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	<h1 id="firstHeading" class="firstHeading" lang="en"><span dir="auto">PulseAudio/Configuration</span></h1>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> This is a work-in-progress page, do not follow (yet). I plan on explaining most of PA's pitfalls, how it works, and ultimately how to make a fully custom PA setup. Feel free to contact me for any issues on this page as this is my first Wiki page ever. (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<div id="toc" class="toc">
<div id="toctitle"><h2>Contents</h2></div>
<ul>
<li class="toclevel-1 tocsection-1"><a href="#Introduction"><span class="tocnumber">1</span> <span class="toctext">Introduction</span></a></li>
<li class="toclevel-1 tocsection-2"><a href="#Architecture"><span class="tocnumber">2</span> <span class="toctext">Architecture</span></a></li>
<li class="toclevel-1 tocsection-3">
<a href="#Configuration_files"><span class="tocnumber">3</span> <span class="toctext">Configuration files</span></a>
<ul>
<li class="toclevel-2 tocsection-4"><a href="#daemon.conf"><span class="tocnumber">3.1</span> <span class="toctext">daemon.conf</span></a></li>
<li class="toclevel-2 tocsection-5"><a href="#default.pa"><span class="tocnumber">3.2</span> <span class="toctext">default.pa</span></a></li>
<li class="toclevel-2 tocsection-6"><a href="#client.conf"><span class="tocnumber">3.3</span> <span class="toctext">client.conf</span></a></li>
</ul>
</li>
<li class="toclevel-1 tocsection-7"><a href="#Client_to_server_connection_.26_authentication"><span class="tocnumber">4</span> <span class="toctext">Client to server connection &amp; authentication</span></a></li>
<li class="toclevel-1 tocsection-8">
<a href="#Modules"><span class="tocnumber">5</span> <span class="toctext">Modules</span></a>
<ul>
<li class="toclevel-2 tocsection-9"><a href="#ALSA"><span class="tocnumber">5.1</span> <span class="toctext">ALSA</span></a></li>
<li class="toclevel-2 tocsection-10"><a href="#JACK"><span class="tocnumber">5.2</span> <span class="toctext">JACK</span></a></li>
<li class="toclevel-2 tocsection-11"><a href="#Protocols"><span class="tocnumber">5.3</span> <span class="toctext">Protocols</span></a></li>
<li class="toclevel-2 tocsection-12"><a href="#Filters_.26_Routing"><span class="tocnumber">5.4</span> <span class="toctext">Filters &amp; Routing</span></a></li>
</ul>
</li>
<li class="toclevel-1 tocsection-13"><a href="#Further_resources"><span class="tocnumber">6</span> <span class="toctext">Further resources</span></a></li>
</ul>
</div>

<h2><span class="mw-headline" id="Introduction">Introduction</span></h2>
<p>By default, PulseAudio is configured to automatically detect all sound cards and manage them, assuming the user will never ever touch ALSA configurations. It takes control of all detected ALSA devices and redirect all audio streams to itself. While this is the intended behavior for user-friendly distributions like Ubuntu where everything is expected to just work, some Arch users may not want to have PulseAudio control everything for various reasons. Fortunately, none of its features are required at all and the server is highly configurable due to its internal design.
</p>
<h2><span class="mw-headline" id="Architecture">Architecture</span></h2>
<p>PulseAudio runs as a server daemon that can run either system-wide or on per-user basis using a client/server architecture. The daemon by itself does nothing without its modules except to provide an API and host dynamically loaded modules. The audio routing and processing tasks are all handled by various modules, including Pulse's native protocol itself (provided by <a rel="nofollow" class="external text" href="http://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Modules/#index22h3">module-native-protocol-unix</a>). Clients reach the server through one of many protocol modules that will accept audio from external sources, route it through PulseAudio and eventually have it go out through a final other module. The output module doesn't have to be an actual sound output: it can dump the stream into a file, stream it to a broadcasting server such as <a href="../../en/Streaming_With_Icecast.html" title="Icecast" class="mw-redirect">Icecast</a>, or even just discard it.
</p>
<h2><span class="mw-headline" id="Configuration_files">Configuration files</span></h2>
<p>PulseAudio can be configured in multiple ways depending on your needs and uses multiple different configuration files. Configuration files are read from <code style="display:inline-block; padding: 0.1em 0.3em;">~/.pulse/</code> first, then from <code style="display:inline-block; padding: 0.1em 0.3em;">/etc/pulse/</code> for system-wide defaults. People usually change the system-wide version unless they intend to have multiple users with different configurations.
