/* 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(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK)) #define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock) #define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad) typedef struct _GstBaseAudioSink GstBaseAudioSink; typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass; struct _GstBaseAudioSink { GstBaseSink element; /*< protected >*/ /* with LOCK */ /* our ringbuffer */ GstRingBuffer *ringbuffer; /* required buffer and latency */ GstClockTime buffer_time; GstClockTime latency_time; /* the next sample to write */ guint64 next_sample; /* clock */ GstClock *clock; /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; struct _GstBaseAudioSinkClass { GstBaseSinkClass parent_class; /* subclass ringbuffer allocation */ GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink); /*< private >*/ gpointer _gst_reserved[GST_PADDING]; }; GType gst_base_audio_sink_get_type(void); GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink); G_END_DECLS #endif /* __GST_BASE_AUDIO_SINK_H__ */