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93daa1435a
Since we started depending on GLib 2.44, we can be sure this macro is defined (it will be a no-op on compilers that don't support it). For plugins we should just start using `G_DECLARE_FINAL_TYPE` which means we no longer need the macro there, but for most types in base/gst-libs we don't want to break ABI, which means it's better to just keep it like it is (and use the `#ifdef` instead).
277 lines
11 KiB
C
277 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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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSink, gst_object_unref)
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G_END_DECLS
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#endif /* __GST_AUDIO_BASE_SINK_H__ */
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