Make sure to always suffix GStreamer tools with -1.0

This commit is contained in:
Thibault Saunier 2016-05-27 12:10:42 -04:00
parent 0ea07b873a
commit a4c04e5c43
14 changed files with 124 additions and 124 deletions

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@ -125,7 +125,7 @@ into an actual pipeline, which is very handy. In fact, this function is
so handy there is a tool built completely around it which you will get
very acquainted with (see [Basic tutorial 10: GStreamer
tools](Basic+tutorial+10+GStreamer+tools.markdown) to
learn about `gst-launch` and the `gst-launch` syntax).
learn about `gst-launch-1.0` and the `gst-launch-1.0` syntax).
### playbin

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@ -35,10 +35,10 @@ and you are ready to start typing the commands given in this tutorial.
In order to allow for multiple versions of GStreamer to coexists in the
same system, these tools are versioned, this is, a GStreamer version
number is appended to their name. This version of the SDK is based on
GStreamer 0.10, so the tools are called `gst-launch-0.10`,
`gst-inspect-0.10` and `gst-discoverer-0.10`
GStreamer 1.0, so the tools are called `gst-launch-1.0`,
`gst-inspect-1.0` and `gst-discoverer-1.0`
# `gst-launch`
# `gst-launch-1.0`
This tool accepts a textual description of a pipeline, instantiates it,
and sets it to the PLAYING state. It allows you to quickly check if a
@ -51,7 +51,7 @@ up to a certain level. In any case, it is extremely handy to test
pipelines quickly, and is used by GStreamer developers around the world
on a daily basis.
Please note that `gst-launch` is primarily a debugging tool for
Please note that `gst-launch-1.0` is primarily a debugging tool for
developers. You should not build applications on top of it. Instead, use
the `gst_parse_launch()` function of the GStreamer API as an easy way to
construct pipelines from pipeline descriptions.
@ -59,12 +59,12 @@ construct pipelines from pipeline descriptions.
Although the rules to construct pipeline descriptions are very simple,
the concatenation of multiple elements can quickly make such
descriptions resemble black magic. Fear not, for everyone learns the
`gst-launch` syntax, eventually.
`gst-launch-1.0` syntax, eventually.
The command line for gst-launch consists of a list of options followed
The command line for gst-launch-1.0 consists of a list of options followed
by a PIPELINE-DESCRIPTION. Some simplified instructions are given next,
se the complete documentation at [the reference page](gst-launch.html)
for `gst-launch`.
se the complete documentation at [the reference page](gst-launch-1.0.html)
for `gst-launch-1.0`.
#### Elements
@ -73,7 +73,7 @@ separated by exclamation marks (\!). Go ahead and type in the following
command:
```
gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc ! ffmpegcolorspace ! autovideosink
```
You should see a windows with an animated video pattern. Use CTRL+C on
@ -92,12 +92,12 @@ to find two compatible Pads.
Properties may be appended to elements, in the form
*property=value *(multiple properties can be specified, separated by
spaces). Use the `gst-inspect` tool (explained next) to find out the
spaces). Use the `gst-inspect-1.0` tool (explained next) to find out the
available properties for an
element.
```
gst-launch-0.10 videotestsrc pattern=11 ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc pattern=11 ! ffmpegcolorspace ! autovideosink
```
You should see a static video pattern, made of circles.
@ -114,7 +114,7 @@ Named elements are referred to using their name followed by a
dot.
```
gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! tee name=t ! queue ! autovideosink t. ! queue ! autovideosink
gst-launch-1.0 videotestsrc ! ffmpegcolorspace ! tee name=t ! queue ! autovideosink t. ! queue ! autovideosink
```
You should see two video windows, showing the same sample video pattern.
@ -141,14 +141,14 @@ Instead of letting GStreamer choose which Pad to use when linking two
elements, you may want to specify the Pads directly. You can do this by
adding a dot plus the Pad name after the name of the element (it must be
a named element). Learn the names of the Pads of an element by using
the `gst-inspect` tool.
the `gst-inspect-1.0` tool.