</p>
<h3><span class="mw-headline" id="daemon.conf"><code style="display:inline-block; padding: 0.1em 0.3em;">daemon.conf</code></span></h3>
<p>This is the main configuration file to configure the daemon itself. It defines base settings like the default sample rates used by modules, resampling methods, realtime scheduling and various other settings related to the server process. These can not be changed at runtime without restarting the PulseAudio daemon. Most defaults make sense here and are self-explaining, see the <code style="display:inline-block; padding: 0.1em 0.3em;">pulse-daemon.conf</code> <a href="../../en/Man_page.html" title="Man" class="mw-redirect">man</a> page for additional information. Boolean options accepts any of these: <code style="display:inline-block; padding: 0.1em 0.3em;">true</code>, <code style="display:inline-block; padding: 0.1em 0.3em;">yes</code>, <code style="display:inline-block; padding: 0.1em 0.3em;">on</code> and <code style="display:inline-block; padding: 0.1em 0.3em;">1</code> as well as <code style="display:inline-block; padding: 0.1em 0.3em;">false</code>, <code style="display:inline-block; padding: 0.1em 0.3em;">no</code>, <code style="display:inline-block; padding: 0.1em 0.3em;">off</code> and <code style="display:inline-block; padding: 0.1em 0.3em;">0</code>. 
</p>
<table class="wikitable">
<caption> Notable configuration options
</caption>
<tr>
<th> Option </th>
<th> Description
</th>
<caption>
</caption>
<td> daemonize </td>
<td> Controls whether the server will daemonize itself and return. Set to <code style="display:inline-block; padding: 0.1em 0.3em;">no</code> when debugging so you can see the debugging information on the terminal.
</td>
<caption>
</caption>
<td> system-instance </td>
<td> If set to <code style="display:inline-block; padding: 0.1em 0.3em;">yes</code>, the daemon will run as a system-wide instance. Highly discouraged as it can introduce security issues. Can be used when multiple users will use sound at the same time, either remotely or locally using <a href="../../en/Xorg_multiseat.html" title="Xorg multiseat">Xorg_multiseat</a>. Defaults to {{ic|no}.
</td>
<caption>
</caption>
<td> nice-level </td>
<td> Since PulseAudio runs in userspace and involves inter-process communication, audio can be subject to dropouts if the daemon doesn't have enough CPU time to process the audio. The default usually is enough, but can be tweaked to give pulse the wanted priority over (or below) other applications.
</td>
<caption>
</caption>
<td> realtime-scheduling </td>
<td> If your kernel supports realtime scheduling (for instance, <a href="../../en/Kernels.html#-rt" title="Kernels">Kernels#-rt</a> or <a href="../../en/Kernels.html#-ck" title="Kernels">Kernels#-ck</a>), set this to <code style="display:inline-block; padding: 0.1em 0.3em;">yes</code> to ensure PulseAudio can deliver low-latency glitch-free playback. You can adjust <code style="display:inline-block; padding: 0.1em 0.3em;">realtime-priority</code> as well to have it use the correct priority, especially when <a href="../../en/JACK_Audio_Connection_Kit.html" title="JACK" class="mw-redirect">JACK</a> is also running on the system.
</td>
<caption>
</caption>
<td> exit-idle-time </td>
<td> If you want to run PulseAudio only when needed and use ALSA otherwise, you can set a delay in seconds after which the daemon will automatically shutdown after all clients are disconnected. Defaults to -1, which disables this feature.
</td>
<caption>
</caption>
<td> log-level </td>
<td> When debugging, you may want to increase the logging level of the daemon to see exactly why a specific module fails to load. High logging levels will sometimes print useful information such as detected minimum latency for the system, which can then be used to tweak <code style="display:inline-block; padding: 0.1em 0.3em;">default-fragments</code> and <code style="display:inline-block; padding: 0.1em 0.3em;">default-fragment-size-msec</code>.
</td>
<caption>
</caption>
<td> resample-method </td>
<td> When PulseAudio needs to pass audio from one module to another using incompatible sample-rate (for example, playback at 96kHz to a card that only supports a maximum of 48kHz), specifies which resampler to use. You can use the command <code style="display:inline-block; padding: 0.1em 0.3em;">$ pulseaudio --dump-resample-methods</code> to list all available resamplers and choose the best tradeoff between CPU usage and audio quality for your taste. <div style="padding: 5px; margin: 0.50em 0; background-color: #DDFFDD; border: thin solid #BBDDBB; overflow: hidden;">
<strong> Tip: </strong>One of the major complaint about PulseAudio is its high CPU usage in some use cases. A lot of these cases happens when pulse needs to resample multiple streams (individually). If you intend to mix multiple sample rates often, consider creating an additional sink at the correct sample rate which you can then feed back to your main sink, resampling only once.</div>
</td>
<caption>
</caption>
<td> enable-remixing </td>
<td> When the input and output have a different channel count (for example, outputting a 6 channel movie into a stereo sink), pulse can either remix all the channels (default, <code style="display:inline-block; padding: 0.1em 0.3em;">yes</code>) or just trivially map the channels by their name (left goes to left, right to right, all others ignored) when <code style="display:inline-block; padding: 0.1em 0.3em;">no</code>
</td>
<caption>
</caption>
<td> flat-volumes </td>
<td> If <code style="display:inline-block; padding: 0.1em 0.3em;">yes</code> (default), all input sources have their volume relative to the maximum volume of the entire sound card. This allows each application to individually adjust their volume so for example, raising your VoIP call volume will raise the hardware volume and adjust your music player volume so it stays where it was, avoiding confusion by having to lower manually the music player then raise the global volume. <div style="padding: 5px; margin: 0.50em 0; background-color: #FFDDDD; border: thin solid #DDBBBB; overflow: hidden;">
<strong> Warning: </strong>Sometimes this can be more confusing than what it solves and some applications unaware of this feature can set their volume at 100% at startup, potentially blowing your speakers or your ears. If unsure, prefer to set this to &lt;codeno&gt;&lt;/code&gt; so pulse will use the classic ALSA behavior instead.</div>
</td>
<caption>
</caption>
<td> default-sample-format </td>
<td> Usually doesn't need change, but if your sound card's native format is diffent performance and quality can be improved by setting the right format here.
