Sending OPUS over RTP using wasapisrc

[Moiz]

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



--
Sent from: http://gstreamer-devel.966125.n4.nabble.com/
_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

[Nirbheek Chauhan]

That pipeline works fine on my Windows 10 machine. You should look at
the debug logs for more information.

On Thu, Oct 15, 2020 at 7:15 AM Moiz <[hidden email]> wrote:

>
> 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
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel

_______________________________________________
gstreamer-devel mailing list
[hidden email]
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel