mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst-inspect/launch: Fixes and updates
This commit is contained in:
Thibault Saunier
parent
e220d85b46
commit
33163869cb
4 changed files with 266 additions and 322 deletions
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@ -253,7 +253,6 @@ and used as follows:
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include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk
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GSTREAMER_PLUGINS := $(GSTREAMER_PLUGINS_CORE) $(GSTREAMER_PLUGINS_CODECS) playbin souphttpsrc
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#### List of categories and included plugins
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| Category | Included plugins |
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@ -48,8 +48,6 @@ Pages to review:
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- Multiplatform+deployment+using+Cerbero.markdown
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- Legal+information.markdown
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- GStreamer+reference.markdown
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- gst-inspect.markdown
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- gst-launch.markdown
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Screenshots:
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@ -74,6 +72,8 @@ Reviewed pages:
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- GStreamer+reference.markdown
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- Playback+tutorial+1+Playbin+usage.markdown
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- Basic+tutorial+1+Hello+world.markdown
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- gst-inspect.markdown
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- gst-launch.markdown
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For-later pages:
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- Qt+tutorials.markdown [tpm: this should all be rewritten from scratch with qmlglsink; QtGStreamer is outdated and unmaintained, we should not promote it]
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@ -1,13 +1,9 @@
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# gst-inspect-1.0
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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><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>
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</tr>
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</tbody>
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</table>
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> ![information] This is the Linux man page for
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> the `gst-inspect-1.0` tool. As such, it is very Linux-centric
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> regarding path specification and plugin names. Please be patient while
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> it is rewritten to be more generic.
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## Name
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@ -20,9 +16,9 @@ gst-inspect-1.0 - print info about a GStreamer plugin or element
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## Description
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*gst-inspect-1.0* is a tool that prints out information on
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available *GStreamer* plugins, information about a particular plugin,
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or information about a particular element. When executed with no PLUGIN
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or ELEMENT argument, *gst-inspect-1.0* will print a list of all plugins and
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available *GStreamer* plugins, information about a particular plugin, or
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information about a particular element. When executed with no PLUGIN or
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ELEMENT argument, *gst-inspect-1.0* will print a list of all plugins and
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elements together with a sumary. When executed with a PLUGIN or ELEMENT
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argument, *gst-inspect-1.0* will print information about that plug-in or
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element.
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@ -79,130 +75,113 @@ Add directories separated with ':' to the plugin search path
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## Example
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```
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gst-inspect-1.0 audiotestsrc
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```
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gst-inspect-1.0 audiotestsrc
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should produce:
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```
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Factory Details:
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Long name: Audio test source
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Class: Source/Audio
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Description: Creates audio test signals of given frequency and volume
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Author(s): Stefan Kost <ensonic@users.sf.net>
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Rank: none (0)
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Plugin Details:
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Name: audiotestsrc
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Description: Creates audio test signals of given frequency and volume
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Filename: I:\gstreamer-sdk\2012.5\x86\lib\gstreamer-1.0\libgstaudiotestsrc.dll
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Version: 0.10.36
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License: LGPL
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Source module: gst-plugins-base
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Source release date: 2012-02-20
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Binary package: GStreamer Base Plug-ins source release
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Origin URL: Unknown package origin
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GObject
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+----GstObject
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+----GstElement
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+----GstBaseSrc
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+----GstAudioTestSrc
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Pad Templates:
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SRC template: 'src'
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Availability: Always
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Capabilities:
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audio/x-raw-int
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endianness: 1234
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signed: true
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width: 16
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depth: 16
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rate: [ 1, 2147483647 ]
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channels: [ 1, 2 ]
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audio/x-raw-int
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endianness: 1234
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signed: true
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width: 32
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depth: 32
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rate: [ 1, 2147483647 ]
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channels: [ 1, 2 ]
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audio/x-raw-float
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endianness: 1234
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width: { 32, 64 }
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rate: [ 1, 2147483647 ]
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channels: [ 1, 2 ]
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Factory Details:
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Rank none (0)
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Long-name Audio test source
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Klass Source/Audio
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Description Creates audio test signals of given frequency and volume
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Author Stefan Kost <ensonic@users.sf.net>
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Element Flags:
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no flags set
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Element Implementation:
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Has change_state() function: gst_base_src_change_state
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Has custom save_thyself() function: gst_element_save_thyself
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Has custom restore_thyself() function: gst_element_restore_thyself
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Element has no clocking capabilities.
