90 KiB
This class is for elements that receive buffers in an undesired size. While for example raw video contains one image per buffer, the same is not true for a lot of other formats, especially those that come directly from a file. So if you have undefined buffer sizes and require a specific size, this object is for you.
An adapter is created with Adapter::new
. It can be freed again with
gobject::ObjectExt::unref
.
The theory of operation is like this: All buffers received are put
into the adapter using Adapter::push
and the data is then read back
in chunks of the desired size using Adapter::map
/Adapter::unmap
and/or Adapter::copy
. After the data has been processed, it is freed
using Adapter::unmap
.
Other methods such as Adapter::take
and Adapter::take_buffer
combine Adapter::map
and Adapter::unmap
in one method and are
potentially more convenient for some use cases.
For example, a sink pad's chain function that needs to pass data to a library in 512-byte chunks could be implemented like this:
static GstFlowReturn
sink_pad_chain (GstPad *pad, GstObject *parent, GstBuffer *buffer)
{
MyElement *this;
GstAdapter *adapter;
GstFlowReturn ret = GST_FLOW_OK;
this = MY_ELEMENT (parent);
adapter = this->adapter;
// put buffer into adapter
gst_adapter_push (adapter, buffer);
// while we can read out 512 bytes, process them
while (gst_adapter_available (adapter) >= 512 && ret == GST_FLOW_OK) {
const guint8 *data = gst_adapter_map (adapter, 512);
// use flowreturn as an error value
ret = my_library_foo (data);
gst_adapter_unmap (adapter);
gst_adapter_flush (adapter, 512);
}
return ret;
}
For another example, a simple element inside GStreamer that uses Adapter
is the libvisual element.
An element using Adapter
in its sink pad chain function should ensure that
when the FLUSH_STOP event is received, that any queued data is cleared using
Adapter::clear
. Data should also be cleared or processed on EOS and
when changing state from gst::State::Paused
to gst::State::Ready
.
Also check the GST_BUFFER_FLAG_DISCONT flag on the buffer. Some elements might need to clear the adapter after a discontinuity.
The adapter will keep track of the timestamps of the buffers
that were pushed. The last seen timestamp before the current position
can be queried with Adapter::prev_pts
. This function can
optionally return the number of bytes between the start of the buffer that
carried the timestamp and the current adapter position. The distance is
useful when dealing with, for example, raw audio samples because it allows
you to calculate the timestamp of the current adapter position by using the
last seen timestamp and the amount of bytes since. Additionally, the
Adapter::prev_pts_at_offset
can be used to determine the last
seen timestamp at a particular offset in the adapter.
The adapter will also keep track of the offset of the buffers
(GST_BUFFER_OFFSET
) that were pushed. The last seen offset before the
current position can be queried with Adapter::prev_offset
. This function
can optionally return the number of bytes between the start of the buffer
that carried the offset and the current adapter position.
Additionally the adapter also keeps track of the PTS, DTS and buffer offset
at the last discontinuity, which can be retrieved with
Adapter::pts_at_discont
, Adapter::dts_at_discont
and
Adapter::offset_at_discont
. The number of bytes that were consumed
since then can be queried with Adapter::distance_from_discont
.
A last thing to note is that while Adapter
is pretty optimized,
merging buffers still might be an operation that requires a malloc()
and
memcpy()
operation, and these operations are not the fastest. Because of
this, some functions like Adapter::available_fast
are provided to help
speed up such cases should you want to. To avoid repeated memory allocations,
Adapter::copy
can be used to copy data into a (statically allocated)
user provided buffer.
Adapter
is not MT safe. All operations on an adapter must be serialized by
the caller. This is not normally a problem, however, as the normal use case
of Adapter
is inside one pad's chain function, in which case access is
serialized via the pad's STREAM_LOCK.
Note that Adapter::push
takes ownership of the buffer passed. Use
gst::Buffer::ref
before pushing it into the adapter if you still want to
access the buffer later. The adapter will never modify the data in the
buffer pushed in it.
Implements
Creates a new Adapter
. Free with gobject::ObjectExt::unref
.
Returns
a new Adapter
Gets the maximum amount of bytes available, that is it returns the maximum
value that can be supplied to Adapter::map
without that function
returning None
.
Returns
number of bytes available in self
Gets the maximum number of bytes that are immediately available without requiring any expensive operations (like copying the data into a temporary buffer).
Returns
number of bytes that are available in self
without expensive
operations
Removes all buffers from self
.
Copies size
bytes of data starting at offset
out of the buffers
contained in Adapter
into an array dest
provided by the caller.
The array dest
should be large enough to contain size
bytes.
The user should check that the adapter has (offset
+ size
) bytes
available before calling this function.
dest
the memory to copy into
offset
the bytes offset in the adapter to start from
size
the number of bytes to copy
Similar to gst_adapter_copy, but more suitable for language bindings. size
bytes of data starting at offset
will be copied out of the buffers contained
in self
and into a new glib::Bytes
structure which is returned. Depending on
the value of the size
argument an empty glib::Bytes
structure may be returned.
offset
the bytes offset in the adapter to start from
size
the number of bytes to copy
Returns
A new glib::Bytes
structure containing the copied data.
Get the DTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT flag, or GST_CLOCK_TIME_NONE.
Feature: v1_10
Returns
The DTS at the last discont or GST_CLOCK_TIME_NONE.
Flushes the first flush
bytes in the self
. The caller must ensure that
at least this many bytes are available.
See also: Adapter::map
, Adapter::unmap
flush
the number of bytes to flush
Returns a gst::Buffer
containing the first nbytes
of the self
, but
does not flush them from the adapter. See Adapter::take_buffer
for details.
Caller owns a reference to the returned buffer. gst::Buffer::unref
after
usage.
Free-function: gst_buffer_unref
nbytes
the number of bytes to get
Returns
a gst::Buffer
containing the first
nbytes
of the adapter, or None
if nbytes
bytes are not available.
gst::Buffer::unref
when no longer needed.
Returns a gst::Buffer
containing the first nbytes
of the self
, but
does not flush them from the adapter. See Adapter::take_buffer_fast
for details.
Caller owns a reference to the returned buffer. gst::Buffer::unref
after
usage.
Free-function: gst_buffer_unref
nbytes
the number of bytes to get
Returns
a gst::Buffer
containing the first
nbytes
of the adapter, or None
if nbytes
bytes are not available.
gst::Buffer::unref
when no longer needed.
Returns a gst::BufferList
of buffers containing the first nbytes
bytes of
the self
but does not flush them from the adapter. See
Adapter::take_buffer_list
for details.
Caller owns the returned list. Call gst::BufferList::unref
to free
the list after usage.
nbytes
the number of bytes to get
Returns
a gst::BufferList
of buffers containing
the first nbytes
of the adapter, or None
if nbytes
bytes are not
available
Returns a glib::List
of buffers containing the first nbytes
bytes of the
self
, but does not flush them from the adapter. See
Adapter::take_list
for details.
Caller owns returned list and contained buffers. gst::Buffer::unref
each
buffer in the list before freeing the list after usage.
nbytes
the number of bytes to get
Returns
a glib::List
of
buffers containing the first nbytes
of the adapter, or None
if nbytes
bytes are not available
Gets the first size
bytes stored in the self
. The returned pointer is
valid until the next function is called on the adapter.
Note that setting the returned pointer as the data of a gst::Buffer
is
incorrect for general-purpose plugins. The reason is that if a downstream
element stores the buffer so that it has access to it outside of the bounds
of its chain function, the buffer will have an invalid data pointer after
your element flushes the bytes. In that case you should use
Adapter::take
, which returns a freshly-allocated buffer that you can set
as gst::Buffer
memory or the potentially more performant
Adapter::take_buffer
.
Returns None
if size
bytes are not available.
size
the number of bytes to map/peek
Returns
a pointer to the first size
bytes of data, or None
Scan for pattern pattern
with applied mask mask
in the adapter data,
starting from offset offset
.
The bytes in pattern
and mask
are interpreted left-to-right, regardless
of endianness. All four bytes of the pattern must be present in the
adapter for it to match, even if the first or last bytes are masked out.
It is an error to call this function without making sure that there is enough data (offset+size bytes) in the adapter.
