gstreamer/gst-libs/gst/audio/gstbaseaudiosink.h
Wim Taymans fa8c2eb659 Make the base audiosink return an error when there is no audiobuffer negotiated.
Original commit message from CVS:
Make the base audiosink return an error when there is no
audiobuffer negotiated.
2005-05-06 16:18:24 +00:00

100 lines
3.7 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 incomming
* 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_BASEAUDIOSINK_H__
#define __GST_BASEAUDIOSINK_H__
#include <gst/gst.h>
#include <gst/base/gstbasesink.h>
#include "gstringbuffer.h"
#include "gstaudioclock.h"
G_BEGIN_DECLS
#define GST_TYPE_BASEAUDIOSINK (gst_baseaudiosink_get_type())
#define GST_BASEAUDIOSINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASEAUDIOSINK,GstBaseAudioSink))
#define GST_BASEAUDIOSINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASEAUDIOSINK,GstBaseAudioSinkClass))
#define GST_BASEAUDIOSINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASEAUDIOSINK, GstBaseAudioSinkClass))
#define GST_IS_BASEAUDIOSINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASEAUDIOSINK))
#define GST_IS_BASEAUDIOSINK_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASEAUDIOSINK))
#define GST_BASEAUDIOSINK_CLOCK(obj) (GST_BASEAUDIOSINK (obj)->clock)
#define GST_BASEAUDIOSINK_PAD(obj) (GST_BASEAUDIOSINK (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;
};
struct _GstBaseAudioSinkClass {
GstBaseSinkClass parent_class;
/* subclass ringbuffer allocation */
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
};
GType gst_baseaudiosink_get_type(void);
GstRingBuffer *gst_baseaudiosink_create_ringbuffer (GstBaseAudioSink *sink);
G_END_DECLS
#endif /* __GST_BASEAUDIOSINK_H__ */