gstreamer/gst-libs/gst/audio/gstaudiobasesink.c

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An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
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* gstaudiobasesink.c:
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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*
* 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.
*/
/**
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* SECTION:gstaudiobasesink
* @short_description: Base class for audio sinks
* @see_also: #GstAudioSink, #GstAudioRingBuffer.
*
* This is the base class for audio sinks. Subclasses need to implement the
* ::create_ringbuffer vmethod. This base class will then take care of
* writing samples to the ringbuffer, synchronisation, clipping and flushing.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
#include <string.h>
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#include "gstaudiobasesink.h"
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
#define GST_CAT_DEFAULT gst_audio_base_sink_debug
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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#define GST_AUDIO_BASE_SINK_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkPrivate))
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struct _GstAudioBaseSinkPrivate
{
/* upstream latency */
GstClockTime us_latency;
/* the clock slaving algorithm in use */
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GstAudioBaseSinkSlaveMethod slave_method;
/* running average of clock skew */
GstClockTimeDiff avg_skew;
/* the number of samples we aligned last time */
gint64 last_align;
gboolean sync_latency;
GstClockTime eos_time;
/* number of microseconds we allow clock slaving to drift
* before resyncing */
guint64 drift_tolerance;
/* number of nanoseconds we allow timestamps to drift
* before resyncing */
GstClockTime alignment_threshold;
/* time of the previous detected discont candidate */
GstClockTime discont_time;
/* number of nanoseconds to wait until creating a discontinuity */
GstClockTime discont_wait;
};
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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/* BaseAudioSink signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
/* FIXME: 0.11, store the buffer_time and latency_time in nanoseconds */
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_PROVIDE_CLOCK TRUE
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#define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SINK_SLAVE_SKEW
/* FIXME, enable pull mode when clock slaving and trick modes are figured out */
#define DEFAULT_CAN_ACTIVATE_PULL FALSE
/* when timestamps drift for more than 40ms we resync. This should
* be anough to compensate for timestamp rounding errors. */
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
/* when clock slaving drift for more than 40ms we resync. This is
* a reasonable default */
#define DEFAULT_DRIFT_TOLERANCE ((40 * GST_MSECOND) / GST_USECOND)
/* allow for one second before resyncing to see if the timestamps drift will
* fix itself, or is a permanent offset */
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
PROP_SLAVE_METHOD,
PROP_CAN_ACTIVATE_PULL,
PROP_ALIGNMENT_THRESHOLD,
PROP_DRIFT_TOLERANCE,
PROP_DISCONT_WAIT,
PROP_LAST
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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};
GType
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gst_audio_base_sink_slave_method_get_type (void)
{
static volatile gsize slave_method_type = 0;
static const GEnumValue slave_method[] = {
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{GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, "GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE",
"resample"},
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{GST_AUDIO_BASE_SINK_SLAVE_SKEW, "GST_AUDIO_BASE_SINK_SLAVE_SKEW", "skew"},
{GST_AUDIO_BASE_SINK_SLAVE_NONE, "GST_AUDIO_BASE_SINK_SLAVE_NONE", "none"},
{0, NULL, NULL},
};
if (g_once_init_enter (&slave_method_type)) {
GType tmp =
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g_enum_register_static ("GstAudioBaseSinkSlaveMethod", slave_method);
g_once_init_leave (&slave_method_type, tmp);
}
return (GType) slave_method_type;
}
#define _do_init \
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GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "audiobasesink", 0, "audiobasesink element");
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#define gst_audio_base_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSink, gst_audio_base_sink,
GST_TYPE_BASE_SINK, _do_init);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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static void gst_audio_base_sink_dispose (GObject * object);
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static void gst_audio_base_sink_set_property (GObject * object, guint prop_id,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_base_sink_get_property (GObject * object, guint prop_id,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GValue * value, GParamSpec * pspec);
#if 0
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static GstStateChangeReturn gst_audio_base_sink_async_play (GstBaseSink *
basesink);
#endif
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static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement *
element, GstStateChange transition);
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static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink,
gboolean active);
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static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
query);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem);
static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
GstAudioBaseSink * sink);
static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
guint8 * data, guint len, gpointer user_data);
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static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GstBuffer * buffer);
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static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GstBuffer * buffer);
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static gboolean gst_audio_base_sink_event (GstBaseSink * bsink,
GstEvent * event);
static GstFlowReturn gst_audio_base_sink_wait_eos (GstBaseSink * bsink,
GstEvent * event);
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static void gst_audio_base_sink_get_times (GstBaseSink * bsink,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink,
GstCaps * caps);
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static GstCaps *gst_audio_base_sink_fixate (GstBaseSink * bsink,
GstCaps * caps);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
GstQuery * query);
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/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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static void
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gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
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g_type_class_add_private (klass, sizeof (GstAudioBaseSinkPrivate));
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gobject_class->set_property = gst_audio_base_sink_set_property;
gobject_class->get_property = gst_audio_base_sink_get_property;
gobject_class->dispose = gst_audio_base_sink_dispose;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in microseconds", 1,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
G_MAXINT64, DEFAULT_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"Audio latency in microseconds", 1,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
G_MAXINT64, DEFAULT_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
g_param_spec_boolean ("provide-clock", "Provide Clock",
"Provide a clock to be used as the global pipeline clock",
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_PROVIDE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
g_param_spec_enum ("slave-method", "Slave Method",
"Algorithm to use to match the rate of the masterclock",
2011-11-11 10:52:47 +00:00
GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
g_param_spec_boolean ("can-activate-pull", "Allow Pull Scheduling",
"Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
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* GstAudioBaseSink:drift-tolerance
*
* Controls the amount of time in microseconds that clocks are allowed
* to drift before resynchronisation happens.
*
* Since: 0.10.26
*/
g_object_class_install_property (gobject_class, PROP_DRIFT_TOLERANCE,
g_param_spec_int64 ("drift-tolerance", "Drift Tolerance",
"Tolerance for clock drift in microseconds", 1,
G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
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* GstAudioBaseSink:alignment_threshold
*
* Controls the amount of time in nanoseconds that timestamps are allowed
* to drift from their ideal time before choosing not to align them.
*
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* Since: 0.10.36
*/
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 1,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
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* GstAudioBaseSink:discont-wait
*
* A window of time in nanoseconds to wait before creating a discontinuity as
* a result of breaching the drift-tolerance.
