/* 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__ */