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406 lines
14 KiB
Markdown
406 lines
14 KiB
Markdown
# GStreamer SDK documentation : gst-launch
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This page last changed on May 30, 2012 by xartigas.
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<table>
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<tbody>
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<tr class="odd">
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<td><img src="images/icons/emoticons/information.png" width="16" height="16" /></td>
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<td><p>This is the Linux man page for the <code>gst-launch</code> tool. As such, it is very Linux-centric regarding path specification and plugin names. Please be patient while it is rewritten to be more generic.</p></td>
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</tr>
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</tbody>
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</table>
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## Name
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gst-launch - build and run a GStreamer pipeline
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## Synopsis
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**gst-launch** *\[OPTION...\]* PIPELINE-DESCRIPTION
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## Description
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*gst-launch* is a tool that builds and runs basic *GStreamer* pipelines.
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In simple form, a PIPELINE-DESCRIPTION is a list of elements separated
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by exclamation marks (\!). Properties may be appended to elements, in
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the form*property=value*.
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For a complete description of possible PIPELINE-DESCRIPTIONS see the
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section*pipeline description* below or consult the GStreamer
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documentation.
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Please note that *gst-launch* is primarily a debugging tool for
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developers and users. You should not build applications on top of it.
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For applications, use the gst\_parse\_launch() function of the GStreamer
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API as an easy way to construct pipelines from pipeline descriptions.
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## Options
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*gst-launch* accepts the following options:
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**--help**
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Print help synopsis and available FLAGS
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**-v, --verbose**
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Output status information and property notifications
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**-q, --quiet**
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Do not print any progress information
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**-m, --messages**
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Output messages posted on the pipeline's bus
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**-t, --tags**
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Output tags (also known as metadata)
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**-o FILE, --output=FILE**
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Save XML representation of pipeline to FILE and exit
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**-f, --no\_fault**
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Do not install a fault handler
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**-T, --trace**
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Print memory allocation traces. The feature must be enabled at compile
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time to work.
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## Gstreamer Options
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*gst-launch* also accepts the following options that are common to all
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GStreamer applications:
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## Pipeline Description
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A pipeline consists *elements* and *links*. *Elements* can be put
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into *bins* of different sorts. *Elements*, *links* and *bins* can be
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specified in a pipeline description in any order.
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**Elements**
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ELEMENTTYPE *\[PROPERTY1 ...\]*
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Creates an element of type ELEMENTTYPE and sets the PROPERTIES.
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**Properties**
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PROPERTY=VALUE ...
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Sets the property to the specified value. You can use **gst-inspect**(1)
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to find out about properties and allowed values of different elements.
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Enumeration properties can be set by name, nick or value.
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**Bins**
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*\[BINTYPE.\]* ( *\[PROPERTY1 ...\]* PIPELINE-DESCRIPTION )
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Specifies that a bin of type BINTYPE is created and the given properties
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are set. Every element between the braces is put into the bin. Please
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note the dot that has to be used after the BINTYPE. You will almost
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never need this functionality, it is only really useful for applications
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using the gst\_launch\_parse() API with 'bin' as bintype. That way it is
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possible to build partial pipelines instead of a full-fledged top-level
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pipeline.
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**Links**
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*\[\[SRCELEMENT\].\[PAD1,...\]\]* \! *\[\[SINKELEMENT\].\[PAD1,...\]\]
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\[\[SRCELEMENT\].\[PAD1,...\]\]* \! CAPS
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\! *\[\[SINKELEMENT\].\[PAD1,...\]\]*
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Links the element with name SRCELEMENT to the element with name
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SINKELEMENT, using the caps specified in CAPS as a filter. Names can be
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set on elements with the name property. If the name is omitted, the
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element that was specified directly in front of or after the link is
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used. This works across bins. If a padname is given, the link is done
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with these pads. If no pad names are given all possibilities are tried
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and a matching pad is used. If multiple padnames are given, both sides
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must have the same number of pads specified and multiple links are done
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in the given order.
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So the simplest link is a simple exclamation mark, that links the
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element to the left of it to the element right of it.
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**Caps**
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MIMETYPE *\[, PROPERTY\[, PROPERTY ...\]\]\] \[; CAPS\[; CAPS ...\]\]*
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Creates a capability with the given mimetype and optionally with given
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properties. The mimetype can be escaped using " or '. If you want to
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chain caps, you can add more caps in the same format afterwards.
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**Properties**
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NAME=*\[(TYPE)\]*VALUE
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in lists and ranges: *\[(TYPE)\]*VALUE
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Sets the requested property in capabilities. The name is an alphanumeric
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value and the type can have the following case-insensitive values:
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\- **i** or **int** for integer values or ranges
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\- **f** or **float** for float values or ranges
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\- **4** or **fourcc** for FOURCC values
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\- **b**, **bool** or **boolean** for boolean values
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\- **s**, **str** or **string** for strings
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\- **fraction** for fractions (framerate, pixel-aspect-ratio)
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\- **l** or **list** for lists
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If no type was given, the following order is tried: integer, float,
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boolean, string.
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Integer values must be parsable by **strtol()**, floats by **strtod()**.
