tdemultimedia/doc/artsbuilder/detail.docbook

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<chapter id="arts-in-detail">
<title>&arts; in Detail</title>

<sect1 id="architecture">
<title>Architecture</title>

<mediaobject>
<imageobject>
<imagedata fileref="arts-structure.png" format="PNG"/>
</imageobject>
<textobject><phrase>The &arts; structure.</phrase></textobject>
</mediaobject>
</sect1>

<sect1 id="modules-ports">
<title>Modules &amp; Ports</title>

<para>
The idea of &arts; is, that synthesis can be done using small modules,
which only do one thing, and then recombine them in complex
structures. The small modules normally have inputs, where they can get
some signals or parameters, and outputs, where they produce some
signals.
</para>

<para>
One module (Synth&lowbar;ADD) for instance just takes the two signals at
it's input and adds them together. The result is available as output
signal. The places where modules provide their input/output signals are
called ports.
</para>

</sect1>

<sect1 id="structures">
<title>Structures</title>

<para>
A structure is a combination of connected modules, some of which may
have parameters coded directly to their input ports, others which may be
connected, and others, which are not connected at all.
</para>

<para>
What you can do with &arts-builder; is describing structures. You
describe, which modules you want to be connected with which other
modules. When you are done, you can save that structure description to a
file, or tell &arts; to create such a structure you described (Execute).
</para>

<para>
Then you'll probably hear some sound, if you did everything the right
way.
</para>
</sect1>

<!-- TODO

<sect1 id="streams">
<title>Streams</title>
<para>
</para>
</sect1>

-->

<sect1 id="latency">
<title>Latency</title>

<sect2 id="what-islatency">
<title>What Is Latency?</title>

<para>
Suppose you have an application called <quote>mousepling</quote>(that
should make a <quote>pling</quote> sound if you click on a button. The
latency is the time between your finger clicking the mouse button and
you hearing the pling. The latency in this setup composes itself out of
certain latencies, that have different causes.
</para>

</sect2>

<sect2 id="latenbcy-simple">
<title>Latency in Simple Applications</title>

<para>
In this simple application, latency occurs at these places:
</para>

<itemizedlist>

<listitem>
<para>
The time until the kernel has notified the X11 server that a mouse
button was pressed.
</para>
</listitem>

<listitem>
<para>
The time until the X11 server has notified your application that a mouse
button was pressed.
</para>
</listitem>

<listitem>
<para>
The time until the mousepling application has decided that this button
is worth playing a pling.
</para>
</listitem>

<listitem>
<para>
The time it takes the mousepling application to tell the soundserver
that it should play a pling.
</para>
</listitem>

<listitem>
<para>
The time it takes for the pling (which the soundserver starts mixing to
the other output at once) to go through the buffered data, until it
really reaches the position where the soundcard plays.
</para>
</listitem>

<listitem>
<para>
The time it takes the pling sound from the speakers to reach your ear.
</para>
</listitem>
</itemizedlist>

<para>
The first three items are latencies external to &arts;.  They are
interesting, but beyond the scope of this document.  Nevertheless be
aware that they exist, so that even if you have optimized everything
else to really low values, you may not necessarily get exactly the
result you calculated.
</para>

<para>
Telling the server to play something involves usually one single &MCOP;
call.  There are benchmarks which confirm that, on the same host with
unix domain sockets, telling the server to play something can be done
about 9000 times in one second with the current implementation. I expect
that most of this is kernel overhead, switching from one application to
another. Of course this value changes with the exact type of the
parameters. If you transfer a whole image with one call, it will be
slower than if you transfer only one long value. For the returncode the
same is true. However for ordinary strings (such as the filename of the
<literal role="extension">wav</literal> file to play) this shouldn't be
a problem.
</para>

<para>
That means, we can approximate this time with 1/9000 sec, that is below
0.15 ms. We'll see that this is not relevant.
</para>

<para>
Next is the time between the server starting playing and the soundcard
getting something.  The server needs to do buffering, so that when other
applications are running, such as your X11 server or
<quote>mousepling</quote> application no dropouts are heard. The way
this is done under &Linux; is that there are a number fragments of a
size. The server will refill fragments, and the soundcard will play
fragments.
</para>

<para>
So suppose there are three fragments. The server refills the first, the
soundcard starts playing it. The server refills the second. The server
refills the third. The server is done, other applications can do
something now.
</para>

<para>
As the soundcard has played the first fragment, it starts playing the
second and the server starts refilling the first. And so on.
</para>

<para>
The maximum latency you get with all that is (number of fragments)*(size
of each fragment)/(samplingrate * (size of each sample)). Suppose we
assume 44kHz stereo, and 7 fragments a 1024 bytes (the current aRts
defaults), we get 40 ms.
</para>

<para>
These values can be tuned according to your needs. However, the
<acronym>CPU</acronym> usage increases with smaller latencies, as the
sound server needs to refill the buffers more often, and in smaller
parts. It is also mostly impossible to reach better values without
giving the soundserver realtime priority, as otherwise you'll often get
drop-outs.
</para>

<para>
However, it is realistic to do something like 3 fragments with 256 bytes
each, which would make this value 4.4 ms. With 4.4ms delay the idle
<acronym>CPU</acronym> usage of &arts; would be about 7.5%. With 40ms delay, it would be
about 3% (of a PII-350, and this value may depend on your soundcard,
kernel version and others).
</para>

<para>
Then there is the time it takes the pling sound to get from the speakers
to your ear.  Suppose your distance from the speakers is 2 meters. Sound
travels at a speed of 330 meters per second. So we can approximate this
time with 6 ms.
</para>

</sect2>

<sect2 id="latency-streaming">
<title>Latency in Streaming Applications</title>

<para>
Streaming applications are those that produce their sound themselves.
Assume a game, which outputs a constant stream of samples, and should
now be adapted to replay things via &arts;. To have an example: when I
press a key, the figure which I am playing jumps, and a boing sound is
played.
</para>

