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371e3e460a
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
163 lines
5 KiB
C
163 lines
5 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstaudiobasesrc.h:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* a base class for audio sources.
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*/
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#ifndef __GST_AUDIO_AUDIO_H__
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#include <gst/audio/audio.h>
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#endif
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#ifndef __GST_AUDIO_BASE_SRC_H__
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#define __GST_AUDIO_BASE_SRC_H__
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#include <gst/gst.h>
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#include <gst/base/gstpushsrc.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type())
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#define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc))
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#define GST_AUDIO_BASE_SRC_CAST(obj) ((GstAudioBaseSrc*)obj)
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#define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass))
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#define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass))
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#define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC))
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#define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC))
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/**
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* GST_AUDIO_BASE_SRC_CLOCK:
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* @obj: a #GstAudioBaseSrc
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*
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* Get the #GstClock of @obj.
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*/
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#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
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/**
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* GST_AUDIO_BASE_SRC_PAD:
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* @obj: a #GstAudioBaseSrc
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*
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* Get the source #GstPad of @obj.
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*/
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#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
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typedef struct _GstAudioBaseSrc GstAudioBaseSrc;
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typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass;
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typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
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/* FIXME 2.0: Should be "retimestamp" not "re-timestamp" */
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/**
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* GstAudioBaseSrcSlaveMethod:
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* @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
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* @GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP: Retimestamp output buffers with master
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* clock time.
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* @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
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* drifts too much.
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* @GST_AUDIO_BASE_SRC_SLAVE_NONE: No adjustment is done.
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*
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* Different possible clock slaving algorithms when the internal audio clock was
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* not selected as the pipeline clock.
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*/
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typedef enum
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{
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GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
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GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP,
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GST_AUDIO_BASE_SRC_SLAVE_SKEW,
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GST_AUDIO_BASE_SRC_SLAVE_NONE
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} GstAudioBaseSrcSlaveMethod;
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#define GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP
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/**
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* GstAudioBaseSrc:
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*
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* Opaque #GstAudioBaseSrc.
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*/
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struct _GstAudioBaseSrc {
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GstPushSrc element;
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/*< protected >*/ /* with LOCK */
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/* our ringbuffer */
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GstAudioRingBuffer *ringbuffer;
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/* required buffer and latency */
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GstClockTime buffer_time;
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GstClockTime latency_time;
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/* the next sample to write */
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guint64 next_sample;
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/* clock */
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GstClock *clock;
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/*< private >*/
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GstAudioBaseSrcPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstAudioBaseSrcClass:
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* @parent_class: the parent class.
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* @create_ringbuffer: create and return a #GstAudioRingBuffer to read from.
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*
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* #GstAudioBaseSrc class. Override the vmethod to implement
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* functionality.
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*/
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struct _GstAudioBaseSrcClass {
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GstPushSrcClass parent_class;
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/* subclass ringbuffer allocation */
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GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_AUDIO_API
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GType gst_audio_base_src_get_type(void);
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GST_AUDIO_API
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GstAudioRingBuffer *
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gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src);
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GST_AUDIO_API
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void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide);
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GST_AUDIO_API
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gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src);
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GST_AUDIO_API
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void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
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GstAudioBaseSrcSlaveMethod method);
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GST_AUDIO_API
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GstAudioBaseSrcSlaveMethod
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gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_AUDIO_BASE_SRC_H__ */
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