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FOURCC values may either be integers or strings. Boolean values are
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(case insensitive) *yes*, *no*, *true* or *false* and may like strings
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be escaped with " or '.
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Ranges are in this format: \[ VALUE, VALUE \]
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Lists use this format: ( VALUE *\[, VALUE ...\]* )
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## Pipeline Control
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A pipeline can be controlled by signals. SIGUSR2 will stop the pipeline
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(GST\_STATE\_NULL); SIGUSR1 will put it back to play
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(GST\_STATE\_PLAYING). By default, the pipeline will start in the
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playing state.
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There are currently no signals defined to go into the ready or pause
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(GST\_STATE\_READY and GST\_STATE\_PAUSED) state explicitely.
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## Pipeline Examples
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The examples below assume that you have the correct plug-ins available.
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In general, "osssink" can be substituted with another audio output
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plug-in such as "directsoundsink", "esdsink", "alsasink",
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"osxaudiosink", or "artsdsink". Likewise, "xvimagesink" can be
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substituted with "d3dvideosink", "ximagesink", "sdlvideosink",
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"osxvideosink", or "aasink". Keep in mind though that different sinks
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might accept different formats and even the same sink might accept
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different formats on different machines, so you might need to add
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converter elements like audioconvert and audioresample (for audio) or
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ffmpegcolorspace (for video) in front of the sink to make things work.
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**Audio playback**
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**gst-launch filesrc location=music.mp3 \! mad \! audioconvert \!
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audioresample \! osssink**
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Play the mp3 music file "music.mp3" using a libmad-based plug-in and
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output to an OSS device
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**gst-launch filesrc location=music.ogg \! oggdemux \! vorbisdec \!
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audioconvert \! audioresample \! osssink**
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Play an Ogg Vorbis format file
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**gst-launch gnomevfssrc location=music.mp3 \! mad \! osssink
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gst-launch gnomevfssrc location=<http://domain.com/music.mp3> \! mad \!
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audioconvert \! audioresample \! osssink**
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Play an mp3 file or an http stream using GNOME-VFS
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**gst-launch gnomevfssrc location=<smb://computer/music.mp3> \! mad \!
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audioconvert \! audioresample \! osssink**
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Use GNOME-VFS to play an mp3 file located on an SMB server
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**Format conversion**
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**gst-launch filesrc location=music.mp3 \! mad \! audioconvert \!
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vorbisenc \! oggmux \! filesink location=music.ogg**
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Convert an mp3 music file to an Ogg Vorbis file
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**gst-launch filesrc location=music.mp3 \! mad \! audioconvert \!
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flacenc \! filesink location=test.flac**
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Convert to the FLAC format
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**Other**
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**gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
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audioresample \! osssink**
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Plays a .WAV file that contains raw audio data (PCM).
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**gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
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vorbisenc \! oggmux \! filesink location=music.ogg
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gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
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lame \! filesink location=music.mp3**
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Convert a .WAV file containing raw audio data into an Ogg Vorbis or mp3
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file
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**gst-launch cdparanoiasrc mode=continuous \! audioconvert \! lame \!
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id3v2mux \! filesink location=cd.mp3**
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rips all tracks from compact disc and convert them into a single mp3
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file
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**gst-launch cdparanoiasrc track=5 \! audioconvert \! lame \! id3v2mux
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\! filesink location=track5.mp3**
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rips track 5 from the CD and converts it into a single mp3 file
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Using **gst-inspect**(1), it is possible to discover settings like the
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above for cdparanoiasrc that will tell it to rip the entire cd or only
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tracks of it. Alternatively, you can use an URI and gst-launch-0.10 will
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find an element (such as cdparanoia) that supports that protocol for
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you, e.g.: **gst-launch [cdda://5]() \! lame vbr=new vbr-quality=6 \!
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filesink location=track5.mp3**
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**gst-launch osssrc \! audioconvert \! vorbisenc \! oggmux \! filesink
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location=input.ogg**
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records sound from your audio input and encodes it into an ogg file
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**Video**
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**gst-launch filesrc location=JB\_FF9\_TheGravityOfLove.mpg \! dvddemux
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\! mpeg2dec \! xvimagesink**
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Display only the video portion of an MPEG-1 video file, outputting to an
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X display window
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**gst-launch filesrc location=/flflfj.vob \! dvddemux \! mpeg2dec \!
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sdlvideosink**
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Display the video portion of a .vob file (used on DVDs), outputting to
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an SDL window
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**gst-launch filesrc location=movie.mpg \! dvddemux name=demuxer
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demuxer. \! queue \! mpeg2dec \! sdlvideosink demuxer. \! queue \! mad
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\! audioconvert \! audioresample \! osssink**
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Play both video and audio portions of an MPEG movie
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**gst-launch filesrc location=movie.mpg \! mpegdemux name=demuxer
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demuxer. \! queue \! mpeg2dec \! ffmpegcolorspace \! sdlvideosink
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demuxer. \! queue \! mad \! audioconvert \! audioresample \! osssink**
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Play an AVI movie with an external text subtitle stream
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This example also shows how to refer to specific pads by name if an
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element (here: textoverlay) has multiple sink or source pads.
