mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-10-05 02:02:26 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-inspect/launch: Fixes and updates

Browse source
This commit is contained in:
Thibault Saunier 2016-05-27 15:06:22 -04:00
parent e220d85b46
commit 33163869cb
4 changed files with 266 additions and 322 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

1
Installing+for+Android+development.markdown
View file

@ -253,7 +253,6 @@ and used as follows:
include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
GSTREAMER_PLUGINS := $(GSTREAMER_PLUGINS_CORE) $(GSTREAMER_PLUGINS_CODECS) playbin souphttpsrc
#### List of categories and included plugins
| Category | Included plugins |

4
TODO.markdown
View file

@ -48,8 +48,6 @@ Pages to review:
- Multiplatform+deployment+using+Cerbero.markdown
- Legal+information.markdown
- GStreamer+reference.markdown
- gst-inspect.markdown
- gst-launch.markdown
Screenshots:
@ -74,6 +72,8 @@ Reviewed pages:
- GStreamer+reference.markdown
- Playback+tutorial+1+Playbin+usage.markdown
- Basic+tutorial+1+Hello+world.markdown
- gst-inspect.markdown
- gst-launch.markdown
For-later pages:
- Qt+tutorials.markdown [tpm: this should all be rewritten from scratch with qmlglsink; QtGStreamer is outdated and unmaintained, we should not promote it]

103
gst-inspect.markdown
View file

@ -1,13 +1,9 @@
# gst-inspect-1.0
<table>
<tbody>
<tr class="odd">
<td><img src="images/icons/emoticons/information.png" width="16" height="16" /></td>
<td><p><span>This is the Linux man page for the </span><code>gst-inspect-1.0</code><span> 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.</span></p></td>
</tr>
</tbody>
</table>
> ![information] This is the Linux man page for
> the `gst-inspect-1.0` 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.
## Name
@ -20,9 +16,9 @@ gst-inspect-1.0 - print info about a GStreamer plugin or element
## Description
*gst-inspect-1.0* is a tool that prints out information on
available *GStreamer* plugins, information about a particular plugin,
or information about a particular element. When executed with no PLUGIN
or ELEMENT argument, *gst-inspect-1.0* will print a list of all plugins and
available *GStreamer* plugins, information about a particular plugin, or
information about a particular element. When executed with no PLUGIN or
ELEMENT argument, *gst-inspect-1.0* will print a list of all plugins and
elements together with a sumary. When executed with a PLUGIN or ELEMENT
argument, *gst-inspect-1.0* will print information about that plug-in or
element.
@ -79,86 +75,68 @@ Add directories separated with ':' to the plugin search path
## Example
```
gst-inspect-1.0 audiotestsrc
```
should produce:
```
Factory Details:
Long name: Audio test source
Class: Source/Audio
Description: Creates audio test signals of given frequency and volume
Author(s): Stefan Kost <ensonic@users.sf.net>
Rank: none (0)
Rank none (0)
Long-name Audio test source
Klass Source/Audio
Description Creates audio test signals of given frequency and volume
Author Stefan Kost <ensonic@users.sf.net>
Plugin Details:
Name: audiotestsrc
Description: Creates audio test signals of given frequency and volume
Filename: I:\gstreamer-sdk\2012.5\x86\lib\gstreamer-1.0\libgstaudiotestsrc.dll
Version: 0.10.36
License: LGPL
Source module: gst-plugins-base
Source release date: 2012-02-20
Binary package: GStreamer Base Plug-ins source release
Origin URL: Unknown package origin
Name audiotestsrc
Description Creates audio test signals of given frequency and volume
Filename /usr/lib/gstreamer-1.0/libgstaudiotestsrc.so
Version 1.8.1
License LGPL
Source module gst-plugins-base
Source release date 2016-04-20
Binary package GStreamer Base Plugins (Arch Linux)
Origin URL http://www.archlinux.org/
GObject
+----GInitiallyUnowned
+----GstObject
+----GstElement
+----GstBaseSrc
+----GstAudioTestSrc
Pad Templates:
SRC template: 'src'
Availability: Always
Capabilities:
audio/x-raw-int
endianness: 1234
signed: true
width: 16
depth: 16
audio/x-raw
format: { S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE, S8, U8 }
layout: interleaved
rate: [ 1, 2147483647 ]
channels: [ 1, 2 ]
audio/x-raw-int
endianness: 1234
signed: true
width: 32
depth: 32
rate: [ 1, 2147483647 ]
channels: [ 1, 2 ]
audio/x-raw-float
endianness: 1234
width: { 32, 64 }
rate: [ 1, 2147483647 ]
channels: [ 1, 2 ]
channels: [ 1, 2147483647 ]
Element Flags:
no flags set
Element Implementation:
Has change_state() function: gst_base_src_change_state
Has custom save_thyself() function: gst_element_save_thyself
Has custom restore_thyself() function: gst_element_restore_thyself
Element has no clocking capabilities.
