Sending OPUS over RTP using wasapisrc

Moiz moiz at playpixel.app
Thu Oct 15 00:34:33 UTC 2020


I have just got wasapi to work on my server 2019 machine which has gstreamer
1.18 installed

I can now capture and encode audio into ogg using opusenc, with the
following pipeline

gst-launch-1.0 wasapisrc ! audioconvert ! opusenc ! filesink
location=test.ogg , this works and I get audio however when I switch to
using rtp I get this error

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
/GstPipeline:pipeline0/GstWasapiSrc:wasapisrc0: actual-buffer-time = 200000
/GstPipeline:pipeline0/GstWasapiSrc:wasapisrc0: actual-latency-time = 10000
Redistribute latency...
/GstPipeline:pipeline0/GstWasapiSrc:wasapisrc0.GstPad:src: caps =
audio/x-raw, format=(string)F32LE, layout=(string)interleaved,
rate=(int)48000, channels=(int)8, channel-mask=(bitmask)0x0000000000000c3f
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps =
audio/x-raw, rate=(int)48000, channels=(int)8, format=(string)S16LE,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000c3f
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps =
audio/x-raw, rate=(int)48000, channels=(int)8, format=(string)S16LE,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000c3f
/GstPipeline:pipeline0/GstOpusEnc:opusenc0.GstPad:sink: caps = audio/x-raw,
rate=(int)48000, channels=(int)8, format=(string)S16LE,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000c3f
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps =
audio/x-raw, rate=(int)48000, channels=(int)8, format=(string)S16LE,
layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000c3f
ERROR: from element /GstPipeline:pipeline0/GstWasapiSrc:wasapisrc0: Internal
data stream error.
Additional debug info:
../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop ():
/GstPipeline:pipeline0/GstWasapiSrc:wasapisrc0:
streaming stopped, reason not-negotiated (-4)
/GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps =
audio/x-raw, format=(string)F32LE, layout=(string)interleaved,
rate=(int)48000, channels=(int)8, channel-mask=(bitmask)0x0000000000000c3f
Execution ended after 0:00:00.082267600
Setting pipeline to NULL ...
Freeing pipeline ...

This is my pipeline: gst-launch-1.0 -v rtpbin name=rtpbin rtp-profile=avpf
wasapisrc ! audioconvert ! audioresample ! opusenc ! rtpopuspay ! udpsink
host=reciver port=5000

Is this an issue with wasapi, does wasapi not support streaming audio over
rtp, how can I get around this?

Thanks



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