Commit 2817bdad authored by Edward Hervey's avatar Edward Hervey
Browse files

libs: Remove "Since" markers and minor doc fixups

parent 666716a0
......@@ -59,8 +59,6 @@
* to avoid polling.
*
* Last reviewed on 2008-12-17 (0.10.22)
*
* Since: 0.10.22
*/
#ifdef HAVE_CONFIG_H
......@@ -820,8 +818,6 @@ gst_app_sink_query (GstBaseSink * bsink, GstQuery * query)
* a copy of the caps structure. After calling this method, the sink will only
* accept caps that match @caps. If @caps is non-fixed, you must check the caps
* on the buffers to get the actual used caps.
*
* Since: 0.10.22
*/
void
gst_app_sink_set_caps (GstAppSink * appsink, const GstCaps * caps)
......@@ -853,8 +849,6 @@ gst_app_sink_set_caps (GstAppSink * appsink, const GstCaps * caps)
* Get the configured caps on @appsink.
*
* Returns: the #GstCaps accepted by the sink. gst_caps_unref() after usage.
*
* Since: 0.10.22
*/
GstCaps *
gst_app_sink_get_caps (GstAppSink * appsink)
......@@ -886,8 +880,6 @@ gst_app_sink_get_caps (GstAppSink * appsink)
* PLAYING state.
*
* Returns: %TRUE if no more samples can be pulled and the appsink is EOS.
*
* Since: 0.10.22
*/
gboolean
gst_app_sink_is_eos (GstAppSink * appsink)
......@@ -930,8 +922,6 @@ not_started:
* Make appsink emit the "new-preroll" and "new-sample" signals. This option is
* by default disabled because signal emission is expensive and unneeded when
* the application prefers to operate in pull mode.
*
* Since: 0.10.22
*/
void
gst_app_sink_set_emit_signals (GstAppSink * appsink, gboolean emit)
......@@ -955,8 +945,6 @@ gst_app_sink_set_emit_signals (GstAppSink * appsink, gboolean emit)
*
* Returns: %TRUE if @appsink is emiting the "new-preroll" and "new-sample"
* signals.
*
* Since: 0.10.22
*/
gboolean
gst_app_sink_get_emit_signals (GstAppSink * appsink)
......@@ -983,8 +971,6 @@ gst_app_sink_get_emit_signals (GstAppSink * appsink)
* Set the maximum amount of buffers that can be queued in @appsink. After this
* amount of buffers are queued in appsink, any more buffers will block upstream
* elements until a sample is pulled from @appsink.
*
* Since: 0.10.22
*/
void
gst_app_sink_set_max_buffers (GstAppSink * appsink, guint max)
......@@ -1011,8 +997,6 @@ gst_app_sink_set_max_buffers (GstAppSink * appsink, guint max)
* Get the maximum amount of buffers that can be queued in @appsink.
*
* Returns: The maximum amount of buffers that can be queued.
*
* Since: 0.10.22
*/
guint
gst_app_sink_get_max_buffers (GstAppSink * appsink)
......@@ -1038,8 +1022,6 @@ gst_app_sink_get_max_buffers (GstAppSink * appsink)
*
* Instruct @appsink to drop old buffers when the maximum amount of queued
* buffers is reached.
*
* Since: 0.10.22
*/
void
gst_app_sink_set_drop (GstAppSink * appsink, gboolean drop)
......@@ -1068,8 +1050,6 @@ gst_app_sink_set_drop (GstAppSink * appsink, gboolean drop)
*
* Returns: %TRUE if @appsink is dropping old buffers when the queue is
* filled.
*
* Since: 0.10.22
*/
gboolean
gst_app_sink_get_drop (GstAppSink * appsink)
......@@ -1110,8 +1090,6 @@ gst_app_sink_get_drop (GstAppSink * appsink)
* element is set to the READY/NULL state.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*
* Since: 0.10.22
*/
GstSample *
gst_app_sink_pull_preroll (GstAppSink * appsink)
......@@ -1180,8 +1158,6 @@ not_started:
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*
* Since: 0.10.22
*/
GstSample *
......@@ -1250,8 +1226,6 @@ not_started:
*
* If callbacks are installed, no signals will be emitted for performance
* reasons.