Element has no indexing capabilities.
Element has no URI handling capabilities.
Pads:
SRC: 'src'
Implementation:
Has getrangefunc(): gst_base_src_pad_get_range
Has custom eventfunc(): gst_base_src_event_handler
Has custom queryfunc(): gst_base_src_query
Has custom iterintlinkfunc(): gst_pad_iterate_internal_links_default
Has getcapsfunc(): gst_base_src_getcaps
Has setcapsfunc(): gst_base_src_setcaps
Has acceptcapsfunc(): gst_pad_acceptcaps_default
Has fixatecapsfunc(): 62B82E10
Pad Template: 'src'
Element Properties:
name : The name of the object
flags: readable, writable
String. Default: "audiotestsrc0"
parent : The parent of the object
flags: readable, writable
Object of type "GstObject"
blocksize : Size in bytes to read per buffer (-1 = default)
flags: readable, writable
Unsigned Long. Range: 0 - 4294967295 Default: 4294967295
Unsigned Integer. Range: 0 - 4294967295 Default: 4294967295
num-buffers : Number of buffers to output before sending EOS (-1 = unlimited)
flags: readable, writable
Integer. Range: -1 - 2147483647 Default: -1
@ -187,9 +165,9 @@ Element Properties:
(10): red-noise - Red (brownian) noise
(11): blue-noise - Blue noise
(12): violet-noise - Violet noise
freq : Frequency of test signal
freq : Frequency of test signal. The sample rate needs to be at least 4 times higher.
flags: readable, writable, controllable
Double. Range: 0 - 20000 Default: 440
Double. Range: 0 - 5.368709e+08 Default: 440
volume : Volume of test signal
flags: readable, writable, controllable
Double. Range: 0 - 1 Default: 0.8
@ -205,4 +183,5 @@ Element Properties:
can-activate-pull : Can activate in pull mode
flags: readable, writable
Boolean. Default: false
```
[information]: images/icons/emoticons/information.png

336
gst-launch.markdown
View file

@ -1,13 +1,9 @@
# gst-launch-1.0
<table>
<tbody>
<tr class="odd">
<td><img src="images/icons/emoticons/information.png" width="16" height="16" /></td>
<td><p>This is the Linux man page for the <code>gst-launch-1.0</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>
</tr>
</tbody>
</table>
> ![information] This is the Linux man page for
> the `gst-inspect-1.0` 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.
## Name
@ -19,11 +15,12 @@ gst-launch-1.0 - build and run a GStreamer pipeline
## Description
*gst-launch-1.0* is a tool that builds and runs basic *GStreamer* pipelines.
*gst-launch-1.0* is a tool that builds and runs
basic *GStreamer* pipelines.
In simple form, a PIPELINE-DESCRIPTION is a list of elements separated
by exclamation marks (\!). Properties may be appended to elements, in
the form*property=value*.
by exclamation marks (!). Properties may be appended to elements, in the
form*property=value*.
For a complete description of possible PIPELINE-DESCRIPTIONS see the
section*pipeline description* below or consult the GStreamer
@ -75,8 +72,8 @@ time to work.
## Gstreamer Options
*gst-launch-1.0* also accepts the following options that are common to all
GStreamer applications:
*gst-launch-1.0* also accepts the following options that are common to
all GStreamer applications:
## Pipeline Description
@ -94,9 +91,10 @@ Creates an element of type ELEMENTTYPE and sets the PROPERTIES.