This is useful, for example, when you want to retrieve one particular
stream out of a
demuxer:
```
gst-launch-0.10.exe souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d d.video_00 ! matroskamux ! filesink location=sintel_video.mkv
gst-launch-1.0.exe souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d d.video_00 ! matroskamux ! filesink location=sintel_video.mkv
```
This fetches a media file from the internet using `souphttpsrc`, which
@ -168,7 +168,7 @@ new matroska file with the video. If we wanted to keep only the
audio:
```
gst-launch-0.10.exe souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d d.audio_00 ! vorbisparse ! matroskamux ! filesink location=sintel_audio.mka
gst-launch-1.0.exe souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d d.audio_00 ! vorbisparse ! matroskamux ! filesink location=sintel_audio.mka
```
The `vorbisparse` element is required to extract some information from
@ -194,7 +194,7 @@ Consider the following
pipeline:
```
gst-launch-0.10 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux ! filesink location=test
gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux ! filesink location=test
```
This is the same media file and demuxer as in the previous example. The
@ -208,7 +208,7 @@ previous sub-section, or by using **Caps
Filters**:
```
gst-launch-0.10 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux ! video/x-vp8 ! matroskamux ! filesink location=sintel_video.mkv
gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux ! video/x-vp8 ! matroskamux ! filesink location=sintel_video.mkv
```
A Caps Filter behaves like a pass-through element which does nothing and
@ -218,9 +218,9 @@ added a `video/x-vp8` Caps Filter to specify that we are interested in
the output pad of `matroskademux` which can produce this kind of video.
To find out the Caps an element accepts and produces, use the
`gst-inspect` tool. To find out the Caps contained in a particular file,
use the `gst-discoverer` tool. To find out the Caps an element is
producing for a particular pipeline, run `gst-launch` as usual, with the
`gst-inspect-1.0` tool. To find out the Caps contained in a particular file,
use the `gst-discoverer-1.0` tool. To find out the Caps an element is
producing for a particular pipeline, run `gst-launch-1.0` as usual, with the
`v` option to print Caps information.
#### Examples
@ -229,7 +229,7 @@ Play a media file using `playbin2` (as in [Basic tutorial 1: Hello
world\!](Basic%2Btutorial%2B1%253A%2BHello%2Bworld%2521.html)):
```
gst-launch-0.10 playbin2 uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm
gst-launch-1.0 playbin2 uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm
```
A fully operation playback pipeline, with audio and video (more or less
@ -237,7 +237,7 @@ the same pipeline that `playbin2` will create
internally):
```
gst-launch-0.10 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d ! queue ! vp8dec ! ffmpegcolorspace ! autovideosink d. ! queue ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! matroskademux name=d ! queue ! vp8dec ! ffmpegcolorspace ! autovideosink d. ! queue ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
```
A transcoding pipeline, which opens the webm container and decodes both
@ -247,7 +247,7 @@ with a different codec, and puts them back together in an Ogg container
it).
```
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm name=d ! queue ! theoraenc ! oggmux name=m ! filesink location=sintel.ogg d. ! queue ! audioconvert ! audioresample ! flacenc ! m.
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm name=d ! queue ! theoraenc ! oggmux name=m ! filesink location=sintel.ogg d. ! queue ! audioconvert ! audioresample ! flacenc ! m.
```
A rescaling pipeline. The `videoscale` element performs a rescaling
@ -256,14 +256,14 @@ output caps. The output caps are set by the Caps Filter to
320x200.
```
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! queue ! videoscale ! video/x-raw-yuv,width=320,height=200 ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! queue ! videoscale ! video/x-raw-yuv,width=320,height=200 ! ffmpegcolorspace ! autovideosink
```
This short description of `gst-launch` should be enough to get you
This short description of `gst-launch-1.0` should be enough to get you
started. Remember that you have the [complete documentation available
here](gst-launch.html).
here](gst-launch-1.0.html).
# `gst-inspect`
# `gst-inspect-1.0`
This tool has three modes of operation:
@ -278,7 +278,7 @@ This tool has three modes of operation:
Let's see an example of the third mode:
```
gst-inspect-0.10 vp8dec
gst-inspect-1.0 vp8dec
 
Factory Details:
Long name: On2 VP8 Decoder
@ -380,9 +380,9 @@ The most relevant sections are:
element, along with their type and accepted values.
For more information, you can check the [documentation
page](http://gst-inspect) of `gst-inspect`.
page](http://gst-inspect-1.0) of `gst-inspect-1.0`.
