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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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a2f8ec4f5a
Make it possible for subclasses to provide the timestamp (as an absolute time against the pipeline clock) of the last read data. Fix up alsa to provide the timestamp received from alsa. Because the alsa timestamps are in monotonic time, we can only do this when the monotonic clock has been selected as the pipeline clock. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=635256
100 lines
3.3 KiB
C
100 lines
3.3 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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* gstaudiosrc.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_AUDIO_SRC_H__
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#define __GST_AUDIO_SRC_H__
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#include <gst/gst.h>
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#include <gst/audio/gstaudiobasesrc.h>
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G_BEGIN_DECLS
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#define GST_TYPE_AUDIO_SRC (gst_audio_src_get_type())
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#define GST_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SRC,GstAudioSrc))
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#define GST_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SRC,GstAudioSrcClass))
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#define GST_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_SRC,GstAudioSrcClass))
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#define GST_IS_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SRC))
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#define GST_IS_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SRC))
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typedef struct _GstAudioSrc GstAudioSrc;
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typedef struct _GstAudioSrcClass GstAudioSrcClass;
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/**
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* GstAudioSrc:
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*
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* Base class for simple audio sources.
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*/
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struct _GstAudioSrc {
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GstAudioBaseSrc element;
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/*< private >*/ /* with LOCK */
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GThread *thread;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstAudioSrcClass:
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* @parent_class: the parent class.
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* @open: open the device with the specified caps
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* @prepare: configure device with format
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* @unprepare: undo the configuration
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* @close: close the device
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* @read: read samples to the audio device
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* @delay: the number of samples queued in the device
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* @reset: unblock a read to the device and reset.
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*
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* #GstAudioSrc class. Override the vmethod to implement
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* functionality.
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*/
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struct _GstAudioSrcClass {
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GstAudioBaseSrcClass parent_class;
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/* vtable */
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/* open the device with given specs */
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gboolean (*open) (GstAudioSrc *src);
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/* prepare resources and state to operate with the given specs */
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gboolean (*prepare) (GstAudioSrc *src, GstAudioRingBufferSpec *spec);
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/* undo anything that was done in prepare() */
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gboolean (*unprepare) (GstAudioSrc *src);
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/* close the device */
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gboolean (*close) (GstAudioSrc *src);
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/* read samples from the device */
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guint (*read) (GstAudioSrc *src, gpointer data, guint length,
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GstClockTime *timestamp);
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/* get number of samples queued in the device */
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guint (*delay) (GstAudioSrc *src);
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/* reset the audio device, unblock from a write */
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void (*reset) (GstAudioSrc *src);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GType gst_audio_src_get_type(void);
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G_END_DECLS
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#endif /* __GST_AUDIO_SRC_H__ */
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