PROPERTY=VALUE ...
Sets the property to the specified value. You can use **gst-inspect-1.0**(1)
to find out about properties and allowed values of different elements.
Enumeration properties can be set by name, nick or value.
Sets the property to the specified value. You can
use **gst-inspect-1.0**(1) to find out about properties and allowed
values of different elements. Enumeration properties can be set by name,
nick or value.
**Bins**
@ -112,9 +110,9 @@ pipeline.
**Links**
*\[\[SRCELEMENT\].\[PAD1,...\]\]* \! *\[\[SINKELEMENT\].\[PAD1,...\]\]
\[\[SRCELEMENT\].\[PAD1,...\]\]* \! CAPS
\! *\[\[SINKELEMENT\].\[PAD1,...\]\]*
*\[\[SRCELEMENT\].\[PAD1,...\]\]* ! *\[\[SINKELEMENT\].\[PAD1,...\]\]
\[\[SRCELEMENT\].\[PAD1,...\]\]* ! CAPS
! *\[\[SINKELEMENT\].\[PAD1,...\]\]*
Links the element with name SRCELEMENT to the element with name
SINKELEMENT, using the caps specified in CAPS as a filter. Names can be
@ -124,9 +122,8 @@ used. This works across bins. If a padname is given, the link is done
with these pads. If no pad names are given all possibilities are tried
and a matching pad is used. If multiple padnames are given, both sides
must have the same number of pads specified and multiple links are done
in the given order.
So the simplest link is a simple exclamation mark, that links the
element to the left of it to the element right of it.
in the given order. So the simplest link is a simple exclamation mark,
that links the element to the left of it to the element right of it.
**Caps**
@ -138,35 +135,30 @@ chain caps, you can add more caps in the same format afterwards.
**Properties**
NAME=*\[(TYPE)\]*VALUE
in lists and ranges: *\[(TYPE)\]*VALUE
NAME=*\[(TYPE)\]*VALUE in lists and ranges: *\[(TYPE)\]*VALUE
Sets the requested property in capabilities. The name is an alphanumeric
value and the type can have the following case-insensitive values:
\- **i** or **int** for integer values or ranges
\- **f** or **float** for float values or ranges
\- **4** or **fourcc** for FOURCC values
\- **b**, **bool** or **boolean** for boolean values
\- **s**, **str** or **string** for strings
\- **fraction** for fractions (framerate, pixel-aspect-ratio)
\- **l** or **list** for lists
If no type was given, the following order is tried: integer, float,
boolean, string.
Integer values must be parsable by **strtol()**, floats by **strtod()**.
FOURCC values may either be integers or strings. Boolean values are
(case insensitive) *yes*, *no*, *true* or *false* and may like strings
be escaped with " or '.
Ranges are in this format: \[ VALUE, VALUE \]
Lists use this format: ( VALUE *\[, VALUE ...\]* )
- **i** or **int** for integer values or ranges - **f** or **float** for
float values or ranges - **4** or **fourcc** for FOURCC values
- **b**, **bool** or **boolean** for boolean values
- **s**, **str** or **string** for strings - **fraction** for fractions
(framerate, pixel-aspect-ratio) - **l** or **list** for lists If no type
was given, the following order is tried: integer, float, boolean,
string. Integer values must be parsable by **strtol()**, floats
by **strtod()**. FOURCC values may either be integers or strings.
Boolean values are (case insensitive) *yes*, *no*, *true* or *false* and
may like strings be escaped with " or '. Ranges are in this format: \[
VALUE, VALUE \] Lists use this format: ( VALUE *\[, VALUE ...\]* )
## Pipeline Control
A pipeline can be controlled by signals. SIGUSR2 will stop the pipeline
(GST\_STATE\_NULL); SIGUSR1 will put it back to play
(GST\_STATE\_PLAYING). By default, the pipeline will start in the
playing state.
There are currently no signals defined to go into the ready or pause
(GST\_STATE\_READY and GST\_STATE\_PAUSED) state explicitely.
playing state. There are currently no signals defined to go into the
ready or pause (GST\_STATE\_READY and GST\_STATE\_PAUSED) state
explicitely.
## Pipeline Examples
@ -183,96 +175,89 @@ ffmpegcolorspace (for video) in front of the sink to make things work.
