<SECTION>
<FILE>gstbaseaudiosink</FILE>
<INCLUDE>gst/audio/gstbaseaudiosink.h</INCLUDE>
-GstBaseAudioSink
-GstBaseAudioSinkClass
-GstBaseAudioSinkSlaveMethod
-
-GST_BASE_AUDIO_SINK_CLOCK
-GST_BASE_AUDIO_SINK_PAD
-gst_base_audio_sink_create_ringbuffer
-gst_base_audio_sink_set_provide_clock
-gst_base_audio_sink_get_provide_clock
-gst_base_audio_sink_set_slave_method
-gst_base_audio_sink_get_slave_method
-gst_base_audio_sink_get_drift_tolerance
-gst_base_audio_sink_set_drift_tolerance
+GstAudioBaseSink
+GstAudioBaseSinkClass
+GstAudioBaseSinkSlaveMethod
+
+GST_AUDIO_BASE_SINK_CLOCK
+GST_AUDIO_BASE_SINK_PAD
+gst_audio_base_sink_create_ringbuffer
+gst_audio_base_sink_set_provide_clock
+gst_audio_base_sink_get_provide_clock
+gst_audio_base_sink_set_slave_method
+gst_audio_base_sink_get_slave_method
+gst_audio_base_sink_get_drift_tolerance
+gst_audio_base_sink_set_drift_tolerance
<SUBSECTION Standard>
-GST_BASE_AUDIO_SINK
-GST_IS_BASE_AUDIO_SINK
-GST_TYPE_BASE_AUDIO_SINK
-gst_base_audio_sink_get_type
-GST_BASE_AUDIO_SINK_CLASS
-GST_IS_BASE_AUDIO_SINK_CLASS
-GST_BASE_AUDIO_SINK_GET_CLASS
-GstBaseAudioSinkPrivate
-gst_base_audio_sink_slave_method_get_type
-GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD
+GST_AUDIO_BASE_SINK
+GST_IS_AUDIO_BASE_SINK
+GST_TYPE_AUDIO_BASE_SINK
+gst_audio_base_sink_get_type
+GST_AUDIO_BASE_SINK_CLASS
+GST_IS_AUDIO_BASE_SINK_CLASS
+GST_AUDIO_BASE_SINK_GET_CLASS
+GstAudioBaseSinkPrivate
+gst_audio_base_sink_slave_method_get_type
+GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD
</SECTION>
<SECTION>
<FILE>gstbaseaudiosrc</FILE>
<INCLUDE>gst/audio/gstbaseaudiosrc.h</INCLUDE>
-GstBaseAudioSrc
-GstBaseAudioSrcClass
-GstBaseAudioSrcSlaveMethod
-
-GST_BASE_AUDIO_SRC_CLOCK
-GST_BASE_AUDIO_SRC_PAD
-gst_base_audio_src_create_ringbuffer
-gst_base_audio_src_set_provide_clock
-gst_base_audio_src_get_provide_clock
-gst_base_audio_src_get_slave_method
-gst_base_audio_src_set_slave_method
+GstAudioBaseSrc
+GstAudioBaseSrcClass
+GstAudioBaseSrcSlaveMethod
+
+GST_AUDIO_BASE_SRC_CLOCK
+GST_AUDIO_BASE_SRC_PAD
+gst_audio_base_src_create_ringbuffer
+gst_audio_base_src_set_provide_clock
+gst_audio_base_src_get_provide_clock
+gst_audio_base_src_get_slave_method
+gst_audio_base_src_set_slave_method
<SUBSECTION Standard>
-GstBaseAudioSrcPrivate
-GST_BASE_AUDIO_SRC
-GST_IS_BASE_AUDIO_SRC
-GST_TYPE_BASE_AUDIO_SRC
-gst_base_audio_src_get_type
-GST_BASE_AUDIO_SRC_CLASS
-GST_IS_BASE_AUDIO_SRC_CLASS
-GST_BASE_AUDIO_SRC_GET_CLASS
-gst_base_audio_src_slave_method_get_type
-GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD
+GstAudioBaseSrcPrivate
+GST_AUDIO_BASE_SRC
+GST_IS_AUDIO_BASE_SRC
+GST_TYPE_AUDIO_BASE_SRC
+gst_audio_base_src_get_type
+GST_AUDIO_BASE_SRC_CLASS
+GST_IS_AUDIO_BASE_SRC_CLASS
+GST_AUDIO_BASE_SRC_GET_CLASS
+gst_audio_base_src_slave_method_get_type
+GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD
</SECTION>
<SECTION>
#include <gst/audio/gstaudiosrc.h>
gst_audio_src_get_type
#include <gst/audio/gstbaseaudiosink.h>
-gst_base_audio_sink_get_type
+gst_audio_base_sink_get_type
#include <gst/audio/gstbaseaudiosrc.h>
-gst_base_audio_src_get_type
+gst_audio_base_src_get_type
#include <gst/audio/gstaudioringbuffer.h>
gst_audio_ring_buffer_get_type
gst_alsasrc_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);
GstAlsaSrc *asrc = GST_ALSA_SRC (element);
GstClock *clk;
/**
* SECTION:gstaudioclock
* @short_description: Helper object for implementing audio clocks
- * @see_also: #GstBaseAudioSink, #GstSystemClock
+ * @see_also: #GstAudioBaseSink, #GstSystemClock
*
* #GstAudioClock makes it easy for elements to implement a #GstClock, they
* simply need to provide a function that returns the current clock time.
*
- * This object is internally used to implement the clock in #GstBaseAudioSink.
+ * This object is internally used to implement the clock in #GstAudioBaseSink.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
/**
* SECTION:gstaudioringbuffer
* @short_description: Base class for audio ringbuffer implementations
- * @see_also: #GstBaseAudioSink, #GstAudioSink
+ * @see_also: #GstAudioBaseSink, #GstAudioSink
*
* <refsect2>
* <para>
/**
* SECTION:gstaudiosink
* @short_description: Simple base class for audio sinks
- * @see_also: #GstBaseAudioSink, #GstAudioRingBuffer, #GstAudioSink.
+ * @see_also: #GstAudioBaseSink, #GstAudioRingBuffer, #GstAudioSink.
*
* This is the most simple base class for audio sinks that only requires
* subclasses to implement a set of simple functions:
* </variablelist>
*
* All scheduling of samples and timestamps is done in this base class
- * together with #GstBaseAudioSink using a default implementation of a
+ * together with #GstAudioBaseSink using a default implementation of a
* #GstAudioRingBuffer that uses threads.
*
* Last reviewed on 2006-09-27 (0.10.12)
GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
#define gst_audio_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioSink, gst_audio_sink,
- GST_TYPE_BASE_AUDIO_SINK, _do_init);
+ GST_TYPE_AUDIO_BASE_SINK, _do_init);
-static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
+static GstAudioRingBuffer *gst_audio_sink_create_ringbuffer (GstAudioBaseSink *
sink);
static void
gst_audio_sink_class_init (GstAudioSinkClass * klass)
{
- GstBaseAudioSinkClass *gstbaseaudiosink_class;
+ GstAudioBaseSinkClass *gstbaseaudiosink_class;
- gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
+ gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
}
static GstAudioRingBuffer *
-gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
+gst_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
GstAudioRingBuffer *buffer;
* Opaque #GstAudioSink.
*/
struct _GstAudioSink {
- GstBaseAudioSink element;
+ GstAudioBaseSink element;
/*< private >*/ /* with LOCK */
GThread *thread;
* #GstAudioSink class. Override the vmethods to implement functionality.