*
* Since: 0.10.23
*/
void
gst_app_sink_set_callbacks (GstAppSink * appsink,
......
......@@ -35,7 +35,6 @@ G_BEGIN_DECLS
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_APP_SINK))
#define GST_IS_APP_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_APP_SINK))
/* Since 0.10.23 */
#define GST_APP_SINK_CAST(obj) \
((GstAppSink*)(obj))
......@@ -60,8 +59,6 @@ typedef struct _GstAppSinkPrivate GstAppSinkPrivate;
*
* A set of callbacks that can be installed on the appsink with
* gst_app_sink_set_callbacks().
*
* Since: 0.10.23
*/
typedef struct {
void (*eos) (GstAppSink *sink, gpointer user_data);
......
......@@ -86,8 +86,6 @@
* happened or the state of the appsrc has gone through READY.
*
* Last reviewed on 2008-12-17 (0.10.10)
*
* Since: 0.10.22
*/
#ifdef HAVE_CONFIG_H
......@@ -259,7 +257,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
gobject_class->get_property = gst_app_src_get_property;
/**
* GstAppSrc::caps
* GstAppSrc::caps:
*
* The GstCaps that will negotiated downstream and will be put
* on outgoing buffers.
......@@ -269,7 +267,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
"The allowed caps for the src pad", GST_TYPE_CAPS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::format
* GstAppSrc::format:
*
* The format to use for segment events. When the source is producing
* timestamped buffers this property should be set to GST_FORMAT_TIME.
......@@ -279,7 +277,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
"The format of the segment events and seek", GST_TYPE_FORMAT,
DEFAULT_PROP_FORMAT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::size
* GstAppSrc::size:
*
* The total size in bytes of the data stream. If the total size is known, it
* is recommended to configure it with this property.
......@@ -290,7 +288,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
-1, G_MAXINT64, DEFAULT_PROP_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::stream-type
* GstAppSrc::stream-type:
*
* The type of stream that this source is producing. For seekable streams the
* application should connect to the seek-data signal.
......@@ -301,7 +299,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
DEFAULT_PROP_STREAM_TYPE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::max-bytes
* GstAppSrc::max-bytes:
*
* The maximum amount of bytes that can be queued internally.
* After the maximum amount of bytes are queued, appsrc will emit the
......@@ -313,7 +311,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
0, G_MAXUINT64, DEFAULT_PROP_MAX_BYTES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::block
* GstAppSrc::block:
*
* When max-bytes are queued and after the enough-data signal has been emitted,
* block any further push-buffer calls until the amount of queued bytes drops
......@@ -325,7 +323,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
DEFAULT_PROP_BLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::is-live
* GstAppSrc::is-live:
*
* Instruct the source to behave like a live source. This includes that it
* will only push out buffers in the PLAYING state.
......@@ -335,7 +333,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
"Whether to act as a live source",
DEFAULT_PROP_IS_LIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::min-latency
* GstAppSrc::min-latency:
*
* The minimum latency of the source. A value of -1 will use the default
* latency calculations of #GstBaseSrc.
......@@ -346,7 +344,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
-1, G_MAXINT64, DEFAULT_PROP_MIN_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::max-latency
* GstAppSrc::max-latency:
*
* The maximum latency of the source. A value of -1 means an unlimited amout
* of latency.
......@@ -358,13 +356,11 @@ gst_app_src_class_init (GstAppSrcClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::emit-signals
* GstAppSrc::emit-signals:
*
* Make appsrc emit the "need-data", "enough-data" and "seek-data" signals.
* This option is by default enabled for backwards compatibility reasons but
* can disabled when needed because signal emission is expensive.
*
* Since: 0.10.23
*/
g_object_class_install_property (gobject_class, PROP_EMIT_SIGNALS,
g_param_spec_boolean ("emit-signals", "Emit signals",
......@@ -373,12 +369,10 @@ gst_app_src_class_init (GstAppSrcClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAppSrc::empty-percent
* GstAppSrc::empty-percent:
*
* Make appsrc emit the "need-data" signal when the amount of bytes in the
* queue drops below this percentage of max-bytes.