**Audio playback**
**gst-launch-1.0 filesrc location=music.mp3 \! mad \! audioconvert \!
audioresample \! osssink**
Play the mp3 music file "music.mp3" using a libmad-based plug-in and
output to an OSS device
**gst-launch-1.0 filesrc location=music.mp3 ! mad ! audioconvert !
audioresample ! osssink** Play the mp3 music file "music.mp3" using a
libmad-based plug-in and output to an OSS device
**gst-launch-1.0 filesrc location=music.ogg \! oggdemux \! vorbisdec \!
audioconvert \! audioresample \! osssink**
Play an Ogg Vorbis format file
**gst-launch-1.0 filesrc location=music.ogg ! oggdemux ! vorbisdec !
audioconvert ! audioresample ! osssink** Play an Ogg Vorbis format file
**gst-launch-1.0 gnomevfssrc location=music.mp3 \! mad \! osssink
gst-launch-1.0 gnomevfssrc location=<http://domain.com/music.mp3> \! mad \!
audioconvert \! audioresample \! osssink**
Play an mp3 file or an http stream using GNOME-VFS
**gst-launch-1.0 gnomevfssrc location=music.mp3 ! mad ! osssink
gst-launch-1.0 gnomevfssrc location=<http://domain.com/music.mp3> ! mad
! audioconvert ! audioresample ! osssink** Play an mp3 file or an http
stream using GNOME-VFS
**gst-launch-1.0 gnomevfssrc location=<smb://computer/music.mp3> \! mad \!
audioconvert \! audioresample \! osssink**
Use GNOME-VFS to play an mp3 file located on an SMB server
**gst-launch-1.0 gnomevfssrc location=<smb://computer/music.mp3> ! mad !
audioconvert ! audioresample ! osssink** Use GNOME-VFS to play an mp3
file located on an SMB server
**Format conversion**
**gst-launch-1.0 filesrc location=music.mp3 \! mad \! audioconvert \!
vorbisenc \! oggmux \! filesink location=music.ogg**
Convert an mp3 music file to an Ogg Vorbis file
**gst-launch-1.0 filesrc location=music.mp3 ! mad ! audioconvert !
vorbisenc ! oggmux ! filesink location=music.ogg** Convert an mp3 music
file to an Ogg Vorbis file
**gst-launch-1.0 filesrc location=music.mp3 \! mad \! audioconvert \!
flacenc \! filesink location=test.flac**
Convert to the FLAC format
**gst-launch-1.0 filesrc location=music.mp3 ! mad ! audioconvert !
flacenc ! filesink location=test.flac** Convert to the FLAC format
**Other**
**gst-launch-1.0 filesrc location=music.wav \! wavparse \! audioconvert \!
audioresample \! osssink**
Plays a .WAV file that contains raw audio data (PCM).
**gst-launch-1.0 filesrc location=music.wav ! wavparse ! audioconvert !
audioresample ! osssink** Plays a .WAV file that contains raw audio data
(PCM).
**gst-launch-1.0 filesrc location=music.wav \! wavparse \! audioconvert \!
vorbisenc \! oggmux \! filesink location=music.ogg
gst-launch-1.0 filesrc location=music.wav \! wavparse \! audioconvert \!
