Difference between revisions of "Gstreamer"

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{{Under_Construction}}
 
{{Under_Construction}}
{{Note|gstreamer has only been introduced recently in Buildroot, so you will need to use our "still in development" migration to Buildroot 2010.xx version.}}
 
  
 
==Installation==
 
==Installation==

Revision as of 18:09, 10 May 2011

Page under construction... Construction.png Informations on this page are not guaranteed !!

Installation

$ make menuconfig
Package Selection for the target  --->
    Audio and video libraries and applications  --->
        ...
        [*] gstreamer
        [*]   require libxml2 for registry and load/save
        -*-   gst-plugins-base  --->
        [*]   gst-plugins-good  --->
        [*]   gst-plugins-bad  --->
        [*]   gst-plugins-ugly  --->

Plugins worth to activate:

  • base:
  • good:
  • bad:
  • ugly:

Usage

Check installed plugins

  • To see all installed plugins:
# gst-inspect
videoscale:  videoscale: Video scaler
queue2:  queue2: Queue               
ffmpegcolorspace:  ffmpegcolorspace: FFMPEG Colorspace converter
audiorate:  audiorate: Audio rate adjuster                      
audioconvert:  audioconvert: Audio converter                    
audioresample:  audioresample: Audio resampler                  
volume:  volume: Volume
...
  • To have more details about one particular plugin (here alsasrc):
# gst-inspect alsasrc
Factory Details:     
  Long name:    Audio source (ALSA)
  Class:        Source/Audio       
  Description:  Read from a sound card via ALSA
  Author(s):    Wim Taymans <wim@fluendo.com>  
  Rank:         primary (256)                  

Plugin Details:
  Name:                 alsa
  Description:          ALSA plugin library
  Filename:             /usr/lib/gstreamer-0.10/libgstalsa.so
  Version:              0.10.25                              
  License:              LGPL                                 
  Source module:        gst-plugins-base                     
  Binary package:       GStreamer Base Plug-ins source release
  Origin URL:           Unknown package origin                
...
<cut>

Audio

  • play a monotic tone to check your sound configuration is working (if not please configure ALSA):
# gst-launch audiotestsrc ! audioconvert ! audioresample ! alsasink
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock
  • play a music file (here a .ogg, supposing the libgstogg plugin was installed):
# gst-launch filesrc location=toto.ogg ! decodebin ! audioconvert ! audioresample ! alsasink
  • record a sound from a mic (APF27 only) and store it as an uncompressed wav file:
# gst-launch alsasrc ! audioconvert ! audioresample ! wavenc ! filesink location=mic.wav
  • record a sound from a mic (APF27 only) and compress it (ogg/worbis):
# gst-launch alsasrc ! audioconvert ! audioresample ! vorbisenc ! oggmux ! filesink location=mic.ogg
  • record a sound from a mic (APF27 only) @ 8KHz (16KHz is too CPU consuming), compress it with Speex and send it to your Host (here 192.168.1.2) at port 6666:
# gst-launch -v alsasrc ! audioconvert ! audioresample ! 'audio/x-raw-int,rate=8000,width=16,channels=1' ! speexenc ! rtpspeexpay ! udpsink host=192.168.1.2 port=6666
$ gst-launch udpsrc port=6666 caps="application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" ! gstrtpjitterbuffer ! rtpspeexdepay ! speexdec ! audioconvert ! audioresample ! autoaudiosink
$ gst-launch udpsrc port=6666 caps="application/x-rtp, media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" ! gstrtpjitterbuffer ! rtpspeexdepay ! speexdec ! audioconvert ! audioresample ! wavenc ! filesink location=toto.wav

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