This function calls Adapter::masked_scan_uint32_peek
passing None
for value.
mask
mask to apply to data before matching against pattern
pattern
pattern to match (after mask is applied)
offset
offset into the adapter data from which to start scanning, returns the last scanned position.
size
number of bytes to scan from offset
Returns
offset of the first match, or -1 if no match was found.
Example:
// Assume the adapter contains 0x00 0x01 0x02 ... 0xfe 0xff
gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x00010203, 0, 256);
// -> returns 0
gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x00010203, 1, 255);
// -> returns -1
gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x01020304, 1, 255);
// -> returns 1
gst_adapter_masked_scan_uint32 (adapter, 0xffff, 0x0001, 0, 256);
// -> returns -1
gst_adapter_masked_scan_uint32 (adapter, 0xffff, 0x0203, 0, 256);
// -> returns 0
gst_adapter_masked_scan_uint32 (adapter, 0xffff0000, 0x02030000, 0, 256);
// -> returns 2
gst_adapter_masked_scan_uint32 (adapter, 0xffff0000, 0x02030000, 0, 4);
// -> returns -1
Scan for pattern pattern
with applied mask mask
in the adapter data,
starting from offset offset
. If a match is found, the value that matched
is returned through value
, otherwise value
is left untouched.
The bytes in pattern
and mask
are interpreted left-to-right, regardless
of endianness. All four bytes of the pattern must be present in the
adapter for it to match, even if the first or last bytes are masked out.
It is an error to call this function without making sure that there is enough data (offset+size bytes) in the adapter.
mask
mask to apply to data before matching against pattern
pattern
pattern to match (after mask is applied)
offset
offset into the adapter data from which to start scanning, returns the last scanned position.
size
number of bytes to scan from offset
value
pointer to uint32 to return matching data
Returns
offset of the first match, or -1 if no match was found.
Get the offset that was on the last buffer with the GST_BUFFER_FLAG_DISCONT flag, or GST_BUFFER_OFFSET_NONE.
Feature: v1_10
Returns
The offset at the last discont or GST_BUFFER_OFFSET_NONE.
Get the dts that was before the current byte in the adapter. When
distance
is given, the amount of bytes between the dts and the current
position is returned.
The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when the adapter is first created or when it is cleared. This also means that before the first byte with a dts is removed from the adapter, the dts and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
distance
pointer to location for distance, or None
Returns
The previously seen dts.
Get the dts that was before the byte at offset offset
in the adapter. When
distance
is given, the amount of bytes between the dts and the current
position is returned.
The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when the adapter is first created or when it is cleared. This also means that before the first byte with a dts is removed from the adapter, the dts and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
offset
the offset in the adapter at which to get timestamp
distance
pointer to location for distance, or None
Returns
The previously seen dts at given offset.
Get the offset that was before the current byte in the adapter. When
distance
is given, the amount of bytes between the offset and the current
position is returned.
The offset is reset to GST_BUFFER_OFFSET_NONE and the distance is set to 0 when the adapter is first created or when it is cleared. This also means that before the first byte with an offset is removed from the adapter, the offset and distance returned are GST_BUFFER_OFFSET_NONE and 0 respectively.
Feature: v1_10
distance
pointer to a location for distance, or None
Returns
The previous seen offset.
Get the pts that was before the current byte in the adapter. When
distance
is given, the amount of bytes between the pts and the current
position is returned.
The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when the adapter is first created or when it is cleared. This also means that before the first byte with a pts is removed from the adapter, the pts and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
distance
pointer to location for distance, or None
Returns
The previously seen pts.
Get the pts that was before the byte at offset offset
in the adapter. When
distance
is given, the amount of bytes between the pts and the current
position is returned.
The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when the adapter is first created or when it is cleared. This also means that before the first byte with a pts is removed from the adapter, the pts and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
offset
the offset in the adapter at which to get timestamp
distance
pointer to location for distance, or None
Returns
The previously seen pts at given offset.
Get the PTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT flag, or GST_CLOCK_TIME_NONE.
Feature: v1_10
Returns
The PTS at the last discont or GST_CLOCK_TIME_NONE.
Adds the data from buf
to the data stored inside self
and takes
ownership of the buffer.
buf
a gst::Buffer
to add to queue in the adapter
Returns a freshly allocated buffer containing the first nbytes
bytes of the
self
. The returned bytes will be flushed from the adapter.
Caller owns returned value. g_free after usage.
Free-function: g_free
nbytes
the number of bytes to take
Returns
oven-fresh hot data, or None
if nbytes
bytes are not available
Returns a gst::Buffer
containing the first nbytes
bytes of the
self
. The returned bytes will be flushed from the adapter.
This function is potentially more performant than
Adapter::take
since it can reuse the memory in pushed buffers
by subbuffering or merging. This function will always return a
buffer with a single memory region.
Note that no assumptions should be made as to whether certain buffer flags such as the DISCONT flag are set on the returned buffer, or not. The caller needs to explicitly set or unset flags that should be set or unset.
Since 1.6 this will also copy over all GstMeta of the input buffers except
for meta with the gst::MetaFlags::Pooled
flag or with the "memory" tag.
Caller owns a reference to the returned buffer. gst::Buffer::unref
after
usage.
Free-function: gst_buffer_unref
nbytes
the number of bytes to take
Returns
a gst::Buffer
containing the first
nbytes
of the adapter, or None
if nbytes
bytes are not available.
gst::Buffer::unref
when no longer needed.
Returns a gst::Buffer
containing the first nbytes
of the self
.
The returned bytes will be flushed from the adapter. This function
is potentially more performant than Adapter::take_buffer
since
it can reuse the memory in pushed buffers by subbuffering or
merging. Unlike Adapter::take_buffer
, the returned buffer may
be composed of multiple non-contiguous gst::Memory
objects, no
copies are made.
Note that no assumptions should be made as to whether certain buffer flags such as the DISCONT flag are set on the returned buffer, or not. The caller needs to explicitly set or unset flags that should be set or unset.
This will also copy over all GstMeta of the input buffers except
for meta with the gst::MetaFlags::Pooled
flag or with the "memory" tag.
This function can return buffer up to the return value of
Adapter::available
without making copies if possible.
Caller owns a reference to the returned buffer. gst::Buffer::unref
after
usage.
Free-function: gst_buffer_unref
nbytes
the number of bytes to take
Returns
a gst::Buffer
containing the first
nbytes
of the adapter, or None
if nbytes
bytes are not available.
gst::Buffer::unref
when no longer needed.
Returns a gst::BufferList
of buffers containing the first nbytes
bytes of
the self
. The returned bytes will be flushed from the adapter.
When the caller can deal with individual buffers, this function is more
performant because no memory should be copied.
Caller owns the returned list. Call gst::BufferList::unref
to free
the list after usage.
nbytes
the number of bytes to take
Returns
a gst::BufferList
of buffers containing
the first nbytes
of the adapter, or None
if nbytes
bytes are not
available
Returns a glib::List
of buffers containing the first nbytes
bytes of the
self
. The returned bytes will be flushed from the adapter.
When the caller can deal with individual buffers, this function is more
performant because no memory should be copied.
Caller owns returned list and contained buffers. gst::Buffer::unref
each
buffer in the list before freeing the list after usage.
nbytes
the number of bytes to take
Returns
a glib::List
of
buffers containing the first nbytes
of the adapter, or None
if nbytes
bytes are not available
Releases the memory obtained with the last Adapter::map
.
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
, aGstPadEventFullFunction
and aGstPadQueryFunction
to queue all serialized data packets per sink pad. Subclasses should not overwrite those, but instead implementAggregatorClass.sink_event
() andAggregatorClass.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. Seegst::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
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, AggregatorExtManual
Trait containing all Aggregator
methods.
Feature: v1_14
Implementors
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.
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
Feature: v1_14
Returns
the instance of the gst::BufferPool
used
by trans
; free it after use it
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
Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if AggregatorClass.negotiate
() fails.
Feature: v1_18
Returns
true
if the negotiation succeeded, else false
.
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
Sets the caps to be used on the src pad.
Feature: v1_14
caps
The gst::Caps
to set on the src pad.