<para>
First of all, you need to know how &arts; does streaming. Its very
similar to the I/O with the soundcard. The game sends some packets with
samples to the sound server. Lets say three packets. As soon as the
sound server is done with the first packet, it sends a confirmation back
to the game that this packet is done.
</para>

<para>
The game creates another packet of sound and sends it to the server.
Meanwhile the server starts consuming the second sound packet, and so
on.  The latency here looks similar like in the simple case:
</para>

<itemizedlist>
<listitem>
<para>
The time until the kernel has notified the X11 server that a key was
pressed.
</para>
</listitem>

<listitem>
<para>
The time until the X11 server has notified the game that a key was
pressed.
</para>
</listitem>

<listitem>
<para>
The time until the game has decided that this key is worth playing a
boing.
</para>
</listitem>

<listitem>
<para>
The time until the packet of sound in which the game has started putting
the boing sound is reaching the sound server.
</para>
</listitem>

<listitem>
<para>
The time it takes for the boing (which the soundserver starts mixing to
the other output at once) to go through the buffered data, until it
really reaches the position where the soundcard plays.
</para>
</listitem>

<listitem>
<para>
The time it takes the boing sound from the speakers to
reach your ear.
</para>
</listitem>

</itemizedlist>

<para>
The external latencies, as above, are beyond the scope of this document.
</para>

<para>
Obviously, the streaming latency depends on the time it takes all
packets that are used for streaming to be played once. So it is (number
of packets)*(size of each packet)/(samplingrate * (size of each sample))
</para>

<para>
As you see that is the same formula as applies for the
fragments. However for games, it makes no sense to do such small delays
as above.  I'd say a realistic configuration for games would be 2048
bytes per packet, use 3 packets. The resulting latency would be 35ms.
</para>

<para>
This is based on the following: assume that the game renders 25 frames
per second (for the display). It is probably safe to assume that you
won't notice a difference of sound output of one frame. Thus 1/25 second
delay for streaming is acceptable, which in turn means 40ms would be
okay.
</para>

<para>
Most people will also not run their games with realtime priority, and
the danger of drop-outs in the sound is not to be neglected. Streaming
with 3 packets a 256 bytes is possible (I tried that) - but causes a lot
of <acronym>CPU</acronym> usage for streaming.
</para>

<para>
For server side latencies, you can calculate these exactly as above.
</para>

</sect2>

<sect2 id="cpu-usage">
<title>Some <acronym>CPU</acronym> usage considerations</title>

<para>
There are a lot of factors which influence _<acronym>CPU</acronym> usage
in a complex scenario, with some streaming applications and some others,
some plugins on the server etc. To name a few:
</para>

<itemizedlist>
<listitem>
<para>
Raw <acronym>CPU</acronym> usage by the calculations necessary.
</para>
</listitem>

<listitem>
<para>
&arts; internal scheduling overhead - how &arts; decides when which
module should calculate what.
</para>
</listitem>

<listitem>
<para>
Integer to float conversion overhead.
</para>
</listitem>

<listitem>
<para>
&MCOP;0 protocol overhead.
</para>
</listitem>

<listitem>
<para>
Kernel: process/context switching.
</para>
</listitem>

<listitem>
<para>
Kernel: communication overhead
</para>
</listitem>
</itemizedlist>

<para>
For raw <acronym>CPU</acronym> usage for calculations, if you play two
streams, simultaneously you need to do additions. If you apply a filter,
some calculations are involved. To have a simplified example, adding two
streams involves maybe four <acronym>CPU</acronym> cycles per addition,
on a 350Mhz processor, this is 44100*2*4/350000000 = 0.1%
<acronym>CPU</acronym> usage.
</para>

<para>
&arts; internal scheduling: &arts; needs to decide which plugin when
calculates what. This takes time.  Take a profiler if you are interested
in that. Generally what can be said is: the less realtime you do
(&ie;. the larger blocks can be calculated at a time) the less
scheduling overhead you have. Above calculating blocks of 128 samples at
a time (thus using fragment sizes of 512 bytes) the scheduling overhead
is probably not worth thinking about it.
</para>

<para>
Integer to float conversion overhead: &arts; uses floats internally as
data format. These are easy to handle and on recent processors not
slower than integer operations. However, if there are clients which play
data which is not float (like a game that should do its sound output via
&arts;), it needs to be converted. The same applies if you want to
replay the sounds on your soundcard. The soundcard wants integers, so
you need to convert.
</para>

<para>
Here are numbers for a Celeron, approx. ticks per sample, with -O2 +egcs
2.91.66 (taken by Eugene Smith <email>hamster@null.ru</email>). This is
of course highly processor dependant:
</para>

<programlisting>
convert_mono_8_float: 14
convert_stereo_i8_2float: 28
convert_mono_16le_float: 40
interpolate_mono_16le_float: 200
convert_stereo_i16le_2float: 80
convert_mono_float_16le: 80
</programlisting>

<para>
So that means 1% <acronym>CPU</acronym> usage for conversion and 5% for
interpolation on this 350 MHz processor.
</para>

<para>
&MCOP; protocol overheadL &MCOP; does, as a rule of thumb, 9000
invocations per second. Much of this is not &MCOP;s fault, but relates
to the two kernel causes named below.  However, this gives a base to do
calculations what the cost of streaming is.
</para>

<para>
Each data packet transferred through streaming can be considered one
&MCOP; invocation. Of course large packets are slower than 9000
packets/s, but its about the idea.
</para>

<para>
Suppose you use packet sizes of 1024 bytes. Thus, to transfer a stream
with 44kHz stereo, you need to transfer 44100*4/1024 = 172 packets per
second. Suppose you could with 100% cpu usage transfer 9000 packets,
then you get (172*100)/9000 = 2% <acronym>CPU</acronym> usage due to
streaming with 1024 byte packets.
</para>

<para>
That are approximations. However, they show, that you would be much
better off (if you can afford it for the latency), to use for instance
packets of 4096 bytes. We can make a compact formula here, by
calculating the packet size which causes 100% <acronym>CPU</acronym> usage as
44100*4/9000 = 19.6 samples, and thus getting the quick formula:
</para>

