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<div class="moz-cite-prefix">On 03/20/2015 05:04 PM, Henrikas
Urbonas wrote:<br>
</div>
<blockquote
cite="mid:CALSKpTkTewKsOFmJFFQ+cJrnNzTUrQ6EJsfL_WANYr3nJ3E3cw@mail.gmail.com"
type="cite">
<div dir="ltr">When you change "latency" while pipeline is
running, nothing happens.
<div>From documentation it looks like latency can be set and
have effect only before pipeline is in playing state.</div>
<div>Because if you change latency when pipeline is in playing
state it actually does not change the latency (we use jitter
buffer).</div>
</div>
</blockquote>
since you've tested this, the next step would be to file a feature
request for the jitter buffer to allow dynamic latency changes.<br>
<blockquote
cite="mid:CALSKpTkTewKsOFmJFFQ+cJrnNzTUrQ6EJsfL_WANYr3nJ3E3cw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div><br>
</div>
<div>Are there other ways to change latency except using rtspsrc
jitter buffer?</div>
</div>
</blockquote>
This is at least one of the elements that in the pipeline that
introduce latency and that can be configured. For other elements the
latency might be static. <br>
<br>
<blockquote
cite="mid:CALSKpTkTewKsOFmJFFQ+cJrnNzTUrQ6EJsfL_WANYr3nJ3E3cw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>And why Android cannot use 48 kHz sample rate because some
devices have lower output latency when using this (48kHz)
sample rate?</div>
</div>
</blockquote>
I am not using gstreamer on android. You would need to start looking
at the audio source and sink elements and figure out why they won't
work at 48kHz (if that is the case).<br>
<br>
Stefan<br>
<br>
<blockquote
cite="mid:CALSKpTkTewKsOFmJFFQ+cJrnNzTUrQ6EJsfL_WANYr3nJ3E3cw@mail.gmail.com"
type="cite">
<div dir="ltr">
<div><br>
</div>
<div>Thanks guys</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-03-20 10:12 GMT+02:00 Stefan Sauer
<span dir="ltr"><<a moz-do-not-send="true"
href="mailto:ensonic@hora-obscura.de" target="_blank">ensonic@hora-obscura.de</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"><span class="">
<div>On 03/18/2015 05:33 PM, Henrikas Urbonas wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">So that means to adjust it without
glithces we have to write a plugin for Gstreamer?</div>
</blockquote>
<br>
</span> I think most part of the latency is the buffering
time (e.g. in the jitter buffer). I would first try what
happens if you change the 'latency' parameter while the
pipeline is running.<span class="HOEnZb"><font
color="#888888"><br>
<br>
STefan <br>
</font></span>
<div>
<div class="h5"> <br>
<blockquote type="cite">
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-03-18 18:23
GMT+02:00 Tim Müller <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:tim@centricular.com"
target="_blank">tim@centricular.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">On Wed, 2015-03-18 at
17:54 +0200, Henrikas Urbonas wrote:<br>
<br>
Hi,<br>
<span><br>
> We work on RTSP protocol and we want to
dynamically change sound<br>
> latency on client side without
re-initializing the pipeline, did<br>
> anyone tried it?<br>
<br>
</span>I think changing the latency on the fly
at runtime should work fine<br>
already, but what we don't have yet is a way
to smoothly adjust the<br>
latency over a period of time so that you can
do adjustments without<br>
glitches.<br>
<br>
Cheers<br>
<span><font color="#888888"> -Tim<br>
<br>
--<br>
Tim Müller, Centricular Ltd - <a
moz-do-not-send="true"
href="http://www.centricular.com"
target="_blank">http://www.centricular.com</a><br>
<br>
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