This is a simple AggregatorClass.get_next_time
() implementation that
just looks at the gst::Segment
on the srcpad of the aggregator and bases
the next time on the running time there.
This is the desired behaviour in most cases where you have a live source and you have a dead line based aggregator subclass.
Feature: v1_16
Returns
The running time based on the position
Subclasses should use this to update the segment on their source pad, instead of directly pushing new segment events downstream.
Feature: v1_18
Force minimum upstream latency (in nanoseconds). When sources with a higher latency are expected to be plugged in dynamically after the aggregator has started playing, this allows overriding the minimum latency reported by the initial source(s). This is only taken into account when larger than the actually reported minimum latency.
Feature: v1_16
Force minimum upstream latency (in nanoseconds). When sources with a higher latency are expected to be plugged in dynamically after the aggregator has started playing, this allows overriding the minimum latency reported by the initial source(s). This is only taken into account when larger than the actually reported minimum latency.
Feature: v1_16
Pads managed by a Aggregator
subclass.
This class used to live in gst-plugins-bad and was moved to core.
Feature: v1_14
Implements
AggregatorPadExt
, gst::PadExt
, gst::ObjectExt
, glib::object::ObjectExt
, AggregatorPadExtManual
Trait containing all AggregatorPad
methods.
Feature: v1_14
Implementors
Drop the buffer currently queued in self
.
Feature: v1_14
Returns
TRUE if there was a buffer queued in self
, or FALSE if not.
This checks if a pad has a buffer available that will be returned by
a call to AggregatorPadExt::peek_buffer
or
AggregatorPadExt::pop_buffer
.
Feature: v1_14_1
Returns
true
if the pad has a buffer available as the next thing.
Feature: v1_14
Returns
true
if the pad is EOS, otherwise false
.
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.
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.
Enables the emission of signals such as AggregatorPad::buffer-consumed
Feature: v1_16
Enables the emission of signals such as AggregatorPad::buffer-consumed
Feature: v1_16
This base class is for parser elements that process data and splits it into separate audio/video/whatever frames.
It provides for:
- provides one sink pad and one source pad
- handles state changes
- can operate in pull mode or push mode
- handles seeking in both modes
- handles events (SEGMENT/EOS/FLUSH)
- handles queries (POSITION/DURATION/SEEKING/FORMAT/CONVERT)
- handles flushing
The purpose of this base class is to provide the basic functionality of a parser and share a lot of rather complex code.
Description of the parsing mechanism:
Set-up phase
-
BaseParse
callsBaseParseClass.start
() to inform subclass that data processing is about to start now. -
BaseParse
class callsBaseParseClass.set_sink_caps
() to inform the subclass about incoming sinkpad caps. Subclass could already set the srcpad caps accordingly, but this might be delayed until callingBaseParse::finish_frame
with a non-queued frame. -
At least at this point subclass needs to tell the
BaseParse
class how big data chunks it wants to receive (minimum frame size ). It can do this withBaseParseExt::set_min_frame_size
. -
BaseParse
class sets up appropriate data passing mode (pull/push) and starts to process the data.
Parsing phase
-
BaseParse
gathers at least min_frame_size bytes of data either by pulling it from upstream or collecting buffers in an internalAdapter
. -
A buffer of (at least) min_frame_size bytes is passed to subclass with
BaseParseClass.handle_frame
(). Subclass checks the contents and can optionally returngst::FlowReturn::Ok
along with an amount of data to be skipped to find a valid frame (which will result in a subsequent DISCONT). If, otherwise, the buffer does not hold a complete frame,BaseParseClass.handle_frame
() can merely return and will be called again when additional data is available. In push mode this amounts to an additional input buffer (thus minimal additional latency), in pull mode this amounts to some arbitrary reasonable buffer size increase.
Of course, BaseParseExt::set_min_frame_size
could also be used if
a very specific known amount of additional data is required. If,
however, the buffer holds a complete valid frame, it can pass the
size of this frame to BaseParse::finish_frame
.
If acting as a converter, it can also merely indicate consumed input
data while simultaneously providing custom output data. Note that
baseclass performs some processing (such as tracking overall consumed
data rate versus duration) for each finished frame, but other state
is only updated upon each call to BaseParseClass.handle_frame
()
(such as tracking upstream input timestamp).
Subclass is also responsible for setting the buffer metadata
(e.g. buffer timestamp and duration, or keyframe if applicable).
(although the latter can also be done by BaseParse
if it is
appropriately configured, see below). Frame is provided with
timestamp derived from upstream (as much as generally possible),
duration obtained from configuration (see below), and offset
if meaningful (in pull mode).
Note that BaseParseClass.handle_frame
() might receive any small
amount of input data when leftover data is being drained (e.g. at
EOS).
-
As part of finish frame processing, just prior to actually pushing the buffer in question, it is passed to
BaseParseClass.pre_push_frame
() which gives subclass yet one last chance to examine buffer metadata, or to send some custom (tag) events, or to perform custom (segment) filtering. -
During the parsing process
BaseParseClass
will handle both srcpad and sinkpad events. They will be passed to subclass ifBaseParseClass.sink_event
() orBaseParseClass.src_event
() implementations have been provided.
Shutdown phase
BaseParse
class callsBaseParseClass.stop
() to inform the subclass that data parsing will be stopped.
Subclass is responsible for providing pad template caps for source and
sink pads. The pads need to be named "sink" and "src". It also needs to
set the fixed caps on srcpad, when the format is ensured (e.g. when
base class calls subclass' BaseParseClass.set_sink_caps
() function).
This base class uses gst::Format::Default
as a meaning of frames. So,
subclass conversion routine needs to know that conversion from
gst::Format::Time
to gst::Format::Default
must return the
frame number that can be found from the given byte position.
BaseParse
uses subclasses conversion methods also for seeking (or
otherwise uses its own default one, see also below).
Subclass start
and stop
functions will be called to inform the beginning
and end of data processing.
Things that subclass need to take care of:
-
Provide pad templates
-
Fixate the source pad caps when appropriate
-
Inform base class how big data chunks should be retrieved. This is done with
BaseParseExt::set_min_frame_size
function. -
Examine data chunks passed to subclass with
BaseParseClass.handle_frame
() and pass proper frame(s) toBaseParse::finish_frame
, and setting src pad caps and timestamps on frame. -
Provide conversion functions
-
Update the duration information with
BaseParse::set_duration
-
Optionally passthrough using
BaseParseExt::set_passthrough
-
Configure various baseparse parameters using
BaseParseExt::set_average_bitrate
,BaseParseExt::set_syncable
andBaseParse::set_frame_rate
. -
In particular, if subclass is unable to determine a duration, but parsing (or specs) yields a frames per seconds rate, then this can be provided to
BaseParse
to enable it to cater for buffer time metadata (which will be taken from upstream as much as possible). Internally keeping track of frame durations and respective sizes that have been pushed providesBaseParse
with an estimated bitrate. A defaultBaseParseClass.convert
() (used if not overridden) will then use these rates to perform obvious conversions. These rates are also used to update (estimated) duration at regular frame intervals.
Implements
BaseParseExt
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, BaseParseExtManual
Trait containing all BaseParse
methods.
Implementors
Adds an entry to the index associating offset
to ts
. It is recommended
to only add keyframe entries. force
allows to bypass checks, such as
whether the stream is (upstream) seekable, another entry is already "close"
to the new entry, etc.
offset
offset of entry
ts
timestamp associated with offset
key
whether entry refers to keyframe
force
add entry disregarding sanity checks
Returns
gboolean
indicating whether entry was added
Default implementation of BaseParseClass.convert
().
src_format
gst::Format
describing the source format.
src_value
Source value to be converted.
dest_format
gst::Format
defining the converted format.
dest_value
Pointer where the conversion result will be put.
Returns
true
if conversion was successful.
Drains the adapter until it is empty. It decreases the min_frame_size to match the current adapter size and calls chain method until the adapter is emptied or chain returns with error.
Feature: v1_12
Collects parsed data and pushes this downstream. Source pad caps must be set when this is called.
If frame
's out_buffer is set, that will be used as subsequent frame data.
