X-Git-Url: http://vcs.maemo.org/git/?a=blobdiff_plain;f=gst-plugins-base-subtitles0.10%2Fgst-libs%2Fgst%2Faudio%2Fgstbaseaudiosink.c;fp=gst-plugins-base-subtitles0.10%2Fgst-libs%2Fgst%2Faudio%2Fgstbaseaudiosink.c;h=60930926bcce83a88dfdff5555c50a568e32884b;hb=57ba96e291a055f69dbfd4ae9f1ae2390e36986e;hp=0000000000000000000000000000000000000000;hpb=be2c98fb83895d10ac44af7b9a9c3e00ca54bf49;p=mafwsubrenderer

diff --git a/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosink.c
new file mode 100644
index 0000000..6093092
--- /dev/null
+++ b/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -0,0 +1,2015 @@
+/* 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"
+
+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;
+
+  gboolean sync_latency;
+
+  GstClockTime eos_time;
+
+  gboolean do_time_offset;
+  /* number of microseconds we alow timestamps or clock slaving to drift
+   * before resyncing */
+  guint64 drift_tolerance;
+};
+
+/* BaseAudioSink signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+/* 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
+
+/* FIXME, enable pull mode when clock slaving and trick modes are figured out */
+#define DEFAULT_CAN_ACTIVATE_PULL FALSE
+
+/* when timestamps or clock slaving drift for more than 40ms we resync. This is
+ * a reasonable default */
+#define DEFAULT_DRIFT_TOLERANCE   ((40 * GST_MSECOND) / GST_USECOND)
+
+enum
+{
+  PROP_0,
+
+  PROP_BUFFER_TIME,
+  PROP_LATENCY_TIME,
+  PROP_PROVIDE_CLOCK,
+  PROP_SLAVE_METHOD,
+  PROP_CAN_ACTIVATE_PULL,
+  PROP_DRIFT_TOLERANCE,
+
+  PROP_LAST
+};
+
+GType
+gst_base_audio_sink_slave_method_get_type (void)
+{
+  static volatile gsize slave_method_type = 0;
+  static const GEnumValue slave_method[] = {
+    {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE",
+        "resample"},
+    {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "GST_BASE_AUDIO_SINK_SLAVE_SKEW", "skew"},
+    {GST_BASE_AUDIO_SINK_SLAVE_NONE, "GST_BASE_AUDIO_SINK_SLAVE_NONE", "none"},
+    {0, NULL, NULL},
+  };
+
+  if (g_once_init_enter (&slave_method_type)) {
+    GType tmp =
+        g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
+    g_once_init_leave (&slave_method_type, tmp);
+  }
+
+  return (GType) 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 gboolean gst_base_audio_sink_query_pad (GstPad * pad, GstQuery * query);
+
+
+/* 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_base_audio_sink_set_property;
+  gobject_class->get_property = gst_base_audio_sink_get_property;
+  gobject_class->dispose = 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_PARAM_STATIC_STRINGS));
+
+  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_PARAM_STATIC_STRINGS));
+
+  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_PARAM_STATIC_STRINGS));
+
+  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_BASE_AUDIO_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
+      g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
+          "Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstBaseAudioSink:drift-tolerance
+   *
+   * Controls the amount of time in milliseconds that timestamps or clocks are allowed
+   * to drift before resynchronisation happens.
