GStreamer 1.0 Library Reference Manual | ||||
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Top | Description | Object Hierarchy | Properties |
#include <gst/base/gstbasesrc.h> struct GstBaseSrc; struct GstBaseSrcClass; enum GstBaseSrcFlags; GstFlowReturn gst_base_src_wait_playing (GstBaseSrc *src
); gboolean gst_base_src_is_live (GstBaseSrc *src
); void gst_base_src_set_live (GstBaseSrc *src
,gboolean live
); void gst_base_src_set_format (GstBaseSrc *src
,GstFormat format
); gboolean gst_base_src_query_latency (GstBaseSrc *src
,gboolean *live
,GstClockTime *min_latency
,GstClockTime *max_latency
); guint gst_base_src_get_blocksize (GstBaseSrc *src
); void gst_base_src_set_blocksize (GstBaseSrc *src
,guint blocksize
); gboolean gst_base_src_get_do_timestamp (GstBaseSrc *src
); void gst_base_src_set_do_timestamp (GstBaseSrc *src
,gboolean timestamp
); void gst_base_src_set_dynamic_size (GstBaseSrc *src
,gboolean dynamic
); gboolean gst_base_src_new_seamless_segment (GstBaseSrc *src
,gint64 start
,gint64 stop
,gint64 time
); gboolean gst_base_src_set_caps (GstBaseSrc *src
,GstCaps *caps
); void gst_base_src_get_allocator (GstBaseSrc *src
,GstAllocator **allocator
,GstAllocationParams *params
); GstBufferPool * gst_base_src_get_buffer_pool (GstBaseSrc *src
); #define GST_BASE_SRC_PAD (obj)
GObject +----GInitiallyUnowned +----GstObject +----GstElement +----GstBaseSrc +----GstPushSrc
"blocksize" guint : Read / Write "do-timestamp" gboolean : Read / Write "num-buffers" gint : Read / Write "typefind" gboolean : Read / Write
This is a generice 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 GstFormat with the
gst_base_src_set_format()
method. The currently set format determines
the format of the internal GstSegment and any GST_EVENT_SEGMENT
events. The default format for GstBaseSrc is GST_FORMAT_BYTES.
GstBaseSrc 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).
GstBaseSrcClass.is_seekable()
returns TRUE
.
If all the conditions are met for operating in pull mode, GstBaseSrc 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:
GstBaseSrcClass.is_seekable()
returns TRUE
.
GstBaseSrcClass.query()
can convert all supported seek formats to the
internal format as set with gst_base_src_set_format()
.
GstBaseSrcClass.do_seek()
is implemented, performs the seek and returns
TRUE
.
When the element does not meet the requirements to operate in pull mode, the
offset and length in the GstBaseSrcClass.create()
method should be ignored.
It is recommended to subclass GstPushSrc 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 GstBaseSrcClass.create()
function.
GstBaseSrc 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 gst_base_src_set_live()
to activate the live source mode.
A live source does not produce data in the PAUSED state. This means that the
GstBaseSrcClass.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_STATE_CHANGE_NO_PREROLL.
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 gst_base_src_wait_playing()
when the
GstBaseSrcClass.create()
function was interrupted by a state change to
PAUSED.
The GstBaseSrcClass.get_times()
method can be used to implement pseudo-live
sources. It only makes sense to implement the GstBaseSrcClass.get_times()
function if the source is a live source. The GstBaseSrcClass.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 GstBaseSrc for exactly one source pad, which should be named "src". A source implementation (subclass of GstBaseSrc) should install a pad template in its class_init function, like so:
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 |
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_pad_template (gstelement_class, gst_static_pad_template_get (&srctemplate)); gst_element_class_set_static_metadata (gstelement_class, "Source name", "Source", "My Source element", "The author <my.sink@my.email>"); } |
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_MESSAGE_SEGMENT_DONE on the bus). This can typically be done
with the gst_element_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.
