7:46 a.m.
Crantila,
Are you ready to push this to docs.fp.o so we can all see how it's going?
--Eric
---------- Forwarded message ----------
From: crantila <crantila(a)fedoraproject.org>
Date: Wed, Jul 28, 2010 at 02:27
Subject: [musicians-guide] added many chapters
To: docs-commits(a)lists.fedoraproject.org
commit 4447fcd58d2f37b6acd03e1936df9194e3a8a5dd
Author: Christopher Antila <crantila(a)fedoraproject.org>
Date: Wed Jul 28 02:27:12 2010 -0400
added many chapters
en-US/Book_Info.xml | 6 +-
en-US/Digital_Audio_Workstations.xml | 244 ++++++++++++++++++++++++++++++++++
en-US/Musicians_Guide.xml | 9 +-
en-US/Planet_CCRMA_at_Home.xml | 139 +++++++++++++++++++
en-US/Real_Time_and_Low_Latency.xml | 77 +++++++++++
en-US/Sound_Cards.xml | 129 ++++++++++++++++++
en-US/Sound_Servers.xml | 141 ++++++++++++++++++++
7 files changed, 741 insertions(+), 4 deletions(-)
---
diff --git a/en-US/Book_Info.xml b/en-US/Book_Info.xml
index a4670a9..2738af5 100644
--- a/en-US/Book_Info.xml
+++ b/en-US/Book_Info.xml
@@ -5,14 +5,14 @@
]>
<bookinfo id="book-Musicians_Guide-Musicians_Guide">
<title>Musicians' Guide</title>
- <subtitle>short description</subtitle>
+ <subtitle>A guide to Fedora Linux's audio creation and music
capabilities.</subtitle>
+<!-- Haydn: <subtitle>A guide to Fedora Linux's audio creation and
music capabilities, written in a new and special way.</subtitle> -->
<productname>Fedora Draft Documentation</productname>
<productnumber></productnumber>
<edition>14.0.1</edition>
<pubsnumber>0</pubsnumber>
<abstract>
- <para>
- A short overview and summary of the book's
subject and purpose, traditionally no more than one paragraph long.
Note: the abstract will appear in the front matter of your book and
will also be placed in the description field of the book's RPM
spec file.
+ <para>This document explores some audio-creation and
music activities possible with Fedora Linux. Computer audio concepts
are explained, and a selection of programs are demonstrated with
tutorials showing a typical usage.
</para>
</abstract>
<corpauthor>
diff --git a/en-US/Digital_Audio_Workstations.xml
b/en-US/Digital_Audio_Workstations.xml
new file mode 100644
index 0000000..3b626f8
--- /dev/null
+++ b/en-US/Digital_Audio_Workstations.xml
@@ -0,0 +1,244 @@
+<?xml version='1.0' encoding='utf-8' ?>
+<!DOCTYPE chapter PUBLIC "-//OASIS//DTD DocBook XML V4.5//EN"
"http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd" [
+<!ENTITY % BOOK_ENTITIES SYSTEM "Musicians_Guide.ent">
+%BOOK_ENTITIES;
+]>
+
+<chapter id="chap-Musicians_Guide-Digital_Audio_Workstations">
+ <title>Digital Audio Workstations</title>
+ <para>
+ The term '''Digital Audio Workstation'''
(henceforth
'''DAW''') refers to the entire hardware and software setup used
for
professional (or professional-quality) audio recording, manipulation,
synthesis, and production. It originally referred to devices
purpose-built for the task, but as personal computers have become more
powerful and wide-spread, certain specially-designed personal
computers can also be thought of as DAWs. The software running on
these computers, especially software capable of multi-track recording,
playback, and synthesis, is simply called "DAW software," which is
often shortened to "DAW." So, the term "DAW" and its usage are
moderately ambiguous, but generally refer to one of the things
mentioned.
+ </para>
+
+ <section id="sect-Musicians_Guide-Knowing_Which_DAW_to_Use">
+ <title>Knowing Which DAW to Use</title>
+ <para>
+ The Musicians' Guide covers three widely-used
DAWs: Ardour, Qtractor, and Rosegarden. All three use JACK
extensively, are highly configurable, share a similar user interface,
and allow users to work with both audio and MIDI signals. Many other
DAWs exist, including a wide selection of commercially-available
solutions. Here is a brief description of the programs documented in
this Guide:
+ <itemizedlist>
+ <listitem><para>Ardour: the open-source
standard for audio manipulation. Flexible and
extensible.</para></listitem>
+ <listitem><para>Qtractor: a relative
new-comer, but easy to use; a "lean and mean," MIDI-focused DAW.
Available from Planet CCRMA at Home or RPM Fusion.</para></listitem>
+ <listitem><para>Rosegarden: a well-tested,
feature-packed workhorse of Linux audio, especially MIDI. Includes a
visual score editor for creating MIDI tracks.</para></listitem>
+ </itemizedlist>
+ </para>
+ <para>
+ If you are unsure of where to start, then you
may not need a DAW at all:
+ <itemizedlist>
+ <listitem><para>If you are looking for a
high-quality recording application, or a tool for manipulating one
audio file at a time, then you would probably be better off with
Audacity. This will be the choice of most computer users, especially
those new to computer audio, or looking for a quick solution requiring
little specialized knowledge. Audacity is also a good way to get your
first computer audio experiences, specifically because it is easier to
use than most other audio software.</para></listitem>
+ <listitem><para>To take full advantage of the
features offered by Ardour, Qtractor, and Rosegarden, your computer
should be equipped with professional-quality audio equipment,
including an after-market audio interface and input devices like
microphones. If you do not have access to such equipment, then
Audacity may be a better choice for you.</para></listitem>
+ <listitem><para>If you are simply hoping to
create a "MIDI recording" of some sheet music, you are probably better
off using LilyPond. This program is designed primarily to create
printable sheet music, but it will produce a MIDI-format version of a
score if you include the following command in the "score" section of
your LilyPond source file: <code>\midi { }</code>. There are a
selection of options that can be put in the "midi" section; refer to
the LilyPond help files for a listing.</para></listitem>
+ </itemizedlist>
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Stages_of_Recording">
+ <title>Stages of Recording</title>
+ <para>
+ There are three main stages involved in the
the process of recording something and preparing it for listeners:
recording, mixing, and mastering. Each step of the process has
distinct characteristics, yet they can sometimes be mixed together.
+ </para>
+ <section id="sect-Musicians_Guide-Recording">
+ <title>Recording</title>
+ <para>
+ Recording is the process of
capturing audio regions (also called "clips" or "segments") into the
DAW software, for later processing. Recording is a complex process,
involving a microphone that captures sound energy, translates it into
electrical energy, and transmits it to an audio interface. The audio
interface converts the electrical energy into digital signals, and
sends it through the operating system to the DAW software. The DAW
stores regions in memory and on the hard drive as required. Every
time the musicians perform some (or all) of the performance to be
recorded, while the DAW is recording, it is considered to be a
'''take'''. A successful recording usually requires several
takes,
due to the inconsistencies of musical performance and of the related
technological aspects.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Mixing">
+ <title>Mixing</title>
+ <para>
+ Mixing is the process through
which recorded audio regions (also called "clips") are coordinated to
produce an aesthetically-appealing musical output. This usually takes
place after recording, but sometimes additional takes will be needed.
Mixing often involves reducing audio from multiple tracks into two
channels, for stereo audio - a process known as "down-mixing," because
it decreases the amount of audio data.
+ </para>
+ <para>
+ Mixing includes the following
procedures, among others:
+ <itemizedlist>
+ <listitem><para>automating
effects,</para></listitem>
+ <listitem><para>adjusting
levels,</para></listitem>
+
<listitem><para>time-shifting,</para></listitem>
+
<listitem><para>filtering,</para></listitem>
+
<listitem><para>panning,</para></listitem>
+ <listitem><para>adding special
effects.</para></listitem>
+ </itemizedlist>
+ </para>
+ <para>
+ When the person performing the
mixing decides that they have finished, their finalized production is
called the '''final mix'''.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Mastering">
+ <title>Mastering</title>
+ <para>
+ Mastering is the process
through which a version of the final mix is prepared for distribution
and listening. Mastering can be performed for many target formats,
including CD, tape, SuperAudio CD, or hard drive. Mastering often
involves a reduction in the information available in an audio file:
audio CDs are commonly recorded with 20- or 24-bit samples, for
example, and reduced to 16-bit samples during mastering. While most
physical formats (like CDs) also specify the audio signal's format,
audio recordings mastered to hard drive can take on many formats,
including OGG, FLAC, AIFF, MP3, and many others. This allows the
person doing the mastering some flexibility in choosing the quality
and file size of the resulting audio.