+   *
+   * Since: 0.10.26
+   */
+  g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
+      g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
+          "Tolerance for timestamp and clock drift in microseconds", 1,
+          G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  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);
+  g_type_class_ref (GST_TYPE_RING_BUFFER);
+
+}
+
+static void
+gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
+    GstBaseAudioSinkClass * g_class)
+{
+  GstPluginFeature *feature;
+  GstBaseSink *basesink;
+
+  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->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
+
+  baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
+      (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
+
+  basesink = GST_BASE_SINK_CAST (baseaudiosink);
+  basesink->can_activate_push = TRUE;
+  basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
+
+  gst_base_sink_set_last_buffer_enabled (basesink, FALSE);
+
+  /* install some custom pad_query functions */
+  gst_pad_set_query_function (GST_BASE_SINK_PAD (baseaudiosink),
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad));
+
+  baseaudiosink->priv->do_time_offset = TRUE;
+
+  /* check the factory, pulsesink < 0.10.17 does the timestamp offset itself so
+   * we should not do ourselves */
+  feature =
+      GST_PLUGIN_FEATURE_CAST (GST_ELEMENT_CLASS (g_class)->elementfactory);
+  GST_DEBUG ("created from factory %p", feature);
+
+  /* HACK for old pulsesink that did the time_offset themselves */
+  if (feature) {
+    if (strcmp (gst_plugin_feature_get_name (feature), "pulsesink") == 0) {
+      if (!gst_plugin_feature_check_version (feature, 0, 10, 17)) {
+        /* we're dealing with an old pulsesink, we need to disable time corection */
+        GST_DEBUG ("disable time offset");
+        baseaudiosink->priv->do_time_offset = FALSE;
+      }
+    }
+  }
+}
+
+static void
+gst_base_audio_sink_dispose (GObject * object)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (object);
+
+  if (sink->provided_clock) {
+    gst_audio_clock_invalidate (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_pad (GstPad * pad, GstQuery * query)
+{
+  gboolean res = FALSE;
+  GstBaseAudioSink *basesink;
+
+  basesink = GST_BASE_AUDIO_SINK (gst_pad_get_parent (pad));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_CONVERT:
+    {
+      GstFormat src_fmt, dest_fmt;
+      gint64 src_val, dest_val;
+
+      GST_LOG_OBJECT (pad, "query convert");
+
+      if (basesink->ringbuffer) {
+        gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
+        res = gst_ring_buffer_convert (basesink->ringbuffer, src_fmt, src_val,
+            dest_fmt, &dest_val);
+        if (res) {
+          gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+        }
+      }
+      break;
+    }
+    default:
+      break;
+  }
+
+  gst_object_unref (basesink);
+
+  return res;
+}
+
+static gboolean
+gst_base_audio_sink_query (GstElement * element, GstQuery * query)
+{
+  gboolean res = FALSE;
+  GstBaseAudioSink *basesink;
+
+  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");
+
+      /* 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;
+
+          GST_OBJECT_LOCK (basesink);
+          if (!basesink->ringbuffer || !basesink->ringbuffer->spec.rate) {
+            GST_OBJECT_UNLOCK (basesink);
+
+            GST_DEBUG_OBJECT (basesink,
+                "we are not yet negotiated, can't report latency yet");
+            res = FALSE;
+            goto done;
+          }
+          spec = &basesink->ringbuffer->spec;
+
+          basesink->priv->us_latency = min_l;
+
+          min_latency =
+              gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
+              GST_SECOND, spec->rate * spec->bytes_per_sample);
+          GST_OBJECT_UNLOCK (basesink);
+
+          /* 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 = min_l;
+          max_latency = max_l;
+        }
+        gst_query_set_latency (query, live, min_latency, max_latency);
+      }
+      break;
+    }
+    case GST_QUERY_CONVERT:
+    {
+      GstFormat src_fmt, dest_fmt;
+      gint64 src_val, dest_val;
+
+      GST_LOG_OBJECT (basesink, "query convert");
+
+      if (basesink->ringbuffer) {
+        gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
+        res = gst_ring_buffer_convert (basesink->ringbuffer, src_fmt, src_val,
+            dest_fmt, &dest_val);
+        if (res) {
+          gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
+        }
+      }
+      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;
+
+  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);
+
+  GST_DEBUG_OBJECT (sink,
+      "processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
+      G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
+      raw, delay, samples, GST_TIME_ARGS (result));
+
+  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
+ */
+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
+ */
+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
+ */
+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
+ */
+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;
+}
+
+
+/**
+ * gst_base_audio_sink_set_drift_tolerance:
+ * @sink: a #GstBaseAudioSink
+ * @drift_tolerance: the new drift tolerance in microseconds
+ *
+ * Controls the sink's drift tolerance.
+ *
+ * Since: 0.10.31
+ */
+void
+gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink,
+    gint64 drift_tolerance)
+{
+  g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+
+  GST_OBJECT_LOCK (sink);
+  sink->priv->drift_tolerance = drift_tolerance;
+  GST_OBJECT_UNLOCK (sink);
+}
+
+/**
+ * gst_base_audio_sink_get_drift_tolerance
+ * @sink: a #GstBaseAudioSink
+ *
+ * Get the current drift tolerance, in microseconds, used by @sink.
+ *
+ * Returns: The current drift tolerance used by @sink.
