additions. Add dbhelper.vim key stroke mappings I use for working with docbook source. There are no xsl or css files for customizing the html output so it will look really boring...this will only be temporary. Support for content localization and generation of pdf's is planned. git-svn-id: svn://localhost/ardour2/trunk@1405 d708f5d6-7413-0410-9779-e7cbd77b26cf
282 lines
12 KiB
XML
282 lines
12 KiB
XML
<?xml version="1.0" standalone="no"?>
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<!DOCTYPE section PUBLIC "-//OASIS//DTD DocBook XML V4.4//EN" "http://www.oasis-open.org/docbook/xml/4.4/docbookx.dtd" [
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]>
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<section id="sn-configuring-jack">
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<title>Getting Audio In, Out and Around Your Computer</title>
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<para>
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Before you can begin to use Ardour, you will need to get the audio
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input/output capabilities of your system working and properly configured.
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There are two aspects to this process: getting your audio interface
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(soundcard) working, and configuring it to work with the Jack Audio
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Connection Kit (<ulink url="http://jackaudio.org/">JACK</ulink>).
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</para>
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<section id="sn-jack">
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<title>JACK</title>
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<para>
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It is extremely important to understand that Ardour does not interact
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directly with your audio interface when it is running. Instead, all of the
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audio data signals that Ardour receives and generates are sent to and from
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JACK, a piece of software that routes audio data between an audio interface
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and audio applications, in real time.
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</para>
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<para>
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Traditionally, most of the audio sources that you would want to record, as
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well as a lot of the more significant effects processing, existed outside
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the computer. Consequently one of the biggest issues in integrating a
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computer into the operation of the studio is how to move audio data in and
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out of the computer.
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</para>
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<para>
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However, it is becoming increasingly common for studios to use audio
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sources and effects processing that are comprised completely of software,
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quite often running on the same machine as an audio sequencer or digital
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audio workstation (DAW). A new problem arises in such situations, because
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moving audio in and out of the DAW no longer involves your hardware audio
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interface. Instead, data has to be moved from one piece of software to
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another, preferably with the same kind of sample synchronisation you’d
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have in a properly configured digital hardware system. This is a problem
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that has been solved at least a couple of times (ReWire from PropellerHeads
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and DirectConnect from Digidesign are the two most common examples), but
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JACK is a new design developed as an open source software project, and is
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thusly available for anyone to use, learn from, extend, *fix or modify.
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</para>
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<para>
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New users may not initially realize that by using Jack, their computer
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becomes an extremely flexible and powerful audio tool - especially with
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Ardour acting as the ’heart’ of the system.
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</para>
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</section>
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<section id="getting-audio-working">
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<title>Getting Your Audio Interface Working</title>
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<note>
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<para>
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Although Ardour runs on OS X as well as Linux, this documentation
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describes only a Linux (ALSA) system. The issues faced on OS X tend to be
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entirely different, and are centered mostly on JACK. There are also
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alternative audio device driver families for Linux but they are also not
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discussed here.
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</para>
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</note>
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<para>
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Getting your audio interface working can be the hardest part of setting
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your computer up to run Ardour, or it could be one of the easiest. The
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level of difficulty you will face depends on the type of audio interface
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("soundcard") you are using, the operating system version you are using,
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and your own understanding of how it all works.
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</para>
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<para>
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In an ideal world, your computer already has a working audio interface, and
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all you need do is to start up qjackctl and run JACK. You can determine if
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you face this ideal situation by doing a few simple tests on your machine.
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The most obvious test is whether you’ve already heard audio coming out of
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your computer. If you are in this situation, you can skip ahead to
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<xref linkend="selecting-capture-source"/>.
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</para>
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</section>
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<section id="checking-for-an-audio-interface">
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<title>Checking For an Audio Interface</title>
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<para>
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If you’ve never tried to play audio on your computer before, you should
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use a basic playback program such as play, aplay or possibly xmms. Find an
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audio file on your machine (<command>locate .wav</command> may help here),
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and try to play it. There are several possibilities:
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</para>
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<itemizedlist>
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<listitem>
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<para>
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You may get an error from the program
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</para>
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</listitem>
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<listitem>
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<para>
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You may hear nothing
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</para>
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</listitem>
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<listitem>
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<para>
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You may hear something, but its too quiet
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</para>
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</listitem>
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<listitem>
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<para>
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you may hear something from the wrong loudspeakers.
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</para>
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</listitem>
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</itemizedlist>
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</section>
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<section id="selecting-capture-source">
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<title>Selecting Capture Source</title>
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<para>
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Many audio interfaces, particularly the cheaper varieties that are often
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found built into computers, have ways to plug in both microphones and
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instruments or other audio equipment to be recorded. This immediately poses
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a question: how does Ardour (or any software) know which signal to record,
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the one coming into the microphone input, or the one arriving at the "line
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in" socket? The same question arises also for "high-end" audio interfaces,
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though in different ways.
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</para>
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<para>
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The short answer is: Ardour doesn’t. Instead, this is a choice you have
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to make using a program a program that understands how to control the
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mixing hardware on the audio interface. Linux/ALSA has a number of such
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programs: alsamixer, gamix, aumix, kmix are just a few of them. Each of
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them offers you a way to select which of the possible recordable signals
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will be used for as the "capture source". How you select the preferred
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signal varies from program to program, so you will have to consult the help
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documentation for whichever program you choose to use.