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*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state);
gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock);
gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_sink_fixate);
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_sink_setcaps);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_sink_event);
gstbasesink_class->wait_eos =
GST_DEBUG_FUNCPTR (gst_audio_base_sink_wait_eos);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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gstbasesink_class->get_times =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_get_times);
gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_audio_base_sink_preroll);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_audio_base_sink_render);
gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query_pad);
gstbasesink_class->activate_pull =
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GST_DEBUG_FUNCPTR (gst_audio_base_sink_activate_pull);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_AUDIO_CLOCK);
g_type_class_ref (GST_TYPE_AUDIO_RING_BUFFER);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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}
static void
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gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
GstBaseSink *basesink;
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audiobasesink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (audiobasesink);
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audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
audiobasesink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
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audiobasesink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, audiobasesink,
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NULL);
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basesink = GST_BASE_SINK_CAST (audiobasesink);
basesink->can_activate_push = TRUE;
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
gst_base_sink_set_last_sample_enabled (basesink, FALSE);
if (DEFAULT_PROVIDE_CLOCK)
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
}
static void
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gst_audio_base_sink_dispose (GObject * object)
{
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GstAudioBaseSink *sink;
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sink = GST_AUDIO_BASE_SINK (object);
if (sink->provided_clock) {
gst_audio_clock_invalidate (sink->provided_clock);
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
}
if (sink->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstClock *
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gst_audio_base_sink_provide_clock (GstElement * elem)
{
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GstAudioBaseSink *sink;
GstClock *clock;
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sink = GST_AUDIO_BASE_SINK (elem);
/* we have no ringbuffer (must be NULL state) */
if (sink->ringbuffer == NULL)
goto wrong_state;
if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
GST_OBJECT_LOCK (sink);
if (!GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK))
goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
GST_OBJECT_UNLOCK (sink);
return clock;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
return NULL;
}
clock_disabled:
{
GST_DEBUG_OBJECT (sink, "clock provide disabled");
GST_OBJECT_UNLOCK (sink);
return NULL;
}
}
static gboolean
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gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
{
gboolean res = FALSE;
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GstAudioBaseSink *basesink;
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basesink = GST_AUDIO_BASE_SINK (bsink);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_LOG_OBJECT (basesink, "query convert");
if (basesink->ringbuffer) {
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res =
gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
src_val, dest_fmt, &dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
}
break;
}
default:
res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
break;
}
return res;
}
static gboolean
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gst_audio_base_sink_query (GstElement * element, GstQuery * query)
{
gboolean res = FALSE;
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GstAudioBaseSink *basesink;
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basesink = GST_AUDIO_BASE_SINK (element);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (basesink, "latency query");
/* ask parent first, it will do an upstream query for us. */
if ((res =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (basesink), &live,
&us_live, &min_l, &max_l))) {
GstClockTime base_latency, min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
GstAudioRingBufferSpec *spec;
GST_OBJECT_LOCK (basesink);
if (!basesink->ringbuffer || !basesink->ringbuffer->spec.info.rate) {
GST_OBJECT_UNLOCK (basesink);
GST_DEBUG_OBJECT (basesink,
"we are not yet negotiated, can't report latency yet");
res = FALSE;
goto done;
}
spec = &basesink->ringbuffer->spec;
basesink->priv->us_latency = min_l;
base_latency =
gst_util_uint64_scale_int (spec->seglatency * spec->segsize,
GST_SECOND, spec->info.rate * spec->info.bpf);
GST_OBJECT_UNLOCK (basesink);
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = base_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
GST_DEBUG_OBJECT (basesink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (basesink,
"peer max %" GST_TIME_FORMAT ", our max latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
GST_TIME_ARGS (max_latency));
} else {
GST_DEBUG_OBJECT (basesink,
"peer or we are not live, don't care about latency");
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
GST_LOG_OBJECT (basesink, "query convert");
if (basesink->ringbuffer) {
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, NULL);
res =
gst_audio_ring_buffer_convert (basesink->ringbuffer, src_fmt,
src_val, dest_fmt, &dest_val);
if (res) {
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
}
}
break;
}
default:
res = GST_ELEMENT_CLASS (parent_class)->query (element, query);
break;
}
done:
return res;
}
static GstClockTime
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gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
{
guint64 raw, samples;
guint delay;
GstClockTime result;
if (sink->ringbuffer == NULL || sink->ringbuffer->spec.info.rate == 0)
return GST_CLOCK_TIME_NONE;
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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/* our processed samples are always increasing */
raw = samples = gst_audio_ring_buffer_samples_done (sink->ringbuffer);
/* the number of samples not yet processed, this is still queued in the
* device (not played for playback). */
delay = gst_audio_ring_buffer_delay (sink->ringbuffer);
if (G_LIKELY (samples >= delay))
samples -= delay;
else
samples = 0;
result = gst_util_uint64_scale_int (samples, GST_SECOND,
sink->ringbuffer->spec.info.rate);
GST_DEBUG_OBJECT (sink,
"processed samples: raw %" G_GUINT64_FORMAT ", delay %u, real %"
G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
raw, delay, samples, GST_TIME_ARGS (result));
return result;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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}
/**
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* gst_audio_base_sink_set_provide_clock:
* @sink: a #GstAudioBaseSink
* @provide: new state
*
* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
*
* Since: 0.10.16
*/
void
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gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
gboolean provide)
{
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g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
if (provide)
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (sink);
}
/**
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* gst_audio_base_sink_get_provide_clock:
* @sink: a #GstAudioBaseSink
*
* Queries whether @sink will provide a clock or not. See also
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* gst_audio_base_sink_set_provide_clock.
*
* Returns: %TRUE if @sink will provide a clock.
*
* Since: 0.10.16
*/
gboolean
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gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
{
gboolean result;
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g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
GST_OBJECT_LOCK (sink);
result = GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
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* gst_audio_base_sink_set_slave_method:
* @sink: a #GstAudioBaseSink
* @method: the new slave method
*
* Controls how clock slaving will be performed in @sink.
*
* Since: 0.10.16
*/
void
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gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
GstAudioBaseSinkSlaveMethod method)
{
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g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->slave_method = method;
GST_OBJECT_UNLOCK (sink);
}
/**
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* gst_audio_base_sink_get_slave_method:
* @sink: a #GstAudioBaseSink
*
* Get the current slave method used by @sink.
*
* Returns: The current slave method used by @sink.
*
* Since: 0.10.16
*/
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GstAudioBaseSinkSlaveMethod
gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
{
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GstAudioBaseSinkSlaveMethod result;
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g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->slave_method;
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
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* gst_audio_base_sink_set_drift_tolerance:
* @sink: a #GstAudioBaseSink
* @drift_tolerance: the new drift tolerance in microseconds
*
* Controls the sink's drift tolerance.
*
* Since: 0.10.31
*/
void
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gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
gint64 drift_tolerance)
{
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g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->drift_tolerance = drift_tolerance;
GST_OBJECT_UNLOCK (sink);
}
/**
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* gst_audio_base_sink_get_drift_tolerance
* @sink: a #GstAudioBaseSink
*
* Get the current drift tolerance, in microseconds, used by @sink.
*
* Returns: The current drift tolerance used by @sink.
*
* Since: 0.10.31
*/
gint64
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gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
{
gint64 result;
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g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->drift_tolerance;
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
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* gst_audio_base_sink_set_alignment_threshold:
* @sink: a #GstAudioBaseSink
* @alignment_threshold: the new alignment threshold in nanoseconds
*
* Controls the sink's alignment threshold.