# `gst-discoverer`
# `gst-discoverer-1.0`
This tool is a wrapper around the `GstDiscoverer` object shown in [Basic
tutorial 9: Media information
@ -392,14 +392,14 @@ regarding the media that GStreamer can extract. It is useful to find out
what container and codecs have been used to produce the media, and
therefore what elements you need to put in a pipeline to play it.
Use `gst-discoverer --help` to obtain the list of available options,
Use `gst-discoverer-1.0 --help` to obtain the list of available options,
which basically control the amount of verbosity of the output.
Let's see an
example:
```
gst-discoverer-0.10 http://docs.gstreamer.com/media/sintel_trailer-480p.webm -v
gst-discoverer-1.0 http://docs.gstreamer.com/media/sintel_trailer-480p.webm -v
Analyzing http://docs.gstreamer.com/media/sintel_trailer-480p.webm
Done discovering http://docs.gstreamer.com/media/sintel_trailer-480p.webm
@ -454,10 +454,10 @@ Properties:
This tutorial has shown:
- How to build and run GStreamer pipelines from the command line using
the `gst-launch` tool.
the `gst-launch-1.0` tool.
- How to find out what GStreamer elements you have available and their
capabilities, using the `gst-inspect` tool.
capabilities, using the `gst-inspect-1.0` tool.
- How to discover the internal structure of media files, using
`gst-discoverer`.
`gst-discoverer-1.0`.
It has been a pleasure having you here, and see you soon\!

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@ -82,7 +82,7 @@ The `'*'` wildcard is also available. For example
starting with the word `audio`. `GST_DEBUG=*:2` is equivalent to
`GST_DEBUG=2`.
Use `gst-launch-0.10 --gst-debug-help` to obtain the list of all
Use `gst-launch-1.0 --gst-debug-help` to obtain the list of all
registered categories. Bear in mind that each plugin registers its own
categories, so, when installing or removing plugins, this list can
change.
@ -189,7 +189,7 @@ learn a bit of GStreamer along the way).
To obtain `.dot` files, simply set
the `GST_DEBUG_DUMP_DOT_DIR` environment variable to point to the
folder where you want the files to be placed. `gst-launch` will create
folder where you want the files to be placed. `gst-launch-1.0` will create
a `.dot` file at each state change, so you can see the evolution of the
caps negotiation. Unset the variable to disable this facility. From
within your application, you can use the

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@ -8,7 +8,7 @@ build complex pipelines easily (like `playbin2`), to little helper
elements which are extremely useful when debugging.
For simplicity, the following examples are given using the
`gst-launch` tool (Learn about it in [Basic tutorial 10: GStreamer
`gst-launch-1.0` tool (Learn about it in [Basic tutorial 10: GStreamer
tools](Basic%2Btutorial%2B10%253A%2BGStreamer%2Btools.html)). Use the
`-v` command line parameter if you want to see the Pad Caps that are
being negotiated.
@ -36,11 +36,11 @@ source pads as streams are found in the
media.
``` lang=bash
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! ffmpegcolorspace ! autovideosink
```
``` lang=bash
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioconvert ! autoaudiosink
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioconvert ! autoaudiosink
```
### `decodebin2`
@ -54,7 +54,7 @@ offers as many source pads as streams are found in the
media.
``` lang=bash
gst-launch-0.10 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin2 ! autovideosink
gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin2 ! autovideosink
```
# File input/output
@ -68,7 +68,7 @@ of `filesrc` to
`TRUE`.
``` lang=c
gst-launch-0.10 filesrc location=f:\\media\\sintel\\sintel_trailer-480p.webm ! decodebin2 ! autovideosink
gst-launch-1.0 filesrc location=f:\\media\\sintel\\sintel_trailer-480p.webm ! decodebin2 ! autovideosink
```
### `filesink`
@ -78,7 +78,7 @@ This element writes to a file all the media it receives. Use the
name.
```
gst-launch-0.10 audiotestsrc ! vorbisenc ! oggmux ! filesink location=test.ogg
gst-launch-1.0 audiotestsrc ! vorbisenc ! oggmux ! filesink location=test.ogg
```
# Network
@ -90,7 +90,7 @@ the SOUP library. Set the URL to retrieve through the `location`
property.
