gstreamer/gst-libs/gst/audio/gstbaseaudiosink.h
Tim-Philipp Müller b9b56ce7d3 gst-libs/gst/: Add padding (you will need to rebuild gst-plugins-base, gst-plugins and all applications afterwards!)
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/net/gstnetbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add padding (you will need to rebuild gst-plugins-base,
gst-plugins and all applications afterwards!)
2005-08-09 17:29:40 +00:00

106 lines
3.8 KiB
C

/* 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(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))
#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
typedef struct _GstBaseAudioSink GstBaseAudioSink;
typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
struct _GstBaseAudioSink {
GstBaseSink element;
/*< protected >*/ /* with LOCK */
/* our ringbuffer */
GstRingBuffer *ringbuffer;
/* required buffer and latency */
GstClockTime buffer_time;
GstClockTime latency_time;
/* clock */
GstClock *clock;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
struct _GstBaseAudioSinkClass {
GstBaseSinkClass parent_class;
/* subclass ringbuffer allocation */
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_base_audio_sink_get_type(void);
GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_SINK_H__ */