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Basic tutorial 14: Handy elements
Goal
This tutorial gives a list of handy GStreamer elements that are worth
knowing. They range from powerful all-in-one elements that allow you to
build complex pipelines easily (like playbin), to little helper
elements which are extremely useful when debugging.
For simplicity, the following examples are given using the
gst-launch-1.0 tool (Learn about it in
). Use the -v command line
parameter if you want to see the Pad Caps that are being negotiated.
Bins
These are Bin elements which you treat as a single element and they take care of instantiating all the necessary internal pipeline to accomplish their task.
playbin
This element has been extensively used throughout the tutorials. It manages all aspects of media playback, from source to display, passing through demuxing and decoding. It is so flexible and has so many options that a whole set of tutorials are devoted to it. See the for more details.
uridecodebin
This element decodes data from a URI into raw media. It selects a source
element that can handle the given URI scheme and connects it to
a decodebin element. It acts like a demuxer, so it offers as many
source pads as streams are found in the
media.
gst-launch-1.0 uridecodebin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! videoconvert ! autovideosink
gst-launch-1.0 uridecodebin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! audioconvert ! autoaudiosink
decodebin
This element automatically constructs a decoding pipeline using
available decoders and demuxers via auto-plugging until raw media is
obtained. It is used internally by uridecodebin which is often more
convenient to use, as it creates a suitable source element as well. It
replaces the old decodebin element. It acts like a demuxer, so it
offers as many source pads as streams are found in the
media.
gst-launch-1.0 souphttpsrc location=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! decodebin ! autovideosink
File input/output
filesrc
This element reads a local file and produces media with ANY Caps. If
you want to obtain the correct Caps for the media, explore the stream by
using a typefind element or by setting the typefind property
of filesrc to
TRUE.
gst-launch-1.0 filesrc location=f:\\media\\sintel\\sintel_trailer-480p.webm ! decodebin ! autovideosink
filesink
This element writes to a file all the media it receives. Use the
location property to specify the file
name.
gst-launch-1.0 audiotestsrc ! vorbisenc ! oggmux ! filesink location=test.ogg
Network
souphttpsrc
This element receives data as a client over the network via HTTP using
the libsoup library. Set the URL to retrieve through the location
property.
gst-launch-1.0 souphttpsrc location=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! decodebin ! autovideosink
Test media generation
These elements are very useful to check if other parts of the pipeline are working, by replacing the source by one of these test sources which are “guaranteed” to work.
videotestsrc
This element produces a video pattern (selectable among many different
options with the pattern property). Use it to test video pipelines.
gst-launch-1.0 videotestsrc ! videoconvert ! autovideosink
audiotestsrc
This element produces an audio wave (selectable among many different
options with the wave property). Use it to test audio pipelines.
gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
Video adapters
videoconvert
This element converts from one color space (e.g. RGB) to another one (e.g. YUV). It can also convert between different YUV formats (e.g. I420, NV12, YUY2 …) or RGB format arrangements (e.g. RGBA, ARGB, BGRA…).
This is normally your first choice when solving negotiation problems. When not needed, because its upstream and downstream elements can already understand each other, it acts in pass-through mode having minimal impact on the performance.
As a rule of thumb, always use videoconvert whenever you use
elements whose Caps are unknown at design time, like autovideosink, or
that can vary depending on external factors, like decoding a
user-provided file.
gst-launch-1.0 videotestsrc ! videoconvert ! autovideosink
videorate
This element takes an incoming stream of time-stamped video frames and produces a stream that matches the source pad's frame rate. The correction is performed by dropping and duplicating frames, no fancy algorithm is used to interpolate frames.
This is useful to allow elements requiring different frame rates to link. As with the other adapters, if it is not needed (because there is a frame rate on which both Pads can agree), it acts in pass-through mode and does not impact performance.
It is therefore a good idea to always use it whenever the actual frame rate is unknown at design time, just in case.
gst-launch-1.0 videotestsrc ! video/x-raw,framerate=30/1 ! videorate ! video/x-raw,framerate=1/1 ! videoconvert ! autovideosink
videoscale
This element resizes video frames. By default the element tries to negotiate to the same size on the source and sink Pads so that no scaling is needed. It is therefore safe to insert this element in a pipeline to get more robust behavior without any cost if no scaling is needed.
This element supports a wide range of color spaces including various YUV and RGB formats and is therefore generally able to operate anywhere in a pipeline.
If the video is to be output to a window whose size is controlled by the
user, it is a good idea to use a videoscale element, since not all
video sinks are capable of performing scaling
operations.
gst-launch-1.0 uridecodebin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! videoscale ! video/x-raw,width=178,height=100 ! videoconvert ! autovideosink
Audio adapters
audioconvert
This element converts raw audio buffers between various possible formats. It supports integer to float conversion, width/depth conversion, signedness and endianness conversion and channel transformations.