<para>
streaming <acronym>CPU</acronym> usage in percent = 1960/(your packet size)
</para>

<para>
which gives us 0.5% <acronym>CPU</acronym> usage when streaming with 4096 byte packets.
</para>

<para>
Kernel process/context switching: this is part of the &MCOP; protocol
overhead.  Switching between two processes takes time. There is new
memory mapping, the caches are invalid, whatever else (if there is a
kernel expert reading this - let me know what exactly are the causes).
This means: it takes time.
</para>

<para>
I am not sure how many context switches &Linux; can do per second, but
that number isn't infinite. Thus, of the &MCOP; protocol overhead I
suppose quite a bit is due to context switching. In the beginning of
&MCOP;, I did tests to use the same communication inside one process,
and it was much faster (four times as fast or so).
</para>

<para>
Kernel: communication overhead: This is part of the &MCOP; protocol
overhead.  Transferring data between processes is currently done via
sockets. This is convenient, as the usual select() methods can be used
to determine when a message has arrived. It can also be combined with
other I/O sources as audio I/O, X11 server or whatever else easily.
</para>

<para>
However, those read and write calls cost certainly processor cycles. For
small invocations (such as transferring one midi event) this is probably
not so bad, for large invocations (such as transferring one video frame
with several megabytes) this is clearly a problem.
</para>

<para>
Adding the usage of shared memory to &MCOP; where appropriate is
probably the best solution. However it should be done transparent to the
application programmer.
</para>

<para>
Take a profiler or do other tests to find out how much exactly
current audio streaming is impacted by the not using sharedmem. However,
its not bad, as audio streaming (replaying mp3) can be done with 6%
total <acronym>CPU</acronym> usage for &artsd; and
<application>artscat</application> (and 5% for the mp3
decoder). However, this includes all things from the necessary
calculations up do the socket overhead, thus I'd say in this setup you
could perhaps save 1% by using sharedmem.
</para>

</sect2>

<sect2 id="hard-numbers">
<title>Some Hard Numbers</title>

<para>
These are done with the current development snapshot. I also wanted to
try out the real hard cases, so this is not what everyday applications
should use.
</para>

<para>
I wrote an application called streamsound which sends streaming data to
&arts;.  Here it is running with realtime priority (without problems),
and one small serverside (volume-scaling and clipping) plugin:
</para>

<programlisting>
 4974 stefan    20   0  2360 2360  1784 S       0 17.7  1.8   0:21 artsd
 5016 stefan    20   0  2208 2208  1684 S       0  7.2  1.7   0:02 streamsound
 5002 stefan    20   0  2208 2208  1684 S       0  6.8  1.7   0:07 streamsound
 4997 stefan    20   0  2208 2208  1684 S       0  6.6  1.7   0:07 streamsound
</programlisting>

<para>
Each of them is streaming with 3 fragments a 1024 bytes (18 ms). There
are three such clients running simultaneously. I know that that does
look a bit too much, but as I said: take a profiler and find out what
costs time, and if you like, improve it.
</para>

<para>
However, I don't think using streaming like that is realistic or makes
sense.  To take it even more to the extreme, I tried what would be the
lowest latency possible. Result: you can do streaming without
interruptions with one client application, if you take 2 fragments of
128 bytes between aRts and the soundcard, and between the client
application and aRts. This means that you have a total maximum latency
of 128*4/44100*4 = 3 ms, where 1.5 ms is generated due to soundcard I/O
and 1.5 ms is generated through communication with &arts;.  Both
applications need to run realtimed.
</para>

<para> 
But: this costs an enormous amount of
<acronym>CPU</acronym>. This example cost you about 45% of my
P-II/350. I also starts to click if you start top, move windows on your
X11 display or do disk I/O. All these are kernel issues. The problem is
that scheduling two or more applications with realtime priority cost you
an enormous amount of effort, too, even more if the communicate, notify
each other &etc;.
</para>

<para>
Finally, a more real life example. This is &arts; with artsd and one
artscat (one streaming client) running 16 fragments a 4096 bytes:
</para>

<programlisting>
 5548 stefan    12   0  2364 2364  1752 R       0  4.9  1.8   0:03 artsd
 5554 stefan     3   0   752  752   572 R       0  0.7  0.5   0:00 top
 5550 stefan     2   0  2280 2280  1696 S       0  0.5  1.7   0:00 artscat
</programlisting>

</sect2>
</sect1>

<!-- TODO

<sect1 id="dynamic-instantiation">
<title>Dynamic Instantiation</title>
<para>
</para>
</sect1>

-->

<sect1 id="busses">
<title>Busses</title>

<para>
Busses are dynamically built connections that transfer audio. Basically,
there are some uplinks and some downlinks. All signals from the uplinks
are added and send to the downlinks.
</para>

<para>
Busses as currently implemented operate in stereo, so you can only
transfer stereo data over busses. If you want mono data, well, transfer
it only over one channel and set the other to zero or whatever.  What
you need to to, is to create one or more Synth&lowbar;BUS&lowbar;UPLINK
objects and tell them a bus name, to which they should talk (&eg;
<quote>audio</quote> or <quote>drums</quote>). Simply throw the data in
there.
</para>

<para>
Then, you'll need to create one or more Synth&lowbar;BUS&lowbar;DOWNLINK
objects, and tell them the bus name (<quote>audio</quote> or
<quote>drums</quote> ... if it matches, the data will get through), and
the mixed data will come out again.
</para>

<para>
The uplinks and downlinks can reside in different structures, you can
even have different &arts-builder;s running and start an uplink in one
and receive the data from the other with a downlink.
</para>

<para>
What is nice about busses is, that they are fully dynamic. Clients can
plug in and out on the fly. There should be no clicking or noise as this
happens.
</para>

<para>
Of course, you should not plug out a client playing a signal, since it
will probably not be a zero level when plugged out the bus, and then it
will click.
</para>
</sect1>

<!-- TODO
<sect1 id="network-ransparency">
<title>Network Transparency</title>
<para>
</para>
</sect1>

