Added gst-plugins-base-subtitles0.10-0.10.34 for Meego Harmattan 1.2

[mafwsubrenderer] / gst-plugins-base-subtitles0.10 / gst-libs / gst / audio / gstbaseaudiosrc.h

diff --git a/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosrc.h b/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosrc.h
new file mode 100644 (file)
index 0000000..b55ac1e
--- /dev/null
+++ b/gst-plugins-base-subtitles0.10/gst-libs/gst/audio/gstbaseaudiosrc.h
@@ -0,0 +1,146 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstbaseaudiosrc.h:
+ *
+ * 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.
+ */
+
+/* a base class for audio sources.
+ */
+
+#ifndef __GST_BASE_AUDIO_SRC_H__
+#define __GST_BASE_AUDIO_SRC_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstpushsrc.h>
+#include "gstringbuffer.h"
+#include "gstaudioclock.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_BASE_AUDIO_SRC                 (gst_base_audio_src_get_type())
+#define GST_BASE_AUDIO_SRC(obj)                 (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc))
+#define GST_BASE_AUDIO_SRC_CLASS(klass)         (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass))
+#define GST_BASE_AUDIO_SRC_GET_CLASS(obj)       (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass))
+#define GST_IS_BASE_AUDIO_SRC(obj)              (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC))
+#define GST_IS_BASE_AUDIO_SRC_CLASS(klass)      (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC))
+
+/**
+ * GST_BASE_AUDIO_SRC_CLOCK:
+ * @obj: a #GstBaseAudioSrc
+ *
+ * Get the #GstClock of @obj.
+ */
+#define GST_BASE_AUDIO_SRC_CLOCK(obj)    (GST_BASE_AUDIO_SRC (obj)->clock)
+/**
+ * GST_BASE_AUDIO_SRC_PAD:
+ * @obj: a #GstBaseAudioSrc
+ *
+ * Get the source #GstPad of @obj.
+ */
+#define GST_BASE_AUDIO_SRC_PAD(obj)      (GST_BASE_SRC (obj)->srcpad)
+
+typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
+typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
+typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate;
+
+/**
+ * GstBaseAudioSrcSlaveMethod:
+ * @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock. 
+ * @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
+ * clock time.
+ * @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
+ * drifts too much.
+ * @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done. 
+ *
+ * Different possible clock slaving algorithms when the internal audio clock was
+ * not selected as the pipeline clock.
+ */
+typedef enum
+{
+  GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
+  GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
+  GST_BASE_AUDIO_SRC_SLAVE_SKEW,
+  GST_BASE_AUDIO_SRC_SLAVE_NONE
+} GstBaseAudioSrcSlaveMethod;
+
+#define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ())
+
+/**
+ * GstBaseAudioSrc:
+ *
+ * Opaque #GstBaseAudioSrc.
+ */
+struct _GstBaseAudioSrc {
+  GstPushSrc     element;
+
+  /*< protected >*/ /* with LOCK */
+  /* our ringbuffer */
+  GstRingBuffer *ringbuffer;
+
+  /* required buffer and latency */
+  GstClockTime   buffer_time;
+  GstClockTime   latency_time;
+
+  /* the next sample to write */
+  guint64        next_sample;
+
+  /* clock */
+  GstClock      *clock;
+
+  /*< private >*/
+  GstBaseAudioSrcPrivate *priv;
+
+  gpointer _gst_reserved[GST_PADDING - 1];
+};
+
+/**
+ * GstBaseAudioSrcClass:
+ * @parent_class: the parent class.
+ * @create_ringbuffer: create and return a #GstRingBuffer to read from.
+ *
+ * #GstBaseAudioSrc class. Override the vmethod to implement
+ * functionality.
+ */
+struct _GstBaseAudioSrcClass {
+  GstPushSrcClass parent_class;
+
+  /* subclass ringbuffer allocation */
+  GstRingBuffer* (*create_ringbuffer)  (GstBaseAudioSrc *src);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_base_audio_src_get_type(void);
+GType gst_base_audio_src_slave_method_get_type (void);
+
+GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
+
+void       gst_base_audio_src_set_provide_clock        (GstBaseAudioSrc *src, gboolean provide);
+gboolean   gst_base_audio_src_get_provide_clock        (GstBaseAudioSrc *src);
+
+void       gst_base_audio_src_set_slave_method         (GstBaseAudioSrc *src,
+                                                        GstBaseAudioSrcSlaveMethod method);
+GstBaseAudioSrcSlaveMethod
+           gst_base_audio_src_get_slave_method         (GstBaseAudioSrc *src);
+
+
+G_END_DECLS
+
+#endif /* __GST_BASE_AUDIO_SRC_H__ */