/**
* gst_event_new_stream_start:
* @stream_id: Identifier for this stream
*
* Create a new STREAM_START event. The stream start event can only
* travel downstream synchronized with the buffer flow. It is expected
* to be the first event that is sent for a new stream.
*
* Source elements, demuxers and other elements that create new streams
* are supposed to send this event as the first event of a new stream. It
* should not be send after a flushing seek or in similar situations
* and is used to mark the beginning of a new logical stream. Elements
* combining multiple streams must ensure that this event is only forwarded
* downstream once and not for every single input stream.
*
* The @stream_id should be a unique string that consists of the upstream
* stream-id, / as separator and a unique stream-id for this specific
* stream. A new stream-id should only be created for a stream if the upstream
* stream is split into (potentially) multiple new streams, e.g. in a demuxer,
* but not for every single element in the pipeline.
* gst_pad_create_stream_id() or gst_pad_create_stream_id_printf() can be
* used to create a stream-id.
*
* Returns: (transfer full): the new STREAM_START event.
*/
GstEvent *
gst_event_new_stream_start (const gchar * stream_id)
{
GstStructure *s;
g_return_val_if_fail (stream_id != NULL, NULL);
s = gst_structure_new_id (GST_QUARK (EVENT_STREAM_START),
GST_QUARK (STREAM_ID), G_TYPE_STRING, stream_id, NULL);
return gst_event_new_custom (GST_EVENT_STREAM_START, s);
}
/**
* gst_event_new_flush_stop:
* @reset_time: if time should be reset
*
* Allocate a new flush stop event. The flush stop event can be sent
* upstream and downstream and travels serialized with the dataflow.
* It is typically sent after sending a FLUSH_START event to make the
* pads accept data again.
*
* Elements can process this event synchronized with the dataflow since
* the preceeding FLUSH_START event stopped the dataflow.
*
* This event is typically generated to complete a seek and to resume
* dataflow.
*
* Returns: (transfer full): a new flush stop event.
*/
GstEvent *
gst_event_new_flush_stop (gboolean reset_time)
{
GstEvent *event;
GST_CAT_INFO (GST_CAT_EVENT, "creating flush stop %d", reset_time);
event = gst_event_new_custom (GST_EVENT_FLUSH_STOP,
gst_structure_new_id (GST_QUARK (EVENT_FLUSH_STOP),
GST_QUARK (RESET_TIME), G_TYPE_BOOLEAN, reset_time, NULL));
return event;
}
static void
send_latency_probe (GstElement * parent, GstPad * pad, guint64 ts)
{
if (parent && (!GST_IS_BIN (parent)) &&
GST_OBJECT_FLAG_IS_SET (parent, GST_ELEMENT_FLAG_SOURCE)) {
GstEvent *latency_probe = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new_id (latency_probe_id,
latency_probe_pad, GST_TYPE_PAD, pad,
latency_probe_ts, G_TYPE_UINT64, ts,
NULL));
gst_pad_push_event (pad, latency_probe);
}
}
/**
* gst_event_new_toc_select:
* @uid: UID in the TOC to start playback from.
*
* Generate a TOC select event with the given @uid. The purpose of the
* TOC select event is to start playback based on the TOC's entry with the
* given @uid.
*
* Returns: a new #GstEvent.
*/
GstEvent *
gst_event_new_toc_select (const gchar * uid)
{
GstStructure *structure;
g_return_val_if_fail (uid != NULL, NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating toc select event for UID: %s", uid);
structure = gst_structure_new_id (GST_QUARK (EVENT_TOC_SELECT),
GST_QUARK (UID), G_TYPE_STRING, uid, NULL);
return gst_event_new_custom (GST_EVENT_TOC_SELECT, structure);
}
/**
* gst_event_new_latency:
* @latency: the new latency value
*
* Create a new latency event. The event is sent upstream from the sinks and
* notifies elements that they should add an additional @latency to the
* running time before synchronising against the clock.
*
* The latency is mostly used in live sinks and is always expressed in
* the time format.
