Pipelining Audio input to CMUSphinx using Gstreamer

The wikipage for CMUSphinx gives a python example (http://cmusphinx.sourceforge.net/wiki/gstreamer) - this is the pipeline setup with gst_parse_launch:
self.pipeline = gst.parse_launch('gconfaudiosrc ! audioconvert ! audioresample '
+ '! vader name=vad auto_threshold=true '
+ '! pocketsphinx name=asr ! fakesink')
The Gstreamer manual gives:
Create a new pipeline based on command line syntax.
I can't seem to get a hold on the pipeline though - here is the command line attempt:
gst-launch-0.10 gconfaudiosrc ! audioconvert ! audioresample '
+ '! vader name=vad auto_threshold=true '
+ '! pocketsphinx name=asr ! fakesink
INFO: cmd_ln.c(691): Parsing command line:
gst-pocketsphinx \
-samprate 8000 \
-cmn prior \
-fwdflat no \
-bestpath no \
-maxhmmpf 2000 \
-maxwpf 20
Current configuration:
[NAME] [DEFLT] [VALUE]
-agc none none
-agcthresh 2.0 2.000000e+00
-alpha 0.97 9.700000e-01
-ascale 20.0 2.000000e+01
-aw 1 1
-backtrace no no
-beam 1e-48 1.000000e-48
-bestpath yes no
-bestpathlw 9.5 9.500000e+00
-bghist no no
-ceplen 13 13
-cmn current prior
-cmninit 8.0 8.0
-compallsen no no
-debug 0
-dict
-dictcase no no
-dither no no
-doublebw no no
-ds 1 1
-fdict
-feat 1s_c_d_dd 1s_c_d_dd
-featparams
-fillprob 1e-8 1.000000e-08
-frate 100 100
-fsg
-fsgusealtpron yes yes
-fsgusefiller yes yes
-fwdflat yes no
-fwdflatbeam 1e-64 1.000000e-64
-fwdflatefwid 4 4
-fwdflatlw 8.5 8.500000e+00
-fwdflatsfwin 25 25
-fwdflatwbeam 7e-29 7.000000e-29
-fwdtree yes yes
-hmm
-input_endian little little
-jsgf
-kdmaxbbi -1 -1
-kdmaxdepth 0 0
-kdtree
-latsize 5000 5000
-lda
-ldadim 0 0
-lextreedump 0 0
-lifter 0 0
-lm
-lmctl
-lmname default default
-logbase 1.0001 1.000100e+00
-logfn
-logspec no no
-lowerf 133.33334 1.333333e+02
-lpbeam 1e-40 1.000000e-40
-lponlybeam 7e-29 7.000000e-29
-lw 6.5 6.500000e+00
-maxhmmpf -1 2000
-maxnewoov 20 20
-maxwpf -1 20
-mdef
-mean
-mfclogdir
-min_endfr 0 0
-mixw
-mixwfloor 0.0000001 1.000000e-07
-mllr
-mmap yes yes
-ncep 13 13
-nfft 512 512
-nfilt 40 40
-nwpen 1.0 1.000000e+00
-pbeam 1e-48 1.000000e-48
-pip 1.0 1.000000e+00
-pl_beam 1e-10 1.000000e-10
-pl_pbeam 1e-5 1.000000e-05
-pl_window 0 0
-rawlogdir
-remove_dc no no
-round_filters yes yes
-samprate 16000 8.000000e+03
-seed -1 -1
-sendump
-senlogdir
-senmgau
-silprob 0.005 5.000000e-03
-smoothspec no no
-svspec
-tmat
-tmatfloor 0.0001 1.000000e-04
-topn 4 4
-topn_beam 0 0
-toprule
-transform legacy legacy
-unit_area yes yes
-upperf 6855.4976 6.855498e+03
-usewdphones no no
-uw 1.0 1.000000e+00
-var
-varfloor 0.0001 1.000000e-04
-varnorm no no
-verbose no no
-warp_params
-warp_type inverse_linear inverse_linear
-wbeam 7e-29 7.000000e-29
-wip 0.65 6.500000e-01
-wlen 0.025625 2.562500e-02
Setting pipeline to PAUSED ...
** (gst-launch-0.10:790): WARNING **: No GConf default audio src key and alsasrc doesn't work
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0: Could not configure supporting library.
Additional debug info:
gstgconfaudiosrc.c(152): do_toggle_element (): /GstPipeline:pipeline0/GstGConfAudioSrc:gconfaudiosrc0:
Failed to render audio src from GConf
Setting pipeline to NULL ...
Freeing pipeline ...
I really think I'm missing something very obvious about how these libraries and devices are supposed to interact. Hopefully someone can point me in the right direction.

So there is a gstreamer plugin autodetect, which allows it to pipeline the default recording using
gst-launch-0.10 autoaudiosrc
. I'd like to mark this as solved but for future reference I actually still don't know where one normally gets the `audio source` name which gst_parse_launch expects (I'm looking at pactl list sources)

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