lame \! filesink location=music.mp3**
Convert a .WAV file containing raw audio data into an Ogg Vorbis or mp3
file
**gst-launch-1.0 filesrc location=music.wav ! wavparse ! audioconvert !
vorbisenc ! oggmux ! filesink location=music.ogg gst-launch-1.0 filesrc
location=music.wav ! wavparse ! audioconvert ! lame ! filesink
location=music.mp3** Convert a .WAV file containing raw audio data into
an Ogg Vorbis or mp3 file
**gst-launch-1.0 cdparanoiasrc mode=continuous \! audioconvert \! lame \!
id3v2mux \! filesink location=cd.mp3**
rips all tracks from compact disc and convert them into a single mp3
file
**gst-launch-1.0 cdparanoiasrc mode=continuous ! audioconvert ! lame !
id3v2mux ! filesink location=cd.mp3** rips all tracks from compact disc
and convert them into a single mp3 file
**gst-launch-1.0 cdparanoiasrc track=5 \! audioconvert \! lame \! id3v2mux
\! filesink location=track5.mp3**
rips track 5 from the CD and converts it into a single mp3 file
**gst-launch-1.0 cdparanoiasrc track=5 ! audioconvert ! lame ! id3v2mux
! filesink location=track5.mp3** rips track 5 from the CD and converts
it into a single mp3 file
Using **gst-inspect-1.0**(1), it is possible to discover settings like the
above for cdparanoiasrc that will tell it to rip the entire cd or only
tracks of it. Alternatively, you can use an URI and gst-launch-1.0 will
find an element (such as cdparanoia) that supports that protocol for
you, e.g.: **gst-launch-1.0 [cdda://5]() \! lame vbr=new vbr-quality=6 \!
filesink location=track5.mp3**
Using **gst-inspect-1.0**(1), it is possible to discover settings like
the above for cdparanoiasrc that will tell it to rip the entire cd or
only tracks of it. Alternatively, you can use an URI and gst-launch-1.0
will find an element (such as cdparanoia) that supports that protocol
for you, e.g.: **gst-launch-1.0 \[cdda://5\] ! lame vbr=new
vbr-quality=6 ! filesink location=track5.mp3**
**gst-launch-1.0 osssrc \! audioconvert \! vorbisenc \! oggmux \! filesink
location=input.ogg**
records sound from your audio input and encodes it into an ogg file
**gst-launch-1.0 osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink
location=input.ogg** records sound from your audio input and encodes it
into an ogg file
**Video**
**gst-launch-1.0 filesrc location=JB\_FF9\_TheGravityOfLove.mpg \! dvddemux
\! mpeg2dec \! xvimagesink**
Display only the video portion of an MPEG-1 video file, outputting to an
X display window
**gst-launch-1.0 filesrc location=JB\_FF9\_TheGravityOfLove.mpg !
dvddemux ! mpeg2dec ! xvimagesink** Display only the video portion of an
MPEG-1 video file, outputting to an X display window
**gst-launch-1.0 filesrc location=/flflfj.vob \! dvddemux \! mpeg2dec \!
sdlvideosink**
Display the video portion of a .vob file (used on DVDs), outputting to
an SDL window
**gst-launch-1.0 filesrc location=/flflfj.vob ! dvddemux ! mpeg2dec !
sdlvideosink** Display the video portion of a .vob file (used on DVDs),
outputting to an SDL window
**gst-launch-1.0 filesrc location=movie.mpg \! dvddemux name=demuxer
demuxer. \! queue \! mpeg2dec \! sdlvideosink demuxer. \! queue \! mad
\! audioconvert \! audioresample \! osssink**
Play both video and audio portions of an MPEG movie
**gst-launch-1.0 filesrc location=movie.mpg ! dvddemux name=demuxer
demuxer. ! queue ! mpeg2dec ! sdlvideosink demuxer. ! queue ! mad !
audioconvert ! audioresample ! osssink** Play both video and audio
portions of an MPEG movie
**gst-launch-1.0 filesrc location=movie.mpg \! mpegdemux name=demuxer
demuxer. \! queue \! mpeg2dec \! ffmpegcolorspace \! sdlvideosink
demuxer. \! queue \! mad \! audioconvert \! audioresample \! osssink**
Play an AVI movie with an external text subtitle stream
**gst-launch-1.0 filesrc location=movie.mpg ! mpegdemux name=demuxer
demuxer. ! queue ! mpeg2dec ! ffmpegcolorspace ! sdlvideosink demuxer. !
queue ! mad ! audioconvert ! audioresample ! osssink** Play an AVI movie
with an external text subtitle stream
This example also shows how to refer to specific pads by name if an
element (here: textoverlay) has multiple sink or source pads.
**gst-launch-1.0 textoverlay name=overlay \! ffmpegcolorspace \! videoscale
\! autovideosink filesrc location=movie.avi \! decodebin2 \!
ffmpegcolorspace \! overlay.video\_sink filesrc location=movie.srt \!
subparse \! overlay.text\_sink**
**gst-launch-1.0 textoverlay name=overlay ! ffmpegcolorspace !
videoscale ! autovideosink filesrc location=movie.avi ! decodebin2 !
ffmpegcolorspace ! overlay.video\_sink filesrc location=movie.srt !
subparse ! overlay.text\_sink**
Play an AVI movie with an external text subtitle stream using playbin
@ -283,45 +268,40 @@ suburi=<file:///path/to/movie.srt>**
Stream video using RTP and network elements.