``` lang=bash
gst-launch-0.10 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin2 ! autovideosink
gst-launch-1.0 souphttpsrc location=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! decodebin2 ! autovideosink
```
# Test media generation
@ -105,7 +105,7 @@ This element produces a video pattern (selectable among many different
options with the `pattern` property). Use it to test video pipelines.
``` lang=bash
gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc ! ffmpegcolorspace ! autovideosink
```
### `audiotestsrc`
@ -114,7 +114,7 @@ This element produces an audio wave (selectable among many different
options with the `wave` property). Use it to test video pipelines.
``` lang=bash
gst-launch-0.10 audiotestsrc ! audioconvert ! autoaudiosink
gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
```
# Video adapters
@ -136,7 +136,7 @@ that can vary depending on external factors, like decoding a
user-provided file.
``` lang=bash
gst-launch-0.10 videotestsrc ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc ! ffmpegcolorspace ! autovideosink
```
### `videorate`
@ -156,7 +156,7 @@ rate is unknown at design time, just in
case.
``` lang=c
gst-launch-0.10 videotestsrc ! video/x-raw-rgb,framerate=30/1 ! videorate ! video/x-raw-rgb,framerate=1/1 ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc ! video/x-raw-rgb,framerate=30/1 ! videorate ! video/x-raw-rgb,framerate=1/1 ! ffmpegcolorspace ! autovideosink
```
### `videoscale`
@ -177,7 +177,7 @@ video sinks are capable of performing scaling
operations.
``` lang=bash
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! videoscale ! video/x-raw-yuv,width=178,height=100 ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! videoscale ! video/x-raw-yuv,width=178,height=100 ! ffmpegcolorspace ! autovideosink
```
# Audio adapters
@ -194,7 +194,7 @@ negotiation problems with audio, and it is generally safe to use it
liberally, since this element does nothing if it is not needed.
``` lang=bash
gst-launch-0.10 audiotestsrc ! audioconvert ! autoaudiosink
gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
```
### `audioresample`
@ -207,7 +207,7 @@ do not fear to use it
generously.
``` lang=bash
gst-launch-0.10 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioresample ! audio/x-raw-float,rate=4000 ! audioconvert ! autoaudiosink
gst-launch-1.0 uridecodebin uri=http://docs.gstreamer.com/media/sintel_trailer-480p.webm ! audioresample ! audio/x-raw-float,rate=4000 ! audioconvert ! autoaudiosink
```
### `audiorate`
@ -294,7 +294,7 @@ branch would stall the other
branches.
```
gst-launch-0.10 audiotestsrc ! tee name=t ! queue ! audioconvert ! autoaudiosink t. ! queue ! wavescope ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 audiotestsrc ! tee name=t ! queue ! audioconvert ! autoaudiosink t. ! queue ! wavescope ! ffmpegcolorspace ! autovideosink
```
# Capabilities
@ -303,14 +303,14 @@ gst-launch-0.10 audiotestsrc ! tee name=t ! queue ! audioconvert ! autoaudiosink
[Basic tutorial 10: GStreamer
tools](Basic%2Btutorial%2B10%253A%2BGStreamer%2Btools.html) already
explained how to use Caps filters with `gst-launch`. When building a
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.
``` lang=bash
gst-launch-0.10 videotestsrc ! video/x-raw-gray ! ffmpegcolorspace ! autovideosink
gst-launch-1.0 videotestsrc ! video/x-raw-gray ! ffmpegcolorspace ! autovideosink
```
### `typefind`
@ -333,11 +333,11 @@ gathering](Basic%2Btutorial%2B9%253A%2BMedia%2Binformation%2Bgathering.html)).
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`, so use the `silent` property to remove any unwanted
of `gst-launch-1.0`, so use the `silent` property to remove any unwanted
noise.
```
gst-launch-0.10 audiotestsrc num-buffers=1000 ! fakesink sync=false
gst-launch-1.0 audiotestsrc num-buffers=1000 ! fakesink sync=false
```
### `identity`
@ -349,7 +349,7 @@ things this seemingly harmless element can
do.
