mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 11:11:08 +00:00
6b070784c4
Not nice to break people's code if we can avoid it. Could add a warning in the next cycle, and then require single includes in the cycle after. https://bugzilla.gnome.org/show_bug.cgi?id=695889
149 lines
4.7 KiB
C
149 lines
4.7 KiB
C
/* GStreamer
|
|
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
|
|
* 2005 Wim Taymans <wim@fluendo.com>
|
|
*
|
|
* gstaudiobasesrc.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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
/* a base class for audio sources.
|
|
*/
|
|
|
|
#ifndef __GST_AUDIO_AUDIO_H__
|
|
#include <gst/audio/audio.h>
|
|
#endif
|
|
|
|
#ifndef __GST_AUDIO_BASE_SRC_H__
|
|
#define __GST_AUDIO_BASE_SRC_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/base/gstpushsrc.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type())
|
|
#define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc))
|
|
#define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass))
|
|
#define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass))
|
|
#define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC))
|
|
#define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC))
|
|
|
|
/**
|
|
* GST_AUDIO_BASE_SRC_CLOCK:
|
|
* @obj: a #GstAudioBaseSrc
|
|
*
|
|
* Get the #GstClock of @obj.
|
|
*/
|
|
#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
|
|
/**
|
|
* GST_AUDIO_BASE_SRC_PAD:
|
|
* @obj: a #GstAudioBaseSrc
|
|
*
|
|
* Get the source #GstPad of @obj.
|
|
*/
|
|
#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
|
|
|
|
typedef struct _GstAudioBaseSrc GstAudioBaseSrc;
|
|
typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass;
|
|
typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
|
|
|
|
/**
|
|
* GstAudioBaseSrcSlaveMethod:
|
|
* @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
|
|
* @GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
|
|
* clock time.
|
|
* @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
|
|
* drifts too much.
|
|
* @GST_AUDIO_BASE_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_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
|
|
GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP,
|
|
GST_AUDIO_BASE_SRC_SLAVE_SKEW,
|
|
GST_AUDIO_BASE_SRC_SLAVE_NONE
|
|
} GstAudioBaseSrcSlaveMethod;
|
|
|
|
#define GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD (gst_audio_base_src_slave_method_get_type ())
|
|
|
|
/**
|
|
* GstAudioBaseSrc:
|
|
*
|
|
* Opaque #GstAudioBaseSrc.
|
|
*/
|
|
struct _GstAudioBaseSrc {
|
|
GstPushSrc element;
|
|
|
|
/*< protected >*/ /* with LOCK */
|
|
/* our ringbuffer */
|
|
GstAudioRingBuffer *ringbuffer;
|
|
|
|
/* required buffer and latency */
|
|
GstClockTime buffer_time;
|
|
GstClockTime latency_time;
|
|
|
|
/* the next sample to write */
|
|
guint64 next_sample;
|
|
|
|
/* clock */
|
|
GstClock *clock;
|
|
|
|
/*< private >*/
|
|
GstAudioBaseSrcPrivate *priv;
|
|
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
/**
|
|
* GstAudioBaseSrcClass:
|
|
* @parent_class: the parent class.
|
|
* @create_ringbuffer: create and return a #GstAudioRingBuffer to read from.
|
|
*
|
|
* #GstAudioBaseSrc class. Override the vmethod to implement
|
|
* functionality.
|
|
*/
|
|
struct _GstAudioBaseSrcClass {
|
|
GstPushSrcClass parent_class;
|
|
|
|
/* subclass ringbuffer allocation */
|
|
GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GType gst_audio_base_src_get_type(void);
|
|
GType gst_audio_base_src_slave_method_get_type (void);
|
|
|
|
GstAudioRingBuffer *
|
|
gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src);
|
|
|
|
void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide);
|
|
gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src);
|
|
|
|
void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
|
|
GstAudioBaseSrcSlaveMethod method);
|
|
GstAudioBaseSrcSlaveMethod
|
|
gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src);
|
|
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_AUDIO_BASE_SRC_H__ */
|