mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-26 18:20:44 +00:00
45e06fe704
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps), (gst_ring_buffer_samples_done): * gst-libs/gst/audio/gstringbuffer.h: Document better the fact that latency_time and buffer_time are values stored in microseconds, and not the usual GStreamer nanoseconds. Change the variables (compatibly) that store them from GstClockTime to guint64 to make it more clear that they're not storing clock times. Also, remove the bogus property description that says the user can specify -1 to get the default value, since that's never been the case. When computing the default segment size for the ring buffer, make it an integer number of samples. When the sub-class indicates a delay greater than the number of samples we've written return 0 from the audio sink get_time method.
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 in microseconds */
|
|
guint64 buffer_time;
|
|
guint64 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__ */
|