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</para>
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<para>
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There are also a few programs that offer ways to control just one
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particular kind of audio interface. For example, the
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<application>hdspmixer</application> program offers control over the very
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powerful matrix mixer present on several RME audio interface.
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<application>envy24ctrl</application> does the same for a number of
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interfaces built around the common ice1712/envy24 chipset, found in devices
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from M-Audio, Terratec and others. Please note that this quite similar to
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the situation for Windows and MacOS users, where each audio interface often
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comes with its own control program that allows certain critical
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configuration choices to be made.
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</para>
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<section id="problems-with-input-signal">
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<title>"I don’t get any signal when I record …"</title>
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<para>
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The most common problem for first-time audio users on Linux is to try to
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record something and get no signal at all, or alternatively, a very low
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signal. The low signal problem typically arises from one or more of the
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following issues:
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</para>
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<itemizedlist>
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<listitem>
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<para>
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a microphone input plugged into the "line in" socket of the interface.
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The signal levels delivered by microphones are very small, and require
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amplification before they can be used by most audio circuitry. In
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professional recording studios, this is done using a dedicated box
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called a "pre-amplifier". If your audio interface has a "mic input"
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socket, then it has its own pre-amplifier built in, although its
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probably not a very good one. If you make the mistake of plugging a
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microphone into the "line in" socket, you will get either an inaudible
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or very quiet signal.
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</para>
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</listitem>
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<listitem>
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<para>
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the wrong capture source selected in the audio interface’s hardware
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mixer (see above)
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</para>
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</listitem>
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<listitem>
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<para>
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the "capture" gain level in the audio interface’s hardware mixer is
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turned down too low. You will need to use a hardware mixer application
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(as described above) to increase this.
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</para>
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</listitem>
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</itemizedlist>
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<note>
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<para>
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You will notice in the mixer strip for each track in ardour that you can
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change the selection of the monitoring source between input/pre/post.
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Adjusting the fader while watching the ’input’ levels will NOT have
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any affect on the levels. As mentioned above, ardour is dependent on
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external mixer settings for a source level.
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</para>
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</note>
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</section>
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</section>
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<section id="monitoring-choices">
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<title>Monitoring Choices</title>
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<para>
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Its unfortunate that we have to raise this issue at a point in the manual
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where you, the reader, may not even knoiw what "monitoring" means. However,
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it is such an absolutely critical aspect of using any digital audio
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workstation that we need to at least cover the basics here. The only people
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who don’t need to care about monitoring are those who will never use
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ardour to record a live performance (even on performed using a software
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synthesizer).
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</para>
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<para>
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Monitoring is the term we use to describe listening to what ardour is
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recording. If you are playing a guitar and recording it with ardour, you
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can probably hear the guitar’s own sound, but there are many situations
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where relying on the sound of the instrument is completely inadequate. For
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example, with an electronic instrument, there is no sound until the
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electrical signal that it generates has been processed by an amplifier and
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fed to a loudspeaker. But if Ardour is recording the instrument’s signal,
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what is responsible for sending it to the amp+loudspeakers? It can get a
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lot more complex than that: if you are recording multiple performers at the
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same time, each performer needs to hear their own playing/singing, but they
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also probably need to hear some of their colleagues’ sound as well. You
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might be overdubbing yourself - playing a new line on an instrument while
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listening to tracks you’ve already recorded - how do you hear the new
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material as well as the existing stuff?
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</para>
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<para>
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Well, hopefully, you’re convinced that there are some questions to be
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dealt with surrounding monitoring, see <xref linkend="sn-monitoring"/> for
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more in depth information.
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</para>
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</section>
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<section id="using-multiple-soundcards">
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<title>Can I use multiple soundcards</title>
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<para>
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There are really lots of great reasons why you should not even attempt to
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do this. But seriously, save your money for a while and buy yourself a
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properly designed multichannel soundcard.
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</para>
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</section>
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<section id="qjackctl">
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<title>Qjackctl</title>
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<para>
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JACK itself does not come with graphical user interface - to start JACK and
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control it you need to have access to a command line and a basic knowledge
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of Unix-like operating systems. However,
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<ulink url="http://qjackctl.sourceforge.net/">qjackctl</ulink> is a
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wonderful application that wraps JACK up with a graphical interface that is
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both nice to look at and useful at same time. qjackctl is the recommended
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way of using JACK.
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</para>
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<mediaobject>
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<imageobject>
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<imagedata fileref="images/qjackctl.png"/>
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</imageobject>
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</mediaobject>
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<para>
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You should be able to start qjackctl from the “application menu” of
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your system, typically found on the panel/appbar/dock or whatever its
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called that lives at the top/bottom/left/right of your screen.
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</para>
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<para>
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[ need screenshot of GNOME/KDE/OSX menus here ]
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</para>
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</section>
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<!--
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<xi:include xmlns:xi="http://www.w3.org/2001/XInclude"
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href="Some_Subsection.xml" />
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-->
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</section>
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