
+    case PROP_PROVIDE_CLOCK:
+      gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value));
+      break;
+    case PROP_SLAVE_METHOD:
+      gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_base_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_BUFFER_TIME:
+      g_value_set_int64 (value, sink->buffer_time);
+      break;
+    case PROP_LATENCY_TIME:
+      g_value_set_int64 (value, sink->latency_time);
+      break;
+    case PROP_PROVIDE_CLOCK:
+      g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink));
+      break;
+    case PROP_SLAVE_METHOD:
+      g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
+{
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+  GstRingBufferSpec *spec;
+
+  if (!sink->ringbuffer)
+    return FALSE;
+
+  spec = &sink->ringbuffer->spec;
+
+  GST_DEBUG_OBJECT (sink, "release old ringbuffer");
+
+  /* release old ringbuffer */
+  gst_ring_buffer_release (sink->ringbuffer);
+
+  GST_DEBUG_OBJECT (sink, "parse caps");
+
+  spec->buffer_time = sink->buffer_time;
+  spec->latency_time = sink->latency_time;
+
+  /* parse new caps */
+  if (!gst_ring_buffer_parse_caps (spec, caps))
+    goto parse_error;
+
+  gst_ring_buffer_debug_spec_buff (spec);
+
+  GST_DEBUG_OBJECT (sink, "acquire new ringbuffer");
+
+  if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
+    goto acquire_error;
+
+  /* calculate actual latency and buffer times. 
+   * FIXME: In 0.11, store the latency_time internally in ns */
+  spec->latency_time = gst_util_uint64_scale (spec->segsize,
+      (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
+
+  spec->buffer_time = spec->segtotal * spec->latency_time;
+
+  gst_ring_buffer_debug_spec_buff (spec);
+
+  return TRUE;
+
+  /* ERRORS */
+parse_error:
+  {
+    GST_DEBUG_OBJECT (sink, "could not parse caps");
+    GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
+        (NULL), ("cannot parse audio format."));
+    return FALSE;
+  }
+acquire_error:
+  {
+    GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
+    return FALSE;
+  }
+}
+
+static void
+gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
+{
+  GstStructure *s;
+  gint width, depth;
+
+  s = gst_caps_get_structure (caps, 0);
+
+  /* fields for all formats */
+  gst_structure_fixate_field_nearest_int (s, "rate", 44100);
+  gst_structure_fixate_field_nearest_int (s, "channels", 2);
+  gst_structure_fixate_field_nearest_int (s, "width", 16);
+
+  /* fields for int */
+  if (gst_structure_has_field (s, "depth")) {
+    gst_structure_get_int (s, "width", &width);
+    /* round width to nearest multiple of 8 for the depth */
+    depth = GST_ROUND_UP_8 (width);
+    gst_structure_fixate_field_nearest_int (s, "depth", depth);
+  }
+  if (gst_structure_has_field (s, "signed"))
+    gst_structure_fixate_field_boolean (s, "signed", TRUE);
+  if (gst_structure_has_field (s, "endianness"))
+    gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
+}
+
+static void
+gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
+    GstClockTime * start, GstClockTime * end)
+{
+  /* our clock sync is a bit too much for the base class to handle so
+   * we implement it ourselves. */
+  *start = GST_CLOCK_TIME_NONE;
+  *end = GST_CLOCK_TIME_NONE;
+}
+
+/* This waits for the drain to happen and can be canceled */
+static gboolean
+gst_base_audio_sink_drain (GstBaseAudioSink * sink)
+{
+  GstClockTime base_time;
+
+  if (!sink->ringbuffer)
+    return TRUE;
+  if (!sink->ringbuffer->spec.rate)
+    return TRUE;
+
+  /* need to start playback before we can drain, but only when
+   * we have successfully negotiated a format and thus acquired the
+   * ringbuffer. */
+  if (gst_ring_buffer_is_acquired (sink->ringbuffer))
+    gst_ring_buffer_start (sink->ringbuffer);
+
+  if (sink->next_sample != -1) {
+    GstClockTime time;
+
+    /* convert next expected sample to time */
+    time =
+        gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
+        sink->ringbuffer->spec.rate);
+
+    GST_DEBUG_OBJECT (sink,
+        "last sample %" G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
+        sink->next_sample, GST_TIME_ARGS (time));
+
+    /* our time already includes the base_time, _wait_eos() wants a running_time
+     * so we have to subtract the base_time again here. FIXME, store an
+     * unadjusted EOS time so that we don't have to do this. */
