/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2005 Wim Taymans * * 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_BASE_SINK_H__ #define __GST_AUDIO_BASE_SINK_H__ #include #include #include "gstaudioringbuffer.h" #include "gstaudioclock.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_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. * * 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 } GstAudioBaseSinkSlaveMethod; #define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ()) typedef struct _GstAudioBaseSink GstAudioBaseSink; typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass; typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate; /** * 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]; }; GType gst_audio_base_sink_get_type(void); GType gst_audio_base_sink_slave_method_get_type (void); GstAudioRingBuffer * gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink); void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide); gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink); void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink, GstAudioBaseSinkSlaveMethod method); GstAudioBaseSinkSlaveMethod gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink); void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink, gint64 drift_tolerance); gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink); void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink, GstClockTime alignment_threshold); GstClockTime gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink); void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink, GstClockTime discont_wait); GstClockTime gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink); G_END_DECLS #endif /* __GST_AUDIO_BASE_SINK_H__ */