+ </para>
+ <para>
+ Even though they are both
distinct activities, mixing and mastering sometimes use the same
techniques. For example, a mastering technician might apply a
specific equalization filter to optimize the audio for a particular
physical medium.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Record_Mix_Master_More_Info">
+ <title>More Information</title>
+ <para>
+ It takes experience and
practice to gain the skills involved in successful recording, mixing,
and mastering. Further information about these procedures is
available from many places, including these web pages:
+ <itemizedlist>
+
<listitem><para>[http://www.64studio.com/howto-mastering "Mastering
your final mix"]</para></listitem>
+
<listitem><para>[http://en.wikipedia.org/wiki/Audio_mixing_%28rec...
"Audio mixing (recorded music)"]</para></listitem>
+
<listitem><para>[http://en.wikipedia.org/wiki/Multitrack_recording
"Multitrack recording"]</para></listitem>
+ </itemizedlist>
+ </para>
+ </section>
+ </section>
+
+ <section
id="sect-Musicians_Guide-DAW_Audio_Vocabulary_Transclusion">
+ <title>Audio Vocabulary</title>
+ <para>
+ This part is going to be transcluded form the
"Audio Vocabulary" page.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-DAW_Interface_Vocabulary">
+ <title>Interface Vocabulary</title>
+ <para>
+ Understanding these concepts is essential to
understanding how to use the DAW software's interface.
+ </para>
+
+ <section id="sect-Musicians_Guide-Session">
+ <title>Session</title>
+ <para>
+ A '''session''' is all of the
tracks,
regions, automation settings, and everything else that goes along with
one "file" saved by the DAW software. Some software DAWs manage to
hide the entire session within one file, but others instead create a
new directory to hold the regions and other data.
+ </para>
+ <para>
+ Typically, one session is used to hold
an entire recording session; it is broken up into individual songs or
movements after recording. Sometimes, as in the tutorial examples
with the Musicians' Guide, one session holds only one song or
movement. There is no strict rule as to how much music should be held
within one session, so your personal preference can determine what you
do here.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Track_and_Multitrack">
+ <title>Track and Multitrack</title>
+ <para>
+ A '''track''' represents one
channel,
or a predetermined collection of simultaneous, inseparable channels
(as is often the case with stereo audio). In the DAW's main window,
tracks are usually represented as rows, whereas time is represented by
columns. A track may hold multiple regions, but usually only one of
those regions can be heard at a time. The '''multitrack'''
capability
of modern software-based DAWs is one of the reasons for their success.
Although each individual track can play only one region at a time,
the use of multiple tracks allows the DAW's outputted audio to contain
a virtually unlimited number of simultaneous regions. The most
powerful aspect of this is that audio does not have to be recorded
simultaneously in order to be played back simultaneously; you could
sing a duet with yourself, for example.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Region_Clip_Segment">
+ <title>Region, Clip, or Segment</title>
+ <para>
+ Region, clip, and segment are
synonyms: different software uses a different word to refer to the
same thing. A '''region''' (or '''clip'''
or '''segment''') is the
portion of audio recorded into one track during one take. Regions are
represented in the main DAW interface window as a rectangle, usually
coloured, and always contained in only one track. Regions containing
audio signal data usually display a spectrographic representation of
that data. Regions containing MIDI signal data usually displayed as
matrix-based representation of that data.
+ </para>
+ <para>
+ For the three DAW applications in the
Musicians' Guide:
+ <itemizedlist>
+ <listitem><para>Ardour calls them
"regions,"</para></listitem>
+ <listitem><para>Qtractor calls them
"clips," and,</para></listitem>
+ <listitem><para>Rosegarden calls them
"segments."</para></listitem>
+ </itemizedlist>
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Session_Track_Region">
+ <title>Relationship of Session, Track, and Region</title>
+ <para>
+ <!--
[[File:Ardour-session_track_region.xcf]] -->
+
[[File:Ardour-session_track_region.png|200px|left|Session, Track, and
Region in Ardour.]]
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Transport_and_Playhead">
+ <title>Transport and Playhead</title>
+ <para>
+ The '''transport''' is responsible
for
managing the current time in a session, and with it the playhead.
The '''playhead''' marks the point on the timeline from where
audio
audio would be played, or to where audio would be recorded. The
transport controls the playhead, and whether it is set for recording
or only playback. The transport can move the playhead forward or
backward, in slow motion, fast motion, or real time. In most
computer-based DAWs, the playhead can also be moved with the cursor.
The playhead is represented on the DAW interface as a vertical line
through all tracks. The transport's buttons and displays are usually
located in a toolbar at the top of the DAW window, but some people
prefer to have the transport controls detached from the main
interface, and this is how they appear by default in Rosegarden.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Automation">
+ <title>Automation</title>
+ <para>
+ Automation of the DAW sounds like it
might be an advanced topic, or something used to replace decisions
made by a human. This is absolutely not the case -
'''automation'''
allows the user to automatically make the same adjustments every time
a session is played. This is superior to manual-only control because
it allows very precise, gradual, and consistent adjustments, because
it relieves you of having to remember the adjustments, and because it
allows many more adjustments to be made simultaneously than you could
make manually. The reality is that automation allows super-human
control of a session. Most settings can be adjusted by means of
automation; the most common are the fader and the panner.
+ </para>
+ <para>
+ The most common method of automating a
setting is with a two-dimensional graph called an '''envelope''',
which is drawn on top of an audio track, or underneath it in an
'''automation track'''. The user adds adjustment points by
adding
and moving points on the graph. This method allows for complex,
gradual changes of the setting, as well as simple, one-time changes.
Automation is often controlled by means of MIDI signals, for both
audio and MIDI tracks. This allows for external devices to adjust
settings in the DAW, and vice-versa - you can actually automate your
own hardware from within a software-based DAW! Of course, not all
hardware supports this, so refer to your device's user manual.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-DAW_User_Interface">
+ <title>User Interface</title>
+ <para>
+ <!--
+File:Qtractor-interface.xcf
+File:Qtractor-interface-clocks.png
+File:Qtractor-interface-messages.png
+File:Qtractor-interface-track.png
+File:Qtractor-interface-track_info.png
+File:Qtractor-interface-transport.png
+-->
+This section describes various components of software-based DAW
interfaces. Although the Qtractor application is visible in the
images, both Ardour and Rosegarden (along with most other DAW
software) have an interface that differs only in details, such as
which buttons are located where.
+ </para>
+
+ <section id="sect-Musicians_Guide-Messages_Pane">
+ <title>"Messages" Pane</title>
+ <para>
+
[[File:Qtractor-interface-messages.png|300px|"Messages" Pane]]
+
+ The "messages" pane, shown in the
above diagram, contains messages produced by the DAW, and sometimes
messages produced by software used by the DAW, such as JACK. If an
error occurs, or if the DAW does not perform as expected, you should
check the "messages" pane for information that may help you to get the
desired results. The "messages" pane can also be used to determine
whether JACK and the DAW were started successfully, with the options
you prefer.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-DAW_Clock">
+ <title>Clock</title>
+ <para>
+
[[File:Qtractor-interface-clocks.png|300px|Clock]]
+
+ The clock shows the current place in
the file, as indicated by the transport. In the image, you can see
that the transport is at the beginning of the session, so the clock
indicates "0". This clock is configured to show time in minutes and
seconds, so it is a "time clock." Other possible settings for clocks
are to show BBT (bars, beats, and ticks - a "MIDI clock"), samples (a
"sample clock"), or an SMPTE timecode (used for high-precision
synchronization, usually with video - a "timecode clock"). Some DAWs
allow the use of multiple clocks simultaneously.
+ </para>
+ <para>
+ Note that this particular time clock
in "Qtractor" also offers information about the MIDI tempo and metre
(120.0 beats per minute, and 4/4 metre), along with a quantization
setting for MIDI recording.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Track_Info_Pane">
+ <title>"Track Info" Pane</title>
+ <para>
+
[[File:Qtractor-interface-track_info.png|300px|"Track Info" Pane]]
+
+ The "track info" pane contains
information and settings for each track and bus in the session. Here,
you can usually adjust settings like the routing of a track's or bus'
input and output routing, the instrument, bank, program, and channel
of MIDI tracks, and the three buttons shown on this image: "R" for
"arm to record," "M" for "mute/silence track's output,"
and "S" for
"solo mode," where only the selected tracks and busses are heard.
