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371e3e460a
We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
280 lines
11 KiB
C
280 lines
11 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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* gstaudiobasesink.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 GstAudioRingBuffer 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_AUDIO_AUDIO_H__
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#include <gst/audio/audio.h>
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#endif
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#ifndef __GST_AUDIO_BASE_SINK_H__
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#define __GST_AUDIO_BASE_SINK_H__
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#include <gst/base/gstbasesink.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_BASE_SINK (gst_audio_base_sink_get_type())
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#define GST_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink))
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#define GST_AUDIO_BASE_SINK_CAST(obj) ((GstAudioBaseSink*)obj)
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#define GST_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass))
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#define GST_AUDIO_BASE_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass))
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#define GST_IS_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK))
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#define GST_IS_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SINK))
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/**
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* GST_AUDIO_BASE_SINK_CLOCK:
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* @obj: a #GstAudioBaseSink
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*
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* Get the #GstClock of @obj.
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*/
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#define GST_AUDIO_BASE_SINK_CLOCK(obj) (GST_AUDIO_BASE_SINK (obj)->clock)
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/**
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* GST_AUDIO_BASE_SINK_PAD:
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* @obj: a #GstAudioBaseSink
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*
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* Get the sink #GstPad of @obj.
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*/
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#define GST_AUDIO_BASE_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
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/**
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* GstAudioBaseSinkSlaveMethod:
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* @GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: Resample to match the master clock
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* @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
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* drifts too much.
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* @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done.
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* @GST_AUDIO_BASE_SINK_SLAVE_CUSTOM: Use custom clock slaving algorithm (Since: 1.6)
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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_AUDIO_BASE_SINK_SLAVE_RESAMPLE,
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GST_AUDIO_BASE_SINK_SLAVE_SKEW,
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GST_AUDIO_BASE_SINK_SLAVE_NONE,
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GST_AUDIO_BASE_SINK_SLAVE_CUSTOM
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} GstAudioBaseSinkSlaveMethod;
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typedef struct _GstAudioBaseSink GstAudioBaseSink;
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typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
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typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
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/**
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* GstAudioBaseSinkDiscontReason:
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* @GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: No discontinuity occurred
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* @GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: New caps are set, causing renegotiotion
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* @GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: Samples have been flushed
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* @GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: Sink was synchronized to the estimated latency (occurs during initialization)
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* @GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: Aligning buffers failed because the timestamps are too discontinuous
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* @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())
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*
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* Different possible reasons for discontinuities. This enum is useful for the custom
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* slave method.
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*
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* Since: 1.6
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*/
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typedef enum
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{
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GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT,
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GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS,
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GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH,
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GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY,
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GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT,
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GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE
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} GstAudioBaseSinkDiscontReason;
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/**
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* GstAudioBaseSinkCustomSlavingCallback:
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* @sink: a #GstAudioBaseSink
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* @etime: external clock time
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* @itime: internal clock time
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* @requested_skew: skew amount requested by the callback
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* @discont_reason: reason for discontinuity (if any)
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* @user_data: user data
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*
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* This function is set with gst_audio_base_sink_set_custom_slaving_callback()
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* and is called during playback. It receives the current time of external and
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* internal clocks, which the callback can then use to apply any custom
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* slaving/synchronization schemes.
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*
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* The external clock is the sink's element clock, the internal one is the
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* internal audio clock. The internal audio clock's calibration is applied to
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* the timestamps before they are passed to the callback. The difference between
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* etime and itime is the skew; how much internal and external clock lie apart
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* from each other. A skew of 0 means both clocks are perfectly in sync.
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* itime > etime means the external clock is going slower, while itime < etime
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* means it is going faster than the internal clock. etime and itime are always
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* valid timestamps, except for when a discontinuity happens.
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*
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* requested_skew is an output value the callback can write to. It informs the
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* sink of whether or not it should move the playout pointer, and if so, by how
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* much. This pointer is only NULL if a discontinuity occurs; otherwise, it is
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* safe to write to *requested_skew. The default skew is 0.
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*
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* The sink may experience discontinuities. If one happens, discont is TRUE,
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* itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
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* This makes it possible to reset custom clock slaving algorithms when a
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* discontinuity happens.
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*
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* Since: 1.6
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*/
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typedef void (*GstAudioBaseSinkCustomSlavingCallback) (GstAudioBaseSink *sink, GstClockTime etime, GstClockTime itime, GstClockTimeDiff *requested_skew, GstAudioBaseSinkDiscontReason discont_reason, gpointer user_data);
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/**
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* GstAudioBaseSink:
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*
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* Opaque #GstAudioBaseSink.
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*/
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struct _GstAudioBaseSink {
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GstBaseSink element;
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/*< protected >*/ /* with LOCK */
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/* our ringbuffer */
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GstAudioRingBuffer *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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GstClock *provided_clock;
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/* with g_atomic_; currently rendering eos */
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gboolean eos_rendering;
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/*< private >*/
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GstAudioBaseSinkPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstAudioBaseSinkClass:
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* @parent_class: the parent class.
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* @create_ringbuffer: create and return a #GstAudioRingBuffer 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.
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*
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* #GstAudioBaseSink class. Override the vmethod to implement
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* functionality.
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*/
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struct _GstAudioBaseSinkClass {
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GstBaseSinkClass parent_class;
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/* subclass ringbuffer allocation */
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GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSink *sink);
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/* subclass payloader */
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GstBuffer* (*payload) (GstAudioBaseSink *sink,
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GstBuffer *buffer);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_AUDIO_API
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GType gst_audio_base_sink_get_type(void);
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GST_AUDIO_API
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GstAudioRingBuffer *
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gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);
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GST_AUDIO_API
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void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide);
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GST_AUDIO_API
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gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);
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GST_AUDIO_API
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void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
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GstAudioBaseSinkSlaveMethod method);
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GST_AUDIO_API
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GstAudioBaseSinkSlaveMethod
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gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);
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GST_AUDIO_API
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void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink,
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gint64 drift_tolerance);
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GST_AUDIO_API
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gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink);
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GST_AUDIO_API
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void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
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GstClockTime alignment_threshold);
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GST_AUDIO_API
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GstClockTime
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gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
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GST_AUDIO_API
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void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
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GstClockTime discont_wait);
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GST_AUDIO_API
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GstClockTime
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gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink);
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GST_AUDIO_API
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void
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gst_audio_base_sink_set_custom_slaving_callback (GstAudioBaseSink * sink,
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GstAudioBaseSinkCustomSlavingCallback callback,
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gpointer user_data,
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GDestroyNotify notify);
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GST_AUDIO_API
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void gst_audio_base_sink_report_device_failure (GstAudioBaseSink * sink);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_AUDIO_BASE_SINK_H__ */
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