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Diffstat (limited to 'gst-libs/gst/audio/gstbaseaudiosink.h')
-rw-r--r-- | gst-libs/gst/audio/gstbaseaudiosink.h | 166 |
1 files changed, 0 insertions, 166 deletions
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.h b/gst-libs/gst/audio/gstbaseaudiosink.h deleted file mode 100644 index cf971527..00000000 --- a/gst-libs/gst/audio/gstbaseaudiosink.h +++ /dev/null @@ -1,166 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2005 Wim Taymans <wim@fluendo.com> - * - * 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 <gst/gst.h> -#include <gst/base/gstbasesink.h> -#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(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK)) - -/** - * GST_BASE_AUDIO_SINK_CLOCK: - * @obj: a #GstBaseAudioSink - * - * Get the #GstClock of @obj. - */ -#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock) -/** - * GST_BASE_AUDIO_SINK_PAD: - * @obj: a #GstBaseAudioSink - * - * Get the sink #GstPad of @obj. - */ -#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad) - -/** - * GstBaseAudioSinkSlaveMethod: - * @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock - * @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock - * drifts too much. - * @GST_BASE_AUDIO_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_BASE_AUDIO_SINK_SLAVE_RESAMPLE, - GST_BASE_AUDIO_SINK_SLAVE_SKEW, - GST_BASE_AUDIO_SINK_SLAVE_NONE -} GstBaseAudioSinkSlaveMethod; - -#define GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD (gst_base_audio_sink_slave_method_get_type ()) - -typedef struct _GstBaseAudioSink GstBaseAudioSink; -typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass; -typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate; - -/** - * GstBaseAudioSink: - * - * Opaque #GstBaseAudioSink. - */ -struct _GstBaseAudioSink { - GstBaseSink element; - - /*< protected >*/ /* with LOCK */ - /* our ringbuffer */ - GstRingBuffer *ringbuffer; - - /* required buffer and latency in microseconds */ - guint64 buffer_time; - guint64 latency_time; - - /* the next sample to write */ - guint64 next_sample; - - /* clock */ - gboolean provide_clock; - GstClock *provided_clock; - - /*< private >*/ - GstBaseAudioSinkPrivate *priv; - - gpointer _gst_reserved[GST_PADDING - 1]; -}; - -/** - * GstBaseAudioSinkClass: - * @parent_class: the parent class. - * @create_ringbuffer: create and return a #GstRingBuffer to write to. - * - * #GstBaseAudioSink class. Override the vmethod to implement - * functionality. - */ -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); -GType gst_base_audio_sink_slave_method_get_type (void); - -GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink); - -void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide); -gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink); - -void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink, - GstBaseAudioSinkSlaveMethod method); -GstBaseAudioSinkSlaveMethod - gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink); - -G_END_DECLS - -#endif /* __GST_BASE_AUDIO_SINK_H__ */ |