*
* Returns: (transfer full): a new #GstEvent
*/
GstEvent *
gst_event_new_latency (GstClockTime latency)
{
GstEvent *event;
GstStructure *structure;
GST_CAT_INFO (GST_CAT_EVENT,
"creating latency event %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
structure = gst_structure_new_id (GST_QUARK (EVENT_LATENCY),
GST_QUARK (LATENCY), G_TYPE_UINT64, latency, NULL);
event = gst_event_new_custom (GST_EVENT_LATENCY, structure);
return event;
}
gboolean
gst_allocator_set_vaapi_video_info (GstAllocator * allocator,
const GstVideoInfo * vip, guint flags)
{
g_return_val_if_fail (GST_IS_ALLOCATOR (allocator), FALSE);
g_return_val_if_fail (vip != NULL, FALSE);
g_object_set_qdata_full (G_OBJECT (allocator), GST_VAAPI_VIDEO_INFO_QUARK,
gst_structure_new_id (GST_VAAPI_VIDEO_INFO_QUARK,
INFO_QUARK, GST_VAAPI_TYPE_VIDEO_INFO, vip,
FLAGS_QUARK, G_TYPE_UINT, flags, NULL),
(GDestroyNotify) gst_structure_free);
return TRUE;
}
/**
* gst_event_new_caps:
* @caps: (transfer none): a #GstCaps
*
* Create a new CAPS event for @caps. The caps event can only travel downstream
* synchronized with the buffer flow and contains the format of the buffers
* that will follow after the event.
*
* Returns: (transfer full): the new CAPS event.
*/
GstEvent *
gst_event_new_caps (GstCaps * caps)
{
GstEvent *event;
g_return_val_if_fail (caps != NULL, NULL);
g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating caps event %" GST_PTR_FORMAT, caps);
event = gst_event_new_custom (GST_EVENT_CAPS,
gst_structure_new_id (GST_QUARK (EVENT_CAPS),
GST_QUARK (CAPS), GST_TYPE_CAPS, caps, NULL));
return event;
}
/**
* gst_event_new_segment_done:
* @format: The format of the position being done
* @position: The position of the segment being done
*
* Create a new segment-done event. This event is sent by elements that
* finish playback of a segment as a result of a segment seek.
*
* Returns: (transfer full): a new #GstEvent
*/
GstEvent *
gst_event_new_segment_done (GstFormat format, gint64 position)
{
GstEvent *event;
GstStructure *structure;
GST_CAT_INFO (GST_CAT_EVENT, "creating segment-done event");
structure = gst_structure_new_id (GST_QUARK (EVENT_SEGMENT_DONE),
GST_QUARK (FORMAT), GST_TYPE_FORMAT, format,
GST_QUARK (POSITION), G_TYPE_INT64, position, NULL);
event = gst_event_new_custom (GST_EVENT_SEGMENT_DONE, structure);
return event;
}
/* FIXME 0.11: take ownership of msg for consistency? */
GstEvent *
gst_event_new_sink_message (const gchar * name, GstMessage * msg)
{
GstEvent *event;
GstStructure *structure;
g_return_val_if_fail (msg != NULL, NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating sink-message event");
structure = gst_structure_new_id (g_quark_from_string (name),
GST_QUARK (MESSAGE), GST_TYPE_MESSAGE, msg, NULL);
event = gst_event_new_custom (GST_EVENT_SINK_MESSAGE, structure);
return event;
}
/**
* gst_event_new_segment:
* @segment: (transfer none): a #GstSegment
*
* Create a new SEGMENT event for @segment. The segment event can only travel
* downstream synchronized with the buffer flow and contains timing information
* and playback properties for the buffers that will follow.
*
* The newsegment event marks the range of buffers to be processed. All
* data not within the segment range is not to be processed. This can be
* used intelligently by plugins to apply more efficient methods of skipping
* unneeded data. The valid range is expressed with the @start and @stop
* values.
*
* The time value of the segment is used in conjunction with the start
* value to convert the buffer timestamps into the stream time. This is
* usually done in sinks to report the current stream_time.
* @time represents the stream_time of a buffer carrying a timestamp of
* @start. @time cannot be -1.
*
* @start cannot be -1, @stop can be -1. If there
* is a valid @stop given, it must be greater or equal the @start, including
* when the indicated playback @rate is < 0.
*
* The @applied_rate value provides information about any rate adjustment that
* has already been made to the timestamps and content on the buffers of the
* stream. (@rate * @applied_rate) should always equal the rate that has been
* requested for playback. For example, if an element has an input segment
* with intended playback @rate of 2.0 and applied_rate of 1.0, it can adjust
* incoming timestamps and buffer content by half and output a newsegment event
* with @rate of 1.0 and @applied_rate of 2.0
*
* After a newsegment event, the buffer stream time is calculated with:
*
* time + (TIMESTAMP(buf) - start) * ABS (rate * applied_rate)
*
* Returns: (transfer full): the new SEGMENT event.