+    GST_OBJECT_LOCK (sink);
+    base_time = GST_ELEMENT_CAST (sink)->base_time;
+    GST_OBJECT_UNLOCK (sink);
+
+    if (time > base_time)
+      time -= base_time;
+    else
+      time = 0;
+
+    /* wait for the EOS time to be reached, this is the time when the last
+     * sample is played. */
+    gst_base_sink_wait_eos (GST_BASE_SINK (sink), time, NULL);
+
+    sink->next_sample = -1;
+  }
+  return TRUE;
+}
+
+static gboolean
+gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
+{
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_FLUSH_START:
+      if (sink->ringbuffer)
+        gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
+      break;
+    case GST_EVENT_FLUSH_STOP:
+      /* always resync on sample after a flush */
+      sink->priv->avg_skew = -1;
+      sink->next_sample = -1;
+      if (sink->ringbuffer)
+        gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
+      break;
+    case GST_EVENT_EOS:
+      /* now wait till we played everything */
+      gst_base_audio_sink_drain (sink);
+      break;
+    case GST_EVENT_NEWSEGMENT:
+    {
+      gdouble rate;
+
+      /* we only need the rate */
+      gst_event_parse_new_segment_full (event, NULL, &rate, NULL, NULL,
+          NULL, NULL, NULL);
+
+      GST_DEBUG_OBJECT (sink, "new segment rate of %f", rate);
+      break;
+    }
+    default:
+      break;
+  }
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
+{
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+
+  if (!gst_ring_buffer_is_acquired (sink->ringbuffer))
+    goto wrong_state;
+
+  /* we don't really do anything when prerolling. We could make a
+   * property to play this buffer to have some sort of scrubbing
+   * support. */
+  return GST_FLOW_OK;
+
+wrong_state:
+  {
+    GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
+    GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+}
+
+static guint64
+gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
+{
+  guint64 sample;
+  gint writeseg, segdone, sps;
+  gint diff;
+
+  /* assume we can append to the previous sample */
+  sample = sink->next_sample;
+  /* no previous sample, try to insert at position 0 */
+  if (sample == -1)
+    sample = 0;
+
+  sps = sink->ringbuffer->samples_per_seg;
+
+  /* figure out the segment and the offset inside the segment where
+   * the sample should be written. */
+  writeseg = sample / sps;
+
+  /* get the currently processed segment */
+  segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
+      - sink->ringbuffer->segbase;
+
+  /* see how far away it is from the write segment */
+  diff = writeseg - segdone;
+  if (diff < 0) {
+    /* sample would be dropped, position to next playable position */
+    sample = (segdone + 1) * sps;
+  }
+
+  return sample;
+}
+
+static GstClockTime
+clock_convert_external (GstClockTime external, GstClockTime cinternal,
+    GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom,
+    GstClockTime us_latency)
+{
+  /* adjust for rate and speed */
+  if (external >= cexternal) {
+    external =
+        gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
+    external += cinternal;
+  } else {
+    external = gst_util_uint64_scale (cexternal - external,
+        crate_denom, crate_num);
+    if (cinternal > external)
+      external = cinternal - external;
+    else
+      external = 0;
+  }
+  /* adjust for offset when slaving started */
+  if (external > us_latency)
+    external -= us_latency;
+  else
+    external = 0;
+
+  return external;
+}
+
+/* algorithm to calculate sample positions that will result in resampling to
+ * match the clock rate of the master */
+static void
+gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
+    GstClockTime render_start, GstClockTime render_stop,
+    GstClockTime * srender_start, GstClockTime * srender_stop)
+{
+  GstClockTime cinternal, cexternal;
+  GstClockTime crate_num, crate_denom;
+
+  /* get calibration parameters to compensate for speed and offset differences
+   * when we are slaved */
+  gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
+      &crate_num, &crate_denom);
+
+  GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
+      GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
+      GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
+      crate_denom, gst_guint64_to_gdouble (crate_num) /
+      gst_guint64_to_gdouble (crate_denom));