*
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* Since: 0.10.36
*/
void
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gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
GstClockTime alignment_threshold)
{
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g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->alignment_threshold = alignment_threshold;
GST_OBJECT_UNLOCK (sink);
}
/**
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* gst_audio_base_sink_get_alignment_threshold
* @sink: a #GstAudioBaseSink
*
* Get the current alignment threshold, in nanoseconds, used by @sink.
*
* Returns: The current alignment threshold used by @sink.
*
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* Since: 0.10.36
*/
GstClockTime
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gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
{
gint64 result;
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g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->alignment_threshold;
GST_OBJECT_UNLOCK (sink);
return result;
}
/**
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* gst_audio_base_sink_set_discont_wait:
* @sink: a #GstAudioBaseSink
* @discont_wait: the new discont wait in nanoseconds
*
* Controls how long the sink will wait before creating a discontinuity.
*
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* Since: 0.10.36
*/
void
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gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
GstClockTime discont_wait)
{
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g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->discont_wait = discont_wait;
GST_OBJECT_UNLOCK (sink);
}
/**
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* gst_audio_base_sink_get_discont_wait
* @sink: a #GstAudioBaseSink
*
* Get the current discont wait, in nanoseconds, used by @sink.
*
* Returns: The current discont wait used by @sink.
*
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* Since: 0.10.36
*/
GstClockTime
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gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
{
GstClockTime result;
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g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->discont_wait;
GST_OBJECT_UNLOCK (sink);
return result;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
static void
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gst_audio_base_sink_set_property (GObject * object, guint prop_id,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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const GValue * value, GParamSpec * pspec)
{
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GstAudioBaseSink *sink;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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sink = GST_AUDIO_BASE_SINK (object);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
switch (prop_id) {
case PROP_BUFFER_TIME:
sink->buffer_time = g_value_get_int64 (value);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
break;
case PROP_LATENCY_TIME:
sink->latency_time = g_value_get_int64 (value);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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break;
case PROP_PROVIDE_CLOCK:
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gst_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
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gst_audio_base_sink_set_slave_method (sink, g_value_get_enum (value));
break;
case PROP_CAN_ACTIVATE_PULL:
GST_BASE_SINK (sink)->can_activate_pull = g_value_get_boolean (value);
break;
case PROP_DRIFT_TOLERANCE:
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gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
break;
case PROP_ALIGNMENT_THRESHOLD:
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gst_audio_base_sink_set_alignment_threshold (sink,
g_value_get_uint64 (value));
break;
case PROP_DISCONT_WAIT:
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gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value));
break;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
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gst_audio_base_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
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GstAudioBaseSink *sink;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
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sink = GST_AUDIO_BASE_SINK (object);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, sink->buffer_time);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, sink->latency_time);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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break;
case PROP_PROVIDE_CLOCK:
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g_value_set_boolean (value, gst_audio_base_sink_get_provide_clock (sink));
break;
case PROP_SLAVE_METHOD:
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g_value_set_enum (value, gst_audio_base_sink_get_slave_method (sink));
break;
case PROP_CAN_ACTIVATE_PULL:
g_value_set_boolean (value, GST_BASE_SINK (sink)->can_activate_pull);
break;
case PROP_DRIFT_TOLERANCE:
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g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink));
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value,
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gst_audio_base_sink_get_alignment_threshold (sink));
break;
case PROP_DISCONT_WAIT:
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g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink));
break;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
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gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
GstAudioRingBufferSpec *spec;
GstClockTime now;
GstClockTime crate_num, crate_denom;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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if (!sink->ringbuffer)
return FALSE;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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spec = &sink->ringbuffer->spec;
GST_DEBUG_OBJECT (sink, "release old ringbuffer");
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/* get current time, updates the last_time. When the subclass has a clock that
* restarts from 0 when a new format is negotiated, it will call
* gst_audio_clock_reset() which will use this last_time to create an offset
* so that time from the clock keeps on increasing monotonically. */
now = gst_clock_get_time (sink->provided_clock);
GST_DEBUG_OBJECT (sink, "time was %" GST_TIME_FORMAT, GST_TIME_ARGS (now));
/* release old ringbuffer */
gst_audio_ring_buffer_pause (sink->ringbuffer);
gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_release (sink->ringbuffer);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GST_DEBUG_OBJECT (sink, "parse caps");
spec->buffer_time = sink->buffer_time;
spec->latency_time = sink->latency_time;
/* parse new caps */
if (!gst_audio_ring_buffer_parse_caps (spec, caps))
goto parse_error;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
gst_audio_ring_buffer_debug_spec_buff (spec);
GST_DEBUG_OBJECT (sink, "acquire ringbuffer");
if (!gst_audio_ring_buffer_acquire (sink->ringbuffer, spec))
goto acquire_error;
2011-11-18 11:37:10 +00:00
if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
}
/* due to possible changes in the spec file we should recalibrate the clock */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
&crate_num, &crate_denom);
gst_clock_set_calibration (sink->provided_clock,
gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
crate_denom);
/* calculate actual latency and buffer times.
* FIXME: In 0.11, store the latency_time internally in ns */
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->info.rate * spec->info.bpf);
spec->buffer_time = spec->segtotal * spec->latency_time;
gst_audio_ring_buffer_debug_spec_buff (spec);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG_OBJECT (sink, "could not parse caps");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
(NULL), ("cannot parse audio format."));
return FALSE;
}
acquire_error:
{
GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
return FALSE;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
}
2012-03-11 18:04:41 +00:00
static GstCaps *
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gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
{
GstStructure *s;
gint width, depth;
2012-03-11 18:04:41 +00:00
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
/* fields for all formats */
gst_structure_fixate_field_nearest_int (s, "rate", 44100);
gst_structure_fixate_field_nearest_int (s, "channels", 2);
gst_structure_fixate_field_nearest_int (s, "width", 16);
/* fields for int */
if (gst_structure_has_field (s, "depth")) {
gst_structure_get_int (s, "width", &width);
/* round width to nearest multiple of 8 for the depth */
depth = GST_ROUND_UP_8 (width);
gst_structure_fixate_field_nearest_int (s, "depth", depth);
}
if (gst_structure_has_field (s, "signed"))
gst_structure_fixate_field_boolean (s, "signed", TRUE);
if (gst_structure_has_field (s, "endianness"))
gst_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
2011-11-10 15:24:12 +00:00
2012-03-11 18:04:41 +00:00
caps = GST_BASE_SINK_CLASS (parent_class)->fixate (bsink, caps);
return caps;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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GstClockTime * start, GstClockTime * end)
{
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
/* our clock sync is a bit too much for the base class to handle so
* we implement it ourselves. */
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
/* This waits for the drain to happen and can be canceled */
static gboolean
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gst_audio_base_sink_drain (GstAudioBaseSink * sink)
{
if (!sink->ringbuffer)
return TRUE;
if (!sink->ringbuffer->spec.info.rate)
return TRUE;
/* if PLAYING is interrupted,
* arrange to have clock running when going to PLAYING again */
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g_atomic_int_set (&sink->eos_rendering, 1);
/* need to start playback before we can drain, but only when
* we have successfully negotiated a format and thus acquired the
* ringbuffer. */
if (gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
gst_audio_ring_buffer_start (sink->ringbuffer);
if (sink->priv->eos_time != -1) {
GST_DEBUG_OBJECT (sink,
"last sample time %" GST_TIME_FORMAT,
GST_TIME_ARGS (sink->priv->eos_time));
/* wait for the EOS time to be reached, this is the time when the last
* sample is played. */
gst_base_sink_wait (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
GST_DEBUG_OBJECT (sink, "drained audio");
}
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g_atomic_int_set (&sink->eos_rendering, 0);
return TRUE;
}
static GstFlowReturn
gst_audio_base_sink_wait_eos (GstBaseSink * bsink, GstEvent * event)
{
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
GstFlowReturn ret;
ret = GST_BASE_SINK_CLASS (parent_class)->wait_eos (bsink, event);
if (ret != GST_FLOW_OK)
return ret;
/* now wait till we played everything */
gst_audio_base_sink_drain (sink);
return ret;
}
More work on the audiosink, mostly debugging and a race in shutdown. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_class_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time), (gst_baseaudiosink_set_property), (build_linear_format), (debug_spec_caps), (debug_spec_buffer), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_stop), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample), (wait_segment), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_advance), (gst_ringbuffer_clear): More work on the audiosink, mostly debugging and a race in shutdown.