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Element has no indexing capabilities.
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Element has no URI handling capabilities.
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Pads:
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SRC: 'src'
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Implementation:
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Has getrangefunc(): gst_base_src_pad_get_range
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Has custom eventfunc(): gst_base_src_event_handler
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Has custom queryfunc(): gst_base_src_query
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Has custom iterintlinkfunc(): gst_pad_iterate_internal_links_default
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Has getcapsfunc(): gst_base_src_getcaps
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Has setcapsfunc(): gst_base_src_setcaps
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Has acceptcapsfunc(): gst_pad_acceptcaps_default
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Has fixatecapsfunc(): 62B82E10
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Pad Template: 'src'
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Element Properties:
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name : The name of the object
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flags: readable, writable
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String. Default: "audiotestsrc0"
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blocksize : Size in bytes to read per buffer (-1 = default)
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flags: readable, writable
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Unsigned Long. Range: 0 - 4294967295 Default: 4294967295
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num-buffers : Number of buffers to output before sending EOS (-1 = unlimited)
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flags: readable, writable
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Integer. Range: -1 - 2147483647 Default: -1
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typefind : Run typefind before negotiating
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flags: readable, writable
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Boolean. Default: false
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do-timestamp : Apply current stream time to buffers
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flags: readable, writable
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Boolean. Default: false
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samplesperbuffer : Number of samples in each outgoing buffer
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flags: readable, writable
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Integer. Range: 1 - 2147483647 Default: 1024
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wave : Oscillator waveform
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flags: readable, writable, controllable
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Enum "GstAudioTestSrcWave" Default: 0, "sine"
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(0): sine - Sine
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(1): square - Square
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(2): saw - Saw
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(3): triangle - Triangle
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(4): silence - Silence
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(5): white-noise - White uniform noise
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(6): pink-noise - Pink noise
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(7): sine-table - Sine table
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(8): ticks - Periodic Ticks
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(9): gaussian-noise - White Gaussian noise
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(10): red-noise - Red (brownian) noise
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(11): blue-noise - Blue noise
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(12): violet-noise - Violet noise
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freq : Frequency of test signal
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flags: readable, writable, controllable
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Double. Range: 0 - 20000 Default: 440
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volume : Volume of test signal
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flags: readable, writable, controllable
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Double. Range: 0 - 1 Default: 0.8
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is-live : Whether to act as a live source
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flags: readable, writable
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Boolean. Default: false
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timestamp-offset : An offset added to timestamps set on buffers (in ns)
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flags: readable, writable
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Integer64. Range: -9223372036854775808 - 9223372036854775807 Default: 0
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can-activate-push : Can activate in push mode
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flags: readable, writable
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Boolean. Default: true
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can-activate-pull : Can activate in pull mode
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flags: readable, writable
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Boolean. Default: false
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```
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Plugin Details:
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Name audiotestsrc
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Description Creates audio test signals of given frequency and volume
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Filename /usr/lib/gstreamer-1.0/libgstaudiotestsrc.so
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Version 1.8.1
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License LGPL
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Source module gst-plugins-base
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Source release date 2016-04-20
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Binary package GStreamer Base Plugins (Arch Linux)
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Origin URL http://www.archlinux.org/
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GObject
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+----GInitiallyUnowned
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+----GstObject
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+----GstElement
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+----GstBaseSrc
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+----GstAudioTestSrc
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Pad Templates:
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SRC template: 'src'
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Availability: Always
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Capabilities:
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audio/x-raw
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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 }
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layout: interleaved
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rate: [ 1, 2147483647 ]
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channels: [ 1, 2147483647 ]
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Element Flags:
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no flags set
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Element Implementation:
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Has change_state() function: gst_base_src_change_state
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Element has no clocking capabilities.
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Element has no URI handling capabilities.