+
+  if (crate_num == 0)
+    crate_denom = crate_num = 1;
+
+  /* bring external time to internal time */
+  render_start = clock_convert_external (render_start, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+
+  GST_DEBUG_OBJECT (sink,
+      "after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
+
+  *srender_start = render_start;
+  *srender_stop = render_stop;
+}
+
+/* algorithm to calculate sample positions that will result in changing the
+ * playout pointer to match the clock rate of the master */
+static void
+gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
+    GstClockTime render_start, GstClockTime render_stop,
+    GstClockTime * srender_start, GstClockTime * srender_stop)
+{
+  GstClockTime cinternal, cexternal, crate_num, crate_denom;
+  GstClockTime etime, itime;
+  GstClockTimeDiff skew, segtime, segtime2;
+  gint segsamples;
+  gint64 last_align;
+
+  /* get calibration parameters to compensate for offsets */
+  gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
+      &crate_num, &crate_denom);
+
+  /* sample clocks and figure out clock skew */
+  etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
+  itime = gst_clock_get_internal_time (sink->provided_clock);
+
+  etime -= cexternal;
+  itime -= cinternal;
+
+  skew = GST_CLOCK_DIFF (etime, itime);
+  if (sink->priv->avg_skew == -1) {
+    /* first observation */
+    sink->priv->avg_skew = skew;
+  } else {
+    /* next observations use a moving average */
+    sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
+  }
+
+  GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
+      GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
+      GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
+
+  /* the max drift we allow is the length of a segment */
+  segtime = sink->ringbuffer->spec.latency_time * 1000;
+  segtime2 = segtime / 2;
+
+  /* adjust playout pointer based on skew */
+  if (sink->priv->avg_skew > segtime2) {
+    /* master is running slower, move internal time forward */
+    GST_WARNING_OBJECT (sink,
+        "correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
+        sink->priv->avg_skew, segtime2);
+    cinternal += segtime;
+    sink->priv->avg_skew -= segtime;
+
+    segsamples =
+        sink->ringbuffer->spec.segsize /
+        sink->ringbuffer->spec.bytes_per_sample;
+    last_align = sink->priv->last_align;
+
+    /* if we were aligning in the wrong direction or we aligned more than what we
+     * will correct, resync */
+    if (last_align < 0 || last_align > segsamples)
+      sink->next_sample = -1;
+
+    GST_DEBUG_OBJECT (sink,
+        "last_align %" G_GINT64_FORMAT " segsamples %u, next %"
+        G_GUINT64_FORMAT, last_align, segsamples, sink->next_sample);
+
+    gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
+        crate_num, crate_denom);
+  } else if (sink->priv->avg_skew < -segtime2) {
+    /* master is running faster, move external time forwards */
+    GST_WARNING_OBJECT (sink,
+        "correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
+        sink->priv->avg_skew, -segtime2);
+    cexternal += segtime;
+    sink->priv->avg_skew += segtime;
+
+    segsamples =
+        sink->ringbuffer->spec.segsize /
+        sink->ringbuffer->spec.bytes_per_sample;
+    last_align = sink->priv->last_align;
+
+    /* if we were aligning in the wrong direction or we aligned more than what we
+     * will correct, resync */
+    if (last_align > 0 || -last_align > segsamples)
+      sink->next_sample = -1;
+
+    GST_DEBUG_OBJECT (sink,
+        "last_align %" G_GINT64_FORMAT " segsamples %u, next %"
+        G_GUINT64_FORMAT, last_align, segsamples, sink->next_sample);
+
+    gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
+        crate_num, crate_denom);
+  }
+
+  /* convert, ignoring speed */
+  render_start = clock_convert_external (render_start, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+
+  *srender_start = render_start;
+  *srender_stop = render_stop;
+}
+
+/* apply the clock offset but do no slaving otherwise */
+static void
+gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink,
+    GstClockTime render_start, GstClockTime render_stop,
+    GstClockTime * srender_start, GstClockTime * srender_stop)
+{
+  GstClockTime cinternal, cexternal, crate_num, crate_denom;
+
+  /* get calibration parameters to compensate for offsets */