+ </para>
+ <para>
+ The information provided, and the
layout of buttons, can change dramatically between DAWs, but they all
offer the same basic functionality. Often, right-clicking on a track
info box will give access to extended configuration options.
Left-clicking on a portion of the track info box that is not a button
allows you to select a track without selecting a particular moment in
"track" pane.
+ </para>
+ <para>
+ The "track info" pane does not scroll out
of view as the "track" pane is adjusted, but is independent.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Track_Pane">
+ <title>"Track" Pane</title>
+ <para>
+
[[File:Qtractor-interface-track.png|300px|"Track" Pane]]
+
+ The "track" pane is the main workspace
in a DAW. It shows regions (also called "clips") with a rough
overview of the audio wave-form or MIDI notes, allows you to adjust
the starting-time and length of regions, and also allows you to assign
or re-assign a region to a track. The "track" pane shows the
transport as a vertical line; in this image it is the left-most red
line in the "track" pane.
+ </para>
+ <para>
+ Scrolling the "track" pane
horizontally allows you to view the regions throughout the session.
The left-most point is the start of the session; the right-most point
is after the end of the session. Most DAWs allow you to scroll well
beyond the end of the session. Scrolling vertically in the "track"
pane allows you to view the regions and tracks in a particular time
range.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-DAW_Transport_Controls">
+ <title>Transport Controls</title>
+ <para>
+
[[File:Qtractor-interface-transport.png|300px|Transport Controls]]
+
+ The transport controls allow you to
manipulate the transport in various ways. The shape of the buttons is
somewhat standardized; a similar-looking button will usually perform
the same function in all DAWs, as well as in consumer electronic
devices like CD players and DVD players.
+ </para>
+ <para>
+ The single, left-pointing arrow with a
vertical line will move the transport to the start of the session,
without playing or recording any material. In "Qtractor," if there is
a blue place-marker between the transport and the start of the
session, the transport will skip to the blue place-marker. You can
press the button again if you wish to skip to the next blue
place-marker or the beginning of the session.
+ </para>
+ <para>
+ The double left-pointing arrows move
the transport in fast motion, towards the start of the session. The
double right-pointing arrows move the transport in fast motion,
towards the end of the session.
+ </para>
+ <para>
+ The single, right-pointing arrow with
a vertical line will move the transport to the end of the last region
currently in a session. In "Qtractor," if there is a blue
place-marker between the transport and the end of the last region in
the session, the transport will skip to the blue place-marker. You
can press the button again if you wish to skip to the next blue
place-marker or the end of the last region in the session.
+ </para>
+ <para>
+ The single, right-pointing arrow is
commonly called "play," but it actually moves the transport forward in
real-time. When it does this, if the transport is armed for
recording, any armed tracks will record. Whether or not the transport
is armed, pressing the "play" button causes all un-armed tracks to
play all existing regions.
+ </para>
+ <para>
+ The circular button arms the transport
for recording. It is conventionally red in colour. In "Qtractor,"
the transport can only be armed ''after'' at least one track has been
armed; to show this, the transport's "arm" button only turns red if a
track is armed.
+ </para>
+ </section>
+ </section>
+</chapter>
+
diff --git a/en-US/Musicians_Guide.xml b/en-US/Musicians_Guide.xml
index ac3a328..60ab481 100644
--- a/en-US/Musicians_Guide.xml
+++ b/en-US/Musicians_Guide.xml
@@ -6,7 +6,14 @@
<book status="draft">
<xi:include href="Book_Info.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
<xi:include href="Preface.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
- <xi:include href="Chapter.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+ <!-- Start Chapters -->
+ <xi:include href="Sound_Cards.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+ <xi:include href="Sound_Servers.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+ <xi:include href="Planet_CCRMA_at_Home.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+ <xi:include href="Real_Time_and_Low_Latency.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+
+ <xi:include href="Digital_Audio_Workstations.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
+ <!-- End Chapters -->
<xi:include href="Revision_History.xml"
xmlns:xi="http://www.w3.org/2001/XInclude" />
<index />
</book>
diff --git a/en-US/Planet_CCRMA_at_Home.xml b/en-US/Planet_CCRMA_at_Home.xml
new file mode 100644
index 0000000..18ce3e3
--- /dev/null
+++ b/en-US/Planet_CCRMA_at_Home.xml
@@ -0,0 +1,139 @@
+<?xml version='1.0' encoding='utf-8' ?>
+<!DOCTYPE chapter PUBLIC "-//OASIS//DTD DocBook XML V4.5//EN"
"http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd" [
+<!ENTITY % BOOK_ENTITIES SYSTEM "Musicians_Guide.ent">
+%BOOK_ENTITIES;
+]>
+
+<chapter id="chap-Musicians_Guide-Planet_CCRMA_at_Home">
+ <title>Planet CCRMA at Home</title>
+
+ <section id="sect-Musicians_Guide-What_Is_Planet_CCRMA">
+ <title>What Planet CCRMA at Home Is</title>
+ <para>
+ As stated on the project's home page, it is
the goal of Planet CCRMA at Home to provide packages which will
transform a Fedora Linux-based computer into an audio workstation.
What this means is that, while the Fedora Project does an excellent
job of providing a general-purpose operating system, a general purpose
operating system is insufficient for audio work of the highest
quality. The contributors to Planet CCRMA at Home provide software
packages which can tune your system specifically for audio work.
+ </para>
+ <para>
+ Users of GNU Solfege and LilyPond should not
concern themselves with Planet CCRMA at Home, unless they also user
other audio software. Neither Solfege nor LilyPond would benefit from
a computer optimzed for audio production.
+ </para>
+ <section id="sect-Musicians_Guide-CCRMA">
+ <title>CCRMA</title>
+ <para>
+ CCRMA stands for "Center for Computer
Research in Music and Acoustics," which is the name of an academic
research initiative and music computing facility at Stanford
University, located in Stanford, California. Its initiatives help
scholars to understand the effects and possibilities of computers and
technology in various musical contexts. They offer academic courses,
hold workshops and concerts, and try to incorporate the work of many
highly-specialized fields.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-CCRMA_Software">
+ <title>The Software</title>
+ <para>
+ The Planet CCRMA at Home website
suggests that they provide most of the software used on the computers
in CCRMA's computing facilities. Much of this software is highly
advanced and complex, and not intended for everyday use. More
adventurous users are encouraged to explore Planet CCRMA's website,
and investigate the software for themselves.
+ </para>
+ </section>
+ </section>
+
+ <section
id="sect-Musicians_Guide-Knowing_Whether_to_Use_Planet_CCRMA">
+ <title>Knowing Whether You Should Use Planet CCRMA at
Home</title>
+ <section
id="sect-Musicians_Guide-CCRMA_Need_Exclusive_Software">
+ <title>Do You Need Exclusive Software?</title>
+ <para>
+ The only useful reason to install an
additional repository is if you intend to install and use its
software. The only software application covered in this guide, which
is available exclusively from the Planet CCRMA at Home repository, is
"SuperCollider". The Planet CCRMA repository also offers many other
audio-related software applications, many of which are available from
the default Fedora Project repositories.
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-CCRMA_Updated_Versions">
+ <title>Do You Need Updated Versions?</title>
+ <para>
+ Most of the audio software currently
available in the default Fedora repositories was initially available
in Fedora from the Planet CCRMA at Home repository. Sometimes, an
updated version of an application is available from the Planet CCRMA
repository before it is available from the Fedora Updates repository.
If you need the newer software version, then you should install the
Planet CCRMA repository.
+ </para>
+ <para>
+ This is also a potential security
weakness, for users who install the Planet CCRMA repository, but do
not install any of its software. When "yum" finds a newer version of
an installed application, it will be installed, regardless of the
repository. This may happen to you without you noticing, so that you
begin using Planet CCRMA software without knowing it.
