<div dir="ltr">Your sending pipeline is sending 8khz data for sure. On the receiving pipeline your not setting the same capabilities from your sending pipeline<div><br></div><div>Sending (change your udpsink host accordingly) :</div><div>gst-launch -v alsasrc ! audioconvert ! audioresample ! audio/x-raw-int,channels=1,rate=8000 ! alawenc ! rtppcmapay ! udpsink port=3001<br></div><div><br></div><div>Receiver:</div><div>gst-launch-1.0 -v udpsrc port=3001 caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, payload=(int)8, ssrc=(uint)2590973091, clock-base=(uint)1591551409, seqnum-base=(uint)8238' ! rtppcmadepay ! alawdec ! audioconvert ! autoaudiosink</div><div><br></div><div>This is working fine for me. If possible always copy the rtppcmapay:src capabilities to the udpsrc caps property. </div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2014-12-18 15:36 GMT+01:00 Footniko <span dir="ltr"><<a href="mailto:Footniko@gmail.com" target="_blank">Footniko@gmail.com</a>></span>:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I'm getting the same (almost):<br>
*gst-launch -v alsasrc ! audioconvert ! audioresample !<br>
<span class="">audio/x-raw-int,channels=1,rate=8000 ! alawenc ! rtppcmapay ! udpsink<br>
</span>host=192.168.1.16 port=3001*<br>
<span class="">Setting pipeline to PAUSED ...<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 =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">Pipeline is live and does not need PREROLL ...<br>
Setting pipeline to PLAYING ...<br>
New clock: GstAudioSrcClock<br>
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:sink: caps =<br>
audio/x-raw-int, rate=(int)8000, channels=(int)1, endianness=(int)1234,<br>
width=(int)16, depth=(int)16, signed=(boolean)true<br>
/GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:src: caps = audio/x-alaw,<br>
rate=(int)8000, channels=(int)1<br>
/GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:sink: caps =<br>
</span>audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,<br>
depth=(int)16, endianness=(int)1234, signed=(boolean)true<br>
<span class="">/GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0.GstPad:src: caps =<br>
application/x-rtp, media=(string)audio, clock-rate=(int)8000,<br>
</span>encoding-name=(string)PCMA, payload=(int)8, ssrc=(uint)2590973091,<br>
clock-base=(uint)1591551409, seqnum-base=(uint)8238<br>
<span class="">/GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0.GstPad:sink: caps =<br>
audio/x-alaw, rate=(int)8000, channels=(int)1<br>
</span>/GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0: timestamp = 1591551409<br>
/GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0: seqnum = 8238<br>
<span class="">/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps =<br>
application/x-rtp, media=(string)audio, clock-rate=(int)8000,<br>
</span>encoding-name=(string)PCMA, payload=(int)8, ssrc=(uint)2590973091,<br>
clock-base=(uint)1591551409, seqnum-base=(uint)8238<br>
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't<br>
record audio fast enough<br>
Additional debug info:<br>
gstbaseaudiosrc.c(840): gst_base_audio_src_create ():<br>
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:<br>
Dropped 2080 samples. This is most likely because downstream can't keep up<br>
and is consuming samples too slowly.<br>
<br>
But i'm not understanding, why to reproduce this stream, i should use<br>
command:<br>
*gst-launch-1.0 udpsrc port=3001 ! application/x-rtp, media=audio,<br>
clock-rate=16000, encoding-name=PCMA, encoding-params=1, channels=1,<br>
payload=8 ! rtppcmadepay ! alawdec ! audioconvert ! autoaudiosink*<br>
with clock-rate 16000 but not 8000 (with 8000 it plays too slowly). It<br>
important for me, because i'm trying to stream it in web browser, that<br>
supports PCMA only with 8000 rate.<br>
<br>
<br>
<br>
--<br>
View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/G-711-decrease-to-8kHz-tp4669957p4669966.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/G-711-decrease-to-8kHz-tp4669957p4669966.html</a><br>
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