+  gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
+      &crate_num, &crate_denom);
+
+  /* convert, ignoring speed */
+  render_start = clock_convert_external (render_start, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom, sink->priv->us_latency);
+
+  *srender_start = render_start;
+  *srender_stop = render_stop;
+}
+
+/* converts render_start and render_stop to their slaved values */
+static void
+gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink,
+    GstClockTime render_start, GstClockTime render_stop,
+    GstClockTime * srender_start, GstClockTime * srender_stop)
+{
+  switch (sink->priv->slave_method) {
+    case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
+      gst_base_audio_sink_resample_slaving (sink, render_start, render_stop,
+          srender_start, srender_stop);
+      break;
+    case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
+      gst_base_audio_sink_skew_slaving (sink, render_start, render_stop,
+          srender_start, srender_stop);
+      break;
+    case GST_BASE_AUDIO_SINK_SLAVE_NONE:
+      gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
+          srender_start, srender_stop);
+      break;
+    default:
+      g_warning ("unknown slaving method %d", sink->priv->slave_method);
+      break;
+  }
+}
+
+static GstFlowReturn
+gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
+{
+  guint64 in_offset;
+  GstClockTime time, stop, render_start, render_stop, sample_offset;
+  GstBaseAudioSink *sink;
+  GstRingBuffer *ringbuf;
+  gint64 diff, align, ctime, cstop;
+  guint8 *data;
+  guint size;
+  guint samples, written;
+  gint bps;
+  gint accum;
+  gint out_samples;
+  GstClockTime base_time = GST_CLOCK_TIME_NONE, latency;
+  GstClock *clock;
+  gboolean sync, slaved, align_next;
+
+  sink = GST_BASE_AUDIO_SINK (bsink);
+
+  ringbuf = sink->ringbuffer;
+
+  /* can't do anything when we don't have the device */
+  if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuf)))
+    goto wrong_state;
+
+  bps = ringbuf->spec.bytes_per_sample;
+
+  size = GST_BUFFER_SIZE (buf);
+  if (G_UNLIKELY (size % bps) != 0)
+    goto wrong_size;
+
+  samples = size / bps;
+  out_samples = samples;
+
+  in_offset = GST_BUFFER_OFFSET (buf);
+  time = GST_BUFFER_TIMESTAMP (buf);
+  stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
+      ringbuf->spec.rate);
+
+  GST_DEBUG_OBJECT (sink,
+      "time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
+      ", samples %u", GST_TIME_ARGS (time), in_offset,
+      GST_TIME_ARGS (bsink->segment.start), samples);
+
+  data = GST_BUFFER_DATA (buf);
+
+  /* if not valid timestamp or we can't clip or sync, try to play
+   * sample ASAP */
+  if (!GST_CLOCK_TIME_IS_VALID (time)) {
+    render_start = gst_base_audio_sink_get_offset (sink);
+    render_stop = render_start + samples;
+    GST_DEBUG_OBJECT (sink,
+        "Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT,
+        GST_BUFFER_SIZE (buf), render_start);
+    goto no_sync;
+  }
+
+  /* samples should be rendered based on their timestamp. All samples
+   * arriving before the segment.start or after segment.stop are to be 
+   * thrown away. All samples should also be clipped to the segment 
+   * boundaries */
+  /* let's calc stop based on the number of samples in the buffer instead
+   * of trusting the DURATION */
+  if (!gst_segment_clip (&bsink->segment, GST_FORMAT_TIME, time, stop, &ctime,
+          &cstop))
+    goto out_of_segment;
+
+  /* see if some clipping happened */
+  diff = ctime - time;
+  if (diff > 0) {
+    /* bring clipped time to samples */
+    diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
+    GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
+        G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
+    samples -= diff;
+    data += diff * bps;
+    time = ctime;
+  }
+  diff = stop - cstop;
+  if (diff > 0) {
+    /* bring clipped time to samples */
+    diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
+    GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
+        G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
+    samples -= diff;
+    stop = cstop;
+  }
+
+  /* figure out how to sync */
+  if ((clock = GST_ELEMENT_CLOCK (bsink)))
+    sync = bsink->sync;
+  else
+    sync = FALSE;
+
+  if (!sync) {
+    /* no sync needed, play sample ASAP */
