why does alsasink sync=false fix audio problems?

[Nicolas Dufresne]

Thanks for sharing, excellent point about first synching element latency
setting.

Le 4 mars 2018 14:11, "Charlie Laub" <charleslaub at sbcglobal.net> a écrit :

Nicolas, thank you for your reply. I discovered another solution to the
dsnoop audio glitch problem that I wanted to share because it illuminates
the magnitude of the dsnoop latency.



I am using a pipeline that includes deinterleave and later audiointerleave
so that I can access and modify the audio in individual channels. A
simplistic example is this pipeline:

gst-launch-1.0 -vm alsasrc device=’dsnoop:CARD=Loopback,DEV=1’ !
deinterleave name=input   \

audiointerleave name=output ! alsasink device=’hw:CARD=PCH,DEV=0’
sync=false   \

input.src_0 ! output.sink_0   \

input.src_1 ! output.sink_1  \

The above pipeline still causes audio glitches/dropouts or freezes.



The element audiointerleave has its own latency property that is used to
accommodate various delays in the streams that are sinked to it. I found
that if I set audiointerleave’s latency to 100 milliseconds (100000000)
even when sync=true (the default of the sync property) the audio was also
rendered without problems. The pipeline looks like this:

gst-launch-1.0 -vm alsasrc device=’dsnoop:CARD=Loopback,DEV=1’ !
deinterleave name=input   \

audiointerleave name=output latency=100000000 ! alsasink
device=’hw:CARD=PCH,DEV=0’   \

input.src_0 ! output.sink_0   \

input.src_1 ! output.sink_1  \



My guess is that the additional latency of dsnoop is accommodated by the
audiointerleave latency, meaning the dsnoop latency is between 10 and 100
milliseconds.



Without increasing the audiointerleave latency, even when I set sync=false
I was still experiencing stuttering audio. It seems that the first element
in the pipeline that checks timing of buffers with respect to the pipeline
clock and can drop buffers that are “too late” must have its parameters
adjusted so that the dsnoop latency can be accommodated.



-Charlie



*From:* gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> *On
Behalf Of *Nicolas Dufresne
*Sent:* Saturday, March 3, 2018 11:28 AM
*To:* Discussion of the development of and with GStreamer <
gstreamer-devel at lists.freedesktop.org>
*Subject:* Re: why does alsasink sync=false fix audio problems?





Just like most alsa plugin, dsnoop does not broadcast the extra latency it
is adding. As a side effect, all samples are submitted late and are
dropped. Disabling the sync turns GST into a dummy audio filler. There is
various settings you can play with that will improve this situation, have a
look at latency and buffer-time on alsasrc/sink.



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