+ *
+ * Since: 0.10.31
+ */
+gint64
+gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink)
+{
+  gint64 result;
+
+  g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+
+  GST_OBJECT_LOCK (sink);
+  result = sink->priv->drift_tolerance;
+  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;
+    case PROP_CAN_ACTIVATE_PULL:
+      GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
+      break;
+    case PROP_DRIFT_TOLERANCE:
+      gst_base_audio_sink_set_drift_tolerance (sink, g_value_get_int64 (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;
+    case PROP_CAN_ACTIVATE_PULL:
+      g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
+      break;
+    case PROP_DRIFT_TOLERANCE:
+      g_value_set_int64 (value, gst_base_audio_sink_get_drift_tolerance (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;
+  GstClockTime now;
+
+  if (!sink->ringbuffer)
+    return FALSE;
+
+  spec = &sink->ringbuffer->spec;
+
+  GST_DEBUG_OBJECT (sink, "release old ringbuffer");
+
+  /* get current time, updates the last_time */
+  now = gst_clock_get_time (sink->provided_clock);
+
+  GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
+
+  /* release old ringbuffer */
+  gst_ring_buffer_pause (sink->ringbuffer);
+  gst_ring_buffer_activate (sink->ringbuffer, FALSE);
+  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 ringbuffer");
+  if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
+    goto acquire_error;
+
+  if (bsink->pad_mode == GST_ACTIVATE_PUSH) {
+    GST_DEBUG_OBJECT (sink, "activate ringbuffer");
+    gst_ring_buffer_activate (sink->ringbuffer, TRUE);
+  }
+
+  /* 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)
+{
+  if (!sink->ringbuffer)
+    return TRUE;
+  if (!sink->ringbuffer->spec.rate)
+    return TRUE;
+
+  /* if PLAYING is interrupted,
+   * arrange to have clock running when going to PLAYING again */
+  g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 1);
+
+  /* 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->priv->eos_time != -1) {
+    GST_DEBUG_OBJECT (sink,
+        "last sample time %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (sink->priv->eos_time));
+
+    /* 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), sink->priv->eos_time, NULL);
+
+    GST_DEBUG_OBJECT (sink, "drained audio");
+  }
+  g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 0);
+  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;
+      sink->priv->eos_time = -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)
+{
+  /* 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;
+  }
+  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;
+
+  /* FIXME, we can sample and add observations here or use the timeouts on the
+   * clock. No idea which one is better or more stable. The timeout seems more
+   * arbitrary but this one seems more demanding and does not work when there is
+   * no data comming in to the sink. */
+#if 0
+  GstClockTime etime, itime;
+  gdouble r_squared;
+
+  /* sample clocks and figure out clock skew */
+  etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
+  itime = gst_audio_clock_get_time (sink->provided_clock);
+
+  /* add new observation */
+  gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
+#endif
+
+  /* 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);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom);
+
+  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, mdrift, mdrift2;
+  gint driftsamples;
+  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_audio_clock_get_time (sink->provided_clock);
+  itime = gst_audio_clock_adjust (sink->provided_clock, itime);
+
+  GST_DEBUG_OBJECT (sink,
+      "internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
+      " cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
+      GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
+
+  /* make sure we never go below 0 */
+  etime = etime > cexternal ? etime - cexternal : 0;
+  itime = itime > cinternal ? itime - cinternal : 0;
+
+  /* do itime - etime.