```
gst-launch-0.10 audiotestsrc ! identity drop-probability=0.1 ! audioconvert ! autoaudiosink
gst-launch-1.0 audiotestsrc ! identity drop-probability=0.1 ! audioconvert ! autoaudiosink
```
# Conclusion

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@ -137,7 +137,7 @@ framework similar to GStreamer. They are different enough, though, so
that their pipelines cannot be interconnected. However, through this
element, GStreamer can benefit from the decoding elements present in
Direct Show. `dshowdecwrapper` wraps multiple Direct Show decoders so
they can be embedded in a GStreamer pipeline. Use the `gst-inspect` tool
they can be embedded in a GStreamer pipeline. Use the `gst-inspect-1.0` tool
(see [Basic tutorial 10: GStreamer
tools](Basic%2Btutorial%2B10%253A%2BGStreamer%2Btools.html)) to see the
available decoders.

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@ -224,7 +224,7 @@ which controls the type of test video the element outputs. Try different
values!
The names and possible values of all the properties an element exposes
can be found using the gst-inspect tool described in [Basic tutorial 10:
can be found using the gst-inspect-1.0 tool described in [Basic tutorial 10:
GStreamer tools](Basic+tutorial+10+GStreamer+tools.markdown).
### Error checking

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@ -336,7 +336,7 @@ case, we pass a pointer to the `CustomData` structure we built specially
for this purpose.
The signals that a `GstElement` generates can be found in its
documentation or using the `gst-inspect` tool as described in [Basic
documentation or using the `gst-inspect-1.0` tool as described in [Basic
tutorial 10: GStreamer
tools](Basic+tutorial+10+GStreamer+tools.markdown).

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@ -464,7 +464,7 @@ This tutorial has shown:
- That Pad Caps are important to know beforehand if two elements can
be linked together.
- That Pad Caps can be found using the `gst-inspect` tool described
- That Pad Caps can be found using the `gst-inspect-1.0` tool described
in [Basic tutorial 10: GStreamer
tools](Basic+tutorial+10+GStreamer+tools.markdown).

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@ -66,7 +66,7 @@ The following code tries to discover the URI provided through the
command line, and outputs the retrieved information (If no URI is
provided it uses a default one).
This is a simplified version of what the `gst-discoverer` tool does
This is a simplified version of what the `gst-discoverer-1.0` tool does
([Basic tutorial 10: GStreamer
tools](Basic%2Btutorial%2B10%253A%2BGStreamer%2Btools.html)), which is
an application that only displays data, but does not perform any

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@ -65,8 +65,8 @@ useful:
- /Library/Frameworks/GStreamer.framework/Headers: path with the
development headers
- /Library/Frameworks/GStreamer.framework/Commands: link to the
commands provided by the framework, such as gst-inspect-0.10 or
gst-launch-0.10
commands provided by the framework, such as gst-inspect-1.0 or
gst-launch-1.0
For more information on OS X Frameworks anatomy, you can consult the
following [link](https://developer.apple.com/library/mac/#documentation/MacOSX/Conceptual/BPFrameworks/Concepts/FrameworkAnatomy.html)

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@ -273,8 +273,8 @@ dependent dynamic libraries
using [otool](https://developer.apple.com/library/mac/#documentation/darwin/reference/manpages/man1/otool.1.html):
``` lang=bash
$ otool -L /Library/Frameworks/GStreamer.framework/Commands/gst-launch-0.10
/Library/Frameworks/GStreamer.framework/Commands/gst-launch-0.10:
$ otool -L /Library/Frameworks/GStreamer.framework/Commands/gst-launch-1.0
/Library/Frameworks/GStreamer.framework/Commands/gst-launch-1.0:
/System/Library/Frameworks/CoreFoundation.framework/Versions/A/CoreFoundation (compatibility version 150.0.0, current version 550.43.0)
/Library/Frameworks/GStreamer.framework/Versions/0.10/x86/lib/libgstreamer-0.10.0.dylib (compatibility version 31.0.0, current version 31.0.0)
/Library/Frameworks/GStreamer.framework/Versions/0.10/x86/lib/libxml2.2.dylib (compatibility version 10.0.0, current version 10.8.0)
@ -300,7 +300,7 @@ $ otool -D /Library/Frameworks/GStreamer.framework/Libraries/libgstreamer-0.10.d
Any object file that links to the dynamic library `gstreamer-0.10` will
use the
path `/Library/Frameworks/GStreamer.framework/Versions/0.10/x86/lib/libgstreamer-0.10.0.dylib` to
locate it, as we saw previously with `gst-launch-0.10`.
locate it, as we saw previously with `gst-launch-1.0`.
