G.711 decrease to 8kHz

[Footniko]

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.



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