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643e5f586c
This allows subclasses to provide a "payload" function to prepare buffers for consumption. The immediate use for this is for sinks that can handle compressed formats - parsers are directly connected to the sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading might be used. API: GstBaseAudioSinkClass:payload() https://bugzilla.gnome.org/show_bug.cgi?id=642730
185 lines
6.5 KiB
C
185 lines
6.5 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbaseaudiosink.h:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* a base class for audio sinks.
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*
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* It uses a ringbuffer to schedule playback of samples. This makes
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* it very easy to drop or insert samples to align incoming
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* buffers to the exact playback timestamp.
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*
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* Subclasses must provide a ringbuffer pointing to either DMA
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* memory or regular memory. A subclass should also call a callback
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* function when it has played N segments in the buffer. The subclass
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* is free to use a thread to signal this callback, use EIO or any
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* other mechanism.
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*
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* The base class is able to operate in push or pull mode. The chain
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* mode will queue the samples in the ringbuffer as much as possible.
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* The available space is calculated in the callback function.
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*
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* The pull mode will pull_range() a new buffer of N samples with a
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* configurable latency. This allows for high-end real time
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* audio processing pipelines driven by the audiosink. The callback
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* function will be used to perform a pull_range() on the sinkpad.
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* The thread scheduling the callback can be a real-time thread.
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*
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* Subclasses must implement a GstRingBuffer in addition to overriding
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* the methods in GstBaseSink and this class.
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*/
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#ifndef __GST_BASE_AUDIO_SINK_H__
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#define __GST_BASE_AUDIO_SINK_H__
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#include <gst/gst.h>
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#include <gst/base/gstbasesink.h>
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#include "gstringbuffer.h"
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#include "gstaudioclock.h"
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G_BEGIN_DECLS
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#define GST_TYPE_BASE_AUDIO_SINK (gst_base_audio_sink_get_type())
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#define GST_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink))
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#define GST_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass))
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#define GST_BASE_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass))
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#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK))
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#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))
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/**
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* GST_BASE_AUDIO_SINK_CLOCK:
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* @obj: a #GstBaseAudioSink
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*
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* Get the #GstClock of @obj.
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*/
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#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
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/**
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* GST_BASE_AUDIO_SINK_PAD:
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* @obj: a #GstBaseAudioSink
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*
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* Get the sink #GstPad of @obj.
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*/
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#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
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/**
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* GstBaseAudioSinkSlaveMethod:
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* @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock
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* @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
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* drifts too much.
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* @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done.
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*
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* Different possible clock slaving algorithms used when the internal audio
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* clock is not selected as the pipeline master clock.
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*/
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typedef enum
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{
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GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,
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GST_BASE_AUDIO_SINK_SLAVE_SKEW,
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GST_BASE_AUDIO_SINK_SLAVE_NONE
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} GstBaseAudioSinkSlaveMethod;
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#define GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD (gst_base_audio_sink_slave_method_get_type ())
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typedef struct _GstBaseAudioSink GstBaseAudioSink;
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typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
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typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate;
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/**
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* GstBaseAudioSink:
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*
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* Opaque #GstBaseAudioSink.
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*/
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struct _GstBaseAudioSink {
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GstBaseSink element;
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/*< protected >*/ /* with LOCK */
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/* our ringbuffer */
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GstRingBuffer *ringbuffer;
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/* required buffer and latency in microseconds */
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guint64 buffer_time;
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guint64 latency_time;
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/* the next sample to write */
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guint64 next_sample;
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/* clock */
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gboolean provide_clock;
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GstClock *provided_clock;
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/*< private >*/
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GstBaseAudioSinkPrivate *priv;
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union {
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struct {
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/*< protected >*/
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/* with g_atomic_; currently rendering eos */
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gboolean eos_rendering;
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} ABI;
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gpointer _gst_reserved[GST_PADDING - 1];
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} abidata;
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};
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/**
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* GstBaseAudioSinkClass:
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* @parent_class: the parent class.
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* @create_ringbuffer: create and return a #GstRingBuffer to write to.
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* @payload: payload data in a format suitable to write to the sink. If no
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* payloading is required, returns a reffed copy of the original
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* buffer, else returns the payloaded buffer with all other metadata
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* copied. (Since: 0.10.35)
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*
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* #GstBaseAudioSink class. Override the vmethod to implement
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* functionality.
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*/
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struct _GstBaseAudioSinkClass {
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GstBaseSinkClass parent_class;
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/* subclass ringbuffer allocation */
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GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
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/* subclass payloader */
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GstBuffer* (*payload) (GstBaseAudioSink *sink,
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GstBuffer *buffer);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING - 1];
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};
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GType gst_base_audio_sink_get_type(void);
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GType gst_base_audio_sink_slave_method_get_type (void);
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GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
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void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide);
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gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink);
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void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink,
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GstBaseAudioSinkSlaveMethod method);
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GstBaseAudioSinkSlaveMethod
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gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink);
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void gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink *sink,
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gint64 drift_tolerance);
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gint64 gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink *sink);
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G_END_DECLS
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#endif /* __GST_BASE_AUDIO_SINK_H__ */
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