*
* Since: 0.10.27
*/
g_object_class_install_property (gobject_class, PROP_MIN_PERCENT,
g_param_spec_uint ("min-percent", "Min Percent",
......@@ -1085,8 +1079,6 @@ seek_error:
* a copy of the caps structure. After calling this method, the source will
* only produce caps that match @caps. @caps must be fixed and the caps on the
* buffers must match the caps or left NULL.
*
* Since: 0.10.22
*/
void
gst_app_src_set_caps (GstAppSrc * appsrc, const GstCaps * caps)
......@@ -1118,8 +1110,6 @@ gst_app_src_set_caps (GstAppSrc * appsrc, const GstCaps * caps)
* Get the configured caps on @appsrc.
*
* Returns: the #GstCaps produced by the source. gst_caps_unref() after usage.
*
* Since: 0.10.22
*/
GstCaps *
gst_app_src_get_caps (GstAppSrc * appsrc)
......@@ -1136,8 +1126,6 @@ gst_app_src_get_caps (GstAppSrc * appsrc)
*
* Set the size of the stream in bytes. A value of -1 means that the size is
* not known.
*
* Since: 0.10.22
*/
void
gst_app_src_set_size (GstAppSrc * appsrc, gint64 size)
......@@ -1162,8 +1150,6 @@ gst_app_src_set_size (GstAppSrc * appsrc, gint64 size)
* not known.
*
* Returns: the size of the stream previously set with gst_app_src_set_size();
*
* Since: 0.10.22
*/
gint64
gst_app_src_get_size (GstAppSrc * appsrc)
......@@ -1192,8 +1178,6 @@ gst_app_src_get_size (GstAppSrc * appsrc)
* be connected to.
*
* A stream_type stream
*
* Since: 0.10.22
*/
void
gst_app_src_set_stream_type (GstAppSrc * appsrc, GstAppStreamType type)
......@@ -1218,8 +1202,6 @@ gst_app_src_set_stream_type (GstAppSrc * appsrc, GstAppStreamType type)
* with gst_app_src_set_stream_type().
*
* Returns: the stream type.
*
* Since: 0.10.22
*/
GstAppStreamType
gst_app_src_get_stream_type (GstAppSrc * appsrc)
......@@ -1247,8 +1229,6 @@ gst_app_src_get_stream_type (GstAppSrc * appsrc)
* Set the maximum amount of bytes that can be queued in @appsrc.
* After the maximum amount of bytes are queued, @appsrc will emit the
* "enough-data" signal.
*
* Since: 0.10.22
*/
void
gst_app_src_set_max_bytes (GstAppSrc * appsrc, guint64 max)
......@@ -1276,8 +1256,6 @@ gst_app_src_set_max_bytes (GstAppSrc * appsrc, guint64 max)
* Get the maximum amount of bytes that can be queued in @appsrc.
*
* Returns: The maximum amount of bytes that can be queued.
*
* Since: 0.10.22
*/
guint64
gst_app_src_get_max_bytes (GstAppSrc * appsrc)
......@@ -1330,8 +1308,6 @@ gst_app_src_set_latencies (GstAppSrc * appsrc, gboolean do_min, guint64 min,
*
* Configure the @min and @max latency in @src. If @min is set to -1, the
* default latency calculations for pseudo-live sources will be used.
*
* Since: 0.10.22
*/
void
gst_app_src_set_latency (GstAppSrc * appsrc, guint64 min, guint64 max)
......@@ -1346,8 +1322,6 @@ gst_app_src_set_latency (GstAppSrc * appsrc, guint64 min, guint64 max)
* @max: the min latency
*
* Retrieve the min and max latencies in @min and @max respectively.
*
* Since: 0.10.22
*/
void
gst_app_src_get_latency (GstAppSrc * appsrc, guint64 * min, guint64 * max)
......@@ -1374,8 +1348,6 @@ gst_app_src_get_latency (GstAppSrc * appsrc, guint64 * min, guint64 * max)
* Make appsrc emit the "new-preroll" and "new-buffer" signals. This option is
* by default disabled because signal emission is expensive and unneeded when
* the application prefers to operate in pull mode.
