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<p class="MsoNormal">Hi,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> I need to increase performance on an application that uses gstreamer. The original pipeline used was:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">“udpsrc name=udpsrc retrieve-sender-address=false reuse=false port=0 caps="application/x-rtp, media=(string)audio, payload=0, clock-rate=(int)8000" ! rtppcmudepay ! queue ! appsink name=appsink emit-signals=true sync=false async=false”<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> With that pipeline, I would get every 20ms a buffer size of 160 bytes. The initial developer has written in the callback a procedure to accumulate that buffer up to a certain size before processing it.<o:p></o:p></p>
<p class="MsoNormal"> Since the app in java (and even if it was not), calling a callback every 20ms is a no go - for me at least.<o:p></o:p></p>
<p class="MsoNormal"> The app does not need to process the audio and it’s main role is to convert the rtp pkt into a grpc pkt to be sent out to Azure cloud for processing etc…<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> I’m looking for such a way that the appsink callback is executed when a certain size (configurable possibly) is available. From my colleagues I got the following pipeline:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">“udpsrc name=udpsrc address="${application.gstreamer.ipAddress}" retrieve-sender-address=false reuse=false port={port} caps="application/x-rtp, media=(string)audio, payload=0, clock-rate=(int)8000" timeout=10000000000 ! rtppcmudepay ! rawaudioparse
sample-rate=48000 num-channels=1 ! audiobuffersplit output-buffer-duration=1/2 ! appsink name=appsink emit-signals=true sync=false async=false”<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> Which improved the buffer size. But I’m uncomfortable with it for few reasons:
<o:p></o:p></p>
<ul style="margin-top:0in" type="disc">
<li class="MsoListParagraph" style="margin-left:0in;mso-list:l0 level1 lfo1">1<sup>st</sup> – I do not see why I need to parse the audio<o:p></o:p></li><li class="MsoListParagraph" style="margin-left:0in;mso-list:l0 level1 lfo1">2<sup>nd</sup> – the audiobuffersplit does not allow me have different buffer size in appsink<o:p></o:p></li></ul>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> I just need a pipeline that extracts the audio from the RTP packet and adds each buffer upto a certain size then pass that buffer to appsink which will be sent out via grpc.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"> I’ve tried different cases but I failed to find the correct pipeline. Any suggestions ?!? Thank you.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">BR,<o:p></o:p></p>
<p class="MsoNormal">Dan S.<o:p></o:p></p>
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