mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 20:21:24 +00:00
277 lines
11 KiB
C
277 lines
11 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_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);
|
|
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref)
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_AUDIO_BASE_SINK_H__ */
|