</td>
<caption>
</caption>
<td> default-sample-rate </td>
<td> The default sample rate user by pulse unless overriden at module level. Change this if your sound card doesn't support 44100Hz or if you wish to upsample all audio. See previous note about CPU usage.
</td>
<caption>
</caption>
<td> alternate-sample-rate </td>
<td> To fix a common limitation where movies at 48000Hz were needlessly downsampled to 44100Hz, some modules support changing their sample rate dynamically to avoid resampling when possible. See manual for more in-depth information. Usually don't need any change.
</td>
<caption>
</caption>
<td> default-channels </td>
<td> The default number of channels when not specified. Usually don't need any change as you can configure more channels on per-module basis.
</td>
<caption>
</caption>
<td> default-fragments </td>
<td> Audio samples are splitted into multiple fragments of <code style="display:inline-block; padding: 0.1em 0.3em;">default-fragment-size-msec</code> each. The larger the buffer is, the less likely audio will skip when the system is overloaded, but will also increase the overall latency. Increase this value if you have issues.
</td>
<caption>
</caption>
<td> default-fragment-size-msec </td>
<td> The size in milliseconds of each fragment. This is the amount of data that will be processed at once by the daemon. TODO: Verify
</td>
</tr>
</table>
<p><br>
</p>
<h3><span class="mw-headline" id="default.pa"><code style="display:inline-block; padding: 0.1em 0.3em;">default.pa</code></span></h3>
<p>This file is called a startup script and is used to configure modules. It is a script because it is actually parsed and read after the daemon has finished initializing and the settings can be sent at runtime using <code style="display:inline-block; padding: 0.1em 0.3em;">$ pactl</code> or <code style="display:inline-block; padding: 0.1em 0.3em;">$ pacmd</code>. The PulseAudio daemon itself can also be started in terminal mode (<code style="display:inline-block; padding: 0.1em 0.3em;">$ pulseaudio -nC</code> where it will accept its configuration from STDIN and output to STDOUT instead of of loading the <code style="display:inline-block; padding: 0.1em 0.3em;">default.pa</code> file. This can be useful when debugging the daemon or just to test various modules before setting them permanently on disk.
</p>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> TODO: Add syntax information and links to the official Wiki which has plenty enough information for every modules (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<p><br>
</p>
<h3><span class="mw-headline" id="client.conf"><code style="display:inline-block; padding: 0.1em 0.3em;">client.conf</code></span></h3>
<p>This is the configuration file read by every PulseAudio client applications. It is used to configure runtime options for individual clients. It can be used to set the configure the default sink and source statically as well as allowing (or disallowing) clients to automatically start the server if not currently running.
</p>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> TODO: Add information about common options (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<p><br>
</p>
<h2><span class="mw-headline" id="Client_to_server_connection_.26_authentication">Client to server connection &amp; authentication</span></h2>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> TODO: Explain how clients find out where the pulse socket is and authentication mechanism used. Talk about environment variables such as PULSE_SINK, PULSE_COOKIE and PULSE_SERVER. Also mention that X11 properties are set to allow clients to discover pulse via their DISPLAY. (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<p><br>
</p>
<h2><span class="mw-headline" id="Modules">Modules</span></h2>
<h3><span class="mw-headline" id="ALSA">ALSA</span></h3>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> TODO: Explain both static ALSA configuration and automatic detection. How to use ALSA plugins. Mention that PulseAudio likes real hardware way better than plugins for latency and power-saving controls (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<h3><span class="mw-headline" id="JACK">JACK</span></h3>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> Jack auto-detection or static Jack client (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<h3><span class="mw-headline" id="Protocols">Protocols</span></h3>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> Local sound, network sound, file sounds, streaming servers (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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<h3><span class="mw-headline" id="Filters_.26_Routing">Filters &amp; Routing</span></h3>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> Collection of useful modules for audio processing (EQ) and routing (combining card, splitting cards, immitating JACK features (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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</p>
<h2><span class="mw-headline" id="Further_resources">Further resources</span></h2>
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<td>  <b>This article is a stub.</b> <div style="font-size: 90%">
<b>Notes:</b> Links to useful documentation, official PA wiki, manual pages, etc. (<a rel="nofollow" class="external text" href="https://wiki.archlinux.org/index.php/Talk:PulseAudio/Configuration">Discuss</a>)</div>
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