<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HTML><HEAD> <META http-equiv=Content-Type content="text/html; charset=us-ascii"> <STYLE>.hmmessage P { PADDING-RIGHT: 0px; PADDING-LEFT: 0px; PADDING-BOTTOM: 0px; MARGIN: 0px; PADDING-TOP: 0px } BODY.hmmessage { FONT-SIZE: 10pt; FONT-FAMILY: Verdana } </STYLE> <META content="MSHTML 6.00.2900.5764" name=GENERATOR></HEAD> <BODY class=hmmessage> <DIV dir=ltr align=left><FONT face=Arial color=#0000ff></FONT> </DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff>Hi Zelalem,</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff>You can use rtppcmapay/depay and rtppcmupay/depay elements with alaw and ulaw.</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff>Correct type for alaw is audio/x-alaw and not audio/x-alaw-int.</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff>--Roope</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=800241108-11062009><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV><BR> <BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px"> <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left> <HR tabIndex=-1> <FONT face=Tahoma><B>From:</B> ext Zelalem Sintayehu [mailto:zelalems@hotmail.com] <BR><B>Sent:</B> 11 June, 2009 11:08<BR><B>To:</B> gstreamer-devel@lists.sourceforge.net<BR><B>Subject:</B> Re: [gst-devel] Problem with Recording Audio from a Network<BR></FONT><BR></DIV> <DIV></DIV>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> <STYLE>.ExternalClass .EC_hmmessage P { PADDING-RIGHT: 0px; PADDING-LEFT: 0px; PADDING-BOTTOM: 0px; PADDING-TOP: 0px } .ExternalClass BODY.EC_hmmessage { FONT-SIZE: 10pt; FONT-FAMILY: Verdana } </STYLE> <DIV dir=ltr align=left><FONT face=Arial color=#0000ff></FONT> </DIV> <DIV dir=ltr align=left><FONT face=Arial color=#0000ff></FONT> </DIV> <DIV dir=ltr align=left><SPAN class=EC_405453407-11062009><FONT face=Arial color=#0000ff>Hi,</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=EC_405453407-11062009><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV dir=ltr align=left><SPAN class=EC_405453407-11062009><FONT face=Arial color=#0000ff>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 face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV dir=ltr align=left><SPAN class=EC_405453407-11062009><FONT face=Arial color=#0000ff>--Roope</FONT></SPAN></DIV> <DIV dir=ltr align=left><SPAN class=EC_405453407-11062009></SPAN> </DIV><BR> <BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(0,0,255) 2px solid; MARGIN-RIGHT: 0px"> <DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left> <HR> <FONT face=Tahoma><B></B><BR></FONT><BR></DIV> <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> </DIV></BLOCKQUOTE></DIV><BR><BR></BLOCKQUOTE><BR> <HR> check out the rest of the Windows Live™. More than mail–Windows Live™ goes way beyond your inbox. <A href="http://www.microsoft.com/windows/windowslive/" target=_new>More than messages</A> </BLOCKQUOTE></BODY></HTML>