Commit 7a7984a6 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Update documentation

parent 12c3d1e9
......@@ -546,6 +546,187 @@ a `glib::List` of
bytes are not available
<!-- impl Adapter::fn unmap -->
Releases the memory obtained with the last `Adapter::map`.
<!-- struct Aggregator -->
Manages a set of pads with the purpose of aggregating their buffers.
Control is given to the subclass when all pads have data.
* Base class for mixers and muxers. Subclasses should at least implement
the `AggregatorClass.aggregate`() virtual method.
* Installs a `GstPadChainFunction`, a `GstPadEventFullFunction` and a
`GstPadQueryFunction` to queue all serialized data packets per sink pad.
Subclasses should not overwrite those, but instead implement
`AggregatorClass.sink_event`() and `AggregatorClass.sink_query`() as
needed.
* When data is queued on all pads, the aggregate vmethod is called.
* One can peek at the data on any given GstAggregatorPad with the
gst_aggregator_pad_peek_buffer () method, and remove it from the pad
with the gst_aggregator_pad_pop_buffer () method. When a buffer
has been taken with pop_buffer (), a new buffer can be queued
on that pad.
* If the subclass wishes to push a buffer downstream in its aggregate
implementation, it should do so through the
gst_aggregator_finish_buffer () method. This method will take care
of sending and ordering mandatory events such as stream start, caps
and segment.
* Same goes for EOS events, which should not be pushed directly by the
subclass, it should instead return GST_FLOW_EOS in its aggregate
implementation.
* Note that the aggregator logic regarding gap event handling is to turn
these into gap buffers with matching PTS and duration. It will also
flag these buffers with GST_BUFFER_FLAG_GAP and GST_BUFFER_FLAG_DROPPABLE
to ease their identification and subsequent processing.
* Subclasses must use (a subclass of) `AggregatorPad` for both their
sink and source pads.
See `gst::ElementClass::add_static_pad_template_with_gtype`.
This class used to live in gst-plugins-bad and was moved to core.
Feature: `v1_14`
# Implements
[`AggregatorExt`](trait.AggregatorExt.html), [`gst::ElementExt`](../gst/trait.ElementExt.html), [`gst::ObjectExt`](../gst/trait.ObjectExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
<!-- trait AggregatorExt -->
Trait containing all `Aggregator` methods.
Feature: `v1_14`
# Implementors
[`Aggregator`](struct.Aggregator.html)
<!-- trait AggregatorExt::fn finish_buffer -->
This method will push the provided output buffer downstream. If needed,
mandatory events such as stream-start, caps, and segment events will be
sent before pushing the buffer.
Feature: `v1_14`
## `buffer`
the `gst::Buffer` to push.
<!-- trait AggregatorExt::fn get_allocator -->
Lets `Aggregator` sub-classes get the memory `allocator`
acquired by the base class and its `params`.
Unref the `allocator` after use it.
Feature: `v1_14`
## `allocator`
the `gst::Allocator`
used
## `params`
the
`gst::AllocationParams` of `allocator`
<!-- trait AggregatorExt::fn get_buffer_pool -->
Feature: `v1_14`
# Returns
the instance of the `gst::BufferPool` used
by `trans`; free it after use it
<!-- trait AggregatorExt::fn get_latency -->
Retrieves the latency values reported by `self` in response to the latency
query, or `GST_CLOCK_TIME_NONE` if there is not live source connected and the element
will not wait for the clock.
Typically only called by subclasses.
Feature: `v1_14`
# Returns
The latency or `GST_CLOCK_TIME_NONE` if the element does not sync
<!-- trait AggregatorExt::fn set_latency -->
Lets `Aggregator` sub-classes tell the baseclass what their internal
latency is. Will also post a LATENCY message on the bus so the pipeline
can reconfigure its global latency.
Feature: `v1_14`
## `min_latency`
minimum latency
## `max_latency`
maximum latency
<!-- trait AggregatorExt::fn set_src_caps -->
Sets the caps to be used on the src pad.
Feature: `v1_14`
## `caps`
The `gst::Caps` to set on the src pad.
<!-- struct AggregatorPad -->
Pads managed by a `GstAggregor` subclass.
This class used to live in gst-plugins-bad and was moved to core.