+    render_start = gst_base_audio_sink_get_offset (sink);
+    render_stop = render_start + samples;
+    GST_DEBUG_OBJECT (sink,
+        "no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
+    goto no_sync;
+  }
+
+  /* bring buffer start and stop times to running time */
+  render_start =
+      gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
+  render_stop =
+      gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
+
+  GST_DEBUG_OBJECT (sink,
+      "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
+
+  base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
+
+  GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (base_time));
+
+  /* add base time to sync against the clock */
+  render_start += base_time;
+  render_stop += base_time;
+
+  /* compensate for latency */
+  latency = gst_base_sink_get_latency (bsink);
+  GST_DEBUG_OBJECT (sink,
+      "compensating for latency %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
+
+  /* add latency to get the timestamp to sync against the pipeline clock */
+  render_start += latency;
+  render_stop += latency;
+
+  GST_DEBUG_OBJECT (sink,
+      "after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
+
+  if ((slaved = clock != sink->provided_clock)) {
+    /* handle clock slaving */
+    gst_base_audio_sink_handle_slaving (sink, render_start, render_stop,
+        &render_start, &render_stop);
+  }
+
+  /* and bring the time to the rate corrected offset in the buffer */
+  render_start = gst_util_uint64_scale_int (render_start,
+      ringbuf->spec.rate, GST_SECOND);
+  render_stop = gst_util_uint64_scale_int (render_stop,
+      ringbuf->spec.rate, GST_SECOND);
+
+  /* always resync after a discont */
+  if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
+    GST_DEBUG_OBJECT (sink, "resync after discont");
+    goto no_align;
+  }
+
+  if (G_UNLIKELY (sink->next_sample == -1)) {
+    GST_DEBUG_OBJECT (sink,
+        "no align possible: no previous sample position known");
+    goto no_align;
+  }
+
+  /* positive playback rate, first sample is render_start, negative rate, first
+   * sample is render_stop */
+  if (bsink->segment.rate >= 1.0)
+    sample_offset = render_start;
+  else
+    sample_offset = render_stop;
+
+  /* now try to align the sample to the previous one */
+  if (sample_offset >= sink->next_sample)
+    diff = sample_offset - sink->next_sample;
+  else
+    diff = sink->next_sample - sample_offset;
+
+  /* we tollerate half a second diff before we start resyncing. This
+   * should be enough to compensate for various rounding errors in the timestamp
+   * and sample offset position. We always resync if we got a discont anyway and
+   * non-discont should be aligned by definition. */
+  if (G_LIKELY (diff < ringbuf->spec.rate / DIFF_TOLERANCE)) {
+    /* calc align with previous sample */
+    align = sink->next_sample - sample_offset;
+    GST_DEBUG_OBJECT (sink,
+        "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %d", align,
+        ringbuf->spec.rate / DIFF_TOLERANCE);
+  } else {
+    /* bring sample diff to seconds for error message */
+    diff = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate);
+    /* timestamps drifted apart from previous samples too much, we need to
+     * resync. We log this as an element warning. */
+    GST_ELEMENT_WARNING (sink, CORE, CLOCK,
+        ("Compensating for audio synchronisation problems"),
+        ("Unexpected discontinuity in audio timestamps of more "
+            "than half a second (%" GST_TIME_FORMAT "), resyncing",
+            GST_TIME_ARGS (diff)));
+    align = 0;
+  }
+  sink->priv->last_align = align;
+
+  /* apply alignment */
+  render_start += align;
+
+  /* only align stop if we are not slaved to resample */
+  if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) {
+    GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
+    goto no_align;
+  }
+  render_stop += align;
+
+no_align:
+  /* number of target samples is difference between start and stop */
+  out_samples = render_stop - render_start;
+
+no_sync:
+  /* we render the first or last sample first, depending on the rate */
+  if (bsink->segment.rate >= 1.0)
+    sample_offset = render_start;
+  else
+    sample_offset = render_stop;
+
+  GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
+      sample_offset, samples, out_samples);