<sect1 id="security">
<title>Security</title>
<para>
</para>
</sect1>


<sect1 id="effects">
<title>Effects and Effect Stacks</title>
<para>
</para>
</sect1>

-->
<sect1 id="trader">
<title>Trader</title>

<para>
&arts;/&MCOP; heavily relies on splitting up things into small
components. This makes things very flexible, as you can extend the
system easily by adding new components, which implement new effects,
fileformats, oscillators, gui elements, ... As almost everything is a
component, almost everything can be extended easily, without changing
existing sources. New components can be simply loaded dynamically to
enhance already existing applications.
</para>

<para>
However, to make this work, two things are required:
</para>

<itemizedlist>

<listitem>
<para>
Components must advertise themselves - they must describe what great
things they offer, so that applications will be able to use them.
</para>
</listitem>

<listitem>
<para>
Applications must actively look for components that they could use,
instead of using always the same thing for some task.
</para>
</listitem>

</itemizedlist>

<para>
The combination of this: components which say <quote>here I am, I am
cool, use me</quote>, and applications (or if you like, other
components) which go out and look which component they could use to get
a thing done, is called trading.
</para>

<para>
In &arts;, components describe themselves by specifying values that they
<quote>support</quote> for properties. A typical property for a
file-loading component could be the extension of the files that it can
process. Typical values could be <literal
role="extension">wav</literal>, <literal role="extension">aiff</literal>
or <literal role="extension">mp3</literal>.
</para>

<para>
In fact, every component may choose to offer many different values for
one property. So one single component could offer reading both, <literal
role="extension">wav</literal> and <literal
role="extension">aiff</literal> files, by specifying that it supports
these values for the property <quote>Extension</quote>.
</para>

<para>
To do so, a component has to place a <literal
role="extension">.mcopclass</literal> file at an appropriate place,
containing the properties it supports, for our example, this could look
like this (and would be installed in
<filename><replaceable>componentdir</replaceable>/Arts/WavPlayObject.mcopclass</filename>):
</para>

<programlisting>
Interface=Arts::WavPlayObject,Arts::PlayObject,Arts::SynthModule,Arts::Object
Author="Stefan Westerfeld &lt;stefan@space.twc.de&gt;"
URL="http://www.arts-project.org"
Extension=wav,aiff
MimeType=audio/x-wav,audio/x-aiff
</programlisting>

<para>
It is important that the filename of the <literal
role="extension">.mcopclass</literal>-file also says what the interface
of the component is called like. The trader doesn't look at the contents
at all, if the file (like here) is called
<filename>Arts/WavPlayObject.mcopclass</filename>, the component
interface is called <interfacename>Arts::WavPlayObject</interfacename>
(modules map to folders).
</para>

<para>
To look for components, there are two interfaces (which are defined in
<filename>core.idl</filename>, so you have them in every application),
called <interfacename>Arts::TraderQuery</interfacename> and
<interfacename>Arts::TraderOffer</interfacename>. You to go on a
<quote>shopping tour</quote> for components like this:
</para>

<orderedlist>
<listitem>
<para>
Create a query object:
</para>
<programlisting>
    Arts::TraderQuery query;
</programlisting>
</listitem>

<listitem>
<para>
Specify what you want. As you saw above, components describe themselves
using properties, for which they offer certain values. So specifying
what you want is done by selecting components that support a certain
value for a property. This is done using the supports method of a
TraderQuery:
</para>

<programlisting>
    query.supports("Interface","Arts::PlayObject");
    query.supports("Extension","wav");
</programlisting>
</listitem>

<listitem>
<para>
Finally, perform the query using the query method. Then, you'll
(hopefully) get some offers:
</para>

<programlisting>
    vector&lt;Arts::TraderOffer&gt; *offers = query.query();
</programlisting>
</listitem>

<listitem>
<para>
Now you can examine what you found. Important is the interfaceName
method of TraderOffer, which will tell you the name of the component,
that matched the query. You can also find out further properties by
getProperty. The following code will simply iterate through all
components, print their interface names (which could be used for
creation), and delete the results of the query again:
</para>
<programlisting>
    vector&lt;Arts::TraderOffer&gt;::iterator i;
    for(i = offers-&gt;begin(); i != offers-&gt;end(); i++)
        cout &lt;&lt; i-&gt;interfaceName() &lt;&lt; endl;
    delete offers;
</programlisting>
</listitem>
</orderedlist>

<para>
For this kind of trading service to be useful, it is important to
somehow agree on what kinds of properties components should usually
define. It is essential that more or less all components in a certain
area use the same set of properties to describe themselves (and the same
set of values where applicable), so that applications (or other
components) will be able to find them.
</para>

<para>
Author (type string, optional): This can be used to ultimately let the
world know that you wrote something.  You can write anything you like in
here, e-mail address is of course helpful.
</para>

<para>
Buildable (type boolean, recommended): This indicates whether the
component is usable with <acronym>RAD</acronym> tools (such as
&arts-builder;) which use components by assigning properties and
connecting ports.  It is recommended to set this value to true for
almost any signal processing component (such as filters, effects,
oscillators, ...), and for all other things which can be used in
<acronym>RAD</acronym> like fashion, but not for internal stuff like for
instance <interfacename>Arts::InterfaceRepo</interfacename>.
</para>

<para>
Extension (type string, used where relevant): Everything dealing with
files should consider using this. You should put the lowercase version
of the file extension without the <quote>.</quote> here, so something
like <userinput>wav</userinput> should be fine.
</para>

<para>
Interface (type string, required): This should include the full list of
(useful) interfaces your components supports, probably including
<interfacename>Arts::Object</interfacename> and if applicable
<interfacename>Arts::SynthModule</interfacename>.
</para>

<para>
Language (type string, recommended): If you want your component to be
dynamically loaded, you need to specify the language here. Currently,
the only allowed value is <userinput>C++</userinput>, which means the
component was written using the normal C++ <acronym>API</acronym>. If
you do so, you'll also need to set the <quote>Library</quote> property
below.
</para>

