mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-25 11:11:08 +00:00
1a2642a1d2
Original commit message from CVS: * ext/pango/gsttextrender.h: * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstaudiosrc.h: * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosrc.h: * gst-libs/gst/audio/gstringbuffer.h: * gst-libs/gst/rtp/gstbasertpdepayload.h: * gst-libs/gst/rtp/gstbasertppayload.h: * gst-libs/gst/video/gstvideofilter.h: * gst-libs/gst/video/gstvideosink.h: * gst/playback/gstplaybasebin.h: * gst/tcp/gstmultifdsink.h: * sys/v4l/gstv4lelement.h: Fix broken GObject macros
110 lines
4 KiB
C
110 lines
4 KiB
C
/* GStreamer
|
|
* 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))
|
|
|
|
#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
|
|
#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
|
|
|
|
typedef struct _GstBaseAudioSink GstBaseAudioSink;
|
|
typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
|
|
|
|
struct _GstBaseAudioSink {
|
|
GstBaseSink 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 */
|
|
gboolean provide_clock;
|
|
GstClock *provided_clock;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
struct _GstBaseAudioSinkClass {
|
|
GstBaseSinkClass parent_class;
|
|
|
|
/* subclass ringbuffer allocation */
|
|
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GType gst_base_audio_sink_get_type(void);
|
|
|
|
GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_BASE_AUDIO_SINK_H__ */
|