*/
struct _GstAudioSinkClass {
- GstBaseAudioSinkClass parent_class;
+ GstAudioBaseSinkClass parent_class;
/* vtable */
/**
* SECTION:gstaudiosrc
* @short_description: Simple base class for audio sources
- * @see_also: #GstBaseAudioSrc, #GstAudioRingBuffer, #GstAudioSrc.
+ * @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer, #GstAudioSrc.
*
* This is the most simple base class for audio sources that only requires
* subclasses to implement a set of simple functions:
* </variablelist>
*
* All scheduling of samples and timestamps is done in this base class
- * together with #GstBaseAudioSrc using a default implementation of a
+ * together with #GstAudioBaseSrc using a default implementation of a
* #GstAudioRingBuffer that uses threads.
*
* Last reviewed on 2006-09-27 (0.10.12)
GST_DEBUG_CATEGORY_INIT (gst_audio_src_debug, "audiosrc", 0, "audiosrc element");
#define gst_audio_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioSrc, gst_audio_src,
- GST_TYPE_BASE_AUDIO_SRC, _do_init);
+ GST_TYPE_AUDIO_BASE_SRC, _do_init);
-static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstBaseAudioSrc *
+static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstAudioBaseSrc *
src);
static void
gst_audio_src_class_init (GstAudioSrcClass * klass)
{
- GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+ GstAudioBaseSrcClass *gstbaseaudiosrc_class;
- gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+ gstbaseaudiosrc_class = (GstAudioBaseSrcClass *) klass;
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer);
}
static GstAudioRingBuffer *
-gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
+gst_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
{
GstAudioRingBuffer *buffer;
* Base class for simple audio sources.
*/
struct _GstAudioSrc {
- GstBaseAudioSrc element;
+ GstAudioBaseSrc element;
/*< private >*/ /* with LOCK */
GThread *thread;
* functionality.
*/
struct _GstAudioSrcClass {
- GstBaseAudioSrcClass parent_class;
+ GstAudioBaseSrcClass parent_class;
/* vtable */
#include "gstbaseaudiosink.h"
-GST_DEBUG_CATEGORY_STATIC (gst_base_audio_sink_debug);
-#define GST_CAT_DEFAULT gst_base_audio_sink_debug
+GST_DEBUG_CATEGORY_STATIC (gst_audio_base_sink_debug);
+#define GST_CAT_DEFAULT gst_audio_base_sink_debug
-#define GST_BASE_AUDIO_SINK_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkPrivate))
+#define GST_AUDIO_BASE_SINK_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkPrivate))
-struct _GstBaseAudioSinkPrivate
+struct _GstAudioBaseSinkPrivate
{
/* upstream latency */
GstClockTime us_latency;
/* the clock slaving algorithm in use */
- GstBaseAudioSinkSlaveMethod slave_method;
+ GstAudioBaseSinkSlaveMethod slave_method;
/* running average of clock skew */
GstClockTimeDiff avg_skew;
/* the number of samples we aligned last time */
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_PROVIDE_CLOCK TRUE
-#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SINK_SLAVE_SKEW
+#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
};
GType
-gst_base_audio_sink_slave_method_get_type (void)
+gst_audio_base_sink_slave_method_get_type (void)
{
static volatile gsize slave_method_type = 0;
static const GEnumValue slave_method[] = {
- {GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE, "GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE",
+ {GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE, "GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE",
"resample"},
- {GST_BASE_AUDIO_SINK_SLAVE_SKEW, "GST_BASE_AUDIO_SINK_SLAVE_SKEW", "skew"},
- {GST_BASE_AUDIO_SINK_SLAVE_NONE, "GST_BASE_AUDIO_SINK_SLAVE_NONE", "none"},
+ {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 =
- g_enum_register_static ("GstBaseAudioSinkSlaveMethod", slave_method);
+ g_enum_register_static ("GstAudioBaseSinkSlaveMethod", slave_method);
g_once_init_leave (&slave_method_type, tmp);
}
#define _do_init \
- GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
-#define gst_base_audio_sink_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSink, gst_base_audio_sink,
+ GST_DEBUG_CATEGORY_INIT (gst_audio_base_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
+#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);
-static void gst_base_audio_sink_dispose (GObject * object);
+static void gst_audio_base_sink_dispose (GObject * object);
-static void gst_base_audio_sink_set_property (GObject * object, guint prop_id,
+static void gst_audio_base_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void gst_base_audio_sink_get_property (GObject * object, guint prop_id,
+static void gst_audio_base_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
#if 0
-static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
+static GstStateChangeReturn gst_audio_base_sink_async_play (GstBaseSink *
basesink);
#endif
-static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
+static GstStateChangeReturn gst_audio_base_sink_change_state (GstElement *
element, GstStateChange transition);
-static gboolean gst_base_audio_sink_activate_pull (GstBaseSink * basesink,
+static gboolean gst_audio_base_sink_activate_pull (GstBaseSink * basesink,
gboolean active);
-static gboolean gst_base_audio_sink_query (GstElement * element, GstQuery *
+static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
query);
-static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
-static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
- GstBaseAudioSink * sink);
-static void gst_base_audio_sink_callback (GstAudioRingBuffer * rbuf,
+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);
-static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink,
+static GstFlowReturn gst_audio_base_sink_preroll (GstBaseSink * bsink,
GstBuffer * buffer);
-static GstFlowReturn gst_base_audio_sink_render (GstBaseSink * bsink,
+static GstFlowReturn gst_audio_base_sink_render (GstBaseSink * bsink,
GstBuffer * buffer);
-static gboolean gst_base_audio_sink_event (GstBaseSink * bsink,
+static gboolean gst_audio_base_sink_event (GstBaseSink * bsink,
GstEvent * event);
-static void gst_base_audio_sink_get_times (GstBaseSink * bsink,
+static void gst_audio_base_sink_get_times (GstBaseSink * bsink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
-static gboolean gst_base_audio_sink_setcaps (GstBaseSink * bsink,
+static gboolean gst_audio_base_sink_setcaps (GstBaseSink * bsink,
GstCaps * caps);
-static void gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
+static void gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps);
-static gboolean gst_base_audio_sink_query_pad (GstBaseSink * bsink,
+static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
GstQuery * query);
-/* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
+/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
static void
-gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
+gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
- g_type_class_add_private (klass, sizeof (GstBaseAudioSinkPrivate));
+ g_type_class_add_private (klass, sizeof (GstAudioBaseSinkPrivate));
- gobject_class->set_property = gst_base_audio_sink_set_property;
- gobject_class->get_property = gst_base_audio_sink_get_property;
- gobject_class->dispose = gst_base_audio_sink_dispose;
+ 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;
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
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",
- GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
+ GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CAN_ACTIVATE_PULL,
"Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseAudioSink:drift-tolerance
+ * GstAudioBaseSink:drift-tolerance
*
* Controls the amount of time in microseconds that clocks are allowed
* to drift before resynchronisation happens.
G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseAudioSink:alignment_threshold
+ * 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.
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseAudioSink:discont-wait
+ * GstAudioBaseSink:discont-wait
*
* A window of time in nanoseconds to wait before creating a discontinuity as
* a result of breaching the drift-tolerance.