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-CCRMA_Security_and_Stability">
+ <title>Security and Stability with Third-Party
Repositories</title>
+ <para>
+ The biggest reason that you should
avoid installing the Planet CCRMA at Home repository unless you
''need'' its software is security. There are two main security issues
with using the Planet CCRMA repositories:
+ <orderedlist>
+ <listitem><para>Planet CCRMA is
intended for specialized audio workstations. The software is packaged
in such a way that creates potential (and unknown) security threats
caused by the optimizations necessary to prepare a computer system for
use in audio work. Furthermore, these optimizations may reveal
software bugs present in non-Planet CCRMA software, and allow them to
do more damage than on a non-optimized system. Finally, a computer
system's "stability" (its ability to run without trouble) may be
compromised by audio optimizations. Regular desktop applications may
perform less well on audio-optimized systems, if the optimization
process unintentionally un-optimized some other
process.</para></listitem>
+ <listitem><para>CCRMA is not a large,
Linux-focussed organization. It is an academic organization, and its
primary intention with the Planet CCRMA at Home repository is to allow
anybody with a computer to do the same kind of work that they do. The
Fedora Project is a relatively large organization, backed by one of
the world's largest commercial Linux providers, which is focussed on
creating a stable and secure operating system for daily use.
Furthermore, thousands of people around the world are working for the
Fedora Project or its corporate sponsor, and it is their
responsibility to proactively solve problems. CCRMA has the same
responsibility, but they do not have the dedicated resources of the
Fedora Project, it would be naive(???) to think that they would be
capable of providing the same level of support.</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-CCRMA_Best_Practices">
+ <title>A "Best Practices" Solution</title>
+ <para>
+ All Fedora Linux users should be
grateful to the people working at CCRMA, who help to provide the
Planet CCRMA at Home repository. Their work has been instrumental in
allowing Fedora to provide the amount of high-quality audio software
that it does. Furthermore, the availability of many of CCRMA's
highly-specialized software applications through the Planet CCRMA at
Home repository is an invaluable resource to audio and music
enthusiasts.
+ </para>
+ <para>
+ On the other hand, Fedora users cannot
expect that Planet CCRMA software is going to meet the same standards
as Fedora software. While the Fedora Project's primary goal is to
provide Linux software, CCRMA's main goal is to advance the state of
knowledge of computer-based music and audio research and art.
+ </para>
+ <para>
+ Where do these two goals meet?
+ </para>
+ <para>
+ If you want to use your computer for
both day-to-day desktop tasks and high-quality audio production, one
good solution is to "dual-boot" your computer. This involves
installing Fedora Linux twice on the same physical computer, but it
will allow you to keep an entirely separate operating system
environment for the Planet CCRMA at Home software. Not only will this
allow you to safely and securely run Planet CCRMA applications in
their most-optimized state, but you can help to further optimize your
system by turning off and even removing some system services that you
do not need for audio work. For example, a GNOME or KDE user might
choose to install only "Openbox" for their audio-optimized
installation.
+ </para>
+ <para>
+ Alternatively, there is the
possibility of going half-way: installing only some Planet CCRMA
applications, but not the fully-optimized kernel and system
components. This would be more suitable for a computer used most
often for typical day-to-day operations (email, word processing, web
browsing, etc.) If you wanted to use SuperCollider, but did not
require other audio software, for example, then this might be the best
solution for you.
+ </para>
+ <para>
+ Ultimately, it is your responsibility
to ensure that your computer and its data is kept safely and securely.
You will need to find the best solution for your own work patterns
and desires.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-Using_Planet_CCRMA_Software">
+ <title>Using Software from Planet CCRMA at Home</title>
+ <para>
+ The Planet CCRMA at Home software is hosted
(stored) on a server at Stanford University. It is separate from the
Fedora Linux servers, so yum (the command-line utility used by
PackageKit and KPackageKit) must be made aware that you wish to use
it. After installing the repository, Planet CCRMA at Home software
can be installed through yum, PackageKit, or KPackageKit just as
easily as any other software.
+ </para>
+ <section
id="sect-Musicians_Guide-CCRMA_Installing_Repository">
+ <title>Installing the Planet CCRMA at Home
Repositories</title>
+ <para>
+ The following steps will install the
Planet CCRMA at Home repository, intended only for Fedora Linux-based
computers.
+ <orderedlist>
+ <listitem><para>Update your computer
with PackageKit, KPackageKit, or by running <code>su -c 'yum
update'</code> and approving the installation.</para></listitem>
+ <listitem><para>You will have to use a
terminal window for the next portion.</para></listitem>
+ <listitem><para>Run the following
commands: <code>su -c 'rpm -Uvh
http://ccrma.stanford.edu/planetccrma/mirror/fedora/linux/planetccrma/12/...
ADMONITION that this will work for Fedora 12, 13, and 14 and 32-bit
and 64-bit</para></listitem>
+ <listitem><para>Update your computer
again.</para></listitem>
+ <listitem><para>You may receive a
warning that the RPM database was altered outside of "yum". This is
normal.</para></listitem>
+ <listitem><para>Your repository
definition will automatically be updated.</para></listitem>
+ <listitem><para>Some packages are
available from Fedora repositories in addition to other repositories
(like Planet CCRMA at Home). If the Planet CCRMA repository has a
newer version of something than the other repositories that you have
installed, then the Planet CCRMA version will be installed at this
point.</para></listitem>
+ </orderedlist>
+ </para>
+ <para>
+ Although it is necessary to use the
"rpm" program directly, all other Planet CCRMA software can be
installed through "yum", like all other applications. Here is an
explanation of the command-line options used above:
+ <itemizedlist>
+ <listitem><para>-U means "upgrade,"
which will install the specified package, and remove any
previously-installed version</para></listitem>
+ <listitem><para>-v means "verbose,"
which will print additional information meessages</para></listitem>
+ <listitem><para>-h means "hash,"
which
will display hash marks (these: #) showing the progress of
installation.</para></listitem>
+ </itemizedlist>
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-CCRMA_Repository_Priorities">
+ <title>Setting Repository Priorities</title>
+ <para>
+ This is optional, and recommended only
for advanced users. Normally, "yum" will install the latest version
of a package, regardless of which repository provides it. Using this
plugin will change this behaviour, so that yum will choose package
versions primarily based on which repository provides it. If a newer
version is available at a repository with lower priority, yum will not
upgrade the package. If you simply wish to prevent a particular
package from being updated, the instructions in "Preventing LINK LINK"
are better-suited to your needs.
+ <orderedlist>
+ <listitem><para>Install the
"yum-plugin-priorities" package.</para></listitem>
+ <listitem><para>Use a text editor or
the "cat" or "less" command to verify that
<code>/etc/yum/pluginconf.d/priorities.conf</code> exists, and
contains the following text:
+ [pre][main]
+ enabled = 1[/pre]If you want to stop
using the plugin, you can edit this file so that <code>enabled =
0</code>. This allows you to keep the priorities as set in the
repository configuration files.</para></listitem>
+ <listitem><para>You can set priorities
for some or all repositories. To add a priority to a repository, edit
its respective file in the <code>/etc/yum.repos.d/*</code> directory,
adding a line like: [pre]priority = N[/pre]where N is a number from 1
to 99, inclusive. A priority of 1 is the highest setting, and 99 is
the lowest. You will need to set priorities of at least two
repositories before this becomes useful.</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-CCRMA_Preventing_Package_Updates">
+ <title>Preventing a Package from Being
Updated</title>
+ <para>
+ This is optional, and recommended only
for advanced users. Normally, "yum" will install the latest version
of a package. Using this plugin will allow you to prevent certain
packages from being updated. If you wish to prevent packages from a
particular repository from being used, then THIS SECTION is
better-suited to your needs.