*
* Since: 0.10.23
*/
void
gst_app_src_set_emit_signals (GstAppSrc * appsrc, gboolean emit)
......@@ -1399,8 +1371,6 @@ gst_app_src_set_emit_signals (GstAppSrc * appsrc, gboolean emit)
*
* Returns: %TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
* signals.
*
* Since: 0.10.23
*/
gboolean
gst_app_src_get_emit_signals (GstAppSrc * appsrc)
......@@ -1522,8 +1492,6 @@ eos:
* Returns: #GST_FLOW_OK when the buffer was successfuly queued.
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
* #GST_FLOW_EOS when EOS occured.
*
* Since: 0.10.22
*/
GstFlowReturn
gst_app_src_push_buffer (GstAppSrc * appsrc, GstBuffer * buffer)
......@@ -1548,8 +1516,6 @@ gst_app_src_push_buffer_action (GstAppSrc * appsrc, GstBuffer * buffer)
*
* Returns: #GST_FLOW_OK when the EOS was successfuly queued.
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
*
* Since: 0.10.22
*/
GstFlowReturn
gst_app_src_end_of_stream (GstAppSrc * appsrc)
......@@ -1596,8 +1562,6 @@ flushing:
*
* If callbacks are installed, no signals will be emitted for performance
* reasons.
*
* Since: 0.10.23
*/
void
gst_app_src_set_callbacks (GstAppSrc * appsrc,
......
......@@ -35,7 +35,6 @@ G_BEGIN_DECLS
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_APP_SRC))
#define GST_IS_APP_SRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_APP_SRC))
/* Since 0.10.23 */
#define GST_APP_SRC_CAST(obj) \
((GstAppSrc*)(obj))
......@@ -57,8 +56,6 @@ typedef struct _GstAppSrcPrivate GstAppSrcPrivate;
*
* A set of callbacks that can be installed on the appsrc with
* gst_app_src_set_callbacks().
*
* Since: 0.10.23
*/
typedef struct {
void (*need_data) (GstAppSrc *src, guint length, gpointer user_data);
......@@ -117,7 +114,7 @@ struct _GstAppSrcClass
GType gst_app_src_get_type(void);
/* GType getter for GstAppStreamType, since 0.10.32 */
/* GType getter for GstAppStreamType */
#define GST_TYPE_APP_STREAM_TYPE (gst_app_stream_type_get_type ())
GType gst_app_stream_type_get_type (void);
......
......@@ -293,7 +293,6 @@ gst_audio_channel_positions_to_mask (const GstAudioChannelPosition * position,
* @channels: The number of channels
* @channel_mask: The input channel_mask
* @position: The %GstAudioChannelPositions
* @caps: a #GstCaps
*
* Convert the @channels present in @channel_mask to a @position array
* (which should have at least @channels entries ensured by caller).
......
......@@ -370,10 +370,10 @@ invalid_channel_positions:
/**
* gst_audio_info_convert:
* @info: a #GstAudioInfo
* @src_format: #GstFormat of the @src_value
* @src_value: value to convert
* @dest_format: #GstFormat of the @dest_value
* @dest_value: pointer to destination value
* @src_fmt: #GstFormat of the @src_val
* @src_val: value to convert
* @dest_fmt: #GstFormat of the @dest_val
* @dest_val: pointer to destination value
*
* Converts among various #GstFormat types. This function handles
* GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
......
......@@ -51,8 +51,6 @@
*
* If the buffer has no timestamp, it is assumed to be inside the segment and
* is not clipped
*
* Since: 0.10.14
*/
GstBuffer *
gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
......
......@@ -234,12 +234,10 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
"Allow pull-based scheduling", DEFAULT_CAN_ACTIVATE_PULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioBaseSink:drift-tolerance
* 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",
......@@ -247,12 +245,10 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
G_MAXINT64, DEFAULT_DRIFT_TOLERANCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioBaseSink: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.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
......@@ -261,12 +257,10 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioBaseSink: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.