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
+ GST_DEBUG_FUNCPTR (gst_audio_base_sink_change_state);
gstelement_class->provide_clock =
- GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
- gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query);
+ GST_DEBUG_FUNCPTR (gst_audio_base_sink_provide_clock);
+ gstelement_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_sink_query);
- gstbasesink_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_sink_fixate);
- gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
- gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
+ 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->get_times =
- GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
- gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
- gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_base_audio_sink_render);
- gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad);
+ 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 =
- GST_DEBUG_FUNCPTR (gst_base_audio_sink_activate_pull);
+ 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 */
}
static void
-gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink)
+gst_audio_base_sink_init (GstAudioBaseSink * baseaudiosink)
{
GstBaseSink *basesink;
- baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
+ baseaudiosink->priv = GST_AUDIO_BASE_SINK_GET_PRIVATE (baseaudiosink);
baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosink->priv->discont_wait = DEFAULT_DISCONT_WAIT;
baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
- (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink,
+ (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time, baseaudiosink,
NULL);
basesink = GST_BASE_SINK_CAST (baseaudiosink);
}
static void
-gst_base_audio_sink_dispose (GObject * object)
+gst_audio_base_sink_dispose (GObject * object)
{
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
- sink = GST_BASE_AUDIO_SINK (object);
+ sink = GST_AUDIO_BASE_SINK (object);
if (sink->provided_clock) {
gst_audio_clock_invalidate (sink->provided_clock);
static GstClock *
-gst_base_audio_sink_provide_clock (GstElement * elem)
+gst_audio_base_sink_provide_clock (GstElement * elem)
{
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
GstClock *clock;
- sink = GST_BASE_AUDIO_SINK (elem);
+ sink = GST_AUDIO_BASE_SINK (elem);
/* we have no ringbuffer (must be NULL state) */
if (sink->ringbuffer == NULL)
}
static gboolean
-gst_base_audio_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
+gst_audio_base_sink_query_pad (GstBaseSink * bsink, GstQuery * query)
{
gboolean res = FALSE;
- GstBaseAudioSink *basesink;
+ GstAudioBaseSink *basesink;
- basesink = GST_BASE_AUDIO_SINK (bsink);
+ basesink = GST_AUDIO_BASE_SINK (bsink);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
}
static gboolean
-gst_base_audio_sink_query (GstElement * element, GstQuery * query)
+gst_audio_base_sink_query (GstElement * element, GstQuery * query)
{
gboolean res = FALSE;
- GstBaseAudioSink *basesink;
+ GstAudioBaseSink *basesink;
- basesink = GST_BASE_AUDIO_SINK (element);
+ basesink = GST_AUDIO_BASE_SINK (element);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
static GstClockTime
-gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
+gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
{
guint64 raw, samples;
guint delay;
}
/**
- * gst_base_audio_sink_set_provide_clock:
- * @sink: a #GstBaseAudioSink
+ * 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,
* Since: 0.10.16
*/
void
-gst_base_audio_sink_set_provide_clock (GstBaseAudioSink * sink,
+gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
gboolean provide)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->provide_clock = provide;
}
/**
- * gst_base_audio_sink_get_provide_clock:
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_get_provide_clock:
+ * @sink: a #GstAudioBaseSink
*
* Queries whether @sink will provide a clock or not. See also
- * gst_base_audio_sink_set_provide_clock.
+ * gst_audio_base_sink_set_provide_clock.
*
* Returns: %TRUE if @sink will provide a clock.
*
* Since: 0.10.16
*/
gboolean
-gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink)
+gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
{
gboolean result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), FALSE);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
GST_OBJECT_LOCK (sink);
result = sink->provide_clock;
}
/**
- * gst_base_audio_sink_set_slave_method:
- * @sink: a #GstBaseAudioSink
+ * 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
-gst_base_audio_sink_set_slave_method (GstBaseAudioSink * sink,
- GstBaseAudioSinkSlaveMethod method)
+gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
+ GstAudioBaseSinkSlaveMethod method)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->slave_method = method;
}
/**
- * gst_base_audio_sink_get_slave_method:
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_get_slave_method:
+ * @sink: a #GstAudioBaseSink
*
* Get the current slave method used by @sink.
*
*
* Since: 0.10.16
*/
-GstBaseAudioSinkSlaveMethod
-gst_base_audio_sink_get_slave_method (GstBaseAudioSink * sink)
+GstAudioBaseSinkSlaveMethod
+gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
{
- GstBaseAudioSinkSlaveMethod result;
+ GstAudioBaseSinkSlaveMethod result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->slave_method;
/**
- * gst_base_audio_sink_set_drift_tolerance:
- * @sink: a #GstBaseAudioSink
+ * 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
-gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink * sink,
+gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
gint64 drift_tolerance)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->drift_tolerance = drift_tolerance;
}
/**
- * gst_base_audio_sink_get_drift_tolerance
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_get_drift_tolerance
+ * @sink: a #GstAudioBaseSink
*
* Get the current drift tolerance, in microseconds, used by @sink.
*
* Since: 0.10.31
*/
gint64
-gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink * sink)
+gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
{
gint64 result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->drift_tolerance;
}
/**
- * gst_base_audio_sink_set_alignment_threshold:
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_set_alignment_threshold:
+ * @sink: a #GstAudioBaseSink
* @alignment_threshold: the new alignment threshold in nanoseconds
*
* Controls the sink's alignment threshold.
* Since: 0.10.36
*/
void
-gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink,
+gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
GstClockTime alignment_threshold)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->alignment_threshold = alignment_threshold;
}
/**
- * gst_base_audio_sink_get_alignment_threshold
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_get_alignment_threshold
+ * @sink: a #GstAudioBaseSink
*
* Get the current alignment threshold, in nanoseconds, used by @sink.
*
* Since: 0.10.36
*/
GstClockTime
-gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink)
+gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
{
gint64 result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->alignment_threshold;
}
/**
- * gst_base_audio_sink_set_discont_wait:
- * @sink: a #GstBaseAudioSink
+ * 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.
* Since: 0.10.36
*/
void
-gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink,
+gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
GstClockTime discont_wait)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SINK (sink));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink);
sink->priv->discont_wait = discont_wait;
}
/**
- * gst_base_audio_sink_get_discont_wait
- * @sink: a #GstBaseAudioSink
+ * gst_audio_base_sink_get_discont_wait
+ * @sink: a #GstAudioBaseSink
*
* Get the current discont wait, in nanoseconds, used by @sink.
