GstBaseSrc — Base class for getrange based source elements



guint blocksize Read / Write
gboolean do-timestamp Read / Write
gint num-buffers Read / Write
gboolean typefind Read / Write

Types and Values

Object Hierarchy

    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstBaseSrc
                    ╰── GstPushSrc


#include <gst/base/gstbasesrc.h>


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 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:

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:

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:

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",
     "My Source element",
     "The author <>");

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_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.


gst_base_src_wait_playing ()

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



GST_FLOW_OK if src is PLAYING and processing can continue. Any other return value should be returned from the create vmethod.

gst_base_src_start_wait ()

gst_base_src_start_wait (GstBaseSrc *basesrc);

Wait until the start operation completes.



base source instance



a GstFlowReturn.

gst_base_src_start_complete ()

gst_base_src_start_complete (GstBaseSrc *basesrc,
                             GstFlowReturn ret);

Complete an asynchronous start operation. When the subclass overrides the start method, it should call gst_base_src_start_complete() when the start operation completes either from the same thread or from an asynchronous helper thread.



base source instance



a GstFlowReturn


gst_base_src_is_live ()

gst_base_src_is_live (GstBaseSrc *src);

Check if an element is in live mode.



base source instance



TRUE if element is in live mode.

gst_base_src_set_live ()

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


gst_base_src_set_format ()

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


gst_base_src_query_latency ()

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 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.



the source



if the source is live.



the min latency of the source.



the max latency of the source.



TRUE if the query succeeded.

gst_base_src_get_blocksize ()

gst_base_src_get_blocksize (GstBaseSrc *src);

Get the number of bytes that src will push out with each buffer.



the source



the number of bytes pushed with each buffer.

gst_base_src_set_blocksize ()

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


gst_base_src_get_do_timestamp ()

gst_base_src_get_do_timestamp (GstBaseSrc *src);

Query if src timestamps outgoing buffers based on the current running_time.



the source



TRUE if the base class will automatically timestamp outgoing buffers.

gst_base_src_set_do_timestamp ()

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


gst_base_src_set_dynamic_size ()

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


gst_base_src_set_automatic_eos ()

gst_base_src_set_automatic_eos (GstBaseSrc *src,
                                gboolean automatic_eos);

If automatic_eos is TRUE, src 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.



base source instance



automatic eos


Since: 1.4

gst_base_src_new_seamless_segment ()

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 segment



TRUE if preparation of the seamless segment succeeded.

gst_base_src_set_caps ()

gst_base_src_set_caps (GstBaseSrc *src,
                       GstCaps *caps);

Set new caps on the basesrc source pad.



a GstBaseSrc



a GstCaps.

[transfer none]


TRUE if the caps could be set

gst_base_src_get_allocator ()

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 usage.



a GstBaseSrc



the GstAllocator used.

[out][allow-none][transfer full]


the GstAllocationParams of allocator .

[out][allow-none][transfer full]

gst_base_src_get_buffer_pool ()

GstBufferPool *
gst_base_src_get_buffer_pool (GstBaseSrc *src);



a GstBaseSrc



the instance of the GstBufferPool used by the src; unref it after usage.

[transfer full]

gst_base_src_is_async ()

gst_base_src_is_async (GstBaseSrc *src);

Get the current async behaviour of src . See also gst_base_src_set_async().



base source instance



TRUE if src is operating in async mode.

gst_base_src_set_async ()

gst_base_src_set_async (GstBaseSrc *src,
                        gboolean async);

Configure async behaviour in src , 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.



base source instance



new async mode



#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






Types and Values

struct GstBaseSrc

struct GstBaseSrc;

The opaque GstBaseSrc data structure.

struct GstBaseSrcClass

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 the format set by
   * gst_base_src_set_format() */
  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 parent_class;

Element parent class


get_caps ()

Called to get the caps to report


negotiate ()

Negotiated the caps with the peer.


fixate ()

Called during negotiation if caps need fixating. Implement instead of setting a fixate function on the source pad.


set_caps ()

Notify subclass of changed output caps


decide_allocation ()

configure the allocation query


start ()

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 ()

Stop processing. Subclasses should use this to close resources.


get_times ()

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.


get_size ()

Return the total size of the resource, in the format set by gst_base_src_set_format().


is_seekable ()

Check if the source can seek


prepare_seek_segment ()

Prepare the GstSegment that will be passed to the GstBaseSrcClass.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.


do_seek ()

Perform seeking on the resource to the indicated segment.


unlock ()

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 GstBaseSrcClass.create() function call should also return GST_FLOW_FLUSHING until the GstBaseSrcClass.unlock_stop() function has been called.


unlock_stop ()

Clear the previous unlock request. Subclasses should clear any state they set during GstBaseSrcClass.unlock(), such as clearing command queues.


query ()

Handle a requested query.


event ()

Override this to implement custom event handling.


create ()

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 GstBaseSrcClass.alloc() and then call GstBaseSrcClass.fill().


alloc ()

Ask the subclass to allocate a buffer with for offset and size. The default implementation will create a new buffer from the negotiated allocator.


fill ()

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.


enum GstBaseSrcFlags

The GstElement flags that a basesrc element may have.



has source is starting



has source been started



offset to define more flags


Property Details

The “blocksize” property

  “blocksize”                guint

Size in bytes to read per buffer (-1 = default).

Flags: Read / Write

Default value: 4096

The “do-timestamp” property

  “do-timestamp”             gboolean

Apply current stream time to buffers.

Flags: Read / Write

Default value: FALSE

The “num-buffers” property

  “num-buffers”              gint

Number of buffers to output before sending EOS (-1 = unlimited).

Flags: Read / Write

Allowed values: >= -1

Default value: -1

The “typefind” property

  “typefind”                 gboolean

Run typefind before negotiating.

Flags: Read / Write

Default value: FALSE

See Also

GstPushSrc, GstBaseTransform, GstBaseSink