<para>
Library (type string, used where relevant): Components written in C++
can be dynamically loaded. To do so, you have to compile them into a
dynamically loadable libtool (<literal role="extension">.la</literal>)
module. Here, you can specify the name of the <literal
role="extension">.la</literal>-File that contains your component.
Remember to use REGISTER_IMPLEMENTATION (as always).
</para>

<para>
MimeType (type string, used where relevant): Everything dealing with
files should consider using this. You should put the lowercase version
of the standard mimetype here, for instance
<userinput>audio/x-wav</userinput>.
</para>

<para>
&URL; (type string, optional): If you like to let people know where they
can find a new version of the component (or a homepage or anything), you
can do it here. This should be standard &HTTP; or &FTP; &URL;.
</para>

</sect1>

<!-- TODO
<sect1 id="midi-synthesis">
<title><acronym>MIDI</acronym> Synthesis</title>
<para>
</para>
</sect1>

<sect1 id="instruments">
<title>Instruments</title>
<para>
</para>
</sect1>

<sect1 id="session-management">
<title>Session Management</title>
<para>
</para>
</sect1>

<sect1 id="full-duplex">
<title>Full duplex Audio</title>
<para>
</para>
</sect1>
-->

<sect1 id="namespaces">
<title>Namespaces in &arts;</title>

<sect2 id="namespaces-intro">
<title>Introduction</title>

<para>
Each namespace declaration corresponds to a <quote>module</quote>
declaration in the &MCOP; &IDL;.
</para>

<programlisting>
// mcop idl

module M {
    interface A
    {
    }
};

interface B;
</programlisting>

<para>
In this case, the generated C++ code for the &IDL; snippet would look
like this:
</para>

<programlisting>
// C++ header

namespace M {
    /* declaration of A_base/A_skel/A_stub and similar */
    class A {        // Smartwrapped reference class
        /* [...] */
    };
}

/* declaration of B_base/B_skel/B_stub and similar */
class B {
    /* [...] */
};
</programlisting>

<para>
So when referring the classes from the above example in your C++ code,
you would have to write <classname>M::A</classname>, but only
B. However, you can of course use <quote>using M</quote> somewhere -
like with any namespace in C++.
</para>

</sect2>

<sect2 id="namespaces-how">
<title>How &arts; uses namespaces</title>

<para>
There is one global namespace called <quote>Arts</quote>, which all
programs and libraries that belong to &arts; itself use to put their
declarations in. This means, that when writing C++ code that depends on
&arts;, you normally have to prefix every class you use with
<classname>Arts::</classname>, like this:
</para>

<programlisting>
int main(int argc, char **argv)
{
    Arts::Dispatcher dispatcher;
    Arts::SimpleSoundServer server(Arts::Reference("global:Arts_SimpleSoundServer"));

    server.play("/var/foo/somefile.wav");
</programlisting>

<para>
The other alternative is to write a using once, like this:
</para>

<programlisting>
using namespace Arts;

int main(int argc, char **argv)
{
    Dispatcher dispatcher;
    SimpleSoundServer server(Reference("global:Arts_SimpleSoundServer"));

    server.play("/var/foo/somefile.wav");
    [...]
</programlisting>

<para>
In &IDL; files, you don't exactly have a choice. If you are writing code
that belongs to &arts; itself, you'll have to put it into module &arts;.
</para>

<programlisting>
// IDL File for aRts code:
#include &lt;artsflow.idl&gt;
module Arts {        // put it into the Arts namespace
    interface Synth_TWEAK : SynthModule
    {
        in audio stream invalue;
        out audio stream outvalue;
        attribute float tweakFactor;
    };
};
</programlisting>

<para>
If you write code that doesn't belong to &arts; itself, you should not
put it into the <quote>Arts</quote> namespace. However, you can make an
own namespace if you like.  In any case, you'll have to prefix classes
you use from &arts;.
</para>

<programlisting>
// IDL File for code which doesn't belong to aRts:
#include &lt;artsflow.idl&gt;

// either write without module declaration, then the generated classes will
// not use a namespace:
interface Synth_TWEAK2 : Arts::SynthModule
{
    in audio stream invalue;
    out audio stream outvalue;
    attribute float tweakFactor;
};

// however, you can also choose your own namespace, if you like, so if you
// write an application "PowerRadio", you could for instance do it like this:
module PowerRadio {
    struct Station {
        string name;
        float frequency;
    };

    interface Tuner : Arts::SynthModule {
        attribute Station station;     // no need to prefix Station, same module
        out audio stream left, right;
    };
};
</programlisting>

</sect2>

<sect2 id="namespaces-implementation">
<title>Internals: How the Implementation Works</title>

<para>
Often, in interfaces, casts, method signatures and similar, &MCOP; needs
to refer to names of types or interfaces. These are represented as
string in the common &MCOP; datastructures, while the namespace is
always fully represented in the C++ style. This means the strings would
contain <quote>M::A</quote> and <quote>B</quote>, following the example
above.
</para>

<para>
Note this even applies if inside the &IDL; text the namespace qualifiers
were not given, since the context made clear which namespace the
interface <interfacename>A</interfacename> was meant to be used in.
</para>

</sect2>
</sect1>

<sect1 id="threads">
<title>Threads in &arts;</title>

<sect2 id="threads-basics">
<title>Basics</title>

<para>
Using threads isn't possible on all platforms. This is why &arts; was
originally written without using threading at all. For almost all
problems, for each threaded solution to the problem, there is a
non-threaded solution that does the same.
</para>

<para>
For instance, instead of putting audio output in a separate thread, and
make it blocking, &arts; uses non-blocking audio output, and figures out
when to write the next chunk of data using
<function>select()</function>.
</para>

<para>
However, &arts; (in very recent versions) at least provides support for
people who do want to implement their objects using threads. For
instance, if you already have code for an <literal
role="extension">mp3</literal> player, and the code expects the <literal
role="extension">mp3</literal> decoder to run in a separate thread, it's
usually the easiest thing to do to keep this design.
</para>