+   * positive value means external clock goes slower
+   * negative value means external clock goes faster */
+  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 */
+  mdrift = sink->priv->drift_tolerance * 1000;
+  mdrift2 = mdrift / 2;
+
+  /* adjust playout pointer based on skew */
+  if (sink->priv->avg_skew > mdrift2) {
+    /* 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, mdrift2);
+    cexternal = cexternal > mdrift ? cexternal - mdrift : 0;
+    sink->priv->avg_skew -= mdrift;
+
+    driftsamples = (sink->ringbuffer->spec.rate * mdrift) / GST_SECOND;
+    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 > driftsamples)
+      sink->next_sample = -1;
+
+    GST_DEBUG_OBJECT (sink,
+        "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
+        G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
+
+    gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
+        crate_num, crate_denom);
+  } else if (sink->priv->avg_skew < -mdrift2) {
+    /* 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, -mdrift2);
+    cexternal += mdrift;
+    sink->priv->avg_skew += mdrift;
+
+    driftsamples = (sink->ringbuffer->spec.rate * mdrift) / GST_SECOND;
+    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 > driftsamples)
+      sink->next_sample = -1;
+
+    GST_DEBUG_OBJECT (sink,
+        "last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
+        G_GUINT64_FORMAT, last_align, driftsamples, 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);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom);
+
+  *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);
+  render_stop = clock_convert_external (render_stop, cinternal, cexternal,
+      crate_num, crate_denom);
+
+  *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;
+  }
+}
+
+/* must be called with LOCK */
+static GstFlowReturn
+gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
+{
+  GstClock *clock;
+  GstClockReturn status;
+  GstClockTime time, render_delay;
+  GstFlowReturn ret;
+  GstBaseAudioSink *sink;
+  GstClockTime itime, etime;
+  GstClockTime rate_num, rate_denom;
+  GstClockTimeDiff jitter;
+
+  sink = GST_BASE_AUDIO_SINK (bsink);
+
+  clock = GST_ELEMENT_CLOCK (sink);
+  if (G_UNLIKELY (clock == NULL))
+    goto no_clock;
+
+  /* we provided the global clock, don't need to do anything special */
+  if (clock == sink->provided_clock)
+    goto no_slaving;
+
+  GST_OBJECT_UNLOCK (sink);
+
+  do {
+    GST_DEBUG_OBJECT (sink, "checking preroll");
+
+    ret = gst_base_sink_do_preroll (bsink, obj);
+    if (ret != GST_FLOW_OK)
+      goto flushing;
+
+    GST_OBJECT_LOCK (sink);
+    time = sink->priv->us_latency;
+    GST_OBJECT_UNLOCK (sink);
+
+    /* Renderdelay is added onto our own latency, and needs
+     * to be subtracted as well */
+    render_delay = gst_base_sink_get_render_delay (bsink);
+
+    if (G_LIKELY (time > render_delay))
+      time -= render_delay;
+    else
+      time = 0;
+
+    /* preroll done, we can sync since we are in PLAYING now. */
+    GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
+        GST_TIME_FORMAT, GST_TIME_ARGS (time));
+
+    /* wait for the clock, this can be interrupted because we got shut down or
+     * we PAUSED. */
+    status = gst_base_sink_wait_clock (bsink, time, &jitter);
+
+    GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
+        GST_TIME_ARGS (jitter));
+
+    /* invalid time, no clock or sync disabled, just continue then */
+    if (status == GST_CLOCK_BADTIME)
+      break;
+
+    /* waiting could have been interrupted and we can be flushing now */
+    if (G_UNLIKELY (bsink->flushing))
+      goto flushing;
+
+    /* retry if we got unscheduled, which means we did not reach the timeout
+     * yet. if some other error occures, we continue. */
+  } while (status == GST_CLOCK_UNSCHEDULED);
+
+  GST_OBJECT_LOCK (sink);
+  GST_DEBUG_OBJECT (sink, "latency synced");
+
+  /* when we prerolled in time, we can accurately set the calibration,
+   * our internal clock should exactly have been the latency (== the running
+   * time of the external clock) */
+  etime = GST_ELEMENT_CAST (sink)->base_time + time;
+  itime = gst_audio_clock_get_time (sink->provided_clock);
+  itime = gst_audio_clock_adjust (sink->provided_clock, itime);
+
+  if (status == GST_CLOCK_EARLY) {
+    /* when we prerolled late, we have to take into account the lateness */
+    GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
+    etime += jitter;