Feature: `v1_14`
# Implements
[`AggregatorPadExt`](trait.AggregatorPadExt.html), [`gst::PadExt`](../gst/trait.PadExt.html), [`gst::ObjectExt`](../gst/trait.ObjectExt.html), [`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
<!-- trait AggregatorPadExt -->
Trait containing all `AggregatorPad` methods.
Feature: `v1_14`
# Implementors
[`AggregatorPad`](struct.AggregatorPad.html)
<!-- trait AggregatorPadExt::fn drop_buffer -->
Drop the buffer currently queued in `self`.
Feature: `v1_14`
# Returns
TRUE if there was a buffer queued in `self`, or FALSE if not.
<!-- trait AggregatorPadExt::fn has_buffer -->
Feature: `v1_14_1`
# Returns
`true` if the pad has a buffer available as the next thing.
<!-- trait AggregatorPadExt::fn is_eos -->
Feature: `v1_14`
# Returns
`true` if the pad is EOS, otherwise `false`.
<!-- trait AggregatorPadExt::fn peek_buffer -->
Feature: `v1_14`
# Returns
A reference to the buffer in `self` or
NULL if no buffer was queued. You should unref the buffer after
usage.
<!-- trait AggregatorPadExt::fn pop_buffer -->
Steal the ref to the buffer currently queued in `self`.
Feature: `v1_14`
# Returns
The buffer in `self` or NULL if no buffer was
queued. You should unref the buffer after usage.
<!-- struct BaseSink -->
`BaseSink` is the base class for sink elements in GStreamer, such as
xvimagesink or filesink. It is a layer on top of `gst::Element` that provides a
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<!-- file * -->
<!-- enum WebRTCDTLSSetup -->
GST_WEBRTC_DTLS_SETUP_NONE: none
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly
<!-- struct WebRTCDTLSTransport -->
......@@ -37,6 +42,14 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
# Implements
[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
<!-- enum WebRTCPeerConnectionState -->
GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/`dom`-rtcpeerconnectionstate`</ulink>`
<!-- struct WebRTCRTPReceiver -->
......@@ -55,6 +68,7 @@ GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling
# Implements
[`glib::object::ObjectExt`](../glib/object/trait.ObjectExt.html)
<!-- enum WebRTCRTPTransceiverDirection -->
<!-- enum WebRTCSDPType -->
GST_WEBRTC_SDP_TYPE_OFFER: offer
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
......@@ -62,7 +76,6 @@ GST_WEBRTC_SDP_TYPE_ANSWER: answer
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
See <ulink url="http://w3c.github.io/webrtc-pc/`rtcsdptype`">http://w3c.github.io/webrtc-pc/`rtcsdptype``</ulink>`
<!-- struct WebRTCSessionDescription -->
sdp: the `gst_sdp::SDPMessage` of the description
See <ulink url="https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class">https://www.w3.org/TR/webrtc/`rtcsessiondescription`-class`</ulink>`
<!-- impl WebRTCSessionDescription::fn new -->
## `type_`
......@@ -81,3 +94,26 @@ a new `WebRTCSessionDescription` from `type_`
a new copy of `self`
<!-- impl WebRTCSessionDescription::fn free -->
Free `self` and all associated resources
<!-- enum WebRTCSignalingState -->
GST_WEBRTC_SIGNALING_STATE_STABLE: stable
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
See <ulink url="http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate">http://w3c.github.io/webrtc-pc/`dom`-rtcsignalingstate`</ulink>`
<!-- enum WebRTCStatsType -->
GST_WEBRTC_STATS_CODEC: codec
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
GST_WEBRTC_STATS_CSRC: csrc
GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
GST_WEBRTC_STATS_STREAM: stream
GST_WEBRTC_STATS_TRANSPORT: transport
GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
GST_WEBRTC_STATS_CERTIFICATE: certificate
......@@ -693,7 +693,8 @@ Copy `size` bytes starting from `offset` in `self` to `dest`.
## `offset`
the offset to extract
## `dest`
the destination address
the destination address
## `size`
the size to extract
......@@ -12157,8 +12158,9 @@ outside of the segment is extrapolated.
When 1 is returned, `running_time` resulted in a positive position returned
in `position`.
When this function returns -1, the returned `position` should be negated
to get the real negative segment position.
When this function returns -1, the returned `position` was < 0, and the value
in the position variable should be negated to get the real negative segment
position.
## `format`
the format of the segment.
## `running_time`
......
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