Otherwise, size
samples will be taken from the input and used for output,
and the output's metadata (timestamps etc) will be taken as (optionally)
set by the subclass on frame
's (input) buffer (which is otherwise
ignored for any but the above purpose/information).
Note that the latter buffer is invalidated by this call, whereas the
caller retains ownership of frame
.
frame
a BaseParseFrame
size
consumed input data represented by frame
Returns
a gst::FlowReturn
that should be escalated to caller (of caller)
Sets the parser subclass's tags and how they should be merged with any
upstream stream tags. This will override any tags previously-set
with BaseParseExt::merge_tags
.
Note that this is provided for convenience, and the subclass is not required to use this and can still do tag handling on its own.
tags
a gst::TagList
to merge, or NULL to unset
previously-set tags
mode
the gst::TagMergeMode
to use, usually gst::TagMergeMode::Replace
Pushes the frame's buffer downstream, sends any pending events and
does some timestamp and segment handling. Takes ownership of
frame's buffer, though caller retains ownership of frame
.
This must be called with sinkpad STREAM_LOCK held.
frame
a BaseParseFrame
Returns
gst::FlowReturn
Optionally sets the average bitrate detected in media (if non-zero), e.g. based on metadata, as it will be posted to the application.
By default, announced average bitrate is estimated. The average bitrate
is used to estimate the total duration of the stream and to estimate
a seek position, if there's no index and the format is syncable
(see BaseParseExt::set_syncable
).
bitrate
average bitrate in bits/second
Sets the duration of the currently playing media. Subclass can use this
when it is able to determine duration and/or notices a change in the media
duration. Alternatively, if interval
is non-zero (default), then stream
duration is determined based on estimated bitrate, and updated every interval
frames.
fmt
gst::Format
.
duration
duration value.
interval
how often to update the duration estimate based on bitrate, or 0.
If frames per second is configured, parser can take care of buffer duration
and timestamping. When performing segment clipping, or seeking to a specific
location, a corresponding decoder might need an initial lead_in
and a
following lead_out
number of frames to ensure the desired segment is
entirely filled upon decoding.
fps_num
frames per second (numerator).
fps_den
frames per second (denominator).
lead_in
frames needed before a segment for subsequent decode
lead_out
frames needed after a segment
Set if frames carry timing information which the subclass can (generally) parse and provide. In particular, intrinsic (rather than estimated) time can be obtained following a seek.
has_timing
whether frames carry timing information
By default, the base class might try to infer PTS from DTS and vice versa. While this is generally correct for audio data, it may not be otherwise. Sub-classes implementing such formats should disable timestamp inferring.
infer_ts
true
if parser should infer DTS/PTS from each other
Sets the minimum and maximum (which may likely be equal) latency introduced by the parsing process. If there is such a latency, which depends on the particular parsing of the format, it typically corresponds to 1 frame duration.
min_latency
minimum parse latency
max_latency
maximum parse latency
Subclass can use this function to tell the base class that it needs to
be given buffers of at least min_size
bytes.
min_size
Minimum size in bytes of the data that this base class should give to subclass.
Set if the nature of the format or configuration does not allow (much)
parsing, and the parser should operate in passthrough mode (which only
applies when operating in push mode). That is, incoming buffers are
pushed through unmodified, i.e. no BaseParseClass.handle_frame
()
will be invoked, but BaseParseClass.pre_push_frame
() will still be
invoked, so subclass can perform as much or as little is appropriate for
passthrough semantics in BaseParseClass.pre_push_frame
().
passthrough
true
if parser should run in passthrough mode
By default, the base class will guess PTS timestamps using a simple interpolation (previous timestamp + duration), which is incorrect for data streams with reordering, where PTS can go backward. Sub-classes implementing such formats should disable PTS interpolation.
pts_interpolate
true
if parser should interpolate PTS timestamps
Set if frame starts can be identified. This is set by default and determines whether seeking based on bitrate averages is possible for a format/stream.
syncable
set if frame starts can be identified
This function should only be called from a handle_frame
implementation.
BaseParse
creates initial timestamps for frames by using the last
timestamp seen in the stream before the frame starts. In certain
cases, the correct timestamps will occur in the stream after the
start of the frame, but before the start of the actual picture data.
This function can be used to set the timestamps based on the offset
into the frame data that the picture starts.
offset
offset into current buffer
If set to true
, baseparse will unconditionally force parsing of the
incoming data. This can be required in the rare cases where the incoming
side-data (caps, pts, dts, ...) is not trusted by the user and wants to
force validation and parsing of the incoming data.
If set to false
, decision of whether to parse the data or not is up to
the implementation (standard behaviour).
If set to true
, baseparse will unconditionally force parsing of the
incoming data. This can be required in the rare cases where the incoming
side-data (caps, pts, dts, ...) is not trusted by the user and wants to
force validation and parsing of the incoming data.
If set to false
, decision of whether to parse the data or not is up to
the implementation (standard behaviour).
Frame (context) data passed to each frame parsing virtual methods. In addition to providing the data to be checked for a valid frame or an already identified frame, it conveys additional metadata or control information from and to the subclass w.r.t. the particular frame in question (rather than global parameters). Some of these may apply to each parsing stage, others only to some a particular one. These parameters are effectively zeroed at start of each frame's processing, i.e. parsing virtual method invocation sequence.
Allocates a new BaseParseFrame
. This function is mainly for bindings,
elements written in C should usually allocate the frame on the stack and
then use BaseParseFrame::init
to initialise it.
buffer
a gst::Buffer
flags
the flags
overhead
number of bytes in this frame which should be counted as metadata overhead, ie. not used to calculate the average bitrate. Set to -1 to mark the entire frame as metadata. If in doubt, set to 0.
Returns
a newly-allocated BaseParseFrame
. Free with
BaseParseFrame::free
when no longer needed.
Copies a BaseParseFrame
.
Returns
A copy of self
Frees the provided self
.
Sets a BaseParseFrame
to initial state. Currently this means
all public fields are zero-ed and a private flag is set to make
sure BaseParseFrame::free
only frees the contents but not
the actual frame. Use this function to initialise a BaseParseFrame
allocated on the stack.
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
simplified interface to plugin writers. BaseSink
handles many details
for you, for example: preroll, clock synchronization, state changes,
activation in push or pull mode, and queries.
In most cases, when writing sink elements, there is no need to implement
class methods from gst::Element
or to set functions on pads, because the
BaseSink
infrastructure should be sufficient.
BaseSink
provides support for exactly one sink pad, which should be
named "sink". A sink implementation (subclass of BaseSink
) should
install a pad template in its class_init function, like so:
static void
my_element_class_init (GstMyElementClass *klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
// sinktemplate should be a #GstStaticPadTemplate with direction
// %GST_PAD_SINK and name "sink"
gst_element_class_add_static_pad_template (gstelement_class, &sinktemplate);
gst_element_class_set_static_metadata (gstelement_class,
"Sink name",
"Sink",
"My Sink element",
"The author <my.sink@my.email>");
}
BaseSink
will handle the prerolling correctly. This means that it will
return gst::StateChangeReturn::Async
from a state change to PAUSED until the first
buffer arrives in this element. The base class will call the
BaseSinkClass.preroll
() vmethod with this preroll buffer and will then
commit the state change to the next asynchronously pending state.
When the element is set to PLAYING, BaseSink
will synchronise on the
clock using the times returned from BaseSinkClass.get_times
(). If this
function returns GST_CLOCK_TIME_NONE
for the start time, no synchronisation
will be done. Synchronisation can be disabled entirely by setting the object
BaseSink:sync
property to false
.
After synchronisation the virtual method BaseSinkClass.render
() will be
called. Subclasses should minimally implement this method.
Subclasses that synchronise on the clock in the BaseSinkClass.render
()
method are supported as well. These classes typically receive a buffer in
the render method and can then potentially block on the clock while
rendering. A typical example is an audiosink.
These subclasses can use BaseSink::wait_preroll
to perform the
blocking wait.
Upon receiving the EOS event in the PLAYING state, BaseSink
will wait
for the clock to reach the time indicated by the stop time of the last
BaseSinkClass.get_times
() call before posting an EOS message. When the
element receives EOS in PAUSED, preroll completes, the event is queued and an
EOS message is posted when going to PLAYING.
