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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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93daa1435a
Since we started depending on GLib 2.44, we can be sure this macro is defined (it will be a no-op on compilers that don't support it). For plugins we should just start using `G_DECLARE_FINAL_TYPE` which means we no longer need the macro there, but for most types in base/gst-libs we don't want to break ABI, which means it's better to just keep it like it is (and use the `#ifdef` instead).
160 lines
4.9 KiB
C
160 lines
4.9 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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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref)
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G_END_DECLS
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#endif /* __GST_AUDIO_BASE_SRC_H__ */
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