Getting lots of noise at the output of this Gstreamer pipeline
Jim Ruxton
jim.ruxton at gmail.com
Mon Feb 10 09:00:11 UTC 2020
I wonder if someone could point me to what I may be doing wrong. The
pipeline itself works except I am getting a lot of noise. The pipeline
takes my mic input and after converting it to unsigned 16 bit audio
resamples it to 8K. I am getting the correct number of samples sent. I
can hear my voice over a pile of noise. If I turn the input level of my
mic all the way down I still get the same noise. I can send other udp
streams to the udpsink address with other programs and it works without
the noise. Could it be because my audio card doesn't natively support 8K
sampling? I thought the pipeline would show the native sampling rate of
44.1K for the audiosrc and then later in the pipeline the audio would
show up at 8K but this line implies that it starts out with 8K. Sorry I
am new to Gstreamer.
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0:
caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)8000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
Below is the the pipeline and full output. Thanks for any help. to sort
this out.
Jim
gst-launch-1.0 -v autoaudiosrc ! audioconvert ! audioresample !
audio/x-raw, rate=8000 ,format=U16LE ! rndbuffersize max=1024 min=1024 !
udpsink host=192.168.1.255 port=3333
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstPulseSrcClock
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
source-output-index = 31
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
actual-buffer-time = 200000
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
actual-latency-time = 10000
Redistribute latency...
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls.GstPad:src:
caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)8000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src:
caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)8000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =
audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps
= audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstRndBufferSize:rndbuffersize0.GstPad:src: caps
= audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =
audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstRndBufferSize:rndbuffersize0.GstPad:sink: caps
= audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps
= audio/x-raw, rate=(int)8000, format=(string)U16LE, channels=(int)2,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =
audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)8000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0.GstGhostPad:src.GstProxyPad:proxypad0:
caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)8000, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
volume = 1
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
mute = false
/GstPipeline:pipeline0/GstAutoAudioSrc:autoaudiosrc0/GstPulseSrc:autoaudiosrc0-actual-src-puls:
current-device = alsa_input.pci-0000_00_1f.3.analog-stereo
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