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static gboolean
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gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
if (sink->ringbuffer)
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
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sink->priv->discont_time = -1;
if (sink->ringbuffer)
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
break;
default:
break;
}
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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}
static GstFlowReturn
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gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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{
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
/* we don't really do anything when prerolling. We could make a
* property to play this buffer to have some sort of scrubbing
* support. */
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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return GST_FLOW_OK;
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
return GST_FLOW_NOT_NEGOTIATED;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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}
static guint64
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_get_offset (GstAudioBaseSink * sink)
{
guint64 sample;
gint writeseg, segdone, sps;
gint diff;
/* assume we can append to the previous sample */
sample = sink->next_sample;
/* no previous sample, try to insert at position 0 */
if (sample == -1)
sample = 0;
sps = sink->ringbuffer->samples_per_seg;
/* figure out the segment and the offset inside the segment where
* the sample should be written. */
writeseg = sample / sps;
/* get the currently processed segment */
segdone = g_atomic_int_get (&sink->ringbuffer->segdone)
- sink->ringbuffer->segbase;
/* see how far away it is from the write segment */
diff = writeseg - segdone;
if (diff < 0) {
/* sample would be dropped, position to next playable position */
sample = (segdone + 1) * sps;
}
return sample;
}
static GstClockTime
clock_convert_external (GstClockTime external, GstClockTime cinternal,
GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom)
{
/* adjust for rate and speed */
if (external >= cexternal) {
external =
gst_util_uint64_scale (external - cexternal, crate_denom, crate_num);
external += cinternal;
} else {
external =
gst_util_uint64_scale (cexternal - external, crate_denom, crate_num);
if (cinternal > external)
external = cinternal - external;
else
external = 0;
}
return external;
}
/* algorithm to calculate sample positions that will result in resampling to
* match the clock rate of the master */
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal;
GstClockTime crate_num, crate_denom;
/* FIXME, we can sample and add observations here or use the timeouts on the
* clock. No idea which one is better or more stable. The timeout seems more
* arbitrary but this one seems more demanding and does not work when there is
* no data comming in to the sink. */
#if 0
GstClockTime etime, itime;
gdouble r_squared;
/* sample clocks and figure out clock skew */
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
itime = gst_audio_clock_get_time (sink->provided_clock);
/* add new observation */
gst_clock_add_observation (sink->provided_clock, itime, etime, &r_squared);
#endif
/* get calibration parameters to compensate for speed and offset differences
* when we are slaved */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
GST_TIME_FORMAT " %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT " = %f",
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal), crate_num,
crate_denom, gst_guint64_to_gdouble (crate_num) /
gst_guint64_to_gdouble (crate_denom));
if (crate_num == 0)
crate_denom = crate_num = 1;
/* bring external time to internal time */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
GST_DEBUG_OBJECT (sink,
"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
*srender_start = render_start;
*srender_stop = render_stop;
}
/* algorithm to calculate sample positions that will result in changing the
* playout pointer to match the clock rate of the master */
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal, crate_num, crate_denom;
GstClockTime etime, itime;
GstClockTimeDiff skew, mdrift, mdrift2;
gint driftsamples;
gint64 last_align;
/* get calibration parameters to compensate for offsets */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
/* sample clocks and figure out clock skew */
etime = gst_clock_get_time (GST_ELEMENT_CLOCK (sink));
itime = gst_audio_clock_get_time (sink->provided_clock);
itime = gst_audio_clock_adjust (sink->provided_clock, itime);
GST_DEBUG_OBJECT (sink,
"internal %" GST_TIME_FORMAT " external %" GST_TIME_FORMAT
" cinternal %" GST_TIME_FORMAT " cexternal %" GST_TIME_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime),
GST_TIME_ARGS (cinternal), GST_TIME_ARGS (cexternal));
/* make sure we never go below 0 */
etime = etime > cexternal ? etime - cexternal : 0;
itime = itime > cinternal ? itime - cinternal : 0;
/* do itime - etime.