+
+  /* we need to accumulate over different runs for when we get interrupted */
+  accum = 0;
+  align_next = TRUE;
+  do {
+    written =
+        gst_ring_buffer_commit_full (ringbuf, &sample_offset, data, samples,
+        out_samples, &accum);
+
+    GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
+    /* if we wrote all, we're done */
+    if (written == samples)
+      break;
+
+    /* else something interrupted us and we wait for preroll. */
+    if (gst_base_sink_wait_preroll (bsink) != GST_FLOW_OK)
+      goto stopping;
+
+    /* if we got interrupted, we cannot assume that the next sample should
+     * be aligned to this one */
+    align_next = FALSE;
+
+    samples -= written;
+    data += written * bps;
+  } while (TRUE);
+
+  if (align_next)
+    sink->next_sample = sample_offset;
+  else
+    sink->next_sample = -1;
+
+  GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
+      sink->next_sample);
+
+  if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
+    GST_DEBUG_OBJECT (sink,
+        "start playback because we are at the end of segment");
+    gst_ring_buffer_start (ringbuf);
+  }
+
+  return GST_FLOW_OK;
+
+  /* SPECIAL cases */
+out_of_segment:
+  {
+    GST_DEBUG_OBJECT (sink,
+        "dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
+        GST_TIME_FORMAT, GST_TIME_ARGS (time),
+        GST_TIME_ARGS (bsink->segment.start));
+    return GST_FLOW_OK;
+  }
+  /* ERRORS */
+wrong_state:
+  {
+    GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
+    GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+wrong_size:
+  {
+    GST_DEBUG_OBJECT (sink, "wrong size");
+    GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
+        (NULL), ("sink received buffer of wrong size."));
+    return GST_FLOW_ERROR;
+  }
+stopping:
+  {
+    GST_DEBUG_OBJECT (sink, "ringbuffer is stopping");
+    return GST_FLOW_WRONG_STATE;
+  }
+}
+
+/**
+ * gst_base_audio_sink_create_ringbuffer:
+ * @sink: a #GstBaseAudioSink.
+ *
+ * Create and return the #GstRingBuffer for @sink. This function will call the
+ * ::create_ringbuffer vmethod and will set @sink as the parent of the returned
+ * buffer (see gst_object_set_parent()).
+ *
+ * Returns: The new ringbuffer of @sink.
+ */
+#ifdef __SYMBIAN32__
+EXPORT_C
+#endif
+
+GstRingBuffer *
+gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
+{
+  GstBaseAudioSinkClass *bclass;
+  GstRingBuffer *buffer = NULL;
+
+  bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
+  if (bclass->create_ringbuffer)
+    buffer = bclass->create_ringbuffer (sink);
+
+  if (buffer)
+    gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
+
+  return buffer;
+}
+
+static gboolean
+gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
+{
+  gboolean ret;
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
+
+  if (active) {
+    gst_ring_buffer_set_callback (sink->ringbuffer,
+        gst_base_audio_sink_callback, sink);
+    ret = gst_ring_buffer_start (sink->ringbuffer);
+  } else {
+    gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
+    /* stop thread */
+    ret = gst_ring_buffer_release (sink->ringbuffer);
+  }
+
+  return ret;
+}
+
+static void
+gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
+    gpointer user_data)
+{
+  GstBaseSink *basesink;
+  GstBaseAudioSink *sink;
+  GstBuffer *buf;
+  GstFlowReturn ret;
+
+  basesink = GST_BASE_SINK (user_data);
+  sink = GST_BASE_AUDIO_SINK (user_data);
+
+  /* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
+     will copy twice, once into data, once into DMA */
+  GST_LOG_OBJECT (basesink, "pulling %d bytes offset %" G_GUINT64_FORMAT
+      " to fill audio buffer", len, basesink->offset);
+  ret = gst_pad_pull_range (basesink->sinkpad, basesink->offset, len, &buf);
+
+  if (ret != GST_FLOW_OK) {
+    if (ret == GST_FLOW_UNEXPECTED)
+      goto eos;
+    else
+      goto error;
+  }
+
+  if (len != GST_BUFFER_SIZE (buf)) {
+    GST_INFO_OBJECT (basesink, "short read pulling from sink pad: %d<%d",
+        len, GST_BUFFER_SIZE (buf));
+    len = MIN (GST_BUFFER_SIZE (buf), len);
+  }
+
+  basesink->offset += len;
+
+  memcpy (data, GST_BUFFER_DATA (buf), len);
+
+  return;
+
+error:
+  {
+    GST_WARNING_OBJECT (basesink, "Got flow error but can't return it: %d",
+        ret);
+    return;
+  }
+eos:
+  {
+    /* FIXME: this is not quite correct; we'll be called endlessly until