Since working exclusively with full paths wouldn't let us install our
binaries anywhere in the path, the linker provides a mechanism of string

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@ -11,9 +11,9 @@ concepts is discussed.
- Debugging: [Basic tutorial 11: Debugging tools]
- Discoverer: [Basic tutorial 9: Media information gathering]
- Elements: [Basic tutorial 2: GStreamer concepts]
- gst-discoverer: [Basic tutorial 10: GStreamer tools]
- gst-inspect: [Basic tutorial 10: GStreamer tools], [gst-inspect]
- gst-launch: [Basic tutorial 10: GStreamer tools], [gst-launch]
- gst-discoverer-1.0: [Basic tutorial 10: GStreamer tools]
- gst-inspect-1.0: [Basic tutorial 10: GStreamer tools], [gst-inspect-1.0]
- gst-launch-1.0: [Basic tutorial 10: GStreamer tools], [gst-launch-1.0]
- GUI: [Basic tutorial 5: GUI toolkit integration]
- Links: [Basic tutorial 2: GStreamer concepts]
- Pads: [Basic tutorial 3: Dynamic pipelines]
@ -36,8 +36,8 @@ concepts is discussed.
[Basic tutorial 11: Debugging tools]: Basic+tutorial+11+Debugging+tools.markdown
[Basic tutorial 9: Media information gathering]: Basic+tutorial+9+Media+information+gathering.markdown
[Basic tutorial 10: GStreamer tools]: Basic+tutorial+10+GStreamer+tools.markdown
[gst-inspect]: gst-inspect.markdown
[gst-launch]: gst-launch.markdown
[gst-inspect-1.0]: gst-inspect.markdown
[gst-launch-1.0]: gst-launch.markdown
[Basic tutorial 5: GUI toolkit integration]: Basic+tutorial+5+GUI+toolkit+integration.markdown
[Basic tutorial 3: Dynamic pipelines]: Basic+tutorial+3+Dynamic+pipelines.markdown
[Basic tutorial 7: Multithreading and Pad Availability]: Basic+tutorial+7+Multithreading+and+Pad+Availability.markdown

View file

@ -1,35 +1,35 @@
# gst-inspect
# 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</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>
<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>
## Name
gst-inspect - print info about a GStreamer plugin or element
gst-inspect-1.0 - print info about a GStreamer plugin or element
## Synopsis
**gst-inspect \[OPTION...\] \[PLUGIN|ELEMENT\]**
**gst-inspect-1.0 \[OPTION...\] \[PLUGIN|ELEMENT\]**
## Description
*gst-inspect* is a tool that prints out information on
*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* will print a list of all plugins and
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* will print information about that plug-in or
argument, *gst-inspect-1.0* will print information about that plug-in or
element.
## Options
*gst-inspect* accepts the following arguments and options:
*gst-inspect-1.0* accepts the following arguments and options:
**PLUGIN**
@ -80,7 +80,7 @@ Add directories separated with ':' to the plugin search path
## Example
```
gst-inspect-0.10 audiotestsrc
gst-inspect-1.0 audiotestsrc
```
should produce:

View file

@ -1,25 +1,25 @@
# gst-launch
# 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</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>
<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>
## Name
gst-launch - build and run a GStreamer pipeline
gst-launch-1.0 - build and run a GStreamer pipeline
## Synopsis
**gst-launch** *\[OPTION...\]* PIPELINE-DESCRIPTION
**gst-launch-1.0** *\[OPTION...\]* PIPELINE-DESCRIPTION
## Description
*gst-launch* 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
@ -29,14 +29,14 @@ For a complete description of possible PIPELINE-DESCRIPTIONS see the
section*pipeline description* below or consult the GStreamer
documentation.
Please note that *gst-launch* is primarily a debugging tool for
Please note that *gst-launch-1.0* is primarily a debugging tool for
developers and users. You should not build applications on top of it.
For applications, use the gst\_parse\_launch() function of the GStreamer
API as an easy way to construct pipelines from pipeline descriptions.
## Options
*gst-launch* accepts the following options:
*gst-launch-1.0* accepts the following options:
**--help**
@ -75,7 +75,7 @@ time to work.
## Gstreamer Options
*gst-launch* also accepts the following options that are common to all
*gst-launch-1.0* also accepts the following options that are common to all
GStreamer applications:
## Pipeline Description
@ -94,7 +94,7 @@ 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)
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.
