1 files changed, 235 insertions, 0 deletions
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
new file mode 100644
index 00000000..d27b4a22
--- /dev/null
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -0,0 +1,235 @@
+/* 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.
+ */
+
+#include "gstbaseaudiosink.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_baseaudiosink_debug);
+#define GST_CAT_DEFAULT gst_baseaudiosink_debug
+
+/* BaseAudioSink signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0,
+};
+
+#define _do_init(bla) \
+    GST_DEBUG_CATEGORY_INIT (gst_baseaudiosink_debug, "baseaudiosink", 0, "baseaudiosink element");
+
+GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_baseaudiosink, GstBaseSink,
+    GST_TYPE_BASESINK, _do_init);
+
+static void gst_baseaudiosink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_baseaudiosink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstElementStateReturn gst_baseaudiosink_change_state (GstElement *
+    element);
+
+static GstFlowReturn gst_baseaudiosink_preroll (GstBaseSink * bsink,
+    GstBuffer * buffer);
+static GstFlowReturn gst_baseaudiosink_render (GstBaseSink * bsink,
+    GstBuffer * buffer);
+static void gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event);
+static void gst_baseaudiosink_get_times (GstBaseSink * bsink,
+    GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
+static gboolean gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps);
+
+//static guint gst_baseaudiosink_signals[LAST_SIGNAL] = { 0 };
+
+static void
+gst_baseaudiosink_base_init (gpointer g_class)
+{
+}
+
+static void
+gst_baseaudiosink_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;
+
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_baseaudiosink_set_property);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_property);
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_baseaudiosink_change_state);
+
+  gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_baseaudiosink_event);
+  gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_baseaudiosink_preroll);
+  gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_baseaudiosink_render);
+  gstbasesink_class->get_times =
+      GST_DEBUG_FUNCPTR (gst_baseaudiosink_get_times);
+  gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_baseaudiosink_setcaps);
+}
+
+static void
+gst_baseaudiosink_init (GstBaseAudioSink * baseaudiosink)
+{
+}
+
+static void
+gst_baseaudiosink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASEAUDIOSINK (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_baseaudiosink_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASEAUDIOSINK (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps)
+{
+  GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
+  GstRingBufferSpec spec;
+
+  spec.caps = caps;
+  spec.segsize = 64;
+  spec.segtotal = 64;
+
+  gst_ringbuffer_release (sink->ringbuffer);
+  gst_ringbuffer_acquire (sink->ringbuffer, &spec);
+
+  return TRUE;
+}
+
+static void
+gst_baseaudiosink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
+    GstClockTime * start, GstClockTime * end)
+{
+  *start = GST_CLOCK_TIME_NONE;
+  *end = GST_CLOCK_TIME_NONE;
+}
+
+static void
+gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
+{
+}
+
+static GstFlowReturn
+gst_baseaudiosink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
+{
+  return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf)
+{
+  GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
+
+  gst_ringbuffer_write (sink->ringbuffer, GST_CLOCK_TIME_NONE,
+      GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+  return GST_FLOW_OK;
+}
+
+GstRingBuffer *
+gst_baseaudiosink_create_ringbuffer (GstBaseAudioSink * sink)
+{
+  GstBaseAudioSinkClass *bclass;
+  GstRingBuffer *buffer = NULL;
+
+  bclass = GST_BASEAUDIOSINK_GET_CLASS (sink);
+  if (bclass->create_ringbuffer)
+    buffer = bclass->create_ringbuffer (sink);
+
+  return buffer;
+}
+
+void
+gst_baseaudiosink_callback (GstRingBuffer * rbuf, guint advance, gpointer data)
+{
+  //GstBaseAudioSink *sink = GST_BASEAUDIOSINK (data);
+}
+
+static GstElementStateReturn
+gst_baseaudiosink_change_state (GstElement * element)
+{
+  GstElementStateReturn ret = GST_STATE_SUCCESS;
+  GstBaseAudioSink *sink = GST_BASEAUDIOSINK (element);
+  GstElementState transition = GST_STATE_TRANSITION (element);
+
+  switch (transition) {
+    case GST_STATE_NULL_TO_READY:
+      break;
+    case GST_STATE_READY_TO_PAUSED:
+      sink->ringbuffer = gst_baseaudiosink_create_ringbuffer (sink);
+      gst_ringbuffer_set_callback (sink->ringbuffer, gst_baseaudiosink_callback,
+          sink);
+      break;
+    case GST_STATE_PAUSED_TO_PLAYING:
+      gst_ringbuffer_play (sink->ringbuffer);
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+  switch (transition) {
+    case GST_STATE_PLAYING_TO_PAUSED:
+      gst_ringbuffer_stop (sink->ringbuffer);
+      break;
+    case GST_STATE_PAUSED_TO_READY:
+      gst_object_unref (GST_OBJECT (sink->ringbuffer));
+      break;
+    case GST_STATE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}