Last reviewed on 2007-12-19 (0.10.16)
struct GstBaseSrcClass { GstElementClass parent_class; /* virtual methods for subclasses */ /* get caps from subclass */ GstCaps* (*get_caps) (GstBaseSrc *src, GstCaps *filter); /* decide on caps */ gboolean (*negotiate) (GstBaseSrc *src); /* called if, in negotiation, caps need fixating */ GstCaps * (*fixate) (GstBaseSrc *src, GstCaps *caps); /* notify the subclass of new caps */ gboolean (*set_caps) (GstBaseSrc *src, GstCaps *caps); /* setup allocation query */ gboolean (*decide_allocation) (GstBaseSrc *src, GstQuery *query); /* start and stop processing, ideal for opening/closing the resource */ gboolean (*start) (GstBaseSrc *src); gboolean (*stop) (GstBaseSrc *src); /* given a buffer, return start and stop time when it should be pushed * out. The base class will sync on the clock using these times. */ void (*get_times) (GstBaseSrc *src, GstBuffer *buffer, GstClockTime *start, GstClockTime *end); /* get the total size of the resource in bytes */ gboolean (*get_size) (GstBaseSrc *src, guint64 *size); /* check if the resource is seekable */ gboolean (*is_seekable) (GstBaseSrc *src); /* Prepare the segment on which to perform do_seek(), converting to the * current basesrc format. */ gboolean (*prepare_seek_segment) (GstBaseSrc *src, GstEvent *seek, GstSegment *segment); /* notify subclasses of a seek */ gboolean (*do_seek) (GstBaseSrc *src, GstSegment *segment); /* unlock any pending access to the resource. subclasses should unlock * any function ASAP. */ gboolean (*unlock) (GstBaseSrc *src); /* Clear any pending unlock request, as we succeeded in unlocking */ gboolean (*unlock_stop) (GstBaseSrc *src); /* notify subclasses of a query */ gboolean (*query) (GstBaseSrc *src, GstQuery *query); /* notify subclasses of an event */ gboolean (*event) (GstBaseSrc *src, GstEvent *event); /* ask the subclass to create a buffer with offset and size, the default * implementation will call alloc and fill. */ GstFlowReturn (*create) (GstBaseSrc *src, guint64 offset, guint size, GstBuffer **buf); /* ask the subclass to allocate an output buffer. The default implementation * will use the negotiated allocator. */ GstFlowReturn (*alloc) (GstBaseSrc *src, guint64 offset, guint size, GstBuffer **buf); /* ask the subclass to fill the buffer with data from offset and size */ GstFlowReturn (*fill) (GstBaseSrc *src, guint64 offset, guint size, GstBuffer *buf); };
Subclasses can override any of the available virtual methods or not, as
needed. At the minimum, the create
method should be overridden to produce
buffers.
GstElementClass |
Element parent class |
Called to get the caps to report | |
Negotiated the caps with the peer. | |
Called during negotiation if caps need fixating. Implement instead of setting a fixate function on the source pad. | |
Notify subclass of changed output caps | |
configure the allocation query | |
Start processing. Subclasses should open resources and prepare
to produce data. Implementation should call gst_base_src_start_complete()
when the operation completes, either from the current thread or any other
thread that finishes the start operation asynchronously. |
|
Stop processing. Subclasses should use this to close resources. | |
Given a buffer, return the start and stop time when it should be pushed out. The base class will sync on the clock using these times. | |
Return the total size of the resource, in the configured format. | |
Check if the source can seek | |
Prepare the GstSegment that will be passed to the do_seek vmethod for executing a seek request. Sub-classes should override this if they support seeking in formats other than the configured native format. By default, it tries to convert the seek arguments to the configured native format and prepare a segment in that format. | |
Perform seeking on the resource to the indicated segment. | |
Unlock any pending access to the resource. Subclasses should
unblock any blocked function ASAP. In particular, any create() function in
progress should be unblocked and should return GST_FLOW_FLUSHING. Any
future create () function call should also return GST_FLOW_FLUSHING
until the unlock_stop () function has been called. |
|
Clear the previous unlock request. Subclasses should clear
any state they set during unlock() , such as clearing command queues. |
|
Handle a requested query. | |
Override this to implement custom event handling. | |
Ask the subclass to create a buffer with offset and size.
When the subclass returns GST_FLOW_OK, it MUST return a buffer of the
requested size unless fewer bytes are available because an EOS condition
is near. No buffer should be returned when the return value is different
from GST_FLOW_OK. A return value of GST_FLOW_EOS signifies that the
end of stream is reached. The default implementation will call alloc and
then call fill . |
|
Ask the subclass to allocate a buffer with for offset and size. The default implementation will create a new buffer from the negotiated allocator. | |
Ask the subclass to fill the buffer with data for offset and size. The passed buffer is guaranteed to hold the requested amount of bytes. |
typedef enum { GST_BASE_SRC_FLAG_STARTING = (GST_ELEMENT_FLAG_LAST << 0), GST_BASE_SRC_FLAG_STARTED = (GST_ELEMENT_FLAG_LAST << 1), /* padding */ GST_BASE_SRC_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 6) } GstBaseSrcFlags;
The GstElement flags that a basesrc element may have.