<para>
The &arts;/&MCOP; implementation is built along sharing state between
separate objects in obvious and non-obvious ways. A small list of shared
state includes:
</para>

<itemizedlist>
<listitem><para>
The Dispatcher object which does &MCOP; communication.
</para>
</listitem>

<listitem>
<para>
The Reference counting (Smartwrappers).
</para>
</listitem>

<listitem>
<para>
The IOManager which does timer and fd watches.
</para>
</listitem>

<listitem>
<para>
The ObjectManager which creates objects and dynamically loads plugins.
</para>
</listitem>

<listitem>
<para>
The FlowSystem which calls calculateBlock in the appropriate situations.
</para>
</listitem>
</itemizedlist>

<para>
All of the above objects don't expect to be used concurrently (&ie;
called from separate threads at the same time). Generally there are two
ways of solving this:
</para>

<itemizedlist>
<listitem>
<para>
Require the caller of any functions on this objects to
acquire a lock before using them.
</para>
</listitem>

<listitem>
<para>
Making these objects really threadsafe and/or create
per-thread instances of them.
</para>
</listitem>
</itemizedlist>

<para>
&arts; follows the first approach: you will need a lock whenever you talk to
any of these objects. The second approach is harder to do. A hack which
tries to achieve this is available at
<ulink url="http://space.twc.de/~stefan/kde/download/arts-mt.tar.gz">
http://space.twc.de/~stefan/kde/download/arts-mt.tar.gz</ulink>, but for
the current point in time, a minimalistic approach will probably work
better, and cause less problems with existing applications.
</para>

</sect2>
<sect2 id="threads-locking">
<title>When/how to acquire the lock?</title>

<para>
You can get/release the lock with the two functions:
</para>

<itemizedlist>
<listitem>
<para>
<ulink
url="http://space.twc.de/~stefan/kde/arts-mcop-doc/arts-reference/headers/Arts__Dispatcher.html#lock"><function>Arts::Dispatcher::lock()</function></ulink>
</para>
</listitem>
<listitem>
<para>
<ulink
url="http://space.twc.de/~stefan/kde/arts-mcop-doc/arts-reference/headers/Arts__Dispatcher.html#unlock"><function>Arts::Dispatcher::unlock()</function></ulink>
</para>
</listitem>
</itemizedlist>

<para>
Generally, you don't need to acquire the lock (and you shouldn't try to
do so), if it is already held. A list of conditions when this is the
case is:
</para>

<itemizedlist>
<listitem>
<para>
You receive a callback from the IOManager (timer or fd).
</para>
</listitem>

<listitem>
<para>
You get call due to some &MCOP; request.
</para>
</listitem>

<listitem>
<para>
You are called from the NotificationManager.
</para>
</listitem>

<listitem>
<para>
You are called from the FlowSystem (calculateBlock)
</para>
</listitem>
</itemizedlist>

<para>
There are also some exceptions of functions. which you can only call in
the main thread, and for that reason you will never need a lock to call
them:
</para>

<itemizedlist>
<listitem>
<para>
Constructor/destructor of Dispatcher/IOManager.
</para>
</listitem>

<listitem>
<para>
<methodname>Dispatcher::run()</methodname> /
<methodname>IOManager::run()</methodname>
</para>
</listitem>

<listitem>
<para><methodname>IOManager::processOneEvent()</methodname></para>
</listitem>
</itemizedlist>

<para>
But that is it. For everything else that is somehow related to &arts;,
you will need to get the lock, and release it again when
done. Always. Here is a simple example:
</para>

<programlisting>
class SuspendTimeThread : Arts::Thread {
public:
    void run() {
        /*
         * you need this lock because:
         *  - constructing a reference needs a lock (as global: will go to
         *    the object manager, which might in turn need the GlobalComm
         *    object to look up where to connect to)
         *  - assigning a smartwrapper needs a lock
         *  - constructing an object from reference needs a lock (because it
         *    might need to connect a server)
         */
        Arts::Dispatcher::lock();
        Arts::SoundServer server = Arts::Reference("global:Arts_SoundServer");
        Arts::Dispatcher::unlock();

        for(;;) {            /*
             * you need a lock here, because
             *  - dereferencing a smartwrapper needs a lock (because it might
             *    do lazy creation)
             *  - doing an MCOP invocation needs a lock
             */
            Arts::Dispatcher::lock();
            long seconds = server.secondsUntilSuspend();
            Arts::Dispatcher::unlock();

            printf("seconds until suspend = %d",seconds);
            sleep(1);
        }
    }
}
</programlisting>


</sect2>

<sect2 id="threads-classes">
<title>Threading related classes</title>

<para>
The following threading related classes are currently available:
</para>

<itemizedlist>
<listitem>
<para>
<ulink
url="http://www.arts-project.org/doc/headers/Arts__Thread.html"><classname>
Arts::Thread</classname></ulink> - which encapsulates a thread.
</para>
</listitem>

<listitem>
<para>
<ulink url="http://www.arts-project.org/doc/headers/Arts__Mutex.html">
<classname>Arts::Mutex</classname></ulink> - which encapsulates a mutex.
</para>
</listitem>

<listitem>
<para>
<ulink
url="http://www.arts-project.org/doc/headers/Arts__ThreadCondition.html">
<classname>Arts::ThreadCondition</classname></ulink> - which provides
support to wake up threads which are waiting for a certain condition to
become true.
</para>
</listitem>

<listitem>
<para>
<ulink
url="http://www.arts-project.org/doc/headers/Arts__SystemThreads.html"><classname>Arts::SystemThreads</classname></ulink>
- which encapsulates the operating system threading layer (which offers
a few helpful functions to application programmers).
</para>
</listitem>
</itemizedlist>

<para>
See the links for documentation.
</para>

</sect2>
</sect1>

<sect1 id="references-errors">
<title>References and Error Handling</title>

<para>
&MCOP; references are one of the most central concepts in &MCOP;
programming.  This section will try to describe how exactly references
are used, and will especially also try to cover cases of failure (server
crashes).
</para>