*
* Since: 0.10.36
*/
GstClockTime
-gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink)
+gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
{
GstClockTime result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SINK (sink), -1);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), -1);
GST_OBJECT_LOCK (sink);
result = sink->priv->discont_wait;
}
static void
-gst_base_audio_sink_set_property (GObject * object, guint prop_id,
+gst_audio_base_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
- sink = GST_BASE_AUDIO_SINK (object);
+ sink = GST_AUDIO_BASE_SINK (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
sink->latency_time = g_value_get_int64 (value);
break;
case PROP_PROVIDE_CLOCK:
- gst_base_audio_sink_set_provide_clock (sink, g_value_get_boolean (value));
+ gst_audio_base_sink_set_provide_clock (sink, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
- gst_base_audio_sink_set_slave_method (sink, g_value_get_enum (value));
+ 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:
- gst_base_audio_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
+ gst_audio_base_sink_set_drift_tolerance (sink, g_value_get_int64 (value));
break;
case PROP_ALIGNMENT_THRESHOLD:
- gst_base_audio_sink_set_alignment_threshold (sink,
+ gst_audio_base_sink_set_alignment_threshold (sink,
g_value_get_uint64 (value));
break;
case PROP_DISCONT_WAIT:
- gst_base_audio_sink_set_discont_wait (sink, g_value_get_uint64 (value));
+ gst_audio_base_sink_set_discont_wait (sink, g_value_get_uint64 (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-gst_base_audio_sink_get_property (GObject * object, guint prop_id,
+gst_audio_base_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
- sink = GST_BASE_AUDIO_SINK (object);
+ sink = GST_AUDIO_BASE_SINK (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, sink->latency_time);
break;
case PROP_PROVIDE_CLOCK:
- g_value_set_boolean (value, gst_base_audio_sink_get_provide_clock (sink));
+ g_value_set_boolean (value, gst_audio_base_sink_get_provide_clock (sink));
break;
case PROP_SLAVE_METHOD:
- g_value_set_enum (value, gst_base_audio_sink_get_slave_method (sink));
+ 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:
- g_value_set_int64 (value, gst_base_audio_sink_get_drift_tolerance (sink));
+ g_value_set_int64 (value, gst_audio_base_sink_get_drift_tolerance (sink));
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value,
- gst_base_audio_sink_get_alignment_threshold (sink));
+ gst_audio_base_sink_get_alignment_threshold (sink));
break;
case PROP_DISCONT_WAIT:
- g_value_set_uint64 (value, gst_base_audio_sink_get_discont_wait (sink));
+ g_value_set_uint64 (value, gst_audio_base_sink_get_discont_wait (sink));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static gboolean
-gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
+gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
{
- GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+ GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
GstAudioRingBufferSpec *spec;
GstClockTime now;
GstClockTime crate_num, crate_denom;
}
static void
-gst_base_audio_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
+gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
{
GstStructure *s;
gint width, depth;
}
static void
-gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
+gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* our clock sync is a bit too much for the base class to handle so
/* This waits for the drain to happen and can be canceled */
static gboolean
-gst_base_audio_sink_drain (GstBaseAudioSink * sink)
+gst_audio_base_sink_drain (GstAudioBaseSink * sink)
{
if (!sink->ringbuffer)
return TRUE;
}
static gboolean
-gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
+gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
{
- GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+ GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
break;
case GST_EVENT_EOS:
/* now wait till we played everything */
- gst_base_audio_sink_drain (sink);
+ gst_audio_base_sink_drain (sink);
break;
default:
break;
}
static GstFlowReturn
-gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
+gst_audio_base_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
{
- GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
+ GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
if (!gst_audio_ring_buffer_is_acquired (sink->ringbuffer))
goto wrong_state;
}
static guint64
-gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
+gst_audio_base_sink_get_offset (GstAudioBaseSink * sink)
{
guint64 sample;
gint writeseg, segdone, sps;
/* algorithm to calculate sample positions that will result in resampling to
* match the clock rate of the master */
static void
-gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
+gst_audio_base_sink_resample_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
/* algorithm to calculate sample positions that will result in changing the
* playout pointer to match the clock rate of the master */
static void
-gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
+gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
/* apply the clock offset but do no slaving otherwise */
static void
-gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink,
+gst_audio_base_sink_none_slaving (GstAudioBaseSink * sink,
GstClockTime render_start, GstClockTime render_stop,
GstClockTime * srender_start, GstClockTime * srender_stop)
{
/* converts render_start and render_stop to their slaved values */
static void
-gst_base_audio_sink_handle_slaving (GstBaseAudioSink * sink,
+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) {
- case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
- gst_base_audio_sink_resample_slaving (sink, render_start, render_stop,
+ case GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE:
+ gst_audio_base_sink_resample_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
- case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
- gst_base_audio_sink_skew_slaving (sink, render_start, render_stop,
+ case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
+ gst_audio_base_sink_skew_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
- case GST_BASE_AUDIO_SINK_SLAVE_NONE:
- gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
+ case GST_AUDIO_BASE_SINK_SLAVE_NONE:
+ gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
srender_start, srender_stop);
break;
default:
/* must be called with LOCK */
static GstFlowReturn
-gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
+gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
{
GstClock *clock;
GstClockReturn status;
GstClockTime time, render_delay;
GstFlowReturn ret;
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
GstClockTime itime, etime;
GstClockTime rate_num, rate_denom;
GstClockTimeDiff jitter;
- sink = GST_BASE_AUDIO_SINK (bsink);
+ sink = GST_AUDIO_BASE_SINK (bsink);
clock = GST_ELEMENT_CLOCK (sink);
if (G_UNLIKELY (clock == NULL))
rate_num, rate_denom);
switch (sink->priv->slave_method) {
- case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
+ 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;
- case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
- case GST_BASE_AUDIO_SINK_SLAVE_NONE:
+ case GST_AUDIO_BASE_SINK_SLAVE_SKEW:
+ case GST_AUDIO_BASE_SINK_SLAVE_NONE:
default:
break;
}
}
static gint64
-gst_base_audio_sink_get_alignment (GstBaseAudioSink * sink,
+gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
GstClockTime sample_offset)
{
GstAudioRingBuffer *ringbuf = sink->ringbuffer;
}
static GstFlowReturn
-gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
+gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstClockTimeDiff sync_offset, ts_offset;
- GstBaseAudioSinkClass *bclass;
- GstBaseAudioSink *sink;
+ GstAudioBaseSinkClass *bclass;
+ GstAudioBaseSink *sink;
GstAudioRingBuffer *ringbuf;
gint64 diff, align;
guint64 ctime, cstop;
gint64 time_offset;
GstBuffer *out = NULL;
- sink = GST_BASE_AUDIO_SINK (bsink);
- bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
+ sink = GST_AUDIO_BASE_SINK (bsink);
+ bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
ringbuf = sink->ringbuffer;
GST_OBJECT_LOCK (sink);
base_time = GST_ELEMENT_CAST (sink)->base_time;
if (G_UNLIKELY (sink->priv->sync_latency)) {
- ret = gst_base_audio_sink_sync_latency (bsink, GST_MINI_OBJECT_CAST (buf));
+ 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;
/* if not valid timestamp or we can't clip or sync, try to play
* sample ASAP */
if (!GST_CLOCK_TIME_IS_VALID (time)) {
- render_start = gst_base_audio_sink_get_offset (sink);
+ 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);
if (!sync) {
/* no sync needed, play sample ASAP */
- render_start = gst_base_audio_sink_get_offset (sink);
+ 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);
if ((slaved = clock != sink->provided_clock)) {
/* handle clock slaving */
- gst_base_audio_sink_handle_slaving (sink, render_start, render_stop,
+ 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 */
- gst_base_audio_sink_none_slaving (sink, render_start, render_stop,
+ gst_audio_base_sink_none_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
}
goto no_align;
}
- align = gst_base_audio_sink_get_alignment (sink, sample_offset);
+ 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 */
- if (slaved && sink->priv->slave_method == GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE) {
+ if (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;
}
}
/**
- * gst_base_audio_sink_create_ringbuffer:
- * @sink: a #GstBaseAudioSink.
+ * 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
* Returns: The new ringbuffer of @sink.