+ <orderedlist>
+ <listitem><para>Install the
"yum-plugin-versionlock" package.</para></listitem>
+ <listitem><para>Use a text editor or
the "cat" or "less" command to verify that
<code>/etc/yum/pluginconf.d/versionlock.conf</code> exists, and
contains the following text:
+ [pre]enabled = 1[/pre]</para></listitem>
+ <listitem><para>Add the list of
packages which you do not want to be updated to
<code>/etc/yum/pluginconf.d/versionlock.list</code>. Each package
should go on its own line. For example:
+ [pre]jack-audio-connect-kit-1.9.4
+ qjackctl-0.3.6[/pre]</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+ </section>
+
+</chapter>
\ No newline at end of file
diff --git a/en-US/Real_Time_and_Low_Latency.xml
b/en-US/Real_Time_and_Low_Latency.xml
new file mode 100644
index 0000000..c5ced01
--- /dev/null
+++ b/en-US/Real_Time_and_Low_Latency.xml
@@ -0,0 +1,77 @@
+<?xml version='1.0' encoding='utf-8' ?>
+<!DOCTYPE chapter PUBLIC "-//OASIS//DTD DocBook XML V4.5//EN"
"http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd" [
+<!ENTITY % BOOK_ENTITIES SYSTEM "Musicians_Guide.ent">
+%BOOK_ENTITIES;
+]>
+
+<chapter id="chap-Musicians_Guide-Real_Time_and_Low_Latency">
+ <title>Real-Time and Low Latency</title>
+ <para>
+ It is perhaps a common perception that computers can
compute things instantaneously. Anybody who has ever waited for a web
page to load has first-hand experience that this is not the case:
computers take time to do things, even if the amount of time is often
imperceptible to human observers. Moreover, a computer doing one
thing can seem like it's acting nearly instantaneously, but a computer
doing fifteen things will have a more difficult time keeping up
appearances.
+ </para>
+
+ <section id="sect-Musicians_Guide-Low_Latency">
+ <title>Why Low Latency Is Desirable</title>
+ <para>
+ When computer audio specialists talk about a
computer acting in '''real-time''', they mean that it is acting
with
only an imperceptible delay. A computer cannot act on something
instantaneously, and the amount of waiting time between an input and
its output is called '''latency'''. In order for the delay
between
input and output to be perceived as non-existant (in other words, for
a computer to "react in real-time,") the latency must be low.
+ </para>
+ <para>
+ For periodic tasks, like processing audio
(which has a consistently recurring amount of data per second), low
latency is desirable, but ''consistent'' latency is usually more
important. Think of it like this: years ago in North America, milk
was delivered to homes by a dedicated delivery person. Imagine if the
milk delivery person had a medium-latency, but consistent schedule,
returning every seven days. You would be able to plan for how much
milk to buy, and to limit your intake so that you don't run out too
soon. Now imagine if the milk delivery person had a low-latency, but
inconsistent schedule, returning every one to four days. You would
never be sure how much milk to buy, and you wouldn't know how to limit
yourself. Sometimes there would be too much milk, and sometimes you
would run out. Audio-processing and synthesis software behaves in a
similar way: if it has a consistent amount of latency, it can plan
accordingly. If it has an inconsistent amount of lat
ency - whether large or small - there will sometimes be too much
data, and sometimes not enough. If your application runs out of audio
data, there will be noise or silence in the audio signal - both bad
things.
+ </para>
+ <para>
+ Relatively low latency is still important, so
that your computer reacts imperceptibly quickly to what's going on.
The point is that the difference between an 8 ms target latency
and a 16 ms target latency is almost certainly imperceptible to
humans, but the higher latency may help your computer to be more
consistent - and that's more important.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Processor_Scheduling">
+ <title>Processor Scheduling</title>
+ <para>
+ If you've ever opened the "System Monitor"
application, you will probably have noticed that there are a lot of
"processes" running all the time. Some of these processes need the
processor, and some of them are just waiting around for something to
happen. To help increase the number of processes that can run at the
same time, many modern CPUs have more than one "core," which allows
for more processes to be evaluated at the same time. Even with these
improvements, there are usually more processes than available cores:
my computer right now has 196 processes and only three cores. There
has to be a way of decided which process gets to run and when, and
this task is left to the operating system.
+ </para>
+ <para>
+ In GNU/Linux systems like Fedora Linux, the
core of the operating system (called the '''kernel''') is
responsible
for deciding which process gets to execute at what time. This
responsibility is called "scheduling." Scheduling access to the
processor is called, '''processor scheduling'''. The kernel also
manages scheduling for a number of other things, like memory access,
video hardware access, audio hardware access, hard drive access, and
so on. The algorithm (procedure) used for each of these scheduling
tasks is different for each, and can be changed depending on the
user's needs and the specific hardware being used. In a hard drive,
for example, it makes sense to consider the physical location of data
on a disk before deciding which process gets to read first. For a
processor this is irrelevant, but there are many other things to
consider.
+ </para>
+ <para>
+ There are a number of scheduling algorithms
that are available with the standard Linux kernel, and for most uses,
a "fair queueing" system is appropriate. This helps to ensure that
all processes get an equal amount of time with the processor, and it's
unacceptable for audio work. If you're recording a live concert, and
the "PackageKit" update manager starts, you don't care if PackageKit
gets a fair share of processing time - it's more important that the
audio is recorded as accurately as possible. For that matter, if
you're recording a live concert, and your computer isn't fast enough
to update the monitor, keyboard, and mouse position while providing
uninterrupted, high-quality audio, you want the audio instead of the
monitor, keyboard, and mouse. After all, once you've missed even the
smallest portion of audio, it's gone for good!
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Real_Time_Linux_Kernel">
+ <title>The Real-Time Linux Kernel</title>
+ <para>
+ There is a "real-time patch" for the Linux
kernel which enables the processor to unfairly schedule certain
processes that ask for higher priority. Although the term "patch" may
make it seem like this is just a partial solution, it really refers to
the fact that the programming code used to enable this kind of unfair
scheduling is not included in standard kernels; the standard kernel
code must have this code "patched" into it.
+ </para>
+ <para>
+ The default behaviour of a real-time kernel is
still to use the "fair queueing" system by default. This is good,
because most processes don't need to have consistently low latencies.
Only specific processes are designed to request high-priority
scheduling. Each process is given (or asks for) a priority number,
and the real-time kernel will always give processing time to the
process with the highest priority number, even if that process uses up
''all'' of the available processing time. This puts regular
applications at a disadvantage: when a high-priority process is
running, the rest of the system may be unable to function properly.
In extreme (and very rare!) cases, a real-time process can encounter
an error, use up all the processing time, and disallow any other
process from running - effectively locking you out of your computer.
Security measures have been taken to help ensure this doesn't happen,
but as with anything, there is no guarantee. If you use a real-tim
e kernel, you are exposing yourself to a slightly higher risk of
system crashes.
+ </para>
+ <para>
+ A real-time kernel should not be used on a
computer that acts as a server, for these reasons.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Hard_and_Soft_Real_Time">
+ <title>Hard and Soft Real-Time</title>
+ <para>
+ Finally, there are two different kinds of
real-time scheduling. The Linux kernel, even at its most extreme,
uses only '''soft real-time'''. This means that, while processor
and
other scheduling algorithms may be optimized to give preference to
higher-priority processes, no absolute guarantee of performance can be
made. A real-time kernel helps to greatly reduce the chance of an
audio process running out of data, but sometimes it can still happen.
+ </para>
+ <para>
+ A '''hard real-time''' computer is
designed
for specialized purposes, where even the smallest amount of latency
can make the difference between life and death. These systems are
implemented in hardware as well as software. Example uses include
triggering airbag deployment in automobile crashes, and monitoring the
heart rate of a patient during an operation. These computers are not
particularly multi-functional, which is part of their means to
accomplishing a guaranteed low latency.
+ </para>
+ </section>
+
+ <section
id="sect-Musicians_Guide-Getting_Real_Time_Kernel_in_Fedora">
+ <title>Getting a Real-Time Kernel in Fedora Linux</title>
+ <para>
+ In Fedora Linux, the real-time kernel is
provided by the Planet CCRMA at Home software repositories. Along
with the warnings in the [[User:Crantila/FSC/CCRMA/Everything|Planet
CCRMA section]], here is one more to consider: the real-time kernel is
used by fewer people than the standard kernel, so it is less
well-tested. The changes of something going wrong are relatively low,
but be aware that using a real-time kernel increases the level of
risk. Always leave a non-real-time option available, in case the
real-time kernel stops working.