* positive value means external clock goes slower
* negative value means external clock goes faster */
skew = GST_CLOCK_DIFF (etime, itime);
if (sink->priv->avg_skew == -1) {
/* first observation */
sink->priv->avg_skew = skew;
} else {
/* next observations use a moving average */
sink->priv->avg_skew = (31 * sink->priv->avg_skew + skew) / 32;
}
GST_DEBUG_OBJECT (sink, "internal %" GST_TIME_FORMAT " external %"
GST_TIME_FORMAT " skew %" G_GINT64_FORMAT " avg %" G_GINT64_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime), skew, sink->priv->avg_skew);
/* the max drift we allow */
mdrift = sink->priv->drift_tolerance * 1000;
mdrift2 = mdrift / 2;
/* adjust playout pointer based on skew */
if (sink->priv->avg_skew > mdrift2) {
/* master is running slower, move internal time forward */
GST_WARNING_OBJECT (sink,
"correct clock skew %" G_GINT64_FORMAT " > %" G_GINT64_FORMAT,
sink->priv->avg_skew, mdrift2);
cexternal = cexternal > mdrift ? cexternal - mdrift : 0;
sink->priv->avg_skew -= mdrift;
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
last_align = sink->priv->last_align;
/* if we were aligning in the wrong direction or we aligned more than what we
* will correct, resync */
if (last_align < 0 || last_align > driftsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
crate_num, crate_denom);
} else if (sink->priv->avg_skew < -mdrift2) {
/* master is running faster, move external time forwards */
GST_WARNING_OBJECT (sink,
"correct clock skew %" G_GINT64_FORMAT " < %" G_GINT64_FORMAT,
sink->priv->avg_skew, -mdrift2);
cexternal += mdrift;
sink->priv->avg_skew += mdrift;
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
last_align = sink->priv->last_align;
/* if we were aligning in the wrong direction or we aligned more than what we
* will correct, resync */
if (last_align > 0 || -last_align > driftsamples)
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink,
"last_align %" G_GINT64_FORMAT " driftsamples %u, next %"
G_GUINT64_FORMAT, last_align, driftsamples, sink->next_sample);
gst_clock_set_calibration (sink->provided_clock, cinternal, cexternal,
crate_num, crate_denom);
}
/* convert, ignoring speed */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
*srender_start = render_start;
*srender_stop = render_stop;
}
/* apply the clock offset but do no slaving otherwise */
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_none_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
GstClockTime cinternal, cexternal, crate_num, crate_denom;
/* get calibration parameters to compensate for offsets */
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom);
/* convert, ignoring speed */
render_start = clock_convert_external (render_start, cinternal, cexternal,
crate_num, crate_denom);
render_stop = clock_convert_external (render_stop, cinternal, cexternal,
crate_num, crate_denom);
*srender_start = render_start;
*srender_stop = render_stop;
}
/* converts render_start and render_stop to their slaved values */
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_handle_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
switch (sink->priv->slave_method) {
2011-11-11 10:52:47 +00:00
case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
gst_audio_base_sink_resample_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
2011-11-11 10:52:47 +00:00
case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
gst_audio_base_sink_skew_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
2011-11-11 10:52:47 +00:00
case GST_AUDIO_BASE_SINK_SLAVE_NONE:
gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
default:
g_warning ("unknown slaving method %d", sink->priv->slave_method);
break;
}
}
/* must be called with LOCK */
static GstFlowReturn
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
{
GstClock *clock;
GstClockReturn status;
GstClockTime time, render_delay;
GstFlowReturn ret;
2011-11-11 10:52:47 +00:00
GstAudioBaseSink *sink;
GstClockTime itime, etime;
GstClockTime rate_num, rate_denom;
GstClockTimeDiff jitter;
2011-11-11 10:52:47 +00:00
sink = GST_AUDIO_BASE_SINK (bsink);
clock = GST_ELEMENT_CLOCK (sink);
if (G_UNLIKELY (clock == NULL))
goto no_clock;
/* we provided the global clock, don't need to do anything special */
if (clock == sink->provided_clock)
goto no_slaving;
GST_OBJECT_UNLOCK (sink);
do {
GST_DEBUG_OBJECT (sink, "checking preroll");
ret = gst_base_sink_do_preroll (bsink, obj);
if (ret != GST_FLOW_OK)
goto flushing;
GST_OBJECT_LOCK (sink);
time = sink->priv->us_latency;
GST_OBJECT_UNLOCK (sink);
/* Renderdelay is added onto our own latency, and needs
* to be subtracted as well */
render_delay = gst_base_sink_get_render_delay (bsink);
if (G_LIKELY (time > render_delay))
time -= render_delay;
else
time = 0;
/* preroll done, we can sync since we are in PLAYING now. */
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT, GST_TIME_ARGS (time));
/* wait for the clock, this can be interrupted because we got shut down or
* we PAUSED. */
status = gst_base_sink_wait_clock (bsink, time, &jitter);
GST_DEBUG_OBJECT (sink, "clock returned %d %" GST_TIME_FORMAT, status,
GST_TIME_ARGS (jitter));
/* invalid time, no clock or sync disabled, just continue then */
if (status == GST_CLOCK_BADTIME)
break;
/* waiting could have been interrupted and we can be flushing now */
if (G_UNLIKELY (bsink->flushing))
goto flushing;
/* retry if we got unscheduled, which means we did not reach the timeout
* yet. if some other error occures, we continue. */
} while (status == GST_CLOCK_UNSCHEDULED);
GST_OBJECT_LOCK (sink);
GST_DEBUG_OBJECT (sink, "latency synced");
/* when we prerolled in time, we can accurately set the calibration,
* our internal clock should exactly have been the latency (== the running
* time of the external clock) */
etime = GST_ELEMENT_CAST (sink)->base_time + time;
itime = gst_audio_clock_get_time (sink->provided_clock);
itime = gst_audio_clock_adjust (sink->provided_clock, itime);
if (status == GST_CLOCK_EARLY) {
/* when we prerolled late, we have to take into account the lateness */
GST_DEBUG_OBJECT (sink, "late preroll, adding jitter");
etime += jitter;
}
/* start ringbuffer so we can start slaving right away when we need to */
gst_audio_ring_buffer_start (sink->ringbuffer);
GST_DEBUG_OBJECT (sink,
"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
/* copy the original calibrated rate but update the internal and external
* times. */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
&rate_denom);
gst_clock_set_calibration (sink->provided_clock, itime, etime,
rate_num, rate_denom);
switch (sink->priv->slave_method) {
2011-11-11 10:52:47 +00:00
case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
/* only set as master when we are resampling */
GST_DEBUG_OBJECT (sink, "Setting clock as master");
gst_clock_set_master (sink->provided_clock, clock);
break;
2011-11-11 10:52:47 +00:00
case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
case GST_AUDIO_BASE_SINK_SLAVE_NONE:
default:
break;
}
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
2011-11-07 10:31:47 +00:00
sink->priv->discont_time = -1;
return GST_FLOW_OK;
/* ERRORS */
no_clock:
{
GST_DEBUG_OBJECT (sink, "we have no clock");
return GST_FLOW_OK;
}
no_slaving:
{
GST_DEBUG_OBJECT (sink, "we are not slaved");
return GST_FLOW_OK;
}
flushing:
{
GST_DEBUG_OBJECT (sink, "we are flushing");
GST_OBJECT_LOCK (sink);
return GST_FLOW_FLUSHING;
}
}
static gint64
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
GstClockTime sample_offset)
{
GstAudioRingBuffer *ringbuf = sink->ringbuffer;
gint64 align;
2011-11-07 10:18:34 +00:00
gint64 sample_diff;
gint64 max_sample_diff;
gint segdone = g_atomic_int_get (&ringbuf->segdone) - ringbuf->segbase;
gint64 samples_done = segdone * ringbuf->samples_per_seg;
gint64 headroom = sample_offset - samples_done;