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**gst-launch textoverlay name=overlay \! ffmpegcolorspace \! videoscale
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\! autovideosink filesrc location=movie.avi \! decodebin2 \!
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ffmpegcolorspace \! overlay.video\_sink filesrc location=movie.srt \!
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subparse \! overlay.text\_sink**
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Play an AVI movie with an external text subtitle stream using playbin2
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**gst-launch playbin2 uri=<file:///path/to/movie.avi>
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suburi=<file:///path/to/movie.srt>**
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**Network streaming**
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Stream video using RTP and network elements.
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**gst-launch v4l2src \!
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video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY \!
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ffmpegcolorspace \! ffenc\_h263 \! video/x-h263 \! rtph263ppay pt=96 \!
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udpsink host=192.168.1.1 port=5000 sync=false**
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Use this command on the receiver
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**gst-launch udpsrc port=5000 \! application/x-rtp,
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clock-rate=90000,payload=96 \! rtph263pdepay queue-delay=0 \!
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ffdec\_h263 \! xvimagesink**
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This command would be run on the transmitter
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**Diagnostic**
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**gst-launch -v fakesrc num-buffers=16 \! fakesink**
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Generate a null stream and ignore it (and print out details).
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**gst-launch audiotestsrc \! audioconvert \! audioresample \!
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osssink**
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Generate a pure sine tone to test the audio output
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**gst-launch videotestsrc \! xvimagesink
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gst-launch videotestsrc \! ximagesink**
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Generate a familiar test pattern to test the video output
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**Automatic linking**
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You can use the decodebin element to automatically select the right
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elements to get a working pipeline.
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**gst-launch filesrc location=musicfile \! decodebin \! audioconvert \!
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audioresample \! osssink**
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Play any supported audio format
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**gst-launch filesrc location=videofile \! decodebin name=decoder
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decoder. \! queue \! audioconvert \! audioresample \! osssink decoder.
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\! ffmpegcolorspace \! xvimagesink**
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Play any supported video format with video and audio output. Threads are
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used automatically. To make this even easier, you can use the playbin
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element:
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**gst-launch playbin uri=<file:///home/joe/foo.avi>**
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**Filtered connections**
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These examples show you how to use filtered caps.
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**gst-launch videotestsrc \!
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'video/x-raw-yuv,format=(fourcc)YUY2;video/x-raw-yuv,format=(fourcc)YV12'
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\! xvimagesink**
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Show a test image and use the YUY2 or YV12 video format for this.
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**gst-launch osssrc \!
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'audio/x-raw-int,rate=\[32000,64000\],width=\[16,32\],depth={16,24,32},signed=(boolean)true'
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\! wavenc \! filesink location=recording.wav**
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record audio and write it to a .wav file. Force usage of signed 16 to 32
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bit samples and a sample rate between 32kHz and 64KHz.
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## Environment Variables
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**GST\_DEBUG**
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Comma-separated list of debug categories and levels, e.g. GST\_DEBUG=
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totem:4,typefind:5
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**GST\_DEBUG\_NO\_COLOR**[](http://totem:4,typefind:5)
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When this environment variable is set, coloured debug output is
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disabled.
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**GST\_DEBUG\_DUMP\_DOT\_DIR**
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When set to a filesystem path, store dot files of pipeline graphs there.
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**GST\_REGISTRY**
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Path of the plugin registry file. Default is
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~/.gstreamer-0.10/registry-CPU.xml where CPU is the machine/cpu type
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GStreamer was compiled for, e.g. 'i486', 'i686', 'x86-64', 'ppc', etc.
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(check the output of "uname -i" and "uname -m" for details).
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**GST\_REGISTRY\_UPDATE**
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Set to "no" to force GStreamer to assume that no plugins have changed,
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been added or been removed. This will make GStreamer skip the initial
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check whether a rebuild of the registry cache is required or not. This
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may be useful in embedded environments where the installed plugins never
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change. Do not use this option in any other setup.
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**GST\_PLUGIN\_PATH**
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Specifies a list of directories to scan for additional plugins. These
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take precedence over the system plugins.
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**GST\_PLUGIN\_SYSTEM\_PATH**
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Specifies a list of plugins that are always loaded by default. If not
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set, this defaults to the system-installed path, and the plugins
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installed in the user's home directory
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**OIL\_CPU\_FLAGS**
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Useful liboil environment variable. Set OIL\_CPU\_FLAGS=0 when valgrind
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or other debugging tools trip over liboil's CPU detection (quite a few
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important GStreamer plugins like videotestsrc, audioconvert or
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audioresample use liboil).
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**G\_DEBUG**
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Useful GLib environment variable. Set G\_DEBUG=fatal\_warnings to make
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GStreamer programs abort when a critical warning such as an assertion
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failure occurs. This is useful if you want to find out which part of the
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code caused that warning to be triggered and under what circumstances.
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Simply set G\_DEBUG as mentioned above and run the program in gdb (or
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let it core dump). Then get a stack trace in the usual way
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<!-- end list -->
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Document generated by Confluence on Oct 08, 2015 10:28
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