+ </para>
+ <para>
+ You can install the real-time kernel, along
with other system optimizations, by following these instructions:
+ <orderedlist>
+ <listitem><para>Install the Planet CCRMA at
Home repositories by following
[[User:Crantila/FSC/CCRMA/Everything#Using Planet CCRMA at Home
Software|these instructions]].</para></listitem>
+ <listitem><para>Run the following command in a
terminal: [pre]su -c 'yum install planetccrma-core'[/pre] Note that
this is a meta-package, which does not install anything by itself, but
causes a number of other packages to be installed, which will
themselves perform the desired installation and
optimization.</para></listitem>
+ <listitem><para>Shut down and reboot your
computer, to test the new kernel. If you decided to modify your GRUB
configuration, be sure that you leave a non-real-time kernel available
for use.</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+
+</chapter>
diff --git a/en-US/Sound_Cards.xml b/en-US/Sound_Cards.xml
new file mode 100644
index 0000000..e8fe2c0
--- /dev/null
+++ b/en-US/Sound_Cards.xml
@@ -0,0 +1,129 @@
+<?xml version='1.0' encoding='utf-8' ?>
+<!DOCTYPE chapter PUBLIC "-//OASIS//DTD DocBook XML V4.5//EN"
"http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd" [
+<!ENTITY % BOOK_ENTITIES SYSTEM "Musicians_Guide.ent">
+%BOOK_ENTITIES;
+]>
+
+<chapter id="chap-Musicians_Guide-Sound_Cards">
+ <title>Sound Cards and Digital Audio</title>
+ <para>
+ Everybody has a vague idea of what sound cards are,
how they work, and what they do. Sometimes, especially when doing
professional-quality audio work, a vague understanding is no longer
sufficient. This chapter introduces the technical vocabulary used
when discussing computer audio hardware.
+ </para>
+
+ <section id="sect-Musicians_Guide-What_Sound_Cards_Are">
+ <title>Defining Sound Cards</title>
+ <para>
+ Broadly defined, a sound card is any
computer-connected device which allows the computer to process audio
in some way. There are two general categories into which most sound
cards fit, described below.
+ </para>
+ <section id="sect-Musicians_Guide-Audio_Interfaces">
+ <title>Audio Interfaces</title>
+ <para>
+ This is a hardware device that allows
audio equipment to be connected to your computer, including
microphones and speakers. Typically audio entering or leaving an
audio interface from/to an external device requires conversion between
digital and analogue formats. However, with the rise of external
digital audio equipment, there are an increasing number of devices
that connect digitally to an audio interface.
+ </para>
+ <para>
+ The conversion between analogue and
digital signals is a prerequisite for computers to be able to process
audio signals, so it is the primary function of audio interfaces. The
real world creates sound with a limitless range of possibilities for
pitch, volume, and duration. The digital nature of computers requires
these limitless possibilities to be reduced to finite limits. The
best digital/analogue converters are capable of using these limits in
such a way that humans don't notice anything missing - much like the
best computer monitors and graphics adapters are able to disguise the
fact that only about half of the colours our eyes can see are
display-able on computers. This problem is discussed further in the
"Bit Rates and Sample Rates" section.
+ </para>
+ <para>
+ Audio interfaces also amplify signals
for directly-connected analogue devices (like headphones). Some offer
power for microphones, too (pre-amplification and/or phantom power).
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-MIDI_Interfaces">
+ <title>MIDI Interfaces</title>
+ <para>
+ MIDI stands for "Musical Instrument
Digital Interface," and is commonly associated with low-quality
imitations of acoustic instruments. This association is unfortunate,
since high-quality audio is indeed possible with MIDI, and MIDI-driven
devices have played a part in many mainstream and non-mainstream audio
environments. Whereas audio signals specify the sounds themselves,
MIDI signals contain instructions on how to make the sounds. It is a
synthesizer's responsibility to follow these instructions, turning
them into sounds. Going even further, the MIDI specification allows
for the control of many audio-related devices, like mixers,
sequencers, and Digital Audio Workstations. Although the signals used
to control these devices (or software applications) do not directly
cause the generation of music, they still follow the definition of
"MIDI signals": instructions on how to make sounds.
+ </para>
+ <para>
+ Whereas audio interfaces allow the
input and output of audio signals ("normal sound") from a computer,
MIDI interfaces allow the input and output of MIDI signals. Some
audio interfaces have MIDI capabilities built-in, and some MIDI
interfaces also transform MIDI signals into audio signals. The latter
kind of device is performing "MIDI synthesis," a task for which there
exist many software-only solutions. "FluidSynth," covered in
[[User:Crantila/FSC/Synthesizers/FluidSynth|this section]] of the
Musicians' Guide, is one such software solution.
+ </para>
+ <para>
+ Having a hardware-based MIDI interface
is not a requirement for working with MIDI signals and applications.
The costly nature of most MIDI hardware makes it impractical for
occasional or beginning MIDI users and computer music enthusiasts.
Much of the software in this Guide is capable of working with MIDI
signals, and supports but does not require MIDI-capable hardware.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-Sound_Card_Connections">
+ <title>Methods of Connection</title>
+ <para>
+ The following connection methods can be used
by either audio or MIDI interfaces, so they are collectively referred
to as "sound cards," in this section.
+ </para>
+
+ <section
id="sect-Musicians_Guide-Motherboard_Integrated">
+ <title>Integrated into the Motherboard</title>
+ <para>
+ These sound cards are built into the
computer's motherboard. In recent years, the quality of audio
produced by these sound cards has greatly increased, but the best
integrated solutions are still not as good as the best non-integrated
solutions. Good integrated sound cards should be good enough for most
audio work; if you want a professional-sounding sound card, or
especially if you want to connect high-quality input devices, then an
additional sound card is recommended.
+ </para>
+ <para>
+ Hardware MIDI interfaces are rarely,
if ever, integrated into the motherboard.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-PCI_Sound_Cards">
+ <title>Internal PCI Connection</title>
+ <para>
+ Sound cards connected to the
motherboard by PCI (or PCI-Express, etc.) will probably offer higher
performance, and lower latencies, than USB- or FireWire-connected
devices. Professional-quality sound cards often have insufficient
space for connectors on the card itself, so they often include a
proprietary, external component specifically for adding connectors.
The biggest disadvantage of PCI-connected sound cards is that they
cannot be used with notebooks or netbooks, and that they are only as
portable as the computer in which they're installed.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-USB_Sound_Cards">
+ <title>External USB Connection</title>
+ <para>
+ USB-connected sound cards are becoming
more popular, especially with the increasing bandwidth possibilities
of USB connections. The quality can be as good as
internally-connected sound cards, although the USB connection may add
additional latency, which may or may not be a concern. The biggest
advantages of USB-connected sound cards is that they can be used with
notebooks and netbooks, and that they are usually easier to transport
than an entire desktop computer.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-FireWire_Sound_Cards">
+ <title>External FireWire Connection</title>
+ <para>
+ FireWire-connected sound cards are not
as popular as USB sound cards, but they tend to be of higher quality.
In addition, unlike USB-connected sound cards, FireWire-connected
sound cards are able to take advantage of FireWire's "guaranteed
bandwidth" and "bus-mastering" capabilities. Having guaranteed
bandwidth ensures that the sound card will be able to send data when
it chooses; the sound card will not have to compete with other devices
connected with the same connection type. Using bus-mastering enables
the FireWire-connected device to read and write directly to and from
the computer's main memory, without first going through the CPU.
High-speed FireWire connections are also available on older computers
where a USB 2.0 connection is not available.
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-Choose_Sound_Card_Connection">
+ <title>Choosing a Connection Type</title>
+ <para>
+ The method of connection should not by
itself determine which sound card is appropriate for you. Which
connection type is right for you will depend on a wide range of
factors, but the actual sound quality is significantly more important
than the theoretical advantages or disadvantages of the connection
type. If possible, you should try out potential devices with your
computer before you buy one.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-Bit_Rates_and_Sample_Rates">
+ <title>Bit Rates and Sample Rates</title>
+ <para>
+ As mentioned in the !!Audio Interface
section!!, the primary job of audio interfaces is to carry out the
transformation of audio signals between digital and analogue forms.
This diagram from Wikipedia illustrates the "digital problem," when it
comes to audio: [http://en.wikipedia.org/wiki/File:Pcm.svg here].
Although the wave-shape of the analogue signal, which is what is
produced by most acoustic instruments and by the human voice, is shown
in red, computers cannot store that information. Instead, they
usually store some approximation, which is represented in that diagram
by the gray, shaded area. Note that the diagram is simply an example,
and not meant to depict a particular real-world recording.
+ </para>
+ <para>
+ It is the conversion between digital and
analogue signals that distinguishes low- and high-quality audio
interfaces. High-quality convertors will be able to record and
reproduce a signal that is nearly identical to the original. Bit and
sample rates are tied to the closeness of approximation that an audio
interface can make, and they are explained below. There are other
factors involved in overall sound quality.