GstFlowReturn gst_base_src_wait_playing (GstBaseSrc *src
);
If the GstBaseSrcClass.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_FLOW_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_FLOW_FLUSHING).
|
the src |
Returns : |
GST_FLOW_OK if src is PLAYING and processing can
continue. Any other return value should be returned from the create vmethod. |
gboolean gst_base_src_is_live (GstBaseSrc *src
);
Check if an element is in live mode.
|
base source instance |
Returns : |
TRUE if element is in live mode. |
void gst_base_src_set_live (GstBaseSrc *src
,gboolean live
);
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.
|
base source instance |
|
new live-mode |
void gst_base_src_set_format (GstBaseSrc *src
,GstFormat format
);
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 GstBaseSrcClass.is_seekable()
returns TRUE.
This function must only be called in states < GST_STATE_PAUSED
.
|
base source instance |
|
the format to use |
gboolean gst_base_src_query_latency (GstBaseSrc *src
,gboolean *live
,GstClockTime *min_latency
,GstClockTime *max_latency
);
Query the source for the latency parameters. live
will be TRUE when src
is
configured as a live source. min_latency
will be set to the difference
between the running time and the timestamp of the first buffer.
max_latency
is always the undefined value of -1.
This function is mostly used by subclasses.
|
the source |
|
if the source is live. [out][allow-none] |
|
the min latency of the source. [out][allow-none] |
|
the max latency of the source. [out][allow-none] |
Returns : |
TRUE if the query succeeded. |
guint gst_base_src_get_blocksize (GstBaseSrc *src
);
Get the number of bytes that src
will push out with each buffer.
|
the source |
Returns : |
the number of bytes pushed with each buffer. |
void gst_base_src_set_blocksize (GstBaseSrc *src
,guint blocksize
);
Set the number of bytes that src
will push out with each buffer. When
blocksize
is set to -1, a default length will be used.
|
the source |
|
the new blocksize in bytes |
gboolean gst_base_src_get_do_timestamp (GstBaseSrc *src
);
Query if src
timestamps outgoing buffers based on the current running_time.
|
the source |
Returns : |
TRUE if the base class will automatically timestamp outgoing buffers. |
void gst_base_src_set_do_timestamp (GstBaseSrc *src
,gboolean timestamp
);
Configure src
to automatically timestamp outgoing buffers based on the
current running_time of the pipeline. This property is mostly useful for live
sources.
|
the source |
|
enable or disable timestamping |
void gst_base_src_set_dynamic_size (GstBaseSrc *src
,gboolean dynamic
);
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.
|
base source instance |
|
new dynamic size mode |
gboolean gst_base_src_new_seamless_segment (GstBaseSrc *src
,gint64 start
,gint64 stop
,gint64 time
);
Prepare a new seamless segment for emission downstream. This function must
only be called by derived sub-classes, and only from the 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 gst_base_src_set_format()
|
The source |
|
The new start value for the segment |
|
Stop value for the new segment |
|
The new time value for the start of the new segent |
Returns : |
TRUE if preparation of the seamless segment succeeded. |
gboolean gst_base_src_set_caps (GstBaseSrc *src
,GstCaps *caps
);
Set new caps on the basesrc source pad.
|
a GstBaseSrc |
|
a GstCaps |
Returns : |
TRUE if the caps could be set |
void gst_base_src_get_allocator (GstBaseSrc *src
,GstAllocator **allocator
,GstAllocationParams *params
);
Lets GstBaseSrc sub-classes to know the memory allocator
used by the base class and its params
.
Unref the allocator
after use it.
|
a GstBaseSrc |
|
the GstAllocator used. [out][allow-none][transfer full] |
|
the
GstAllocatorParams of allocator . [out][allow-none][transfer full]
|
GstBufferPool * gst_base_src_get_buffer_pool (GstBaseSrc *src
);
|
a GstBaseSrc |
Returns : |
the instance of the GstBufferPool used by the src; free it after use it. [transfer full] |
#define GST_BASE_SRC_PAD(obj) (GST_BASE_SRC_CAST (obj)->srcpad)
Gives the pointer to the GstPad object of the element.
|
base source instance |
"blocksize"
property"blocksize" guint : Read / Write
Size in bytes to read per buffer (-1 = default).
Default value: 4096
"do-timestamp"
property"do-timestamp" gboolean : Read / Write
Apply current stream time to buffers.
Default value: FALSE
"num-buffers"
property"num-buffers" gint : Read / Write
Number of buffers to output before sending EOS (-1 = unlimited).
Allowed values: >= G_MAXULONG
Default value: -1
"typefind"
property"typefind" gboolean : Read / Write
Run typefind before negotiating.
Default value: FALSE