*/
GstAudioRingBuffer *
-gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
+gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
- GstBaseAudioSinkClass *bclass;
+ GstAudioBaseSinkClass *bclass;
GstAudioRingBuffer *buffer = NULL;
- bclass = GST_BASE_AUDIO_SINK_GET_CLASS (sink);
+ bclass = GST_AUDIO_BASE_SINK_GET_CLASS (sink);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (sink);
}
static void
-gst_base_audio_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
+gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
guint len, gpointer user_data)
{
GstBaseSink *basesink;
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
GstBuffer *buf;
GstFlowReturn ret;
gsize size;
basesink = GST_BASE_SINK (user_data);
- sink = GST_BASE_AUDIO_SINK (user_data);
+ sink = GST_AUDIO_BASE_SINK (user_data);
GST_PAD_STREAM_LOCK (basesink->sinkpad);
* 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");
- gst_base_audio_sink_drain (sink);
+ 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)));
}
static gboolean
-gst_base_audio_sink_activate_pull (GstBaseSink * basesink, gboolean active)
+gst_audio_base_sink_activate_pull (GstBaseSink * basesink, gboolean active)
{
gboolean ret;
- GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (basesink);
+ GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (basesink);
if (active) {
GST_DEBUG_OBJECT (basesink, "activating pull");
gst_audio_ring_buffer_set_callback (sink->ringbuffer,
- gst_base_audio_sink_callback, sink);
+ gst_audio_base_sink_callback, sink);
ret = gst_audio_ring_buffer_activate (sink->ringbuffer, TRUE);
} else {
#if 0
/* should be called with the LOCK */
static GstStateChangeReturn
-gst_base_audio_sink_async_play (GstBaseSink * basesink)
+gst_audio_base_sink_async_play (GstBaseSink * basesink)
{
- GstBaseAudioSink *sink;
+ GstAudioBaseSink *sink;
- sink = GST_BASE_AUDIO_SINK (basesink);
+ sink = GST_AUDIO_BASE_SINK (basesink);
GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
sink->priv->sync_latency = TRUE;
#endif
static GstStateChangeReturn
-gst_base_audio_sink_change_state (GstElement * element,
+gst_audio_base_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (element);
+ GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
- sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
+ sink->ringbuffer = gst_audio_base_sink_create_ringbuffer (sink);
}
if (!gst_audio_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
* 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 ==
- (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
+ (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));
* 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 ==
- (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
+ (GstAudioClockGetTimeFunc) gst_audio_base_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
sink->provided_clock));
* the methods in GstBaseSink and this class.
*/
-#ifndef __GST_BASE_AUDIO_SINK_H__
-#define __GST_BASE_AUDIO_SINK_H__
+#ifndef __GST_AUDIO_BASE_SINK_H__
+#define __GST_AUDIO_BASE_SINK_H__
#include <gst/gst.h>
#include <gst/base/gstbasesink.h>
G_BEGIN_DECLS
-#define GST_TYPE_BASE_AUDIO_SINK (gst_base_audio_sink_get_type())
-#define GST_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink))
-#define GST_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass))
-#define GST_BASE_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass))
-#define GST_IS_BASE_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK))
-#define GST_IS_BASE_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))
+#define GST_TYPE_AUDIO_BASE_SINK (gst_audio_base_sink_get_type())
+#define GST_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink))
+#define GST_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass))
+#define GST_AUDIO_BASE_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass))
+#define GST_IS_AUDIO_BASE_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK))
+#define GST_IS_AUDIO_BASE_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SINK))
/**
- * GST_BASE_AUDIO_SINK_CLOCK:
- * @obj: a #GstBaseAudioSink
+ * GST_AUDIO_BASE_SINK_CLOCK:
+ * @obj: a #GstAudioBaseSink
*
* Get the #GstClock of @obj.
*/
-#define GST_BASE_AUDIO_SINK_CLOCK(obj) (GST_BASE_AUDIO_SINK (obj)->clock)
+#define GST_AUDIO_BASE_SINK_CLOCK(obj) (GST_AUDIO_BASE_SINK (obj)->clock)
/**
- * GST_BASE_AUDIO_SINK_PAD:
- * @obj: a #GstBaseAudioSink
+ * GST_AUDIO_BASE_SINK_PAD:
+ * @obj: a #GstAudioBaseSink
*
* Get the sink #GstPad of @obj.
*/
-#define GST_BASE_AUDIO_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
+#define GST_AUDIO_BASE_SINK_PAD(obj) (GST_BASE_SINK (obj)->sinkpad)
/**
- * GstBaseAudioSinkSlaveMethod:
- * @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock
- * @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
+ * GstAudioBaseSinkSlaveMethod:
+ * @GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE: Resample to match the master clock
+ * @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
* drifts too much.
- * @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done.
+ * @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done.
*
* Different possible clock slaving algorithms used when the internal audio
* clock is not selected as the pipeline master clock.
*/
typedef enum
{
- GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,
- GST_BASE_AUDIO_SINK_SLAVE_SKEW,
- GST_BASE_AUDIO_SINK_SLAVE_NONE
-} GstBaseAudioSinkSlaveMethod;
+ GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE,
+ GST_AUDIO_BASE_SINK_SLAVE_SKEW,
+ GST_AUDIO_BASE_SINK_SLAVE_NONE
+} GstAudioBaseSinkSlaveMethod;
-#define GST_TYPE_BASE_AUDIO_SINK_SLAVE_METHOD (gst_base_audio_sink_slave_method_get_type ())
+#define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ())
-typedef struct _GstBaseAudioSink GstBaseAudioSink;
-typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;
-typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate;
+typedef struct _GstAudioBaseSink GstAudioBaseSink;
+typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
+typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
/**
- * GstBaseAudioSink:
+ * GstAudioBaseSink:
*
- * Opaque #GstBaseAudioSink.
+ * Opaque #GstAudioBaseSink.
*/
-struct _GstBaseAudioSink {
+struct _GstAudioBaseSink {
GstBaseSink element;
/*< protected >*/ /* with LOCK */
gboolean eos_rendering;
/*< private >*/
- GstBaseAudioSinkPrivate *priv;
+ GstAudioBaseSinkPrivate *priv;
gpointer _gst_reserved[GST_PADDING];
};
/**
- * GstBaseAudioSinkClass:
+ * GstAudioBaseSinkClass:
* @parent_class: the parent class.
* @create_ringbuffer: create and return a #GstAudioRingBuffer to write to.
* @payload: payload data in a format suitable to write to the sink. If no
* buffer, else returns the payloaded buffer with all other metadata
* copied. (Since: 0.10.36)
*
- * #GstBaseAudioSink class. Override the vmethod to implement
+ * #GstAudioBaseSink class. Override the vmethod to implement
* functionality.