+ </para>
+ <section id="sect-Musicians_Guide-Bit_Rate">
+ <title>Bit Rate (Sample Format)</title>
+ <para>
+ This is the number of bits used to
describe the audio in a length of time. The higher the number of
bits, the greater the detail that will be stored. For most uses, the
bit-rate is usually measured in "bits per second," as in the
often-used 128 kb/s bit-rate for MP3 audio. Professional audio
is more often referred to as "bits per sample," which is usually
simply called "bits." CDs have a 16 bit/sample bit-rate,
professional audio is usually recorded at a 24 bit/sample
bit-rate, and a 32 bit/sample bit-rate is supported by some
hardware and software, but not widely used. Due to technical
limitations, 20-bit audio is also widely used. See Wikipedia for more
information (get a link??)
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Sample_Rate">
+ <title>Sample Rate</title>
+ <para>
+ A sample is a collection of a number
of bits, representing a sound at an instantaneous point in time. The
number of bits contained in a sample is determined by the bit-rate
(usually 16 or 24 bits per sample). The sample rate is a measure of
how many samples occupy one second - that is, how many "instants" of
sound are catalogued for each second. Theoretically, a higher sample
rate results in a higher-quality audio signal. The sample rate is
measured in Hertz, which means "samples per second." CDs have a
44 100 Hz sample rate, but audio is often recorded at
48 000 Hz, 96 000 Hz, or even
192 000 Hz. These are often indicated as 44.1 kHz,
48 kHz, 96 kHz, and 192 kHz, respectively.
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-Bit_and_Sample_Rate_Conclusions">
+ <title>Conclusions</title>
+ <para>
+ Both of these factors have an impact
on potential sound quality. Depending on the limitations and
capabilities of your equipment, you may be more inclined to use
particular settings than others. Here are some comparisons:
+ <itemizedlist>
+ <listitem><para>16-bit bit rate, and
44.1 kHz sample rate (CD audio; good for wide distribution and
maximum compatibility; 705.6 kb/s)</para></listitem>
+ <listitem><para>24-bit bit rate, and
96 kHz sample rate (CDs are usually recorded at these rates, then
"down-mixed" later; 2304 kb/s)</para></listitem>
+ <listitem><para>24-bit bit rate, and
192 kHz sample rate (DVD Audio; not widely compatible;
4608 kb/s)</para></listitem>
+ <listitem><para>1-bit bit rate, and
2822.4 kHz sample rate (Super Audio CD; not widely compatible;
2822.4 kb/s)</para></listitem>
+ </itemizedlist>
+ </para>
+ <para>
+ In the end, bit rate and sample rate
are only part of what determines overall sound quality. Moreover,
sound quality is subjective, and you will need to experiment to find
the equipment and rates that work best for what you do.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-Sound_Cards_Audio_Vocabular">
+ <title>Audio Vocabulary</title>
+ <para>
+ This part will transclude the "audio vocabulary" file.
+ </para>
+ </section>
+
+</chapter>
\ No newline at end of file
diff --git a/en-US/Sound_Servers.xml b/en-US/Sound_Servers.xml
new file mode 100644
index 0000000..d8b6b74
--- /dev/null
+++ b/en-US/Sound_Servers.xml
@@ -0,0 +1,141 @@
+<?xml version='1.0' encoding='utf-8' ?>
+<!DOCTYPE chapter PUBLIC "-//OASIS//DTD DocBook XML V4.5//EN"
"http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd" [
+<!ENTITY % BOOK_ENTITIES SYSTEM "Musicians_Guide.ent">
+%BOOK_ENTITIES;
+]>
+
+<chapter id="chap-Musicians_Guide-How_Computers_Deal_with_Hardware">
+ <title>How Computers Deal with Audio and MIDI Hardware</title>
+ <para>
+ One of the techniques consistently used in computer
science is abstraction. Abstraction is the process of creating a
generic model for something (or some things) that are actually unique.
The "driver" for a hardware device in a computer is one form of
dealing with abstraction: the computer's software interacts with all
sound cards in a similar way, and it is the driver which translates
the universal instructions given by the software into specific
instructions for operating that hardware device. Consider this
real-world comparison: you know how to operate doors because of
abstracted instructions. You don't know how to open and close every
door that exists, but from the ones that you do know how to operate,
your brain automatically creates abstracted instructions, like "turn
the handle," and "push the door," which apply with all or most doors.
When you see a new door, you have certain expectations about how it
works, based on the abstract behaviour of doors, and you qu
ickly figure out how to operate that specific door with a simple
visual inspection. The principle is the same with computer hardware
drivers: since the computer already knows how to operate "sound
cards," it just needs a few simple instructions (the driver) in order
to know how to operate any particular sound card.
+ </para>
+
+ <section id="sect-Musicians_Guide-Sound_Servers-ALSA">
+ <title>How Linux Deals with Audio: ALSA</title>
+ <para>
+ In Linux, the core of the operating system
provides hardware drivers for most audio hardware. The hardware
drivers, and the instructions that other software can use to connect
to those drivers, are collectively called "ALSA," which stands for
"Advanced Linux Sound Architecture." ALSA is the most direct way that
software applications can interact with audio and MIDI hardware, and
it used to be the most common way. However, in order to include all
of the features that a software application might want to use, ALSA is
quite complex, and can be error-prone. For this and many other
reasons, another level of abstraction is normally used, and this makes
it easier for software applications to take advantage of the features
they need.
+ </para>
+ </section>
+
+ <section id="sect-Musicians_Guide-Sound_Servers_Section">
+ <title>Sound Servers</title>
+ <para>
+ Sound servers are software applications that
run "in the background," meaning they are rarely seen by users. They
are used to provide another level of abstraction - essentially to
automatically take care of certain aspects of using ALSA, thereby
making it easier for software applications to use the audio hardware.
The three sound servers discussed in this guide have distinctly
different goals, provide distinctly different features and
capabilities, and should not be viewed as though one is universally
better than the others.
+ </para>
+ <section
id="sect-Musicians_Guide-Sound_Servers-PulseAudio">
+ <title>PulseAudio</title>
+ <para>
+ PulseAudio is an advanced sound
server, intended to make audio programming in GNU/Linux operating
systems as easy as possible. The idea behind its design is that an
audio application needs only to output audio to PulseAudio, and
PulseAudio will take care of the rest: choosing and controlling a
particular device, adjusting the volume, working with other
applications, and so on. PulseAudio even has the ability to use
"networked sound," which allows two computers using PulseAudio to
communicate as though they were one computer - either computer can
input from or output to either computer's audio hardware just as
easily as its own audio hardware. This is all controlled within
PulseAudio, so no further complication is added to the software.
+ </para>
+ <para>
+ The Fedora Project's integration of
PulseAudio as a vital part of the operating system has helped to
ensure that audio applications can "just work" for most people under
most circumstances. This has made it much easier for users to carry
out basic audio tasks.
+ </para>
+ <!--
+ Fernando Lopez-Lezcano: I don't think that at
this point applications can be written for the pulse audio API.
Lennart I think discourages it. AFAIK applications still use the ALSA
API when talking to pulse audio, it is just that pulse is in the
middle, and can manage connections from many applications that
basically share the audio hardware (which is only managed by PA). So
what is made easier is not "audio programming" but perhaps "shared
access to the sound card". You would not want to program for PA (it is
very complex, perhaps even more than ALSA as it is all asynchronous
and callback based).
+
+ Christopher Antila: This is useful and
interesting information, but I don't think it belongs here. I''m
keeping it as a comment, in case somebody decides to use it in the
future.
+ -->
+ </section>
+ <section id="sect-Musicians_Guide-Sound_Servers-JACK">
+ <title>JACK Audio Connection Kit</title>
+ <para>
+ The JACK sound server offers fewer
features than other sound servers, but they are tailor-made to allow
the functionality required by audio creation applications. JACK also
makes it easier for users to configure the options that are most
important for such situations. The server supports only one sample
rate and format at a time, and allows applications and hardware to
easily [[User:Crantila/FSC/Sound_Cards#Routing_and_Multiplexing|connect
and multiplex]] in ways that other sound servers do not. It is also
optimized to run with consistently low latencies. Although using JACK
requires a better understanding of the underlying hardware, the
"QjackCtl" application provides a graphical user interface to ease the
process.