+  }
+
+  /* start ringbuffer so we can start slaving right away when we need to */
+  gst_ring_buffer_start (sink->ringbuffer);
+
+  GST_DEBUG_OBJECT (sink,
+      "internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
+
+  /* copy the original calibrated rate but update the internal and external
+   * times. */
+  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 when we are resampling */
+      GST_DEBUG_OBJECT (sink, "Setting clock as master");
+      gst_clock_set_master (sink->provided_clock, clock);
+      break;
+    case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
+    case GST_BASE_AUDIO_SINK_SLAVE_NONE:
+    default:
+      break;
+  }
+
+  sink->priv->avg_skew = -1;
+  sink->next_sample = -1;
+  sink->priv->eos_time = -1;
+
+  return GST_FLOW_OK;
+
+  /* ERRORS */
+no_clock:
+  {
+    GST_DEBUG_OBJECT (sink, "we have no clock");
+    return GST_FLOW_OK;
+  }
+no_slaving:
+  {
+    GST_DEBUG_OBJECT (sink, "we are not slaved");
+    return GST_FLOW_OK;
+  }
+flushing:
+  {
+    GST_DEBUG_OBJECT (sink, "we are flushing");
+    GST_OBJECT_LOCK (sink);
+    return GST_FLOW_WRONG_STATE;
+  }
+}
+
+static gint64
+gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink, GstClockTime sample_offset)
+{
+  GstRingBuffer *ringbuf = sink->ringbuffer;
+  gint64 align;
+  gint64 diff;
+  gint64 maxdrift;
+  gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
+  gint64 samples_done = segdone * ringbuf->samples_per_seg;
+  gint64 headroom = sample_offset - samples_done;
+  gboolean allow_align = TRUE;
+
+  /* now try to align the sample to the previous one, first see how big the
+   * difference is. */
+  if (sample_offset >= sink->next_sample)
+    diff = sample_offset - sink->next_sample;
+  else
+    diff = sink->next_sample - sample_offset;
+
+  /* calculate the max allowed drift in units of samples. By default this is
+   * 20ms and should be anough to compensate for timestamp rounding errors. */
+  maxdrift = (ringbuf->spec.rate * sink->priv->drift_tolerance) / GST_MSECOND;
+
+  /* calc align with previous sample */
+  align = sink->next_sample - sample_offset;
+
+  /* don't align if it means writing behind the read-segment */
+  if (diff > headroom && align < 0)
+    allow_align = FALSE;
+
+  if (G_LIKELY (diff < maxdrift && allow_align)) {
+    GST_DEBUG_OBJECT (sink,
+        "align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
+        G_GINT64_FORMAT, align, maxdrift);
+  } else {
+    /* calculate sample diff in seconds for error message */
+    gint64 diff_s = 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_WARNING_OBJECT (sink,
+        "Unexpected discontinuity in audio timestamps of "
+        "%s%" GST_TIME_FORMAT ", resyncing",
+        sample_offset > sink->next_sample ? "+" : "-",
+        GST_TIME_ARGS (diff_s));
+    align = 0;
+  }
+
+  return align;
+}
+
+static GstFlowReturn
+gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
+{
+  guint64 in_offset;
+  GstClockTime time, stop, render_start, render_stop, sample_offset;
+  GstClockTimeDiff sync_offset, ts_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, render_delay, latency;
+  GstClock *clock;
+  gboolean sync, slaved, align_next;
+  GstFlowReturn ret;
+  GstSegment clip_seg;
+  gint64 time_offset;
+
+  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;
+
+  /* Wait for upstream latency before starting the ringbuffer, we do this so
+   * that we can align the first sample of the ringbuffer to the base_time +
+   * latency. */
+  GST_OBJECT_LOCK (sink);
+  base_time = GST_ELEMENT_CAST (sink)->base_time;
+  if (G_UNLIKELY (sink->priv->sync_latency)) {
+    ret = gst_base_audio_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
+    GST_OBJECT_UNLOCK (sink);
+    if (G_UNLIKELY (ret != GST_FLOW_OK))
+      goto sync_latency_failed;
+    /* only do this once until we are set back to PLAYING */
+    sink->priv->sync_latency = FALSE;
+  } else {
+    GST_OBJECT_UNLOCK (sink);
+  }
+
+  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);
+
+  GST_DEBUG_OBJECT (sink,
+      "time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", 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);
+    /* we don't have a start so we don't know stop either */
+    stop = -1;
+    goto no_sync;
+  }
+
+  /* let's calc stop based on the number of samples in the buffer instead
+   * of trusting the DURATION */
+  stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
+      ringbuf->spec.rate);