*/
GstEvent *
gst_event_new_segment (const GstSegment * segment)
{
GstEvent *event;
g_return_val_if_fail (segment != NULL, NULL);
g_return_val_if_fail (segment->rate != 0.0, NULL);
g_return_val_if_fail (segment->applied_rate != 0.0, NULL);
g_return_val_if_fail (segment->format != GST_FORMAT_UNDEFINED, NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating segment event %" GST_SEGMENT_FORMAT,
segment);
event = gst_event_new_custom (GST_EVENT_SEGMENT,
gst_structure_new_id (GST_QUARK (EVENT_SEGMENT),
GST_QUARK (SEGMENT), GST_TYPE_SEGMENT, segment, NULL));
return event;
}
/**
* gst_event_new_gap:
* @timestamp: the start time (pts) of the gap
* @duration: the duration of the gap
*
* Create a new GAP event. A gap event can be thought of as conceptually
* equivalent to a buffer to signal that there is no data for a certain
* amount of time. This is useful to signal a gap to downstream elements
* which may wait for data, such as muxers or mixers or overlays, especially
* for sparse streams such as subtitle streams.
*
* Returns: (transfer full): the new GAP event.
*/
GstEvent *
gst_event_new_gap (GstClockTime timestamp, GstClockTime duration)
{
GstEvent *event;
g_return_val_if_fail (GST_CLOCK_TIME_IS_VALID (timestamp), NULL);
GST_CAT_TRACE (GST_CAT_EVENT, "creating gap %" GST_TIME_FORMAT " - "
"%" GST_TIME_FORMAT " (duration: %" GST_TIME_FORMAT ")",
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (timestamp + duration),
GST_TIME_ARGS (duration));
event = gst_event_new_custom (GST_EVENT_GAP,
gst_structure_new_id (GST_QUARK (EVENT_GAP),
GST_QUARK (TIMESTAMP), GST_TYPE_CLOCK_TIME, timestamp,
GST_QUARK (DURATION), GST_TYPE_CLOCK_TIME, duration, NULL));
return event;
}
/**
* gst_event_new_buffer_size:
* @format: buffer format
* @minsize: minimum buffer size
* @maxsize: maximum buffer size
* @async: thread behavior
*
* Create a new buffersize event. The event is sent downstream and notifies
* elements that they should provide a buffer of the specified dimensions.
*
* When the @async flag is set, a thread boundary is preferred.
*
* Returns: (transfer full): a new #GstEvent
*/
GstEvent *
gst_event_new_buffer_size (GstFormat format, gint64 minsize,
gint64 maxsize, gboolean async)
{
GstEvent *event;
GstStructure *structure;
GST_CAT_INFO (GST_CAT_EVENT,
"creating buffersize format %s, minsize %" G_GINT64_FORMAT
", maxsize %" G_GINT64_FORMAT ", async %d", gst_format_get_name (format),
minsize, maxsize, async);
structure = gst_structure_new_id (GST_QUARK (EVENT_BUFFER_SIZE),
GST_QUARK (FORMAT), GST_TYPE_FORMAT, format,
GST_QUARK (MINSIZE), G_TYPE_INT64, minsize,
GST_QUARK (MAXSIZE), G_TYPE_INT64, maxsize,
GST_QUARK (ASYNC), G_TYPE_BOOLEAN, async, NULL);
event = gst_event_new_custom (GST_EVENT_BUFFERSIZE, structure);
return event;
}
/**
* gst_event_new_step:
* @format: the format of @amount
* @amount: the amount of data to step
* @rate: the step rate
* @flush: flushing steps
* @intermediate: intermediate steps
*
* Create a new step event. The purpose of the step event is to instruct a sink
* to skip @amount (expressed in @format) of media. It can be used to implement
* stepping through the video frame by frame or for doing fast trick modes.
*
* A rate of <= 0.0 is not allowed. Pause the pipeline, for the effect of rate
* = 0.0 or first reverse the direction of playback using a seek event to get
* the same effect as rate < 0.0.
*
* The @flush flag will clear any pending data in the pipeline before starting
* the step operation.
*
* The @intermediate flag instructs the pipeline that this step operation is
* part of a larger step operation.