@ -183,85 +183,85 @@ ffmpegcolorspace (for video) in front of the sink to make things work.
**Audio playback**
**gst-launch filesrc location=music.mp3 \! mad \! audioconvert \!
**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 filesrc location=music.ogg \! oggdemux \! vorbisdec \!
**gst-launch-1.0 filesrc location=music.ogg \! oggdemux \! vorbisdec \!
audioconvert \! audioresample \! osssink**
Play an Ogg Vorbis format file
**gst-launch gnomevfssrc location=music.mp3 \! mad \! osssink
gst-launch gnomevfssrc location=<http://domain.com/music.mp3> \! mad \!
**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 gnomevfssrc location=<smb://computer/music.mp3> \! mad \!
**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 filesrc location=music.mp3 \! mad \! audioconvert \!
**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 filesrc location=music.mp3 \! mad \! audioconvert \!
**gst-launch-1.0 filesrc location=music.mp3 \! mad \! audioconvert \!
flacenc \! filesink location=test.flac**
Convert to the FLAC format
**Other**
**gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
**gst-launch-1.0 filesrc location=music.wav \! wavparse \! audioconvert \!
audioresample \! osssink**
Plays a .WAV file that contains raw audio data (PCM).
**gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
**gst-launch-1.0 filesrc location=music.wav \! wavparse \! audioconvert \!
vorbisenc \! oggmux \! filesink location=music.ogg
gst-launch filesrc location=music.wav \! wavparse \! audioconvert \!
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 cdparanoiasrc mode=continuous \! audioconvert \! lame \!
**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 cdparanoiasrc track=5 \! audioconvert \! lame \! id3v2mux
**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), it is possible to discover settings like the
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-0.10 will
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 [cdda://5]() \! lame vbr=new vbr-quality=6 \!
you, e.g.: **gst-launch-1.0 [cdda://5]() \! lame vbr=new vbr-quality=6 \!
filesink location=track5.mp3**
**gst-launch osssrc \! audioconvert \! vorbisenc \! oggmux \! filesink
**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 filesrc location=JB\_FF9\_TheGravityOfLove.mpg \! dvddemux
**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 filesrc location=/flflfj.vob \! dvddemux \! mpeg2dec \!
**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 filesrc location=movie.mpg \! dvddemux name=demuxer
**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 filesrc location=movie.mpg \! mpegdemux name=demuxer
**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
@ -269,42 +269,42 @@ 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 textoverlay name=overlay \! ffmpegcolorspace \! videoscale
**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 playbin2
**gst-launch playbin2 uri=<file:///path/to/movie.avi>
**gst-launch-1.0 playbin2 uri=<file:///path/to/movie.avi>
suburi=<file:///path/to/movie.srt>**
**Network streaming**
Stream video using RTP and network elements.
**gst-launch v4l2src \!
**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 udpsrc port=5000 \! application/x-rtp,
**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 -v fakesrc num-buffers=16 \! fakesink**
**gst-launch-1.0 -v fakesrc num-buffers=16 \! fakesink**
Generate a null stream and ignore it (and print out details).
**gst-launch audiotestsrc \! audioconvert \! audioresample \!
**gst-launch-1.0 audiotestsrc \! audioconvert \! audioresample \!
osssink**
Generate a pure sine tone to test the audio output
**gst-launch videotestsrc \! xvimagesink
gst-launch videotestsrc \! ximagesink**
**gst-launch-1.0 videotestsrc \! xvimagesink
gst-launch-1.0 videotestsrc \! ximagesink**
Generate a familiar test pattern to test the video output
**Automatic linking**
@ -312,29 +312,29 @@ Generate a familiar test pattern to test the video output
You can use the decodebin element to automatically select the right
elements to get a working pipeline.
**gst-launch filesrc location=musicfile \! decodebin \! audioconvert \!
**gst-launch-1.0 filesrc location=musicfile \! decodebin \! audioconvert \!
audioresample \! osssink**
Play any supported audio format
**gst-launch filesrc location=videofile \! decodebin name=decoder
**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 playbin uri=<file:///home/joe/foo.avi>**
**gst-launch-1.0 playbin uri=<file:///home/joe/foo.avi>**
**Filtered connections**
These examples show you how to use filtered caps.
**gst-launch 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.
**gst-launch 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