/* 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_AUDIO_H__ #include #endif #ifndef __GST_AUDIO_BASE_SINK_H__ #define __GST_AUDIO_BASE_SINK_H__ #include 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); #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref) #endif G_END_DECLS #endif /* __GST_AUDIO_BASE_SINK_H__ */