/usr/include/gstreamer-0.10/gst/audio/gstbaseaudiosink.h is in libgstreamer-plugins-base0.10-dev 0.10.36-1ubuntu0.2.
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* 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 used when the internal audio
* clock is not selected as the pipeline master clock.
*/
typedef enum
{
GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,
GST_BASE_AUDIO_SINK_SLAVE_SKEW,
GST_BASE_AUDIO_SINK_SLAVE_NONE
} GstBaseAudioSinkSlaveMethod;
#define GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD (gst_base_audio_sink_slave_method_get_type ())
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;
union {
struct {
/*< protected >*/
/* with g_atomic_; currently rendering eos */
gboolean eos_rendering;
} ABI;
gpointer _gst_reserved[GST_PADDING - 1];
} abidata;
};
/**
* GstBaseAudioSinkClass:
* @parent_class: the parent class.
* @create_ringbuffer: create and return a #GstRingBuffer to write to.
* @payload: payload data in a format suitable to write to the sink. If no
* payloading is required, returns a reffed copy of the original
* buffer, else returns the payloaded buffer with all other metadata
* copied. (Since: 0.10.36)
*
* #GstBaseAudioSink class. Override the vmethod to implement
* functionality.
*/
struct _GstBaseAudioSinkClass {
GstBaseSinkClass parent_class;
/* subclass ringbuffer allocation */
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
/* subclass payloader */
GstBuffer* (*payload) (GstBaseAudioSink *sink,
GstBuffer *buffer);
/*< private >*/
gpointer _gst_reserved[GST_PADDING - 1];
};
GType gst_base_audio_sink_get_type(void);
GType gst_base_audio_sink_slave_method_get_type (void);
GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide);
gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink);
void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink,
GstBaseAudioSinkSlaveMethod method);
GstBaseAudioSinkSlaveMethod
gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink);
void gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink *sink,
gint64 drift_tolerance);
gint64 gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink *sink);
void gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink,
GstClockTime alignment_threshold);
GstClockTime
gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink);
void gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink,
GstClockTime discont_wait);
GstClockTime
gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_SINK_H__ */
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