mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
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
191 lines
7 KiB
C
191 lines
7 KiB
C
/* GStreamer
|
|
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
|
|
* 2005 Wim Taymans <wim@fluendo.com>
|
|
*
|
|
* gstaudiobasesink.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 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 GstAudioRingBuffer in addition to overriding
|
|
* the methods in GstBaseSink and this class.
|
|
*/
|
|
|
|
#ifndef __GST_AUDIO_AUDIO_H__
|
|
#include <gst/audio/audio.h>
|
|
#endif
|
|
|
|
#ifndef __GST_AUDIO_BASE_SINK_H__
|
|
#define __GST_AUDIO_BASE_SINK_H__
|
|
|
|
#include <gst/base/gstbasesink.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_AUDIO_BASE_SINK (gst_audio_base_sink_get_type())
|
|
#define GST_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink))
|
|
#define GST_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass))
|
|
#define GST_AUDIO_BASE_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass))
|
|
#define GST_IS_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK))
|
|
#define GST_IS_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SINK))
|
|
|
|
/**
|
|
* GST_AUDIO_BASE_SINK_CLOCK:
|
|
* @obj: a #GstAudioBaseSink
|
|
*
|
|
* Get the #GstClock of @obj.
|
|
*/
|
|
#define GST_AUDIO_BASE_SINK_CLOCK(obj) (GST_AUDIO_BASE_SINK (obj)->clock)
|
|
/**
|
|
* GST_AUDIO_BASE_SINK_PAD:
|
|
* @obj: a #GstAudioBaseSink
|
|
*
|
|
* Get the sink #GstPad of @obj.
|
|
*/
|
|
#define GST_AUDIO_BASE_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
|
|
|
|
/**
|
|
* GstAudioBaseSinkSlaveMethod:
|
|
* @GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: Resample to match the master clock
|
|
* @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
|
|
* drifts too much.
|
|
* @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done.
|
|
*
|
|
* Different possible clock slaving algorithms used when the internal audio
|
|
* clock is not selected as the pipeline master clock.
|
|
*/
|
|
typedef enum
|
|
{
|
|
GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE,
|
|
GST_AUDIO_BASE_SINK_SLAVE_SKEW,
|
|
GST_AUDIO_BASE_SINK_SLAVE_NONE
|
|
} GstAudioBaseSinkSlaveMethod;
|
|
|
|
#define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ())
|
|
|
|
typedef struct _GstAudioBaseSink GstAudioBaseSink;
|
|
typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
|
|
typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
|
|
|
|
/**
|
|
* GstAudioBaseSink:
|
|
*
|
|
* Opaque #GstAudioBaseSink.
|
|
*/
|
|
struct _GstAudioBaseSink {
|
|
GstBaseSink element;
|
|
|
|
/*< protected >*/ /* with LOCK */
|
|
/* our ringbuffer */
|
|
GstAudioRingBuffer *ringbuffer;
|
|
|
|
/* required buffer and latency in microseconds */
|
|
guint64 buffer_time;
|
|
guint64 latency_time;
|
|
|
|
/* the next sample to write */
|
|
guint64 next_sample;
|
|
|
|
/* clock */
|
|
GstClock *provided_clock;
|
|
|
|
/* with g_atomic_; currently rendering eos */
|
|
gboolean eos_rendering;
|
|
|
|
/*< private >*/
|
|
GstAudioBaseSinkPrivate *priv;
|
|
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
/**
|
|
* GstAudioBaseSinkClass:
|
|
* @parent_class: the parent class.
|
|
* @create_ringbuffer: create and return a #GstAudioRingBuffer to write to.
|
|
* @payload: payload data in a format suitable to write to the sink. If no
|
|
* payloading is required, returns a reffed copy of the original
|
|
* buffer, else returns the payloaded buffer with all other metadata
|
|
* copied.
|
|
*
|
|
* #GstAudioBaseSink class. Override the vmethod to implement
|
|
* functionality.
|
|
*/
|
|
struct _GstAudioBaseSinkClass {
|
|
GstBaseSinkClass parent_class;
|
|
|
|
/* subclass ringbuffer allocation */
|
|
GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSink *sink);
|
|
|
|
/* subclass payloader */
|
|
GstBuffer* (*payload) (GstAudioBaseSink *sink,
|
|
GstBuffer *buffer);
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GType gst_audio_base_sink_get_type(void);
|
|
GType gst_audio_base_sink_slave_method_get_type (void);
|
|
|
|
GstAudioRingBuffer *
|
|
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);
|
|
|
|
void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide);
|
|
gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);
|
|
|
|
void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
|
|
GstAudioBaseSinkSlaveMethod method);
|
|
GstAudioBaseSinkSlaveMethod
|
|
gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);
|
|
|
|
void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink,
|
|
gint64 drift_tolerance);
|
|
gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink);
|
|
|
|
void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
|
|
GstClockTime alignment_threshold);
|
|
GstClockTime
|
|
gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
|
|
|
|
void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
|
|
GstClockTime discont_wait);
|
|
GstClockTime
|
|
gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink);
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_AUDIO_BASE_SINK_H__ */
|