*
* Returns: (transfer full): a new #GstEvent
*/
GstEvent *
gst_event_new_step (GstFormat format, guint64 amount, gdouble rate,
gboolean flush, gboolean intermediate)
{
GstEvent *event;
GstStructure *structure;
g_return_val_if_fail (rate > 0.0, NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating step event");
structure = gst_structure_new_id (GST_QUARK (EVENT_STEP),
GST_QUARK (FORMAT), GST_TYPE_FORMAT, format,
GST_QUARK (AMOUNT), G_TYPE_UINT64, amount,
GST_QUARK (RATE), G_TYPE_DOUBLE, rate,
GST_QUARK (FLUSH), G_TYPE_BOOLEAN, flush,
GST_QUARK (INTERMEDIATE), G_TYPE_BOOLEAN, intermediate, NULL);
event = gst_event_new_custom (GST_EVENT_STEP, structure);
return event;
}
/**
* gst_event_new_toc:
* @toc: (transfer none): #GstToc structure.
* @updated: whether @toc was updated or not.
*
* Generate a TOC event from the given @toc. The purpose of the TOC event is to
* inform elements that some kind of the TOC was found.
*
* Returns: (transfer full): a new #GstEvent.
*/
GstEvent *
gst_event_new_toc (GstToc * toc, gboolean updated)
{
GstStructure *toc_struct;
GQuark id;
g_return_val_if_fail (toc != NULL, NULL);
GST_CAT_INFO (GST_CAT_EVENT, "creating toc event");
/* need different structure names so sticky_multi event stuff on pads
* works, i.e. both TOC events are kept around */
if (gst_toc_get_scope (toc) == GST_TOC_SCOPE_GLOBAL)
id = GST_QUARK (EVENT_TOC_GLOBAL);
else
id = GST_QUARK (EVENT_TOC_CURRENT);
toc_struct = gst_structure_new_id (id,
GST_QUARK (TOC), GST_TYPE_TOC, toc,
GST_QUARK (UPDATED), G_TYPE_BOOLEAN, updated, NULL);
return gst_event_new_custom (GST_EVENT_TOC, toc_struct);
}
/**
* gst_event_new_seek:
* @rate: The new playback rate
* @format: The format of the seek values
* @flags: The optional seek flags
* @start_type: The type and flags for the new start position
* @start: The value of the new start position
* @stop_type: The type and flags for the new stop position
* @stop: The value of the new stop position
*
* Allocate a new seek event with the given parameters.
*
* The seek event configures playback of the pipeline between @start to @stop
* at the speed given in @rate, also called a playback segment.
* The @start and @stop values are expressed in @format.
*
* A @rate of 1.0 means normal playback rate, 2.0 means double speed.
* Negatives values means backwards playback. A value of 0.0 for the
* rate is not allowed and should be accomplished instead by PAUSING the
* pipeline.
*
* A pipeline has a default playback segment configured with a start
* position of 0, a stop position of -1 and a rate of 1.0. The currently
* configured playback segment can be queried with #GST_QUERY_SEGMENT.
*
* @start_type and @stop_type specify how to adjust the currently configured
* start and stop fields in playback segment. Adjustments can be made relative
* or absolute to the last configured values. A type of #GST_SEEK_TYPE_NONE
* means that the position should not be updated.
*
* When the rate is positive and @start has been updated, playback will start
* from the newly configured start position.
*
* For negative rates, playback will start from the newly configured stop
* position (if any). If the stop position is updated, it must be different from
* -1 (#GST_CLOCK_TIME_NONE) for negative rates.
*
* It is not possible to seek relative to the current playback position, to do
* this, PAUSE the pipeline, query the current playback position with
* #GST_QUERY_POSITION and update the playback segment current position with a
* #GST_SEEK_TYPE_SET to the desired position.
*
* Returns: (transfer full): a new seek event.
