--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosink.h Thu Dec 17 08:53:32 2009 +0200
@@ -0,0 +1,186 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2005 Wim Taymans <wim@fluendo.com>
+ *
+ * gstbaseaudiosink.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 sinks.
+ *
+ * It uses a ringbuffer to schedule playback of samples. This makes
+ * it very easy to drop or insert samples to align incoming
+ * buffers to the exact playback timestamp.
+ *
+ * Subclasses must provide a ringbuffer pointing to either DMA
+ * memory or regular memory. A subclass should also call a callback
+ * function when it has played N segments in the buffer. The subclass
+ * is free to use a thread to signal this callback, use EIO or any
+ * other mechanism.
+ *
+ * The base class is able to operate in push or pull mode. The chain
+ * mode will queue the samples in the ringbuffer as much as possible.
+ * The available space is calculated in the callback function.
+ *
+ * The pull mode will pull_range() a new buffer of N samples with a
+ * configurable latency. This allows for high-end real time
+ * audio processing pipelines driven by the audiosink. The callback
+ * function will be used to perform a pull_range() on the sinkpad.
+ * The thread scheduling the callback can be a real-time thread.
+ *
+ * Subclasses must implement a GstRingBuffer in addition to overriding
+ * the methods in GstBaseSink and this class.
+ */
+
+#ifndef __GST_BASE_AUDIO_SINK_H__
+#define __GST_BASE_AUDIO_SINK_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasesink.h>
+#include "gstringbuffer.h"
+#include "gstaudioclock.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_BASE_AUDIO_SINK (gst_base_audio_sink_get_type())
+#define GST_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink))
+#define GST_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass))
+#define GST_BASE_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass))
+#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK))
+#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))
+
+/**
+ * GST_BASE_AUDIO_SINK_CLOCK:
+ * @obj: a #GstBaseAudioSink
+ *
+ * Get the #GstClock of @obj.
+ */
+#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
+/**
+ * GST_BASE_AUDIO_SINK_PAD:
+ * @obj: a #GstBaseAudioSink
+ *
+ * Get the sink #GstPad of @obj.
+ */
+#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
+
+/**
+ * GstBaseAudioSinkSlaveMethod:
+ * @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock
+ * @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
+ * drifts too much.
+ * @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done.
+ *
+ * Different possible clock slaving algorithms
+ */
+typedef enum
+{
+ GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,
+ GST_BASE_AUDIO_SINK_SLAVE_SKEW,
+ GST_BASE_AUDIO_SINK_SLAVE_NONE
+} GstBaseAudioSinkSlaveMethod;
+
+typedef struct _GstBaseAudioSink GstBaseAudioSink;
+typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
+typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate;
+
+/**
+ * GstBaseAudioSink:
+ *
+ * Opaque #GstBaseAudioSink.
+ */
+struct _GstBaseAudioSink {
+ GstBaseSink element;
+
+ /*< protected >*/ /* with LOCK */
+ /* our ringbuffer */
+ GstRingBuffer *ringbuffer;
+
+ /* required buffer and latency in microseconds */
+ guint64 buffer_time;
+ guint64 latency_time;
+
+ /* the next sample to write */
+ guint64 next_sample;
+
+ /* clock */
+ gboolean provide_clock;
+ GstClock *provided_clock;
+
+ /*< private >*/
+ GstBaseAudioSinkPrivate *priv;
+
+ gpointer _gst_reserved[GST_PADDING - 1];
+};
+
+/**
+ * GstBaseAudioSinkClass:
+ * @parent_class: the parent class.
+ * @create_ringbuffer: create and return a #GstRingBuffer to write to.
+ *
+ * #GstBaseAudioSink class. Override the vmethod to implement
+ * functionality.
+ */
+struct _GstBaseAudioSinkClass {
+ GstBaseSinkClass parent_class;
+
+ /* subclass ringbuffer allocation */
+ GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GType gst_base_audio_sink_get_type(void);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+
+void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink,
+ GstBaseAudioSinkSlaveMethod method);
+#ifdef __SYMBIAN32__
+IMPORT_C
+#endif
+
+GstBaseAudioSinkSlaveMethod
+ gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink);
+
+G_END_DECLS
+
+#endif /* __GST_BASE_AUDIO_SINK_H__ */