Error resolution or example for webrtcdsp with webrtcechoprobe

ankit ankit.patel1 at einfochips.com
Wed Apr 3 15:39:23 UTC 2019


>As for every embedded system, drivers aren't made for multiple purpose. They
are generally written in a rush for a specific use case. Whenever you
deviate from the original use case, it often fail or miss-behave. In this
case, with the echo canceller you now have a hard dependency on correct
latency reporting, and a round-trip latency being less the. 200ms.
>Also, it may not be latency, maybe it picks a different a different rate or
format, and your driver might not be handling it properly. A large amount of
android audio drivers ignores the rate and always assume 48kHz as an
example.
>You could also simply have a fpu that is too slow for this library.
Remember that this is a PC solution. So you need PC like fpu to make it
work.
I am not aware of fpu part. Do you mean to libav? How to check with that?

>It is really easy to bump your GST version in Yocto, so that should not be
an argument to not use the latest debugging tools available. audiolatency
have not external depency, you could even build it on the side using a
simple gcc call in its sub-directory.
I am able to port audiolatency with Yocto. But when I run the pipeline
suggested by you it is giving me error.

root at dev:~# gst-launch-1.0 alsasrc ! audiolatency print-latency=true !
alsasink buffer-time=80000
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Internal
data stream error.
Additional debug info:
../../../../gstreamer-1.12.4/libs/gst/base/gstbasesrc.c(2939):
gst_base_src_loop (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:

After that I have tried with random buffer-time
With value 10000, 25000, 50000 = Sound is distorted and not audible
With value 65000, 80000, 100000 = Sound ok but with echo(echo is maybe minor
less but still echo is there)
Do you have any idea?





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