*/
-struct _GstBaseAudioSinkClass {
+struct _GstAudioBaseSinkClass {
GstBaseSinkClass parent_class;
/* subclass ringbuffer allocation */
- GstAudioRingBuffer* (*create_ringbuffer) (GstBaseAudioSink *sink);
+ GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSink *sink);
/* subclass payloader */
- GstBuffer* (*payload) (GstBaseAudioSink *sink,
+ GstBuffer* (*payload) (GstAudioBaseSink *sink,
GstBuffer *buffer);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
-GType gst_base_audio_sink_get_type(void);
-GType gst_base_audio_sink_slave_method_get_type (void);
+GType gst_audio_base_sink_get_type(void);
+GType gst_audio_base_sink_slave_method_get_type (void);
GstAudioRingBuffer *
- gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);
+ gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink *sink);
-void gst_base_audio_sink_set_provide_clock (GstBaseAudioSink *sink, gboolean provide);
-gboolean gst_base_audio_sink_get_provide_clock (GstBaseAudioSink *sink);
+void gst_audio_base_sink_set_provide_clock (GstAudioBaseSink *sink, gboolean provide);
+gboolean gst_audio_base_sink_get_provide_clock (GstAudioBaseSink *sink);
-void gst_base_audio_sink_set_slave_method (GstBaseAudioSink *sink,
- GstBaseAudioSinkSlaveMethod method);
-GstBaseAudioSinkSlaveMethod
- gst_base_audio_sink_get_slave_method (GstBaseAudioSink *sink);
+void gst_audio_base_sink_set_slave_method (GstAudioBaseSink *sink,
+ GstAudioBaseSinkSlaveMethod method);
+GstAudioBaseSinkSlaveMethod
+ gst_audio_base_sink_get_slave_method (GstAudioBaseSink *sink);
-void gst_base_audio_sink_set_drift_tolerance (GstBaseAudioSink *sink,
+void gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink *sink,
gint64 drift_tolerance);
-gint64 gst_base_audio_sink_get_drift_tolerance (GstBaseAudioSink *sink);
+gint64 gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink *sink);
-void gst_base_audio_sink_set_alignment_threshold (GstBaseAudioSink * sink,
+void gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
GstClockTime alignment_threshold);
GstClockTime
- gst_base_audio_sink_get_alignment_threshold (GstBaseAudioSink * sink);
+ gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink);
-void gst_base_audio_sink_set_discont_wait (GstBaseAudioSink * sink,
+void gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
GstClockTime discont_wait);
GstClockTime
- gst_base_audio_sink_get_discont_wait (GstBaseAudioSink * sink);
+ gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink);
G_END_DECLS
-#endif /* __GST_BASE_AUDIO_SINK_H__ */
+#endif /* __GST_AUDIO_BASE_SINK_H__ */
#include "gst/gst-i18n-plugin.h"
-GST_DEBUG_CATEGORY_STATIC (gst_base_audio_src_debug);
-#define GST_CAT_DEFAULT gst_base_audio_src_debug
+GST_DEBUG_CATEGORY_STATIC (gst_audio_base_src_debug);
+#define GST_CAT_DEFAULT gst_audio_base_src_debug
GType
-gst_base_audio_src_slave_method_get_type (void)
+gst_audio_base_src_slave_method_get_type (void)
{
static volatile gsize slave_method_type = 0;
/* FIXME 0.11: nick should be "retimestamp" not "re-timestamp" */
static const GEnumValue slave_method[] = {
- {GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
- "GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE", "resample"},
- {GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
- "GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP", "re-timestamp"},
- {GST_BASE_AUDIO_SRC_SLAVE_SKEW, "GST_BASE_AUDIO_SRC_SLAVE_SKEW", "skew"},
- {GST_BASE_AUDIO_SRC_SLAVE_NONE, "GST_BASE_AUDIO_SRC_SLAVE_NONE", "none"},
+ {GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
+ "GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE", "resample"},
+ {GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP,
+ "GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP", "re-timestamp"},
+ {GST_AUDIO_BASE_SRC_SLAVE_SKEW, "GST_AUDIO_BASE_SRC_SLAVE_SKEW", "skew"},
+ {GST_AUDIO_BASE_SRC_SLAVE_NONE, "GST_AUDIO_BASE_SRC_SLAVE_NONE", "none"},
{0, NULL, NULL},
};
if (g_once_init_enter (&slave_method_type)) {
GType tmp =
- g_enum_register_static ("GstBaseAudioSrcSlaveMethod", slave_method);
+ g_enum_register_static ("GstAudioBaseSrcSlaveMethod", slave_method);
g_once_init_leave (&slave_method_type, tmp);
}
return (GType) slave_method_type;
}
-#define GST_BASE_AUDIO_SRC_GET_PRIVATE(obj) \
- (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcPrivate))
+#define GST_AUDIO_BASE_SRC_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcPrivate))
-struct _GstBaseAudioSrcPrivate
+struct _GstAudioBaseSrcPrivate
{
gboolean provide_clock;
/* the clock slaving algorithm in use */
- GstBaseAudioSrcSlaveMethod slave_method;
+ GstAudioBaseSrcSlaveMethod slave_method;
};
/* BaseAudioSrc signals and args */
#define DEFAULT_ACTUAL_BUFFER_TIME -1
#define DEFAULT_ACTUAL_LATENCY_TIME -1
#define DEFAULT_PROVIDE_CLOCK TRUE
-#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_SKEW
+#define DEFAULT_SLAVE_METHOD GST_AUDIO_BASE_SRC_SLAVE_SKEW
enum
{
static void
_do_init (GType type)
{
- GST_DEBUG_CATEGORY_INIT (gst_base_audio_src_debug, "baseaudiosrc", 0,
+ GST_DEBUG_CATEGORY_INIT (gst_audio_base_src_debug, "baseaudiosrc", 0,
"baseaudiosrc element");
#ifdef ENABLE_NLS
#endif /* ENABLE_NLS */
}
-#define gst_base_audio_src_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSrc, gst_base_audio_src, GST_TYPE_PUSH_SRC,
+#define gst_audio_base_src_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstAudioBaseSrc, gst_audio_base_src, GST_TYPE_PUSH_SRC,
_do_init (g_define_type_id));
-static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
+static void gst_audio_base_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static void gst_base_audio_src_get_property (GObject * object, guint prop_id,
+static void gst_audio_base_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static void gst_base_audio_src_dispose (GObject * object);
+static void gst_audio_base_src_dispose (GObject * object);
-static GstStateChangeReturn gst_base_audio_src_change_state (GstElement *
+static GstStateChangeReturn gst_audio_base_src_change_state (GstElement *
element, GstStateChange transition);
-static GstClock *gst_base_audio_src_provide_clock (GstElement * elem);
-static GstClockTime gst_base_audio_src_get_time (GstClock * clock,
- GstBaseAudioSrc * src);
+static GstClock *gst_audio_base_src_provide_clock (GstElement * elem);
+static GstClockTime gst_audio_base_src_get_time (GstClock * clock,
+ GstAudioBaseSrc * src);
-static GstFlowReturn gst_base_audio_src_create (GstBaseSrc * bsrc,
+static GstFlowReturn gst_audio_base_src_create (GstBaseSrc * bsrc,
guint64 offset, guint length, GstBuffer ** buf);
-static gboolean gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event);
-static void gst_base_audio_src_get_times (GstBaseSrc * bsrc,
+static gboolean gst_audio_base_src_event (GstBaseSrc * bsrc, GstEvent * event);
+static void gst_audio_base_src_get_times (GstBaseSrc * bsrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
-static gboolean gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
-static gboolean gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query);
-static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
+static gboolean gst_audio_base_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
+static gboolean gst_audio_base_src_query (GstBaseSrc * bsrc, GstQuery * query);
+static void gst_audio_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
-/* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
+/* static guint gst_audio_base_src_signals[LAST_SIGNAL] = { 0 }; */
static void
-gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
+gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
- g_type_class_add_private (klass, sizeof (GstBaseAudioSrcPrivate));
+ g_type_class_add_private (klass, sizeof (GstAudioBaseSrcPrivate));
- gobject_class->set_property = gst_base_audio_src_set_property;
- gobject_class->get_property = gst_base_audio_src_get_property;
- gobject_class->dispose = gst_base_audio_src_dispose;
+ gobject_class->set_property = gst_audio_base_src_set_property;
+ gobject_class->get_property = gst_audio_base_src_get_property;
+ gobject_class->dispose = gst_audio_base_src_dispose;
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseAudioSrc:actual-buffer-time:
+ * GstAudioBaseSrc:actual-buffer-time:
*
* Actual configured size of audio buffer in microseconds.