gboolean allow_align = TRUE;
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
2011-11-07 10:31:47 +00:00
gboolean discont = FALSE;
gint rate;
/* now try to align the sample to the previous one, first see how big the
* difference is. */
if (sample_offset >= sink->next_sample)
2011-11-07 10:18:34 +00:00
sample_diff = sample_offset - sink->next_sample;
else
2011-11-07 10:18:34 +00:00
sample_diff = sink->next_sample - sample_offset;
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
/* calculate the max allowed drift in units of samples. */
max_sample_diff = gst_util_uint64_scale_int (sink->priv->alignment_threshold,
rate, GST_SECOND);
/* calc align with previous sample */
align = sink->next_sample - sample_offset;
/* don't align if it means writing behind the read-segment */
2011-11-07 10:18:34 +00:00
if (sample_diff > headroom && align < 0)
allow_align = FALSE;
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
2011-11-07 10:31:47 +00:00
if (G_UNLIKELY (sample_diff >= max_sample_diff)) {
/* wait before deciding to make a discontinuity */
if (sink->priv->discont_wait > 0) {
GstClockTime time = gst_util_uint64_scale_int (sample_offset,
GST_SECOND, rate);
if (sink->priv->discont_time == -1) {
/* discont candidate */
sink->priv->discont_time = time;
} else if (time - sink->priv->discont_time >= sink->priv->discont_wait) {
/* discont_wait expired, discontinuity detected */
discont = TRUE;
sink->priv->discont_time = -1;
}
} else {
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
2011-11-07 10:31:47 +00:00
discont = TRUE;
}
} else if (G_UNLIKELY (sink->priv->discont_time != -1)) {
/* we have had a discont, but are now back on track! */
sink->priv->discont_time = -1;
}
if (G_LIKELY (!discont && allow_align)) {
GST_DEBUG_OBJECT (sink,
"align with prev sample, ABS (%" G_GINT64_FORMAT ") < %"
2011-11-07 10:18:34 +00:00
G_GINT64_FORMAT, align, max_sample_diff);
} else {
gint64 diff_s G_GNUC_UNUSED;
/* calculate sample diff in seconds for error message */
diff_s = gst_util_uint64_scale_int (sample_diff, GST_SECOND, rate);
/* timestamps drifted apart from previous samples too much, we need to
* resync. We log this as an element warning. */
GST_WARNING_OBJECT (sink,
"Unexpected discontinuity in audio timestamps of "
"%s%" GST_TIME_FORMAT ", resyncing",
sample_offset > sink->next_sample ? "+" : "-", GST_TIME_ARGS (diff_s));
align = 0;
}
return align;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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static GstFlowReturn
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gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
{
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstClockTimeDiff sync_offset, ts_offset;
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GstAudioBaseSinkClass *bclass;
GstAudioBaseSink *sink;
GstAudioRingBuffer *ringbuf;
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gint64 diff, align;
guint64 ctime, cstop;
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gsize offset;
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GstMapInfo info;
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gsize size;
guint samples, written;
gint bpf, rate;
gint accum;
gint out_samples;
GstClockTime base_time, render_delay, latency;
GstClock *clock;
gboolean sync, slaved, align_next;
GstFlowReturn ret;
GstSegment clip_seg;
gint64 time_offset;
GstBuffer *out = NULL;
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sink = GST_AUDIO_BASE_SINK (bsink);
bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
ringbuf = sink->ringbuffer;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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/* can't do anything when we don't have the device */
if (G_UNLIKELY (!gst_audio_ring_buffer_is_acquired (ringbuf)))
goto wrong_state;
/* Wait for upstream latency before starting the ringbuffer, we do this so
* that we can align the first sample of the ringbuffer to the base_time +
* latency. */
GST_OBJECT_LOCK (sink);
base_time = GST_ELEMENT_CAST (sink)->base_time;
if (G_UNLIKELY (sink->priv->sync_latency)) {
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ret = gst_audio_base_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
GST_OBJECT_UNLOCK (sink);
if (G_UNLIKELY (ret != GST_FLOW_OK))
goto sync_latency_failed;
/* only do this once until we are set back to PLAYING */
sink->priv->sync_latency = FALSE;
} else {
GST_OBJECT_UNLOCK (sink);
}
/* Before we go on, let's see if we need to payload the data. If yes, we also
* need to unref the output buffer before leaving. */
if (bclass->payload) {
out = bclass->payload (sink, buf);
if (!out)
goto payload_failed;
buf = out;
}
bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
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size = gst_buffer_get_size (buf);
if (G_UNLIKELY (size % bpf) != 0)
goto wrong_size;
samples = size / bpf;
out_samples = samples;
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %"
GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset,
GST_TIME_ARGS (bsink->segment.start), samples);
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
2011-03-27 11:55:15 +00:00
offset = 0;
/* if not valid timestamp or we can't clip or sync, try to play
* sample ASAP */
if (!GST_CLOCK_TIME_IS_VALID (time)) {
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render_start = gst_audio_base_sink_get_offset (sink);
render_stop = render_start + samples;
GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
" Using render_start=%" G_GUINT64_FORMAT, size, render_start);
/* we don't have a start so we don't know stop either */
stop = -1;
goto no_align;
}
/* let's calc stop based on the number of samples in the buffer instead
* of trusting the DURATION */
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND, rate);
/* prepare the clipping segment. Since we will be subtracting ts-offset and
* device-delay later we scale the start and stop with those values so that we
* can correctly clip them */
clip_seg.format = GST_FORMAT_TIME;
clip_seg.start = bsink->segment.start;
clip_seg.stop = bsink->segment.stop;
clip_seg.duration = -1;
/* the sync offset is the combination of ts-offset and device-delay */
latency = gst_base_sink_get_latency (bsink);
ts_offset = gst_base_sink_get_ts_offset (bsink);
render_delay = gst_base_sink_get_render_delay (bsink);
sync_offset = ts_offset - render_delay + latency;
GST_DEBUG_OBJECT (sink,
"sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
", ts-offset %" G_GINT64_FORMAT, sync_offset,
GST_TIME_ARGS (render_delay), ts_offset);
/* compensate for ts-offset and device-delay when negative we need to
* clip. */
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if (G_UNLIKELY (sync_offset < 0)) {
clip_seg.start += -sync_offset;
if (clip_seg.stop != -1)
clip_seg.stop += -sync_offset;
}
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
/* samples should be rendered based on their timestamp. All samples
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* boundaries */
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if (G_UNLIKELY (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop,
&ctime, &cstop)))
goto out_of_segment;
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
/* see if some clipping happened */
diff = ctime - time;
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if (G_UNLIKELY (diff > 0)) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
samples -= diff;
offset += diff * bpf;
time = ctime;
}
diff = stop - cstop;
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if (G_UNLIKELY (diff > 0)) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
samples -= diff;
stop = cstop;
}
/* figure out how to sync */
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if (G_LIKELY ((clock = GST_ELEMENT_CLOCK (bsink))))