*/
GstEvent *
gst_event_new_seek (gdouble rate, GstFormat format, GstSeekFlags flags,
GstSeekType start_type, gint64 start, GstSeekType stop_type, gint64 stop)
{
GstEvent *event;
GstStructure *structure;
g_return_val_if_fail (rate != 0.0, NULL);
if (format == GST_FORMAT_TIME) {
GST_CAT_INFO (GST_CAT_EVENT,
"creating seek rate %lf, format TIME, flags %d, "
"start_type %d, start %" GST_TIME_FORMAT ", "
"stop_type %d, stop %" GST_TIME_FORMAT,
rate, flags, start_type, GST_TIME_ARGS (start),
stop_type, GST_TIME_ARGS (stop));
} else {
GST_CAT_INFO (GST_CAT_EVENT,
"creating seek rate %lf, format %s, flags %d, "
"start_type %d, start %" G_GINT64_FORMAT ", "
"stop_type %d, stop %" G_GINT64_FORMAT,
rate, gst_format_get_name (format), flags, start_type, start, stop_type,
stop);
}
structure = gst_structure_new_id (GST_QUARK (EVENT_SEEK),
GST_QUARK (RATE), G_TYPE_DOUBLE, rate,
GST_QUARK (FORMAT), GST_TYPE_FORMAT, format,
GST_QUARK (FLAGS), GST_TYPE_SEEK_FLAGS, flags,
GST_QUARK (CUR_TYPE), GST_TYPE_SEEK_TYPE, start_type,
GST_QUARK (CUR), G_TYPE_INT64, start,
GST_QUARK (STOP_TYPE), GST_TYPE_SEEK_TYPE, stop_type,
GST_QUARK (STOP), G_TYPE_INT64, stop, NULL);
event = gst_event_new_custom (GST_EVENT_SEEK, structure);
return event;
}
/**
* gst_event_new_qos:
* @type: the QoS type
* @proportion: the proportion of the qos message
* @diff: The time difference of the last Clock sync
* @timestamp: The timestamp of the buffer
*
* Allocate a new qos event with the given values.
* The QOS event is generated in an element that wants an upstream
* element to either reduce or increase its rate because of
* high/low CPU load or other resource usage such as network performance or
* throttling. Typically sinks generate these events for each buffer
* they receive.
*
* @type indicates the reason for the QoS event. #GST_QOS_TYPE_OVERFLOW is
* used when a buffer arrived in time or when the sink cannot keep up with
* the upstream datarate. #GST_QOS_TYPE_UNDERFLOW is when the sink is not
* receiving buffers fast enough and thus has to drop late buffers.
* #GST_QOS_TYPE_THROTTLE is used when the datarate is artificially limited
* by the application, for example to reduce power consumption.
*
* @proportion indicates the real-time performance of the streaming in the
* element that generated the QoS event (usually the sink). The value is
* generally computed based on more long term statistics about the streams
* timestamps compared to the clock.
* A value < 1.0 indicates that the upstream element is producing data faster
* than real-time. A value > 1.0 indicates that the upstream element is not
* producing data fast enough. 1.0 is the ideal @proportion value. The
* proportion value can safely be used to lower or increase the quality of
* the element.
*
* @diff is the difference against the clock in running time of the last
* buffer that caused the element to generate the QOS event. A negative value
* means that the buffer with @timestamp arrived in time. A positive value
* indicates how late the buffer with @timestamp was. When throttling is
* enabled, @diff will be set to the requested throttling interval.
*
* @timestamp is the timestamp of the last buffer that cause the element
* to generate the QOS event. It is expressed in running time and thus an ever
* increasing value.
*
* The upstream element can use the @diff and @timestamp values to decide
* whether to process more buffers. For possitive @diff, all buffers with
* timestamp <= @timestamp + @diff will certainly arrive late in the sink
* as well. A (negative) @diff value so that @timestamp + @diff would yield a
* result smaller than 0 is not allowed.
*
* The application can use general event probes to intercept the QoS
* event and implement custom application specific QoS handling.
*
* Returns: (transfer full): a new QOS event.
*/
GstEvent *
gst_event_new_qos (GstQOSType type, gdouble proportion,
GstClockTimeDiff diff, GstClockTime timestamp)
{
GstEvent *event;
GstStructure *structure;
/* diff must be positive or timestamp + diff must be positive */
g_return_val_if_fail (diff >= 0 || -diff <= timestamp, NULL);
GST_CAT_LOG (GST_CAT_EVENT,
"creating qos type %d, proportion %lf, diff %" G_GINT64_FORMAT
", timestamp %" GST_TIME_FORMAT, type, proportion,
diff, GST_TIME_ARGS (timestamp));
structure = gst_structure_new_id (GST_QUARK (EVENT_QOS),
GST_QUARK (TYPE), GST_TYPE_QOS_TYPE, type,
GST_QUARK (PROPORTION), G_TYPE_DOUBLE, proportion,
GST_QUARK (DIFF), G_TYPE_INT64, diff,
GST_QUARK (TIMESTAMP), G_TYPE_UINT64, timestamp, NULL);
event = gst_event_new_custom (GST_EVENT_QOS, structure);
return event;
}