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Sound_Servers-Phonon">
+ <title>Phonon</title>
+ <para>
+ Phonon is a sound server built into
the KDE Software Compilation, and is one of the core components of
KDE. By default on Fedora Linux, Phonon feeds output to PulseAudio,
but on other platforms (like Mac OS X, Windows, other versions of
Linux, FreeBSD, and any other system that supports KDE), Phonon can be
configured to feed its output anywhere. This is its greatest strength
- that KDE applications like Amarok and Dragon Player need only be
programmed to use Phonon, and they can rely on Phonon to take care of
everything else. As KDE applications increasingly find their place in
Windows and especially Mac OS X, this cross-platform capability is
turning out to be very useful.
+ </para>
+ </section>
+ </section>
+
+ <section id="sect-Musicians_Guide-Using_JACK">
+ <title>Using the JACK Audio Connection Kit</title>
+ <para>
+ !! What to say here depends on whether jack2
will be available with Fedora 14. If it is, no need for CCRMA
solution. If it isn't, need for CCRMA solution. !!
+ </para>
+ <section
id="sect-Musicians_Guide-Install_and_Configure_JACK">
+ <title>Installing and Configuring JACK</title>
+ <para>
+ <orderedlist>
+ <listitem><para>Ensure that you have
installed the Planet CCRMA at Home repositories. For instructions,
refer to [[User:Crantila/FSC/CCRMA/Everything#Installing_the_Repository|this
section]].</para></listitem>
+ <listitem><para>Use PackageKit or
KPackageKit to install the "jack-audio-connection-kit" and "qjackctl"
packages, or run the following command in a terminal: [pre]sudo -c
'yum install jack-audio-connection-kit
qjackctl'[/pre]</para></listitem>
+ <listitem><para>Review and approve the
installation, making sure that it completes
correctly.</para></listitem>
+ <listitem><para>Run QjackCtl from the
KMenu or the Applications menu.</para></listitem>
+ <listitem><para>To start the JACK
server, press the 'Start' button; to stop it, press the 'Stop'
button.</para></listitem>
+ <listitem><para>Use the 'Messages'
button to see messages, which are usually errors or
warnings.</para></listitem>
+ <listitem><para>Use the 'Status'
button to see various statistics about the currently-running
server.</para></listitem>
+ <listitem><para>Use the 'Connections'
button to see and adjust the connections between applications and
audio hardware.</para></listitem>
+ </orderedlist>
+ </para>
+ <para>
+ JACK will operate without following
this procedure, but users are strongly encouraged to follow these
three steps, for security reasons. They will help to allow optimal
performance of the JACK sound server, while greatly reducing the risk
that an application or user will accidentally or malicious take
advantage of the capability.
+ <orderedlist>
+ <listitem><para>Add all of the users
who will use JACK to the "audio" group. For help with this, see the
!!Fedora 14 Deployment Guide, Chapter 22!!</para></listitem>
+ <listitem><para>The default
installation automatically enables real-time priority to be requested
by any user or process. This is undesirable, so we will edit it.
+ <orderedlist>
+ <listitem><para>Open a
terminal, and run the following command: [pre]sudo -c 'gedit
/etc/security/limits.conf'[/pre]</para></listitem>
+ <listitem><para>Be careful!
You're editing this important system file as the root
user!</para></listitem>
+ <listitem><para>Edit the last
lines, so that they read:[pre]
+ @audio - rtprio 99
+ @audio - memlock 4194304
+ @audio - nice
-10[/pre]</para></listitem>
+ </orderedlist></para></listitem>
+ </orderedlist>
+ </para>
+ <para>
+ With the default configuration of
QjackCtl, it chooses the "default" sound card, which actually goes
through the ALSA sound server. We can avoid this, and use the ALSA
drivers without the sound server, which will help JACK to maintain
accurately low latencies. The following procedure configures JACK to
connect to the ALSA driver directly.
+ <orderedlist>
+ <listitem><para>In a terminal, run the
following command: [pre]cat /proc/asound/cards[/pre]</para></listitem>
+ <listitem><para>The command will
output a list of sound cards in your system, similar to this:
+ [pre]
+0 [SB ]: HDA-Intel - HDA ATI SB
+ HDA ATI SB at 0xf7ff4000 irq 16
+1 [MobilePre ]: USB-Audio - MobilePre
+ M Audio MobilePre at usb-0000:00:13.0-2
+[/pre]
+ The left-most number is the sound
card's number. The portion in square brackets is the name (these
cards are called "SB" and "MobilePre").</para></listitem>
+ <listitem><para>Identify your desired
sound card in the list. If it's not in the list, then it's not
currently detected and configured by your system.</para></listitem>
+ <listitem><para>Open the "QjackCtl"
application, and press the 'Setup' button.</para></listitem>
+ <listitem><para>In the "Interface"
field, input the name of your preferred sound card with "hw:" in
front. For example, with the sound card information posted above, you
might write <code>hw:MobilePre</code></para></listitem>
+ <listitem><para>To save your settings,
close "QjackCtl".</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+ <section id="sect-Musicians_Guide-Using_QjackCtl">
+ <title>Using QjackCtl</title>
+ <para>
+ The QjackCtl application offers many
more features and configuration options. The patch bay is a notable
feature, which lets users save configurations of the "Connections"
window, and restore them later, to help avoid the lengthy set-up time
that might be required in complicated routing and multiplexing
situations.
+ </para>
+ <para>
+ For more information on QjackCtl, you
can refer to [http://www.64studio.com/manual/audio/jack this] web
site.
+ </para>
+ </section>
+ <section
id="sect-Musicians_Guide-Integrating_PulseAudio_with_JACK">
+ <title>Integrating PulseAudio with JACK</title>
+ <para>
+ The default configuration of
PulseAudio yields control of the audio equipment to JACK when the JACK
server starts. PulseAudio will not be able to receive input or send
output of any audio signals on the audio interface used by JACK. This
is fine for occasional users of JACK, but many users will want to use
JACK and PulseAudio simultaneously, or switch between the two
frequently. The following instructions will configure PulseAudio so
that its input and output is routed through JACK.
+ <orderedlist>
+ <listitem><para>Use PackageKit or
KPackageKit to install the "pulseaudio-module-jack" package, or in a
terminal run the following command: [pre]su -c 'yum install
pulseaudio-module-jack'[/pre]</para></listitem>
+ <listitem><para>Approve the
installation and ensure that it is carried out
properly.</para></listitem>
+ <listitem><para>You'll need to edit
the PulseAudio configuration file to use the JACK module.
+ <orderedlist>
+ <listitem><para>Be careful!
You will be editing an important system file as the root
user!</para></listitem>
+ <listitem><para>Run the
following command in a terminal: [pre]sudo -c 'gedit
/etc/pulse/default.pa'[/pre]</para></listitem>
+ <listitem><para>Add the
following lines, underneath the line that says [code]#load-module
module-alsa-sink[/code]:
+ [pre]load-module module-jack-sink
+ load-module
module-jack-source[/pre]</para></listitem>
+
</orderedlist></para></listitem>
+ <listitem><para>Restart PulseAudio by
running the following command in a terminal: [pre]killall
pulseaudio[/pre] It will start again automatically.</para></listitem>
+ <listitem><para>Confirm that this has
worked by opening QjackCtl. The display should confirm that JACK is
"Active".</para></listitem>
+ <listitem><para>In the "Connect"
window, on the "Audio" tab, there should be PulseAudio devices on each
side, and they should be connected to "system" devices on the opposite
sides.</para></listitem>
+ <listitem><para>Open QjackCtl's
"Setup" window, then click on the "Options" tab. Uncheck
"Execute
script after Shutdown: killall jackd". If you did not make this
change, then QjackCtl would stop the JACK server from running every
time the program quits. Since PulseAudio is still expecting to use
JACK after that, you shouldn't do this any more.</para></listitem>
+ <listitem><para>When PulseAudio starts
JACK, it uses the command found in the <code>~/.jackdrc</code> file.
QjackCtl automatically updates this file when you change settings, but
you may have to restart both PulseAudio and JACK in order to get the
new changes to take effect. If they refuse to take effect, you can
edit that file yourself.</para></listitem>
+ <listitem><para>Be careful about using
a very high sample rate with PulseAudio, since it will tend to use a
lot of CPU power.</para></listitem>
+ </orderedlist>
+ </para>
+ </section>
+ </section>
+
+</chapter>
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