sync = bsink->sync;
else
sync = FALSE;
2012-03-09 16:15:38 +00:00
if (G_UNLIKELY (!sync)) {
/* no sync needed, play sample ASAP */
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render_start = gst_audio_base_sink_get_offset (sink);
render_stop = render_start + samples;
GST_DEBUG_OBJECT (sink,
"no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
goto no_align;
}
/* bring buffer start and stop times to running time */
render_start =
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
render_stop =
gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
GST_DEBUG_OBJECT (sink,
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* store the time of the last sample, we'll use this to perform sync on the
* last sample when draining the buffer */
2012-03-09 16:15:38 +00:00
if (G_LIKELY (bsink->segment.rate >= 0.0)) {
sink->priv->eos_time = render_stop;
} else {
sink->priv->eos_time = render_start;
}
2012-03-09 16:15:38 +00:00
if (G_UNLIKELY (sync_offset != 0)) {
/* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */
GST_DEBUG_OBJECT (sink,
"compensating for sync-offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (sync_offset));
render_start += sync_offset;
render_stop += sync_offset;
}
2012-03-09 16:15:38 +00:00
if (base_time != 0) {
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
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/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
}
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if (G_UNLIKELY ((slaved = (clock != sink->provided_clock)))) {
/* handle clock slaving */
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_handle_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
} else {
/* no slaving needed but we need to adapt to the clock calibration
* parameters */
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
}
GST_DEBUG_OBJECT (sink,
"final timestamps: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
/* bring to position in the ringbuffer */
2011-06-21 16:17:59 +00:00
time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
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if (G_UNLIKELY (time_offset != 0)) {
GST_DEBUG_OBJECT (sink,
"apply time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
if (render_start > time_offset)
render_start -= time_offset;
else
render_start = 0;
if (render_stop > time_offset)
render_stop -= time_offset;
else
render_stop = 0;
}
/* in some clock slaving cases, all late samples end up at 0 first,
* and subsequent ones align with that until threshold exceeded,
* and then sync back to 0 and so on, so avoid that altogether */
if (G_UNLIKELY (render_start == 0 && render_stop == 0))
goto too_late;
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start, rate, GST_SECOND);
render_stop = gst_util_uint64_scale_int (render_stop, rate, GST_SECOND);
/* positive playback rate, first sample is render_start, negative rate, first
* sample is render_stop. When no rate conversion is active, render exactly
* the amount of input samples to avoid aligning to rounding errors. */
2012-03-09 16:15:38 +00:00
if (G_LIKELY (bsink->segment.rate >= 0.0)) {
sample_offset = render_start;
2012-03-09 16:15:38 +00:00
if (G_LIKELY (bsink->segment.rate == 1.0))
render_stop = sample_offset + samples;
} else {
sample_offset = render_stop;
if (bsink->segment.rate == -1.0)
render_start = sample_offset + samples;
}
/* always resync after a discont */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
GST_DEBUG_OBJECT (sink, "resync after discont");
goto no_align;
}
/* resync when we don't know what to align the sample with */
if (G_UNLIKELY (sink->next_sample == -1)) {
GST_DEBUG_OBJECT (sink,
"no align possible: no previous sample position known");
goto no_align;
}
2011-11-11 10:52:47 +00:00
align = gst_audio_base_sink_get_alignment (sink, sample_offset);
sink->priv->last_align = align;
/* apply alignment */
render_start += align;
/* only align stop if we are not slaved to resample */
2012-03-09 16:15:38 +00:00
if (G_UNLIKELY (slaved
&& sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE)) {
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
goto no_align;
}
render_stop += align;
no_align:
/* number of target samples is difference between start and stop */
out_samples = render_stop - render_start;
/* we render the first or last sample first, depending on the rate */
2012-03-09 16:15:38 +00:00
if (G_LIKELY (bsink->segment.rate >= 0.0))
sample_offset = render_start;
else
sample_offset = render_stop;
GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
sample_offset, samples, out_samples);
/* we need to accumulate over different runs for when we get interrupted */
accum = 0;
align_next = TRUE;
2012-01-20 15:11:54 +00:00
gst_buffer_map (buf, &info, GST_MAP_READ);
do {
written =
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gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
2012-01-20 15:11:54 +00:00
info.data + offset, samples, out_samples, &accum);
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
2012-03-09 16:15:38 +00:00
if (G_LIKELY (written == samples))
break;
/* else something interrupted us and we wait for preroll. */
if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
goto stopping;
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
/* update the output samples. FIXME, this will just skip them when pausing
* during trick mode */
if (out_samples > written) {
out_samples -= written;
accum = 0;
} else
break;
samples -= written;
offset += written * bpf;
} while (TRUE);
2012-01-20 15:11:54 +00:00
gst_buffer_unmap (buf, &info);
2012-03-09 16:15:38 +00:00
if (G_LIKELY (align_next))
sink->next_sample = sample_offset;
else
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
sink->next_sample);
2012-03-09 16:15:38 +00:00
if (G_UNLIKELY (GST_CLOCK_TIME_IS_VALID (stop)
&& stop >= bsink->segment.stop)) {
GST_DEBUG_OBJECT (sink,
"start playback because we are at the end of segment");
gst_audio_ring_buffer_start (ringbuf);
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
ret = GST_FLOW_OK;
done:
if (out)
gst_buffer_unref (out);
return ret;
/* SPECIAL cases */
out_of_segment:
{
GST_DEBUG_OBJECT (sink,
"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
GST_TIME_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (bsink->segment.start));
ret = GST_FLOW_OK;
goto done;
}
too_late:
{
GST_DEBUG_OBJECT (sink, "dropping late sample");
2011-12-13 12:55:45 +00:00
ret = GST_FLOW_OK;
goto done;
}
/* ERRORS */
payload_failed:
{
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("failed to payload."));
ret = GST_FLOW_ERROR;
goto done;
}
wrong_state:
{
GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
GST_ELEMENT_ERROR (sink, STREAM, FORMAT, (NULL), ("sink not negotiated."));
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
wrong_size:
{
GST_DEBUG_OBJECT (sink, "wrong size");
GST_ELEMENT_ERROR (sink, STREAM, WRONG_TYPE,
(NULL), ("sink received buffer of wrong size."));
ret = GST_FLOW_ERROR;
goto done;
}
stopping:
{
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GST_DEBUG_OBJECT (sink, "preroll got interrupted: %d (%s)", ret,
gst_flow_get_name (ret));
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gst_buffer_unmap (buf, &info);
goto done;
}
sync_latency_failed:
{
GST_DEBUG_OBJECT (sink, "failed waiting for latency");
goto done;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
}
/**
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* gst_audio_base_sink_create_ringbuffer:
* @sink: a #GstAudioBaseSink.
*
* Create and return the #GstAudioRingBuffer for @sink. This function will call the
* ::create_ringbuffer vmethod and will set @sink as the parent of the returned
* buffer (see gst_object_set_parent()).
*
* Returns: The new ringbuffer of @sink.