Like videoconvert does for video, you use this to solve
negotiation problems with audio, and it is generally safe to use it
liberally, since this element does nothing if it is not needed.
gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
audioresample
This element resamples raw audio buffers to different sampling rates using a configurable windowing function to enhance quality
Again, use it to solve negotiation problems regarding sampling rates and do not fear to use it generously.
gst-launch-1.0 uridecodebin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm ! audioresample ! audio/x-raw-float,rate=4000 ! audioconvert ! autoaudiosink
audiorate
This element takes an incoming stream of time-stamped raw audio frames
and produces a perfect stream by inserting or dropping samples as
needed. It does not allow the sample rate to be changed
as videorate does, it just fills gaps and removes overlapped samples
so the output stream is continuous and “clean”.
It is useful in situations where the timestamps are going to be lost (when storing into certain file formats, for example) and the receiver will require all samples to be present. It is cumbersome to exemplify this, so no example is given.
Most of the time, audiorate is not what you want.
Multithreading
queue
Queues have been explained in . Basically, a queue performs two tasks:
- Data is queued until a selected limit is reached. Any attempt to push more buffers into the queue blocks the pushing thread until more space becomes available.
- The queue creates a new thread on the source Pad to decouple the processing on sink and source Pads.
Additionally, queue triggers signals when it is about to become empty
or full (according to some configurable thresholds), and can be
instructed to drop buffers instead of blocking when it is full.
As a rule of thumb, prefer the simpler queue element
over queue2 whenever network buffering is not a concern to you.
See
for an example.
queue2
This element is not an evolution of queue. It has the same design
goals but follows a different implementation approach, which results in
different features. Unfortunately, it is often not easy to tell which
queue is the best choice.
queue2 performs the two tasks listed above for queue, and,
additionally, is able to store the received data (or part of it) on a
disk file, for later retrieval. It also replaces the signals with the
more general and convenient buffering messages described in
.
As a rule of thumb, prefer queue2 over queue whenever network
buffering is a concern to you. See
for an example (queue2 is hidden inside playbin).
multiqueue
This element provides queues for multiple streams simultaneously, and eases their management, by allowing some queues to grow if no data is being received on other streams, or by allowing some queues to drop data if they are not connected to anything (instead of returning an error, as a simpler queue would do). Additionally, it synchronizes the different streams, ensuring that none of them goes too far ahead of the others.
This is an advanced element. It is found inside decodebin, but you
will rarely need to instantiate it yourself in a normal playback
application.
tee
already
showed how to use a tee element, which splits data to multiple pads.
Splitting the data flow is useful, for example, when capturing a video
where the video is shown on the screen and also encoded and written to a
file. Another example is playing music and hooking up a visualization
module.
One needs to use separate queue elements in each branch to provide
separate threads for each branch. Otherwise a blocked dataflow in one
branch would stall the other
branches.
gst-launch-1.0 audiotestsrc ! tee name=t ! queue ! audioconvert ! autoaudiosink t. ! queue ! wavescope ! videoconvert ! autovideosink
Capabilities
capsfilter
already
explained how to use Caps filters with gst-launch-1.0. When building a
pipeline programmatically, Caps filters are implemented with
the capsfilter element. This element does not modify data as such,
but enforces limitations on the data format.
gst-launch-1.0 videotestsrc ! video/x-raw, format=GRAY8 ! videoconvert ! autovideosink
typefind
This element determines the type of media a stream contains. It applies
typefind functions in the order of their rank. Once the type has been
detected it sets its source Pad Caps to the found media type and emits
the have-type signal.
It is instantiated internally by decodebin, and you can use it too to
find the media type, although you can normally use the
GstDiscoverer which provides more information (as seen in
).
Debugging
fakesink
This sink element simply swallows any data fed to it. It is useful when
debugging, to replace your normal sinks and rule them out of the
equation. It can be very verbose when combined with the -v switch
of gst-launch-1.0, so use the silent property to remove any unwanted
noise.
gst-launch-1.0 audiotestsrc num-buffers=1000 ! fakesink sync=false
identity
This is a dummy element that passes incoming data through unmodified. It has several useful diagnostic functions, such as offset and timestamp checking, or buffer dropping. Read its documentation to learn all the things this seemingly harmless element can do.
gst-launch-1.0 audiotestsrc ! identity drop-probability=0.1 ! audioconvert ! autoaudiosink
Conclusion
This tutorial has listed a few elements which are worth knowing, due to their usefulness in the day-to-day work with GStreamer. Some are valuable for production pipelines, whereas others are only needed for debugging purposes.
It has been a pleasure having you here, and see you soon!