diff options
author | Lennart Poettering <lennart@poettering.net> | 2006-04-16 13:34:09 +0000 |
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committer | Lennart Poettering <lennart@poettering.net> | 2006-04-16 13:34:09 +0000 |
commit | 7871f41f2e49978b8c5451516e7a464b0985828b (patch) | |
tree | 788a2c3b04e0d2b94bef958052fe07510bd56122 /doc/FAQ.html.in | |
parent | 2f3fa42ca6dddc56c4ddab1d7d8ac89ff6eb75d6 (diff) |
add documentation for the new RTP modules
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@731 fefdeb5f-60dc-0310-8127-8f9354f1896f
Diffstat (limited to 'doc/FAQ.html.in')
-rw-r--r-- | doc/FAQ.html.in | 97 |
1 files changed, 93 insertions, 4 deletions
diff --git a/doc/FAQ.html.in b/doc/FAQ.html.in index 0e738217..7adc2441 100644 --- a/doc/FAQ.html.in +++ b/doc/FAQ.html.in @@ -67,7 +67,7 @@ <tt>realtime</tt>, or increase the fragment sizes of the audio drivers. The former will allow Polypaudio to activate <tt>SCHED_FIFO</tt> high priority scheduling (root rights are dropped - immediately after this) Keep in mind that this is a potential security hole!</p></li> + immediately after this). Keep in mind that this is a potential security hole!</p></li> <li><p><b>The <tt>polypaudio</tt> executable is installed SUID root by default. Why this? Isn't this a potential security hole?</b></p> @@ -103,7 +103,12 @@ in <tt>~/.polypaudio/</tt>.</p></li> <li><p><b>How do I use polypaudio over the network?</b></p> -<p>Just set <tt>$POLYP_SERVER</tt> to the host name of the polypaudio server.</p> +<p>Just set <tt>$POLYP_SERVER</tt> to the host name of the polypaudio +server. For authentication you need the same auth cookies on all sides. For +that copy <tt>~./polypaudio-cookie</tt> to all clients that shall +be allowed to connect.</p> + +<p>Alternatively the authorization cookies can be stored in the X11 server.</p></li> <li><p><b>Is polypaudio capable of providing synchronized audio playback over the network for movie players like <tt>mplayer</tt>?</b></p> @@ -126,7 +131,7 @@ connect to a running polypaudio daemon try using the following commands:</p> <pre>killall -USR2 polypaudio bidilink unix-client:/tmp/polypaudio/cli</pre> -<p><i>BTW: Someone should package that great tool for Debian!</i></p> +<p><i>BTW: Someone should package this great tool for Debian!</i></p> <p><b>New:</b> There's now a tool <tt>pacmd</tt> that automates sending SIGUSR2 to the daemon and running a bidilink like tool for you.</p> </li> @@ -146,7 +151,91 @@ bidilink unix-client:/tmp/polypaudio/cli</pre> </li> <li><p><b>Why the heck does libpolyp link against libX11?</b></p> -<p>The Polypaudio client libraries look for some X11 root window properties for the credentials of the Polypaudio server to access. You may compile Polypaudio without X11 for disabling this.</p></li> +<p>The Polypaudio client libraries look for some X11 root window +properties for the credentials of the Polypaudio server to access. You +may compile Polypaudio without X11 for disabling this feature.</p></li> + +<li><p><b>How can I use Polypaudio as an RTP based N:N multicast +conferencing solution for the LAN?</b></p> <p>After loading all the +necessary audio drivers for recording and playback, just load the RTP +reciever and sender modules with default parameters:</p> + +<pre> +load-module module-rtp-send +load-module module-rtp-recv +</pre> + +<p>As long as the Polypaudio daemon runs, the microphone data will be +streamed to the network and the data from other hosts is played back +locally. Please note that this may cause quite a lot of traffic. Hence +consider passing <tt>rate=8000 format=ulaw channels=1</tt> to the +sender module to save bandwith while still maintaining good quality +for speech transmission.</p></li> + +<li><p><b>What is this RTP/SDP/SAP thing all about?</b></p> + +<p>RTP is the <i>Realtime Transfer Protocol</i>. It is a well-known +protocol for transferring audio and video data over IP. SDP is the <i>Session +Description Protocol</i> and can be used to describe RTP sessions. SAP +is the <i>Session Announcement Protocol</i> and can be used to +announce RTP sessions that are described with SDP. (Modern SIP based VoIP phones use RTP/SDP for their sessions, too)</p> + +<p>All three protocols are defined in IETF RFCs (RFC3550, RFC3551, +RFC2327, RFC2327). They can be used in both multicast and unicast +fashions. Polypaudio exclusively uses multicast RTP/SDP/SAP containing audio data.</p> + +<p>For more information about using these technologies with Polypaudio have a look on the <a href="modules.html#rtp">respective module's documentation</a>. + +<li><p><b>How can I use Polypaudio to stream music from my main PC to my LAN with multiple PCs with speakers?</b></p> + +<p>On the sender side create an RTP sink:</p> + +<pre> +load-module module-null-sink sink_name=rtp +load-module module-rtp-send source=rtp_monitor +set-default-sink rtp +</pre> + +<p>This will make <tt>rtp</tt> the default sink, i.e. all applications will write to this virtual RTP device by default.</p> + +<p>On the client sides just load the reciever module:</p> +<pre> +load-module module-rtp-recv +</pre> + +<p>Now you can play your favourite music on the sender side and all clients will output it simultaneously.</p> + + +<p>BTW: You can have more than one sender machine set up like this. The audio data will be mixed on the client side.</p></li> + +<li><p><b>How can I use Polypaudio to share a single LINE-IN/MIC jack on the entire LAN?</b></p> + +<p>On the sender side simply load the RTP sender module:</p> + +<pre> +load-module module-rtp-send +</pre> + +<p>On the reciever sides, create an RTP source:</p> + +<pre> +load-module module-null-sink sink_name=rtp +load-module module-rtp-recv sink=rtp +set-default-source rtp_monitor +</pre> + +<p>Now the audio data will be available from the default source <tt>rtp_monitor</tt>.</p> + +<li><p><b>When sending multicast RTP traffic it is recieved on the entire LAN but not by the sender machine itself!</b></p> + +<p>Pass <tt>loop=1</tt> to the sender module!</p></li> + +<li><p><b>Can I have more than one multicast RTP group?</b></p> + +<p>Yes! Simply use a new multicast group address. Use +the <tt>destination</tt>/<tt>sap_address</tt> arguments of the RTP +modules to select them. Choose your group addresses from the range +<tt>225.0.0.x</tt> to make sure the audio data never leaves the LAN.</p></li> </ol> |