*/
GstAudioRingBuffer *
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gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
{
2011-11-11 10:52:47 +00:00
GstAudioBaseSinkClass *bclass;
GstAudioRingBuffer *buffer = NULL;
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
2011-11-11 10:52:47 +00:00
bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (sink);
if (buffer)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
return buffer;
}
static void
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
guint len, gpointer user_data)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
{
GstBaseSink *basesink;
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GstAudioBaseSink *sink;
GstBuffer *buf = NULL;
GstFlowReturn ret;
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gsize size;
basesink = GST_BASE_SINK (user_data);
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sink = GST_AUDIO_BASE_SINK (user_data);
GST_PAD_STREAM_LOCK (basesink->sinkpad);
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
will copy twice, once into data, once into DMA */
GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
" to fill audio buffer", len, basesink->offset);
ret =
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gst_pad_pull_range (basesink->sinkpad, basesink->segment.position, len,
&buf);
if (ret != GST_FLOW_OK) {
if (ret == GST_FLOW_EOS)
goto eos;
else
goto error;
}
2011-03-04 16:25:46 +00:00
GST_BASE_SINK_PREROLL_LOCK (basesink);
if (basesink->flushing)
goto flushing;
/* complete preroll and wait for PLAYING */
ret = gst_base_sink_do_preroll (basesink, GST_MINI_OBJECT_CAST (buf));
if (ret != GST_FLOW_OK)
goto preroll_error;
2011-03-27 11:55:15 +00:00
size = gst_buffer_get_size (buf);
if (len != size) {
GST_INFO_OBJECT (basesink,
"got different size than requested from sink pad: %u"
" != %" G_GSIZE_FORMAT, len, size);
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len = MIN (size, len);
}
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basesink->segment.position += len;
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gst_buffer_extract (buf, 0, data, len);
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GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
error:
{
GST_WARNING_OBJECT (basesink, "Got flow '%s' but can't return it: %d",
gst_flow_get_name (ret), ret);
gst_audio_ring_buffer_pause (rbuf);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
eos:
{
/* FIXME: this is not quite correct; we'll be called endlessly until
* the sink gets shut down; maybe we should set a flag somewhere, or
* set segment.stop and segment.duration to the last sample or so */
GST_DEBUG_OBJECT (sink, "EOS");
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gst_audio_base_sink_drain (sink);
gst_audio_ring_buffer_pause (rbuf);
gst_element_post_message (GST_ELEMENT_CAST (sink),
gst_message_new_eos (GST_OBJECT_CAST (sink)));
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
}
flushing:
{
GST_DEBUG_OBJECT (sink, "we are flushing");
gst_audio_ring_buffer_pause (rbuf);
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GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
preroll_error:
{
GST_DEBUG_OBJECT (sink, "error %s", gst_flow_get_name (ret));
gst_audio_ring_buffer_pause (rbuf);
2011-03-04 16:25:46 +00:00
GST_BASE_SINK_PREROLL_UNLOCK (basesink);
GST_PAD_STREAM_UNLOCK (basesink->sinkpad);
return;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
}
static gboolean
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gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active)
{
gboolean ret;
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink);
if (active) {
GST_DEBUG_OBJECT (basesink, "activating pull");
gst_audio_ring_buffer_set_callback (sink->ringbuffer,
2011-11-11 10:52:47 +00:00
gst_audio_base_sink_callback, sink);
ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
} else {
GST_DEBUG_OBJECT (basesink, "deactivating pull");
gst_audio_ring_buffer_set_callback (sink->ringbuffer, NULL, NULL);
ret = gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
}
return ret;
}
#if 0
/* should be called with the LOCK */
static GstStateChangeReturn
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gst_audio_base_sink_async_play (GstBaseSink * basesink)
{
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GstAudioBaseSink *sink;
2011-11-11 10:52:47 +00:00
sink = GST_AUDIO_BASE_SINK (basesink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2011-11-18 11:37:10 +00:00
if (basesink->pad_mode == GST_PAD_MODE_PULL) {
/* we always start the ringbuffer in pull mode immediatly */
gst_audio_ring_buffer_start (sink->ringbuffer);
}
return GST_STATE_CHANGE_SUCCESS;
}
#endif
static GstStateChangeReturn
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gst_audio_base_sink_change_state (GstElement * element,
GstStateChange transition)
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
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sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink);
}
if (!gst_audio_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
sink->next_sample = -1;
sink->priv->last_align = -1;
sink->priv->eos_time = -1;
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.
2011-11-07 10:31:47 +00:00
sink->priv->discont_time = -1;
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2011-11-11 10:52:47 +00:00
(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
sink->provided_clock, TRUE));
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
gboolean eos;
GST_OBJECT_LOCK (sink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
eos = GST_BASE_SINK (sink)->eos;
GST_OBJECT_UNLOCK (sink);
gst_audio_ring_buffer_may_start (sink->ringbuffer, TRUE);
2011-11-18 11:37:10 +00:00
if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_PAD_MODE_PULL ||
2011-06-21 16:17:59 +00:00
g_atomic_int_get (&sink->eos_rendering) || eos) {
/* we always start the ringbuffer in pull mode immediatly */
/* sync rendering on eos needs running clock,
* and others need running clock when finished rendering eos */
gst_audio_ring_buffer_start (sink->ringbuffer);
}
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* ringbuffer cannot start anymore */
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_pause (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
sink->priv->sync_latency = FALSE;
GST_OBJECT_UNLOCK (sink);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* Only post clock-lost messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
2011-11-11 10:52:47 +00:00
(GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
sink->provided_clock));
/* make sure we unblock before calling the parent state change
* so it can grab the STREAM_LOCK */
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop slaving ourselves to the master, if any */
gst_clock_set_master (sink->provided_clock, NULL);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
2005-04-20 10:19:54 +00:00
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_release (sink->ringbuffer);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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break;
case GST_STATE_CHANGE_READY_TO_NULL:
/* we release again here because the aqcuire happens when setting the
* caps, which happens before we commit the state to PAUSED and thus the
* PAUSED->READY state change (see above, where we release the ringbuffer)
* might not be called when we get here. */
gst_audio_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_release (sink->ringbuffer);
gst_audio_ring_buffer_close_device (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
GST_OBJECT_UNLOCK (sink);
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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break;
default:
break;
}
return ret;
/* ERRORS */
open_failed:
{
/* subclass must post a meaningful error message */
GST_DEBUG_OBJECT (sink, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
An attempt at a set of audio base classes together with some design docs. Original commit message from CVS: * docs/design-audiosinks.txt: * gst-libs/gst/audio/Makefile.am: * gst-libs/gst/audio/TODO: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init), (audioringbuffer_thread_func), (gst_audioringbuffer_init), (gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire), (gst_audioringbuffer_release), (gst_audioringbuffer_play), (gst_audioringbuffer_stop), (gst_audioringbuffer_delay), (gst_audiosink_base_init), (gst_audiosink_class_init), (gst_audiosink_init), (gst_audiosink_create_ringbuffer): * gst-libs/gst/audio/gstaudiosink.h: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init), (gst_baseaudiosink_init), (gst_baseaudiosink_set_property), (gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times), (gst_baseaudiosink_event), (gst_baseaudiosink_preroll), (gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer), (gst_baseaudiosink_callback), (gst_baseaudiosink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: * gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type), (gst_ringbuffer_class_init), (gst_ringbuffer_init), (gst_ringbuffer_dispose), (gst_ringbuffer_finalize), (gst_ringbuffer_set_callback), (gst_ringbuffer_acquire), (gst_ringbuffer_release), (gst_ringbuffer_play_unlocked), (gst_ringbuffer_play), (gst_ringbuffer_pause), (gst_ringbuffer_resume), (gst_ringbuffer_stop), (gst_ringbuffer_callback), (gst_ringbuffer_delay), (gst_ringbuffer_played_samples), (gst_ringbuffer_commit), (gst_ringbuffer_prepare_read), (gst_ringbuffer_clear): * gst-libs/gst/audio/gstringbuffer.h: An attempt at a set of audio base classes together with some design docs.
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}