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Pads:
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SRC: 'src'
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Pad Template: 'src'
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Element Properties:
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name : The name of the object
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flags: readable, writable
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String. Default: "audiotestsrc0"
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parent : The parent of the object
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flags: readable, writable
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Object of type "GstObject"
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blocksize : Size in bytes to read per buffer (-1 = default)
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flags: readable, writable
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Unsigned Integer. Range: 0 - 4294967295 Default: 4294967295
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num-buffers : Number of buffers to output before sending EOS (-1 = unlimited)
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flags: readable, writable
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Integer. Range: -1 - 2147483647 Default: -1
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typefind : Run typefind before negotiating
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flags: readable, writable
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Boolean. Default: false
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do-timestamp : Apply current stream time to buffers
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flags: readable, writable
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Boolean. Default: false
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samplesperbuffer : Number of samples in each outgoing buffer
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flags: readable, writable
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Integer. Range: 1 - 2147483647 Default: 1024
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wave : Oscillator waveform
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flags: readable, writable, controllable
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Enum "GstAudioTestSrcWave" Default: 0, "sine"
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(0): sine - Sine
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(1): square - Square
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(2): saw - Saw
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(3): triangle - Triangle
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(4): silence - Silence
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(5): white-noise - White uniform noise
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(6): pink-noise - Pink noise
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(7): sine-table - Sine table
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(8): ticks - Periodic Ticks
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(9): gaussian-noise - White Gaussian noise
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(10): red-noise - Red (brownian) noise
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(11): blue-noise - Blue noise
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(12): violet-noise - Violet noise
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freq : Frequency of test signal. The sample rate needs to be at least 4 times higher.
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flags: readable, writable, controllable
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Double. Range: 0 - 5.368709e+08 Default: 440
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volume : Volume of test signal
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flags: readable, writable, controllable
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Double. Range: 0 - 1 Default: 0.8
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is-live : Whether to act as a live source
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flags: readable, writable
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Boolean. Default: false
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timestamp-offset : An offset added to timestamps set on buffers (in ns)
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flags: readable, writable
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Integer64. Range: -9223372036854775808 - 9223372036854775807 Default: 0
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can-activate-push : Can activate in push mode
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flags: readable, writable
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Boolean. Default: true
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can-activate-pull : Can activate in pull mode
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flags: readable, writable
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Boolean. Default: false
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[information]: images/icons/emoticons/information.png
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@ -1,13 +1,9 @@
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# gst-launch-1.0
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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-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>
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</tr>
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</tbody>
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</table>
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> ![information] This is the Linux man page for
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> the `gst-inspect-1.0` tool. As such, it is very Linux-centric
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> regarding path specification and plugin names. Please be patient while
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> it is rewritten to be more generic.
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## Name
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@ -19,11 +15,12 @@ gst-launch-1.0 - build and run a GStreamer pipeline
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## Description
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*gst-launch-1.0* is a tool that builds and runs basic *GStreamer* pipelines.
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*gst-launch-1.0* is a tool that builds and runs
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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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by exclamation marks (!). Properties may be appended to elements, in the
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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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@ -75,8 +72,8 @@ time to work.
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## Gstreamer Options
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*gst-launch-1.0* also accepts the following options that are common to all
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GStreamer applications:
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*gst-launch-1.0* also accepts the following options that are common to
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all GStreamer applications:
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## Pipeline Description
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@ -94,9 +91,10 @@ Creates an element of type ELEMENTTYPE and sets the PROPERTIES.
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PROPERTY=VALUE ...
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Sets the property to the specified value. You can use **gst-inspect-1.0**(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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Sets the property to the specified value. You can
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use **gst-inspect-1.0**(1) to find out about properties and allowed
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values of different elements. Enumeration properties can be set by name,
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nick or value.
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**Bins**
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@ -112,9 +110,9 @@ 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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*\[\[SRCELEMENT\].\[PAD1,...\]\]* ! *\[\[SINKELEMENT\].\[PAD1,...\]\]
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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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@ -124,9 +122,8 @@ 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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in the given order. So the simplest link is a simple exclamation mark,
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that links the element to the left of it to the element right of it.
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**Caps**
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@ -138,35 +135,30 @@ 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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NAME=*\[(TYPE)\]*VALUE 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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- **i** or **int** for integer values or ranges - **f** or **float** for
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float values or ranges - **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 - **fraction** for fractions
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(framerate, pixel-aspect-ratio) - **l** or **list** for lists If no type
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was given, the following order is tried: integer, float, boolean,
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string. Integer values must be parsable by **strtol()**, floats
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by **strtod()**. FOURCC values may either be integers or strings.
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Boolean values are (case insensitive) *yes*, *no*, *true* or *false* and
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may like strings be escaped with " or '. Ranges are in this format: \[
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VALUE, VALUE \] 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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playing state. There are currently no signals defined to go into the
|
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ready or pause (GST\_STATE\_READY and GST\_STATE\_PAUSED) state
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explicitely.
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## Pipeline Examples
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|
|
@ -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
|
||||
|
|
|
|||
Loading…
Reference in a new issue