*
* Since: 0.10.36
*/
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
......@@ -561,8 +555,6 @@ gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
* 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
gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
......@@ -586,8 +578,6 @@ gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
* gst_audio_base_sink_set_provide_clock.
*
* Returns: %TRUE if @sink will provide a clock.
*
* Since: 0.10.16
*/
gboolean
gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
......@@ -609,8 +599,6 @@ gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
* @method: the new slave method
*
* Controls how clock slaving will be performed in @sink.
*
* Since: 0.10.16
*/
void
gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
......@@ -630,8 +618,6 @@ gst_audio_base_sink_set_slave_method (GstAudioBaseSink * sink,
* Get the current slave method used by @sink.
*
* Returns: The current slave method used by @sink.
*
* Since: 0.10.16
*/
GstAudioBaseSinkSlaveMethod
gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
......@@ -654,8 +640,6 @@ gst_audio_base_sink_get_slave_method (GstAudioBaseSink * sink)
* @drift_tolerance: the new drift tolerance in microseconds
*
* Controls the sink's drift tolerance.
*
* Since: 0.10.31
*/
void
gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
......@@ -669,14 +653,12 @@ gst_audio_base_sink_set_drift_tolerance (GstAudioBaseSink * sink,
}
/**
* gst_audio_base_sink_get_drift_tolerance
* 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
gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
......@@ -698,8 +680,6 @@ gst_audio_base_sink_get_drift_tolerance (GstAudioBaseSink * sink)
* @alignment_threshold: the new alignment threshold in nanoseconds
*
* Controls the sink's alignment threshold.
*
* Since: 0.10.36
*/
void
gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
......@@ -713,14 +693,12 @@ gst_audio_base_sink_set_alignment_threshold (GstAudioBaseSink * sink,
}
/**
* gst_audio_base_sink_get_alignment_threshold
* 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.
*
* Since: 0.10.36
*/
GstClockTime
gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
......@@ -742,8 +720,6 @@ gst_audio_base_sink_get_alignment_threshold (GstAudioBaseSink * sink)
* @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_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
......@@ -757,14 +733,12 @@ gst_audio_base_sink_set_discont_wait (GstAudioBaseSink * sink,
}
/**
* gst_audio_base_sink_get_discont_wait
* 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.
*
* Since: 0.10.36
*/
GstClockTime
gst_audio_base_sink_get_discont_wait (GstAudioBaseSink * sink)
......@@ -1981,7 +1955,7 @@ sync_latency_failed:
* ::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.
* Returns: (transfer none): The new ringbuffer of @sink.
*/
GstAudioRingBuffer *
gst_audio_base_sink_create_ringbuffer (GstAudioBaseSink * sink)
......
......@@ -139,7 +139,7 @@ struct _GstAudioBaseSink {
* @payload: payload data in a format suitable to write to the sink. If no
* payloading is required, returns a reffed copy of the original
* buffer, else returns the payloaded buffer with all other metadata
* copied. (Since: 0.10.36)
* copied.
*
* #GstAudioBaseSink class. Override the vmethod to implement
* functionality.
......
......@@ -179,8 +179,6 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
* GstAudioBaseSrc:actual-buffer-time:
*
* Actual configured size of audio buffer in microseconds.
*
* Since: 0.10.20
**/
g_object_class_install_property (gobject_class, PROP_ACTUAL_BUFFER_TIME,
g_param_spec_int64 ("actual-buffer-time", "Actual Buffer Time",
......@@ -192,8 +190,6 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
* GstAudioBaseSrc:actual-latency-time:
*
* Actual configured audio latency in microseconds.
*
* Since: 0.10.20
**/
g_object_class_install_property (gobject_class, PROP_ACTUAL_LATENCY_TIME,
g_param_spec_int64 ("actual-latency-time", "Actual Latency Time",
......@@ -357,8 +353,6 @@ gst_audio_base_src_get_time (GstClock * clock, GstAudioBaseSrc * src)
* Controls whether @src will provide a clock or not. If @provide is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
*
* Since: 0.10.16
*/
void
gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src, gboolean provide)
......@@ -381,8 +375,6 @@ gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src, gboolean provide)
* gst_audio_base_src_set_provide_clock.