+
+  /* prepare the clipping segment. Since we will be subtracting ts-offset and
+   * device-delay later we scale the start and stop with those values so that we
+   * can correctly clip them */
+  clip_seg.format = GST_FORMAT_TIME;
+  clip_seg.start = bsink->segment.start;
+  clip_seg.stop = bsink->segment.stop;
+  clip_seg.duration = -1;
+
+  /* the sync offset is the combination of ts-offset and device-delay */
+  latency = gst_base_sink_get_latency (bsink);
+  ts_offset = gst_base_sink_get_ts_offset (bsink);
+  render_delay = gst_base_sink_get_render_delay (bsink);
+  sync_offset = ts_offset - render_delay + latency;
+
+  GST_DEBUG_OBJECT (sink,
+      "sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
+      ", ts-offset %" G_GINT64_FORMAT, sync_offset,
+      GST_TIME_ARGS (render_delay), ts_offset);
+
+  /* compensate for ts-offset and device-delay when negative we need to
+   * clip. */
+  if (sync_offset < 0) {
+    clip_seg.start += -sync_offset;
+    if (clip_seg.stop != -1)
+      clip_seg.stop += -sync_offset;
+  }
+
+  /* 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 */
+  if (!gst_segment_clip (&clip_seg, 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));
+
+  /* store the time of the last sample, we'll use this to perform sync on the
+   * last sample when draining the buffer */
+  if (bsink->segment.rate >= 0.0) {
+    sink->priv->eos_time = render_stop;
+  } else {
+    sink->priv->eos_time = render_start;
+  }
+
+  /* compensate for ts-offset and delay we know this will not underflow because we
+   * clipped above. */
+  GST_DEBUG_OBJECT (sink,
+      "compensating for sync-offset %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (sync_offset));
+  render_start += sync_offset;
+  render_stop += sync_offset;
+
+  GST_DEBUG_OBJECT (sink, "adding 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;
+
+  GST_DEBUG_OBJECT (sink,
+      "after compensation: 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);
+  } else {
+    /* no slaving needed but we need to adapt to the clock calibration
+     * parameters */
+    gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
+        &render_start, &render_stop);
+  }
+
+  GST_DEBUG_OBJECT (sink,
+      "final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
+
+  /* bring to position in the ringbuffer */
+  if (sink->priv->do_time_offset) {
+    time_offset =
+        GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset;
+    GST_DEBUG_OBJECT (sink,
+        "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
+    if (render_start > time_offset)
+      render_start -= time_offset;
+    else
+      render_start = 0;
+    if (render_stop > time_offset)
+      render_stop -= time_offset;
+    else
+      render_stop = 0;
+  }
+
+  /* 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);
+
+  /* positive playback rate, first sample is render_start, negative rate, first
+   * sample is render_stop. When no rate conversion is active, render exactly
+   * the amount of input samples to avoid aligning to rounding errors. */
+  if (bsink->segment.rate >= 0.0) {
+    sample_offset = render_start;
+    if (bsink->segment.rate == 1.0)
+      render_stop = sample_offset + samples;
+  } else {
+    sample_offset = render_stop;
+    if (bsink->segment.rate == -1.0)
+      render_start = sample_offset + samples;
+  }
+
+  /* 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;
+  }
+
+  /* resync when we don't know what to align the sample with */
+  if (G_UNLIKELY (sink->next_sample == -1)) {
+    GST_DEBUG_OBJECT (sink,
+        "no align possible: no previous sample position known");
+    goto no_align;
+  }
+
+  align = gst_base_audio_sink_get_alignment (sink, sample_offset);
+  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 >= 0.0)
+    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 ((ret = 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;
+
+    /* update the output samples. FIXME, this will just skip them when pausing
+     * during trick mode */
+    if (out_samples > written) {
+      out_samples -= written;
+      accum = 0;
+    } else
+      break;
+
+    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, "preroll got interrupted: %d (%s)", ret,
+        gst_flow_get_name (ret));
+    return ret;
+  }
+sync_latency_failed:
+  {
+    GST_DEBUG_OBJECT (sink, "failed waiting for latency");
+    return ret;
+  }
+}
+
+/**
+ * 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.