+     * the sink gets shut down; maybe we should set a flag somewhere, or
+     * set segment.stop and segment.duration to the last sample or so */
+    GST_DEBUG_OBJECT (sink, "EOS");
+    gst_element_post_message (GST_ELEMENT_CAST (sink),
+        gst_message_new_eos (GST_OBJECT_CAST (sink)));
+    gst_base_audio_sink_drain (sink);
+  }
+}
+
+/* should be called with the LOCK */
+static GstStateChangeReturn
+gst_base_audio_sink_async_play (GstBaseSink * basesink)
+{
+  GstClock *clock;
+  GstBaseAudioSink *sink;
+  GstClockTime itime, etime;
+  GstClockTime rate_num, rate_denom;
+
+  sink = GST_BASE_AUDIO_SINK (basesink);
+
+  GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
+  gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
+
+  clock = GST_ELEMENT_CLOCK (sink);
+  if (clock == NULL)
+    goto done;
+
+  /* we provided the global clock, don't need to do anything special */
+  if (clock == sink->provided_clock)
+    goto done;
+
+  /* if we are slaved to a clock, we need to set the initial
+   * calibration */
+  /* get external and internal time to set as calibration params */
+  etime = gst_clock_get_time (clock);
+  itime = gst_clock_get_internal_time (sink->provided_clock);
+
+  sink->priv->avg_skew = -1;
+  sink->next_sample = -1;
+
+  GST_DEBUG_OBJECT (sink,
+      "internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
+
+  gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
+      &rate_denom);
+  gst_clock_set_calibration (sink->provided_clock, itime, etime,
+      rate_num, rate_denom);
+
+  switch (sink->priv->slave_method) {
+    case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
+      /* only set as master if we need to resample */
+      GST_DEBUG_OBJECT (sink, "Setting clock as master");
+      gst_clock_set_master (sink->provided_clock, clock);
+      break;
+    default:
+      break;
+  }
+
+  /* start ringbuffer so we can start slaving right away when we need to */
+  gst_ring_buffer_start (sink->ringbuffer);
+
+done:
+  return GST_STATE_CHANGE_SUCCESS;
+}
+
+static GstStateChangeReturn
+gst_base_audio_sink_do_play (GstBaseAudioSink * sink)
+{
+  GstStateChangeReturn ret;
+
+  GST_OBJECT_LOCK (sink);
+  ret = gst_base_audio_sink_async_play (GST_BASE_SINK_CAST (sink));
+  GST_OBJECT_UNLOCK (sink);
+
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_base_audio_sink_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (sink->ringbuffer == NULL) {
+        sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
+      }
+      if (!gst_ring_buffer_open_device (sink->ringbuffer))
+        goto open_failed;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      sink->next_sample = -1;
+      sink->priv->last_align = -1;
+      gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
+      gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      gst_base_audio_sink_do_play (sink);
+      break;
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* need to take the lock so we don't interfere with an
+       * async play */
+      GST_OBJECT_LOCK (sink);
+      /* ringbuffer cannot start anymore */
+      gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
+      gst_ring_buffer_pause (sink->ringbuffer);
+      GST_OBJECT_UNLOCK (sink);
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      /* make sure we unblock before calling the parent state change
+       * so it can grab the STREAM_LOCK */
+      gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* stop slaving ourselves to the master, if any */
+      gst_clock_set_master (sink->provided_clock, NULL);
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_ring_buffer_release (sink->ringbuffer);
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      /* we release again here because the aqcuire happens when setting the
+       * caps, which happens before we commit the state to PAUSED and thus the
+       * PAUSED->READY state change (see above, where we release the ringbuffer)
+       * might not be called when we get here. */
+      gst_ring_buffer_release (sink->ringbuffer);
+      gst_ring_buffer_close_device (sink->ringbuffer);
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+
+  /* ERRORS */
+open_failed:
+  {
+    /* subclass must post a meaningfull error message */
+    GST_DEBUG_OBJECT (sink, "open failed");
+    return GST_STATE_CHANGE_FAILURE;
+  }
+}