1 # Basic tutorial 14: Handy elements
5 This tutorial gives a list of handy GStreamer elements that are worth
6 knowing. They range from powerful all-in-one elements that allow you to
7 build complex pipelines easily (like `playbin`), to little helper
8 elements which are extremely useful when debugging.
10 For simplicity, the following examples are given using the
11 `gst-launch-1.0` tool (Learn about it in
12 [](sdk-basic-tutorial-gstreamer-tools.md)). Use the `-v` command line
13 parameter if you want to see the Pad Caps that are being negotiated.
17 These are Bin elements which you treat as a single element and they take
18 care of instantiating all the necessary internal pipeline to accomplish
23 This element has been extensively used throughout the tutorials. It
24 manages all aspects of media playback, from source to display, passing
25 through demuxing and decoding. It is so flexible and has so many options
26 that a whole set of tutorials are devoted to it. See the [](sdk-playback-tutorials.md) for more details.
30 This element decodes data from a URI into raw media. It selects a source
31 element that can handle the given URI scheme and connects it to
32 a `decodebin` element. It acts like a demuxer, so it offers as many
33 source pads as streams are found in the
37 gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! videoconvert ! autovideosink
41 gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioconvert ! autoaudiosink
46 This element automatically constructs a decoding pipeline using
47 available decoders and demuxers via auto-plugging until raw media is
48 obtained. It is used internally by `uridecodebin` which is often more
49 convenient to use, as it creates a suitable source element as well. It
50 replaces the old `decodebin` element. It acts like a demuxer, so it
51 offers as many source pads as streams are found in the
55 gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin ! autovideosink
62 This element reads a local file and produces media with `ANY` Caps. If
63 you want to obtain the correct Caps for the media, explore the stream by
64 using a `typefind` element or by setting the `typefind` property
69 gst-launch-1.0 filesrc location=f:\\media\\sintel\\sintel_trailer-480p.webm ! decodebin ! autovideosink
74 This element writes to a file all the media it receives. Use the
75 `location` property to specify the file
79 gst-launch-1.0 audiotestsrc ! vorbisenc ! oggmux ! filesink location=test.ogg
86 This element receives data as a client over the network via HTTP using
87 the [libsoup](https://wiki.gnome.org/Projects/libsoup) library. Set the URL to retrieve through the `location`
91 gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin ! autovideosink
94 ## Test media generation
96 These elements are very useful to check if other parts of the pipeline
97 are working, by replacing the source by one of these test sources which
98 are “guaranteed” to work.
102 This element produces a video pattern (selectable among many different
103 options with the `pattern` property). Use it to test video pipelines.
106 gst-launch-1.0 videotestsrc ! videoconvert ! autovideosink
111 This element produces an audio wave (selectable among many different
112 options with the `wave` property). Use it to test video pipelines.
115 gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
122 This element converts from one color space (e.g. RGB) to another one
123 (e.g. YUV). It can also convert between different YUV formats (e.g.
124 I420, NV12, YUY2 …) or RGB format arrangements (e.g. RGBA, ARGB, BGRA…).
126 This is normally your first choice when solving negotiation problems.
127 When not needed, because its upstream and downstream elements can
128 already understand each other, it acts in pass-through mode having
129 minimal impact on the performance.
131 As a rule of thumb, always use `videoconvert` whenever you use
132 elements whose Caps are unknown at design time, like `autovideosink`, or
133 that can vary depending on external factors, like decoding a
137 gst-launch-1.0 videotestsrc ! videoconvert ! autovideosink
142 This element takes an incoming stream of time-stamped video frames and
143 produces a stream that matches the source pad's frame rate. The
144 correction is performed by dropping and duplicating frames, no fancy
145 algorithm is used to interpolate frames.
147 This is useful to allow elements requiring different frame rates to
148 link. As with the other adapters, if it is not needed (because there is
149 a frame rate on which both Pads can agree), it acts in pass-through mode
150 and does not impact performance.
152 It is therefore a good idea to always use it whenever the actual frame
153 rate is unknown at design time, just in
157 gst-launch-1.0 videotestsrc ! video/x-raw,framerate=30/1 ! videorate ! video/x-raw,framerate=1/1 ! videoconvert ! autovideosink
162 This element resizes video frames. By default the element tries to
163 negotiate to the same size on the source and sink Pads so that no
164 scaling is needed. It is therefore safe to insert this element in a
165 pipeline to get more robust behavior without any cost if no scaling is
168 This element supports a wide range of color spaces including various YUV
169 and RGB formats and is therefore generally able to operate anywhere in a
172 If the video is to be output to a window whose size is controlled by the
173 user, it is a good idea to use a `videoscale` element, since not all
174 video sinks are capable of performing scaling
178 gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! videoscale ! video/x-raw,width=178,height=100 ! videoconvert ! autovideosink
185 This element converts raw audio buffers between various possible
186 formats. It supports integer to float conversion, width/depth
187 conversion, signedness and endianness conversion and channel
190 Like `videoconvert` does for video, you use this to solve
191 negotiation problems with audio, and it is generally safe to use it
192 liberally, since this element does nothing if it is not needed.