BaseSink
will internally use the gst::EventType::Segment
events to schedule
synchronisation and clipping of buffers. Buffers that fall completely outside
of the current segment are dropped. Buffers that fall partially in the
segment are rendered (and prerolled). Subclasses should do any subbuffer
clipping themselves when needed.
BaseSink
will by default report the current playback position in
gst::Format::Time
based on the current clock time and segment information.
If no clock has been set on the element, the query will be forwarded
upstream.
The BaseSinkClass.set_caps
() function will be called when the subclass
should configure itself to process a specific media type.
The BaseSinkClass.start
() and BaseSinkClass.stop
() virtual methods
will be called when resources should be allocated. Any
BaseSinkClass.preroll
(), BaseSinkClass.render
() and
BaseSinkClass.set_caps
() function will be called between the
BaseSinkClass.start
() and BaseSinkClass.stop
() calls.
The BaseSinkClass.event
() virtual method will be called when an event is
received by BaseSink
. Normally this method should only be overridden by
very specific elements (such as file sinks) which need to handle the
newsegment event specially.
The BaseSinkClass.unlock
() method is called when the elements should
unblock any blocking operations they perform in the
BaseSinkClass.render
() method. This is mostly useful when the
BaseSinkClass.render
() method performs a blocking write on a file
descriptor, for example.
The BaseSink:max-lateness
property affects how the sink deals with
buffers that arrive too late in the sink. A buffer arrives too late in the
sink when the presentation time (as a combination of the last segment, buffer
timestamp and element base_time) plus the duration is before the current
time of the clock.
If the frame is later than max-lateness, the sink will drop the buffer
without calling the render method.
This feature is disabled if sync is disabled, the
BaseSinkClass.get_times
() method does not return a valid start time or
max-lateness is set to -1 (the default).
Subclasses can use BaseSinkExt::set_max_lateness
to configure the
max-lateness value.
The BaseSink:qos
property will enable the quality-of-service features of
the basesink which gather statistics about the real-time performance of the
clock synchronisation. For each buffer received in the sink, statistics are
gathered and a QOS event is sent upstream with these numbers. This
information can then be used by upstream elements to reduce their processing
rate, for example.
The BaseSink:async
property can be used to instruct the sink to never
perform an ASYNC state change. This feature is mostly usable when dealing
with non-synchronized streams or sparse streams.
Implements
BaseSinkExt
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, BaseSinkExtManual
Trait containing all BaseSink
methods.
Implementors
If the self
spawns its own thread for pulling buffers from upstream it
should call this method after it has pulled a buffer. If the element needed
to preroll, this function will perform the preroll and will then block
until the element state is changed.
This function should be called with the PREROLL_LOCK held.
obj
the mini object that caused the preroll
Returns
gst::FlowReturn::Ok
if the preroll completed and processing can
continue. Any other return value should be returned from the render vmethod.
Get the number of bytes that the sink will pull when it is operating in pull mode.
Returns
the number of bytes self
will pull in pull mode.
Checks if self
is currently configured to drop buffers which are outside
the current segment
Feature: v1_12
Returns
true
if the sink is configured to drop buffers outside the
current segment.
Get the last sample that arrived in the sink and was used for preroll or for rendering. This property can be used to generate thumbnails.
The gst::Caps
on the sample can be used to determine the type of the buffer.
Free-function: gst_sample_unref
Returns
a gst::Sample
. gst::Sample::unref
after
usage. This function returns None
when no buffer has arrived in the
sink yet or when the sink is not in PAUSED or PLAYING.
Get the currently configured latency.
Returns
The configured latency.
Get the maximum amount of bits per second that the sink will render.
Returns
the maximum number of bits per second self
will render.
Gets the max lateness value. See BaseSinkExt::set_max_lateness
for
more details.
Returns
The maximum time in nanoseconds that a buffer can be late before it is dropped and not rendered. A value of -1 means an unlimited time.
Get the processing deadline of self
. see
BaseSinkExt::set_processing_deadline
for more information about
the processing deadline.
Feature: v1_16
Returns
the processing deadline
Get the render delay of self
. see BaseSinkExt::set_render_delay
for more
information about the render delay.
Returns
the render delay of self
.
Return various BaseSink
statistics. This function returns a gst::Structure
with name application/x-gst-base-sink-stats
with the following fields:
- "average-rate" G_TYPE_DOUBLE average frame rate
- "dropped" G_TYPE_UINT64 Number of dropped frames
- "rendered" G_TYPE_UINT64 Number of rendered frames
Feature: v1_18
Returns
pointer to gst::Structure
Checks if self
is currently configured to synchronize against the
clock.
Returns
true
if the sink is configured to synchronize against the clock.
Get the time that will be inserted between frames to control the maximum buffers per second.
Returns
the number of nanoseconds self
will put between frames.
Get the synchronisation offset of self
.
Returns
The synchronisation offset.
Checks if self
is currently configured to perform asynchronous state
changes to PAUSED.
Returns
true
if the sink is configured to perform asynchronous state
changes.
Checks if self
is currently configured to store the last received sample in
the last-sample property.
Returns
true
if the sink is configured to store the last received sample.
Checks if self
is currently configured to send Quality-of-Service events
upstream.
Returns
true
if the sink is configured to perform Quality-of-Service.
Query the sink for the latency parameters. The latency will be queried from
the upstream elements. live
will be true
if self
is configured to
synchronize against the clock. upstream_live
will be true
if an upstream
element is live.
If both live
and upstream_live
are true
, the sink will want to compensate
for the latency introduced by the upstream elements by setting the
min_latency
to a strictly positive value.
This function is mostly used by subclasses.
live
if the sink is live
upstream_live
if an upstream element is live
min_latency
the min latency of the upstream elements
max_latency
the max latency of the upstream elements
Returns
true
if the query succeeded.
Configures self
to perform all state changes asynchronously. When async is
disabled, the sink will immediately go to PAUSED instead of waiting for a
preroll buffer. This feature is useful if the sink does not synchronize
against the clock or when it is dealing with sparse streams.
enabled
the new async value.
Set the number of bytes that the sink will pull when it is operating in pull mode.
blocksize
the blocksize in bytes
Configure self
to drop buffers which are outside the current segment
Feature: v1_12
drop_out_of_segment
drop buffers outside the segment
Configures self
to store the last received sample in the last-sample
property.
enabled
the new enable-last-sample value.
Set the maximum amount of bits per second that the sink will render.
max_bitrate
the max_bitrate in bits per second
Sets the new max lateness value to max_lateness
. This value is
used to decide if a buffer should be dropped or not based on the
buffer timestamp and the current clock time. A value of -1 means
an unlimited time.
max_lateness
the new max lateness value.
Maximum amount of time (in nanoseconds) that the pipeline can take for processing the buffer. This is added to the latency of live pipelines.
This function is usually called by subclasses.
Feature: v1_16
processing_deadline
the new processing deadline in nanoseconds.
Configures self
to send Quality-of-Service events upstream.
enabled
the new qos value.
Set the render delay in self
to delay
. The render delay is the time
between actual rendering of a buffer and its synchronisation time. Some
devices might delay media rendering which can be compensated for with this
function.
After calling this function, this sink will report additional latency and other sinks will adjust their latency to delay the rendering of their media.
This function is usually called by subclasses.
delay
the new delay
Configures self
to synchronize on the clock or not. When
sync
is false
, incoming samples will be played as fast as
possible. If sync
is true
, the timestamps of the incoming
buffers will be used to schedule the exact render time of its
contents.
sync
the new sync value.
Set the time that will be inserted between rendered buffers. This can be used to control the maximum buffers per second that the sink will render.
throttle
the throttle time in nanoseconds
Adjust the synchronisation of self
with offset
. A negative value will
render buffers earlier than their timestamp. A positive value will delay
rendering. This function can be used to fix playback of badly timestamped
buffers.
offset
the new offset
This function will wait for preroll to complete and will then block until time
is reached. It is usually called by subclasses that use their own internal
synchronisation but want to let some synchronization (like EOS) be handled
by the base class.
This function should only be called with the PREROLL_LOCK held (like when receiving an EOS event in the ::event vmethod or when handling buffers in ::render).
