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dm355 problem of cpu overload

seif ferjani
Prodigy 205 points

 

hi,

I am trying to stream audio over network

 

Test 1:

 

I tried this 2 pipe (alaw+rtp+udp)

 

##gst-launch -v alsasrc ! audioconvert ! volume volume=9 ! alawenc ! rtppcmapay ! udpsink host=192.168.14.45 port=5000 sync=false

 

##gst-launch -v udpsrc port=5000 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)772587109, clock-base=(guint)3652151312, seqnum-base=(guint)60451" ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! volume volume=3 !  alsasink

 

 

Test 2:

 

raw format+udp

 

##gst-launch -v alsasrc ! audioconvert ! volume volume=9  ! udpsink host=192.168.14.45 port=5000 sync=false

 

##gst-launch -v udpsrc port=5000 caps=" audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2"   ! audioconvert ! audioresample ! volume volume=3 ! alsasink

 

In the 2 cases the cpu is always overloaded (between 95% and 99%)

 

i didn't found a solution for this issue

 

Any help will be welcome

 

 

Regards,

 

  • 0
    Prodigy 90 points

    I have the same issue with DM368

    I am using the following pipe:

    gst-launch -v &> output5.log gstrtpbin name=rtpbin \
        alsasrc ! queue ! audioconvert ! TIAudenc1 codecName=aaclcenc engineName=codecServer ! dmaiperf print-arm-load=true ! rtpmp4apay ! rtpbin.send_rtp_sink_1                                             \
            rtpbin.send_rtp_src_1 ! udpsink port=5002 host=$DEST ts-offset=$AOFFSET name=artpsink                                                 \
            rtpbin.send_rtcp_src_1 ! udpsink port=5003 host=$DEST sync=false async=false name=artcpsink                                               \
          udpsrc port=5007 name=artpsrc ! rtpbin.recv_rtcp_sink_1

    Was any solution found regarding this issue?

  • 0 in reply to Guillaume Frigon
    Prodigy 90 points

    Try specifying the sample rate you want in your alsasrc caps. It solved the issue for me. e.g.

    gst-launch -v alsasrc ! audio/x-raw-int, endianness=1234, signed=true, width=16, depth=16, rate=44100, channels=2 ! volume volume=9 ! alawenc ! rtppcmapay ! udpsink host=192.168.14.45 port=5000 sync=false

    Hope it helps, even if the post is 2 years old