+ */
+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 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);
+
+  GST_PAD_STREAM_LOCK (basesink->sinkpad);
+
+  /* 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->segment.last_stop, len,
+      &buf);
+
+  if (ret != GST_FLOW_OK) {
+    if (ret == GST_FLOW_UNEXPECTED)
+      goto eos;
+    else
+      goto error;
+  }
+
+  GST_PAD_PREROLL_LOCK (basesink->sinkpad);
+  if (basesink->flushing)
+    goto flushing;
+
+  /* complete preroll and wait for PLAYING */
+  ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
+  if (ret != GST_FLOW_OK)
+    goto preroll_error;
+
+  if (len != GST_BUFFER_SIZE (buf)) {
+    GST_INFO_OBJECT (basesink,
+        "got different size than requested from sink pad: %u != %u", len,
+        GST_BUFFER_SIZE (buf));
+    len = MIN (GST_BUFFER_SIZE (buf), len);
+  }
+
+  basesink->segment.last_stop += len;
+
+  memcpy (data, GST_BUFFER_DATA (buf), len);
+  GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
+
+  GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
+
+  return;
+
+error:
+  {
+    GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
+        gst_flow_get_name (ret), ret);
+    gst_ring_buffer_pause (rbuf);
+    GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
+    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_base_audio_sink_drain (sink);
+    gst_ring_buffer_pause (rbuf);
+    gst_element_post_message (GST_ELEMENT_CAST (sink),
+        gst_message_new_eos (GST_OBJECT_CAST (sink)));
+    GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
+  }
+flushing:
+  {
+    GST_DEBUG_OBJECT (sink, "we are flushing");
+    gst_ring_buffer_pause (rbuf);
+    GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
+    GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
+    return;
+  }
+preroll_error:
+  {
+    GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
+    gst_ring_buffer_pause (rbuf);
+    GST_PAD_PREROLL_UNLOCK (basesink->sinkpad);
+    GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
+    return;
+  }
+}
+
+static gboolean
+gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
+{
+  gboolean ret;
+  GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
+
+  if (active) {
+    GST_DEBUG_OBJECT (basesink, "activating pull");
+
+    gst_ring_buffer_set_callback (sink->ringbuffer,
+        gst_base_audio_sink_callback, sink);
+
+    ret = gst_ring_buffer_activate (sink->ringbuffer, TRUE);
+  } else {
+    GST_DEBUG_OBJECT (basesink, "deactivating pull");
+    gst_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
+    ret = gst_ring_buffer_activate (sink->ringbuffer, FALSE);
+  }
+
+  return ret;
+}
+
+/* should be called with the LOCK */
+static GstStateChangeReturn
+gst_base_audio_sink_async_play (GstBaseSink * basesink)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (basesink);
+
+  GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
+  sink->priv->sync_latency = TRUE;
+  gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
+  if (basesink->pad_mode == GST_ACTIVATE_PULL) {
+    /* we always start the ringbuffer in pull mode immediatly */
+    gst_ring_buffer_start (sink->ringbuffer);
+  }
+
+  return GST_STATE_CHANGE_SUCCESS;
+}
+
+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) {
+        gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
+        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;
+      sink->priv->eos_time = -1;
+      gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
+      gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
+
+      /* Only post clock-provide messages if this is the clock that
+       * we've created. If the subclass has overriden it the subclass
+       * should post this messages whenever necessary */
+      if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
+          GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
+          (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
+        gst_element_post_message (element,
+            gst_message_new_clock_provide (GST_OBJECT_CAST (element),
+                sink->provided_clock, TRUE));
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+    {
+      gboolean eos;
+
+      GST_OBJECT_LOCK (sink);
+      GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
+      sink->priv->sync_latency = TRUE;
+      eos = GST_BASE_SINK (sink)->eos;
+      GST_OBJECT_UNLOCK (sink);
+
+      gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
+      if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL ||
+          g_atomic_int_get (&sink->abidata.ABI.eos_rendering) || eos) {
+        /* we always start the ringbuffer in pull mode immediatly */
+        /* sync rendering on eos needs running clock,
+         * and others need running clock when finished rendering eos */
+        gst_ring_buffer_start (sink->ringbuffer);
+      }
+      break;
+    }
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* ringbuffer cannot start anymore */
+      gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
+      gst_ring_buffer_pause (sink->ringbuffer);
+
+      GST_OBJECT_LOCK (sink);
+      sink->priv->sync_latency = FALSE;
+      GST_OBJECT_UNLOCK (sink);
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      /* Only post clock-lost messages if this is the clock that
+       * we've created. If the subclass has overriden it the subclass
+       * should post this messages whenever necessary */
+      if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
+          GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
+          (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
+        gst_element_post_message (element,
+            gst_message_new_clock_lost (GST_OBJECT_CAST (element),
+                sink->provided_clock));
+
+      /* 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_activate (sink->ringbuffer, FALSE);
+      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_activate (sink->ringbuffer, FALSE);
+      gst_ring_buffer_release (sink->ringbuffer);
+      gst_ring_buffer_close_device (sink->ringbuffer);
+      GST_OBJECT_LOCK (sink);
+      gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
+      sink->ringbuffer = NULL;
+      GST_OBJECT_UNLOCK (sink);
+      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;
+  }
+}