The time
argument should be the running_time of when the timeout should happen
and will be adjusted with any latency and offset configured in the sink.
time
the running_time to be reached
jitter
the jitter to be filled with time diff, or None
Returns
gst::FlowReturn
This function will block until time
is reached. It is usually called by
subclasses that use their own internal synchronisation.
If time
is not valid, no synchronisation is done and gst::ClockReturn::Badtime
is
returned. Likewise, if synchronisation is disabled in the element or there
is no clock, no synchronisation is done and gst::ClockReturn::Badtime
is returned.
This function should only be called with the PREROLL_LOCK held, like when
receiving an EOS event in the BaseSinkClass.event
() vmethod or when
receiving a buffer in
the BaseSinkClass.render
() vmethod.
The time
argument should be the running_time of when this method should
return and is not adjusted with any latency or offset configured in the
sink.
time
the running_time to be reached
jitter
the jitter to be filled with time diff, or None
Returns
gst::ClockReturn
If the BaseSinkClass.render
() method performs its own synchronisation
against the clock it must unblock when going from PLAYING to the PAUSED state
and call this method before continuing to render the remaining data.
If the BaseSinkClass.render
() method can block on something else than
the clock, it must also be ready to unblock immediately on
the BaseSinkClass.unlock
() method and cause the
BaseSinkClass.render
() method to immediately call this function.
In this case, the subclass must be prepared to continue rendering where it
left off if this function returns gst::FlowReturn::Ok
.
This function will block until a state change to PLAYING happens (in which
case this function returns gst::FlowReturn::Ok
) or the processing must be stopped due
to a state change to READY or a FLUSH event (in which case this function
returns gst::FlowReturn::Flushing
).
This function should only be called with the PREROLL_LOCK held, like in the render function.
Returns
gst::FlowReturn::Ok
if the preroll completed and processing can
continue. Any other return value should be returned from the render vmethod.
If set to true
, the basesink will perform asynchronous state changes.
When set to false
, the sink will not signal the parent when it prerolls.
Use this option when dealing with sparse streams or when synchronisation is
not required.
If set to true
, the basesink will perform asynchronous state changes.
When set to false
, the sink will not signal the parent when it prerolls.
Use this option when dealing with sparse streams or when synchronisation is
not required.
The amount of bytes to pull when operating in pull mode.
The amount of bytes to pull when operating in pull mode.
Enable the last-sample property. If false
, basesink doesn't keep a
reference to the last buffer arrived and the last-sample property is always
set to None
. This can be useful if you need buffers to be released as soon
as possible, eg. if you're using a buffer pool.
Enable the last-sample property. If false
, basesink doesn't keep a
reference to the last buffer arrived and the last-sample property is always
set to None
. This can be useful if you need buffers to be released as soon
as possible, eg. if you're using a buffer pool.
The last buffer that arrived in the sink and was used for preroll or for
rendering. This property can be used to generate thumbnails. This property
can be None
when the sink has not yet received a buffer.
Control the maximum amount of bits that will be rendered per second. Setting this property to a value bigger than 0 will make the sink delay rendering of the buffers when it would exceed to max-bitrate.
Control the maximum amount of bits that will be rendered per second. Setting this property to a value bigger than 0 will make the sink delay rendering of the buffers when it would exceed to max-bitrate.
Maximum amount of time (in nanoseconds) that the pipeline can take for processing the buffer. This is added to the latency of live pipelines.
Feature: v1_16
Maximum amount of time (in nanoseconds) that the pipeline can take for processing the buffer. This is added to the latency of live pipelines.
Feature: v1_16
The additional delay between synchronisation and actual rendering of the media. This property will add additional latency to the device in order to make other sinks compensate for the delay.
The additional delay between synchronisation and actual rendering of the media. This property will add additional latency to the device in order to make other sinks compensate for the delay.
Various BaseSink
statistics. This property returns a gst::Structure
with name application/x-gst-base-sink-stats
with the following fields:
- "average-rate" G_TYPE_DOUBLE average frame rate
- "dropped" G_TYPE_UINT64 Number of dropped frames
- "rendered" G_TYPE_UINT64 Number of rendered frames
Feature: v1_18
The time to insert between buffers. This property can be used to control the maximum amount of buffers per second to render. Setting this property to a value bigger than 0 will make the sink create THROTTLE QoS events.
The time to insert between buffers. This property can be used to control the maximum amount of buffers per second to render. Setting this property to a value bigger than 0 will make the sink create THROTTLE QoS events.
Controls the final synchronisation, a negative value will render the buffer earlier while a positive value delays playback. This property can be used to fix synchronisation in bad files.
Controls the final synchronisation, a negative value will render the buffer earlier while a positive value delays playback. This property can be used to fix synchronisation in bad files.
This is a generic base class for source elements. The following types of sources are supported:
- random access sources like files
- seekable sources
- live sources
The source can be configured to operate in any gst::Format
with the
BaseSrcExt::set_format
method. The currently set format determines
the format of the internal gst::Segment
and any gst::EventType::Segment
events. The default format for BaseSrc
is gst::Format::Bytes
.
BaseSrc
always supports push mode scheduling. If the following
conditions are met, it also supports pull mode scheduling:
- The format is set to
gst::Format::Bytes
(default). BaseSrcClass.is_seekable
() returnstrue
.
If all the conditions are met for operating in pull mode, BaseSrc
is
automatically seekable in push mode as well. The following conditions must
be met to make the element seekable in push mode when the format is not
gst::Format::Bytes
:
BaseSrcClass.is_seekable
() returnstrue
.BaseSrcClass.query
() can convert all supported seek formats to the internal format as set withBaseSrcExt::set_format
.BaseSrcClass.do_seek
() is implemented, performs the seek and returnstrue
.
When the element does not meet the requirements to operate in pull mode, the
offset and length in the BaseSrcClass.create
() method should be ignored.
It is recommended to subclass PushSrc
instead, in this situation. If the
element can operate in pull mode but only with specific offsets and
lengths, it is allowed to generate an error when the wrong values are passed
to the BaseSrcClass.create
() function.
BaseSrc
has support for live sources. Live sources are sources that when
paused discard data, such as audio or video capture devices. A typical live
source also produces data at a fixed rate and thus provides a clock to publish
this rate.
Use BaseSrcExt::set_live
to activate the live source mode.
A live source does not produce data in the PAUSED state. This means that the
BaseSrcClass.create
() method will not be called in PAUSED but only in
PLAYING. To signal the pipeline that the element will not produce data, the
return value from the READY to PAUSED state will be
gst::StateChangeReturn::NoPreroll
.
A typical live source will timestamp the buffers it creates with the current running time of the pipeline. This is one reason why a live source can only produce data in the PLAYING state, when the clock is actually distributed and running.
Live sources that synchronize and block on the clock (an audio source, for
example) can use BaseSrc::wait_playing
when the
BaseSrcClass.create
() function was interrupted by a state change to
PAUSED.
The BaseSrcClass.get_times
() method can be used to implement pseudo-live
sources. It only makes sense to implement the BaseSrcClass.get_times
()
function if the source is a live source. The BaseSrcClass.get_times
()
function should return timestamps starting from 0, as if it were a non-live
source. The base class will make sure that the timestamps are transformed
into the current running_time. The base source will then wait for the
calculated running_time before pushing out the buffer.
For live sources, the base class will by default report a latency of 0. For pseudo live sources, the base class will by default measure the difference between the first buffer timestamp and the start time of get_times and will report this value as the latency. Subclasses should override the query function when this behaviour is not acceptable.
There is only support in BaseSrc
for exactly one source pad, which
should be named "src". A source implementation (subclass of BaseSrc
)
should install a pad template in its class_init function, like so:
static void
my_element_class_init (GstMyElementClass *klass)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
// srctemplate should be a #GstStaticPadTemplate with direction
// %GST_PAD_SRC and name "src"
gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
gst_element_class_set_static_metadata (gstelement_class,
"Source name",
"Source",
"My Source element",
"The author <my.sink@my.email>");
}
Controlled shutdown of live sources in applications
Applications that record from a live source may want to stop recording in a controlled way, so that the recording is stopped, but the data already in the pipeline is processed to the end (remember that many live sources would go on recording forever otherwise). For that to happen the application needs to make the source stop recording and send an EOS event down the pipeline. The application would then wait for an EOS message posted on the pipeline's bus to know when all data has been processed and the pipeline can safely be stopped.
