mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-01 14:11:15 +00:00
20adaa1328
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosrc.h: * gst-libs/gst/audio/gstbaseaudiosink.h: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
146 lines
4.6 KiB
C
146 lines
4.6 KiB
C
/* GStreamer
|
|
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
|
|
* 2005 Wim Taymans <wim@fluendo.com>
|
|
*
|
|
* gstbaseaudiosrc.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 sources.
|
|
*/
|
|
|
|
#ifndef __GST_BASE_AUDIO_SRC_H__
|
|
#define __GST_BASE_AUDIO_SRC_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/base/gstpushsrc.h>
|
|
#include "gstringbuffer.h"
|
|
#include "gstaudioclock.h"
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_BASE_AUDIO_SRC (gst_base_audio_src_get_type())
|
|
#define GST_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc))
|
|
#define GST_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass))
|
|
#define GST_BASE_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass))
|
|
#define GST_IS_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC))
|
|
#define GST_IS_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC))
|
|
|
|
/**
|
|
* GST_BASE_AUDIO_SRC_CLOCK:
|
|
* @obj: a #GstBaseAudioSrc
|
|
*
|
|
* Get the #GstClock of @obj.
|
|
*/
|
|
#define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock)
|
|
/**
|
|
* GST_BASE_AUDIO_SRC_PAD:
|
|
* @obj: a #GstBaseAudioSrc
|
|
*
|
|
* Get the source #GstPad of @obj.
|
|
*/
|
|
#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
|
|
|
|
typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
|
|
typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
|
|
typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate;
|
|
|
|
/**
|
|
* GstBaseAudioSrcSlaveMethod:
|
|
* @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
|
|
* @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
|
|
* clock time.
|
|
* @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
|
|
* drifts too much.
|
|
* @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done.
|
|
*
|
|
* Different possible clock slaving algorithms when the internal audio clock was
|
|
* not selected as the pipeline clock.
|
|
*/
|
|
typedef enum
|
|
{
|
|
GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
|
|
GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
|
|
GST_BASE_AUDIO_SRC_SLAVE_SKEW,
|
|
GST_BASE_AUDIO_SRC_SLAVE_NONE
|
|
} GstBaseAudioSrcSlaveMethod;
|
|
|
|
#define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ())
|
|
|
|
/**
|
|
* GstBaseAudioSrc:
|
|
*
|
|
* Opaque #GstBaseAudioSrc.
|
|
*/
|
|
struct _GstBaseAudioSrc {
|
|
GstPushSrc element;
|
|
|
|
/*< protected >*/ /* with LOCK */
|
|
/* our ringbuffer */
|
|
GstRingBuffer *ringbuffer;
|
|
|
|
/* required buffer and latency */
|
|
GstClockTime buffer_time;
|
|
GstClockTime latency_time;
|
|
|
|
/* the next sample to write */
|
|
guint64 next_sample;
|
|
|
|
/* clock */
|
|
GstClock *clock;
|
|
|
|
/*< private >*/
|
|
GstBaseAudioSrcPrivate *priv;
|
|
|
|
gpointer _gst_reserved[GST_PADDING - 1];
|
|
};
|
|
|
|
/**
|
|
* GstBaseAudioSrcClass:
|
|
* @parent_class: the parent class.
|
|
* @create_ringbuffer: create and return a #GstRingBuffer to read from.
|
|
*
|
|
* #GstBaseAudioSrc class. Override the vmethod to implement
|
|
* functionality.
|
|
*/
|
|
struct _GstBaseAudioSrcClass {
|
|
GstPushSrcClass parent_class;
|
|
|
|
/* subclass ringbuffer allocation */
|
|
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GType gst_base_audio_src_get_type(void);
|
|
GType gst_base_audio_src_slave_method_get_type (void);
|
|
|
|
GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
|
|
|
|
void gst_base_audio_src_set_provide_clock (GstBaseAudioSrc *src, gboolean provide);
|
|
gboolean gst_base_audio_src_get_provide_clock (GstBaseAudioSrc *src);
|
|
|
|
void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src,
|
|
GstBaseAudioSrcSlaveMethod method);
|
|
GstBaseAudioSrcSlaveMethod
|
|
gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src);
|
|
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_BASE_AUDIO_SRC_H__ */
|