G.711 decrease to 8kHz

[Sérgio Agostinho]

Your sending pipeline is sending 8khz data for sure. On the receiving
pipeline your not setting the same capabilities from your sending pipeline

Sending (change your udpsink host accordingly) :
gst-launch -v alsasrc ! audioconvert ! audioresample !
audio/x-raw-int,channels=1,rate=8000 ! alawenc ! rtppcmapay ! udpsink
port=3001

Receiver:
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

This is working fine for me. If possible always copy the rtppcmapay:src
capabilities to the udpsrc caps property.


2014-12-18 15:36 GMT+01:00 Footniko <Footniko at gmail.com>:
>
> I'm getting the same (almost):
> *gst-launch -v alsasrc ! audioconvert ! audioresample !
> audio/x-raw-int,channels=1,rate=8000 ! alawenc ! rtppcmapay ! udpsink
> host=192.168.1.16 port=3001*
> Setting pipeline to PAUSED ...
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstAudioSrcClock
> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:sink: caps =
> audio/x-raw-int, rate=(int)8000, channels=(int)1, endianness=(int)1234,
> width=(int)16, depth=(int)16, signed=(boolean)true
> /GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:src: caps = audio/x-alaw,
> rate=(int)8000, channels=(int)1
> /GstPipeline:pipeline0/GstALawEnc:alawenc0.GstPad:sink: caps =
> audio/x-raw-int, channels=(int)1, rate=(int)8000, width=(int)16,
> depth=(int)16, endianness=(int)1234, signed=(boolean)true
> /GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0.GstPad:src: 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
> /GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0.GstPad:sink: caps =
> audio/x-alaw, rate=(int)8000, channels=(int)1
> /GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0: timestamp = 1591551409
> /GstPipeline:pipeline0/GstRtpPcmaPay:rtppcmapay0: seqnum = 8238
> /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=(uint)2590973091,
> clock-base=(uint)1591551409, seqnum-base=(uint)8238
> WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't
> record audio fast enough
> Additional debug info:
> gstbaseaudiosrc.c(840): gst_base_audio_src_create ():
> /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
> Dropped 2080 samples. This is most likely because downstream can't keep up
> and is consuming samples too slowly.
>
> But i'm not understanding, why to reproduce this stream, i should use
> command:
> *gst-launch-1.0 udpsrc port=3001 ! application/x-rtp, media=audio,
> clock-rate=16000, encoding-name=PCMA, encoding-params=1, channels=1,
> payload=8 ! rtppcmadepay ! alawdec ! audioconvert ! autoaudiosink*
> with clock-rate 16000 but not 8000 (with 8000 it plays too slowly). It
> important for me, because i'm trying to stream it in web browser, that
> supports PCMA only with 8000 rate.
>
>
>
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/G-711-decrease-to-8kHz-tp4669957p4669966.html
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