An application may send an EOS event to a source element to make it
perform the EOS logic (send EOS event downstream or post a
gst::MessageType::SegmentDone
on the bus). This can typically be done
with the gst::ElementExt::send_event
function on the element or its parent bin.
After the EOS has been sent to the element, the application should wait for an EOS message to be posted on the pipeline's bus. Once this EOS message is received, it may safely shut down the entire pipeline.
Implements
BaseSrcExt
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, BaseSrcExtManual
Trait containing all BaseSrc
methods.
Implementors
Lets BaseSrc
sub-classes to know the memory allocator
used by the base class and its params
.
Unref the allocator
after usage.
allocator
the gst::Allocator
used
params
the
gst::AllocationParams
of allocator
Get the number of bytes that self
will push out with each buffer.
Returns
the number of bytes pushed with each buffer.
Returns
the instance of the gst::BufferPool
used
by the src; unref it after usage.
Query if self
timestamps outgoing buffers based on the current running_time.
Returns
true
if the base class will automatically timestamp outgoing buffers.
Get the current async behaviour of self
. See also BaseSrcExt::set_async
.
Returns
true
if self
is operating in async mode.
Check if an element is in live mode.
Returns
true
if element is in live mode.
Negotiates src pad caps with downstream elements.
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
if BaseSrcClass.negotiate
() fails.
Do not call this in the BaseSrcClass.fill
() vmethod. Call this in
BaseSrcClass.create
() or in BaseSrcClass.alloc
(), before any
buffer is allocated.
Feature: v1_18
Returns
true
if the negotiation succeeded, else false
.
Prepare a new seamless segment for emission downstream. This function must
only be called by derived sub-classes, and only from the BaseSrcClass::create
function,
as the stream-lock needs to be held.
The format for the new segment will be the current format of the source, as
configured with BaseSrcExt::set_format
start
The new start value for the segment
stop
Stop value for the new segment
time
The new time value for the start of the new segment
Returns
true
if preparation of the seamless segment succeeded.
Query the source for the latency parameters. live
will be true
when self
is
configured as a live source. min_latency
and max_latency
will be set
to the difference between the running time and the timestamp of the first
buffer.
This function is mostly used by subclasses.
live
if the source is live
min_latency
the min latency of the source
max_latency
the max latency of the source
Returns
true
if the query succeeded.
Configure async behaviour in self
, no state change will block. The open,
close, start, stop, play and pause virtual methods will be executed in a
different thread and are thus allowed to perform blocking operations. Any
blocking operation should be unblocked with the unlock vmethod.
async
new async mode
If automatic_eos
is true
, self
will automatically go EOS if a buffer
after the total size is returned. By default this is true
but sources
that can't return an authoritative size and only know that they're EOS
when trying to read more should set this to false
.
When self
operates in gst::Format::Time
, BaseSrc
will send an EOS
when a buffer outside of the currently configured segment is pushed if
automatic_eos
is true
. Since 1.16, if automatic_eos
is false
an
EOS will be pushed only when the BaseSrcClass.create
() implementation
returns gst::FlowReturn::Eos
.
automatic_eos
automatic eos
Set the number of bytes that self
will push out with each buffer. When
blocksize
is set to -1, a default length will be used.
blocksize
the new blocksize in bytes
Set new caps on the basesrc source pad.
caps
a gst::Caps
Returns
true
if the caps could be set
Configure self
to automatically timestamp outgoing buffers based on the
current running_time of the pipeline. This property is mostly useful for live
sources.
timestamp
enable or disable timestamping
If not dynamic
, size is only updated when needed, such as when trying to
read past current tracked size. Otherwise, size is checked for upon each
read.
dynamic
new dynamic size mode
Sets the default format of the source. This will be the format used for sending SEGMENT events and for performing seeks.
If a format of GST_FORMAT_BYTES is set, the element will be able to
operate in pull mode if the BaseSrcClass.is_seekable
() returns true
.
This function must only be called in states < gst::State::Paused
.
format
the format to use
If the element listens to a live source, live
should
be set to true
.
A live source will not produce data in the PAUSED state and will therefore not be able to participate in the PREROLL phase of a pipeline. To signal this fact to the application and the pipeline, the state change return value of the live source will be GST_STATE_CHANGE_NO_PREROLL.
live
new live-mode
Complete an asynchronous start operation. When the subclass overrides the
start method, it should call BaseSrc::start_complete
when the start
operation completes either from the same thread or from an asynchronous
helper thread.
ret
a gst::FlowReturn
Wait until the start operation completes.
Returns
a gst::FlowReturn
.
Subclasses can call this from their create virtual method implementation
to submit a buffer list to be pushed out later. This is useful in
cases where the create function wants to produce multiple buffers to be
pushed out in one go in form of a gst::BufferList
, which can reduce overhead
drastically, especially for packetised inputs (for data streams where
the packetisation/chunking is not important it is usually more efficient
to return larger buffers instead).
Subclasses that use this function from their create function must return
gst::FlowReturn::Ok
and no buffer from their create virtual method implementation.
If a buffer is returned after a buffer list has also been submitted via this
function the behaviour is undefined.
Subclasses must only call this function once per create function call and subclasses must only call this function when the source operates in push mode.
Feature: v1_14
buffer_list
a gst::BufferList
If the BaseSrcClass.create
() method performs its own synchronisation
against the clock it must unblock when going from PLAYING to the PAUSED state
and call this method before continuing to produce the remaining data.
This function will block until a state change to PLAYING happens (in which
case this function returns gst::FlowReturn::Ok
) or the processing must be stopped due
to a state change to READY or a FLUSH event (in which case this function
returns gst::FlowReturn::Flushing
).
Returns
gst::FlowReturn::Ok
if self
is PLAYING and processing can
continue. Any other return value should be returned from the create vmethod.
This base class is for filter elements that process data. Elements
that are suitable for implementation using BaseTransform
are ones
where the size and caps of the output is known entirely from the input
caps and buffer sizes. These include elements that directly transform
one buffer into another, modify the contents of a buffer in-place, as
well as elements that collate multiple input buffers into one output buffer,
or that expand one input buffer into multiple output buffers. See below
for more concrete use cases.
It provides for:
-
one sinkpad and one srcpad
-
Possible formats on sink and source pad implemented with custom transform_caps function. By default uses same format on sink and source.
-
Handles state changes
-
Does flushing
-
Push mode
-
Pull mode if the sub-class transform can operate on arbitrary data
Use Cases
Passthrough mode
-
Element has no interest in modifying the buffer. It may want to inspect it, in which case the element should have a transform_ip function. If there is no transform_ip function in passthrough mode, the buffer is pushed intact.
-
The
BaseTransformClass.passthrough_on_same_caps
variable will automatically set/unset passthrough based on whether the element negotiates the same caps on both pads. -
BaseTransformClass.passthrough_on_same_caps
on an element that doesn't implement a transform_caps function is useful for elements that only inspect data (such as level) -
Example elements
-
Level
-
Videoscale, audioconvert, videoconvert, audioresample in certain modes.
Modifications in-place - input buffer and output buffer are the same thing.
-
The element must implement a transform_ip function.
-
Output buffer size must <= input buffer size
-
If the always_in_place flag is set, non-writable buffers will be copied and passed to the transform_ip function, otherwise a new buffer will be created and the transform function called.
-
Incoming writable buffers will be passed to the transform_ip function immediately.
-
only implementing transform_ip and not transform implies always_in_place =
true
-
Example elements:
-
Volume
-
Audioconvert in certain modes (signed/unsigned conversion)
-
videoconvert in certain modes (endianness swapping)
Modifications only to the caps/metadata of a buffer
-
The element does not require writable data, but non-writable buffers should be subbuffered so that the meta-information can be replaced.
-
Elements wishing to operate in this mode should replace the prepare_output_buffer method to create subbuffers of the input buffer and set always_in_place to
true
-
Example elements
-
Capsfilter when setting caps on outgoing buffers that have none.
