diff -r 000000000000 -r 0e761a78d257 gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosink.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/gst_plugins_base/gst-libs/gst/audio/gstbaseaudiosink.h Thu Dec 17 08:53:32 2009 +0200 @@ -0,0 +1,186 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen + * 2005 Wim Taymans + * + * gstbaseaudiosink.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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, 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 GstRingBuffer in addition to overriding + * the methods in GstBaseSink and this class. + */ + +#ifndef __GST_BASE_AUDIO_SINK_H__ +#define __GST_BASE_AUDIO_SINK_H__ + +#include +#include +#include "gstringbuffer.h" +#include "gstaudioclock.h" + +G_BEGIN_DECLS + +#define GST_TYPE_BASE_AUDIO_SINK (gst_base_audio_sink_get_type()) +#define GST_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink)) +#define GST_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass)) +#define GST_BASE_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass)) +#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK)) +#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK)) + +/** + * GST_BASE_AUDIO_SINK_CLOCK: + * @obj: a #GstBaseAudioSink + * + * Get the #GstClock of @obj. + */ +#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock) +/** + * GST_BASE_AUDIO_SINK_PAD: + * @obj: a #GstBaseAudioSink + * + * Get the sink #GstPad of @obj. + */ +#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad) + +/** + * GstBaseAudioSinkSlaveMethod: + * @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock + * @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock + * drifts too much. + * @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done. + * + * Different possible clock slaving algorithms + */ +typedef enum +{ + GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, + GST_BASE_AUDIO_SINK_SLAVE_SKEW, + GST_BASE_AUDIO_SINK_SLAVE_NONE +} GstBaseAudioSinkSlaveMethod; + +typedef struct _GstBaseAudioSink GstBaseAudioSink; +typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass; +typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate; + +/** + * GstBaseAudioSink: + * + * Opaque #GstBaseAudioSink. + */ +struct _GstBaseAudioSink { + GstBaseSink element; + + /*< protected >*/ /* with LOCK */ + /* our ringbuffer */ + GstRingBuffer *ringbuffer; + + /* required buffer and latency in microseconds */ + guint64 buffer_time; + guint64 latency_time; + + /* the next sample to write */ + guint64 next_sample; + + /* clock */ + gboolean provide_clock; + GstClock *provided_clock; + + /*< private >*/ + GstBaseAudioSinkPrivate *priv; + + gpointer _gst_reserved[GST_PADDING - 1]; +}; + +/** + * GstBaseAudioSinkClass: + * @parent_class: the parent class. + * @create_ringbuffer: create and return a #GstRingBuffer to write to. + * + * #GstBaseAudioSink class. Override the vmethod to implement + * functionality. + */ +struct _GstBaseAudioSinkClass { + GstBaseSinkClass parent_class; + + /* subclass ringbuffer allocation */ + GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink); + + /*< private >*/ + gpointer _gst_reserved[GST_PADDING]; +}; +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + + +GType gst_base_audio_sink_get_type(void); +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + + +GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink); +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + + +void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide); +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + +gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink); +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + + +void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink, + GstBaseAudioSinkSlaveMethod method); +#ifdef __SYMBIAN32__ +IMPORT_C +#endif + +GstBaseAudioSinkSlaveMethod + gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink); + +G_END_DECLS + +#endif /* __GST_BASE_AUDIO_SINK_H__ */