**gst-launch-1.0 v4l2src \!
video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY \!
ffmpegcolorspace \! ffenc\_h263 \! video/x-h263 \! rtph263ppay pt=96 \!
udpsink host=192.168.1.1 port=5000 sync=false**
Use this command on the receiver
**gst-launch-1.0 v4l2src !
video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY !
ffmpegcolorspace ! ffenc\_h263 ! video/x-h263 ! rtph263ppay pt=96 !
udpsink host=192.168.1.1 port=5000 sync=false** Use this command on the
receiver
**gst-launch-1.0 udpsrc port=5000 \! application/x-rtp,
clock-rate=90000,payload=96 \! rtph263pdepay queue-delay=0 \!
ffdec\_h263 \! xvimagesink**
This command would be run on the transmitter
**gst-launch-1.0 udpsrc port=5000 ! application/x-rtp,
clock-rate=90000,payload=96 ! rtph263pdepay queue-delay=0 ! ffdec\_h263
! xvimagesink** This command would be run on the transmitter
**Diagnostic**
**gst-launch-1.0 -v fakesrc num-buffers=16 \! fakesink**
Generate a null stream and ignore it (and print out details).
**gst-launch-1.0 -v fakesrc num-buffers=16 ! fakesink** Generate a null
stream and ignore it (and print out details).
**gst-launch-1.0 audiotestsrc \! audioconvert \! audioresample \!
osssink**
**gst-launch-1.0 audiotestsrc ! audioconvert ! audioresample ! osssink**
Generate a pure sine tone to test the audio output
**gst-launch-1.0 videotestsrc \! xvimagesink
gst-launch-1.0 videotestsrc \! ximagesink**
Generate a familiar test pattern to test the video output
**gst-launch-1.0 videotestsrc ! xvimagesink gst-launch-1.0 videotestsrc
! ximagesink** Generate a familiar test pattern to test the video output
**Automatic linking**
You can use the decodebin element to automatically select the right
elements to get a working pipeline.
**gst-launch-1.0 filesrc location=musicfile \! decodebin \! audioconvert \!
audioresample \! osssink**
Play any supported audio format
**gst-launch-1.0 filesrc location=musicfile ! decodebin ! audioconvert !
audioresample ! osssink** Play any supported audio format
**gst-launch-1.0 filesrc location=videofile \! decodebin name=decoder
decoder. \! queue \! audioconvert \! audioresample \! osssink decoder.
\! ffmpegcolorspace \! xvimagesink**
Play any supported video format with video and audio output. Threads are
used automatically. To make this even easier, you can use the playbin
element:
**gst-launch-1.0 filesrc location=videofile ! decodebin name=decoder
decoder. ! queue ! audioconvert ! audioresample ! osssink decoder. !
ffmpegcolorspace ! xvimagesink** Play any supported video format with
video and audio output. Threads are used automatically. To make this
even easier, you can use the playbin element:
**gst-launch-1.0 playbin uri=<file:///home/joe/foo.avi>**
@ -329,73 +309,59 @@ element:
These examples show you how to use filtered caps.
**gst-launch-1.0 videotestsrc \!
**gst-launch-1.0 videotestsrc !
'video/x-raw-yuv,format=(fourcc)YUY2;video/x-raw-yuv,format=(fourcc)YV12'
\! xvimagesink**
Show a test image and use the YUY2 or YV12 video format for this.
! xvimagesink** Show a test image and use the YUY2 or YV12 video format
for this.
**gst-launch-1.0 osssrc \!
**gst-launch-1.0 osssrc !
'audio/x-raw-int,rate=\[32000,64000\],width=\[16,32\],depth={16,24,32},signed=(boolean)true'
\! wavenc \! filesink location=recording.wav**
record audio and write it to a .wav file. Force usage of signed 16 to 32
bit samples and a sample rate between 32kHz and 64KHz.
