/* 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_CAST(obj) ((GstAudioBaseSink*)obj)
#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.
* @GST_AUDIO_BASE_SINK_SLAVE_CUSTOM: Use custom clock slaving algorithm (Since: 1.6)
*
* 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,
GST_AUDIO_BASE_SINK_SLAVE_CUSTOM
} GstAudioBaseSinkSlaveMethod;
typedef struct _GstAudioBaseSink GstAudioBaseSink;
typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
/**
* GstAudioBaseSinkDiscontReason:
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: No discontinuity occurred
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: New caps are set, causing renegotiotion
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: Samples have been flushed
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: Sink was synchronized to the estimated latency (occurs during initialization)
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: Aligning buffers failed because the timestamps are too discontinuous
* @GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE: Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure())
*
* Different possible reasons for discontinuities. This enum is useful for the custom
* slave method.
*
* Since: 1.6
*/
typedef enum
{
GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT,
GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS,
GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH,
GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY,
GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT,
GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE
} GstAudioBaseSinkDiscontReason;
/**
* GstAudioBaseSinkCustomSlavingCallback:
* @sink: a #GstAudioBaseSink
* @etime: external clock time
* @itime: internal clock time
* @requested_skew: skew amount requested by the callback
* @discont_reason: reason for discontinuity (if any)
* @user_data: user data
*
* This function is set with gst_audio_base_sink_set_custom_slaving_callback()
* and is called during playback. It receives the current time of external and
* internal clocks, which the callback can then use to apply any custom
* slaving/synchronization schemes.
*
* The external clock is the sink's element clock, the internal one is the
* internal audio clock. The internal audio clock's calibration is applied to
* the timestamps before they are passed to the callback. The difference between
* etime and itime is the skew; how much internal and external clock lie apart
* from each other. A skew of 0 means both clocks are perfectly in sync.
* itime > etime means the external clock is going slower, while itime < etime
* means it is going faster than the internal clock. etime and itime are always
* valid timestamps, except for when a discontinuity happens.
*
* requested_skew is an output value the callback can write to. It informs the
* sink of whether or not it should move the playout pointer, and if so, by how
* much. This pointer is only NULL if a discontinuity occurs; otherwise, it is
* safe to write to *requested_skew. The default skew is 0.
*
* The sink may experience discontinuities. If one happens, discont is TRUE,
* itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
* This makes it possible to reset custom clock slaving algorithms when a
* discontinuity happens.
*
* Since: 1.6
*/
typedef void (*GstAudioBaseSinkCustomSlavingCallback) (GstAudioBaseSink *sink, GstClockTime etime, GstClockTime itime, GstClockTimeDiff *requested_skew, GstAudioBaseSinkDiscontReason discont_reason, gpointer user_data);
/**
* 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];
};
GST_AUDIO_API
GType gst_audio_base_sink_get_type(void);
GST_AUDIO_API
GstAudioRingBuffer *
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);
GST_AUDIO_API
void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide);
GST_AUDIO_API
gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);
GST_AUDIO_API
void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
GstAudioBaseSinkSlaveMethod method);
GST_AUDIO_API
GstAudioBaseSinkSlaveMethod
gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);
GST_AUDIO_API
void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink,
gint64 drift_tolerance);
GST_AUDIO_API
gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink);
GST_AUDIO_API
void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
GstClockTime alignment_threshold);
GST_AUDIO_API
GstClockTime
gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
GST_AUDIO_API
void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
GstClockTime discont_wait);
GST_AUDIO_API
GstClockTime
gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink);
GST_AUDIO_API
void
gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink,
GstAudioBaseSinkCustomSlavingCallback callback,
gpointer user_data,
GDestroyNotify notify);
GST_AUDIO_API
void gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink);
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref)
#endif
G_END_DECLS
#endif /* __GST_AUDIO_BASE_SINK_H__ */