-
identity when it is going to re-timestamp buffers by datarate.
Normal mode
- always_in_place flag is not set, or there is no transform_ip function
- Element will receive an input buffer and output buffer to operate on.
- Output buffer is allocated by calling the prepare_output_buffer function.
- Example elements:
- Videoscale, videoconvert, audioconvert when doing scaling/conversions
Special output buffer allocations
-
Elements which need to do special allocation of their output buffers beyond allocating output buffers via the negotiated allocator or buffer pool should implement the prepare_output_buffer method.
-
Example elements:
-
efence
Sub-class settable flags on GstBaseTransform
-
passthrough
-
Implies that in the current configuration, the sub-class is not interested in modifying the buffers.
-
Elements which are always in passthrough mode whenever the same caps has been negotiated on both pads can set the class variable passthrough_on_same_caps to have this behaviour automatically.
-
always_in_place
-
Determines whether a non-writable buffer will be copied before passing to the transform_ip function.
-
Implied
true
if no transform function is implemented. -
Implied
false
if ONLY transform function is implemented.
Implements
BaseTransformExt
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, BaseTransformExtManual
Trait containing all BaseTransform
methods.
Implementors
Lets BaseTransform
sub-classes know the memory allocator
used by the base class and its params
.
Unref the allocator
after use.
allocator
the gst::Allocator
used
params
the
gst::AllocationParams
of allocator
Returns
the instance of the gst::BufferPool
used
by self
; free it after use
See if self
is configured as a in_place transform.
Returns
true
if the transform is configured in in_place mode.
MT safe.
See if self
is configured as a passthrough transform.
Returns
true
if the transform is configured in passthrough mode.
MT safe.
Queries if the transform will handle QoS.
Returns
true
if QoS is enabled.
MT safe.
Negotiates src pad caps with downstream elements if the source pad is marked as needing reconfiguring. Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again if negotiation fails.
Do not call this in the BaseTransformClass.transform
() or
BaseTransformClass.transform_ip
() vmethod. Call this in
BaseTransformClass.submit_input_buffer
(),
BaseTransformClass.prepare_output_buffer
() or in
BaseTransformClass.generate_output
() before any output buffer is
allocated.
It will be default be called when handling an ALLOCATION query or at the
very beginning of the default BaseTransformClass.submit_input_buffer
()
implementation.
Feature: v1_18
Returns
true
if the negotiation succeeded, else false
.
Instructs self
to request renegotiation upstream. This function is
typically called after properties on the transform were set that
influence the input format.
Instructs self
to renegotiate a new downstream transform on the next
buffer. This function is typically called after properties on the transform
were set that influence the output format.
If gap_aware
is false
(the default), output buffers will have the
gst::BufferFlags::Gap
flag unset.
If set to true
, the element must handle output buffers with this flag set
correctly, i.e. it can assume that the buffer contains neutral data but must
unset the flag if the output is no neutral data.
MT safe.
gap_aware
New state
Determines whether a non-writable buffer will be copied before passing to the transform_ip function.
- Always
true
if no transform function is implemented. - Always
false
if ONLY transform function is implemented.
MT safe.
in_place
Boolean value indicating that we would like to operate on in_place buffers.
Set passthrough mode for this filter by default. This is mostly useful for filters that do not care about negotiation.
Always true
for filters which don't implement either a transform
or transform_ip or generate_output method.
MT safe.
passthrough
boolean indicating passthrough mode.
If prefer_passthrough
is true
(the default), self
will check and
prefer passthrough caps from the list of caps returned by the
transform_caps vmethod.
If set to false
, the element must order the caps returned from the
transform_caps function in such a way that the preferred format is
first in the list. This can be interesting for transforms that can do
passthrough transforms but prefer to do something else, like a
capsfilter.
MT safe.
prefer_passthrough
New state
Enable or disable QoS handling in the transform.
MT safe.
enabled
new state
Set the QoS parameters in the transform. This function is called internally when a QOS event is received but subclasses can provide custom information when needed.
MT safe.
proportion
the proportion
diff
the diff against the clock
timestamp
the timestamp of the buffer generating the QoS expressed in running_time.
Updates the srcpad caps and sends the caps downstream. This function can be used by subclasses when they have already negotiated their caps but found a change in them (or computed new information). This way, they can notify downstream about that change without losing any buffer.
updated_caps
An updated version of the srcpad caps to be pushed downstream
Returns
true
if the caps could be sent downstream false
otherwise
Utility struct to help handling gst::FlowReturn
combination. Useful for
gst::Element
s that have multiple source pads and need to combine
the different gst::FlowReturn
for those pads.
FlowCombiner
works by using the last gst::FlowReturn
for all gst::Pad
it has in its list and computes the combined return value and provides
it to the caller.
To add a new pad to the FlowCombiner
use FlowCombiner::add_pad
.
The new gst::Pad
is stored with a default value of gst::FlowReturn::Ok
.
In case you want a gst::Pad
to be removed, use FlowCombiner::remove_pad
.
Please be aware that this struct isn't thread safe as its designed to be used by demuxers, those usually will have a single thread operating it.
These functions will take refs on the passed gst::Pad
s.
Aside from reducing the user's code size, the main advantage of using this
helper struct is to follow the standard rules for gst::FlowReturn
combination.
These rules are:
gst::FlowReturn::Eos
: only if all returns are EOS toogst::FlowReturn::NotLinked
: only if all returns are NOT_LINKED toogst::FlowReturn::Error
or below: if at least one returns an error returngst::FlowReturn::NotNegotiated
: if at least one returns a not-negotiated returngst::FlowReturn::Flushing
: if at least one returns flushinggst::FlowReturn::Ok
: otherwise
gst::FlowReturn::Error
or below, GST_FLOW_NOT_NEGOTIATED and GST_FLOW_FLUSHING are
returned immediately from the FlowCombiner::update_flow
function.
Creates a new FlowCombiner
, use FlowCombiner::free
to free it.
Returns
A new FlowCombiner
Adds a new gst::Pad
to the FlowCombiner
.
pad
the gst::Pad
that is being added
Removes all pads from a FlowCombiner
and resets it to its initial state.
Frees a FlowCombiner
struct and all its internal data.
Increments the reference count on the FlowCombiner
.
Feature: v1_12_1
Returns
the FlowCombiner
.
Removes a gst::Pad
from the FlowCombiner
.
pad
the gst::Pad
to remove
Reset flow combiner and all pads to their initial state without removing pads.
Decrements the reference count on the FlowCombiner
.
Feature: v1_12_1
Computes the combined flow return for the pads in it.
The gst::FlowReturn
parameter should be the last flow return update for a pad
in this FlowCombiner
. It will use this value to be able to shortcut some
combinations and avoid looking over all pads again. e.g. The last combined
return is the same as the latest obtained gst::FlowReturn
.
fret
the latest gst::FlowReturn
received for a pad in this FlowCombiner
Returns
The combined gst::FlowReturn
Sets the provided pad's last flow return to provided value and computes the combined flow return for the pads in it.
The gst::FlowReturn
parameter should be the last flow return update for a pad
in this FlowCombiner
. It will use this value to be able to shortcut some
combinations and avoid looking over all pads again. e.g. The last combined
return is the same as the latest obtained gst::FlowReturn
.
pad
the gst::Pad
whose gst::FlowReturn
to update
fret
the latest gst::FlowReturn
received for a pad in this FlowCombiner
Returns
The combined gst::FlowReturn
This class is mostly useful for elements that cannot do random access, or at least very slowly. The source usually prefers to push out a fixed size buffer.
Subclasses usually operate in a format that is different from the
default GST_FORMAT_BYTES format of BaseSrc
.
Classes extending this base class will usually be scheduled in a push based mode. If the peer accepts to operate without offsets and within the limits of the allowed block size, this class can operate in getrange based mode automatically. To make this possible, the subclass should implement and override the SCHEDULING query.
The subclass should extend the methods from the baseclass in addition to the ::create method.
Seeking, flushing, scheduling and sync is all handled by this base class.
Implements
BaseSrcExt
, gst::ElementExt
, gst::ObjectExt
, glib::object::ObjectExt
, BaseSrcExtManual