*
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
- * GstBaseAudioSrc:actual-latency-time:
+ * GstAudioBaseSrc:actual-latency-time:
*
* Actual configured audio latency in microseconds.
*
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",
- GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
+ GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_base_audio_src_change_state);
+ GST_DEBUG_FUNCPTR (gst_audio_base_src_change_state);
gstelement_class->provide_clock =
- GST_DEBUG_FUNCPTR (gst_base_audio_src_provide_clock);
+ GST_DEBUG_FUNCPTR (gst_audio_base_src_provide_clock);
- gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_src_setcaps);
- gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_src_event);
- gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_base_audio_src_query);
+ gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_base_src_setcaps);
+ gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_audio_base_src_event);
+ gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_base_src_query);
gstbasesrc_class->get_times =
- GST_DEBUG_FUNCPTR (gst_base_audio_src_get_times);
- gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_base_audio_src_create);
- gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_base_audio_src_fixate);
+ GST_DEBUG_FUNCPTR (gst_audio_base_src_get_times);
+ gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_base_src_create);
+ gstbasesrc_class->fixate = GST_DEBUG_FUNCPTR (gst_audio_base_src_fixate);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
}
static void
-gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc)
+gst_audio_base_src_init (GstAudioBaseSrc * baseaudiosrc)
{
- baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc);
+ baseaudiosrc->priv = GST_AUDIO_BASE_SRC_GET_PRIVATE (baseaudiosrc);
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
GST_BASE_SRC (baseaudiosrc)->blocksize = 0;
baseaudiosrc->clock = gst_audio_clock_new ("GstAudioSrcClock",
- (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time, baseaudiosrc,
+ (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, baseaudiosrc,
NULL);
/* we are always a live source */
}
static void
-gst_base_audio_src_dispose (GObject * object)
+gst_audio_base_src_dispose (GObject * object)
{
- GstBaseAudioSrc *src;
+ GstAudioBaseSrc *src;
- src = GST_BASE_AUDIO_SRC (object);
+ src = GST_AUDIO_BASE_SRC (object);
GST_OBJECT_LOCK (src);
if (src->clock) {
}
static GstClock *
-gst_base_audio_src_provide_clock (GstElement * elem)
+gst_audio_base_src_provide_clock (GstElement * elem)
{
- GstBaseAudioSrc *src;
+ GstAudioBaseSrc *src;
GstClock *clock;
- src = GST_BASE_AUDIO_SRC (elem);
+ src = GST_AUDIO_BASE_SRC (elem);
/* we have no ringbuffer (must be NULL state) */
if (src->ringbuffer == NULL)
}
static GstClockTime
-gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
+gst_audio_base_src_get_time (GstClock * clock, GstAudioBaseSrc * src)
{
guint64 raw, samples;
guint delay;
}
/**
- * gst_base_audio_src_set_provide_clock:
- * @src: a #GstBaseAudioSrc
+ * gst_audio_base_src_set_provide_clock:
+ * @src: a #GstAudioBaseSrc
* @provide: new state
*
* Controls whether @src will provide a clock or not. If @provide is %TRUE,
* Since: 0.10.16
*/
void
-gst_base_audio_src_set_provide_clock (GstBaseAudioSrc * src, gboolean provide)
+gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src, gboolean provide)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src));
GST_OBJECT_LOCK (src);
src->priv->provide_clock = provide;
}
/**
- * gst_base_audio_src_get_provide_clock:
- * @src: a #GstBaseAudioSrc
+ * gst_audio_base_src_get_provide_clock:
+ * @src: a #GstAudioBaseSrc
*
* Queries whether @src will provide a clock or not. See also
- * gst_base_audio_src_set_provide_clock.
+ * gst_audio_base_src_set_provide_clock.
*
* Returns: %TRUE if @src will provide a clock.
*
* Since: 0.10.16
*/
gboolean
-gst_base_audio_src_get_provide_clock (GstBaseAudioSrc * src)
+gst_audio_base_src_get_provide_clock (GstAudioBaseSrc * src)
{
gboolean result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), FALSE);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), FALSE);
GST_OBJECT_LOCK (src);
result = src->priv->provide_clock;
}
/**
- * gst_base_audio_src_set_slave_method:
- * @src: a #GstBaseAudioSrc
+ * gst_audio_base_src_set_slave_method:
+ * @src: a #GstAudioBaseSrc
* @method: the new slave method
*
* Controls how clock slaving will be performed in @src.
* Since: 0.10.20
*/
void
-gst_base_audio_src_set_slave_method (GstBaseAudioSrc * src,
- GstBaseAudioSrcSlaveMethod method)
+gst_audio_base_src_set_slave_method (GstAudioBaseSrc * src,
+ GstAudioBaseSrcSlaveMethod method)
{
- g_return_if_fail (GST_IS_BASE_AUDIO_SRC (src));
+ g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src));
GST_OBJECT_LOCK (src);
src->priv->slave_method = method;
}
/**
- * gst_base_audio_src_get_slave_method:
- * @src: a #GstBaseAudioSrc
+ * gst_audio_base_src_get_slave_method:
+ * @src: a #GstAudioBaseSrc
*
* Get the current slave method used by @src.
*
*
* Since: 0.10.20
*/
-GstBaseAudioSrcSlaveMethod
-gst_base_audio_src_get_slave_method (GstBaseAudioSrc * src)
+GstAudioBaseSrcSlaveMethod
+gst_audio_base_src_get_slave_method (GstAudioBaseSrc * src)
{
- GstBaseAudioSrcSlaveMethod result;
+ GstAudioBaseSrcSlaveMethod result;
- g_return_val_if_fail (GST_IS_BASE_AUDIO_SRC (src), -1);
+ g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), -1);
GST_OBJECT_LOCK (src);
result = src->priv->slave_method;
}
static void
-gst_base_audio_src_set_property (GObject * object, guint prop_id,
+gst_audio_base_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
- GstBaseAudioSrc *src;
+ GstAudioBaseSrc *src;
- src = GST_BASE_AUDIO_SRC (object);
+ src = GST_AUDIO_BASE_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
src->latency_time = g_value_get_int64 (value);
break;
case PROP_PROVIDE_CLOCK:
- gst_base_audio_src_set_provide_clock (src, g_value_get_boolean (value));
+ gst_audio_base_src_set_provide_clock (src, g_value_get_boolean (value));
break;
case PROP_SLAVE_METHOD:
- gst_base_audio_src_set_slave_method (src, g_value_get_enum (value));
+ gst_audio_base_src_set_slave_method (src, g_value_get_enum (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-gst_base_audio_src_get_property (GObject * object, guint prop_id,
+gst_audio_base_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
- GstBaseAudioSrc *src;
+ GstAudioBaseSrc *src;
- src = GST_BASE_AUDIO_SRC (object);
+ src = GST_AUDIO_BASE_SRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
GST_OBJECT_UNLOCK (src);
break;
case PROP_PROVIDE_CLOCK:
- g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
+ g_value_set_boolean (value, gst_audio_base_src_get_provide_clock (src));
break;
case PROP_SLAVE_METHOD:
- g_value_set_enum (value, gst_base_audio_src_get_slave_method (src));
+ g_value_set_enum (value, gst_audio_base_src_get_slave_method (src));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
static void
-gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
+gst_audio_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
{
GstStructure *s;
}
static gboolean
-gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
+gst_audio_base_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
{
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc);
GstAudioRingBufferSpec *spec;
gint bpf, rate;
}
static void
-gst_base_audio_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
+gst_audio_base_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* no need to sync to a clock here, we schedule the samples based
}
static gboolean
-gst_base_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
+gst_audio_base_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
}
static gboolean
-gst_base_audio_src_event (GstBaseSrc * bsrc, GstEvent * event)
+gst_audio_base_src_event (GstBaseSrc * bsrc, GstEvent * event)
{
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc);
gboolean res, forward;
res = FALSE;
* If the next sample is too far away, this function will position itself to the
* next most recent sample, creating discontinuity */
static guint64
-gst_base_audio_src_get_offset (GstBaseAudioSrc * src)
+gst_audio_base_src_get_offset (GstAudioBaseSrc * src)
{
guint64 sample;
gint readseg, segdone, segtotal, sps;
}
static GstFlowReturn
-gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
+gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstFlowReturn ret;
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (bsrc);
GstBuffer *buf;
guchar *data, *ptr;
guint samples, total_samples;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
- sample = gst_base_audio_src_get_offset (src);
+ sample = gst_audio_base_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT " length %u",
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
- case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
+ case GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
- case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
+ case GST_AUDIO_BASE_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
}
break;
}
- case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
+ case GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
else
timestamp = 0;
}
- case GST_BASE_AUDIO_SRC_SLAVE_NONE:
+ case GST_AUDIO_BASE_SRC_SLAVE_NONE:
break;
}
} else {
}
/**
- * gst_base_audio_src_create_ringbuffer:
- * @src: a #GstBaseAudioSrc.