195 gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
200 This element resamples raw audio buffers to different sampling rates
201 using a configurable windowing function to enhance quality
203 Again, use it to solve negotiation problems regarding sampling rates and
204 do not fear to use it
208 gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioresample ! audio/x-raw-float,rate=4000 ! audioconvert ! autoaudiosink
213 This element takes an incoming stream of time-stamped raw audio frames
214 and produces a perfect stream by inserting or dropping samples as
215 needed. It does not allow the sample rate to be changed
216 as `videorate` does, it just fills gaps and removes overlapped samples
217 so the output stream is continuous and “clean”.
219 It is useful in situations where the timestamps are going to be lost
220 (when storing into certain file formats, for example) and the receiver
221 will require all samples to be present. It is cumbersome to exemplify
222 this, so no example is given.
224 ![Warning](images/icons/emoticons/warning.png)
225 Most of the time, `audiorate` is not what you want.
231 Queues have been explained in [](sdk-basic-tutorial-multithreading-and-pad-availability.md). Basically, a queue performs two tasks:
233 - Data is queued until a selected limit is reached. Any attempt to
234 push more buffers into the queue blocks the pushing thread until
235 more space becomes available.
236 - The queue creates a new thread on the source Pad to decouple the
237 processing on sink and source Pads.
239 Additionally, `queue` triggers signals when it is about to become empty
240 or full (according to some configurable thresholds), and can be
241 instructed to drop buffers instead of blocking when it is full.
243 As a rule of thumb, prefer the simpler `queue` element
244 over `queue2` whenever network buffering is not a concern to you.
245 See [](sdk-basic-tutorial-multithreading-and-pad-availability.md)
250 This element is not an evolution of `queue`. It has the same design
251 goals but follows a different implementation approach, which results in
252 different features. Unfortunately, it is often not easy to tell which
253 queue is the best choice.
255 `queue2` performs the two tasks listed above for `queue`, and,
256 additionally, is able to store the received data (or part of it) on a
257 disk file, for later retrieval. It also replaces the signals with the
258 more general and convenient buffering messages described in
259 [](sdk-basic-tutorial-streaming.md).
261 As a rule of thumb, prefer `queue2` over `queue` whenever network
262 buffering is a concern to you. See [](sdk-basic-tutorial-streaming.md)
263 for an example (`queue2` is hidden inside `playbin`).
267 This element provides queues for multiple streams simultaneously, and
268 eases their management, by allowing some queues to grow if no data is
269 being received on other streams, or by allowing some queues to drop data
270 if they are not connected to anything (instead of returning an error, as
271 a simpler queue would do). Additionally, it synchronizes the different
272 streams, ensuring that none of them goes too far ahead of the others.
274 This is an advanced element. It is found inside `decodebin`, but you
275 will rarely need to instantiate it yourself in a normal playback
280 [](sdk-basic-tutorial-multithreading-and-pad-availability.md) already
281 showed how to use a `tee` element, which splits data to multiple pads.
282 Splitting the data flow is useful, for example, when capturing a video
283 where the video is shown on the screen and also encoded and written to a
284 file. Another example is playing music and hooking up a visualization
287 One needs to use separate `queue` elements in each branch to provide
288 separate threads for each branch. Otherwise a blocked dataflow in one
289 branch would stall the other
293 gst-launch-1.0 audiotestsrc ! tee name=t ! queue ! audioconvert ! autoaudiosink t. ! queue ! wavescope ! videoconvert ! autovideosink
299 [](sdk-basic-tutorial-gstreamer-tools.md) already
300 explained how to use Caps filters with `gst-launch-1.0`. When building a
301 pipeline programmatically, Caps filters are implemented with
302 the `capsfilter` element. This element does not modify data as such,
303 but enforces limitations on the data format.
306 gst-launch-1.0 videotestsrc ! video/x-raw, format=GRAY8 ! videoconvert ! autovideosink
311 This element determines the type of media a stream contains. It applies
312 typefind functions in the order of their rank. Once the type has been
313 detected it sets its source Pad Caps to the found media type and emits
314 the `have-type` signal.
316 It is instantiated internally by `decodebin`, and you can use it too to
317 find the media type, although you can normally use the
318 `GstDiscoverer` which provides more information (as seen in
319 [](sdk-basic-tutorial-media-information-gathering.md)).
325 This sink element simply swallows any data fed to it. It is useful when
326 debugging, to replace your normal sinks and rule them out of the
327 equation. It can be very verbose when combined with the `-v` switch
328 of `gst-launch-1.0`, so use the `silent` property to remove any unwanted
332 gst-launch-1.0 audiotestsrc num-buffers=1000 ! fakesink sync=false
337 This is a dummy element that passes incoming data through unmodified. It
338 has several useful diagnostic functions, such as offset and timestamp
339 checking, or buffer dropping. Read its documentation to learn all the
340 things this seemingly harmless element can
344 gst-launch-1.0 audiotestsrc ! identity drop-probability=0.1 ! audioconvert ! autoaudiosink
349 This tutorial has listed a few elements which are worth knowing, due to
350 their usefulness in the day-to-day work with GStreamer. Some are
351 valuable for production pipelines, whereas others are only needed for
354 It has been a pleasure having you here, and see you soon!