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Hi Roope, you are right, it allows alaw,mulaw,ac3,mpeg and raw. But alaw and mulaw don't have payloader and depayloader. Anyway, I tried to recieve without depayloader but it still produced the same error. The following is what tried. Please help me.<br><br>gst-launch-0.10 -v udpsrc port=5002 caps="audio/x-rtp,rate=1000,channels=1,depth=8" ! queue ! audio/x-alaw-int,rate=1000,channels=1,depth=8 ! avimux ! filesink location=audio.avi sync=false . I also changed the caps for udpsrc with "audio/x-alaw-int,rate=1000,channels=1,depth=8" but didn't work.<br><br>Thank you.<br><br>- Zelalem S. <br><br><hr id="stopSpelling">From: roope.jarvinen@nokia.com<br>To: gstreamer-devel@lists.sourceforge.net<br>Date: Thu, 11 Jun 2009 09:35:58 +0200<br>Subject: Re: [gst-devel] Problem with Recording Audio from a Network<br><br>
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<div dir="ltr" align="left"><span class="EC_405453407-11062009"><font color="#0000ff" face="Arial">Hi,</font></span></div>
<div dir="ltr" align="left"><span class="EC_405453407-11062009"><font color="#0000ff" face="Arial"></font></span> </div>
<div dir="ltr" align="left"><span class="EC_405453407-11062009"><font color="#0000ff" face="Arial">You cannot mux gsm-encoded audio into AVI container. Check avimux
description for allowed formats.</font></span></div>
<div dir="ltr" align="left"><span class="EC_405453407-11062009"><font color="#0000ff" face="Arial"></font></span> </div>
<div dir="ltr" align="left"><span class="EC_405453407-11062009"><font color="#0000ff" face="Arial">--Roope</font></span></div>
<div dir="ltr" align="left"><span class="EC_405453407-11062009"></span> </div><br>
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<div></div>Hi Jyoti, thank you for your prompt response. I added the following
caps statement, but it is still the same. The following is the modified
receiver side code.<br><br>gst-launch-0.10 -v udpsrc port=5002
caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)GSM,encoding-params=(string)1,octet-align=(string)1"
! queue ! rtpgsmdepay ! audio/x-raw-int,rate=8000,channels=1,depth=8 ! avimux
! filesink location=audio.avi sync=false<br><br>The error is the same:
"WARNING: erroneous pipeline: could not link rtpgsmdepay0 to
avimux0"<br><br>Thank you.<br><br>- Zelalem S.
<br>-----------------------------------<br>From: jyoti.d@allaboutif.com<br>To:
gstreamer-devel@lists.sourceforge.net<br>Subject: Re: [gst-devel] Problem with
Recording Audio from a Network<br><br>You should set caps property on udpsrc
element at receiver side.<br><br>
<div class="EC_EC_gmail_quote">On Wed, Jun 10, 2009 at 7:26 PM, Zelalem Sintayehu
<span dir="ltr"><<a href="mailto:zelalems@hotmail.com">zelalems@hotmail.com</a>></span>
wrote:<br>
<blockquote class="EC_EC_gmail_quote" style="padding-left: 1ex;">
<div>Hi all, I was trying to record audio and video over the network. I
wanted to check that separately and succeeded with the video recording, but
i couldn't record the audio part. BTW, I want to record into avi file and I
used h263p for the video and wanted to use either amr or gsm for the audio.
I hope this is possible. The following is my code both from teh sender and
reciever side.<br><br>Sender:<br>gst-launch-0.10 -v alsasrc ! queue !
audio/x-raw-int,rate=8000,channels=1,depth=8 ! audioconvert ! gsmenc !
rtpgsmpay ! queue ! udpsink
port=5002<br><br>Receiver:<br><br>gst-launch-0.10 -v udpsrc port=5002 !
queue ! rtpgsmdepay ! audio/x-raw-int,rate=8000,channels=1,depth=8 ! avimux
! filesink location=audio.avi sync=false<br><br>and I got the following
error and it terminates. "WARNING: erroneous pipeline: could not link
rtpgsmdepay0 to avimux0. By the way, when i store both the video later,
should i still put "sync=false" (the last statement in the reciever code)?
<br><br><br>Please help me.<br><br>
<hr>
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