! wavenc ! filesink location=recording.wav** record audio and write it
to a .wav file. Force usage of signed 16 to 32 bit samples and a sample
rate between 32kHz and 64KHz.
## Environment Variables
**GST\_DEBUG**
**GST\_DEBUG**: Comma-separated list of debug categories and levels,
e.g. GST\_DEBUG= totem:4,typefind:5
Comma-separated list of debug categories and levels, e.g. GST\_DEBUG=
totem:4,typefind:5
**GST\_DEBUG\_NO\_COLOR**: When this environment variable is set,
coloured debug output is disabled.
**GST\_DEBUG\_NO\_COLOR**[](http://totem:4,typefind:5)
**GST\_DEBUG\_DUMP\_DOT\_DIR**: When set to a filesystem path, store dot
files of pipeline graphs there.
When this environment variable is set, coloured debug output is
disabled.
**GST\_DEBUG\_DUMP\_DOT\_DIR**
When set to a filesystem path, store dot files of pipeline graphs there.
**GST\_REGISTRY**
Path of the plugin registry file. Default is
~/.gstreamer-1.0/registry-CPU.xml where CPU is the machine/cpu type
**GST\_REGISTRY**: Path of the plugin registry file. Default is
\~/.gstreamer-1.0/registry-CPU.xml where CPU is the machine/cpu type
GStreamer was compiled for, e.g. 'i486', 'i686', 'x86-64', 'ppc', etc.
(check the output of "uname -i" and "uname -m" for details).
**GST\_REGISTRY\_UPDATE**
**GST\_REGISTRY\_UPDATE**: Set to "no" to force GStreamer to assume that
no plugins have changed, been added or been removed. This will make
GStreamer skip the initial check whether a rebuild of the registry cache
is required or not. This may be useful in embedded environments where
the installed plugins never change. Do not use this option in any other
setup.
Set to "no" to force GStreamer to assume that no plugins have changed,
been added or been removed. This will make GStreamer skip the initial
check whether a rebuild of the registry cache is required or not. This
may be useful in embedded environments where the installed plugins never
change. Do not use this option in any other setup.
**GST\_PLUGIN\_PATH**: Specifies a list of directories to scan for
additional plugins. These take precedence over the system plugins.
**GST\_PLUGIN\_PATH**
**GST\_PLUGIN\_SYSTEM\_PATH**: Specifies a list of plugins that are
always loaded by default. If not set, this defaults to the
system-installed path, and the plugins installed in the user's home
directory
Specifies a list of directories to scan for additional plugins. These
take precedence over the system plugins.
**OIL\_CPU\_FLAGS**: Useful liboil environment variable. Set
OIL\_CPU\_FLAGS=0 when valgrind or other debugging tools trip over
liboil's CPU detection (quite a few important GStreamer plugins like
videotestsrc, audioconvert or audioresample use liboil).
**GST\_PLUGIN\_SYSTEM\_PATH**
**G\_DEBUG**: Useful GLib environment variable. Set
G\_DEBUG=fatal\_warnings to make GStreamer programs abort when a
critical warning such as an assertion failure occurs. This is useful if
you want to find out which part of the code caused that warning to be
triggered and under what circumstances. Simply set G\_DEBUG as mentioned
above and run the program in gdb (or let it core dump). Then get a stack
trace in the usual way
Specifies a list of plugins that are always loaded by default. If not
set, this defaults to the system-installed path, and the plugins
installed in the user's home directory
**OIL\_CPU\_FLAGS**
Useful liboil environment variable. Set OIL\_CPU\_FLAGS=0 when valgrind
or other debugging tools trip over liboil's CPU detection (quite a few
important GStreamer plugins like videotestsrc, audioconvert or
audioresample use liboil).
**G\_DEBUG**
Useful GLib environment variable. Set G\_DEBUG=fatal\_warnings to make
GStreamer programs abort when a critical warning such as an assertion
failure occurs. This is useful if you want to find out which part of the
code caused that warning to be triggered and under what circumstances.
Simply set G\_DEBUG as mentioned above and run the program in gdb (or
let it core dump). Then get a stack trace in the usual way
<!-- end list -->
[information]: images/icons/emoticons/information.png