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Diffstat (limited to 'gst-libs/gst/audio/gstbaseaudiosink.h')
-rw-r--r-- | gst-libs/gst/audio/gstbaseaudiosink.h | 87 |
1 files changed, 87 insertions, 0 deletions
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.h b/gst-libs/gst/audio/gstbaseaudiosink.h new file mode 100644 index 00000000..cd8bb61a --- /dev/null +++ b/gst-libs/gst/audio/gstbaseaudiosink.h @@ -0,0 +1,87 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> + * 2005 Wim Taymans <wim@fluendo.com> + * + * gstbaseaudosink.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 processed N samples 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. + */ + +#ifndef __GST_BASEAUDIOSINK_H__ +#define __GST_BASEAUDIOSINK_H__ + +#include <gst/gst.h> +#include <gst/base/gstbasesink.h> +#include "gstringbuffer.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; + + GstRingBuffer *ringbuffer; +}; + +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__ */ |