<sect2 id="references-properties">
<title>Basic properties of references</title>

<itemizedlist>
<listitem>
<para>
An &MCOP; reference is not an object, but a reference to an object: Even
though the following declaration

<programlisting>
   Arts::Synth_PLAY p;
</programlisting>

looks like a definition of an object, it only declares a reference to an
object. As C++ programmer, you might also think of it as Synth_PLAY *, a
kind of pointer to a Synth_PLAY object. This especially means, that p
can be the same thing as a NULL pointer.
</para>
</listitem>

<listitem>
<para>
You can create a NULL reference by assigning it explicitly
</para>
<programlisting>
   Arts::Synth_PLAY p = Arts::Synth_PLAY::null();
</programlisting>
</listitem>

<listitem>
<para>
Invoking things on a NULL reference leads to a core dump
</para>
<programlisting>
   Arts::Synth_PLAY p = Arts::Synth_PLAY::null();
   string s = p.toString();
</programlisting>
<para>
will lead to a core dump. Comparing this to a pointer, it is essentially
the same as 
<programlisting>
   QWindow* w = 0;
   w-&gt;show();
</programlisting>
which every C++ programmer would know to avoid.
</para>
</listitem>

<listitem>
<para>
Uninitialized objects try to lazy-create themselves upon first use
</para>

<programlisting>
   Arts::Synth_PLAY p;
   string s = p.toString();
</programlisting>
<para>
is something different than dereferencing a NULL pointer. You didn't tell
the object at all what it is, and now you try to use it. The guess here
is that you want to have a new local instance of a Arts::Synth_PLAY
object. Of course you might have wanted something else (like creating the
object somewhere else, or using an existing remote object). However, it
is a convenient short cut to creating objects. Lazy creation will not work
once you assigned something else (like a null reference).
</para>

<para>
The equivalent C++ terms would be
<programlisting>
   TQWidget* w;
   w-&gt;show();
</programlisting>

which obviously in C++ just plain segfaults. So this is different here.
This lazy creation is tricky especially as not necessarily an implementation
exists for your interface.
</para>

<para>
For instance, consider an abstract thing like a
Arts::PlayObject. There are certainly concrete PlayObjects like those for
playing mp3s or wavs, but

<programlisting>
   Arts::PlayObject po;
   po.play();
</programlisting>

will certainly fail. The problem is that although lazy creation kicks
in, and tries to create a PlayObject, it fails, because there are only
things like Arts::WavPlayObject and similar. Thus, use lazy creation
only when you are sure that an implementation exists.
</para>
</listitem>

<listitem>
<para>
References may point to the same object
</para>

<programlisting>
   Arts::SimpleSoundServer s = Arts::Reference("global:Arts_SimpleSoundServer");
   Arts::SimpleSoundServer s2 = s;
</programlisting>

<para>
creates two references referring to the same object. It doesn't copy any
value, and doesn't create two objects.  
</para>
</listitem>

<listitem>
<para>
All objects are reference counted So once an object isn't referred any
longer by any references, it gets deleted.  There is no way to
explicitly delete an object, however, you can use something like this
<programlisting>
   Arts::Synth_PLAY p;
   p.start();
   [...]
   p = Arts::Synth_PLAY::null();
</programlisting>
to make the Synth_PLAY object go away in the end. Especially, it should never
be necessary to use new and delete in conjunction with references.
</para>
</listitem>
</itemizedlist>

</sect2>

<sect2 id="references-failure">
<title>The case of failure</title>

<para>
As references can point to remote objects, the servers containing these
objects can crash. What happens then?
</para>

<itemizedlist>

<listitem>
<para>
A crash doesn't change whether a reference is a null reference.  This
means that if <function>foo.isNull()</function> was
<returnvalue>true</returnvalue> before a server crash then it is also
<returnvalue>true</returnvalue> after a server crash (which is
clear). It also means that if <function>foo.isNull()</function> was
<returnvalue>false</returnvalue> before a server crash (foo referred to
an object) then it is also <returnvalue>false</returnvalue> after the
server crash.
</para>
</listitem>

<listitem>
<para>
Invoking methods on a valid reference stays safe
Suppose the server containing the object calc crashed. Still calling things
like
<programlisting>
   int k = calc.subtract(i,j)
</programlisting>
are safe. Obviously subtract has to return something here, which it
can't because the remote object no longer exists. In this case (k == 0)
would be true. Generally, operations try to return something
<quote>neutral</quote> as result, such as 0.0, a null reference for
objects or empty strings, when the object no longer exists.
</para>
</listitem>

<listitem>
<para>
Checking <function>error()</function> reveals whether something worked.
</para>

<para>
In the above case,
<programlisting>
   int k = calc.subtract(i,j)
   if(k.error()) {
      printf("k is not i-j!\n");
   }
</programlisting>
would print out <computeroutput>k is not i-j</computeroutput> whenever
the remote invocation didn't work. Otherwise <varname>k</varname> is
really the result of the subtract operation as performed by the remote
object (no server crash). However, for methods doing things like
deleting a file, you can't know for sure whether it really happened. Of
course it happened if <function>.error()</function> is
<returnvalue>false</returnvalue>.  However, if
<function>.error()</function> is <returnvalue>true</returnvalue>, there
are two possibilities:
</para>

<itemizedlist>
<listitem>
<para>
The file got deleted, and the server crashed just after deleting it, but
before transferring the result.
</para>
</listitem>

<listitem>
<para>
The server crashed before being able to delete the file.
</para>
</listitem>
</itemizedlist>
</listitem>

<listitem>
<para>
Using nested invocations is dangerous in crash resistant programs
</para>

<para>
Using something like
<programlisting>
   window.titlebar().setTitle("foo");
</programlisting>
is not a good idea. Suppose you know that window contains a valid Window
reference. Suppose you know that <function>window.titlebar()</function>
will return a Titlebar reference because the Window object is
implemented properly. However, still the above statement isn't safe. 
</para>

<para>
What could happen is that the server containing the Window object has
crashed.  Then, regardless of how good the Window implementation is, you
will get a null reference as result of the window.titlebar()
operation. And then of course invoking setTitle on that null reference
will lead to a crash as well.
</para>

