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📄 commreceive.htm

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2 Final//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" Content="text/html; charset=Windows-1252">
<TITLE>Receiving sound</TITLE>
</HEAD>

<BODY BGCOLOR="#FFFFFF" TEXT="#000000">

<OBJECT TYPE="application/x-oleobject" CLASSID="clsid:1e2a7bd0-dab9-11d0-b93a-00c04fc99f9e">
	<PARAM NAME="Keyword" VALUE="Sound, receiving">
	<PARAM NAME="Keyword" VALUE="Break Input mode">
	<PARAM NAME="Keyword" VALUE="jitter compensation">
</OBJECT>

<H2><A NAME="commreceive"></A>Receiving sound</H2>

<P>As sound packets arrive, they're immediately sent to the
audio output device.  If a packet arrives from a host with
which you haven't <A HREF="newconnection.htm">opened a
connection</A>, a new temporary connection to that host is
automatically opened.  If you respond to the host by either <A
HREF="commsend.htm">sending live audio</A> or a <A
HREF="commsendfile.htm">sound file</A>, the connection becomes
permanent.  Otherwise, a temporary connection is automatically
closed if no sound is received from that host for 30
seconds.</P>

<P>If sound arrives while you're <A HREF="commsend.htm">sending
live audio</A> and your audio hardware is <A
HREF="hardduplex.htm">half-duplex</A>, <i>Speak Freely</i>, by
default, simply discards the incoming sound and increments the
number of lost input packets shown in the <A
HREF="propellerhead.htm">Extended Status</A> dialogue.  If you
check the Options/Break Input menu item, sound that arrives
while you're sending will, instead, interrupt your
transmission, letting you know that the other person wants to
say something.</P>

<P>Network traffic congestion and the fact that packets can
travel on a variety of routes between two sites can lead to <A
HREF="problemsrandpauses.htm">random pauses</A> (jitter) in the
sound you receive.  To reduce the severity of the pauses, <i>Speak
Freely</i> usually delays playback of the first in a sequence of
sound packets to provide some margin for subsequent packets to
arrive, even if slightly delayed.  This improves the quality of
the sound, but at the cost of introducing an additional delay
before you start to hear a transmission from another user.  The
Options/Jitter Compensation menu allows you to select a variety
of anti-jitter delays ranging from none at all to three
seconds.  If you're communicating across a local network,
"None" is the best setting.  The default, 1 second, generally
gives much better results across Internet connections than no
delay.  If you have severe delay problems, you might want to
try a higher setting.  Lower jitter compensation times are
usable when communicating between sites with high-bandwidth
connectivity to the Internet.</P>

<P>If sound arrives simultaneously from more than one host, the
packets are interleaved.  This makes it difficult to
understand, but it does permit interrupting a long winded
speaker in a conference call.</P>

</BODY>
</HTML>

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