<div>Hi,</div>
<div>I dont think there is any bug in gstreamer.</div>
<div>I have successfully run "audio capture - UDP send" on one board and "UDP receive - audio playout" on other board. </div>
<div>But we didnt specify ssrc, clock-base, seqnum-base etc.. parameters for udpsrc.</div>
<div>Try this once (your command with minimal udpsrc parameters)</div>
<div> </div>
<div>gst-launch-0.10 -v udpsrc port=1234 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA" ! rtppcmadepay ! audio/x-alaw, rate=8000, channels=1 ! alawdec ! alsasink </div>
<div><br>Regards,</div>
<div>Viraj<br></div>
<div class="gmail_quote">On Mon, Oct 5, 2009 at 7:28 PM, john oliver <span dir="ltr"><<a href="mailto:johnwpoliver@hotmail.com">johnwpoliver@hotmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>Hi,<br><br>I managed to play back the stream live using VLC ( buffer time 100ms) . On the server I'm using<br><br> gst-launch-0.10 -v filesrc location=./audiotest1.pcma blocksize=800 do-timestamp=true ! audio/x-alaw, rate=8000, channels=1 ! queue ! rtppcmapay ! udpsink host=x.x.x.x port=1234<br>
<br><br>Perhaps I can't play it back live using gstreamer because of a bug ?<br><br>John<br>
<hr>
From: <a href="mailto:johnwpoliver@hotmail.com" target="_blank">johnwpoliver@hotmail.com</a><br>To: <a href="mailto:gstreamer-devel@lists.sourceforge.net" target="_blank">gstreamer-devel@lists.sourceforge.net</a><br>Date: Mon, 5 Oct 2009 00:20:39 +0100<br>
Subject: [gst-devel] alsasrc ! alawenc ! rtppcmapay ! udpsink - can receive, record then playback, but not playback live.<br><br>Hi all,<br><br>I'm trying to stream a live source encoded as pcm ulaw over rtp over udp. I'm able to send it out on one side using gst-launch, and on the client side I can save the stream to a file, then play it back. But I'd like to be able to play it back live on the client side, which I'm unable to do. <br>
<br>My server<br><br> gst-launch-0.10 -v alsasrc do-timestamp=true ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=1234 <br><br>/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000<br>
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000<br>/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)8000, channels=(int)2<br>
Pipeline is live and does not need PREROLL ...<br>Setting pipeline to PLAYING ...<br>New clock: GstAudioSrcClock<br>[snip]<br>/GstPipeline:pipeline0/GstRtpPmcaPay:rtppmcapay0.GstPad:sink: caps = audio/x-alaw, rate=(int)8000, channels=(int)1<br>
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)2636653180, clock-base=(guint)4163613013, seqnum-base=(guint)41237<br>
<br>On the client<br>receive and record:<br><br>gst-launch-0.10 -v udpsrc port=1234 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)3682605153, clock-base=(guint)600325920, seqnum-base=(guint)34884" ! rtppcmadepay ! filesink location=./received.pcma<br>
<br>then playback:<br> gst-launch-0.10 -v filesrc location=./received.pcma ! audio/x-alaw, rate=8000, channels=1 ! alawdec ! alsasink blocksize=32<br><br>The above works ok. <br><br>But when I'm trying to use the following to playback live ..<br>
<br> gst-launch-0.10 -v udpsrc do-timestamp=true port=1234 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)3682605153, clock-base=(guint)600325920, seqnum-base=(guint)34884" ! rtppcmadepay ! audio/x-alaw, rate=8000, channels=1 ! alawdec ! alsasink <br>
<br><br>I don't hear anything ( apart from the odd glitch) . It tells me .. <br><br>Setting pipeline to PAUSED ...<br>Pipeline is live and does not need PREROLL ...<br>Setting pipeline to PLAYING ...<br>New clock: GstSystemClock<br>
/GstPipeline:pipeline0/GstRtpPcmaDepay:rtppcmadepay0.GstPad:src: caps = audio/x-alaw, channels=(int)1, rate=(int)8000<br>/GstPipeline:pipeline0/GstRtpPcmaDepay:rtppcmadepay0.GstPad:sink: caps = application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(guint)3682605153, clock-base=(guint)600325920, seqnum-base=(guint)34884<br>
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-alaw, channels=(int)1, rate=(int)8000<br>/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-alaw, channels=(int)1, rate=(int)8000<br>
/GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)8000, channels=(int)1<br>/GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:sink: caps = audio/x-alaw, channels=(int)1, rate=(int)8000<br>
/GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)8000, channels=(int)1<br><br>This is the same caps I see when I playback from the recorded file, almost, where the alawdec sink buffer size is 4096 ..<br>
<br><br><br>/GstPipeline:pipeline0/GstALawDec:alawdec0.GstPad:sink: caps = audio/x-alaw, rate=(int)8000, channels=(int)1<br>alawdec: buffer size: 4096<br>/GstPipeline:pipeline0/GstAlsaSink:alsasink0.GstPad:sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, rate=(int)8000, channels=(int)1<br>
<br>Can anybody spot any glaring mistakes or perhaps give me any pointers, please.<br><br>John<br><br></div><br>------------------------------------------------------------------------------<br>Come build with us! The BlackBerry® Developer Conference in SF, CA<br>
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<div></div><br>-- <br>- Viraj<br>Reality is merely an illusion, albeit a very persistent one.<br>