<para>
So a safe variant of this would be
<programlisting>
   Titlebar titlebar = window.titlebar();
   if(!window.error())
      titlebar.setTitle("foo");
</programlisting>
add the appropriate error handling if you like. If you don't trust the
Window implementation, you might as well use
<programlisting>
   Titlebar titlebar = window.titlebar();
   if(!titlebar.isNull())
      titlebar.setTitle("foo");
</programlisting>
which are both safe.
</para>
</listitem>
</itemizedlist>

<para>
There are other conditions of failure, such as network disconnection
(suppose you remove the cable between your server and client while your
application runs). However their effect is the same like a server crash.
</para>

<para>
Overall, it is of course a consideration of policy how strictly you try
to trap communication errors throughout your application. You might
follow the <quote>if the server crashes, we need to debug the server
until it never crashes again</quote> method, which would mean you need
not bother about all these problems.
</para>

</sect2>

<sect2 id="references-internals">
<title>Internals: Distributed Reference Counting</title>

<para>
An object, to exist, must be owned by someone. If it isn't, it will
cease to exist (more or less) immediately. Internally, ownership is
indicated by calling <function>_copy()</function>, which increments an
reference count, and given back by calling
<function>_release()</function>. As soon as the reference count drops to
zero, a delete will be done.
</para>

<para>
As a variation of the theme, remote usage is indicated by
<function>_useRemote()</function>, and dissolved by
<function>_releaseRemote()</function>. These functions lead a list which
server has invoked them (and thus owns the object). This is used in case
this server disconnects (&ie; crash, network failure), to remove the
references that are still on the objects. This is done in
<function>_disconnectRemote()</function>.
</para>

<para>
Now there is one problem. Consider a return value. Usually, the return
value object will not be owned by the calling function any longer. It
will however also not be owned by the caller, until the message holding
the object is received. So there is a time of
<quote>ownershipless</quote> objects.
</para>

<para>
Now, when sending an object, one can be reasonable sure that as soon as
it is received, it will be owned by somebody again, unless, again, the
receiver dies.  However this means that special care needs to be taken
about object at least while sending, probably also while receiving, so
that it doesn't die at once.
</para>

<para>
The way &MCOP; does this is by <quote>tagging</quote> objects that are
in process of being copied across the wire. Before such a copy is
started, <function>_copyRemote</function> is called. This prevents the
object from being freed for a while (5 seconds). Once the receiver calls
<function>_useRemote()</function>, the tag is removed again. So all
objects that are send over wire are tagged before transfer.
</para>

<para>
If the receiver receives an object which is on their server, of course
they will not <function>_useRemote()</function> it. For this special case,
<function>_cancelCopyRemote()</function> exists to remove the tag
manually. Other than that, there is also timer based tag removal, if
tagging was done, but the receiver didn't really get the object (due to
crash, network failure). This is done by the
<classname>ReferenceClean</classname> class.
</para>

</sect2>

</sect1>

<sect1 id="detail-gui-elements">
<title>&GUI; Elements</title>

<para>
&GUI; elements are currently in the experimental state. However, this
section will describe what is supposed to happen here, so if you are a
developer, you will be able to understand how &arts; will deal with
&GUI;s in the future.  There is some code there already, too.
</para>

<para>
&GUI; elements should be used to allow synthesis structures to interact
with the user. In the simplest case, the user should be able to modify
some parameters of a structure directly (such as a gain factor which is
used before the final play module).
</para>

<para>
In more complex settings, one could imagine the user modifying
parameters of groups of structures and/or not yet running structures,
such as modifying the <acronym>ADSR</acronym> envelope of the currently
active &MIDI; instrument. Another thing would be setting the filename of
some sample based instrument.
</para> 

<para>
On the other hand, the user could like to monitor what the synthesizer
is doing. There could be oscilloscopes, spectrum analyzers, volume
meters and <quote>experiments</quote> that figure out the frequency
transfer curve of some given filter module.
</para>

<para>
Finally, the &GUI; elements should be able to control the whole
structure of what is running inside &arts; and how. The user should be
able to assign instruments to midi channels, start new effect
processors, configure his main mixer desk (which is built of &arts;
structures itself) to have one channel more and use another strategy for
its equalizers.
</para>

<para>
You see - the <acronym>GUI</acronym> elements should bring all
possibilities of the virtual studio &arts; should simulate to the
user. Of course, they should also gracefully interact with midi inputs
(such as sliders should move if they get &MIDI; inputs which also change
just that parameter), and probably even generate events themselves, to
allow the user interaction to be recorded via sequencer.
</para>

<para>
Technically, the idea is to have an &IDL; base class for all widgets
(<classname>Arts::Widget</classname>), and derive a number of commonly
used widgets from there (like <classname>Arts::Poti</classname>,
<classname>Arts::Panel</classname>, <classname>Arts::Window</classname>,
...).
</para>

<para>
Then, one can implement these widgets using a toolkit, for instance &Qt;
or Gtk. Finally, effects should build their &GUI;s out of existing
widgets. For instance, a freeverb effect could build it's &GUI; out of
five <classname>Arts::Poti</classname> thingies and an
<classname>Arts::Window</classname>.  So IF there is a &Qt;
implementation for these base widgets, the effect will be able to
display itself using &Qt;. If there is Gtk implementation, it will also
work for Gtk (and more or less look/work the same).
</para>

<para>
Finally, as we're using &IDL; here, &arts-builder; (or other tools) will
be able to plug &GUI;s together visually, or autogenerate &GUI;s given
hints for parameters, only based on the interfaces. It should be
relatively straight forward to write a <quote>create &GUI; from
description</quote> class, which takes a &GUI; description (containing
the various parameters and widgets), and creates a living &GUI; object
out of it.
</para>

<para>
Based on &IDL; and the &arts;/&MCOP; component model, it should be easy
to extend the possible objects which can be used for the &GUI; just as
easy as it is to add a plugin implementing a new filter to &arts;.
</para>

</sect1>

</chapter>