+ * gst_audio_base_src_create_ringbuffer:
+ * @src: a #GstAudioBaseSrc.
*
* Create and return the #GstAudioRingBuffer for @src. This function will call the
* ::create_ringbuffer vmethod and will set @src as the parent of the returned
* Returns: The new ringbuffer of @src.
*/
GstAudioRingBuffer *
-gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
+gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc * src)
{
- GstBaseAudioSrcClass *bclass;
+ GstAudioBaseSrcClass *bclass;
GstAudioRingBuffer *buffer = NULL;
- bclass = GST_BASE_AUDIO_SRC_GET_CLASS (src);
+ bclass = GST_AUDIO_BASE_SRC_GET_CLASS (src);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (src);
}
static GstStateChangeReturn
-gst_base_audio_src_change_state (GstElement * element,
+gst_audio_base_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);
+ GstAudioBaseSrc *src = GST_AUDIO_BASE_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
- src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
+ src->ringbuffer = gst_audio_base_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_audio_ring_buffer_open_device (src->ringbuffer))
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
- (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
+ (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
src->clock, TRUE));
* should post this messages whenever necessary */
if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
GST_AUDIO_CLOCK_CAST (src->clock)->func ==
- (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
+ (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
gst_audio_ring_buffer_set_flushing (src->ringbuffer, TRUE);
/* a base class for audio sources.
*/
-#ifndef __GST_BASE_AUDIO_SRC_H__
-#define __GST_BASE_AUDIO_SRC_H__
+#ifndef __GST_AUDIO_BASE_SRC_H__
+#define __GST_AUDIO_BASE_SRC_H__
#include <gst/gst.h>
#include <gst/base/gstpushsrc.h>
G_BEGIN_DECLS
-#define GST_TYPE_BASE_AUDIO_SRC (gst_base_audio_src_get_type())
-#define GST_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc))
-#define GST_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass))
-#define GST_BASE_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass))
-#define GST_IS_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC))
-#define GST_IS_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC))
+#define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type())
+#define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc))
+#define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass))
+#define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass))
+#define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC))
+#define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC))
/**
- * GST_BASE_AUDIO_SRC_CLOCK:
- * @obj: a #GstBaseAudioSrc
+ * GST_AUDIO_BASE_SRC_CLOCK:
+ * @obj: a #GstAudioBaseSrc
*
* Get the #GstClock of @obj.
*/
-#define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock)
+#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
/**
- * GST_BASE_AUDIO_SRC_PAD:
- * @obj: a #GstBaseAudioSrc
+ * GST_AUDIO_BASE_SRC_PAD:
+ * @obj: a #GstAudioBaseSrc
*
* Get the source #GstPad of @obj.
*/
-#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
+#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
-typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
-typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
-typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate;
+typedef struct _GstAudioBaseSrc GstAudioBaseSrc;
+typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass;
+typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
/**
- * GstBaseAudioSrcSlaveMethod:
- * @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
- * @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
+ * GstAudioBaseSrcSlaveMethod:
+ * @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
+ * @GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
* clock time.
- * @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
+ * @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
* drifts too much.
- * @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done.
+ * @GST_AUDIO_BASE_SRC_SLAVE_NONE: No adjustment is done.
*
* Different possible clock slaving algorithms when the internal audio clock was
* not selected as the pipeline clock.
*/
typedef enum
{
- GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
- GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
- GST_BASE_AUDIO_SRC_SLAVE_SKEW,
- GST_BASE_AUDIO_SRC_SLAVE_NONE
-} GstBaseAudioSrcSlaveMethod;
+ GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
+ GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP,
+ GST_AUDIO_BASE_SRC_SLAVE_SKEW,
+ GST_AUDIO_BASE_SRC_SLAVE_NONE
+} GstAudioBaseSrcSlaveMethod;
-#define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ())
+#define GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD (gst_audio_base_src_slave_method_get_type ())
/**
- * GstBaseAudioSrc:
+ * GstAudioBaseSrc:
*
- * Opaque #GstBaseAudioSrc.
+ * Opaque #GstAudioBaseSrc.
*/
-struct _GstBaseAudioSrc {
+struct _GstAudioBaseSrc {
GstPushSrc element;
/*< protected >*/ /* with LOCK */
GstClock *clock;
/*< private >*/
- GstBaseAudioSrcPrivate *priv;
+ GstAudioBaseSrcPrivate *priv;
gpointer _gst_reserved[GST_PADDING - 1];
};
/**
- * GstBaseAudioSrcClass:
+ * GstAudioBaseSrcClass:
* @parent_class: the parent class.
* @create_ringbuffer: create and return a #GstAudioRingBuffer to read from.
*
- * #GstBaseAudioSrc class. Override the vmethod to implement
+ * #GstAudioBaseSrc class. Override the vmethod to implement
* functionality.
*/
-struct _GstBaseAudioSrcClass {
+struct _GstAudioBaseSrcClass {
GstPushSrcClass parent_class;
/* subclass ringbuffer allocation */
- GstAudioRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src);
+ GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
-GType gst_base_audio_src_get_type(void);
-GType gst_base_audio_src_slave_method_get_type (void);
+GType gst_audio_base_src_get_type(void);
+GType gst_audio_base_src_slave_method_get_type (void);
GstAudioRingBuffer *
- gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
+ gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src);
-void gst_base_audio_src_set_provide_clock (GstBaseAudioSrc *src, gboolean provide);
-gboolean gst_base_audio_src_get_provide_clock (GstBaseAudioSrc *src);
+void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide);
+gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src);
-void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src,
- GstBaseAudioSrcSlaveMethod method);
-GstBaseAudioSrcSlaveMethod
- gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src);
+void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
+ GstAudioBaseSrcSlaveMethod method);
+GstAudioBaseSrcSlaveMethod
+ gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src);
G_END_DECLS
-#endif /* __GST_BASE_AUDIO_SRC_H__ */
+#endif /* __GST_AUDIO_BASE_SRC_H__ */
projects / platform / upstream / gstreamer.git / commitdiff